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TEHO/IPIPGW configuration assistance

will.alvord
Contributor
Contributor

I'm testing some TEHO configs in lab (for redundancy/disaster recovery, not really for toll-bypass). I have 2 2811's on the same network segment, each with a single POTS. I configured an h.323 gateway to a ccm 4.1 cluster for the sole purpose of placing test calls and testing inbound POTS delivery.

The purpose of my testing is to pass a call to router1 with 2 dial-peers - pref 1 routes to router2, pref 2 routes out the POTS on router1. I have the same configs in place on router2. I haven't gotten around to adding RSVP configs because I'm not getting RTP on all call legs.

From a phone registered to ccm, I place a test call to router1. The pref 1 dial-peer routes the call to router2. Router1 shows 2 active call legs - both h.323 (ccm to gateway & gateway to gateway). Router2 shows 2 active call legs - 1 telephony and 1 h.323 (router1 to router2 & POTS). The call rings to a 2nd test phone via the PSTN. I answer, and get no RTP either way and the call disconnects after 7-8secs every time.

I've attached the configs for both routers. Both are running c2800nm-adventerprisek9_ivs-mz.124-24.T.bin.

Any assistance will be greatly appreciated.

thanks,

Will

8 Replies 8

will.alvord
Contributor
Contributor

bump

If your RTP is failing you probably have a codec negotiation issues.

Also, some CUBE guidelines:

SIP-H323:

voice service voip

h323

emptycapability

SIP-SIP

voice service voip

sip

midcall-signaling passthru

H323-H323

voice service voip

h323

h225 connect-passthru

hth,

nick

I've tried adding the h225 connect-passthru statement to both routers but it's still the same -- no audio and the call disconnects 7-8 secs after answering it.

I can dial either of the 2 POTS (1 to each router) and ring my desk phone with 2 way audio. Right now the 2 dial-peers are setup as pref 1 - ipipgw, pref 2 - pstn. If I flip the preferences (cucm phone - isr - pots), I get 2 way audio.

It's only when I'm routing from cucm - isr 1 - isr 2 - pots (or vice versa) that I get no audio and the call disconnects.

I've only got g711ulaw in the voice class codec ref'd by all voip dial-peers. I found a Cisco page (can't find it now though) which specifically covered 1/no way audio issues, but didn't help me.

I'm not sure where to go from here, what debugs to run, etc.

A debug voice rtp shows this though which I'm sure is a problem:

*May 6 16:36:36.389: voip_rtp_create_session: callID=41, dstCallID=42 laddr=10.255.111.3, lport=17464,raddr=0.0.0.0, rport=0, type=3, sig_tos=3, ip_tos=5

...

...

*May 6 16:36:46.449: voip_rtcp_stop_session: ERROR - Dest IP Addr=0, no RTCP pkt sent

The remote address is all zeroes. Not sure how to fix it though. Any advice is welcome.

thanks,

Will

Hi Will,

It sounds like you're going to need to run a considerable amount of debugs. I suggest a TAC case.

-nick

Roger that. I just opened a TAC case. Thanks for your help!

will

Hi Will,

did you end up with a resolution for this one?

think I'm having exact same issue.

Cheers,

Tim

I don't recall that this was ever resolved.  Shortly after opening the tac case, my contract with that company ended.  I doubt anything was done on this after I left since it was sort of a pet project of mine.  I'll try to reach out to the remaining engineer who I believe is still working for that company and post again if he ever got it working.

thanks,

will

have you try these command?

voice service voip
h323
h245 caps mode restricted --> hidden command
h245 tunnel disable --> hidden command

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