trixbox CE - The Open Platform For Business Telephony

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01/07/12--01:25: Unable to Trunk two trixbox (chan 2127936)

I have created two trixbox server on two different locations which can ping to each other. Giving you the details of them.

hotrixbox - 192.168.2.36
trixbox1 - 192.168.4.20

[hotrixbox ~]# ping trixbox1
PING trixbox1 (192.168.4.20) 56(84) bytes of data.
64 bytes from trixbox1 (192.168.4.20): icmp_seq=1 ttl=62 time=90.5 ms
64 bytes from trixbox1 (192.168.4.20): icmp_seq=2 ttl=62 time=88.1 ms
64 bytes from trixbox1 (192.168.4.20): icmp_seq=3 ttl=62 time=89.5 ms

[trixbox1.localdomain etc]# ping hotrixbox
PING hotrixbox (192.168.2.36) 56(84) bytes of data.
64 bytes from hotrixbox (192.168.2.36): icmp_seq=1 ttl=62 time=92.2 ms
64 bytes from hotrixbox (192.168.2.36): icmp_seq=2 ttl=62 time=261 ms
64 bytes from hotrixbox (192.168.2.36): icmp_seq=3 ttl=62 time=88.3 ms

Now I want to trunk this two servers (Hotrixbox extensions can call trixbox1 extensions via LAN ONLY). I am totally newbie for this. All the extensions on both servers are working very fine (Have tested with softphone as well as hardphones).
After several research over google, I have tried to trunk them. For that I have created a extension on both the servers. 1017 for trixbox1 & 2001 on hotrixbox. Here I am also giving you the settings I have made on both servers -

HOTRIXBOX -

Added SIP trunk with following settings -

Outbound Caller ID: 2001

Trunk Name - RGH
PEER DETAILS -
context=from-pstn
fromuser=2001
host=192.168.2.36
insecure=port,invite
secret=2001
type=peer
defaultuser=2001
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw

Register String: 1017:1017@192.168.4.20/1017

Incoming Settings - LEFT BLANK.

OUTBOUND ROUTE -

Route name - RGH

Dial Pattern - 7|.

Trunk sequence - SIP/RGH

INBOUND ROUTE -

Description - RGH
DID Number - 2001

Set Destination -

Extensions: 2001

Then added following on sip_general_custom.conf file -

context=from-pstn
srvlookup=yes
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas

Then on other side (TRIXBOX1), I have configured the following -

Added SIP trunk with following settings -

Outbound Caller ID: 1017

Trunk Name - HO
PEER DETAILS -
context=from-pstn
fromuser=1017
host=192.168.4.20
insecure=port,invite
secret=1017
type=peer
defaultuser=1017
disallowed_methods=UPDATE
directmedia=no
videosupport=no
disallow=all
allow=ulaw

Register String: 2001:2001@192.168.2.36/2001

Incoming Settings - LEFT BLANK.

OUTBOUND ROUTE -

Route name - HO

Dial Pattern - 7|.

Trunk sequence - SIP/HO

INBOUND ROUTE -

Description - HO
DID Number - 1017

Set Destination -

Extensions: 1017

Then added following on sip_general_custom.conf file -

context=from-pstn
srvlookup=yes
session-timers=refuse
session-expires=180
session-minse=90
session-refresher=uas

This are the outputs of SIP registration from both server -

hotrixbox*CLI> sip show registry
Host Username Refresh State Reg.Time
192.168.4.20:5060 1017 105 Registered Sat, 07 Jan 2012 14:51:10
1 SIP registrations.

trixbox1*CLI> sip show registry
Host Username Refresh State Reg.Time
192.168.2.36:5060 2001 105 Registered Sat, 07 Jan 2012 16:20:29

Now when I am trying to call from hotrixbox extension (say 2004) to other side trixbox1 extension 1015, I am getting following msg -

all circuits are busy now

How to diagnosis this issue? What more do I need to do... Please help.




01/08/12--14:52: Hold Music on Transfer but not on internal calls? (chan 2127936)

Hi All,

I noticed that the default behaviour of Asterisk when transferring calls is that the caller fist goes on hold (when you press the transfer key on the IP phone) and then once the destination has been entered and the transfer completed, the caller is taken off hold and ringing tones are played.

I want on hold music to play the whole time, so I read this thread: http://fonality.com/trixbox/forums/trixbox-forums/open-discussion...

So I've made the alterations and yes now hold music plays during a transfer. The down side is that now when you call an internal extension, you get played hold music instead of the IP phone ringing, which is not ideal, I'd far prefer that the phone rang rather then sang you a song when you're calling internally.

Is there anyway to change this behaviour so that transferring calls plays hold music, but internal calls just ring?




01/09/12--12:58: Second M670 extension, only one row blinking lamps (chan 2127936)

I'm running into an issue with my second Aastra m670 extension module. While all the lights light up fine on the first module, on the second, only the left row of LED's light up with the xfer/BLF option tied to any extension.

I swapped out the second M670 with another, and yet, I have the same problem. This leads me to believe it is something in the software not working. I haven't the foggiest idea of where to look.

I know linux somewhat well, but not XML scripting. But, I can easily learn if needed.

On the second module, keys 19-20 are assigned as blf/xfer, with global as the line noted
keys, 21-25 are left blank
key 26-36 as also assigned as blf/xfer. Also global as the line chose.

The other side, 1-18, are all blf/xfer, as is the first module, 1-36.




01/09/12--16:35: Inbound call routing on second server (chan 2127936)

I am a newbie regarding Asterix.

We have three servers of TrxBox Free PBX installed.
Version : Asterisk 1.2.13 svn rev 47264

Current scenario

Main Server
1.1.1.1
BRI card installed, incoming and local NWD calls are routed through it.
Most of the extensions are configured on it
3 dgits extensions
XXX
1XX
2XX
3XX
4XX
5XX
6XX
7XX

Second Server
1.1.1.2
4 digits extensions
8XXX
For NWD calls use route for main

Third Server
2.2.2.1
Located in separate building
Few extension are configured there.
4 digits extension
16XX
For NWD calls use route for main

4 ZAP Trunks for BRI channels

UAN and 4 DID’s are configured on BRI card one interface.
All servers are interlinked with each other through IAX2 Trunk and can easily make calls among each other.
Users on other servers make NWD calls through Main Server

Issue
Extensions configured on server 2 and 3 can’t receive the calls from NWD interface.
We have the option enabled that the caller on DID through IVR can dial his extension.
I need to change that option and configure that if caller on DIDs want to dial the extension it should be #
Make the IVR options that calls forward to another server extensions, ringgroup.

