sccp to sip in uc500

[motasemkhater]

Dear all,

I am trying to make a sip call from uc500 to ASTRIK which iam connected to it throw site to site vpn.

i was doing somthing stupied by trying to dial from my 7940 phone without change it to sip ..i have tried alot of cisco guides to convert 7940 to sip but nothing worked out may be its the UC500.. attached is my running config ..

in my phone when i check status from Network configuration i have VERSION 8.0(5.0)

i follow this link http://www.cisco.com/univercd/cc/td/doc/product/voice/its/cmeadm/cmeinstl.htm#wp1070512

and do every step to change SCCP to SIP

but nothing work out... i dont know if i need to downgrade from SCCP 8.x to 5.x then upgrade to sip 4.x ... and is the firmware in call manager same as call manager express or uc500

[paolo bevilacqua]

Hi,

you don't need the 7940 to be converted to SIP in order to call from UC500 to asterisk and vice-versa. Cisco phones work better in SCCP when used with CME.UC/CCM, etc.

Let us know what you want to do exactly and we can give you indications how to do that.

[motasemkhater]

I have a site to site vpn , in my LAN i have UC500 and in other site there is a Asterisk with sip server i can ping this server , i want to register with this server so i can call there extentions, for test i have 7940 in my site with ext 618 and in the Asterisk site they have 277 ext , so i make dial-peer 3001 and sip-ua configuration and open all ports in my WAN port

please check my running configuration

[paolo bevilacqua]

Hi, do "debug ccsip message" with "term mon" to see why the call is failing. If it is a registration problem, remove the registration from asterisk (recommended), or configure sip-register under sip-ua so that extension will also register to asterisk.

[motasemkhater]

ok sir i ll try it and give you my feed back today, but i what do u mean by [remove the registration from asterisk (recommended)] and how to do it

and there is no command sip-register under sip-ua ??

[paolo bevilacqua]

I meant, configure asterisk so that no registration is required.

I was referring to the registrar server command.

[motasemkhater]

Hi Sir,

I cant get yoyr point ,!!

my customer is the one that have Asterisk and want me to give him a Cisco solution , by connection to Asterisk Sip server so he can call Asterisk site ,

They have done a test using Linksys and sip phones and every thing work fine with them .

we offer the uc500 solution .

first is this doable?

second what i understand till now that the UC500 will register with sip server (astersik) and allow calls to sip users .

is that right ?

all configuration that i have made is the sip-ua and dial -peer ,, what else i need sir

dial-peer voice 3001 voip

destination-pattern 277

session protocol sipv2

session target ipv4:192.168.253.35

dtmf-relay rtp-nte

codec g711ulaw

no vad

!

!

no dial-peer outbound status-check pots

sip-ua

authentication username xxx password xxx

sip-server ipv4:192.168.253.35

[ashok_boin]

Hi,

If your goal is to communicate between your SCCP phones (Cisco callmanager) & SIP phones (Asterisk), then you just need to create SIP trunk interaces to your SIP proxy server in Cisco callmanager. And CCM will take care of signaling conversion.

No need to convert your phone signaling to SIP.

Please follow this link...

http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/4_0_1/ccmsys/a08sip.html

Regards...

-Ashok.

[paolo bevilacqua]

Ashok, he is using CME, not CM, so the link provided does not apply.

However the logic is correct (configure sip trunk). Then if doesn't work you have to find why with "debug ccsip message" and "term mon".

[ashok_boin]

Oh ok. I thought he was talking about Unified communications manager 5.0. Anyways, thanks for the correction.

Regards...

-Ashok.

[motasemkhater]

thanks all,

so here is the ccsip debug and my last sh run

and when i run debug ccsip error i get nothing

[paolo bevilacqua]

Hi,

you get:

Received:

SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 192.168.253.35:5060;branch=z9hG4bK0489;received=192.168.0.121

From: "618" <618>;tag=12AF0C-18C6

To: <277>;tag=as1b41f6e1

This means asterisk doesn't know anything about number 277. Configure asterisk accordingly.

[motasemkhater]

thanks sir ,, i have check ith them and they call my ext,,and give them SIP/2.0 400 Bad Request - 'Invalid Host' comming from the Cisco

this the debug :-

14:05:08.641785 IP (tos 0x68, ttl 252, id 223, offset 0, flags [none], proto: UDP (17), length: 474) 192.168.0.121.5060 > 192.168.253.35.5060: SIP, length: 446

SIP/2.0 400 Bad Request - 'Invalid Host'

Via: SIP/2.0/UDP 192.168.253.35:5060;branch=z9hG4bK3effce61;rport

From: "Emmanuel Sovaridis" <277>;tag=as32434bc0

To: <618>;tag=996CA8-20E

Date: Mon, 23 Jun 2008 12:02:17 GMT

Call-ID: 5ce5a4c276d55dd8529ea6463912d2e1@192.168.253.35

Server: Cisco-SIPGateway/IOS-12.x

CSeq: 102 INVITE

Allow-Events: telephone-event

Reason: Q.850;cause=100

Content-Length: 0

[paolo bevilacqua]

Please take the complete "debug ccsip mesage" also for incoming call.

Note DP 3001 should have codec g711u, not g771a, and also "incoming called-number ."

[motasemkhater]

ok .. here is my sh run and the debug ...thank you very much

i just want to be sure that we dont need in sip-ua (register command or using dns not ip )

please check my sh run i did what you asked about DP, and incoming called number