iTalk SIP trunks on Asterisk stop working

Forums › VoIP › iTalk SIP trunks on Asterisk stop working - other SIP trunks OK

paul21019

19 posts

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#125724 17-Jul-2013 13:04

Fault,

Cannot make Incoming or Outgoing calls via Italk on Asterisk, stopped suddenly on Monday 15th

My Italk lines on my Asterisk PBX ( Elastix FreePBX), these lines were working and have been for years, I have two other lines 2talk & Voipbuster, both work fine, I can "7777" and my ivr works fine as well

I rang Italk and they wont help as Asterisk is not a supported item...

I see some errors in the logs and I googled them

[Jul 17 12:58:26] VERBOSE[15772] app_dial.c: -- Called Italk /0800800xxx
[Jul 17 12:58:26] WARNING[2593] chan_sip.c: Received response: "Forbidden" from '"200" <sip:649271xxxx@203.184.52.xx>;tag=as204f031a'
[Jul 17 12:58:26] VERBOSE[15772] app_dial.c: -- SIP/Italk -0000007f is circuit-busy
[Jul 17 12:58:26] VERBOSE[15772] app_dial.c: == Everyone is busy/congested at this time (1:0/1/0)
[Jul 17 12:58:26] VERBOSE[15772] pbx.c: -- Executing [s@macro-dialout-trunk:21] NoOp("SIP/200-0000007e", "Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21") in new stack
[Jul 17 12:58:26] VERBOSE[15772] pbx.c: -- Executing [s@macro-dialout-trunk:22] Goto("SIP/200-0000007e", "s-CONGESTION,1") in new stack
[Jul 17 12:58:26] VERBOSE[15772] pbx.c: -- Goto (macro-dialout-trunk,s-CONGESTION,1)
[Jul 17 12:58:26] VERBOSE[15772] pbx.c: -- Executing [s-CONGESTION@macro-dialout-trunk:1] Set("SIP/200-0000007e", "RC=21") in new stack
[Jul 17 12:58:26] VERBOSE[15772] pbx.c: -- Executing [s-CONGESTION@macro-dialout-trunk:2] Goto("SIP/200-0000007e", "21,1") in new stack
[Jul 17 12:58:26] VERBOSE[15772] pbx.c: -- Goto (macro-dialout-trunk,21,1)
[Jul 17 12:58:26] VERBOSE[15772] pbx.c: -- Executing [21@macro-dialout-trunk:1] Goto("SIP/200-0000007e", "continue,1") in new stack
[Jul 17 12:58:26] VERBOSE[15772] pbx.c: -- Goto (macro-dialout-trunk,continue,1)
[Jul 17 12:58:26] VERBOSE[15772] pbx.c: -- Executing [continue@macro-dialout-trunk:1] GotoIf("SIP/200-0000007e", "1?noreport") in new stack
[Jul 17 12:58:26] VERBOSE[15772] pbx.c: -- Goto (macro-dialout-trunk,continue,3)
[Jul 17 12:58:26] VERBOSE[15772] pbx.c: -- Executing [continue@macro-dialout-trunk:3] NoOp("SIP/200-0000007e", "TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 21 - failing through to other trunks") in new stack

Trunk

type=friend
secret=xxxxx
username=64997xxxx
fromuser=64997xxxx
host=akl.italk.co.nz
dtmfmode=rfc2833
insecure=port,invite
nat=yes
canreinvite=no
disallow=all
allow=ulaw&alaw
qualify=yes

64997xxxxx:[password]@akl.italk.co.nz

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#858021 17-Jul-2013 13:24

From that log it appears they're returning a FORBIDDEN response to your INVITE. There could be a number of reasons for this.

Worth capturing a network packet trace with tcpdump and posting it here.

paul21019

19 posts

Geek

#858022 17-Jul-2013 13:25

Forgot to add....

I can register to Italk with xlite and a hardware SIP phone - works perfectly - it just the trunks on the Asterisk which have stopped working

ubergeeknz

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#858024 17-Jul-2013 13:26

Definitely worth comparing a SIP capture of what is working and what is not working. As an aside, 403 Forbidden responses will often include a text reason.

paul21019

19 posts

Geek

#858046 17-Jul-2013 13:49

network packet trace with tcpdump

Because I am a newbie I cannot load a file, what I did was Root> tcpdump -w italk and I got a file, it is virtually unreadable

I cannot upload the image of the file either

Did I use the correct command?

