[Asterisk-Users] FAX over SIP
Hello. Has someone been able to make work faxes over sip, i have one mp108 fxo and one mp108 fxs, my setup is : telco analog line -> mp108fxo -> Asterisk --> mp108fxs ---> fax machine 1) Asterisk detects the tone from the sending fax ( i am receiving ) but looks for extension 'ff' not 'fax', ( at least that's what * complaint, invalid extension ff in context ). 2) I add 'ff' extension and dial to SIP/faxdev, Asterisk routes the call to this extension but when the fax answers the call is dropped ( don't have here the SIP debug output ) but seems that when * tries to make the bridge the mp108 fxo sends a BYE. 3) I dialed in to * with a phone ( external line and internal extension ) and dial extension for the fax and i cann hear the fax, so the call is not dropped, the bridge is established successfully. 4) If I pickup the call on the fax machine ( it has a phone set ) and then pressed the 'start' button to start de fax receiver, then, the two faxes talked to each other and the fax is received well. Seems that the problem is only when the fax answer automatically ( could be the tones the receiving fax plays ? ), the same problem happens when i try to use hylafax to receive the fax. Any hints ? Thank's. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FAX over SIP
On Thu, 2003-09-04 at 01:18, Ing. Angel Gomez Garcia wrote: > Hello. > > Has someone been able to make work faxes over sip, i have one mp108 > fxo and one mp108 fxs, my setup is : > > telco analog line -> mp108fxo -> Asterisk --> mp108fxs > ---> fax machine > > 1) Asterisk detects the tone from the sending fax ( i am receiving ) but > looks for extension 'ff' not 'fax', ( at least that's what * complaint, > invalid extension ff in context ). > > 2) I add 'ff' extension and dial to SIP/faxdev, Asterisk routes the call > to this extension but when the fax answers the call is dropped ( don't > have here the SIP debug output ) but seems that when * tries to make the > bridge the mp108 fxo sends a BYE. > > 3) I dialed in to * with a phone ( external line and internal extension > ) and dial extension for the fax and i cann hear the fax, so the call is > not dropped, the bridge is established successfully. > > 4) If I pickup the call on the fax machine ( it has a phone set ) and > then pressed the 'start' button to start de fax receiver, then, the two > faxes talked to each other and the fax is received well. By listening to #4 it sounds like you answered your own question, yes it is possible. Now you need to find out why in #2 that it sends a bye. My guess is that you have a difference in the dial command options that keep asterisk listening to the line when dialing the extension that isn't there on the ff extension. This may have asterisk trying to issue a reinvite to connect the call legs together without asterisk in the middle. This is causing the BYE, and then everything fall apart. Maybe you need to make sure the canreinvite is turned off for this device in the sip.conf and try some more. > Seems that the problem is only when the fax answer automatically ( > could be the tones the receiving fax plays ? ), the same problem happens > when i try to use hylafax to receive the fax. > > Any hints ? > > Thank's. > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax over SIP alaw/ulaw
Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave -- Dave Weis "I believe there are more instances of the abridgment [EMAIL PROTECTED] of the freedom of the people by gradual and silent encroachments of those in power than by violent and sudden usurpations."- James Madison ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP alaw/ulaw
At 20:34 14-11-2003 -0600, you wrote: Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. I have gotten this to work, but it seems to be very dependant on your infrastructure. Fax over alaw or ulaw clear channels will work as long as the latency of the network is within acceptable range. Give it a try :-) Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP alaw/ulaw
Am Sam, 2003-11-15 um 11.22 schrieb Florian Overkamp: > At 20:34 14-11-2003 -0600, you wrote: > >Should I expect a standard fax machine connected to an ata-188 connected > >to an asterisk server, connected to a pri fed from a cisco 7206vxr to work > >correctly? It needs to have a standard fax machine, receiving and emailing > >it won't be acceptable. > > I have gotten this to work, but it seems to be very dependant on your > infrastructure. Fax over alaw or ulaw clear channels will work as long as > the latency of the network is within acceptable range. Give it a try :-) > > Florian On a LAN this should work without problems. I am even faxing over the wild internet with ping times between 100 and 120 ms. Both locations have an AVM Fritz card connected to an internal S0 bus of an isdn pbx and run chan_capi. best regards kapejod -- Klaus-Peter Junghanns CEO,CTO Junghanns.NET GmbH Breite Strasse 13 - 12167 Berlin - Germany fon:+49 30 79705392 fax:+49 30 79705391 iaxtel: 1-700-157-8753 email: [EMAIL PROTECTED] http://www.junghanns.net/asterisk ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP alaw/ulaw
