Kai Militzer
2004-Nov-23 09:03 UTC
[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
L?tticher Stra?e 10 Tel 0241/701333-11
km@westend.com D-52064 Aachen Fax 0241/911879
Elman Efendiyev
2004-Nov-23 10:23 UTC
[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
elman@protechtele.com
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] 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
L?tticher Stra?e 10 Tel 0241/701333-11
km@westend.com D-52064 Aachen Fax 0241/911879
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