[Asterisk-Users] SIP Connection to a Cisco AS5xxx gateway

[Emilio Panighetti]

Thanks.

I looked at your config and it looks very simple (I would use 
dial-peers instead of plar, but that's not the issue).

It seems that the gateway is sending the codec information 
mime-encoded, which is what I see from an ethereal trace.
On the other hand, Asterisk doesn't seem to understand that. The Cisco 
gateway is sending RPT packets, but apparently, Asterisk is ignoring 
them.

Asterisk, in debug mode, shows me the following errors:

     -- Executing Dial("SIP/1234-a56a", "SIP/PSTN/4567") in new stack
     -- Called PSTN/4567
Oct 12 18:46:09 NOTICE[1107327920]: chan_sip.c:2617 process_sdp: 
Content is 'multipart/mixed;boundary=uniqueBoundary', not 
'application/sdp'
     -- SIP/PSTN-934f answered SIP/1234-a56a
     -- Attempting native bridge of SIP/1234-a56a and SIP/PSTN-934f
Oct 12 18:46:09 WARNING[1109470128]: rtp.c:460 ast_rtp_read: RTP Read 
too short (20, expecting 77524)
RFC3389: 1 bytes, level 8...
Oct 12 18:46:12 NOTICE[1109470128]: rtp.c:289 process_rfc3389: RFC3389 
support incomplete.  Turn off on client if possible
   == Spawn extension (from-sip, 4567, 1) exited non-zero on 
'SIP/1234-a56a'

Your gateway seems to have a comparable IOS. I don't think Cisco would 
change the SIP stack from platform to plaform.

I'm using a CVS version of asterisk (Asterisk 
CVS-HEAD-10/11/04-01:13:29). Should I go back and use a 1.0 release? 
Everything else seems to be working so far.

On Oct 12, 2004, at 5:36 PM, Asterisk wrote:

> Here is what works for me. It is currently working and in service on 
> an MC3810.
>
> plar is needed so incoming calls ring an extension in asterisk.  
> extension 102 sends call to my IVR root. Please remember to configure 
> default gateway. This especially important if you have nat specified 
> in asterisk.
>
> This is for an MC3810, but you should be able to get enough out of it 
> to make your AS5300 work.
>
> Jojo