opensips and asterisk retransmits

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opensips and asterisk retransmits

troxlinux
Hi list , I have some days fighting with asterisk and opensips to
solve this problem,  when I use asterisk to listen my voicemail and to
call to the pstn, asterisk shows me this error message:

WARNING[3196]: chan_sip.c:1976 retrans_pkt: Maximum retries exceeded
on transmission d5a57aa528f5c8f2@192.168.10.30 for seqno 45371
(Critical Response) -- See doc/sip-retransmit.txt.
[Apr 28 19:34:44] WARNING[3196]: chan_sip.c:1998 retrans_pkt: Hanging
up call d5a57aa528f5c8f2@192.168.10.30 - no reply to our critical
packet (see doc/sip-retransmit.txt).

I read the documentation in asterisk, and there are possibly several
factors for those that I could give this problem:

Firewall - (I Don`t have)
A badly configured SIP proxy - ( with the version 1.3.4 of openser I
work me well and I never had this problem )
A SIP middlebox (SBC) - (I Don`t have)


I use opensips with asteriks in the same server but in different port,
and I have asterisk set in mode comedia

any idea?

some person that has presented him previously this problem?

help!...


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Re: opensips and asterisk retransmits

Alex Balashov
You may wish to consider posting this to the SER-Asterisk-Interwork list.

troxlinux wrote:

> Hi list , I have some days fighting with asterisk and opensips to
> solve this problem,  when I use asterisk to listen my voicemail and to
> call to the pstn, asterisk shows me this error message:
>
> WARNING[3196]: chan_sip.c:1976 retrans_pkt: Maximum retries exceeded
> on transmission d5a57aa528f5c8f2@192.168.10.30 for seqno 45371
> (Critical Response) -- See doc/sip-retransmit.txt.
> [Apr 28 19:34:44] WARNING[3196]: chan_sip.c:1998 retrans_pkt: Hanging
> up call d5a57aa528f5c8f2@192.168.10.30 - no reply to our critical
> packet (see doc/sip-retransmit.txt).
>
> I read the documentation in asterisk, and there are possibly several
> factors for those that I could give this problem:
>
> Firewall - (I Don`t have)
> A badly configured SIP proxy - ( with the version 1.3.4 of openser I
> work me well and I never had this problem )
> A SIP middlebox (SBC) - (I Don`t have)
>
>
> I use opensips with asteriks in the same server but in different port,
> and I have asterisk set in mode comedia
>
> any idea?
>
> some person that has presented him previously this problem?
>
> help!...
>
>


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Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
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Re: opensips and asterisk retransmits

Bogdan-Andrei Iancu
In reply to this post by troxlinux
Hi,

Get a ngrep capture of the SIP traffic between * and OSIPS . Typically a
retransmission is triggered by a lack of response from the other party,
but to see what response is lacking, you need to see the ngrep capture
of the SIP traffic.

Regards,
Bogdan

troxlinux wrote:

> Hi list , I have some days fighting with asterisk and opensips to
> solve this problem,  when I use asterisk to listen my voicemail and to
> call to the pstn, asterisk shows me this error message:
>
> WARNING[3196]: chan_sip.c:1976 retrans_pkt: Maximum retries exceeded
> on transmission d5a57aa528f5c8f2@192.168.10.30 for seqno 45371
> (Critical Response) -- See doc/sip-retransmit.txt.
> [Apr 28 19:34:44] WARNING[3196]: chan_sip.c:1998 retrans_pkt: Hanging
> up call d5a57aa528f5c8f2@192.168.10.30 - no reply to our critical
> packet (see doc/sip-retransmit.txt).
>
> I read the documentation in asterisk, and there are possibly several
> factors for those that I could give this problem:
>
> Firewall - (I Don`t have)
> A badly configured SIP proxy - ( with the version 1.3.4 of openser I
> work me well and I never had this problem )
> A SIP middlebox (SBC) - (I Don`t have)
>
>
> I use opensips with asteriks in the same server but in different port,
> and I have asterisk set in mode comedia
>
> any idea?
>
> some person that has presented him previously this problem?
>
> help!...
>
>
>  


