Re: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.

2009-09-17 Thread Yehavi Bourvine 
I've solved the problem: I am running it on a Fedora-10 system. Once I've
installed a vanilla kernel (from kernel.org) the problem went away.

BTW, can someone shed the light on the kernel's bug which I see mentions
of it in this list?

 Thanks! __Yehavi:
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Re: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.

2009-09-15 Thread Yehavi Bourvine 
Hello Jason,

  Sorry for the delay in answering - I saw your reply only now as it got
burried with some other stuff...

  Anyway, I attach bellow the relevant sip trace. Phone 80678 (132.64.4.137)
is calling 80679 (132.64.4.135) which answers. When 80679 presses the Hold
or Transfer button the call is disconnected.

   Thanks! __Yehavi:

2009/9/8 Jason White ja...@jasonjgw.net

 Yehavi Bourvine yehavi.bourv...@gmail.com wrote:
 
I have a problem when trying to put a call on hold: I get the above
  message and  the call is disconnected. Any idea where to look for the
 source
  of the problem?

 My next step in your situation would be to obtain a Sip trace and post
 relevant details from it to the list.


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=== HERE IS THE INITIAL INVITE =

   
recv 1407 bytes from udp/[132.64.4.137]:2048 at 06:31:26.925580:
   
   INVITE sip:80...@pbx-dev.cc.huji.ac.il;user=phone SIP/2.0
   Via: SIP/2.0/UDP 132.64.4.137:2048;branch=z9hG4bK-j6ogg8cwv5nm;rport
   From: Test Yehavi SNOM sip:80...@pbx-dev.cc.huji.ac.il;tag=j1fjgvgf7e
   To: sip:80...@pbx-dev.cc.huji.ac.il;user=phone
   Call-ID: 3c2696db6dfb-z5x2h00d9zcw
   CSeq: 2 INVITE
   Max-Forwards: 70
   Contact: sip:80...@132.64.4.137:2048;reg-id=1
   P-Key-Flags: keys=3
   User-Agent: snom320/7.3.14
   Accept: application/sdp
   Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, 
MESSAGE, INFO
   Allow-Events: talk, hold, refer, call-info
   Supported: timer, 100rel, replaces, from-change, Remote-Aprty-ID
   Session-Expires: 3600;refresher=uas
   Min-SE: 90
   Proxy-Authorization: Digest 
username=80678,realm=pbx-dev.cc.huji.ac.il,nonce=27cd67de-dfbf-4a05-8e19-edfc00d159b5,uri=sip:80...@pbx-dev.cc.huji.ac.il;user=phone,qop=auth,nc=0001,cnonce=044d5d78,response=a29e4873f5e72ebbd5e526cc45e1de0d,algorithm=MD5
   Content-Type: application/sdp
   Content-Length: 388
   
   v=0
   o=root 1073374100 1073374100 IN IP4 132.64.4.137
   s=call
   c=IN IP4 132.64.4.137
   t=0 0
   m=audio 60606 RTP/AVP 8 0 9 99 3 18 4 101
   a=direction:both
   a=rtpmap:8 pcma/8000
   a=rtpmap:0 pcmu/8000
   a=rtpmap:9 g722/8000
   a=rtpmap:99 g726-32/8000
   a=rtpmap:3 gsm/8000
   a=rtpmap:18 g729/8000
   a=rtpmap:4 g723/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   a=sendrecv
   
send 342 bytes to udp/[132.64.4.137]:2048 at 06:31:26.937621:
   
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 132.64.4.137:2048;branch=z9hG4bK-j6ogg8cwv5nm;rport=2048
   From: Test Yehavi SNOM sip:80...@pbx-dev.cc.huji.ac.il;tag=j1fjgvgf7e
   To: sip:80...@pbx-dev.cc.huji.ac.il;user=phone
   Call-ID: 3c2696db6dfb-z5x2h00d9zcw
   CSeq: 2 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
   Content-Length: 0
   
   
** invite **80678 80679
2009-09-15 09:31:27.214557 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/80...@pbx-dev.cc.huji.ac.il 
[98e4ed2c-fb37-4a19-9fa7-268bb04413f8]
2009-09-15 09:31:27.275406 [INFO] mod_dialplan_xml.c:315 Processing Test 
Yehavi SNOM-80679 in context huji
--
send 1233 bytes to udp/[132.64.4.135]:5060 at 06:31:29.313289:
   
   INVITE sip:80...@132.64.4.135 SIP/2.0
   Via: SIP/2.0/UDP 132.64.9.164;rport;branch=z9hG4bKFrN8j6K4Ncjea
   Max-Forwards: 69
   From: n8 l8 sip:80...@132.64.9.164;tag=9aSUyZB0m7y8N
   To: sip:80...@132.64.4.135
   Call-ID: 3e2a8eb9-1c64-122d-4aa2-0002b35fc481
   CSeq: 120379808 INVITE
   Contact: sip:mod_so...@132.64.9.164:5060
   User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, 
REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, 
include-session-description, presence.winfo, message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 376
   P-Key-Flags: keys=3
   Remote-Party-ID: n8 l8 
sip:80...@132.64.9.164;party=calling;screen=yes;privacy=off
   
   v=0
   o=root 1073374100 1073374100 IN IP4 132.64.4.137
   s=call
   c=IN IP4 132.64.4.137
   t=0 0
   m=audio 60606 RTP/AVP 8 0 9 99 3 18 4 101
   a=rtpmap:8 pcma/8000
   a=rtpmap:0 pcmu/8000
   a=rtpmap:9 g722/8000
   a=rtpmap:99 g726-32/8000

Re: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.

2009-09-15 Thread Brian West 
Do you have Late Negotiation on?  Also is this the only FreeSWITCH log  
output you have in this transfer?

/b

On Sep 15, 2009, at 1:55 AM, Yehavi Bourvine wrote:

 Hello Jason,

   Sorry for the delay in answering - I saw your reply only now as it  
 got burried with some other stuff...

   Anyway, I attach bellow the relevant sip trace. Phone 80678  
 (132.64.4.137) is calling 80679 (132.64.4.135) which answers. When  
 80679 presses the Hold or Transfer button the call is disconnected.

Thanks! __Yehavi:


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Re: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.

2009-09-15 Thread Yehavi Bourvine 
No, I have late negotiation commented out.

This is the only log from the beginning of the session until it disconnects.
Shall I turn on more debugging (if available)?

  Thanks, __Yehavi:

2009/9/15 Brian West br...@freeswitch.org

 Do you have Late Negotiation on?  Also is this the only FreeSWITCH log
 output you have in this transfer?






 On Sep 15, 2009, at 1:55 AM, Yehavi Bourvine wrote:

  Hello Jason,
 
Sorry for the delay in answering - I saw your reply only now as it
  got burried with some other stuff...
 
Anyway, I attach bellow the relevant sip trace. Phone 80678
  (132.64.4.137) is calling 80679 (132.64.4.135) which answers. When
  80679 presses the Hold or Transfer button the call is disconnected.
 
 Thanks! __Yehavi:


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[Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.

2009-09-08 Thread Yehavi Bourvine 
Hello,

  I have a problem when trying to put a call on hold: I get the above
message and  the call is disconnected. Any idea where to look for the source
of the problem?

  One thing I've tried is limiting all phones to use only one codec, but it
doesn't help...

  Thanks! __Yehavi:
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Re: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec.

2009-09-08 Thread Jason White 
Yehavi Bourvine yehavi.bourv...@gmail.com wrote:
 
   I have a problem when trying to put a call on hold: I get the above
 message and  the call is disconnected. Any idea where to look for the source
 of the problem?

My next step in your situation would be to obtain a Sip trace and post
relevant details from it to the list.


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