Re: [asterisk-users] Asterisk and hylafax: how to debug ...
Am 08.05.2013 01:12, schrieb James Cloos: SN == Sebastian Niehaus nieh...@web.de writes: SN Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the SN same maschine: Hylafax fax server. I want hylafax to use t38modem (a SN virtual T.38 modem) for sending faxes. t38modem schould connect to SN asterisk on the same host. SN If hylafax sends a fax it should use t38modem which ist connected to SN asterisk. Asterik is expected to establish an outbound connection to my SN SIP provider which supports T38. The asterisk box is behind nat. Silly question: Not so silly ... If you want to use T38 to the remote provider, and have t38modem, do you /need/ the asterisk in the middle? The hylafax server is behind NAT and I did not succeed to get t38modem running behind NAT. So I wanted to give asterisk a try since it has some features to make it work behind a natted connection. And if you /do/ need something between the two, might a sip proxy work better than a pbx I am behind a IAD which does NAT. I cannot install my own software on the IAD. Therefore I did nit consider a SIP proxy. Thanks! Sebastian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and hylafax: how to debug ...
Hi, Am Dienstag, den 07.05.2013, 21:48 +0200 schrieb Sebastian Niehaus: Am 07.05.2013 18:23, schrieb Sebastian Niehaus: For some reason, t38modem tells hylafax the line is BUSY so there is no fax send. Well, I may add the log of t38modem (sorry for the ugly formatting) Parts I consider as most important are: ModemConnection::SetUpConnection dstNum=189659 srcNum=30 srcName=root ... denied (all modems busy) [ snip ] it seems to me, that the call is routed from the modem to the modem (and not to asterisk). t38modem has some config options for call routing. Something like: route=modem:.*=sip:dn@ip:port route=sip:.*=modem:dn The first rule routes calls from the modem to a sip destination. I think in Your setup it should be route=modem:.*=sip:dn@127.0.0.1:5060. (I never used localhost in a setup like this, it should work with the IP of Your ethernet too). HTH, Karsten -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and hylafax: how to debug ...
Am 07.05.2013 18:23, schrieb Sebastian Niehaus: For some reason, t38modem tells hylafax the line is BUSY so there is no fax send. Well, I may add the log of t38modem (sorry for the ugly formatting) Parts I consider as most important are: ModemConnection::SetUpConnection dstNum=189659 srcNum=30 srcName=root ... denied (all modems busy) SIP OnSetUp failed for INVITE from sip:06814003340@127.0.0.1:6060 for Call[Cf64671db22]-EPsip ModemConnection::OnReleased Call -EPmodem[modem:/C8670792521/0] (Call cleared because the line is out of service) Thanks for any suggestions here is all the log: --- 2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 -- AT+FCLASS=1 2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 -- { 0d 0a 4f 4b 0d 0a ..OK.. } 2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 -- ATDT0681/4003340 2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 ModemEndPoint::OnMyCallback command=dial extra=5 2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 PseudoModemQ::Dequeue ttyT38-0 2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 MyManager::OnMyCallback SetUpCall(modem:, 06814003340@+/dev/ttyT38-0) 2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 ModemEndPoint::MakeConnection modem: 2013/05/07 21:30:03.574 ttyT38-0(e...0xb1778700 ModemConnection::SetUpConnection Call[Cdd70c36e1]-EPmodem[modem:/Cdd70c36e1/0] Call[Cdd70c36e1] from modem:T38modem to 06814003340@+/dev/ttyT38-0, route to sip:06814003340@127.0.0.1:6060 2013/05/07 21:30:03.575 ttyT38-0(e...0xb1778700 Call[Cdd70c36e1] from modem:T38modem to 06814003340@+/dev/ttyT38-0, route to sip:06814003340@127.0.0.1:6060 2013/05/07 