I am very much thankful to you for your cooperation and assistance in this scenario.




01/09/12--23:00: Asterisk always restart on Tribox (chan 2127936)

Dear All,

Have any one know how to fix the asterisk restart itself.

Please help me.

Thanks,
Leakhena




01/10/12--01:34: I want to replace Trixbox (chan 2127936)

for sure we all know that trixbox is coming to an end
no support
old forum posts
a lot of problems with no solution
...

any suggestions for a FREE replaceable Asterisk system with a good updated and up-to-date support ??
what is your advice




01/10/12--10:20: 911 calls get "call can not be completed as dialed" (chan 2127936)

This morning I discovered that dialing 911 is not ringing the 911 center. Instead I get "your call cannot be completed as dialed".
Before I call the provider I thought I'd see if anyone here recognoized a problem in the call progress that may be corrected on my side.
I have only 2 outbound routes. Route 0 is 911 9|911 Route 1 is everything else. All trunks are SIP.
I don't see anything usual compared to other calls into the trunk.

Below is the CLI log:
Connected to Asterisk 1.6.0.26-FONCORE-r78 currently running on trixbox1 (pid = 30228)
Verbosity was 3 and is now 8
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Executing [911@from-internal:1] Macro("SIP/200-00000887", "user-callerid,SKIPTTL,") in new stack
-- Executing [s@macro-user-callerid:1] Set("SIP/200-00000887", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:2] GotoIf("SIP/200-00000887", "0?report") in new stack
-- Executing [s@macro-user-callerid:3] ExecIf("SIP/200-00000887", "1?Set(REALCALLERIDNUM=200)") in new stack
-- Executing [s@macro-user-callerid:4] Set("SIP/200-00000887", "AMPUSER=200") in new stack
-- Executing [s@macro-user-callerid:5] Set("SIP/200-00000887", "AMPUSERCIDNAME=Ms Smith") in new stack
-- Executing [s@macro-user-callerid:6] GotoIf("SIP/200-00000887", "0?report") in new stack
-- Executing [s@macro-user-callerid:7] Set("SIP/200-00000887", "AMPUSERCID=200") in new stack
-- Executing [s@macro-user-callerid:8] Set("SIP/200-00000887", "CALLERID(all)="Ms Smith" <200>") in new stack
-- Executing [s@macro-user-callerid:9] ExecIf("SIP/200-00000887", "0?Set(CHANNEL(language)=)") in new stack
-- Executing [s@macro-user-callerid:10] GotoIf("SIP/200-00000887", "1?continue") in new stack
-- Goto (macro-user-callerid,s,19)
-- Executing [s@macro-user-callerid:19] NoOp("SIP/200-00000887", "Using CallerID "Ms Smith" <200>") in new stack
-- Executing [911@from-internal:2] Set("SIP/200-00000887", "_NODEST=") in new stack
-- Executing [911@from-internal:3] Macro("SIP/200-00000887", "record-enable,200,OUT,") in new stack
-- Executing [s@macro-record-enable:1] GotoIf("SIP/200-00000887", "1?check") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s@macro-record-enable:4] AGI("SIP/200-00000887", "recordingcheck,20120110-103448,1326213288.10206") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20120110-103448,1326213288.10206: Outbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [s@macro-record-enable:5] MacroExit("SIP/200-00000887", "") in new stack
-- Executing [911@from-internal:4] Macro("SIP/200-00000887", "dialout-trunk,2,911,,") in new stack
-- Executing [s@macro-dialout-trunk:1] Set("SIP/200-00000887", "DIAL_TRUNK=2") in new stack
-- Executing [s@macro-dialout-trunk:2] GosubIf("SIP/200-00000887", "0?sub-pincheck,s,1") in new stack
-- Executing [s@macro-dialout-trunk:3] GotoIf("SIP/200-00000887", "0?disabletrunk,1") in new stack
-- Executing [s@macro-dialout-trunk:4] Set("SIP/200-00000887", "DIAL_NUMBER=911") in new stack
-- Executing [s@macro-dialout-trunk:5] Set("SIP/200-00000887", "DIAL_TRUNK_OPTIONS=trwW") in new stack
-- Executing [s@macro-dialout-trunk:6] Set("SIP/200-00000887", "OUTBOUND_GROUP=OUT_2") in new stack
-- Executing [s@macro-dialout-trunk:7] GotoIf("SIP/200-00000887", "0?nomax") in new stack
-- Executing [s@macro-dialout-trunk:8] GotoIf("SIP/200-00000887", "0?chanfull") in new stack
-- Executing [s@macro-dialout-trunk:9] GotoIf("SIP/200-00000887", "0?skipoutcid") in new stack
-- Executing [s@macro-dialout-trunk:10] Set("SIP/200-00000887", "DIAL_TRUNK_OPTIONS=twW") in new stack
-- Executing [s@macro-dialout-trunk:11] Macro("SIP/200-00000887", "outbound-callerid,2") in new stack
-- Executing [s@macro-outbound-callerid:1] ExecIf("SIP/200-00000887", "0?Set(CALLERPRES()=)") in new stack
-- Executing [s@macro-outbound-callerid:2] ExecIf("SIP/200-00000887", "0?Set(REALCALLERIDNUM=200)") in new stack
-- Executing [s@macro-outbound-callerid:3] GotoIf("SIP/200-00000887", "1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,6)
-- Executing [s@macro-outbound-callerid:6] Set("SIP/200-00000887", "USEROUTCID=") in new stack
-- Executing [s@macro-outbound-callerid:7] Set("SIP/200-00000887", "EMERGENCYCID=") in new stack
-- Executing [s@macro-outbound-callerid:8] Set("SIP/200-00000887", "TRUNKOUTCID=gensip comp <555-888-0000>") in new stack
-- Executing [s@macro-outbound-callerid:9] GotoIf("SIP/200-00000887", "1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,12)
-- Executing [s@macro-outbound-callerid:12] ExecIf("SIP/200-00000887", "1?Set(CALLERID(all)=gensip comp <555-888-0000>)") in new stack
-- Executing [s@macro-outbound-callerid:13] ExecIf("SIP/200-00000887", "0?Set(CALLERID(all)=)") in new stack
-- Executing [s@macro-outbound-callerid:14] ExecIf("SIP/200-00000887", "0?Set(CALLERPRES()=prohib_passed_screen)") in new stack
-- Executing [s@macro-dialout-trunk:12] ExecIf("SIP/200-00000887", "0?AGI(fixlocalprefix)") in new stack
-- Executing [s@macro-dialout-trunk:13] Set("SIP/200-00000887", "OUTNUM=911") in new stack
-- Executing [s@macro-dialout-trunk:14] Set("SIP/200-00000887", "custom=SIP/sip-gensip-cons-01") in new stack
-- Executing [s@macro-dialout-trunk:15] ExecIf("SIP/200-00000887", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^)twW)") in new stack
-- Executing [s@macro-dialout-trunk:16] Macro("SIP/200-00000887", "dialout-trunk-predial-hook,") in new stack
-- Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit("SIP/200-00000887", "") in new stack
-- Executing [s@macro-dialout-trunk:17] GotoIf("SIP/200-00000887", "0?bypass,1") in new stack
-- Executing [s@macro-dialout-trunk:18] GotoIf("SIP/200-00000887", "0?customtrunk") in new stack
-- Executing [s@macro-dialout-trunk:19] Dial("SIP/200-00000887", "SIP/sip-gensip-cons-01/911,300,twW") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
-- Called sip-gensip-cons-01/911
-- SIP/sip-gensip-cons-01-00002333 is making progress passing it to SIP/200-00000887
trixbox1*CLI>
trixbox1*CLI>
trixbox1*CLI>
trixbox1*CLI> ---- Your call cannot be completed as dialed message heard here
trixbox1*CLI> ---- Then we hang up
trixbox1*CLI>
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/200-00000887' in macro 'dialout-trunk'
== Spawn extension (from-internal, 911, 4) exited non-zero on 'SIP/200-00000887'
-- Executing [h@from-internal:1] Macro("SIP/200-00000887", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] GotoIf("SIP/200-00000887", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,4)
-- Executing [s@macro-hangupcall:4] GotoIf("SIP/200-00000887", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,7)
-- Executing [s@macro-hangupcall:7] GotoIf("SIP/200-00000887", "1?theend") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] Hangup("SIP/200-00000887", "") in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on 'SIP/200-00000887' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/200-00000887'
trixbox1*CLI> exit




