Hi, This is my setup; 1. PSTN ==> Cisco ==> Internet ==> Asterisks ==> Grandstream Phone 2. Grandstream ATA =>SIP Proxy ==> Internet ==> Asterisks ==> Grandstream Phone In both cases above when I dialed the DID (say) 1-213-444-1234 from either the PSTN or Grandstream ATA the response I "see" on the asterisks is somthing like this below; "SIP/kkk.kkk.kkk.kkk-084cfc38" ==> where kkk.kkk.kkk.kkk is the IP address on the Cisco or SIP Proxy I was actually expecting something like SIP/12134441234 that will allow me the opportunity to route the incoming calls by DID to different context base on each DID. How do I achieve this? ------------------------------------------------------------- Olusoji (Soji) Oyenuga Senior VoIP Project Manager Modern Digital Communications Inc Phone: 1-306-683-2089 Email: soji@mdci.ca MSN: sogi@mdci.ca http://www.mdci.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050721/02105fad/attachment.htm
In sip.conf you specify a context right? In extensions.conf, in that context, you route the call... exten => 12134441234,1,Dial(whatever) --On Thursday, July 21, 2005 11:30 AM -0600 "Olusoji (soji) Oyenuga" <soji@mdci.ca> wrote:> > Hi, > > This is my setup; > > 1. PSTN ==> Cisco ==> Internet ==> Asterisks ==> Grandstream Phone > 2. Grandstream ATA =>SIP Proxy ==> Internet ==> Asterisks ==> > Grandstream Phone > > In both cases above when I dialed the DID (say) 1-213-444-1234 from > either the PSTN or Grandstream ATA the response I "see" on the asterisks > is somthing like this below; > > "SIP/kkk.kkk.kkk.kkk-084cfc38" ==> where kkk.kkk.kkk.kkk is the IP > address on the Cisco or SIP Proxy > > I was actually expecting something like > > SIP/12134441234 > > that will allow me the opportunity to route the incoming calls by DID to > different context base on each DID. > > How do I achieve this? > > ------------------------------------------------------------- > Olusoji (Soji) Oyenuga > Senior VoIP Project Manager > Modern Digital Communications Inc > Phone: 1-306-683-2089 > Email: soji@mdci.ca > MSN: sogi@mdci.ca > http://www.mdci.ca
This only works if you DONT have: insecure=very in your SIP section. Rene Kluwen Chimit> In sip.conf you specify a context right? > > In extensions.conf, in that context, you route the call... > > exten => 12134441234,1,Dial(whatever) > > > > --On Thursday, July 21, 2005 11:30 AM -0600 "Olusoji (soji) Oyenuga" > <soji@mdci.ca> wrote: > >> >> Hi, >> >> This is my setup; >> >> 1. PSTN ==> Cisco ==> Internet ==> Asterisks ==> Grandstream Phone >> 2. Grandstream ATA =>SIP Proxy ==> Internet ==> Asterisks ==> >> Grandstream Phone >> >> In both cases above when I dialed the DID (say) 1-213-444-1234 from >> either the PSTN or Grandstream ATA the response I "see" on the asterisks >> is somthing like this below; >> >> "SIP/kkk.kkk.kkk.kkk-084cfc38" ==> where kkk.kkk.kkk.kkk is the IP >> address on the Cisco or SIP Proxy >> >> I was actually expecting something like >> >> SIP/12134441234 >> >> that will allow me the opportunity to route the incoming calls by DID to >> different context base on each DID. >> >> How do I achieve this? >> >> ------------------------------------------------------------- >> Olusoji (Soji) Oyenuga >> Senior VoIP Project Manager >> Modern Digital Communications Inc >> Phone: 1-306-683-2089 >> Email: soji@mdci.ca >> MSN: sogi@mdci.ca >> http://www.mdci.ca > > > > > _______________________________________________ > 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 > > >
This only works if you DONT have: insecure=very in your SIP section. Rene Kluwen Chimit> In sip.conf you specify a context right? > > In extensions.conf, in that context, you route the call... > > exten => 12134441234,1,Dial(whatever) > > > > --On Thursday, July 21, 2005 11:30 AM -0600 "Olusoji (soji) Oyenuga" > <soji@mdci.ca> wrote: > >> >> Hi, >> >> This is my setup; >> >> 1. PSTN ==> Cisco ==> Internet ==> Asterisks ==> Grandstream Phone >> 2. Grandstream ATA =>SIP Proxy ==> Internet ==> Asterisks ==> >> Grandstream Phone >> >> In both cases above when I dialed the DID (say) 1-213-444-1234 from >> either the PSTN or Grandstream ATA the response I "see" on the asterisks >> is somthing like this below; >> >> "SIP/kkk.kkk.kkk.kkk-084cfc38" ==> where kkk.kkk.kkk.kkk is the IP >> address on the Cisco or SIP Proxy >> >> I was actually expecting something like >> >> SIP/12134441234 >> >> that will allow me the opportunity to route the incoming calls by DID to >> different context base on each DID. >> >> How do I achieve this? >> >> ------------------------------------------------------------- >> Olusoji (Soji) Oyenuga >> Senior VoIP Project Manager >> Modern Digital Communications Inc >> Phone: 1-306-683-2089 >> Email: soji@mdci.ca >> MSN: sogi@mdci.ca >> http://www.mdci.ca > > > > > _______________________________________________ > 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 > > >