asterisk users - Sep 2006 - One way audio problem on gateway to PSTN after some time, no NAT involved

Hello everyone,

since some weeks I experience strange problems on my gateways to the
PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that

SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN

What happens is, that after a while (uptime was a least two days) the
gateway starts to not transmit audio to the PSTN on outgoing calls, but
the caller can still hear the called party. There is no NAT involved and
firewall rules allow the RTP ports defined in rtp.conf on both asterisk
(A and B) machines. The SIP packages look good, no errors messages from
asterisk or anything else, so I have really no idea what causes it and I
cannot reproduce it except by waiting till it happens again. :(

Now the strange thing is, that if I restart the asterisk all works fine
again. A reload does not help, only a restart. Until now I came across
this phenomenon two times on different machines and it all started about
three weeks ago. Before that I ran asterisk 1.2.10 on the machines and
then updated to 1.2.11. I looked through the Changelog but coulnd't find
anything that seems related, but I guess it's a bug that was introduced
somewhere between 1.2.10 and 1.2.11 ...

Does anyone else have similar problems?

Regards,
Kai

-- 
Kai Militzer                 WESTEND GmbH  |  Internet-Business-Provider
Technik                      CISCO Systems Partner - Authorized Reseller
                             L?tticher Stra?e 10      Tel 0241/701333-14
km@westend.com               D-52064 Aachen              Fax 0241/911879

Hi Kai,
we had a similar problem with a PBX which had PSTN lines and SIP phones: 
sometimes some phones had one way calls...the caller couldn't hear. We 
hadn't tried to restart but we reduced the number of RTP ports (rtp.conf 
if memory helps!) to a range of 200 (it depends from the number of 
simultaneous calls you have).
That seemed to work!
Hope it may help!


Giorgio Incantalupo




Kai Militzer wrote:
> Hello everyone,
>
> since some weeks I experience strange problems on my gateways to the
> PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that
>
> SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN
>
> What happens is, that after a while (uptime was a least two days) the
> gateway starts to not transmit audio to the PSTN on outgoing calls, but
> the caller can still hear the called party. There is no NAT involved and
> firewall rules allow the RTP ports defined in rtp.conf on both asterisk
> (A and B) machines. The SIP packages look good, no errors messages from
> asterisk or anything else, so I have really no idea what causes it and I
> cannot reproduce it except by waiting till it happens again. :(
>
> Now the strange thing is, that if I restart the asterisk all works fine
> again. A reload does not help, only a restart. Until now I came across
> this phenomenon two times on different machines and it all started about
> three weeks ago. Before that I ran asterisk 1.2.10 on the machines and
> then updated to 1.2.11. I looked through the Changelog but coulnd't
find
> anything that seems related, but I guess it's a bug that was introduced
> somewhere between 1.2.10 and 1.2.11 ...
>
> Does anyone else have similar problems?
>
> Regards,
> Kai
>
>

Has anyone used the Polycom expansion module with multiple lines?

My application is for 20 lines and read there was a limit of 7 at one point.

Thanks

Did you solved this Problem?

I have the same problem, and i can't solve it, did you know anything 
about?

Thanks

Nico


On Thu, 14 Sep 2006, Kai Militzer wrote:
> Hello everyone,
>
> since some weeks I experience strange problems on my gateways to the
> PSTN. The gateways use chan_ss7 and SIP. My setup is roughly like that
>
> SER --> Asterisk A --> Asterisk B (chan_ss7) --> PSTN
>
> What happens is, that after a while (uptime was a least two days) the
> gateway starts to not transmit audio to the PSTN on outgoing calls, but
> the caller can still hear the called party. There is no NAT involved and
> firewall rules allow the RTP ports defined in rtp.conf on both asterisk
> (A and B) machines. The SIP packages look good, no errors messages from
> asterisk or anything else, so I have really no idea what causes it and I
> cannot reproduce it except by waiting till it happens again. :(
>
> Now the strange thing is, that if I restart the asterisk all works fine
> again. A reload does not help, only a restart. Until now I came across
> this phenomenon two times on different machines and it all started about
> three weeks ago. Before that I ran asterisk 1.2.10 on the machines and
> then updated to 1.2.11. I looked through the Changelog but coulnd't
find
> anything that seems related, but I guess it's a bug that was introduced
> somewhere between 1.2.10 and 1.2.11 ...
>
> Does anyone else have similar problems?
>
> Regards,
> Kai
>
> -- 
> Kai Militzer                 WESTEND GmbH  |  Internet-Business-Provider
> Technik                      CISCO Systems Partner - Authorized Reseller
>                             L?tticher Stra?e 10      Tel 0241/701333-14
> km@westend.com               D-52064 Aachen              Fax 0241/911879
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>