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SIP trunk on a Cisco 2811 - problem

hi guys... wondering if someone could help a bit...

i've been trying to set up a sip trunk to an alt. provider on a Cisco 2811 router with a 124-20.T1 IOS.

the number translation has been tested and the digits go through as they are supposed to.

when i try to phone out the call is cut off almost immediately and I get a msg in my debug output:

SIP/2.0 403 Do not use late offer-answer model

When I phone in from my mobile phone I get the IP Phone ringing, I get the ringback on my mobile… but when I IP Phone picks up my mobile phone continues to get the ringback and the call can not complete…

I have a CCM 4.1 in the background and it does not do any translations. All the translation is done on the 2811 which is a h323 gateway…

If you need any more info just say the word…

For reference:

I got this working with the same router and IOS, the same alt. provider. The only difference is that the working config has a CME on the router and no CCM in the back.

If anyone has got any ideas on how to work this out I would be very gratefull…

2 ACCEPTED SOLUTIONS

Accepted Solutions

Re: SIP trunk on a Cisco 2811 - problem

Hi,

Two things:

1) For the incoming call, it appears that they may not be receiving our messages. Check for potential firewall / ACL issues.

2) For the outgoing call, we send an INVITE without SDP. Generally this only happens if you're using H323 on the other side of the connection. If your provider does not support late offer media, then you will need to make sure that the incoming H323 leg is fast start. To do this in CUCM, you must have 'MTP Required' checked with an available MTP in the MRGL for the H323 gateway. Then towards the bottom, check 'Enable Outbound Fast Start'.

hth,

nick

Re: SIP trunk on a Cisco 2811 - problem

For the outgoing calls you must still not be sending H323 fast start. This is very easy to check - the first INVITE you send to the provider should have an SDP - it will have "Content Length: " and then a whole bunch of lines like a=, m=, t=. That's the SDP.

If you're not sending the SDP in the invite it's because you're not getting fast start from CUCM.

You may need to check your codec used in outbound fast start regarding your ISDN problem.

-nick

4 REPLIES

Re: SIP trunk on a Cisco 2811 - problem

Hi,

Two things:

1) For the incoming call, it appears that they may not be receiving our messages. Check for potential firewall / ACL issues.

2) For the outgoing call, we send an INVITE without SDP. Generally this only happens if you're using H323 on the other side of the connection. If your provider does not support late offer media, then you will need to make sure that the incoming H323 leg is fast start. To do this in CUCM, you must have 'MTP Required' checked with an available MTP in the MRGL for the H323 gateway. Then towards the bottom, check 'Enable Outbound Fast Start'.

hth,

nick

New Member

Re: SIP trunk on a Cisco 2811 - problem

thnx for the help nick

i tried doing that with no success.

i did get inbound calls to work by enabling fast-start on inbound calls though.

outbound calls still fail due to the late-offer answer model and i can't seem to get this figured out... when i enable outbound fast-start even my ISDN calls fail...

i have an MTP in the MRGL. The CCM is the MTP.

and i also have an ASA 5520 but i put a permit ip any any for testing and it isn't blocking anything...

Re: SIP trunk on a Cisco 2811 - problem

For the outgoing calls you must still not be sending H323 fast start. This is very easy to check - the first INVITE you send to the provider should have an SDP - it will have "Content Length: " and then a whole bunch of lines like a=, m=, t=. That's the SDP.

If you're not sending the SDP in the invite it's because you're not getting fast start from CUCM.

You may need to check your codec used in outbound fast start regarding your ISDN problem.

-nick

New Member

Re: SIP trunk on a Cisco 2811 - problem

thanks guys... works like a charm... the inbound fast start solved the inbound call problem... the outbound fast-start the other way with a lil playing with the codecs...

:)

thanks again for your help.

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