No Voice Path over SIP Trunk from G450 Analog Stations or DS1s

[CChambleeJr]

Are there any major differences in the way that G450 media gateway stations and trunks vs. G650 port network stations and trunks handle the RTP when calling out over a SIP trunk? All of our IP codec sets specify G.711MU as the primary, and G.729 as second choice. I can't see where it would be a codec selection issue, but it sure acts like a codec issue. When I do a list trace on the SIP trunk TAC, the major difference I see between a G450 analog station calling out over the trunk and a G650 analog station calling out over the trunk is, on the G450 station test call that has no audio it appears that the DSP being used is one of the DSP resources in the same G450, while the call made from the G650 station that has audio is using one of the med pros in one of the G650s. All of the G450's are assigned to NR1. Now, if I go into the G450 config for the MG that serves the analog station and I set its NR to "2", the test call WILL have audio. The list trace though, shows that a med pro is being used for the media resource. What is the thought process used to troubleshoot this?

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Chuck Chamblee
ACSS (CM) | CTP | A+ ce | Security+ ce

 

[kyle555]

What region are your SIP trunks in?

How does NR1 connect to that region vs NR2?

Let's just say 192.168.1.0 is NR1, 192.168.2.0 is NR2, .3.0 NR3, etc
If your trunks are in NR10, for example, and NR1 is direct to NR10, maybe 192.168.1.0 and 192.168.10.0 can't route
If NR2 has no direct media allowed and proxies through intervening region 3 in 192.168.3.0 to get to NR10, and 192.168.3.0 and .10.0 can route, then maybe that's why the calls pass.

In any case, CM will absolutely reject a call between regions it cannot set up audio between - barring things like IGAR and DPT. If it let you setup dead air, it let you setup audio between two endpoints that cannot communicate.