01/11/12--04:49: Configuration failure (chan 2127936)

Hi

I've been running Trixbox Pro Standard Edition for several years, mostly problem free. This morning however, changes are no longer being synchronised.

"We have tried to transfer the recent configuration changes you have made in this software down to your premise trixbox Pro server"

I've got a user who increasingly desperately wants to undivert their extension. I'd be extremely grateful if anybody could offer any advice or help in resolving the issue.
I've checked the network at my end, and everything looks fine. I've also rebooted the trixbox server, checked for disk errors etc.

The server ID is 113861

Many thanks,
Alex




01/11/12--11:21: How to Use Google to Perform Free Cell Phone Number Search Without Google Phonebook (chan 2127936)

One of the most well known services provided by Google was reverse phone lookup. If one used the ‘phonebook:’ or ‘rphonebook:’ operators along with a USA phone number, Google showed the owner of the phone number for sure, unless it was unlisted on their servers.

On a related note read about free cell phone number lookup – a business angle of the story.

As we all know, since November 2010, Google shut down its Phonebook service. And that too, without any press release. The reason was stated as “causing an endless source of hassles for people”. One question we must ask is why? Well, the obvious reason being that some googlers were shocked to see their very own personal numbers and addresses listed on the most well known search engine on this planet. So, Google had to handle endless numbers of “takedown requests and outraged letters” from these very users.

Google, in the end, came up with a solution to drop the “phonebook:” and “rphonebook:” operators. If you want your information removed from Google, visit http://www.google.com/intl/en/help/pbremoval.html.

But, as we say, the end is not near. You can still search for phone numbers using Google. Now, don’t get too ahead of yourself.

Some Googlers are annoyed by this. One Google’s support forum poter, billdickason comments

“I have tried about six other phone search engines. None of them have the search capability of Google Phonebook. I just finished finding classmates for a 50-yr high school reunion. I couldn’t have done it with the other phonebooks. Most of them are not as searchable and are terribly out of date with their listings. I haven’t found any way to send something to Google. At some point in the not too distant future it will probably not matter. My kids don’t have land lines only cell phones. Good luck in finding them.”

Now, to the meat and potatoes of this article. Doing a reverse phone lookup using Google is still hard. But we make it easy as 5 simple tips:

1. Type the number in Google with the dashes in between. e.g. 666-666-6666
2.If their number is listed in phone lookup services, try to find out the sites. If it is a business, it will most likely be paid. If you got the money, pay for it and you’ll have gotten access to the address.
3.It might be listed on a social network. If so, you’ll have got the username or even the real name of the person.
4.Their number might still be on google. If you know the name, find the zip code by checking the social website or any other source.
5.Type their name and zip code on google for their complete information – name,phone number,address and a Google map with directions to the residence.

Note you might never find the number on Google. This method is not error-proof and you might not find the number. But who knows, you might get lucky. Sites asking for payment may not have all the information about the number so beware of such shady sites. Before payment, you might want to enter the site’s name followed by ‘scam’ in Google, just to check.




01/11/12--15:24: Cisco 7960G Sip Firmware (chan 2127936)

Anyone that needs SIP firmware for 7960 or 7960G email or msg me and
I'll send you a copy... I have sample .cnf the SIP Firmware everything

I'm also selling a bunch of 7960's pre-flashed to SIP from an office that shut down.

check them out here: http://http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=1208411...




01/12/12--01:09: Registering cisco phone-7911 procedure. (chan 2127936)

hi
my cisco phone-7911 is not registering and
i don,t know whats ta problem.




