Geekzone: technology news, blogs, forums
Guest
Welcome Guest.
You haven't logged in yet. If you don't have an account you can register now.


19 posts

Geek


Topic # 125724 17-Jul-2013 13:04
Send private message

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

Filter this topic showing only the reply marked as answer View this topic in a long page with up to 500 replies per page Create new topic
 1 | 2 | 3 | 4 | 5
3343 posts

Uber Geek
+1 received by user: 1089

Trusted
Vocus

  Reply # 858021 17-Jul-2013 13:24
Send private message

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.



19 posts

Geek


  Reply # 858022 17-Jul-2013 13:25
Send private message

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

3343 posts

Uber Geek
+1 received by user: 1089

Trusted
Vocus

  Reply # 858024 17-Jul-2013 13:26
Send private message

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.




19 posts

Geek


  Reply # 858046 17-Jul-2013 13:49
Send private message

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


 

3343 posts

Uber Geek
+1 received by user: 1089

Trusted
Vocus

  Reply # 858053 17-Jul-2013 13:56
Send private message

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.



19 posts

Geek


  Reply # 858061 17-Jul-2013 14:09
Send private message

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>


3343 posts

Uber Geek
+1 received by user: 1089

Trusted
Vocus

  Reply # 858067 17-Jul-2013 14:24
Send private message

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.



19 posts

Geek


  Reply # 858088 17-Jul-2013 15:10
Send private message

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

3343 posts

Uber Geek
+1 received by user: 1089

Trusted
Vocus

  Reply # 858100 17-Jul-2013 15:35
Send private message

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?



19 posts

Geek


  Reply # 858105 17-Jul-2013 15:55
Send private message



"Did you enable OPTIONS ping since before?"

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

3343 posts

Uber Geek
+1 received by user: 1089

Trusted
Vocus

  Reply # 858106 17-Jul-2013 15:57
Send private message

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



19 posts

Geek


  Reply # 858108 17-Jul-2013 16:01
Send private message

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..



3343 posts

Uber Geek
+1 received by user: 1089

Trusted
Vocus

  Reply # 858110 17-Jul-2013 16:02
Send private message

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.



19 posts

Geek


  Reply # 858113 17-Jul-2013 16:07
Send private message

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 

3343 posts

Uber Geek
+1 received by user: 1089

Trusted
Vocus

  Reply # 858115 17-Jul-2013 16:11
Send private message

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

 1 | 2 | 3 | 4 | 5
Filter this topic showing only the reply marked as answer View this topic in a long page with up to 500 replies per page Create new topic

Twitter »

Follow us to receive Twitter updates when new discussions are posted in our forums:



Follow us to receive Twitter updates when news items and blogs are posted in our frontpage:



Follow us to receive Twitter updates when tech item prices are listed in our price comparison site:



Geekzone Live »

Try automatic live updates from Geekzone directly in your browser, without refreshing the page, with Geekzone Live now.



Are you subscribed to our RSS feed? You can download the latest headlines and summaries from our stories directly to your computer or smartphone by using a feed reader.

Alternatively, you can receive a daily email with Geekzone updates.