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Re: opensips and asterisk retransmits

troxlinux
In reply to this post by Alex Balashov
excuseme , I didn't remember that there was a list

2009/4/27 Alex Balashov <[hidden email]>:
> You may wish to consider posting this to the SER-Asterisk-Interwork list.
>

regardss

--
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Re: opensips and asterisk retransmits

troxlinux
In reply to this post by Bogdan-Andrei Iancu
it is strange, I thought that it was asterisk the problem, but I
upgrade to a version that I consider stable 1.4.24



2009/4/28 Bogdan-Andrei Iancu <[hidden email]>:

> Hi,
>
> Get a ngrep capture of the SIP traffic between * and OSIPS . Typically a
> retransmission is triggered by a lack of response from the other party, but
> to see what response is lacking, you need to see the ngrep capture of the
> SIP traffic.
>
> Regards,
> Bogdan
>
>

--
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http://gnuforever.homelinux.com

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Re: opensips and asterisk retransmits

Bogdan-Andrei Iancu
It seams you have an ACK routing problem. The caller (.30:5064)
correctly sends ACK with:
        ACK sip:*981@192.168.10.3:5070 SIP/2.0
        Route: <sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3>

but opensips (.3:5060),sends it out as:
       ACK sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3 SIP/2.0

this means that OSIPS tinks that 192.168.10.3:5070 (RURI of received
ACK) is a  local IP (either alias in cfg, either domain in domains
module) and does strict routing....

So, do you have the :5070 set as alias or domain in your opensips setup?

Regards,
Bogdan

troxlinux wrote:

> it is strange, I thought that it was asterisk the problem, but I
> upgrade to a version that I consider stable 1.4.24
>
>
>
> 2009/4/28 Bogdan-Andrei Iancu <[hidden email]>:
>  
>> Hi,
>>
>> Get a ngrep capture of the SIP traffic between * and OSIPS . Typically a
>> retransmission is triggered by a lack of response from the other party, but
>> to see what response is lacking, you need to see the ngrep capture of the
>> SIP traffic.
>>
>> Regards,
>> Bogdan
>>
>>
>>    
>
>
>  


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Re: opensips and asterisk retransmits

troxlinux
2009/4/28 Bogdan-Andrei Iancu <[hidden email]>:

> It seams you have an ACK routing problem. The caller (.30:5064) correctly
> sends ACK with:
>       ACK sip:*981@192.168.10.3:5070 SIP/2.0
>       Route: <sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3>
>
> but opensips (.3:5060),sends it out as:
>      ACK sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3 SIP/2.0
>
> this means that OSIPS tinks that 192.168.10.3:5070 (RURI of received ACK) is
> a  local IP (either alias in cfg, either domain in domains module) and does
> strict routing....
>
> So, do you have the :5070 set as alias or domain in your opensips setup?
>

Hi Bogdan , I don't have any alias en mi opensips.cfg , the only thing
that I have is that when they make calls to the pstn they leave to
that ip port

route[4] {
        rewritehostport("192.168.10.3:5070");
        route(1);
}

###  example my routes ###




append_hf("P-hint: inbound->inbound \r\n");
        if (uri=~"^sip:9[0-9]*@") {
        if (is_user_in("credentials", "local")){
        route(4);
        exit;


regards , and many thanks for you help ...


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Re: opensips and asterisk retransmits

Bogdan-Andrei Iancu
I see...Could you please get the opensips logs in full debug (debug=6)
for the ACK processing? I can take a look to see what exactly is going on.

