Thanks for your answer. The logs where posted at pastebin, here the links:
- Working Phone: http://pastebin.com/q3pHcwna - Not working phone: http://pastebin.com/iiCHPMmn 2012/8/20 Rusty Newton <rnew...@digium.com> > On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote: > >> Hi >> >> I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of >> ATA on the network who autenticate the phones: Linksys PAP2, Overtek >> OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at >> the same network all with g729 codecs and rfc2833 for the DTMF. Making >> calls via the Overtek ATA the DTMF works fine but at the others ATA it >> doesn't. >> >> My config: >> >> - asterisk 1.6.2.13 >> - dahdi 2.3.0.1 >> - The phones connected are all physical phones >> > There is additional data you can provide to make it easier for others to > help out: > If you can pastebin an Asterisk log including all message types plus > VERBOSE,DEBUG,DTMF [1] during a working call and a failed call that would > be very helpful. > A step beyond that is to also provide a SIP and RTP packet trace so that > whoever wants to help can look through it in Wireshark. > > If you can get the packet trace for the same calls you gather log data > for, that would be best. > > Thanks! > > [1] https://wiki.asterisk.org/**wiki/display/AST/Collecting+** > Debug+Information<https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information> > > -- > Rusty Newton > Digium, Inc | Open Source Community Support Manager > Check us out at: www.digium.com www.asterisk.org > > > -- > ______________________________**______________________________**_________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users> > -- Att.* ***
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