Gareth Blades
2005-Aug-10 02:20 UTC
[Asterisk-Users] No audio when calling between internal phones
I am running the latest CVS version of Asterisk. Calls between an IAX client and SIP phones (Grandstream SP2000 and Sipura SPA-841) works fine and so do external call over the Internet from the SIP desk phones. However when I call from either the Grandstream/Sipura phones to another one I get no audio. I have the G711 ulaw codec defined as the preferred on on all phones. Any idea what is going wrong? I am guessing it is something to do with native transfers which is performed in this situation.
Tom Hayden
2005-Aug-10 06:14 UTC
[Asterisk-Users] No audio when calling between internal phones
I encountered a similar problem with CVS-HEAD and sip2sip calls between our Polycom IP500s. I attempted to diagnose the problem and there are a few patches on mantis, but none of them worked for me. I flipped back to stable and have had no problems since. Anyone got any ideas? -- Tom On 8/10/05, Gareth Blades <list-asterisk@linguaphone.co.uk> wrote:> I am running the latest CVS version of Asterisk. > Calls between an IAX client and SIP phones (Grandstream SP2000 and > Sipura SPA-841) works fine and so do external call over the Internet > from the SIP desk phones. > > However when I call from either the Grandstream/Sipura phones to another > one I get no audio. I have the G711 ulaw codec defined as the preferred > on on all phones. > > Any idea what is going wrong? > I am guessing it is something to do with native transfers which is > performed in this situation. > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Tom
Gareth Blades
2005-Aug-10 06:28 UTC
[Asterisk-Users] No audio when calling between internal phones
I did try installing the 1.0.9 version but I have the same problem with that release aswell. On Wed, 2005-08-10 at 14:14, Tom Hayden wrote:> I encountered a similar problem with CVS-HEAD and sip2sip calls > between our Polycom IP500s. I attempted to diagnose the problem and > there are a few patches on mantis, but none of them worked for me. I > flipped back to stable and have had no problems since. > > Anyone got any ideas? > > -- > Tom > > On 8/10/05, Gareth Blades <list-asterisk@linguaphone.co.uk> wrote: > > I am running the latest CVS version of Asterisk. > > Calls between an IAX client and SIP phones (Grandstream SP2000 and > > Sipura SPA-841) works fine and so do external call over the Internet > > from the SIP desk phones. > > > > However when I call from either the Grandstream/Sipura phones to another > > one I get no audio. I have the G711 ulaw codec defined as the preferred > > on on all phones. > > > > Any idea what is going wrong? > > I am guessing it is something to do with native transfers which is > > performed in this situation. > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Tom Hayden
2005-Aug-10 07:08 UTC
[Asterisk-Users] No audio when calling between internal phones
Then perhaps you have a NAT problem or some other issue. -- Tom On 8/10/05, Gareth Blades <list-asterisk@linguaphone.co.uk> wrote:> I did try installing the 1.0.9 version but I have the same problem with > that release aswell. > > On Wed, 2005-08-10 at 14:14, Tom Hayden wrote: > > I encountered a similar problem with CVS-HEAD and sip2sip calls > > between our Polycom IP500s. I attempted to diagnose the problem and > > there are a few patches on mantis, but none of them worked for me. I > > flipped back to stable and have had no problems since. > > > > Anyone got any ideas? > > > > -- > > Tom > > > > On 8/10/05, Gareth Blades <list-asterisk@linguaphone.co.uk> wrote: > > > I am running the latest CVS version of Asterisk. > > > Calls between an IAX client and SIP phones (Grandstream SP2000 and > > > Sipura SPA-841) works fine and so do external call over the Internet > > > from the SIP desk phones. > > > > > > However when I call from either the Grandstream/Sipura phones to another > > > one I get no audio. I have the G711 ulaw codec defined as the preferred > > > on on all phones. > > > > > > Any idea what is going wrong? > > > I am guessing it is something to do with native transfers which is > > > performed in this situation. > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > Asterisk-Users@lists.digium.com > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Tom
Gareth Blades
2005-Aug-10 07:16 UTC
[Asterisk-Users] No audio when calling between internal phones
Both phones are on our internal network so not a NAT issue. However your email did prompt me to check iptables as I have rebooted the machine since it last worked. Dropping the firewall has fixed the fault so it looks like I will need to have a look at the ruleset. Thanks On Wed, 2005-08-10 at 15:08, Tom Hayden wrote:> Then perhaps you have a NAT problem or some other issue. > > -- > Tom > > On 8/10/05, Gareth Blades <list-asterisk@linguaphone.co.uk> wrote: > > I did try installing the 1.0.9 version but I have the same problem with > > that release aswell. > > > > On Wed, 2005-08-10 at 14:14, Tom Hayden wrote: > > > I encountered a similar problem with CVS-HEAD and sip2sip calls > > > between our Polycom IP500s. I attempted to diagnose the problem and > > > there are a few patches on mantis, but none of them worked for me. I > > > flipped back to stable and have had no problems since. > > > > > > Anyone got any ideas? > > > > > > -- > > > Tom > > > > > > On 8/10/05, Gareth Blades <list-asterisk@linguaphone.co.uk> wrote: > > > > I am running the latest CVS version of Asterisk. > > > > Calls between an IAX client and SIP phones (Grandstream SP2000 and > > > > Sipura SPA-841) works fine and so do external call over the Internet > > > > from the SIP desk phones. > > > > > > > > However when I call from either the Grandstream/Sipura phones to another > > > > one I get no audio. I have the G711 ulaw codec defined as the preferred > > > > on on all phones. > > > > > > > > Any idea what is going wrong? > > > > I am guessing it is something to do with native transfers which is > > > > performed in this situation. > > > > > > > > _______________________________________________ > > > > Asterisk-Users mailing list > > > > Asterisk-Users@lists.digium.com > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > To UNSUBSCRIBE or update options visit: > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > >