CallManager Express - intermitent inbound calls / no DTMF

[glacious]

I have Viatalk setup on CME, and all outbound calls are working just fine.

Inbound calls work only 'right' after I add the line:

registrar dns:newyork.vtnoc.net expires 3600

It will then work for a couple minutes, I see the SIP OPTION requests come in from Viatalk in the ccSIP debugs on CME. CME replies with a 200 ok/ACK and all is well.

Usually around the 3rd or 4th OPTION message, I stop seeing them, and I can no longer make outbound calls. The OPTION messages come in once every 60 seconds when working normally.

If I do a 'no registrar' and then add the registrar back in, the OPTION messages will come in for another 2-4 minutes, then stop again.

Turning on UDP debugs, I can see 5 UDP packets come in after the OPTION never makes it (thus my side never sends an ACK/200 OK). The tech said they will send 5 in a row if they miss the 200 OK. They just show up as UDP, not appearing in the SIP debugs.

Any idea what can be causing this? I've forwarded all required ports, tried it with no ports forwarded, put the CME as the DMZ address in my router (Linksys running DD-WRT), all yield the same results.


Also, DTMF does not work. With the defaults, nothing, with it set to RFC 2833, nothing. Waiting for them to set it to 'Auto'.


One last hurdle and I'll be keeping Viatalk for a good while, my first long call sounded great today, happy with the quality so far.


Thanks everyone,

- Bill

[bbrindle]

Hi Bill,

Even though you've put your CME in the DMZ and forwarded the ports your problem sounds very much like a NAT issue with the UDP timeout on your inbound ports. Any way you could post the debug info you have? Also for kicks try adjusting the registrar timeout for less than 5 minutes and see if that makes a difference.

Brian

[glacious]

Brian,

Right now it is Cable modem --- Linksys (DD-WRT firmware) --- 2811 (CME)

There are no ports forwarded anymore, still had the same issue.

I changed my config to the below for sip-ua:

mwi-server dns:sanfrancisco-1.vtnoc.net expires 200 port 5060 transport udp unsolicited
registrar dns:sanfrancisco-1.vtnoc.net expires 200
sip-server dns:sanfrancisco-1.vtnoc.net

Placed a call it worked. (as usual)
Waited 5 or 6 minutes, placed a call. (worked, usually wouldn't)
Waited 20 minutes, placed a call. (worked, never would have before)

Result: Adjusting registrar timeout (expire) to 200 seconds makes it work perfectly, definitely seems as if the UDP session was timing out (probably on the router's side, even tho sip option messages make it to keep it alive, they eventually stop making it). This may however give a small (1-2 second max) service interruption while it re-registers every 3 minutes and 20 seconds, but only if I am making an outbound call during that 1-2 second period. Established calls shouldn't be affected.

It looks like I am keeping ViaTalk (wonder if I can use you as a referrer after I joined?)!

I really appreciate your input Brian, and thank you so much. I just hope ViaTalk doesn't complain that I register too often now :)



On a side note, DTMF works fine if I call ViaTalk's support number, or Comcast, etc... If I call voicemail, it does not. I had them change DTMF to RFC 2833, now to auto on their end.

dial-peer voice 15 voip
description Viatalk Voicemail
destination-pattern *123
session protocol sipv2
session target dns:sanfrancisco-1.vtnoc.net
dtmf-relay rtp-nte
codec g711ulaw

With or without the dtmf-relay line I get the same results. I know they use inband only for their VM, so 2833 shouldn't work anyway, but without the dtmf-relay option at all, it should be pure inband.



Thanks again Brian!

-Bill

[bbrindle]

Bill,

Try doing a "debug ccsip all" and look to see what the Negotiated DTMF relay is showing or at least make sure it's negotiated as inband. If it's showing as rtp-nte then your inband is being converted to RFC2833.

Brian

[glacious]

Brian,

With dtmf-relay rtp-nte taken out of the *123 dial peer, I see:

payload_type=0, codec_bytes=160, codec=g711ulaw, dtmf_relay=inband-voice

I also see a few lines after it: *Jun 21 04:06:16.991: //43/9764AC1D803F/SIP/Info/sipSPISetDTMFRelayMode: Set DSP for dtmf-relay = CC_CAP_DTMF_RELAY_INBAND_VOICE_AND_OOB

So it looks like it is going for true in-band and not 2833. I don't have any DSPs installed (gonna have to pickup a PVDM module or just toss in the T1 and use those resources). Without DSP resources, I don't think true in-band is possible, but I thought the IP phones can use their built-in DSP resources, maybe not for DTMF I guess.

Thanks again Brian,

Bill

[cstatton]

Bill,

The standard timeout on sip-ua is five minutes of inactivity. You can increase the limit under sip-ua by using the "timers connection aging #" where # is the number of minutes.

[glacious]

Right now everything is working 100% (except DTMF for ViaTalk Voicemail). I think the UDP session is getting timed out on my router as the SIP OPTION messages being sent by ViaTalk for some reason to not keep it alive. Increasing the connection aging timer on the router would ensure that the CME would not close the connection, but my Linksys router (/w DD-WRT) will still be killing it.

My current workaround as stated above is to set the registrar expire timer to 200 seconds, so it pretty much creates a new connection every 200 seconds, well before the Linksys thinks it dies.

Thanks again for the input, hopefully the next firmware fixes some additional firewall issues in DD-WRT, either way it is still a kick-ass firmware, I highly recommend, if not for its QoS settings alone.


-Bill

[glacious]

anyone have any success with DTMF to VT voicemail with callmanager express?

using rtp-nte for the VM dial-peer with no luck.

DTMF to all other IVR's work just fine.