Regards,
Bogdan

troxlinux wrote:

> 2009/4/28 Bogdan-Andrei Iancu <[hidden email]>:
>  
>> It seams you have an ACK routing problem. The caller (.30:5064) correctly
>> sends ACK with:
>>       ACK sip:*981@192.168.10.3:5070 SIP/2.0
>>       Route: <sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3>
>>
>> but opensips (.3:5060),sends it out as:
>>      ACK sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3 SIP/2.0
>>
>> this means that OSIPS tinks that 192.168.10.3:5070 (RURI of received ACK) is
>> a  local IP (either alias in cfg, either domain in domains module) and does
>> strict routing....
>>
>> So, do you have the :5070 set as alias or domain in your opensips setup?
>>
>>    
>
> Hi Bogdan , I don't have any alias en mi opensips.cfg , the only thing
> that I have is that when they make calls to the pstn they leave to
> that ip port
>
> route[4] {
>         rewritehostport("192.168.10.3:5070");
>         route(1);
> }
>
> ###  example my routes ###
>
>
>
>
> append_hf("P-hint: inbound->inbound \r\n");
>         if (uri=~"^sip:9[0-9]*@") {
>         if (is_user_in("credentials", "local")){
>         route(4);
>         exit;
>
>
> regards , and many thanks for you help ...
>
>
>  


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Re: opensips and asterisk retransmits

Iñaki Baz Castillo
In reply to this post by troxlinux
El Miércoles, 29 de Abril de 2009, troxlinux escribió:
> Hi Bogdan , I don't have any alias en mi opensips.cfg , the only thing
> that I have is that when they make calls to the pstn they leave to
> that ip port
>
> route[4] {
>         rewritehostport("192.168.10.3:5070");
>         route(1);
> }

You *should not* use the above route[4] for in-dialog ACK (after 200 OK), are
you using it for that?

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Re: opensips and asterisk retransmits

Iñaki Baz Castillo
El Miércoles, 29 de Abril de 2009, Iñaki Baz Castillo escribió:

> El Miércoles, 29 de Abril de 2009, troxlinux escribió:
> > Hi Bogdan , I don't have any alias en mi opensips.cfg , the only thing
> > that I have is that when they make calls to the pstn they leave to
> > that ip port
> >
> > route[4] {
> >         rewritehostport("192.168.10.3:5070");
> >         route(1);
> > }
>
> You *should not* use the above route[4] for in-dialog ACK (after 200 OK),
> are you using it for that?

Check this (as Bogdan said) and try first to understand which the problem is.
Later you could look for a solution:


# PHONE -> OPENSIPS
U +0.038456 192.168.10.30:5064 -> 192.168.10.3:5060
ACK sip:*981@192.168.10.3:5070 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.30:5064;branch=z9hG4bKb1e3a5d2b3f28c71
Route: <sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3>

# OPENSIPS -> OPENSIPS ?
U +0.000634 192.168.10.3:5060 -> 192.168.10.3:5060
ACK sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3 SIP/2.0
Record-Route: <sip:192.168.10.3;lr=on;ftag=d7f613072c3769a3>


This ACK is an in-dialog request (confirmation for 200 OK) so RURI
***shouldn't*** be changed by the proxy, but you do change it and this is your
error.

Most probably, as I said in othermail, you are modifying the RURI of the ACK
while what you should do is bypass it as an in-dialog request (lose_route()).


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Re: opensips and asterisk retransmits

troxlinux
In reply to this post by Bogdan-Andrei Iancu
Hi Bogdan , something stranger happens when I put the debug in 6 I
don't see that it shows me the opensips log

tail -f /var/log/openser.log

twoxserver /sbin/opensips[3744]: INFO:core:sig_usr: signal 15 received
twoxserver /sbin/opensips[3733]: INFO:core:sig_usr: signal 15 received
twoxserver /sbin/opensips[3719]: NOTICE:presence:destroy: destroy module ...

any idea?

2009/4/29 Bogdan-Andrei Iancu <[hidden email]>:
> I see...Could you please get the opensips logs in full debug (debug=6) for
> the ACK processing? I can take a look to see what exactly is going on.
>
> Regards,
> Bogdan
>


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