You will need to check with Cisco to see if the ATA188 has the same issues with faxing as the ATA186. http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml Dave Weis wrote: Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP alaw/ulaw
At 21:20 16-11-2003 -0600, you wrote: You will need to check with Cisco to see if the ATA188 has the same issues with faxing as the ATA186. http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_field_notice09186a0080094af7.shtml Be advised this item regards the ATA186-L series, and the issues should be resolved with the I1 and I2 models. (the L series are End Of Life and have been since mid-2002). Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax over SIP Problems (sorry for this topic ...)
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway
is external). Whenever I start sending a Fax to a PSTN destination, the
Call gets answered and asterisk tries to build a native bridging:
-- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and
SIP/xxx-3ef8
Then the following messages start to appear (about 100 of them)
Nov 23 16:27:25 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP
codec 127 received
After that asterisk gets totaly confused:
Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too
short
Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too
short
Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too
short
Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too
short
Nov 23 16:27:35 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP
codec 1 received
Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein:
Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP
(15)?
Nov 23 16:27:35 NOTICE[1061908]: channel.c:1724 ast_set_read_format:
Unable to find a path from G723 to ALAW
Nov 23 16:27:35 NOTICE[1061908]: channel.c:1691 ast_set_write_format:
Unable to find a path from GSM to G723
Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein:
Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP
(20)?
Nov 23 16:27:35 WARNING[1061908]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/sip.westend.com-082fd1b8(8) to SIP/xxx-3ef8(1)
Nov 23 16:27:35 WARNING[1061908]: channel.c:2633 ast_channel_bridge:
Can't make SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 compatible
Nov 23 16:27:35 WARNING[1061908]: res_features.c:358 ast_bridge_call:
Bridge failed on channels SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8
== Spawn extension (macro-enumcall, s, 211) exited non-zero on
'SIP/sip.westend.com-082fd1b8' in macro 'enumcall'
== Spawn extension (xxx, 911879, 7) exited non-zero on
'SIP/sip.westend.com-082fd1b8'
-- Executing NoOp("SIP/sip.westend.com-082fd1b8", "") in new stack
> cdr_odbc: Query Successful!
Then the call gets hung up. I cannot explain why this happens. I would
have explanations for the fax-machines not able to synchronize or faxes
not being transmitted correctly, as the communication is SIP only, but
this seems a bit strange to me. I can't even tell, if my asterisk
produces these messages, or the other side (aka PSTN-Gateway).
If anyone can bring some light into this behavior I would be very
greatful.
Best regards
Kai
--
Kai Militzer WESTEND GmbH | Internet-Business-Provider
Technik CISCO Systems Partner - Authorized Reseller
Lütticher Straße 10 Tel 0241/701333-11
[EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...)
Are you trying to send fax over T.38?
As far I understand * don't support T.38 event when passing packets
trouth.
I'm interested in T.38 support too, so if anybody could explain why *
can't just pass theese packets (as i undrstand there is no need foe
recoding etc.) I would be very appreciative.
Are anybody currently working on T.38 support for * ?
I don't mean T.38 support on zap interfaces, just passing T.38 packets
trouth asterisk
--
Sincerely,
Elman Efendiyev
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai
Militzer
Sent: Tuesday, November 23, 2004 6:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Fax over SIP Problems (sorry for this topic
...)
