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Summary:ASTERISK-01049: Dial with SIP and IAX target not indicating ring to caller
Reporter:vaewyn (vaewyn)Labels:
Date Opened:2004-02-18 13:56:18.000-0600Date Closed:2011-06-07 14:10:12
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Core/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:Dial(SIP/6103&IAX2/user:pass@143.207.x.y/s@bell,15)
Does not indicate ringing to caller even though both:
Dial(SIP/6103,15)
and
Dial(IAX2/user:pass@143.207.x.y/s@bell,15)
indicate ringing when they are used by themselves and can be answered and handle correctly.

****** ADDITIONAL INFORMATION ******

Local setup:
Linux eventhorizon 2.6.2 #1 SMP Wed Feb 4 08:58:59 EST 2004 i686 GNU/Linux
Asterisk CVS-02/18/04-10:01:38 built by root@eventhorizon on a i686 running Linux

Remote setup:
Linux londo 2.6.2 #1 Wed Feb 4 08:58:59 EST 2004 i686 GNU/Linux
Asterisk CVS-02/18/04-10:01:38 built by root@londo on a i686 running Linux


Transcript of call on local machine:
-- Called 6103
-- Called user:pass@143.207.x.y/s@bell
-- SIP/6103-5850 is ringing
-- Call accepted by 143.207.x.y (format GSM)
-- Format for call is GSM
-- IAX2[blah]/7 stopped sounds
-- IAX2[blah]/7 is ringing
-- Nobody picked up in 15000 ms
-- Hungup 'IAX2[blah]/7'


Transcript on remote machine:
-- Accepting AUTHENTICATED call from 143.207.z.w, requested format = 2, actual format = 2
-- Executing Dial("IAX2[user@bridge]/7", "Zap/1") in new stack
-- Called 1
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Hungup 'Zap/1-1'
== Spawn extension (bell, s, 1) exited non-zero on 'IAX2[user@bridge]/7'
-- Hungup 'IAX2[user@bridge]/7'

Call is entering through a FXO interface... remote dials Zap/1 and SIP/6103 is a barbietone
Comments:By: vaewyn (vaewyn) 2004-02-19 21:13:36.000-0600

Additional information:

When a ,r is placed on the dial it speradically works.

When a ,m is placed on the dial I always (as far as I have tried) get music.

By: scaredycat (scaredycat) 2004-02-20 03:57:04.000-0600

We have a similar problem but going the other way:

Sip phone -> *1 -> *2 -> PSTN

r gives ring, but it's useless in this situation since if the number is busy the phone rings a few times then gives busy tone confusing the user. Without the r the user gets no ring tone at all :/

SC

By: Brian West (bkw918) 2004-02-20 14:07:06.000-0600

Try using chan_local to dial all your targets and see if you get ringing

Local/exten@context

bkw

By: vaewyn (vaewyn) 2004-02-22 15:45:27.000-0600

Using chan_local still does not result in caller getting a ring indication.

By: Olle Johansson (oej) 2004-03-21 16:59:17.000-0600

This is a known problem that needs a fix. Any coder?

By: cmslaght (cmslaght) 2004-03-21 23:05:37.000-0600

I am not sure if this problem is the same as what I am seeing but I have seen caller ID pop up on my phone but my phone does not ring.  For sure I have seen this when a SIP device calls my IAX.

By: hmodes (hmodes) 2004-03-23 00:04:43.000-0600

does 'known problem' == 180/183 sip message confusion?  mark suggested this might be fixed in -head.  can someone with the issue test or provide a sip debug to confirm?  i've been able to isolate this in the past by copying case 183 to case 180 in chan_sip or vice versa, but with less then desireable results for endpoints that don't experience the problem.

By: vaewyn (vaewyn) 2004-03-23 08:30:49.000-0600

W000! Kram rocks

Havn't been able to reproduce the bug at all with CVS as of this morning.

I consider issue resolved. :}

By: Olle Johansson (oej) 2004-03-23 08:50:14.000-0600

Bug closed. Contact a bug marshal if you still have problems, then we'll reopen.