SIP call from ITSP to ITSP failing

[nareh84]

hi,

I have a setup of business edition 6000 9.x. There is h323 link between CUCM to GW (2911). and GW is connected to ITSP via SIP Trunk. The outgoing calls from IP phones are successful. But when i dial from my mobile to ip phone, the phone rings but when i attend the call, there is no voice and after some time the call gets disconnected. I looked up the GW traces and found the cause code cause Q.850;cause=47. I also searched this error and found out that there is problem with codec negotiation. I removed codec related command in dial peer. I also checked"mtp required" checkbox (and assign MRGL (software MTP) to Phones and GW) but both incoming and outgoing calls were failing. I also tried changing the Trunk b/w CUCM and GW from H323 to SIP but both incoming and outgoing calls were failing. any ideas??. find the attached config of router and sip trace of GW.

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following queries are just to clear my mind.

1. In case of SIP. ITSP dont know my inside network subnet but still the calls are successfull because GW acts as a Proxy/MTP and accepts the request from  CUCM and reintiate it to ITSP because of which ITSP dont have to know the IP address/network of CUCM/IP Phones. Can we do it with H323?

[nareh84]

hi,

for outgoing..dial-peer 10 is selected (sip-dial-peer).

[Nishant Savalia]

Hey sorry,

but i mean that which incoming dial-peer is being selected when call comes from CUCM to your gateway.

[nareh84]

hi,

incoming dial peer for outoging call is 601

[Nishant Savalia]

Please try by applying below to that dial-peer and check

dial-peer voice 601 voip

incoming called number .

dtmf-relay h245-alphanumeric

[Jorge Armijo]

All problems you've been facing are common on SIP-to-H323 call flows, you may want to use SIP-to-SIP CUBE configuration, it will overcome the DTMF problem and more.

Here you're having issues becuase H323 uses OOB DTMF method, and SIP uses RFC2833 (most of the time), that is like a hibrid DMTF, it is a RTP payload that is send to the ITSP, if you were using SIP-to-SIP you wouldn't have any issues.

HTH

--
Jorge Armijo

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