Sip trunking 3300 to a Cisco CCM

[drsmith45]

Hello Everyone, we have a Mitel 3300 MCD 4.0 connected to a Cisco Call manager using SIP trunks everything works good except when you dial a Cisco extension that routes the Unity voice mail system the call gets rejected because of the G729 codec used by the 3300in the Sip profile I have enabled the option Renegotiate SDP to enforce symmetric code I can not see how on the Mitel side we are using compression G729?or how to change it so we are locked down to use G711 There is a interop from Mitel and Cisco of course it was written when the 3300 was 7.0 and the Call Manager was 6 any Ideas are appreciated ! SIP

 

[Supernova99]

When programming make sure that the SIP trunks to the Cisco are in a non compressed zone e.g. Zone 1 whih is by default uncompressed
I have spent countless hours trying to get Cisco to Mitel to work flawlessly using SIP trunks but there is always something else that seems to break somewhere.
The last customer gave up and installed a Cisco 3800 and we connected to it via Q-sig, the IP part back to the CM was all handled by the CM so the demarkation was easy, and all our issues with voicemail, IVR etc. were solved.

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[drsmith45]

Thank you for your response , I believe you on the hours for trouble shooting the Sip protocol seems to change for every Vendor on their new release
I have the Network element assignment set for zone 1
which then is grayed out in the Sip peer form
from every thing that I can see in programming it should be uncompressed 711 but the traces in the CCM say we are using 729 ?

 

[Supernova99]

you could create another zone that is un compressed and then add this zone ID to the SIP trunks

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[drsmith45]

the issue was resolved by unchecking the “Asserted Identity” field on the SIP trunk page. on the CCM
calls now reroute correctly to the users greeting instead of being dropped because of no reply from the Mitel 3300
Thanks to all the replies

 

[swinstation]

@drsmith45


hi
please tell me where is the arrested Identity?in mitel or CCM?


I have your problem too,when I rerouted cisco user to RAD the cisco user gets busy tone.

 

[drsmith45]

Hello swinstation the option is in the cisco call manager for sip trunks , the issue was when the mitel users dialed a cisco extension and the call rerouted to the unity voice mail we got a fast busy
unchecking asserted identity fixed it

 

[swinstation]

@drsmith45
Hi thanks for your reply,I was in my site and talked with cisco expert and he said this feature dose not exist in cisco.(call manager)
can you tell me where is it?(exactly)

actually i have your problem ,but when a cisco user calls to my RAD directory they get busy tone.I do not know your mentioned feature helps me?

 

[drsmith45]

Hello swinstation below is from the Cisco web site
it is called P-Asserted-Identity that we turned off so the Call manager would not reject our call because it thought we wanted to change the codec and make call uncompressed


New SIP trunk enhancements increase interoperability with many other SIP applications and call agents which also support SIP:
• SIP Trunk Device Identification allows features such as location-based CAC and Media Resource Selection to work for calls initiated by devices behind a SIP Proxy/B2BUA.

• Privacy, P-Asserted-Identity, and P-Preferred-Identity headers introduce standard support (RFC 3323, RFC 3324, and RFC 3325, respectively) to convey calling, ringing, answering, connected identity, and request privacy.

• Early Offer signaling support for G.729, enables initiation of SIP trunk calls that have a preallocated media termination point (MTP) with low-bandwidth codecs.

• Secure Real-Time Transport Protocol (SRTP) over SIP trunk enables encryption and authentication of media over a SIP trunk.

 

[swinstation]

THANK YOU.