[cisco-voip] router to router (SCCP/h323 to SIP) calls don't work in SRST
Lelio Fulgenzi
lelio at uoguelph.ca
Sat Mar 27 16:40:35 EDT 2010
Thanks Nick. There were four things I needed to do to make things work (got some help from the forums):
• allow h323 to sip connections
• add the codec on the inbound call leg
• add the codec on the outbound call leg
• add the dtmf-relay on the outbound call leg
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.
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?
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.
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.
on the terminating router:
!
voice service voip
allow-connections h323 to sip
!
dial-peer voice 11112 voip
codec g711ulaw
plus normal stuff
!
on the originating router:
!
dial-peer voice 11111 voip
plus normal stuff
dtmf-relay h245-alphanumeric
codec g711ulaw
!
---
Lelio Fulgenzi, B.A.
Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
(519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
Cooking with unix is easy. You just sed it and forget it.
- LFJ (with apologies to Mr. Popeil)
----- Original Message -----
From: "Nick Matthews" <matthnick at gmail.com>
To: "Lelio Fulgenzi" <lelio at uoguelph.ca>
Cc: "cisco-voip voyp list" <cisco-voip at puck.nether.net>
Sent: Saturday, March 27, 2010 4:15:10 PM GMT -05:00 US/Canada Eastern
Subject: Re: [cisco-voip] router to router (SCCP/h323 to SIP) calls don't work in SRST
Technically you should be able to point it at the other CME, and that
is what I would do. Make sure on router B you have the proper
incoming dial peer, outgoing dial peer, and that you have
allow-connections for h323-to-sip, etc.
-nick
On Fri, Mar 26, 2010 at 10:16 PM, Lelio Fulgenzi <lelio at uoguelph.ca> wrote:
> ok, after some more reading, it looks like the default inbound dial peer
> won't work with SIP calls.
>
> which makes more sense, the configuration should really happen on the
> terminating router, not the originating router.
>
> thanks to Ed for some pointers.
>
> ---
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> Cooking with unix is easy. You just sed it and forget it.
> - LFJ (with apologies to Mr. Popeil)
>
>
> ----- Original Message -----
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> To: "cisco-voip voyp list" <cisco-voip at puck.nether.net>
> Sent: Friday, March 26, 2010 7:59:55 PM GMT -05:00 US/Canada Eastern
> Subject: Re: [cisco-voip] router to router (SCCP/h323 to SIP) calls don't
> work in SRST
>
> ya know, I think I just answered my own question after sending this off....
>
> i'm guessing I need a more specific dial-peer on the far end router which
> more closely matches the dial-peer on local router.
>
> so, something like this? the question is, can i use the router address or
> should i use the CUE ip address?
>
> hmmm, something to try later
>
> ________________________________
> Router A:
> !
> dial-peer voice 37063 voip
> description Cisco Unity Express AutoAttendant (Default)
> destination-pattern 37063
> session protocol sipv2
> session target ipv4:10.104.13.66
> dtmf-relay sip-notify
> codec g711ulaw
> no vad
> !
> dial-peer voice 11111 voip
> description Wild Card to vgw-jnhn-b
> destination-pattern [1234567]....
> session target ipv4:10.104.13.202
> !
> dial-peer voice 37000 voip
> description SIP Wild Card to vgw-jnhn-b
> destination-pattern 37...
> session protocol sipv2
> session target ipv4:10.104.13.202 (OR CUE IP address?)
> dtmf-relay sip-notify
> codec g711ulaw
> no vad
> !
> ________________________________
>
>
> ----- Original Message -----
> From: "Lelio Fulgenzi" <lelio at uoguelph.ca>
> To: "cisco-voip voyp list" <cisco-voip at puck.nether.net>
> Sent: Friday, March 26, 2010 7:53:10 PM GMT -05:00 US/Canada Eastern
> Subject: [cisco-voip] router to router (SCCP/h323 to SIP) calls don't work
> in SRST
>
> Does anyone know if there is anything special you have to do to make calls
> from an SCCP phone on one SRST router to a SIP endpoint on another SRST
> router work?
>
> Here's what I have and can do:
>
> two routers in SRST mode
> all phones register properly, some to one router, some to another
> I can make a call from phone A on router A to phone B on router B (and vice
> versa)
> I can make a call from phone A on router A to Unity Express A on router A
> and be transferred to phone B on router B
>
> I can NOT place a call from phone A on router A to Unity Express B on router
> B.
>
> I'm pretty sure Router B is getting the call, because a "debug voice
> dialpeer all" started spewing out stuff on Router B like it was going out of
> style. It even showed matches. I can post the full debug next week, but just
> thought there would be a quick(?) answer.
>
> I think these are the relevant configs, but will post more if needed:
>
> ________________________________
> Router A:
> !
> dial-peer voice 37063 voip
> description Cisco Unity Express AutoAttendant (Default)
> destination-pattern 37063
> session protocol sipv2
> session target ipv4:10.104.13.66
> dtmf-relay sip-notify
> codec g711ulaw
> no vad
> !
> dial-peer voice 11111 voip
> description Wild Card to vgw-jnhn-b
> destination-pattern [1234567]....
> session target ipv4:10.104.13.202
> !
> ________________________________
> Router B:
> !
> dial-peer voice 37073 voip
> description Cisco Unity Express AutoAttendant (Default)
> destination-pattern 37073
> session protocol sipv2
> session target ipv4:10.104.13.70
> dtmf-relay sip-notify
> codec g711ulaw
> no vad
> !
> dial-peer voice 11111 voip
> description Wild Card to vgw-jnhn-a
> destination-pattern [1234567]....
> session target ipv4:10.104.13.201
> !
> ________________________________
>
>
>
>
> ---
> Lelio Fulgenzi, B.A.
> Senior Analyst (CCS) * University of Guelph * Guelph, Ontario N1G 2W1
> (519) 824-4120 x56354 (519) 767-1060 FAX (JNHN)
> ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^
> Cooking with unix is easy. You just sed it and forget it.
> - LFJ (with apologies to Mr. Popeil)
>
>
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