Better read up on why a sip phone should register with asterisk. Do a 'sip
show peers' and that will be the list of phones that can "receive"
calls.
-------------------
I've double checked this. Everything is logging in fine, because I can
make calls, check my voicemail, everything except recieve calls on the
SIP devices.
David Phelan wrote:
>
>
>________________________________
>
>From: asterisk-users-bounces@lists.digium.com
>[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jeremi
Bergman
>Sent: Friday, 8 July 2005 9:12 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: [Asterisk-Users] Extension Problems
>
>
>The extensions I've created in AAH, when dialed, always go straight
to
>voicemail.
>
>I may be missing a step... I'm simply adding it in the
"Extensions" part of
>AAH.
>
>I can dial out with my extension, and recieve the voicemail
notification, so
>I know i'm logged in, or so I thought...
>
>This is SIP 210 logging in and 220 making a call to 210
>
>
>
>
><--SNIP -->
>
>
>
>Looks Like an Authentication Issue to me....
>Chack the Username and password on the sip device and AAH
>
>Dave
>
>
>
>
>
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Thanks, but those extensions are listed in that list! I'm stumped Richard Adamson wrote:>Better read up on why a sip phone should register with asterisk. Do a 'sip show peers' and that will be the list of phones that can "receive" calls. >------------------- > > I've double checked this. Everything is logging in fine, because I can > make calls, check my voicemail, everything except recieve calls on the > SIP devices. > > David Phelan wrote: > > > > > > >________________________________ > > > >From: asterisk-users-bounces@lists.digium.com > >[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Jeremi Bergman > >Sent: Friday, 8 July 2005 9:12 AM > >To: Asterisk Users Mailing List - Non-Commercial Discussion > >Subject: [Asterisk-Users] Extension Problems > > > > > >The extensions I've created in AAH, when dialed, always go straight to > >voicemail. > > > >I may be missing a step... I'm simply adding it in the "Extensions" part of > >AAH. > > > >I can dial out with my extension, and recieve the voicemail notification, so > >I know i'm logged in, or so I thought... > > > >This is SIP 210 logging in and 220 making a call to 210 > > > > > > > > > ><--SNIP --> > > > > > > > >Looks Like an Authentication Issue to me.... > >Chack the Username and password on the sip device and AAH > > > >Dave > > > > > > > > > > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >_______________________________________________ >Asterisk-Users mailing list >Asterisk-Users@lists.digium.com >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
Hi all I have a question: i am trying to make a dial plan with IVR with option to call some phone. exten => 3,1,Dial(SIP/"phonenumber"@xxx.xxx.xxx.xxx,,r) and i have the next problem : INVITE sip:"phonenumber"@xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK7f9a1d08..From: "Unknown" <sip:Unk nown@xxx.xxx.xxx.xxx>;tag=as75542381..To: <sip:phonenumber@xxx.xxx.xxx.xxx>..Contact: <sip:Unknown@xxx.xxx.xxx.xxx>..Call-ID: 50e2f6c317 fad67974bfdad353ea22e0@xxx.xxx.xxx.xxx..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Tue, 09 Aug 2005 05:47:10 GMT..Allow: I NVITE, ACK, CANCEL, OPTIONS, BYE, REFER..Content-Type: application/sdp..Content-Length: 218....v=0..o=root 18529 18529 IN IP4 xxx.xxx.xxx.xxx ..s=session..c=IN IP4 xxx.xxx.xxx.xxx..t=0 0..m=audio 12712 RTP/AVP 8 101..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telepho ne-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -.. "Unknown" in the "From" making the problem i am trying to change this value with i puting these 2 lines before the Dial with correct order. exten => 212,5,SetCallerID(221222) exten => 212,6,SetCIDNum(221222) and no success. any help will be appreciated -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050809/73c9a7d4/attachment.htm