On Mon, Mar 16, 2009 at 12:10 AM, Steve Totaro <stot...@totarotechnologies.com> wrote: > Again, if I am interpreting this correctly, he is not using SIP. A > four port card 2fxo/2fxs means to me that he is not using SIP at all.
You are correct. I was confused. It is Zap (zaptel) channel > > If by card, you mean some kind of SIP gateway, then I misunderstood > and the problem, but seeing DAHDI channels leads me to believe that > SIP is not required and actually causing your problems. > > SIP is a protocol for VoIP, DAHDI/Zaptel is TDM (analog POTS in this > case)... If you had a SIP device, it would be connected to the data > network, not a phone line. Can you just plug your phone into a > regular landline jack and get dialtone? If so, forget SIP for now. > > Comment out or delete all your sip.conf peers since you are not using SIP. > > Change your dialplan to not (Dial/SIP" but (Dial/DAHDI/1,10) and the > correct channel to your FXS port that the phone is connected to. Dial(Zap/1) worked like a charm. Thanks all for your help > > Thanks, > Steve Totaro > > On Sun, Mar 15, 2009 at 9:20 PM, Marco Mouta <marco.mo...@gmail.com> wrote: >> Hi, >> >> problem is that you are saying that phone in sip.conf is at the same >> ip address of your asterisk box so you are dialing into a loop to your >> self asterisk box >> >> [phone] >> type=friend >> context=phone1 >> secret=g00dpazzwerd >> bindport=5060 >> host=192.168.1.106 >> dtmfmode=rfc2833 >> >> what you need is: >> >> [phone] >> type=friend >> context=phone1 >> secret=g00dpazzwerd >> dtmfmode=rfc2833 >> host=dynamic >> ;configuring your codecs (i don't know what else you have configured, >> just preventing audio for you) >> disallow=all >> allow=ulaw >> allow=alaw >> allow=gsm >> >> >> Dial sip/phone is enough too.. >> >> [from-pstn] >> ;include => default >> exten => s,1,Dial(SIP/phone,10) >> exten => s,2,Voicemail(line) >> exten => s,3,Hangup >> >> >> hope it helps. >> >> don't forget to asterisk reload on cli. >> >> Looking forward to hearing from you. >> >> cheers >> >> -- >> Marco Mouta >> >> >> >> On Sun, Mar 15, 2009 at 10:28 PM, Asif Iqbal <vad...@gmail.com> wrote: >>> Hi I looked at few emails related to this subject. And still not sure >>> how to solve the loop detect problem for my case >>> >>> iqb...@improvise:/etc/asterisk$ cat sip.conf >>> >>> [general] >>> context=line1 >>> >>> [phone] >>> type=friend >>> context=phone1 >>> secret=g00dpazzwerd >>> bindport=5060 >>> host=192.168.1.106 >>> dtmfmode=rfc2833 >>> >>> [line] >>> type=friend >>> context=line1 >>> secret=anothers33cret >>> bindport=5061 >>> host=192.168.1.106 >>> dtmfmode=rfc2833 >>> >>> iqb...@improvise:/etc/asterisk$ cat extensions.conf >>> [default] >>> exten => s,1,Answer >>> exten => s,2,Wait(2) >>> exten => s,3,Playback(tt-monkeys) >>> exten => s,4,Hangup >>> >>> [from-internal] >>> include => default >>> >>> [phone1] >>> >>> [from-pstn] >>> ;include => default >>> exten => s,1,Dial(SIP/ph...@phone,10) >>> exten => s,2,Voicemail(line) >>> exten => s,3,Hangup >>> >>> [line1] >>> >>> >>> So my home land line is going to the FXO port and my home phone is >>> hanging off of FXS port. >>> >>> Here are the contexts for my fxo/fxs card >>> >>> >>> improvise*CLI> dahdi show channels >>> Chan Extension Context Language MOH Interpret >>> pseudo default default >>> 1 from-internal default >>> 2 from-internal default >>> 3 from-pstn default >>> 4 from-pstn default >>> >>> >>> I want to call from my cell and make my home phone ring and if I dont >>> pickup in 10 secs I want the call >>> go to my voicemail. But I am getting a loop detect. The debug output >>> is attached. >>> >>> What am I doing wrong? >>> >>> -- >>> Asif Iqbal >>> PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu >>> A: Because it messes up the order in which people normally read text. >>> Q: Why is top-posting such a bad thing? >>> >>> >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Asif Iqbal PGP Key: 0xE62693C5 KeyServer: pgp.mit.edu A: Because it messes up the order in which people normally read text. Q: Why is top-posting such a bad thing? _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users