21:30:03.576 ttyT38-0(e...0xb1778700 ModemEndPoint::GetMediaFormats 2013/05/07 21:30:03.585 ttyT38-0(e...0xb1778700 ModemEndPoint::OnMyCallback request={ calltoken=modem:/Cdd70c36e1/0 localpartyname= command=dial response=confirm number=06814003340 modemtoken=ttyT38-0 } 2013/05/07 21:30:03.588 Pool:0xb1674700 ModemEndPoint::MakeConnection modem:06814003340 Call[C8e34455a2] from sip:06814003340@127.0.0.1:6060 to sip:06814003340@127.0.0.1:6060, route to modem:06814003340 2013/05/07 21:30:03.588 Pool:0xb1674700 Call[C8e34455a2] from sip:06814003340@127.0.0.1:6060 to sip:06814003340@127.0.0.1:6060, route to modem:06814003340 2013/05/07 21:30:03.588 Pool:0xb1674700 ModemEndPoint::GetMediaFormats 2013/05/07 21:30:03.593 Pool:0xb1674700 ModemConnection::SetUpConnection Call[C8e34455a2]-EPmodem[modem:06814003340/C8e34455a2/0] 2013/05/07 21:30:03.593 Pool:0xb1674700 ModemConnection::SetUpConnection dstNum=06814003340 srcNum=30 srcName=root ... 2013/05/07 21:30:03.593 Pool:0xb1674700 ... denied (all modems busy) 2013/05/07 21:30:03.593 Pool:0xb1674700 ModemConnection::OnReleased Call[C8e34455a2]-EPmodem[modem:06814003340/C8e34455a2/0] 2013/05/07 21:30:03.593 Pool:0xb1674700 SIP OnSetUp failed for INVITE from sip:06814003340@127.0.0.1:6060 for Call[C8e34455a2]-EPsip[b8e7402f-bab5-e211-972b-6c626db69c09] 2013/05/07 21:30:03.596OnRelease:0xb15b1700 ModemConnection::OnReleased Call[Cdd70c36e1]-EPmodem[modem:/Cdd70c36e1/0] Call[Cdd70c36e1] cleared (Call cleared because the line is out of service) 2013/05/07 21:30:03.596OnRelease:0xb15b1700 Call[Cdd70c36e1] cleared (EndedByOutOfService) Call[C8e34455a2] cleared (Local party cleared call) 2013/05/07 21:30:03.695OnRelease:0xb15f2700 Call[C8e34455a2] cleared (EndedByLocalUser) 2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700 ttyT38-0 AudioEngine Attach 2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700 ttyT38-0 AudioEngine OnAttach Attached 2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700 ttyT38-0 AudioEngine OnResetModemState 2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700 ttyT38-0 AudioEngine ChangeModemClass to mcFax 2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700 ttyT38-0 AudioEngine OnChangeModemClass to mcFax 2013/05/07 21:30:04.229 Opal Garbage:0xc71ba700 ModemEngineBody::_AttachEngine Attached mceAudio 2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700 ttyT38-0 AudioEngine Detach 2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700 ttyT38-0 AudioEngine OnChangeModemClass to mcUndefined 2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700 ttyT38-0 AudioEngine OnDetach Detached 2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700 ttyT38-0 AudioEngine OnResetModemState 2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700 ModemEngineBody::_DetachEngine Detached mceAudio 2013/05/07 21:30:04.229 ttyT38-0(e...0xb1778700 -- { 0d 0a 42 55 53 59 0d 0a..BUSY.. } -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and hylafax: how to debug ...
SN == Sebastian Niehaus nieh...@web.de writes: SN Running Asterisk (version: 1.8.13.1~dfsg-3) on Debian Wheezy. On the SN same maschine: Hylafax fax server. I want hylafax to use t38modem (a SN virtual T.38 modem) for sending faxes. t38modem schould connect to SN asterisk on the same host. SN If hylafax sends a fax it should use t38modem which ist connected to SN asterisk. Asterik is expected to establish an outbound connection to my SN SIP provider which supports T38. The asterisk box is behind nat. Silly question: If you want to use T38 to the remote provider, and have t38modem, do you /need/ the asterisk in the middle? And if you /do/ need something between the two, might a sip proxy work better than a pbx? -JimC -- James Cloos cl...@jhcloos.com OpenPGP: 1024D/ED7DAEA6 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users