<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'>Thanks Nick. There were four things I needed to do to make things work (got some help from the forums):<br><ul><li>allow h323 to sip connections</li><li>add the codec on the inbound call leg</li><li>add the codec on the outbound call leg</li><li>add the dtmf-relay on the outbound call leg</li></ul>I'm totally on-board for making any changes on the terminating router, but I am curious about making the changes on the originating router. <br><br>Let's say the two routers belonged to different organizations...is it normal for this type of information to be passed pre-configuration, i.e. what codec and dtmf relay is needed?<br><br>I'm also wondering what I might be breaking with the "dtmf-relay" command. And what I might break if I add other commands. For example, modem passthrough.<br><br>I'm guessing I might have to make a more specific outbound dial-peer for just those 3 unity express ports on the other router.<br><br><div style="margin-left: 40px;"><hr style="width: 100%; height: 2px;">on the terminating router:<br><br>!<br>voice service voip <br> allow-connections h323 to sip<br>!<br>dial-peer voice 11112 voip<br> codec g711ulaw<br><span style="font-style: italic;"> plus normal stuff</span><br>!<br><br><hr style="width: 100%; height: 2px;">on the originating router:<br><br>!<br>dial-peer voice 11111 voip<br> <span style="font-style: italic;">plus normal stuff</span><br> dtmf-relay h245-alphanumeric<br> codec g711ulaw<br>!<br><hr style="width: 100%; height: 2px;"><br><br></div>---<br>Lelio Fulgenzi, B.A.<br>Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>Cooking with unix is easy. You just sed it and forget it. <br> - LFJ (with apologies to Mr. Popeil)<br><br><br>----- Original Message -----<br>From: "Nick Matthews" <matthnick@gmail.com><br>To: "Lelio Fulgenzi" <lelio@uoguelph.ca><br>Cc: "cisco-voip voyp list" <cisco-voip@puck.nether.net><br>Sent: Saturday, March 27, 2010 4:15:10 PM GMT -05:00 US/Canada Eastern<br>Subject: Re: [cisco-voip] router to router (SCCP/h323 to SIP) calls don't work in SRST<br><br>Technically you should be able to point it at the other CME, and that<br>is what I would do. Make sure on router B you have the proper<br>incoming dial peer, outgoing dial peer, and that you have<br>allow-connections for h323-to-sip, etc.<br><br>-nick<br><br>On Fri, Mar 26, 2010 at 10:16 PM, Lelio Fulgenzi <lelio@uoguelph.ca> wrote:<br>> ok, after some more reading, it looks like the default inbound dial peer<br>> won't work with SIP calls.<br>><br>> which makes more sense, the configuration should really happen on the<br>> terminating router, not the originating router.<br>><br>> thanks to Ed for some pointers.<br>><br>> ---<br>> Lelio Fulgenzi, B.A.<br>> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>> Cooking with unix is easy. You just sed it and forget it.<br>> - LFJ (with apologies to Mr. Popeil)<br>><br>><br>> ----- Original Message -----<br>> From: "Lelio Fulgenzi" <lelio@uoguelph.ca><br>> To: "cisco-voip voyp list" <cisco-voip@puck.nether.net><br>> Sent: Friday, March 26, 2010 7:59:55 PM GMT -05:00 US/Canada Eastern<br>> Subject: Re: [cisco-voip] router to router (SCCP/h323 to SIP) calls don't<br>> work in SRST<br>><br>> ya know, I think I just answered my own question after sending this off....<br>><br>> i'm guessing I need a more specific dial-peer on the far end router which<br>> more closely matches the dial-peer on local router.<br>><br>> so, something like this? the question is, can i use the router address or<br>> should i use the CUE ip address?<br>><br>> hmmm, something to try later<br>><br>> ________________________________<br>> Router A:<br>> !