Asterisk to CME Calling Issue

[Cerp]

We have merged two small offices that are using different VoIP solutions.  One office is using a 3825 running CME (8.1) and the other office is running FreePBX (Asterisk 2.9).  I believe that I have correctly configured the SIP trunk/Dial Peers so the CME and FreePBX can talk to each other.  I can initiate a call from the CME side to the FreePBX side with no issue.  Calls from a Cisco Phone to a SIP Phone connects and sounds good (two way audio).  However, if I initiate from the Asterisk side to the CME the call fails to connect.  I debugged the CME and placed a packet sniffer and I am seeing a Q.850 Cause Code 65 disconnect error (SIP_UNACCEPTABLE_MEDIA_ERR) coming from the CME.  I am matching an "incoming" dial-peer that I created and all dial-peers are built for g711ulaw, as are all the extensions/trunks on the Asterisk.  Long term we will probably move all SIP phones onto the CME but for now I need to get these sites talking.  Any tips/assistance would be appreciated.  Here is a diagram and a debug capture for further clarification. 

Thanks

Chad

[thisisshanky]

can you post your dialpeer configs please

[Cerp]

Dialpeer snippet

-----------------------------------

dial-peer voice 1 voip

description to Asterisk

destination-pattern 51[1-7].

session protocol sipv2

session target ipv4:10.30.2.2

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 2 voip

description Star codes

destination-pattern *..

session protocol sipv2

session target ipv4:10.30.2.2

codec g711ulaw

no vad

!

dial-peer voice 100 voip

session protocol sipv2

incoming called-number 518.

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

dial-peer voice 101 voip

session protocol sipv2

incoming called-number 519.

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

[thisisshanky]

Have you checked if your Asterisk settings are Codec and DTMF match the Cisco side?

[Cerp]

I'm new to the Asterisk side but the trunk and extensions are set for g711ulaw only.  When I place a packet sniffer between the CME and Asterisk I see the SIP Invite message from the Asterisk is set with g711 ulaw.  Now as to the DTMF settings I'm sure everyhting is set to default, I haven't changed anything there.

[thisisshanky]

Do you have any dial plan pattern commands in your telephony-services...please paste that portion of the config..

here are some additional links that may help stare and compare configs..

http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Express+Integration

HTH

Please rate helpful posts!!

[Cerp]

No dial plan patterns on the Cisco side.  On the asterisk side I have an outbound pattern that matches the 518X and 519X and sends calls out the SIP trunk to the CME.

[thisisshanky]

can you try enabling vad and see if that makes a difference ? and changing it to g729 ..etc..apparently there is some media negotiation problems happening here...

[Cerp]

1) Ok, I enabled/disabled VAD....no change.

2) On the asterisk side I enabled all codecs and then "sniffed" the call setup from the Asterisk to the CME.  On the SIP INVITE message from the Asterisk to the CME, I see all the relevant codecs that the asterisk can support.  The CME is now built with a voice class that can select from g711ulaw, g711alaw, g726-32, and g729.  I should negotiate one of these but instead the CME kicks out a "Media Type Unavailable" and kills the call.

3) A call initiated from the CME to the Asterisk, SIP INVITE message lists g711ulaw, g711alaw, g726-32, and g729.  The Asterisk accepts and negotiates g711ulaw and the call builds.  All of the options appear the same in both INVITE messages so I'm wondering what the CME is choking on when the Asterisk initiates the call.

[lusandi]

Tracey,

I hope you are doing great,

I would like to confirm if you can provide the sniffer captures you are talking about.

Regards,

Luis Sandi

[hzaben]

we need to check what dial-peer is being matched for incoming and the media thats being negotiated .

can you get the following from CME /

debug ccsip messages

debug voice ccapi inout

you've  registered the distination phone to CME over sip or skinny ?

thanks

Haitham

[thisisshanky]

Tracey try putting the incoming called-number patterns at the top (reorder the dial peers)...although i really dont see a reason why the first two dial-peers should be matched since you are really specific on the patterns..except the last digit...worth a try...