SIP Trunk Problem

[Philipp Hoeffker]

Hi ,

we use CME 8.6 on 2811 with 4 x BRi 

Now we want an additional sip-trunk.

Register with :

sip-ua
credentials username "hidden" password "hidden" realm fpbx.de
authentication username "hidden" password "hidden" realm fpbx.de
calling-info pstn-to-sip from number set "hidden"
no remote-party-id
registrar 1 dns:fpbx.de expires 3600
sip-server dns:fpbx.de
presence enable

Strange SIP-UA Registration is happen :

--------------------- Registrar-Index 1 ---------------------

Line peer expires(sec) registered P-Associ-URI
================================ ========== ============ ========== ============
0* 30 347 yes
07 20054 345 yes
08 20003 345 yes
09 20004 346 yes
10 20005 346 yes
11 20006 347 yes
12 20007 346 yes
13 20008 346 yes
15 20010 346 yes
16 20011 346 yes
17 20012 346 yes
18 20013 346 yes
19 20014 346 yes
21 20015 346 yes
22 20016 346 yes
23 20017 348 yes
24 20018 347 yes
25 20019 347 yes
26 20020 347 yes
27 20021 347 yes
28 20022 347 yes
29 20023 348 yes
30 20024 347 yes
317 20030 348 yes
40 20026 346 yes
50 20027 346 yes
500 20032 348 yes
51 20028 347 yes
53 20029 348 yes
"username of siptrunk" -1 36 yes

88 20055 348 yes
xxx 20025 348 yes

other Details :

voice service voip
ip address trusted list
ipv4 62.134.52.212 (this is the fpbx.de)
gcid
cti message device-id suppress-conversion
no cti shutdown
callmonitor
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
redirect ip2ip
sip
registrar server expires max 600 min 60
no call service stop

dial-peer voice 800 voip
translation-profile incoming mit-vorwahl
session protocol sipv2
session target dns:fpbx.de
incoming called-number "complete incomming string"
dtmf-relay sip-notify
codec g711ulaw

i cannot dial in or dial out.

Can someone help me ?

Philipp

[Ayodeji Okanlawon]

Your gateway is sending a 404 not found. Which means that the number that was dialled doesnt exist in the gateway

Sent: 
SIP/2.0 404 Not Found

Via: SIP/2.0/UDP 62.134.52.212;branch=z9hG4bK6a7.182f36c5.0,SIP/2.0/UDP 212.82.247.40:7010;branch=z9hG4bK77cbbd48;rport=7010

From: "+49hidden" <sip:+49hidden@fpbx.de>;tag=as74f76623

To: <sip:hidden2@fpbx.de:5060>;tag=91D54F18-1066

Date: Wed, 23 Mar 2016 14:46:08 GMT

Call-ID: 59c2ba8c7cda995508846e7205dea913@fpbx.de

CSeq: 103 INVITE

Allow-Events: telephone-event

Server: Cisco-SIPGateway/IOS-12.x

Reason: Q.850;cause=1

Content-Length: 0

[Philipp Hoeffker]

that was one of the hints i need :-)

[Philipp Hoeffker]

So the imbound works fine, but now my sip provider is rejecting the Call outbound.

The From Header of the sip message is wrong.

The Provider need : username@fpbx.de

but the system is using : username@cmeipaddress

Please Help a secound time.

Philipp

[Tagir Temirgaliyev]

translation-profile incoming mit-vorwahl  ?

[Marcelo Morais]

Hi,

take a look at the following link: CME SIP Trunking Configuration Example.

Take extra care to the following commands:
- incoming called-number
- destination-pattern

Hope this helps

[Philipp Hoeffker]

HI,

thanks i used the the wrong , destination-pattern ....... was not the right command.

But the same time there was the num-exp command configured ...

sins of the past :-)

Philipp

[Marcelo Morais]

Hi Philipp,

 excellent news ... is everything ok now?

Note: if everything is ok, then please mark the question as correct for future references.

Best regards.