01/12/12--10:19: No clue.. (chan 2127936)

I have googled this, and small pieces of it to find what the problem may be. I cannot find anything. Thanks guys, anything helps :)

Here is a piece of my log file:

[2012-01-12 10:16:17] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:18] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:19] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:20] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:21] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:31] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:32] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:33] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:34] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:35] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:45] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cd768 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:46] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cd768 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:47] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cd768 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:48] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cd768 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:49] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cd768 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:16:59] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cd768 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:17:00] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cd768 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:17:01] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cd768 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:17:02] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cd768 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:17:03] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cd768 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:17:13] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:17:14] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:17:15] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:17:16] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted
[2012-01-12 10:17:17] WARNING[3726] chan_sip.c: sip_xmit of 0xa1cdd80 (len 491) to 10.1.10.37:5061 returned -1: Operation not permitted




01/16/12--00:35: Bad Audio Timing (chan 2127936)

Hi everyone,

I am currently running trixbox version 2.6.0.0 on a dedicated AMD 64 300+, 1GB, 80HDD, and have a X100P FXO card installed.

Audio playback of recorded messages in calls (ie: IVR, voice commands etc) the audio seems very choppy and can often skip. I believe this may be a timing source issue.

I have just run the dahdi_test command and found some shocking results:

Best: 100.000 -- Worst: 0.629 -- Average: 66.985243, Difference: 66.986553

I am sure I have done something wrong because, from what I understand, those are really bad results. What am I doing wrong? Is the X100P meant to be used as a timing source? Do I need to set this as the source in the settings somewhere?




01/16/12--02:26: I've connected to Trixbox FTP can I back up files to Trixbox public folder? (chan 2127936)

Hi,

I want to backup Trixbox quickly rather than over to my FTP server online as the files are getting very large now!

I have connected to Trixbox FTP using the ftpuser and asteriskftp username and password and it connects me to a public and config folder. Is there away I can backup my Trixbox to this public folder so I can then download it to another PC on the network?

Many thanks
Joe




01/16/12--07:13: Zoiper (or any soft-phone) DTMF not working while all hard-phones are (chan 2127936)

Good morning all!

We are having issues with setting up our soft-phones. We currently have Trixbox 2.6.2.5 with a very simple setup. We have Aastra i57 hard-phones and are have no issues at all with them. We are moving to have a few soft-phones installed but are having issues once the call is made the touch tones are not being recognized.

We are using Zoiper and are interesting in purchasing it once the demo proves to be a success. We have trouble shot and tried the basic items that we could find by searching forums. We have forced the DTMF to rfc2833 we have tried inline but still nothing. The Zoiper soft-phone will register and make/receive calls but once the call is active no tones are recognized. This proves to be a major issue as IVR or transferring is not working.

Please let me know if there are any things that I can try...
Thank you in advanced!!!




01/17/12--00:18: Caller ID (chan 2127936)

Hi,

I have problems with setting caller ID. Could anybody help me with that? I've tried to put this combination , but each time the Outbound route caller ID is displaying my voip account number.

Please help me with that, cause it's becoming frustrating...

Regards,
majenka




01/17/12--05:09: Access points and coverage (chan 2127936)

Hi everyone

My boss has a brilliant idee in his office building. I will explain what is going to happen. We have 7 hotspots areas in one office building with 3 floors, but if you walk from point A ( access point number 6) to B (acces point number 7) on the second and same floor you need to connect to the other access point even if you go downstairs. So my boss want that you can walk with your laptop from point A to B without logging in to the other point and he ask me to find it. I searched for 2 days and I can't find a good solutions. I can't find a solution to connect 6 DWL-3200 AP with each other ,so I hope that someone of you can help with this if you don't understand my English let me know and I will try to explain it in an other way

Thank you already if you want to help me
Stan




01/18/12--13:51: Using Trix as a SIP Gateway forward all calls to CUCM (chan 2127936)

Hey all,

Have a dilemma here, we want to use trixbox as a SIP gateway that will forward all calls to our CUCM servers.

We've got 800+ phones and even more DIDs, including 800s and local numbers. We constantly are adding new customers and I would prefer not to manage dial plans and translation patterns in multiple spots.

We have PRIs to our CUCM and then the newest are the Trix Boxes to CUCM for SIP.

What I would like to do is forward all calls to the CUCM server to be translated and forwarded off to a phone.

For Example:
555-555-5555 calls 800-555-555
I want to forward the call over a SIP trunk that is already setup and working but not involve an extension or inbound route for the 800-555-5555 specifically.
I want something similar to a route that says to forward calls with a wildcard like SIP/XXXXXXXXXX@10.110.101.15

Currently for a small group of numbers we are using an inbound route, that forward to a device that has a device address like SIP/extension@10.110.101.15

Can anyone point me in the correct direction here?

Thanks,




01/18/12--14:04: Polycom Soundpoint 330 phones not registering (chan 2127936)

I'm tinkering around with Trixbox 2.8.0.4 on a PowerEdge 2650 to feed my curiosity. Don't have a SIP Trunk yet, as my provider is still working on it. For now, just want to make calls from one Soundpoint 330 to another on my network. Very simple setup.

I've followed the instructions for registering my handsets, however when I dial the extension from one Polycom 330 to another, I receive the message "the number you have dialed is not in service." Consequently, when viewing the Line Information & Server Status menu on my phones, it shows the correct information for each, followed by "(Not Registered)." I know that the phones communicate fine with the PBX, as the phones were able to pull the latest bootrom from the TFTP of the PBX just fine.

Here's what I did, step by step.

1. Installed "firmware-polycom" package in the "Packages" menu in the Trixbox GUI
2. typed "setup-polycom" then selected the appropriate ETH interface on my PBX.
3. Inputted MAC address/other required info in the "Endpoint Manager" for my Polycoms.
4. Ensured phones had correct TCP-IP/Server IP info. Saved settings.
5. Unplugged/replugged both phones. Phones saw the PBX and were able to download/install the latest BootROM.
6. Disabled my firewall to no avail.

Any idea why they are not registering?




01/18/12--15:44: 1 extension returns dialstatus busy rather than dialstatus noanswer (chan 2127936)

I have one extension that after the phone rings for 3 or 4 times, it returns DIALSTATUS=BUSY and the voicemail plays the busy message. What it is supposed to do is return DIALSTATUS=NOANSWER. If I create a new extension and use the same phone it works fine -- so it is not a phone problem signaling wrong. If I delete the offending extension and re-add it, it still has the same wrong behavior. What is the problem? How can I fix it?

