Re: [asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected
On 04/06/2017 at 08:33 PM Joshua Colp wrote: > On Thu, Apr 6, 2017, at 03:15 PM, Michael Maier wrote: >> Hello! >> >> I'm trying to send a fax via T.38 to a destination, which should be T.38 >> capable. My provider supports T.38, too. Unfortunately, it doesn't work. >> This means: >> >> Call is started and SDP is negotiated w/ alaw. Callee sends reinvite - >> for alaw again (and not for T.38)!! After about 30s, callee hangs up >> because of missing data (this is true, because I don't send alaw coded >> fax data. >> >> Tracing the signaling shows, that the callee doesn't have any >> possibility to recognize, if I'm supporting T.38, because it is never >> sent during Invite process. >> >> I'm missing the media feature tag sip.fax in the contact header. Did I >> miss some configuration? > > This is not currently supported in either chan_sip or chan_pjsip. > There's no configuration which will enable it. It would need to be > written. Have you confirmed this is what is needed by them? It turned out, that there is a third provider in between, which doesn't support T.38 ... . Thanks, Michael -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected
On 04/06/2017 at 08:33 PM, Joshua Colp wrote: > On Thu, Apr 6, 2017, at 03:15 PM, Michael Maier wrote: >> Hello! >> >> I'm trying to send a fax via T.38 to a destination, which should be T.38 >> capable. My provider supports T.38, too. Unfortunately, it doesn't work. >> This means: >> >> Call is started and SDP is negotiated w/ alaw. Callee sends reinvite - >> for alaw again (and not for T.38)!! After about 30s, callee hangs up >> because of missing data (this is true, because I don't send alaw coded >> fax data. >> >> Tracing the signaling shows, that the callee doesn't have any >> possibility to recognize, if I'm supporting T.38, because it is never >> sent during Invite process. >> >> I'm missing the media feature tag sip.fax in the contact header. Did I >> miss some configuration? > > This is not currently supported in either chan_sip or chan_pjsip. > There's no configuration which will enable it. It would need to be > written. Have you confirmed this is what is needed by them? No - I have to confirm it. But this may take some time :-). Thanks, Michaal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected
On Thu, Apr 6, 2017, at 03:15 PM, Michael Maier wrote: > Hello! > > I'm trying to send a fax via T.38 to a destination, which should be T.38 > capable. My provider supports T.38, too. Unfortunately, it doesn't work. > This means: > > Call is started and SDP is negotiated w/ alaw. Callee sends reinvite - > for alaw again (and not for T.38)!! After about 30s, callee hangs up > because of missing data (this is true, because I don't send alaw coded > fax data. > > Tracing the signaling shows, that the callee doesn't have any > possibility to recognize, if I'm supporting T.38, because it is never > sent during Invite process. > > I'm missing the media feature tag sip.fax in the contact header. Did I > miss some configuration? This is not currently supported in either chan_sip or chan_pjsip. There's no configuration which will enable it. It would need to be written. Have you confirmed this is what is needed by them? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
