Steve Underwood wrote:
Spandsp doesn't support those features. I don't know anything which
does. It seems they can only be used with TCP. Spandsp does support
T38FaxFillBitRemoval
which the FAX for Asterisk package does not (according to Commetrex).
I added indication of
Hello everyone.
I'm struggling to get T.38 faxing to work in Asterisk 1.6.1.13 with a SIP DID
provider here in Brazil (GVT - Vox IP service). Here's my scenario:
When faxes arrive by a specific DID, they are routed thru this simple macro:
[macro-recebefax]
exten =
Vinícius Fontes wrote:
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP v=0... UNSUPPORTED.
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP o=PVG 1265107000170 1265107000170 IN IP4 10.152.0.164...
Hi Kevin,
On 02/02/2010 09:12 PM, Kevin P. Fleming wrote:
Vinícius Fontes wrote:
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP v=0... UNSUPPORTED.
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP
Steve Underwood wrote:
Hi Kevin,
On 02/02/2010 09:12 PM, Kevin P. Fleming wrote:
Vinícius Fontes wrote:
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP v=0... UNSUPPORTED.
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp:
On 02/02/2010 10:11 PM, Kevin P. Fleming wrote:
Steve Underwood wrote:
Hi Kevin,
On 02/02/2010 09:12 PM, Kevin P. Fleming wrote:
Vinícius Fontes wrote:
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589 process_sdp: Processing
session-level SDP v=0... UNSUPPORTED.
[Feb
Steve Underwood wrote:
That's how T.38 calls normally start. They mostly start as audio, and
switch into T.38 mode later. We have only seen an initial fragment in
the log. We haven't seen anything that's actually wrong. We see an offer
to do telephony events, and from there things might
- Steve Underwood ste...@coppice.org escreveu:
On 02/02/2010 10:11 PM, Kevin P. Fleming wrote:
Steve Underwood wrote:
Hi Kevin,
On 02/02/2010 09:12 PM, Kevin P. Fleming wrote:
Vinícius Fontes wrote:
[Feb 2 08:38:06] DEBUG[21032]: chan_sip.c:7589
Vinícius Fontes wrote:
I couldn't agree more Steve.
Is there any other info I could provide in order to help you find out what's
wrong? I could even open an issue on Mantis if the Digium staff think it's
worth it.
Post a 'sip set debug' capture of the failing call in this thread; that
- Kevin P. Fleming kpflem...@digium.com escreveu:
Vinícius Fontes wrote:
I couldn't agree more Steve.
Is there any other info I could provide in order to help you find
out what's wrong? I could even open an issue on Mantis if the Digium
staff think it's worth it.
Post a 'sip
- Vinícius Fontes vinic...@canall.com.br escreveu:
- Kevin P. Fleming kpflem...@digium.com escreveu:
Vinícius Fontes wrote:
I couldn't agree more Steve.
Is there any other info I could provide in order to help you find
out what's wrong? I could even open an issue on
On 02/03/2010 12:45 AM, Vinícius Fontes wrote:
- Kevin P. Flemingkpflem...@digium.com escreveu:
Vinícius Fontes wrote:
I couldn't agree more Steve.
Is there any other info I could provide in order to help you find
out what's wrong? I could even open an issue on
Steve Underwood wrote:
I wonder why Asterisk would say:
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 344
v=0
o=root 44350963 44350964 IN IP4 10.153.66.146
s=Asterisk PBX 1.6.1.13
c=IN IP4 10.153.66.146
t=0 0
m=image 4819 udptl
- Steve Underwood ste...@coppice.org escreveu:
On 02/03/2010 12:45 AM, Vinícius Fontes wrote:
- Kevin P. Flemingkpflem...@digium.com escreveu:
Vinícius Fontes wrote:
I couldn't agree more Steve.
Is there any other info I could provide in order to help you
Vinícius Fontes wrote:
I've put it on pastebin because is was a lot of text. Here's the link:
http://pastebin.com/m7467cea1. That's all the information on the CLI with
verbose=3 and sip set debug peer voxip.
OK, with the complete capture we can see that the problem is actually
quite
On 02/03/2010 03:14 AM, Kevin P. Fleming wrote:
Steve Underwood wrote:
I wonder why Asterisk would say:
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 344
v=0
o=root 44350963 44350964 IN IP4 10.153.66.146
s=Asterisk PBX 1.6.1.13
- Kevin P. Fleming kpflem...@digium.com escreveu:
Vinícius Fontes wrote:
I've put it on pastebin because is was a lot of text. Here's the
link: http://pastebin.com/m7467cea1. That's all the information on the
CLI with verbose=3 and sip set debug peer voxip.
OK, with the complete
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