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Summary:ASTERISK-12246: DTMF not reproduced towards ZAP T1 Port after connection has arrive as SIP Trunk
Reporter:Bart Fisher (bhfisher)Labels:
Date Opened:2008-06-22 10:47:17Date Closed:2011-06-07 14:02:40
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Core/PBX
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sample-call.txt
( 1) sample-call-full-debug.txt
Description:When I dial from a standard telephone set to a DID that's provides SIP Trunking and the call is routed (bridged SIP to ZAP T1 (TE410P)), I see the call arriving and setup on ZAP port, but afterwards, any DTMF presses are ignored.

I use e&m signalling toward ZAP. I hear the asterisk placing the DTMF properly toward ZAP. Call connects, any further DTMF pressed digits from SIP Trunk is very short sounding on the ZAP and ignored. ZAP to ZAP connection are perfect. Version 1.2 this was not an issue as it's working now.  

I have a test system to demonstrate this behavior.  And I can give you full access to system to see / test yourself.

I'm using:
* Asterisk Source Version : 1.4.21 (released)
* Zaptel Source Version : 1.4.11
* Libpri Source Version : 1.4.4
* Addons Source Version : 1.4.7
Comments:By: Bart Fisher (bhfisher) 2008-06-23 11:23:11

Just for fun, I had SIP provider change to inband and retested. Now after first press the connection goes 1 way and has double digits.  I switch back since this appears to be another bug with DTMF.

By: Bart Fisher (bhfisher) 2008-06-25 00:02:14

Please help - I'm stuck on 1.2 until this works!

By: Tilghman Lesher (tilghman) 2008-06-25 17:51:34

What are the contents of sip.conf for this user/peer?

By: Bart Fisher (bhfisher) 2008-06-25 22:56:16

[VITELITY-IN]
type=friend
username=innov
secret=secretpass
context=from-pstn
insecure=port,invite
canreinvite=no
host=inbound5.vitelity.net
qualify=yes

I'm using PBX in a Flash, but today I made a trixbox install on another box, and it too has this same behavior -

Also, I found if I press a single dtmf rapidly 2 out of four might be heard.

Also note: After the call is connected (bridge), voice transmission is normal - only the regeneration of dtmf is a problem.



By: Leif Madsen (lmadsen) 2008-11-21 09:14:12.000-0600

What dtmfmode are you using here? I'm presuming you are using RFC2833? Please confirm.

By: Tilghman Lesher (tilghman) 2009-01-25 15:38:19.000-0600

No response from reporter.