<br>> dial-peer voice 37063 voip<br>> description Cisco Unity Express AutoAttendant (Default)<br>> destination-pattern 37063<br>> session protocol sipv2<br>> session target ipv4:10.104.13.66<br>> dtmf-relay sip-notify<br>> codec g711ulaw<br>> no vad<br>> !<br>> dial-peer voice 11111 voip<br>> description Wild Card to vgw-jnhn-b<br>> destination-pattern [1234567]....<br>> session target ipv4:10.104.13.202<br>> !<br>> dial-peer voice 37000 voip<br>> description SIP Wild Card to vgw-jnhn-b<br>> destination-pattern 37...<br>> session protocol sipv2<br>> session target ipv4:10.104.13.202 (OR CUE IP address?)<br>> dtmf-relay sip-notify<br>> codec g711ulaw<br>> no vad<br>> !<br>> ________________________________<br>><br>><br>> ----- Original Message -----<br>> From: "Lelio Fulgenzi" <lelio@uoguelph.ca><br>> To: "cisco-voip voyp list" <cisco-voip@puck.nether.net><br>> Sent: Friday, March 26, 2010 7:53:10 PM GMT -05:00 US/Canada Eastern<br>> Subject: [cisco-voip] router to router (SCCP/h323 to SIP) calls don't work<br>> in SRST<br>><br>> Does anyone know if there is anything special you have to do to make calls<br>> from an SCCP phone on one SRST router to a SIP endpoint on another SRST<br>> router work?<br>><br>> Here's what I have and can do:<br>><br>> two routers in SRST mode<br>> all phones register properly, some to one router, some to another<br>> I can make a call from phone A on router A to phone B on router B (and vice<br>> versa)<br>> I can make a call from phone A on router A to Unity Express A on router A<br>> and be transferred to phone B on router B<br>><br>> I can NOT place a call from phone A on router A to Unity Express B on router<br>> B.<br>><br>> I'm pretty sure Router B is getting the call, because a "debug voice<br>> dialpeer all" started spewing out stuff on Router B like it was going out of<br>> style. It even showed matches. I can post the full debug next week, but just<br>> thought there would be a quick(?) answer.<br>><br>> I think these are the relevant configs, but will post more if needed:<br>><br>> ________________________________<br>> Router A:<br>> !<br>> dial-peer voice 37063 voip<br>> description Cisco Unity Express AutoAttendant (Default)<br>> destination-pattern 37063<br>> session protocol sipv2<br>> session target ipv4:10.104.13.66<br>> dtmf-relay sip-notify<br>> codec g711ulaw<br>> no vad<br>> !<br>> dial-peer voice 11111 voip<br>> description Wild Card to vgw-jnhn-b<br>> destination-pattern [1234567]....<br>> session target ipv4:10.104.13.202<br>> !<br>> ________________________________<br>> Router B:<br>> !<br>> dial-peer voice 37073 voip<br>> description Cisco Unity Express AutoAttendant (Default)<br>> destination-pattern 37073<br>> session protocol sipv2<br>> session target ipv4:10.104.13.70<br>> dtmf-relay sip-notify<br>> codec g711ulaw<br>> no vad<br>> !<br>> dial-peer voice 11111 voip<br>> description Wild Card to vgw-jnhn-a<br>> destination-pattern [1234567]....<br>> session target ipv4:10.104.13.201<br>> !<br>> ________________________________<br>><br>><br>><br>><br>> ---<br>> Lelio Fulgenzi, B.A.<br>> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1<br>> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)<br>> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^<br>> Cooking with unix is easy. You just sed it and forget it.<br>> - LFJ (with apologies to Mr. Popeil)<br>><br>><br>> _______________________________________________ cisco-voip mailing list<br>> cisco-voip@puck.nether.net<br>> https://puck.nether.net/mailman/listinfo/cisco-voip<br>> _______________________________________________ cisco-voip mailing list<br>> cisco-voip@puck.nether.net<br>> https://puck.nether.net/mailman/listinfo/cisco-voip<br>> _______________________________________________<br>> cisco-voip mailing list<br>> cisco-voip@puck.nether.net<br>> https://puck.nether.net/mailman/listinfo/cisco-voip<br>><br>><br></div></body></html>