>> Cyprus VoIP wrote:
>>
>>> This is the reINVITE SDP received from the SIP Proxy:
>>> ---
>>> Content-Type: application/sdp
>>> Content-Length: 353
>>>
>>> v=0
>>> o=root 30427 30428 IN IP4 194.98.xxx.xxx
>>> s=session
>>> c=IN IP4 194.98.xxx.xxx
>>> t=0 0
>>> m=image 17548 udptl t38
>>> a
Cyprus VoIP wrote:
> Yes. I saw the message and the required addition in the sip.conf. The
> problem is that if I set it to 72, other terminating gateways that
> support 400 or more would also be limited to 72.
This is incorrect. First, you would not set it to 72, since the endpoint
is already
> Cyprus VoIP wrote:
>
>> This is the reINVITE SDP received from the SIP Proxy:
>> ---
>> Content-Type: application/sdp
>> Content-Length: 353
>>
>> v=0
>> o=root 30427 30428 IN IP4 194.98.xxx.xxx
>> s=session
>> c=IN IP4 194.98.xxx.xxx
>> t=0 0
>> m=image 17548 udptl t38
>> a=T38FaxVersio
Cyprus VoIP wrote:
> This is the reINVITE SDP received from the SIP Proxy:
> ---
> Content-Type: application/sdp
> Content-Length: 353
>
> v=0
> o=root 30427 30428 IN IP4 194.98.xxx.xxx
> s=session
> c=IN IP4 194.98.xxx.xxx
> t=0 0
> m=image 17548 udptl t38
> a=T38FaxVersion:0
> a=T38MaxB
al Message ----
Subject: Re: [asterisk-users] Fax throughput - Asterisk 1.6.1.9
From: Cyprus VoIP
To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Friday, 04 December, 2009 18:21:59
>> It's probably because you are using 1.6.1.9; that release (and older)
>> had a &
> It's probably because you are using 1.6.1.9; that release (and older)
> had a 'feature' that allowed automatic switching back to audio from T.38
> if one of the endpoints sent an audio packet. It turns out that wasn't a
> good idea, and it's been removed... but in later versions. You'll have
> to
Cyprus VoIP wrote:
> If it's not related, why does Asterisk send again INVITE messages to
> both parties? How can this be prevented? I don't see more debug data
> prior to the new INVITE.
It's probably because you are using 1.6.1.9; that release (and older)
had a 'feature' that allowed automati
> Cyprus VoIP wrote:
>
>> So, I enabled the full logger, and the strange thing I see is this message:
>> "Got T.38 Re-invite without audio. Keeping RTP active during T.38 session"
>>
>> It seems that this might be the reason Asterisk initiates a reINVITE
>> with voice codecs, after connecting the
Cyprus VoIP wrote:
> So, I enabled the full logger, and the strange thing I see is this message:
> "Got T.38 Re-invite without audio. Keeping RTP active during T.38 session"
>
> It seems that this might be the reason Asterisk initiates a reINVITE
> with voice codecs, after connecting the 2 parti
> Set 'canreinvite=no' on all applicable peers?
>
I tried with yes and no. No difference. I'm almost certain it's related
to the "Keeping RTP active during T.38 session" issue.
___
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> Cyprus VoIP wrote:
>
>> Thank you for your answer. The 'internal extension' is indeed a T.38
>> capable device that works perfectly when connected directly to the
>> Proxy/ITSP.
>>
>> As you said, the key to debugging/resolving this issue is the logger. I
>> wasn't aware of this file. this is
Set 'canreinvite=no' on all applicable peers?
Cyprus VoIP wrote:
> Hello,
>
> We are trying to send faxes by T.38 protocol to a remote SIP proxy from
> a local extension. The local extension sends the INVITE, Asterisk sends
> the call to the Proxy the call is connected with a regular audio cod
Cyprus VoIP wrote:
> Thank you for your answer. The 'internal extension' is indeed a T.38
> capable device that works perfectly when connected directly to the
> Proxy/ITSP.
>
> As you said, the key to debugging/resolving this issue is the logger. I
> wasn't aware of this file. this is what I h
>> We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the
>> local extension and remote Proxy, but it still forces the call to go
>> back to a voice call.
>
> Define 'internal extension'. Is this a T.38-capable device? If not,
> Asterisk doesn't support TDM-to-T.38 FAX relay (yet
Cyprus VoIP wrote:
> We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the
> local extension and remote Proxy, but it still forces the call to go
> back to a voice call.
That's correct behavior if T.38 cannot autonegotiate.
What happens in the reverse direction, trying to send fa
Cyprus VoIP wrote:
> We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the
> local extension and remote Proxy, but it still forces the call to go
> back to a voice call.
Define 'internal extension'. Is this a T.38-capable device? If not,
Asterisk doesn't support TDM-to-T.38 FAX
Hello,
We are trying to send faxes by T.38 protocol to a remote SIP proxy from
a local extension. The local extension sends the INVITE, Asterisk sends
the call to the Proxy the call is connected with a regular audio codec.
After a few seconds the remote proxy sends an INVITE with UDPTL and the
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