[cisco-voip] CME: SIP Trunk with Asteriks

[Syed Khalid Ali]

Thank you for the reply.

What about SIP normalization? Can it be used to alter the INVITE message. I
am trying change the INVITE contents from " From: "Phone-5" <sip:"406100"@
124.109.50.25>; "  to
"406100 <sip:406100 at 124.109.50.25 <sip%3A406100 at 124.109.50.25>>".

voice class sip-profiles 1
request INVITE sip-header From modify "(<.*:)(.*@)" "\1\"406100\"@"
request INVITE sip-header From modify "(From: ".*") (<.*>)" "From:
406100"\2"

the rule-1 is working fine and it changes the <sip:.....> part perfectly and
I did test it on the IOS. but the starting part "From: Phone-X" how to
change, I could come up with rule-2 but could get the chance to test on the
IOS.

One more thing, is there any RE-INVITE message available in SIP? i tried to
find this message type but could not able locate. The book that I am
referencing are:
1- Cisco Press, SIP Trunking
2- McGraw, SIP, Controlling Convergent Networks

lastly what IOS release to use on GNS/Dynamips for learning and testing SIP
normalization. Right now, we have 12.4(15) Adv. IP Services. Do we need the
other feature set release.

Regards,
Khalid



On Fri, Oct 22, 2010 at 5:57 PM, Dennis Heim <Dennis.Heim at cdw.com> wrote:

> your dn’s should have no-reg on them, and the only thing that usually
> registers is your e.164 number.
>
>
>
> Dennis Heim
> Network Voice Engineer
> CDW  Advanced Technology Services
> 11711 N. Meridian Street, Suite 225
> Carmel, IN  46032
>
> 317.569.4255 Office/Home Office
> 317.569.4201 Fax
> 317.694.6070 Cell
>
> dennis.heim at cdw.com
> cdw.com/content/solutions/unified-communications/<http://www.cdw.com/content/solutions/unified-communications/>
>
>
>
>
>
> *From:* cisco-voip-bounces at puck.nether.net [mailto:
> cisco-voip-bounces at puck.nether.net] *On Behalf Of *Syed Khalid Ali
> *Sent:* Friday, October 22, 2010 7:43 AM
> *To:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] CME: SIP Trunk with Asteriks
>
>
>
> The Service Provider also sent me these traces:
>
> Oct 22 16:47:48 NOTICE[13901]: chan_sip.c:11385 handle_request_register:
> Registration from '<sip:60001 at 124.109.50.25>' failed for '172.21.5.134' -
> ACL error (permit/deny)
>
> Oct 22 16:47:48 NOTICE[13901]: chan_sip.c:11385 handle_request_register:
> Registration from '<sip:60002 at 124.109.50.25>' failed for '172.21.5.134' -
> ACL error (permit/deny)
>
> Oct 22 16:47:48 NOTICE[13901]: chan_sip.c:11385 handle_request_register:
> Registration from '<sip:60003 at 124.109.50.25>' failed for '172.21.5.134' -
> ACL error (permit/deny)
>
> Oct 22 16:47:48 NOTICE[13901]: chan_sip.c:11385 handle_request_register:
> Registration from '<sip:60004 at 124.109.50.25>' failed for '172.21.5.134' -
> ACL error (permit/deny)
>
> Oct 22 16:47:48 NOTICE[13901]: chan_sip.c:11385 handle_request_register:
> Registration from '<sip:60005 at 124.109.50.25>' failed for '172.21.5.134' -
> ACL error (permit/deny)
>
> Oct 22 16:47:48 NOTICE[13901]: chan_sip.c:11385 handle_request_register:
> Registration from '<sip:60006 at 124.109.50.25>' failed for '172.21.5.134' -
> ACL error (permit/deny)
>
>
>
>
>
> On Fri, Oct 22, 2010 at 5:31 PM, Syed Khalid Ali <
> syed.khalid.khursheed at gmail.com> wrote:
>
> hi,
>
> I am trying to establish SIP trunk with a local service provide. They are
> using Asteriks as the SIP Proxy. The INVITE is going from my CME (SCCP)
> extension to the SIP Server which in turns reject outbound the call with
> Forbidden 403 message.
>
> Right now I am scratching my head what I missed during configuration.
> Attached is the configuration and debug for reference.
>
> --
> Thanks,
> Syed Khalid Ali
>
>
>
>
> --
> Thanks,
> Syed Khalid Ali
>



-- 
Thanks,
Syed Khalid Ali
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