All other extensions work fine.

Thank you -- see below for relevant log entries.

This is the bad extension (4209):
-- Executing Dial("Zap/18-1", "SIP/4209|15|trwW") in new stack
-- Called 4209I>
-- SIP/4209-0a07d0c8 is ringing
-- Playing 'conf-hasjoin' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'phpagi' logged on from 127.0.0.1
== Manager 'phpagi' logged off from 127.0.0.1
-- Nobody picked up in 15000 ms
-- Executing Set("Zap/18-1", "DIALSTATUS=BUSY") in new stack
-- Executing GosubIf("Zap/18-1", "0?BUSY|1") in new stack
-- Executing GotoIf("Zap/18-1", "0?exit") in new stack
-- Executing Set("Zap/18-1", "SV_DIALSTATUS=BUSY") in new stack

This is the good extension (4211):
-- Executing Dial("Zap/20-1", "SIP/4211|22|trwWM(auto-blkvm)") in new stack
-- Called 4211
-- SIP/4211-0a07d0c8 is ringing
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'phpagi' logged on from 127.0.0.1
== Manager 'phpagi' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_additional.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'phpagi' logged on from 127.0.0.1
== Manager 'phpagi' logged off from 127.0.0.1
-- Nobody picked up in 22000 ms
-- Executing Set("Zap/20-1", "DIALSTATUS=NOANSWER") in new stack
-- Executing GosubIf("Zap/20-1", "0?NOANSWER|1") in new stack
-- Executing Goto("Zap/20-1", "nextstep") in new stack
-- Goto (from-did-direct,4211,21)




01/18/12--19:05: Trouble setting Misc destination to external phone number and using for an IVR call forward. (chan 2127936)

system status version: 2.6.2.5
v2.8.0.4

Firstly, thanks to all of you! Without this forum, I don't know where I'd be.
I have set up an IVR with several options, one of which I want to be able to forward incoming calls to a series of external cell numbers. If I understand correctly, the way to do this is to setup a misc destination and set the IVR to go directly to it (I'm using "t" as the code, since I want the forward to occur after the voice recording plays). I have set up the misc destination with the phone number I want it to go to. 214xxxxxxx
The IVR does everything it's supposed to, but I get the message, "All circuits are busy now. Please try again later." everytime I test it. Any ideas? I'm sure it's probably a number format issue, but I can't find anything about how I should put the number in differently. Thanks again,

I've tried the following formats:
214xxxxxxx
1214xxxxxxx
0214xxxxxxx
01214xxxxxxx
091214xxxxxxx
091214xxxxxxx
91214xxxxxxx
9214xxxxxxx

I even tried to set up a ring group with 214xxxxxxx# in the extension list (supposedly this is what is needed in order to dial an outside number). When I do this, it rings once (though not on the other end--just hear a ringing tone on the trixbox end) and immediately disconnects. Any ideas?




01/19/12--11:05: Cannot dial to sip accounts (internally from sip or from outside) (chan 2127936)

Hi,

i can do calls from inside to outside world, but i cannot call from outside to inside or internally with sip ...

i also deleted and recreated the extensions ...

