Hello everybody, I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? Thanks. I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers The one that works: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer The one that doen't work: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces -- -- -- Marc LEURENT lftsy at leurent.eu
Klaus Darilion
2009-Oct-23 14:15 UTC
[asterisk-users] How to generate 183 Session Progress
If the outgoing channel receives progress indication from the far end (e.g. ISDN PROGRESS message or 183 response from an ITSP) then Asterisk will relay the progress message. If there is no progress indication received - that means that early media is not available - Asterisk does not send 183 as this would make the client listing for early media although it is not available. regards klaus Marc Leurent schrieb:> Hello everybody, > I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. > > For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? > Thanks. > > I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers > > The one that works: > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > > The one that doen't work: > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Supported: replaces >
You can call application Progress() from within dialplan and it will cause the Asterisk to send a SIP reply 183 on the call that came to Asterisk. Martin On Fri, Oct 23, 2009 at 6:36 AM, Marc Leurent <lftsy at leurent.eu> wrote:> Hello everybody, > I have 2 users connected on the same Asterisk server that are connected with 2 different Asterisk servers. > > For outgoing calls, one in receiving 183 Session Progress and the other not! Do you have any idea why? > Thanks. > > I have tried to understand by myself and in their INVITE they have almost the same Allow and Supported SIP Headers > > The one that works: > ? ? ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > ? ? ? ?Supported: replaces, timer > > The one that doen't work: > ? ? ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > ? ? ? ?Supported: replaces > > -- > -- -- > Marc LEURENT > lftsy at leurent.eu > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > ? http://lists.digium.com/mailman/listinfo/asterisk-users >
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