[Asterisk-Dev] SIP Extrange Problem

Fri Feb 27 02:09:41 MST 2004

I try sip debug, but nothing happen. 
	Extension 705 is behind a NAT. Extension 707 is in Intranet.
Problem is always with 707 Extension.

I attach /var/log/asterisk/debug and messages

Feb 27 09:46:23 DEBUG[6151]: Call from user '705' is 1 out of 0
Feb 27 09:46:23 DEBUG[6151]: build_route: Contact hop:
<sip:705 at 192.168.1.160>
Feb 27 09:46:23 DEBUG[24595]: SIMPLE DIAL (NO URL)
Feb 27 09:46:23 DEBUG[24595]: Setting NAT on RTP to 0
Feb 27 09:46:23 DEBUG[24595]: Outgoing Call for 707
Feb 27 09:46:23 DEBUG[24595]: Call from user '707' is 1 out of 0
Feb 27 09:46:23 DEBUG[24595]: Prodding channel 'SIP/705-083d'
Feb 27 09:46:23 DEBUG[6151]: (Provisional) Stopping retransmission (but
retaining packet) on '186e62c321bde78f11ee6f385778061f at 192.168.0.207'
Request 102: Found
Feb 27 09:46:23 DEBUG[6151]: (Provisional) Stopping retransmission (but
retaining packet) on '186e62c321bde78f11ee6f385778061f at 192.168.0.207'
Request 102: Found
Feb 27 09:46:23 DEBUG[24595]: RTP NAT: Using address 82.223.7.185:5004
Feb 27 09:46:26 DEBUG[6151]: Acked pending invite 102
Feb 27 09:46:26 DEBUG[6151]: Stopping retransmission on
'186e62c321bde78f11ee6f385778061f at 192.168.0.207' of Request 102: Found
Feb 27 09:46:26 DEBUG[6151]: build_route: Contact hop:
<sip:707 at 192.168.0.157:52478>
Feb 27 09:46:26 DEBUG[6151]: Stopping retransmission on
'39d77c53f34ee760 at 192.168.1.160' of Response 28042: Found
Feb 27 09:46:26 DEBUG[24595]: Ooh, format changed from UNKN to G729A
Feb 27 09:46:26 DEBUG[24595]: Ooh, format changed from UNKN to G729A
Feb 27 09:46:50 DEBUG[24595]: Didn't get a frame from channel:
SIP/705-083d
Feb 27 09:46:50 DEBUG[24595]: Bridge stops bridging channels
SIP/705-083d and SIP/707-ba16
Feb 27 09:46:50 DEBUG[24595]: find_user(707) - decrement outUse counter
Feb 27 09:46:50 DEBUG[24595]: cdr_mysql: inserting a CDR record.
Feb 27 09:46:50 DEBUG[24595]: cdr_mysql: SQL command as follows:  INSERT
INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode) VALUES ('2004-02-27
09:46:50','\"705\" <705>','705','707','voip-h323',
'SIP/705-083d','SIP/707-ba16','Dial','SIP/707|30|tTrm',27,24,'ANSWERED',
3,'')
Feb 27 09:46:50 DEBUG[24595]: find_user(705) - decrement inUse counter

Any idea?
srsergio

-----Mensaje original-----
De: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] En nombre de Conroy,
Lawrence (SMTP)
Enviado el: viernes, 27 de febrero de 2004 1:02
Para: asterisk-dev at lists.digium.com
Asunto: Re: [Asterisk-Dev] SIP Extrange Problem

Hi Sergio, Folks,
  "after a few seconds" is interesting.
I wonder if this is a re-INVITE issue?

First, switch on "sip debug" in the asterisk console, and
see if there are sip messages approximately when the audio dies.

One quick way to *TEST* this is to:
*   set the phones in sip.conf to canreinvite=no,
AND
*   edit rtp.c to always return -2, (a couple of lines before the 
existing conditional -2 return).

That way, * should always stay in the call and should NOT try to drop 
out by sending a re-INVITE to the
phones after the initial call setup.
Then retry the calls to see if the problem has "gone away".

If so, then the re-INVITE issue has bitten you. It can cause strange 
effects just like you're seeing (especially with some kinds of NAT/ALG 
between the phones). The problem is in the NAT/ALG, but it's triggered
by the re-INVITE behaviour of *.

all the best,
   Lawrence

On 26 Feb 2004, at 10:07 am, Sergio Serrano Revuelto wrote:

> Hi all,
>     For a few days we have a very extrange problem. We have an
> intranet with Budgetone and others SIP Phones.
>  In the extranet We Have Budgetone Phones. The whole system was 
> working well between the extranet and the intranet until a few days 
> ago.
>  When we try to speak with a Budgetone of the intranet, we can speak 
> during a few seconds but after a time the audio is cut in the sense of

> intranet-extranet.
>  The problem is not only it, but if a budgetone of the intranet speaks

> with another phone of the intranet the same thing happens.
>  After a time of conversation the audio is cut in the sense of the 
> budgetone to another phone. I see the next meesage in debug:
>  
> Feb 26 10:50:04 DEBUG[50193]: Didn't get a frame from channel:
> SIP/707-996a
> I have checked the files of configuration. It does not appear at all 
> any more in the files of logs and I do not know that to do.
>  Can it be a problem of the internal network? of the switches? Is 
> there any bug in the budgetones?
>
>  
> Any idea?
>
>  
>  
> Thanks,
>  
> srsergio
>  

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