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Beginner

Voice Services VOIP SIP Commands, VG224 vs VG350

Can someone explain to me what the command "midcall-signaling passthru media-change" does on a VG350?

Also, on a VG224, the command "midcall-signaling passthru" doesn't allow  you to add "media-change" to the command.

Is "midcall-signaling passthru" on a vg224 the same as "midcall-signaling passthru media-change" on a VG350?

1 ACCEPTED SOLUTION

Accepted Solutions
Highlighted

Re: Voice Services VOIP SIP Commands, VG224 vs VG350


@chuck47172 wrote:

Thanks for your patience with me, much appreciated.

I’m new at this and I’m trying to take a config from a production VG350 and make it work on a VG224 that is on an older version of IOS that doesn’t support same features that my VG350 does.

I should probably upgrade the VG224 IOS but for reasons beyond my control, I’m not allowed to.

Having said that, can you explain to me in simple terms what the bind statement actually does for the call processing and why I need it?

Also, regarding the dial peers, does the following look like the fix you are suggesting?

dial-peer voice 100 voip
 description To/From CUCM
 huntstop
 destination-pattern .T
 session protocol sipv2
 session target ipv4: (CUCM SUB 1 IP Address)
 session transport udp
 incoming uri via CUCM
 incoming uri from CUCM
 voice-class codec 1
 voice-class sip options-keepalive up-interval 20 down-interval 10 retry 3
 dtmf-relay rtp-nte
 no vad

dial-peer voice 200 voip
 description To/From CUCM
 huntstop
 destination-pattern .T
 session protocol sipv2
 session target ipv4: (CUCM SUB 2 IP Address)
 session transport udp
 incoming uri via CUCM
 incoming uri from CUCM
 voice-class codec 1
 voice-class sip options-keepalive up-interval 20 down-interval 10 retry 3
 dtmf-relay rtp-nte
 no vad

dial-peer voice 300 voip
 description To/From CUCM
 huntstop
 destination-pattern .T
 session protocol sipv2
 session target ipv4: (CUCM SUB 3 IP Address)
 session transport udp
 incoming uri via CUCM
 incoming uri from CUCM
 voice-class codec 1
 voice-class sip options-keepalive up-interval 20 down-interval 10 retry 3
 dtmf-relay rtp-nte
 no vad

The bind command controls what interface that will be used for the source for communication. Normally you would use the interface that you connect with the network. Loopback interface does work, but you should in such case assign an IP address to it.

Your dial peers looks okay, apart from that you should not have huntstop on 100 and 200 as that would stop the router from using a second or third option CM if the previous are unreachable for some reason. If you want to control the order of connection with your CMs you should also add preference to the dial peers, lower is more favoured than a higher number.

 

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8 REPLIES 8
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Re: Voice Services VOIP SIP Commands, VG224 vs VG350

This is a function that you’d normally use on an SBC, aka a CUBE. There should be little reason for this to be used on an VG used for analog ports.

For more information about this function have a look at this url, https://www.cisco.com/c/en/us/td/docs/ios-xml/ios/voice/cube/configuration/cube-book/voi-cube-midcall-reinvite.html

 

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Highlighted
Beginner

Re: Voice Services VOIP SIP Commands, VG224 vs VG350

Thanks for the reply.

 

Does this section of code look correct for SIP on a vg224?

I'm wondering if the bind commands and early offer are needed.

Also will this work without using "voice class server-group" on the dial peer?

I'm on IOS version Version 15.1(4)M10 which I don't believe supports server groups.

 

interface Loopback0
 no ip address

voice service voip ip address trusted list ipv4 (CUCM PUB IP ADDRESS) ipv4 (CUCM SUB IP ADDRESS) dtmf-interworking rtp-nte allow-connections sip to sip signaling forward unconditional fax protocol t38 version 0 ls-redundancy 0 hs-redundancy 0 fallback pass-through g711ulaw sip bind control source-interface Loopback 0 bind media source-interface Loopback 0 early-offer forced voice class uri CUCM sip host ipv4: (CUCM PUB IP ADDRESS) host ipv4: (CUCM SUB IP ADDRESS) voice class codec 1 codec preference 1 g711ulaw dial-peer voice 100 voip description To/From CUCM huntstop destination-pattern .T session protocol sipv2 session transport udp incoming uri via CUCM incoming uri from CUCM voice-class codec 1 voice-class sip options-keepalive up-interval 20 down-interval 10 retry 3 dtmf-relay rtp-nte no vad sip-ua retry invite 2 retry response 2 timers trying 400
Highlighted

Re: Voice Services VOIP SIP Commands, VG224 vs VG350

Are there any specific reason for why you want to setup your analog voice gateway with SIP? Normally it would be configured for SCCP controlled ports.