here is a log from a sip -> sip call

[Jan 19 19:41:44] VERBOSE[21672] logger.c: Extension Changed 24[ext-local] new state InUse for Notify User 24
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [26@from-internal:1] Macro("SIP/24-08e0b188", "exten-vm|novm|26") in new stack
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:1] Macro("SIP/24-08e0b188", "user-callerid") in new stack
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:1] NoOp("SIP/24-08e0b188", "user-callerid: device 24") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Noop
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:2] Set("SIP/24-08e0b188", "AMPUSER=24") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:3] GotoIf("SIP/24-08e0b188", "0?report") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: GotoIf
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:4] ExecIf("SIP/24-08e0b188", "1|Set|REALCALLERIDNUM=24") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: ExecIf
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:5] NoOp("SIP/24-08e0b188", "REALCALLERIDNUM is 24") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Noop
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:6] Set("SIP/24-08e0b188", "AMPUSER=24") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:7] Set("SIP/24-08e0b188", "AMPUSERCIDNAME=Juergen Arendt1") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:8] GotoIf("SIP/24-08e0b188", "0?report") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: GotoIf
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:9] Set("SIP/24-08e0b188", "AMPUSERCID=24") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:10] Set("SIP/24-08e0b188", "CALLERID(all)="Juergen Arendt1" <24>") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:11] Set("SIP/24-08e0b188", "REALCALLERIDNUM=24") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:12] ExecIf("SIP/24-08e0b188", "0|Set|CHANNEL(language)=") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: ExecIf
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:13] NoOp("SIP/24-08e0b188", "TTL: ARG1: novm") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Noop
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:14] GotoIf("SIP/24-08e0b188", "0?continue") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: GotoIf
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:15] Set("SIP/24-08e0b188", "__TTL=64") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:16] GotoIf("SIP/24-08e0b188", "1?continue") in new stack
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Goto (macro-user-callerid,s,23)
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: GotoIf
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-user-callerid:23] NoOp("SIP/24-08e0b188", "Using CallerID "Juergen Arendt1" <24>") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Noop
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Macro
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:2] Set("SIP/24-08e0b188", "RingGroupMethod=none") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:3] Set("SIP/24-08e0b188", "VMBOX=novm") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:4] Set("SIP/24-08e0b188", "EXTTOCALL=26") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] DEBUG[31630] func_db.c: DB: CFU/26 not found in database.
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:5] Set("SIP/24-08e0b188", "CFUEXT=") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] DEBUG[31630] func_db.c: DB: CFB/26 not found in database.
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:6] Set("SIP/24-08e0b188", "CFBEXT=") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:7] Set("SIP/24-08e0b188", "RT=""") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:8] Macro("SIP/24-08e0b188", "record-enable|26|IN") in new stack
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-record-enable:1] GotoIf("SIP/24-08e0b188", "0?2:4") in new stack
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Goto (macro-record-enable,s,4)
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: GotoIf
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-record-enable:4] AGI("SIP/24-08e0b188", "recordingcheck|20120119-194144|1326998504.127") in new stack
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- AGI Script recordingcheck completed, returning 0
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: AGI
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-record-enable:5] NoOp("SIP/24-08e0b188", "No recording needed") in new stack
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Noop
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: Macro
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:9] Macro("SIP/24-08e0b188", "dial||tr|26") in new stack
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-dial:1] GotoIf("SIP/24-08e0b188", "1?dial") in new stack
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Goto (macro-dial,s,3)
[Jan 19 19:41:44] DEBUG[31630] app_macro.c: Executed application: GotoIf
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Executing [s@macro-dial:3] AGI("SIP/24-08e0b188", "dialparties.agi") in new stack
[Jan 19 19:41:44] VERBOSE[31630] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- AGI Script dialparties.agi completed, returning 0
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: AGI
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-dial:4] NoOp("SIP/24-08e0b188", "Returned from dialparties with no extensions to call and DIALSTATUS: ") in new stack
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: NoOp
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: Macro
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:10] Set("SIP/24-08e0b188", "SV_DIALSTATUS=") in new stack
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:11] GosubIf("SIP/24-08e0b188", "0?docfu|1") in new stack
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: GosubIf
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:12] GosubIf("SIP/24-08e0b188", "0?docfb|1") in new stack
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: GosubIf
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:13] Set("SIP/24-08e0b188", "DIALSTATUS=") in new stack
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: Set
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:14] NoOp("SIP/24-08e0b188", "Voicemail is novm") in new stack
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: NoOp
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-exten-vm:15] GotoIf("SIP/24-08e0b188", "1?s-|1") in new stack
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Goto (macro-exten-vm,s-,1)
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: GotoIf
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [26@from-internal:2] Hangup("SIP/24-08e0b188", "") in new stack
[Jan 19 19:41:45] VERBOSE[31630] logger.c: == Spawn extension (from-internal, 26, 2) exited non-zero on 'SIP/24-08e0b188'
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [h@from-internal:1] Macro("SIP/24-08e0b188", "hangupcall") in new stack
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-hangupcall:1] ResetCDR("SIP/24-08e0b188", "w") in new stack
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: ResetCDR
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-hangupcall:2] NoCDR("SIP/24-08e0b188", "") in new stack
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: NoCDR
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-hangupcall:3] GotoIf("SIP/24-08e0b188", "1?skiprg") in new stack
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Goto (macro-hangupcall,s,6)
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: GotoIf
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-hangupcall:6] GotoIf("SIP/24-08e0b188", "1?skipblkvm") in new stack
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Goto (macro-hangupcall,s,9)
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: GotoIf
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-hangupcall:9] GotoIf("SIP/24-08e0b188", "1?theend") in new stack
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Goto (macro-hangupcall,s,11)
[Jan 19 19:41:45] DEBUG[31630] app_macro.c: Executed application: GotoIf
[Jan 19 19:41:45] VERBOSE[31630] logger.c: -- Executing [s@macro-hangupcall:11] Hangup("SIP/24-08e0b188", "") in new stack
[Jan 19 19:41:45] VERBOSE[31630] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/24-08e0b188' in macro 'hangupcall'
[Jan 19 19:41:45] VERBOSE[31630] logger.c: == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/24-08e0b188'
[Jan 19 19:41:45] VERBOSE[21672] logger.c: Extension Changed 24[ext-local] new state Idle for Notify User 24

and this is a call from outside to a sip ...

[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [89540238@from-pstn:1] Set("mISDN/3-u169", "__FROM_DID=89540238") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [89540238@from-pstn:2] Gosub("mISDN/3-u169", "app-blacklist-check|s|1") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@app-blacklist-check:1] LookupBlacklist("mISDN/3-u169", "") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@app-blacklist-check:2] GotoIf("mISDN/3-u169", "0?blacklisted") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@app-blacklist-check:3] Return("mISDN/3-u169", "") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [89540238@from-pstn:3] GotoIf("mISDN/3-u169", "0 ?cidok") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [89540238@from-pstn:4] Set("mISDN/3-u169", "CALLERID(name)=01632587799") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [89540238@from-pstn:5] NoOp("mISDN/3-u169", "CallerID is "01632587799" <01632587799>") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [89540238@from-pstn:6] Set("mISDN/3-u169", "__CALLINGPRES_SV=allowed_not_screened") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [89540238@from-pstn:7] SetCallerPres("mISDN/3-u169", "allowed_not_screened") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [89540238@from-pstn:8] Goto("mISDN/3-u169", "from-did-direct|26|1") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Goto (from-did-direct,26,1)
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [26@from-did-direct:1] Macro("mISDN/3-u169", "exten-vm|novm|26") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:1] Macro("mISDN/3-u169", "user-callerid") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:1] NoOp("mISDN/3-u169", "user-callerid: 01632587799 01632587799") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Noop
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:2] Set("mISDN/3-u169", "AMPUSER=01632587799") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:3] GotoIf("mISDN/3-u169", "0?report") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: GotoIf
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:4] ExecIf("mISDN/3-u169", "1|Set|REALCALLERIDNUM=01632587799") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: ExecIf
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:5] NoOp("mISDN/3-u169", "REALCALLERIDNUM is 01632587799") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Noop
[Jan 19 20:02:59] DEBUG[1230] func_db.c: DB: DEVICE/01632587799/user not found in database.
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:6] Set("mISDN/3-u169", "AMPUSER=") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] DEBUG[1230] func_db.c: DB: AMPUSER//cidname not found in database.
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:7] Set("mISDN/3-u169", "AMPUSERCIDNAME=") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:8] GotoIf("mISDN/3-u169", "1?report") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Goto (macro-user-callerid,s,13)
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: GotoIf
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:13] NoOp("mISDN/3-u169", "TTL: ARG1: novm") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Noop
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:14] GotoIf("mISDN/3-u169", "0?continue") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: GotoIf
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:15] Set("mISDN/3-u169", "__TTL=64") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:16] GotoIf("mISDN/3-u169", "1?continue") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Goto (macro-user-callerid,s,23)
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: GotoIf
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-user-callerid:23] NoOp("mISDN/3-u169", "Using CallerID "01632587799" <01632587799>") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Noop
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Macro
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:2] Set("mISDN/3-u169", "RingGroupMethod=none") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:3] Set("mISDN/3-u169", "VMBOX=novm") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:4] Set("mISDN/3-u169", "EXTTOCALL=26") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] DEBUG[1230] func_db.c: DB: CFU/26 not found in database.
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:5] Set("mISDN/3-u169", "CFUEXT=") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] DEBUG[1230] func_db.c: DB: CFB/26 not found in database.
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:6] Set("mISDN/3-u169", "CFBEXT=") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:7] Set("mISDN/3-u169", "RT=""") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:8] Macro("mISDN/3-u169", "record-enable|26|IN") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-record-enable:1] GotoIf("mISDN/3-u169", "0?2:4") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Goto (macro-record-enable,s,4)
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: GotoIf
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-record-enable:4] AGI("mISDN/3-u169", "recordingcheck|20120119-200259|1326999779.130") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- AGI Script recordingcheck completed, returning 0
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: AGI
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-record-enable:5] NoOp("mISDN/3-u169", "No recording needed") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Noop
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Macro
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:9] Macro("mISDN/3-u169", "dial||tr|26") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-dial:1] GotoIf("mISDN/3-u169", "1?dial") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Goto (macro-dial,s,3)
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: GotoIf
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-dial:3] AGI("mISDN/3-u169", "dialparties.agi") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- AGI Script dialparties.agi completed, returning 0
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: AGI
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-dial:4] NoOp("mISDN/3-u169", "Returned from dialparties with no extensions to call and DIALSTATUS: ") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: NoOp
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Macro
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:10] Set("mISDN/3-u169", "SV_DIALSTATUS=") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:11] GosubIf("mISDN/3-u169", "0?docfu|1") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: GosubIf
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:12] GosubIf("mISDN/3-u169", "0?docfb|1") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: GosubIf
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:13] Set("mISDN/3-u169", "DIALSTATUS=") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: Set
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:14] NoOp("mISDN/3-u169", "Voicemail is novm") in new stack
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: NoOp
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [s@macro-exten-vm:15] GotoIf("mISDN/3-u169", "1?s-|1") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Goto (macro-exten-vm,s-,1)
[Jan 19 20:02:59] DEBUG[1230] app_macro.c: Executed application: GotoIf
[Jan 19 20:02:59] VERBOSE[1230] logger.c: -- Executing [26@from-did-direct:2] Hangup("mISDN/3-u169", "") in new stack
[Jan 19 20:02:59] VERBOSE[1230] logger.c: == Spawn extension (from-did-direct, 26, 2) exited non-zero on 'mISDN/3-u169'
[Jan 19 20:02:59] DEBUG[1230] chan_misdn.c: misdn_hangup(mISDN/3-u169)

