Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/c131f743/attachment.htm
On Thu, 2006-12-07 at 17:51 +0100, FaberK wrote:> Hi to all, > I got a Cisco 2651XM wired to an E1 PRI. > What I want to do is to pass all incoming calls to my asterisk. > This is my actual conf: > http://pastebin.ca/270677 > with this I'm able to call my number from outside, but the call stop > on the 2600, infact I can hear the tone, but I'm not able to forward > calls to my asterisk. > > Anyone got an idea of my errors?Your config says "session transport tcp" for SIP. Asterisk does not support SIP over TCP, only SIP over UDP so change that to UDP. Not sure about these: Your config says "no dspfarm" but you have specified g729br8 codecs. Is g729br8 supported on a Cisco 2600 without the PVDM2 or NM-HDV2 modules that have the dsp's on board to do the g729br8<->alaw/ulaw transcoding? Your config says "clock source internal". Why don't you use the clock of the telco that provides the E1? That would prevent clock slips as the telco's clock is bound to be more reliable and precise than the internal clock in the Cisco. Regards, Patrick
voice service voip sip session transport tcp Last I checked, asterisk doesn't support TCP SIP signaling (or RTP over TCP). See what happens if you change it back to the UDP default. On 12/7/06, FaberK <f.faberk@gmail.com> wrote:> Hi to all, > I got a Cisco 2651XM wired to an E1 PRI. > What I want to do is to pass all incoming calls to my asterisk. > This is my actual conf: > http://pastebin.ca/270677 > with this I'm able to call my number from outside, but the call stop on the > 2600, infact I can hear the tone, but I'm not able to forward calls to my > asterisk. > > Anyone got an idea of my errors? > > Thanks to all. > -- > .:FaberK:. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-- "I am Dyslexic of Borg. Fusistance is retile. Your ass will be laminated!"
Hi In dial-peer voice 697617664 voip your must specify into voip dial peer session protocol sipv2 and check if session target sip-server is corect doing a ping to sip-server . I think you must configure it with ipv4:ip_addres or map a host entry with ip host sip-server x.x.x.x in global configuration mode you have forgotten to configure a pots dial peer for your controler. put something like this dial-peer voice 10 pots destination-pattern 0T fax rate disable direct-inward-dial port 1/0:15 and try if you can write authentication username "asterisk-uername" password XXXXXX this last command should allow dial-peer voice 10 to register within asterisk I hope it will help you best regards 2006/12/7, FaberK <f.faberk@gmail.com>:> Hi to all, > I got a Cisco 2651XM wired to an E1 PRI. > What I want to do is to pass all incoming calls to my asterisk. > This is my actual conf: > http://pastebin.ca/270677 > with this I'm able to call my number from outside, but the call stop on > the 2600, infact I can hear the tone, but I'm not able to forward calls to > my asterisk. > > Anyone got an idea of my errors? > > Thanks to all. > -- > .:FaberK:. > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com <http://easynews.com/>-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/ea46052a/attachment.htm
http://pastebin.ca/270840 This is the newone with some changements. Unfortunately, always the same problem. Fran, if I add the "dial-peer voice 10 pots", I receive the advise that the number does not exist. Also, I do not find the way to add "authentication username "asterisk-uername" password XXXXXX". The story continues... Thanks F. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/1c443f9d/attachment.htm
As I understand your configuration , dial-peer voice 697617664 voip, only forward the pattern 697617664( destination-pattern 697617664) to XXX.XXX.XXX .115:5060 ( session target ipv4:XXX.XXX.XXX.115:5060) that I think is your Asterisk box. An incoming call in your E1 must much a destination pattern, your only destination pattern is 697617664. Usually an E1 has several DID associated it in a consecutive range, 91 5344XXX for example. otherwise, for outgoing calls you must configure a pots dial peer ,you can put a randon name to the dial peer. You can configure asterisk , without user registration with the sip.confinsecure option when I copied dial-peer voice 10 pots destination-pattern 0T should be .T it tells cisco 26xx router what patterns can be reached throught E1 I?ll take a look into the cisco web site for sip user authentication, I have a configuration done, but with FXS interfaces and worsk fine. best regards 2006/12/7, FaberK <f.faberk@gmail.com>:> > http://pastebin.ca/270840 > This is the newone with some changements. > Unfortunately, always the same problem. > > Fran, if I add the "dial-peer voice 10 pots", I receive the advise that > the number does not exist. > Also, I do not find the way to add "authentication username > "asterisk-uername" password XXXXXX". > > The story continues... > > Thanks > > F. >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061208/fdad29b6/attachment.htm