 

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Beginner

Re: Voice Services VOIP SIP Commands, VG224 vs VG350

It works best with our fax server.


Highlighted

Re: Voice Services VOIP SIP Commands, VG224 vs VG350

The bind statements you have is set to use loopback and your loopback interface doesn’t have an IP address. So I would say it’s not really doing anything for you.

Server group is not needed on the dial peer, but if you don’t have it you’ll need to use the older style of configuration with session target and have multiple dial peers to be able to define more than one target IP.

 

Please rate all useful posts
Highlighted
Beginner

Re: Voice Services VOIP SIP Commands, VG224 vs VG350

Thanks for your patience with me, much appreciated.

I’m new at this and I’m trying to take a config from a production VG350 and make it work on a VG224 that is on an older version of IOS that doesn’t support same features that my VG350 does.

I should probably upgrade the VG224 IOS but for reasons beyond my control, I’m not allowed to.

Having said that, can you explain to me in simple terms what the bind statement actually does for the call processing and why I need it?

Also, regarding the dial peers, does the following look like the fix you are suggesting?

dial-peer voice 100 voip
 description To/From CUCM
 huntstop
 destination-pattern .T
 session protocol sipv2
 session target ipv4: (CUCM SUB 1 IP Address)
 session transport udp
 incoming uri via CUCM
 incoming uri from CUCM
 voice-class codec 1
 voice-class sip options-keepalive up-interval 20 down-interval 10 retry 3
 dtmf-relay rtp-nte
 no vad

dial-peer voice 200 voip
 description To/From CUCM
 huntstop
 destination-pattern .T
 session protocol sipv2
 session target ipv4: (CUCM SUB 2 IP Address)
 session transport udp
 incoming uri via CUCM
 incoming uri from CUCM
 voice-class codec 1
 voice-class sip options-keepalive up-interval 20 down-interval 10 retry 3
 dtmf-relay rtp-nte
 no vad

dial-peer voice 300 voip
 description To/From CUCM
 huntstop
 destination-pattern .T
 session protocol sipv2
 session target ipv4: (CUCM SUB 3 IP Address)
 session transport udp
 incoming uri via CUCM
 incoming uri from CUCM
 voice-class codec 1
 voice-class sip options-keepalive up-interval 20 down-interval 10 retry 3
 dtmf-relay rtp-nte
 no vad
Highlighted

Re: Voice Services VOIP SIP Commands, VG224 vs VG350


@chuck47172 wrote:

Thanks for your patience with me, much appreciated.

I’m new at this and I’m trying to take a config from a production VG350 and make it work on a VG224 that is on an older version of IOS that doesn’t support same features that my VG350 does.

I should probably upgrade the VG224 IOS but for reasons beyond my control, I’m not allowed to.

Having said that, can you explain to me in simple terms what the bind statement actually does for the call processing and why I need it?

Also, regarding the dial peers, does the following look like the fix you are suggesting?

dial-peer voice 100 voip
 description To/From CUCM
 huntstop
 destination-pattern .T
 session protocol sipv2
 session target ipv4: (CUCM SUB 1 IP Address)
 session transport udp
 incoming uri via CUCM
 incoming uri from CUCM
 voice-class codec 1
 voice-class sip options-keepalive up-interval 20 down-interval 10 retry 3
 dtmf-relay rtp-nte
 no vad

dial-peer voice 200 voip
 description To/From CUCM
 huntstop
 destination-pattern .T
 session protocol sipv2
 session target ipv4: (CUCM SUB 2 IP Address)
 session transport udp
 incoming uri via CUCM
 incoming uri from CUCM
 voice-class codec 1
 voice-class sip options-keepalive up-interval 20 down-interval 10 retry 3
 dtmf-relay rtp-nte
 no vad

dial-peer voice 300 voip
 description To/From CUCM
 huntstop
 destination-pattern .T
 session protocol sipv2
 session target ipv4: (CUCM SUB 3 IP Address)
 session transport udp
 incoming uri via CUCM
 incoming uri from CUCM
 voice-class codec 1
 voice-class sip options-keepalive up-interval 20 down-interval 10 retry 3
 dtmf-relay rtp-nte
 no vad

The bind command controls what interface that will be used for the source for communication. Normally you would use the interface that you connect with the network. Loopback interface does work, but you should in such case assign an IP address to it.

Your dial peers looks okay, apart from that you should not have huntstop on 100 and 200 as that would stop the router from using a second or third option CM if the previous are unreachable for some reason. If you want to control the order of connection with your CMs you should also add preference to the dial peers, lower is more favoured than a higher number.

 

Please rate all useful posts

View solution in original post

Highlighted
Beginner

Re: Voice Services VOIP SIP Commands, VG224 vs VG350

Thanks sooooooooooo much for your help!
This is the best communication I have had in the Cisco Community!