PDE

ubergeeknz

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#858053 17-Jul-2013 13:56

It's readable with something like wireshark. Stick it on google drive or dropbox or something and share it.

Or just enable sip debug ("sip debug" or "sip set debug on") in your asterisk console and get the response from there, it might be enough of a clue.

paul21019

19 posts

Geek

#858061 17-Jul-2013 14:09

Here is the SIP call from 200 to 0800800800

The italk file works in wireshark but there is a lot of truncated messages, i dont know dropbox but here is a link to the file

http://www.sendspace.com/file/fkc30g

===============================================================
Connected to Asterisk 1.6.2.13 currently running on localhost (pid = 2458)
Verbosity is at least 3
localhost*CLI> sip set dubug
No such command 'sip set dubug' (type 'core show help sip set dubug' for other possible commands)
localhost*CLI> sip set debug
No such command 'sip set debug' (type 'core show help sip set debug' for other possible commands)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0800800800@from-internal:1] ResetCDR("SIP/200-00000083", "") in new stack
-- Executing [0800800800@from-internal:2] NoCDR("SIP/200-00000083", "") in new stack
-- Executing [0800800800@from-internal:3] Progress("SIP/200-00000083", "") in new stack
-- Executing [0800800800@from-internal:4] Wait("SIP/200-00000083", "1") in new stack
-- Executing [0800800800@from-internal:5] Progress("SIP/200-00000083", "") in new stack
-- Executing [0800800800@from-internal:6] Playback("SIP/200-00000083", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- <SIP/200-00000083> Playing 'silence/1.gsm' (language 'en')
-- <SIP/200-00000083> Playing 'cannot-complete-as-dialed.gsm' (language 'en')
-- <SIP/200-00000083> Playing 'check-number-dial-again.gsm' (language 'en')
-- Executing [0800800800@from-internal:7] Wait("SIP/200-00000083", "1") in new stack
-- Executing [0800800800@from-internal:8] Congestion("SIP/200-00000083", "20") in new stack
== Spawn extension (from-internal, 0800800800, 8) exited non-zero on 'SIP/200-00000083'
-- Executing [h@from-internal:1] Hangup("SIP/200-00000083", "") in new stack
== Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/200-00000083'
localhost*CLI>

ubergeeknz

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#858067 17-Jul-2013 14:24

Add an "-s 0" to your tcpdump command line and try again. Without the SIP headers it doesn't tell me much we didn't know already.

Also it looks like you're running the wrong command to enable sip tracing in your asterisk console.

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paul21019

19 posts

Geek

#858088 17-Jul-2013 15:10

Here is the new trace

http://www.sendspace.com/file/dhtkvu

There is two calls in there one to italk ( 1st one) and one to my second provider Voipbuster , both calls are to 0800800800 (test number)

Appreciate your help so far

PDE

ubergeeknz

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#858100 17-Jul-2013 15:35

Hmm that capture has no INVITE going to italk. I see it's sending OPTIONS and getting back Forbidden which may be causing it to not attempt routing via that trunk.

Did you enable OPTIONS ping since before?

paul21019

19 posts

Geek

#858105 17-Jul-2013 15:55

"Did you enable OPTIONS ping since before?"

Sorry but I dont know what you mean, what is OPTIONS and where would I enable them?

ubergeeknz

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#858106 17-Jul-2013 15:57

OK forget that.. but your PBX is not sending an INVITE to italk in that capture. So it's not really helpful for troubleshooting :/

paul21019

19 posts

Geek

#858108 17-Jul-2013 16:01

Ok .. the only remedy I can think of is to reload the ISO, I am currently using Elastix , I have downloaded Asterisk Now - so I will do a rebuild from scratch.

I did try to create a new trunk but the problem was the same, perhaps a clean rebuild will help, its not a big job..

ubergeeknz

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#858110 17-Jul-2013 16:02

I don't hold out too much hope for that, but good luck anyway. If you could get a good trace I can probably point you to the problem quick smart.

paul21019

19 posts

Geek

#858113 17-Jul-2013 16:07

I am happy to try ...another trace, when you say a good trace, just the same as I did before with maybe more calls - would that do it?

On the subject of the Italk2 trace, I made several calls and then went to the wireshark voip decoder, it clearly showed the "rejected call" and the good calls - not sure what I am trying to add here, but this is bizarre.. something has changed but what.. I wonder if any other Asterisk users are finding the same?

PDE

ubergeeknz

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#858115 17-Jul-2013 16:11

In that trace, your server is rejecting the call, not italk. It never got sent to italk. Look at the IP addresses.

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