Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1, Asterisk is running strictly VoIP over the network and using CCM as the trunk. Calls from the SIP phones connected to Asterisk work fine. They can call both external numbers and any Cisco extensions attached to CCM. Calls from CCM to Asterisk fail without any notification in Asterisk (and I DID have this working at one point, but I suspect that our Telco may have pooched the config somehow, since they're in the process of connecting us to another CCM site). I have verified: Media Termination point exists, Calling Search Space exists, Trunk is properly defined (uLaw 711, UDP, ip address & port, etc), and a route pattern exists to take calls to the right trunk. The system will let me complete the dialing sequence to the Asterisk server, but as soon as I enter the last digit I get a busy signal. Thoughts anyone? Here's my sip.conf if that helps... [callman] type=peer context=incoming insecure=very host=(ip of my call manager server) disallow=all allow=ulaw allow=alaw nat=no canreinvite=yes qualify=yes Thanks! Aaron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080311/b35cc0cf/attachment.htm
On Tue, 2008-03-11 at 06:11 -0600, Aaron Fransen wrote: [snip]> > Here's my sip.conf if that helps... > > [callman] > type=peer > context=incoming > insecure=very > host=(ip of my call manager server) > disallow=all > allow=ulaw > allow=alaw > nat=no > canreinvite=yes > qualify=yesWithout any logs it's difficult to say what's going on. Assuming that you want the signaling & media from the CCM to the IP phones behind the Asterisk box to go through your Asterisk box set canreinvite to "no" instead of "yes". Regards, Patrick
I've noticed two differences in what you described and my working CM setup: 1. My sip trunk in CM is defined as 711alaw, you have ulaw. 2. My sip.conf defines CM as a type=friend instead of a peer. Do you have any SIP phones connected to Asterisk (you could use a softphone like the free xten)? Can you call the phone from CallManager? Peter Pauly http://www.usbtests.com On 3/11/08, Aaron Fransen <aaron.fransen at gmail.com> wrote:> > Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1, > Asterisk is running strictly VoIP over the network and using CCM as the > trunk. > > Calls from the SIP phones connected to Asterisk work fine. They can call > both external numbers and any Cisco extensions attached to CCM. > > Calls from CCM to Asterisk fail without any notification in Asterisk (and I > DID have this working at one point, but I suspect that our Telco may have > pooched the config somehow, since they're in the process of connecting us to > another CCM site). > > I have verified: Media Termination point exists, Calling Search Space > exists, Trunk is properly defined (uLaw 711, UDP, ip address & port, etc), > and a route pattern exists to take calls to the right trunk. > > The system will let me complete the dialing sequence to the Asterisk > server, but as soon as I enter the last digit I get a busy signal. > > Thoughts anyone? > > Here's my sip.conf if that helps... > > [callman] > type=peer > context=incoming > insecure=very > host=(ip of my call manager server) > disallow=all > allow=ulaw > allow=alaw > nat=no > canreinvite=yes > qualify=yes > > Thanks! Aaron > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users >
One interesting thing: If I call the Cisco system from the Aastra 57i connected to Asterisk, then pick up the Cisco it works. I then hit the "CONFRN" button on the Cisco and dial my cell phone, then hit "CONFRN" again to conference in, the Asterisk log shows that it puts me on "music on hold" and never takes me off again. Odd? On Tue, Mar 11, 2008 at 6:11 AM, Aaron Fransen <aaron.fransen at gmail.com> wrote:> > Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1, > Asterisk is running strictly VoIP over the network and using CCM as the > trunk. > > Calls from the SIP phones connected to Asterisk work fine. They can call > both external numbers and any Cisco extensions attached to CCM. > > Calls from CCM to Asterisk fail without any notification in Asterisk (and > I DID have this working at one point, but I suspect that our Telco may have > pooched the config somehow, since they're in the process of connecting us to > another CCM site). > > I have verified: Media Termination point exists, Calling Search Space > exists, Trunk is properly defined (uLaw 711, UDP, ip address & port, etc), > and a route pattern exists to take calls to the right trunk. > > The system will let me complete the dialing sequence to the Asterisk > server, but as soon as I enter the last digit I get a busy signal. > > Thoughts anyone? > > Here's my sip.conf if that helps... > > [callman] > type=peer > context=incoming > insecure=very > host=(ip of my call manager server) > disallow=all > allow=ulaw > allow=alaw > nat=no > canreinvite=yes > qualify=yes > > Thanks! Aaron >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080311/b2ccd7b0/attachment.htm
More information: Wireshark running on the * box shows that the request is being received, but rejected: Request: INVITE sip:2247@(ipaddress of *):5060, with session description Status: 407 Proxy Authentication Required Request: ACK sip:2247@(ipaddress of *):5060 Did I miss a security setting? On Tue, Mar 11, 2008 at 6:11 AM, Aaron Fransen <aaron.fransen at gmail.com> wrote:> > Running Cisco Call Manager 6.1 and Asterisk 1.4. CCM is connected to a T1, > Asterisk is running strictly VoIP over the network and using CCM as the > trunk. > > Calls from the SIP phones connected to Asterisk work fine. They can call > both external numbers and any Cisco extensions attached to CCM. > > Calls from CCM to Asterisk fail without any notification in Asterisk (and > I DID have this working at one point, but I suspect that our Telco may have > pooched the config somehow, since they're in the process of connecting us to > another CCM site). > > I have verified: Media Termination point exists, Calling Search Space > exists, Trunk is properly defined (uLaw 711, UDP, ip address & port, etc), > and a route pattern exists to take calls to the right trunk. > > The system will let me complete the dialing sequence to the Asterisk > server, but as soon as I enter the last digit I get a busy signal. > > Thoughts anyone? > > Here's my sip.conf if that helps... > > [callman] > type=peer > context=incoming > insecure=very > host=(ip of my call manager server) > disallow=all > allow=ulaw > allow=alaw > nat=no > canreinvite=yes > qualify=yes > > Thanks! Aaron >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080313/cd5168ee/attachment.htm