CME - SIP Trunk - Can't get incoming calls.

Answered Question
Sep 17th, 2010

Hello


I am having some problems trying to receive calls on my UCME using SIP trunk. I am using a 2801 with UCME and managed to "successfuly" configure it. The fact is that I can place calls from my phone but when I receive calls nothing happens and the caller phone gets a "network busy" or "network failed" notification. I use a simple configuration where :

internet access <--> 2801 <--> Switch <--> 7912 phone


My 2801 configuration is as follows :

version 15.1

voice service voip
gcid
no cti shutdown
callmonitor
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
  registrar server expires max 120 min 60

voice translation-rule 1
rule 1 /.*/ /0975172273/

voice translation-rule 2
rule 1 /^.*/ /404/
!
!
voice translation-profile inclid
translate called 2
!
voice translation-profile outclid
translate calling 1

dial-peer voice 1 voip

translation-profile incoming inclid
translation-profile outgoing outclid
destination-pattern ..........
session protocol sipv2
session target dns:sip5.voip-centrex.net
codec g711ulaw
!
!
sip-ua
credentials username mysiplogin password 7 09051D5C414C0E081B1733 realm sip5.voip-centrex.net
authentication username mysiplogin password 7 155B58595C63323E382802 realm sip5.voip-centrex.net
retry register 10
registrar dns:sip5.voip-centrex.net expires 3600
registrar dns:sip5.voip-centrex.net expires 3600 secondary

telephony-service
xml user test password test 15
max-ephones 8
max-dn 8

ip source-address 192.168.0.251 port 2000
auto assign 1 to 8
load 7910 P00403020214
load 7912 CP7912080003SCCP070409A
load 7960-7940 P00305000301
time-format 24
date-format dd-mm-yy
dialplan-pattern 1 4.. extension-length 3
max-conferences 4 gain -6
call-forward pattern .T
moh Terre.wav
transfer-system full-consult
log table retain-timer 30
log table max-size 100
create cnf-files version-stamp Jan 01 2002 00:00:00

ephone-dn  5  dual-line
number 404
cti watch
[/code]


where 0975172273 is my public SIP phone number and 404 is the 7912 extension.


The SIP parameters appear to be ok given that the account is registered.

Cisco2801#show sip-ua register status
Line                             peer       expires(sec) registered P-Associ-URI


================================ ========== ============ ========== ============

mylogin                          -1         1873         yes


When I place and outgoing call, the called phoned show the 0975xxxxxx number so I guess it's still good.

When I get an incoming call

Received:
INVITE sip:975172273@109.69.196.140:60365 SIP/2.0
Via: SIP/2.0/UDP 212.155.199.204:5060;branch=z9hG4bK155bee4e;rport
Max-Forwards: 70
From: "cellphonenumber" <sip:"cellphonenumber"@212.155.199.204>;tag=as782b22ce
To: <sip:975172273@109.69.196.140:60365>
Contact: <sip:"cellphonenumber"@212.155.199.204>
Call-ID: 4e08276313e481034635cf6026be227d@212.155.199.204
CSeq: 102 INVITE
User-Agent: OpenVoice-Atlantis
Date: Thu, 16 Sep 2010 15:28:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 269


v=0
o=root 238446103 238446103 IN IP4 212.155.199.204
s=Asterisk PBX 1.6.0-rc6
c=IN IP4 212.155.199.204
t=0 0
m=audio 13704 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv


*Sep 16 14:59:00.199: //310/C5B4B91A815C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.155.199.204:5060;branch=z9hG4bK155bee4e;rport
From: "cellphonenumber" <sip:"cellphonenumber"@212.155.199.204>;tag=as782b22ce
To: <sip:975172273@109.69.196.140:60365>
Date: Thu, 16 Sep 2010 14:59:00 GMT
Call-ID: 4e08276313e481034635cf6026be227d@212.155.199.204
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0

*Sep 16 14:59:00.199: //310/C5B4B91A815C/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 212.155.199.204:5060;branch=z9hG4bK155bee4e;rport
From: "cellphonenumber" <sip:"cellphonenumber"@212.155.199.204>;tag=as782b22ce
To: <sip:975172273@109.69.196.140:60365>;tag=5879A0-0
To: <sip:975172273@109.69.196.140:60365>;tag=5879A0-0
Call-ID: 4e08276313e481034635cf6026be227d@212.155.199.204
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=21
Content-Length: 0

*Sep 16 14:59:00.215: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:975172273@109.69.196.140:60365 SIP/2.0
Via: SIP/2.0/UDP 212.155.199.204:5060;branch=z9hG4bK155bee4e;rport
Max-Forwards: 70
From: "cellphonenumber" <sip:"cellphonenumber"@212.155.199.204>;tag=as782b22ce
To: <sip:975172273@109.69.196.140:60365>;tag=5879A0-0
Contact: <sip:"cellphonenumber"@212.155.199.204>
Call-ID: 4e08276313e481034635cf6026be227d@212.155.199.204
CSeq: 102 ACK
User-Agent: OpenVoice-Atlantis
Content-Length: 0


I guess the 403 Forbidden error is the problem but I don't see how to fix it. Any help is welcomed


Thanks.

