Still no luck.... Br, Mohit
-----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Peder Sent: Monday, March 15, 2010 6:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways exten=07028XXXXXX,1,Dial(SIP/${ext...@pccw-kpn) You aren't sending an outbound DID with just SIP/PCCW-KPN. -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Mohit Saxena Sent: Monday, March 15, 2010 12:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways Sip.comf [PCCW-KPN] type=peer host=41.205.190.15 allow=ulaw qualify=100 nat=no canreinvite=no user=07028000709 extension.conf exten=07028XXXXXX,1,Dial(SIP/PCCW-KPN) Cisco Gateway: dial-peer voice 110 voip description Voip peer to test the server destination-pattern 1234 session protocol sipv2 session target ipv4:196.3.60.24 session transport udp incoming called-number .T dtmf-relay rtp-nte codec g711ulaw fax-relay ecm disable fax rate 9600 fax protocol t38 ls-redundancy 1 hs-redundancy 1 fallback pass-through g711ulaw clid strip Br, Mohit C. Saxena I Data/ISP Department Starcomms Plc. 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709 email:[email protected] www.starcomms.com -----Original Message----- From: [email protected] [mailto:[email protected]] On Behalf Of Tim Nelson Sent: Monday, March 15, 2010 6:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media gateways Continuing with the top posting parade... Can you post your {sanitized} sip.conf and your extensions.conf for inspection? --Tim ----- "Mohit Saxena" <[email protected]> wrote: > The problem is not with cisco as the SIP header on debug doesn't > contain the called number. It only says To:sip:ip add of cisco gw. It > should say number:ip addr of cisco gw. > > Br, > Mohit C. Saxena I Data/ISP Department > Starcomms Plc. > 1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, > +234-702-8000-709 email:[email protected] > www.starcomms.com > > > -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of David > Backeberg > Sent: Monday, March 15, 2010 5:48 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk to be used with Ciscs media > gateways > > On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena <[email protected]> > wrote: > > I have been trying to do this since 2 days but couldn't make > it....need your help.. > > Well, you could certainly ask Cisco for help. > You did pay Cisco money, right? > > > PSTN-----Cisco AS5350-------Asterisk Box--------VoIP Providers > > > I am able to place call from cisco gateway to the asterisk box and > also to some softphones extensions but >when making a call from > softphone from asterisk box to PSTN, it fails. While I debug on Cisco > gateway I found >that the To field is SIP header is coming as > sip:41.205.190.15 which is not correct, instead it should be dialed > >number:41.205.190.15 > > Then the problem seems to be between your asterisk box and your > Cisco. > Perhaps if you told us what you were trying to SIP dial, we would be > able to tell us what you did wrong. > > > Has any one of you tried using Asterisk in this scenario > > yes. > > > and also to do LCR and Quality based routing of International > calls? > > I don't know what that means. > > > Please let me know if there is any documentation /example of this > kind available > > There is. > cisco.com > you pay them, then you can use their documentation. > > -- > _____________________________________________________________________ > -- 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 > > -- > _____________________________________________________________________ > -- 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? 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