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PSTN SIP and Transcoding

Getting very stuck and frustrated with some testing of a new PSTN SIP trunk to replace a large estate of E1's. Hopefully someone far cleverer than me in all things MGR's related can help!

I'am having real trouble getting the SIP trunks to invoke a transcoder and have some MOH issues also which may be related.


The setup for testing...

SIP trunks to telco, have followed their CUCM config guide, early offer - mandatory (insert MTP if needed) and have adjusted some timers

Some IP phones in different regions. Voice calls G729 and region for Faxing G711

MoH from a CUCM server

Hardware universal transcoder on a 2951 router

3 x Software MTP's (1 for 711a, 711u and g729) on the same 2951 router

Have the SIP trunk in a MRGL containing the 5 MRG's, one for each resource mentioned above

For testing, have put the phones in the same MRGL as the SIP trunk.


What works...

Can make calls out to PSTN from IP phones and the codec negotiates to G711 or G729 depending on your phone or fax region

Inbound calls (from most) sources do the same. Telco offer 729 and 711a and 711u so we can do inbound faxing

Can, in the main, place calls on hold, resume, transfer etc


The issues...

1. There is one upstream partner of the telco who only offer G711 codec, my telco then just pass that codec list. If from this provider I call my G711 test phone the call connects no problem. However I cannot get calls into anything on the G729 region. Wiresharking the SIP trunk to the telco I see the call offer (with only g711) and CUCM sends back a 503 message, telco then retries another trunk (we have multiple telco trunks to different cucm subs) and the same thing before the call fails.

It is at this point that I should be invoking my transcoder, so the telco can offer the call in as G711 and I transcode it to become G729 for my phones to work. For some reason, perhaps my own blindness, I cannot get this working. If I offer the same transcoder to another internal resource it works OK so I know the transcoder itself is good.

2. Music on Hold. If for either an inbound or outbound call I place the call on hold I get no MoH at all from CUCM. Looking at the wireshark I see RTP media coming from my MoH server out towards the telco proxies.What makes matters worse, is that for an inbound call after the call is on hold for 15/20 seconds the telco send a bye and drop the call off. However for an outbound call I can place a call to the pstn and leave them on hold (in silence) for as long as needed.

I have tried setting MOH service parameters for  G729 and G711u/a with no luck


Just to give an overview of the topology. I have a ASA firewall between my network and the telco, this is allowing all media ports from my endpoints so there is no SIP inspection happening.

There is no cube, the telco provide me two proxies in their network that all my signalling and RTP media goes to/from

No slow links, QoS trusted end to end, all routing OK

MTP config (sccp groups excluded)

dspfarm profile 4 transcode universal
 description Hardware Transcoder
 codec g729abr8
 codec g729ar8
 codec g711alaw
 codec g711ulaw
 codec pass-through
 codec g729br8
 codec g729r8
 maximum sessions 51
 associate application SCCP
dspfarm profile 1 mtp
 description Soft MTP - G.711ulaw
 codec g711ulaw
 codec pass-through
 maximum sessions software 100
 associate application SCCP
dspfarm profile 2 mtp
 description Soft MTP - G.711alaw
 codec pass-through
 codec g711alaw
 maximum sessions software 100
 associate application SCCP
dspfarm profile 3 mtp
 description Soft MTP - G.729a
 codec g729ar8
 maximum sessions software 200
 associate application SCCP



Any help gratefully received.



Gaurav Purohit


you need to collect call manager detail traces (and perhaps packet captures through CLI). some pointers:

1) you have "use MTP if needed", which means if they send SDP in invite (which they must be with G.711 only), you are going to forward it as it is.   

2) whats the MOH type? if Multicast, is it allowed?


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