asterisk users - Oct 2003 - SIP Calls Don't Properly Connect (Continue Ringing) After CVS Update

Hi,

I just updated my image from CVS, compiled and reinstalled it.  Now
whenever I make calls from my Grandstream phone to my X-Lite soft-phone,
the call does not complete correctly.

Scenario:

1.	I take the GS off hook and dial 1100 (the extension of the
x-lite phone).
2.	The x-lite phone rings properly.
3.	The user at the x-lite site answers the call.
4.	The GS phone continues to "ringback" and does not detect that
the call is complete.
5.	After about 10 seconds the GS plays busy and the x-lite detects
hangup.
6.    The x-lite goes back on hook.

This scenario was working properly (the call completed as expected)
prior to the CVS update.  Oddly, calls from x-lite to the GS complete
properly and without incident.  The big difference is that there is a
"precursor" script on the GS extension that answers and plays the
use's
name using the name file in the voicemail folder.  THEN it uses Dial to
send the call to the SIP device.

I swear there was a thread about this last week but I can't find it for
the life of me.  Perhaps it was in the error log at Digium.

Any thoughts?

Thanks - Steve

I just noticed that messages like this:

WARNING[1142106560]: File chan_sip.c, Line 451 (retrans_pkt): Maximum
retries exceeded on call
aa3556d0-2b4a-39fa-be42-cfe62377e2d7@192.168.1.111 for seqno 8338
(Response)

And this:

WARNING[1142106560]: File chan_sip.c, Line 451 (retrans_pkt): Maximum
retries exceeded on call
e846e829-69d5-9003-4e9d-532466a22506@192.168.1.111 for seqno 53696
(Response)

Show up each time I try to place a call from the GS to the x-Lite.
Never saw these before the update.

I hope this helps.  If anybody wants, I can pull a trace of the SIP
stuff as well.

Thanks,

Steve

-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Steven M.
Sokol
Sent: Tuesday, October 28, 2003 10:39 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SIP Calls Don't Properly Connect (Continue
Ringing) After CVS Update

Hi,

I just updated my image from CVS, compiled and reinstalled it.  Now
whenever I make calls from my Grandstream phone to my X-Lite soft-phone,
the call does not complete correctly.

Scenario:

1.	I take the GS off hook and dial 1100 (the extension of the
x-lite phone).
2.	The x-lite phone rings properly.
3.	The user at the x-lite site answers the call.
4.	The GS phone continues to "ringback" and does not detect that
the call is complete.
5.	After about 10 seconds the GS plays busy and the x-lite detects
hangup.
6.    The x-lite goes back on hook.

This scenario was working properly (the call completed as expected)
prior to the CVS update.  Oddly, calls from x-lite to the GS complete
properly and without incident.  The big difference is that there is a
"precursor" script on the GS extension that answers and plays the
use's
name using the name file in the voicemail folder.  THEN it uses Dial to
send the call to the SIP device.

I swear there was a thread about this last week but I can't find it for
the life of me.  Perhaps it was in the error log at Digium.

Any thoughts?

Thanks - Steve



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>Hi,
>
>I just updated my image from CVS, compiled and reinstalled it.  Now
>whenever I make calls from my Grandstream phone to my X-Lite soft-phone,
>the call does not complete correctly.
>
>Scenario:
>
>1.	I take the GS off hook and dial 1100 (the extension of the
>x-lite phone).
>2.	The x-lite phone rings properly.
>3.	The user at the x-lite site answers the call.
>4.	The GS phone continues to "ringback" and does not detect that
>the call is complete.
>5.	After about 10 seconds the GS plays busy and the x-lite detects
>hangup.
>6.    The x-lite goes back on hook.
>
>This scenario was working properly (the call completed as expected)
>prior to the CVS update.  Oddly, calls from x-lite to the GS complete
>properly and without incident.  The big difference is that there is a
>"precursor" script on the GS extension that answers and plays the
use's
>name using the name file in the voicemail folder.  THEN it uses Dial to
>send the call to the SIP device.
>
>I swear there was a thread about this last week but I can't find it for
>the life of me.  Perhaps it was in the error log at Digium.
>
>Any thoughts?
>
>Thanks - Steve
Try exhaustively checking  combinations of codec permissions to see 
if that makes a difference.

Set disallow=all allow=ulaw  in both the [general] section, and in 
each end device.  Change those settings around a bit, trying a call 
each time you change a setting.  Both the GS and the x-lite phone 
have some quirks with how they handle codecs, especially with *.

JT