Any idea ?

best wishes

Niels




01/20/12--15:58: Boot TRIXBOX CE (chan 2127936)

Some one could help me?

When I boot TRIXBOX CE CD I loose all HD documents and programs?

It means that I need to have an exclusive TRIXBOX HD?




01/23/12--02:14: Abuse asterisk based PBX (chan 2127936)

Hello All,

In the last months I saw different abuses of Asterisk based PBXes. All of my PBXes are secured by some good firewall but I have got the experience that when u shutdown the firewall and letting a PBX like Trixbox for some time directly connected with Internet, It will be abused within a short time.

Abusers can login on some arbitrary extension within a short time and use the calling routes for this extension. It doesn't matter what is the username or how long and how strong the password is, they will login and start calling to expensive numbers in Africa.

Can somebody tell me how they can do this? Can they login without any password?

It doesn't matter witch system, the abuse will happening by Trixbox, FreePBX, 4psa with kamilio SIP server and lot of other.

Most important for this is how can we stop this kind of abuse.

Thanks for each reply!




01/23/12--17:00: GrandStream GXV3175 visual alerts (chan 2127936)

We are changing from centrex to a trixbox ce system using GrandStream GXV3175's. Everything works fine for me, I would like to replicate a feature in ring groups in centrex. Basically a line rings in a ring group, on the primary phone it rings and the others can also pickup but their phone does not ring just lights up on the line. Any ideas?




01/25/12--12:25: Call routing inconsistent (chan 2127936)

Hi,

I've got Trixbox 2.8.0.4 running on a PhoneBochs mini w/ 5 analog lines in and about 25 extensions. The system works fine most of the time, but I receive complaints on a regular basis that the extension entered gets routed to the wrong phone when calling from an outside line to the IVR. For example, someone dials extension 223 and gets connected to the person at extension 225. Other times, the system reports that it is an invalid extension. There is no discernable pattern to when it happens, but it happens much too frequently to be chalked up to "fat finger" dialing. Any suggestions on what could cause this behavior or where I should be looking to troubleshoot?

Thanks in advance for your help!




01/26/12--12:14: Configuration error message (chan 2127936)

Hello,

After making some configuration changes to our server, I recently received this message:
Configuration failure: We have tried to transfer the recent configuration changes you have made in this software down to your premise Fonality, Inc. server. Normally, this occurs instantly, but for an unknown network reason, the changes have not yet reached your server.

Your changes will not be complete and you will continue to see this message until you press the "Complete Changes" button.

Internet connectivity is fine, DNS is fine and VPN tunnels to Fonality are established. Any ideas?

Server ID: 131256

Thanks,
Louis




01/27/12--08:19: Llamadas entrantes no se cuelgan (chan 2127936)

Hola buenas, soy nuevo en esto de la VoIP y tengo implementado un server con trixbox 2.8 y tengo ya registrados 3 extensiones. Mi problema es que cada vez que un cliente no termina la llamada, mi terminal vuelve a sonar. Ej:

El cliente llama, la operadora la atiende, pueden hablar sin problemas, y a la hora de terminar la llamada, si la operadora, cuelga el teléfono y el cliente todavía sigue en línea, entonces la terminal vuelve a sonar, no se si me explico

El tema es que al parecer el comando Hangup no está haciendo su trabajo.