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway
is external). Whenever I start sending a Fax to a PSTN destination, the
Call gets answered and asterisk tries to build a native bridging:
-- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and
SIP/xxx-3ef8
Then the following messages start to appear (about 100 of them)
Nov 23 16:27:25 NOTICE[1061908]: rtp.c:489 ast_rtp_read: Unknown RTP
codec 127 received
After that asterisk gets totaly confused:
Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read too
short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP Read
too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read: RTP
Read too short Nov 23 16:27:33 WARNING[1061908]: rtp.c:428 ast_rtp_read:
RTP Read too short Nov 23 16:27:35 NOTICE[1061908]: rtp.c:489
ast_rtp_read: Unknown RTP codec 1 received Nov 23 16:27:35
WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein: Huh? A GSM frame
that isn't a multiple of 33 or 65 bytes long from RTP (15)? Nov 23
16:27:35 NOTICE[1061908]: channel.c:1724 ast_set_read_format: Unable to
find a path from G723 to ALAW Nov 23 16:27:35 NOTICE[1061908]:
channel.c:1691 ast_set_write_format: Unable to find a path from GSM to
G723 Nov 23 16:27:35 WARNING[1061908]: codec_gsm.c:135 gsmtolin_framein:
Huh? A GSM frame that isn't a multiple of 33 or 65 bytes long from RTP
(20)? Nov 23 16:27:35 WARNING[1061908]: channel.c:2115
ast_channel_make_compatible: No path to translate from
SIP/sip.westend.com-082fd1b8(8) to SIP/xxx-3ef8(1)
Nov 23 16:27:35 WARNING[1061908]: channel.c:2633 ast_channel_bridge:
Can't make SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 compatible Nov
23 16:27:35 WARNING[1061908]: res_features.c:358 ast_bridge_call: Bridge
failed on channels SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8
== Spawn extension (macro-enumcall, s, 211) exited non-zero on
'SIP/sip.westend.com-082fd1b8' in macro 'enumcall'
== Spawn extension (xxx, 911879, 7) exited non-zero on
'SIP/sip.westend.com-082fd1b8'
-- Executing NoOp("SIP/sip.westend.com-082fd1b8", "") in new stack
> cdr_odbc: Query Successful!
Then the call gets hung up. I cannot explain why this happens. I would
have explanations for the fax-machines not able to synchronize or faxes
not being transmitted correctly, as the communication is SIP only, but
this seems a bit strange to me. I can't even tell, if my asterisk
produces these messages, or the other side (aka PSTN-Gateway).
If anyone can bring some light into this behavior I would be very
greatful.
Best regards
Kai
--
Kai Militzer WESTEND GmbH | Internet-Business-Provider
Technik CISCO Systems Partner - Authorized Reseller
Lьtticher StraЯe 10 Tel 0241/701333-11
[EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...)
Hello! Elman Efendiyev wrote: Are you trying to send fax over T.38? As far I understand * don't support T.38 event when passing packets trouth. No, I'm not trying to send the fax over T.38, I am trying to send it in the "voice path" by using the G711 alaw codec. This should work, I think, but it doesn't. Best regards Kai ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...)
Log you posted looks like sitll T.38 problemm Which gates you use? Gateways able to support T.38 will try to use it by default no matter what codec in use. I'd suggest check gateway setup if T.38 is completely disabled Fax call with G711 passtrouth (without T.38) havent any difference comparing to voice call, you can (and probaby will) have troubles with fax transmission (quality, line drops etc) but not with * complain about codecs. -- Sincerely, Elman Efendiyev [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kai Militzer Sent: Tuesday, November 23, 2004 8:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fax over SIP Problems (sorry for this topic ...) Hello! Elman Efendiyev wrote: > Are you trying to send fax over T.38? > As far I understand * don't support T.38 event when passing packets > trouth. No, I'm not trying to send the fax over T.38, I am trying to send it in the "voice path" by using the G711 alaw codec. This should work, I think, but it doesn't. Best regards Kai ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
