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
[asterisk-users] Outbound T.38 via RTP with pjsip does not work as expected
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? That's my setup: Hylafax sends fax to t38modem and t38modem is connected via SIP to asterisk as extension. The extension is bound to an outbound route, which uses the t.38 capable ISP. pjsip.endpoint.conf: [ISP] t38_udptl=yes t38_udptl_nat=no # there is no nat necessary t38_udptl_ec=fec [t38endpoint] t38_udptl=yes t38_udptl_ec=fec Any idea? 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
