Tom Moore, thanks for the reply!

Changing the rfcmode to anything doesn't work.  I tried info, inband,
rfc2833, and auto with the exact same results.

I can hear the button being pressed with I listen on either phone.


On Fri, 2008-02-15 at 13:04, Tom Moore wrote:
> Try
> dtmfmode=rfc2833
> 
> I use this on all my sip trunks.
> 
> Tom
>  
> 
> -----Original Message-----
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gerald Cox
> Sent: Friday, February 15, 2008 2:55 PM
> To: Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Callweaver-users] Problems with callweaver handling VoIP calls
> 
> Tom Moore, you're my new best friend!!!!  :)  The change worked great
> for passing calling data through VoIP to callweaver.
> 
> Everything is working except callweaver isn't recognizing DTMF signals. 
> Any ideas on that aspect?
> 
> My configuration for the VoIP is:
> 
>  [VoIP]
> type=peer
> nat=yes
> host=<Qwest's VoIP IP>
> context=ht488
> dtmfmode=info
> insecure=port,invite
> externip = <my external IP>
> canreinvite=yes
> 
> 
> On Fri, 2008-02-15 at 12:30, Tom Moore wrote:
> > Not sure if the setting is the same in Callweaver as it is in Asterisk,
> but
> > in sip.conf you'll either want to put an entry such as:
> > externip 1.2.3.4
> > or
> > externhost my.host
> > 
> > Right now your sip server is sending out a private address as its
> interface
> > to send back packets too and you need it to hand off an address that is
> the
> > public interface of your router.
> > 
> > Tom
> >  
> > 
> > -----Original Message-----
> > From: [EMAIL PROTECTED]
> > [mailto:[EMAIL PROTECTED] On Behalf Of Gerald Cox
> > Sent: Friday, February 15, 2008 2:21 PM
> > To: Users Mailing List - Non-Commercial Discussion
> > Subject: [Callweaver-users] Problems with callweaver handling VoIP calls
> > 
> > I got callweaver setup with a few SIP phones, an FXO phone, and Qwest as
> > a VoIP provider.
> > 
> > Direct SIP calls to the callweaver from an external IP works great. 
> > Calls from SIP phone to SIP phone in callweaver works great.  The t.38
> > part of callweaver works great.
> > 
> > When calling through VoIP, I cannot hear what's being said on the
> > regular phone that dialed our 8xx number that's directed straight to a
> > SIP phone on our callweaver server, but I can hear, on either phone,
> > what is being said on the SIP phone.  DTMF info is not passed into
> > callweaver from the regular phone dialed/dialing via VoIP.
> > I experience the exact same problems when placing an outgoing call as I
> > do when getting an incoming call.  The only difference is, when getting
> > an incoming call, it drops the call after about 30 seconds.  It never
> > drops the call on outgoing calls.  Also, on incoming calls, the call
> > doesn't get dropped/closed when the regular phone, that dialed
> > callweaver through VoIP, gets hung up.
> > 
> > The only log info that I could think of that might help follows:
> > 
> > Pulled from the callweaver CLI when placing an incoming call to our 8xx
> > #:
> > 
> > Retransmitting #6 (no NAT) to 65.115.130.151:5060:
> > SIP/2.0 200 OK
> > Via: SIP/2.0/UDP
> >
> 65.115.130.151:5060;branch=z9hG4bK0olmbr107go0ab0ko780.1;received=65.115.130
> > .151
> > From:
> >
> <sip:[EMAIL PROTECTED]:5060;isup-oli=0;affo6642p801=Affo6642P801-62p
> > 8g55gnns02>;tag=SD1kj3e01-gK07346d05
> > To: <sip:[EMAIL PROTECTED]:5060>;tag=as1c61fbbd
> > Call-ID: SD1kj3e01-988bc3631bb5f1e8413ded042ff878df-v300050
> > CSeq: 11694 INVITE
> > User-Agent: CallWeaver
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> > Max-Forwards: 70
> > Contact: <sip:[EMAIL PROTECTED]>
> > Content-Type: application/sdp
> > Content-Length: 218
> >  
> > 
> > v=0
> > o=root 25882 25882 IN IP4 192.168.1.160
> > s=session
> > c=IN IP4 192.168.1.160
> > t=0 0
> > m=audio 10344 RTP/AVP 0 100
> > a=rtpmap:0 PCMU/8000
> > a=rtpmap:100 telephone-event/8000
> > a=fmtp:100 0-16
> > a=silenceSupp:off - - - -
> > 
> > 
> > 
> > I get a, about, 6 or 7 of those messages before it drops the call and
> > spits out these messages:
> > 
> > Feb 15 06:14:07 WARNING[3075787664]: chan_sip.c:1554 retrans_pkt:
> > Maximum retries exceeded on transmission
> > SD1kj3e01-988bc3631bb5f1e8413ded042ff878df-v300050 for seqno 11694
> > (Critical Response)
> > Feb 15 06:14:07 WARNING[3075787664]: chan_sip.c:1576 retrans_pkt:
> > Hanging up call SD1kj3e01-988bc3631bb5f1e8413ded042ff878df-v300050 - no
> > reply to our critical packet.
> >   == Spawn extension (menuOptions, s, 13) exited non-zero on
> > 'SIP/5060-09cea5c8'
> > 
> > 
> > 
> > Any ideas or other places to look for problems is much appreciated.
> > 
> > _______________________________________________
> > Callweaver-users mailing list
> > [email protected]
> > http://lists.callweaver.org/mailman/listinfo/callweaver-users
> > 
> > No virus found in this incoming message.
> > Checked by AVG Free Edition. 
> > Version: 7.5.516 / Virus Database: 269.20.6/1280 - Release Date: 2/15/2008
> > 9:00 AM
> >  
> > 
> > No virus found in this outgoing message.
> > Checked by AVG Free Edition. 
> > Version: 7.5.516 / Virus Database: 269.20.6/1280 - Release Date: 2/15/2008
> > 9:00 AM
> >  
> > 
> > _______________________________________________
> > Callweaver-users mailing list
> > [email protected]
> > http://lists.callweaver.org/mailman/listinfo/callweaver-users
> > 
> 
> _______________________________________________
> Callweaver-users mailing list
> [email protected]
> http://lists.callweaver.org/mailman/listinfo/callweaver-users
> 
> No virus found in this incoming message.
> Checked by AVG Free Edition. 
> Version: 7.5.516 / Virus Database: 269.20.6/1280 - Release Date: 2/15/2008
> 9:00 AM
>  
> 
> No virus found in this outgoing message.
> Checked by AVG Free Edition. 
> Version: 7.5.516 / Virus Database: 269.20.6/1280 - Release Date: 2/15/2008
> 9:00 AM
>  
> 
> _______________________________________________
> Callweaver-users mailing list
> [email protected]
> http://lists.callweaver.org/mailman/listinfo/callweaver-users
> 

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