03-07-2020 06:49 AM
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?
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03-08-2020 07:42 AM
@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.
03-07-2020 09:07 AM
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
03-07-2020 11:29 AM
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
03-07-2020 11:53 AM
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.
03-07-2020 12:27 PM
03-08-2020 12:37 AM
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.
03-08-2020 06:45 AM
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
03-08-2020 07:42 AM
@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.
03-08-2020 08:39 AM
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