<html><head><style type='text/css'>p { margin: 0; }</style></head><body><div style='font-family: Verdana; font-size: 10pt; color: #000000'>Thanks again Nick. This is great information. I'm gonna have to read and digest.<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: "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 5:18:43 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>You could do it two different ways:<br><br>Pass everything to the other router, let it deal with the details.<br>This would require for you to use g711 for all calls, or configure a<br>transcoder on the remote router since CUE doesn't support g729. DTMF<br>relay is fairly inconsequential as long as it matches on both ends.<br>Under ccn subsystem sip on the CUE you can change the DTMF method, and<br>I usually feel better about the config if the trunking side DTMF<br>matches the CUE DTMF.<br><br>Make a specific dial peer for CUE directly from router A. You could<br>use g729 for all voice calls if its a concern, and just g711 for CUE<br>without the necessity of a transcoder - unless calls are transferred<br>between router A/B to CUE (likely). You can configure the DTMF<br>setting directly from the router A dial peer to CUE, and it would be<br>SIP. I believe there may be problems with SRST A sending calls to CUE<br>B because CUE B will assume that SRST A is SRST B. Not 100% on that,<br>but I seem to remember some problems from doing that (especially if<br>there are overlapping extensions).<br><br>I would say best practices are:<br>-Configure SIP between the two routers. This simplifies the transfer to CUE.<br>-Configure a transcoder on both routers (if they both have CUE,<br>otherwise just the router with CUE) if you're going to use g729.<br>-If you don't configure a transcoder, use g711.<br>-Try to have dtmf methods match between your SRST trunk and CUE. For<br>h323-sip your option is rtp-nte.<br>-Point your dial peer for SRST A to CUE B to SRST B and make sure the<br>correct allow-connections is configured.<br><br>In general, yes you will agree on a dtmf and codec type, just like<br>with SIP trunks. The alternative is to configure a voice-class codec<br>with all the options you could expect, and to do debugging to<br>determine the DTMF method in use.<br><br>-nick<br><br>On Sat, Mar 27, 2010 at 4:40 PM, Lelio Fulgenzi <lelio@uoguelph.ca> wrote:<br>> Thanks Nick. There were four things I needed to do to make things work (got<br>> some help from the forums):<br>><br>> allow h323 to sip connections<br>> add the codec on the inbound call leg<br>> add the codec on the outbound call leg<br>> add the dtmf-relay on the outbound call leg<br>><br>> I'm totally on-board for making any changes on the terminating router, but I<br>> 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<br>> for this type of information to be passed pre-configuration, i.e. what codec<br>> and dtmf relay is needed?<br>><br>> I'm also wondering what I might be breaking with the "dtmf-relay" command.<br>> And what I might break if I add other commands. For example, modem<br>> passthrough.<br>><br>> I'm guessing I might have to make a more specific outbound dial-peer for<br>> just those 3 unity express ports on the other router.<br>><br>> ________________________________<br>> 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>> plus normal stuff<br>> !<br>><br>> ________________________________<br>> on the originating router:<br>><br>> !<br>> dial-peer voice 11111 voip<br>> plus normal stuff<br>> dtmf-relay h245-alphanumeric<br>> codec g711ulaw<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>> ----- 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<br>> 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<br>>> 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<br>>> 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<br>>> 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<br>>> of<br>>> style. It even showed matches. I can post the full debug next week, but<br>>> 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>><br></div></body></html>