Mi configuración es esta:

extensions.conf
[from-sip-external]
;give external sip users congestion and hangup
; Yes. This is _really_ meant to be _. - I know asterisk whinges about it, but
; I do know what I'm doing. This is correct.

exten => _.,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => _.,n,Set(DID=${IF($["${EXTEN:1:2}"=""]?s:${EXTEN})})
exten => _.,n,Goto(snom305,1)

exten => snom305,1,NoOp(Received incoming SIP connection from unknown peer to ${EXTEN})
exten => snom305,n,GotoIf($["${ALLOW_SIP_ANON}"="yes"]?from-trunk,${DID},1)
exten => snom305,n,Dial(Sip/500)
exten => snom305,n,Hangup

sip_additional.conf
[snom305]
disallow=all
type=peer
context=from-sip-external
host=**.**.**.*
username=snom305
fromuser=snom305
secret=****
qualify=yes
canreinvite=no
insecure=port,invite
dtmfmode=rfc2833
allow=g729

Si alguien por favor me puede ayudar con esto se lo agradeceré de todo corazón!




01/27/12--09:05: Aastra 9112i with tb 2.8.0.4 no audio on outgoing calls, incoming are OK (chan 2127936)

Trying to configure tb 2.8.0.4 with Aastra 9112i phones.

When I first set up the phones tb was not "seeing" them as being registered, however, I was able to make outgoing calls just fine.

In my efforts to get the phones registered, I managed to get tb to see them and the phones were able to call each other and with audio both directions. Everything's cool. Then I noticed that only incoming calls were working as expected. Outgoing calls now have no audio either way. I don't believe this is limited to a config issue with the phones as incoming calls routed to custom extensions (e.g. LOCAL/14055551212@outbound-allroutes) have no audio either.

My best guess is that this may be a codec issue.

My router settings mirror that of my two other working tb installs.
My sip_nat.conf file mirrors that of my other two working tb installs.

What I am needing is a nudge in the right direction on what to check next. Like I said, incoming calls are fine. Outgoing calls have no audio either way.

I'm not sure which config details I should post but I am happy to post any/all config details needed to sort this out.

Here's my aastra.cfg:

#===================================================
# "aastra.cfg" file

alert emergency: 3
alert group: 1
alert internal: 2
alert priority: 4
date format: 7
dhcp: 1
dndkey value: 0

http server: 0.0.0.0
http path:
ring tone: 2

sip dial plan terminator: 0
sip dial plan: "[2-9]11|[56]1[1-5]|958|8502|[2-9]xxxxxx|1[2-9]xxxxxxxxx|*1[2-9]xxxxxxxxx|011x+#|xx*|*xx+#|x+#"
sip digit timeout: 5
sip dtmf method: 2
sip nortel nat support: 1
sip outbound proxy port: 0
sip outbound proxy: 0.0.0.0
sip proxy ip: 192.168.1.88
sip proxy port: 5060
sip registrar ip: 192.168.1.88
sip registrar port: 5060
sip backup proxy ip: xxxx.xxxxxxx.xxx - this has a legit server defined
sip backup proxy port: 5060
sip backup registrar ip: xxxx.xxxxxxx.xxx - this has a legit server defined
sip backup registrar port: 5060
sip registration period: 3600
sip rtp port: 3000
sip session timer: 0
sip transport protocol: 1
time format: 1
time server1: 129.6.15.28
time server2: tick.uh.edu
time server3: server 1.us.pool.ntp.org
time zone code: CST
time zone minutes: 0
time zone name: US-Central
#===================================================

Here's one of my extension .cfg files:

#===================================================
# "MAC.cfg" file

displayName1: "Kris Smith x201"
############################
displayName2: "My Company Name"
sip screen name: "Kris Smith"
sip user name: 201
sip line1 auth name: 201
sip line1 password: xxxxxxxxxxxxxxxx
sip line1 user name: 201
#===================================================

What am I missing?




01/27/12--14:09: Need help with cisco 8961 (chan 2127936)

I am having issues with a new Cisco 8961 phones I am trying out. I have loaded the most recent firmware on the phone. it is 9-2-3-27. I have a working SEP file but I can't make incoming or outgoing phone calls. I get a dial tone but when I try to dial a number nothing happens. It does not matter if it is a local extension or outside number. I have SSH into the trixbox machine and watch the asterisk -r prompt but I don't see any logs when I try the call. Inside of trixbox on config file editor I have edited the sip_custom file to add these lines:
tcpenable=yes
transport=tcp,udp

For the extension I have the follwing settings...

This device uses sip technology.
secret=pass
dtmfmode=rfc2833
canreinvite =yes
context=from-internal
host=dynamic
type=friend
nat=never
port=5060
qualify=no
callgroug=
pickupgroup=
disallow=
allow=
dial=SIP/108
accountcode
mailbox=108@device
deny=0.0.0.0/0.0.0.0
permit=0.0.0.0/0.0.0.0

The phone Status Message screen shows.....

7:00:59p DNS Timeout
7:01:00p DNS Unknown Host example.domain.ext
7:01:00p Updating Trust List
7:01:00p No Trust List installed
7:01:01p SEPE80462EB0E37.cnf.xml (TFTP)
7:01:02p VPN Error: VPN is not Configured.
7:01:25p File Not Found : /gh-sip.jar
7:01:25p Error Updating Locale
7:01:25p File Not Found : /g4-tones.xml
7:01:25p Error Updating Locale

The xml config is....

SIP
admin
password


D/M/YA
Eastern Standard/Daylight Time


192.168.10.4
Unicast








2000
5060
5061
192.168.10.4





false

true
x-serviceuri-cfwdall
x-cisco-serviceuri-pickup
x-cisco-serviceuri-opickup
x-cisco-serviceuri-gpickup
x-cisco-serviceuri-meetme
x-cisco-serviceuri-abbrdial
false
2
true
true
2
2
0
true


6
10
180
3600
5
120
120
5
500
4000
70
false
None

1
false
true

false false

101 3 avt false
false
3

Home 1
false
10
false


9
108

192.168.10.4 5060 108
108

2

3
108
pass
false
1
*99
4
5
108

true
false
false
true


5060
16348
20134
184
0
dialplan.xml




true
2

sip8961.9-2-3-27

false
false

0 1
0
0
0
0
0
0
1,2,3,4,5,6,7
00:00
00:00
00:00
1
1





en_US
1.0.0.0-1
iso-8859-1





1.0.0.0-1

1
http://example.domain.ext/services/authenticate.php
http://example.domain.ext/services/directory.php
http://example.domain.ext/services/menu.xml




96
0
96

4 0



3804


false


I have a feeling the problem is with NAT. I have added false under the sipProfile but still nothing. Please help. I am at the end of my rope.

thanks