I have this problem too.
0 votes
Correct Answer by Steven Holl about 3 years 7 months ago

Probably the inbound call is matching dial-peer 0, so the translation doesn't take effect, so the gateway doesn't know how to route the call.

For the original poster, use this to force the inbound dial-peer match:

dial-peer voice 1 voip

incoming called-number .

Read this document like it's going out of style:

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml

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infologic Fri, 09/17/2010 - 05:32

Thank you I checked that link and fixed the ip address trusted list which is now :

Cisco2801#show ip address  trusted list
IP Address Trusted Authentication
Administration State: UP
Operation State:      UP

IP Address Trusted Call Block Cause: call-reject (21)

VoIP Dial-peer IPv4 Session Targets:
Peer Tag        Oper State      Session Target
--------        ----------      --------------

IP Address Trusted List:
ipv4 0.0.0.0 0.0.0.0

But I still have the same problem, incoming calls can't reach my phone. The voice iec syslog shows

%VOICE_IEC-3-GW: C SCRIPTS: Internal Error (No dialpeer match): IEC=1.1.128.11.5.0 on callID 2164 GUID=72B7DA78C19611DF8838BFB4F5810008

So my guess is that there is a translation problem but my translation rules should be ok

malnaima Fri, 09/17/2010 - 06:00

Hello,

It seems that your not facing the "403 Forbidden" message anymore

Now it's a dial-peer issue as per "No dialpeer match" that you posted.

As i can see from your config there is no dial-peer to match the incomming "975172273" number...

I guess your SIP provider is striping the leading zero which is causing the confusion as i can see from your translation - rule 1 /.*/ /0975172273/ -

I think you might want to change your incomming dia-peer and translation rules to match "975172273" whithout the leading zero.

Kindly Rate if info was helpfull

Best Regards,

Mahmoud

infologic Fri, 09/17/2010 - 06:35

Yes the 0 is stripped in the messages I receive my translation rules are as follows :

voice translation-rule 1
rule 1 /.*/ /975172273/

voice translation-rule 2
rule 1 /^.*/ /404/

voice translation-profile inclid
translate called 2

voice translation-profile outclid
translate calling 1


dial-peer voice 1 voip
translation-profile incoming inclid
translation-profile outgoing outclid
destination-pattern ..........
session protocol sipv2
session target dns:sip5.voip-centrex.net
codec g711ulaw

Cisco2801#test voice  translation-rule 1 404
Matched with rule 1
Original number: 404    Translated number: 975172273
Original number type: none      Translated number type: none
Original number plan: none      Translated number plan: none

Cisco2801#test voice  translation-rule 2 1234567890
Matched with rule 1
Original number: 1234567890     Translated number: 404
Original number type: none      Translated number type: none
Original number plan: none      Translated number plan: none

So basicaly, all outgoing calls are translated to 975172273. This rule is sucessful given that this number is shown on the cellphone when called and appears as "unknown" is I remove the translation rule.

The result of the second test shows that any incoming call should be translated to 404 right ?

Should I put

destination-pattern 975172273

on my dial-peer voice 1 voip ?

Anyway thanks for your answers

malnaima Fri, 09/17/2010 - 12:59

Hello,

Yes, you are correct; any incomming number according to this translation should be translated to "404" and i guess this matches an internal DN of yours.

Can you kindly update us if everything works fine now.

Best Regards,

Mahmoud

infologic Sun, 09/19/2010 - 23:52

Situation is still the same, I get the %VOICE_IEC-3-GW: C SCRIPTS: Internal Error (No dialpeer match): IEC=1.1.128.11.5.0 on callID 2164 GUID=72B7DA78C19611DF8838BFB4F5810008 message everytime I was supposed to receive a call. Since the call is detected I don't think it could be a provider problem but more likely a configuration problem. I'm still looking into it, any help is welcome.

Correct Answer
Steven Holl Mon, 09/20/2010 - 07:53

Probably the inbound call is matching dial-peer 0, so the translation doesn't take effect, so the gateway doesn't know how to route the call.

For the original poster, use this to force the inbound dial-peer match:

dial-peer voice 1 voip

incoming called-number .

Read this document like it's going out of style:

http://www.cisco.com/en/US/tech/tk652/tk90/technologies_tech_note09186a008010fed1.shtml

infologic Tue, 09/21/2010 - 06:15

Ok, thanks a lot, the "incoming called-number ."

command fixed it, everything now works fine.

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Posted September 17, 2010 at 1:35 AM
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