(2801/CCME) SIP Trunk with VoIP provider

[Tiago Santos]

Hello,

I'm having trouble configuring the SIP Trunk with my ISP.
Here attached the logs when I try to connect to my ISP.
Note the following line extracted from the logs attached:

SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.3:5060;branch=z9hG4bK2E212C5 To: <sip:898600779@201.12.106.139> From: <sip:231@201.12.106.139>;tag=15A82AC-1379 Call-ID: 89D8BC12-95A11E0-81BFF001-5D9EB1A4@192.168.0.3 CSeq: 101 INVITE Content-Length: 0

192.168.0.3 (CCME)

898600779 (dialer number)

201.12.106.139 (SIP Server ISP)

231 (CCME extension)

From: <sip:231@201.12.106.139> (Is now the CCME should go to the provider <sip: <sip user> @ 201.12.106.139> instead of <sip:231@201.12.106.139> why can not complete the call?)

Thanks for your help!

T.Santos

[Muthurani Lavanya Paneerselvam]

Hi

Pl provide the "show run" from your 2801 CME.

Thanks.

[Tiago Santos]

Hi mpaneers,

Follow the attached file.

Thanks.

[Muthurani Lavanya Paneerselvam]

Hi

Thanks for providing.  Your configuraiton seems to be ok, one thing what you can do is create a dial-peer for "898600779" outbound calls instead of .T

just try routing the full destination called no. thru the outbound dial peer directly to your sip-server ip.

dial-peer voice 10 voip
description Testing
destination-pattern 898600779

redirect ip2ip
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711alaw
no vad

Also check the sip-server resolves the ip-address you have configured under sip-ua.  this you check routing the session target :ipv4 address defined under the dial-peer.

If still you find some issues, Test the voice translation rule and then take out the "debug voip ccapi inout " for one call alone.

Thanks.

[Tiago Santos]

Hi,

I still can not. I noticed that at no time in debug I see Sip Username and Password:
user id: 750495501
password is: 5660

In my case, all connections must go through a single username and password for my ISP.

Thanks.