RE: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme
Hi I have a Cisco 2801 with CME and I'm trying to connect it to Asterisk through SIP. Can you please send me the Dial-peer configuration that creates a trunk between the Cisco router and Asterisk. Thank you -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Jones Sent: Wednesday, March 16, 2005 5:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Possible SPAM] :[Asterisk-Users] about sip,asterisk and cisco ccme I am starting to work on a similar solution, but with full call manager rather than CME. I am going to use Asterisk to accept POTS calls through PCI FXO ports (winmodems) and then forward the calls through to call manager via SIP. I don't have my FXO cards yet (waiting for UPS man!!) but I have * talking to the CM through SIP just fine. I am testing with the Cisco softphone, connected as a call manager extension, and using the dial-plan to direct the call to *, and I do successfully get the * voicemail. Why do you want to use h323/skinny rather than SIP? -Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Riela Sent: Wednesday, March 16, 2005 5:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, I would create a structure like this: external sip server \ external sip server |-| Asterisk |--| Cisco CME |---| ip phones | external sip server / I would use Asterisk as SIP client for some SIP accounts on external servers ... then register those via H323 (if possible; skynny?) on Cisco CME ... Then I would use Asterisk to add the voicemail feature to Cisco CME. I don't know if that's possible, I'm really newbie on Asterisk, I know only Cisco world, and just a little bit. Any advice will be appreciated. Thanks for your support Regards dott. Andrea Riela -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCOLJtMakHrsrHP9wRAmDfAJ9AgcMf1CmdrLBk4HEdlvWKZiht7QCfcgns GbTX2LxGxO3ZR7iMIPqreJA= =eKlT -END PGP SIGNATURE- ___ 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 ___ 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
RE: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme
Gilbert Abboud wrote: I have a Cisco 2801 with CME and I'm trying to connect it to Asterisk through SIP. Can you please send me the Dial-peer configuration that creates a trunk between the Cisco router and Asterisk. You can try something like this: dial-peer voice 900 voip destination-pattern 9... session protocol sipv2 !(the address of the Asterisk server) session target ipv4:192.168.0.100 !(in Asterisk use dtmfmode=rfc2833) dtmf-relay rtp-nte codec g711ulaw --TWH ___ 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
RE: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme
I am starting to work on a similar solution, but with full call manager rather than CME. I am going to use Asterisk to accept POTS calls through PCI FXO ports (winmodems) and then forward the calls through to call manager via SIP. I don't have my FXO cards yet (waiting for UPS man!!) but I have * talking to the CM through SIP just fine. I am testing with the Cisco softphone, connected as a call manager extension, and using the dial-plan to direct the call to *, and I do successfully get the * voicemail. Why do you want to use h323/skinny rather than SIP? -Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrea Riela Sent: Wednesday, March 16, 2005 5:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Possible SPAM] :[Asterisk-Users] about sip, asterisk and cisco ccme -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, I would create a structure like this: external sip server \ external sip server |-| Asterisk |--| Cisco CME |---| ip phones | external sip server / I would use Asterisk as SIP client for some SIP accounts on external servers ... then register those via H323 (if possible; skynny?) on Cisco CME ... Then I would use Asterisk to add the voicemail feature to Cisco CME. I don't know if that's possible, I'm really newbie on Asterisk, I know only Cisco world, and just a little bit. Any advice will be appreciated. Thanks for your support Regards dott. Andrea Riela -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCOLJtMakHrsrHP9wRAmDfAJ9AgcMf1CmdrLBk4HEdlvWKZiht7QCfcgns GbTX2LxGxO3ZR7iMIPqreJA= =eKlT -END PGP SIGNATURE- ___ 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
