CME Sip-ua outbound calls problems with messagenet VOIP provider

_Mike_ · ‎05-26-2016

Hi all,

I have an existing old cme 7.1 based on a UC540 with this topology connections: cme____firewall____adslrouter___SIPprovider

Once bought the phone number, the provider has sent me the following parameters: URI, Password and SIP Server (sip.messagenet.it:5061)

I configured a sip-ua profile to permit all my cisco phones to place voip calls via "messagenet" provider.

Doing tests with my CIPC softphone, I can receive calls configuring the command "number 112 secondary 5406042832 no-reg primary" under the ephone-dn, but trying to make outbound calls I got always the fast busy tone.

What am I doing wrong, can anyone help me?

Here is my "debug ccsip messages" and the cme conf. in attach, thank you in advance.

035188: May 25 10:34:41.107: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:9347xxxxx9@sip.messagenet.it:5061 SIP/2.0
Date: Wed, 25 May 2016 10:34:41 GMT
Call-Info: <sip:192.168.2.254:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: "05xxxxxxx2" <sip:112@sip.messagenet.it>;tag=EDCEE9AC-26E6
Allow-Events: telephone-event
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Remote-Party-ID: "IFI Tests" <sip:112@192.168.2.254>;party=calling;screen=no;privacy=off
Cisco-Guid: 455890549-563810790-2880145261-787559590
Timestamp: 1464172481
Content-Length: 303
User-Agent: Cisco-SIPGateway/IOS-12.x
To: <sip:9347xxxxx9@sip.messagenet.it>
Contact: <sip:112@192.168.2.254:5060>
Expires: 180
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 1F5A5458-219B11E6-ABB0876D-2EF134A6@192.168.2.254
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK3E33B2040
CSeq: 101 INVITE
Max-Forwards: 70

v=0
o=CiscoSystemsSIP-GW-UserAgent 2357 9567 IN IP4 192.168.2.254
s=SIP Call
c=IN IP4 192.168.2.254
t=0 0
m=audio 18002 RTP/AVP 0 8 18 100
c=IN IP4 192.168.2.254
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194

035189: May 25 10:34:41.131: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 407 Proxy Authentication Required
From: "05xxxxxxx2" <sip:112@sip.messagenet.it>;tag=EDCEE9AC-26E6
To: <sip:9347xxxxx9@sip.messagenet.it>;tag=ac2f3091de46227321b1af258e10b75e-3c84
Call-ID: 1F5A5458-219B11E6-ABB0876D-2EF134A6@192.168.2.254
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK3E33B2040;rport=56043;received=79.58.183.41
CSeq: 101 INVITE
Proxy-Authenticate: Digest realm="sip.messagenet.it", nonce="V0WA61dFf78BY5IQto2DBbbF2rNhKuIkCKrgqfvMxS4snvgPYozQYGCmlkOC", qop="auth"
Server: sip.messagenet.it SIP Proxy
Content-Length: 0
Warning: 392 212.97.59.76:5061 "Noisy feedback tells: pid=24839 req_src_ip=79.58.183.41 req_src_port=56043 in_uri=sip:9347xxxxx9@sip.messagenet.it:5061 out_uri=sip:9347xxxxx9@sip.messagenet.it:5061 via_cnt==1"

035190: May 25 10:34:41.135: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:9347xxxxx9@sip.messagenet.it:5061 SIP/2.0
Date: Wed, 25 May 2016 10:34:41 GMT
From: "05xxxxxxx2" <sip:112@sip.messagenet.it>;tag=EDCEE9AC-26E6
Allow-Events: telephone-event
Content-Length: 0
To: <sip:9347xxxxx9@sip.messagenet.it>;tag=ac2f3091de46227321b1af258e10b75e-3c84
Call-ID: 1F5A5458-219B11E6-ABB0876D-2EF134A6@192.168.2.254
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK3E33B2040
CSeq: 101 ACK
Max-Forwards: 70

035191: May 25 10:34:41.135: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:9347xxxxx9@sip.messagenet.it:5061 SIP/2.0
Date: Wed, 25 May 2016 10:34:41 GMT
Call-Info: <sip:192.168.2.254:5060>;method="NOTIFY;Event=telephone-event;Duration=2000"
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
From: "05xxxxxxx2" <sip:112@sip.messagenet.it>;tag=EDCEE9AC-26E6
Allow-Events: telephone-event
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Remote-Party-ID: "IFI Tests" <sip:112@192.168.2.254>;party=calling;screen=no;privacy=off
Cisco-Guid: 455890549-563810790-2880145261-787559590
Timestamp: 1464172481
Content-Length: 303
User-Agent: Cisco-SIPGateway/IOS-12.x
Proxy-Authorization: Digest username="5406042832",realm="sip.messagenet.it",uri="sip:9347xxxxx9@sip.messagenet.it:5061",response="bb7956866aec9477983c390c07e5d74a",nonce="V0WA61dFf78BY5IQto2DBbbF2rNhKuIkCKrgqfvMxS4snvgPYozQYGCmlkOC",cnonce="51E8C7B7",qop=auth,algorithm=md5,nc=00000001
To: <sip:9347xxxxx9@sip.messagenet.it>
Contact: <sip:112@192.168.2.254:5060>
Expires: 180
Content-Disposition: session;handling=required
Content-Type: application/sdp
Call-ID: 1F5A5458-219B11E6-ABB0876D-2EF134A6@192.168.2.254
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK3E33C2583
CSeq: 102 INVITE
Max-Forwards: 70

v=0
o=CiscoSystemsSIP-GW-UserAgent 2357 9567 IN IP4 192.168.2.254
s=SIP Call
c=IN IP4 192.168.2.254
t=0 0
m=audio 18002 RTP/AVP 0 8 18 100
c=IN IP4 192.168.2.254
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194

035192: May 25 10:34:41.159: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Auth/From Username Matching Policy Failed
From: "05xxxxxxx2" <sip:112@sip.messagenet.it>;tag=EDCEE9AC-26E6
To: <sip:9347xxxxx9@sip.messagenet.it>;tag=98df93dff07c6bc0cd4e22344f9aa5a7.101e
Call-ID: 1F5A5458-219B11E6-ABB0876D-2EF134A6@192.168.2.254
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK3E33C2583;rport=56043;received=79.58.183.41
CSeq: 102 INVITE
Server: sip.messagenet.it SIP Proxy
Content-Length: 0
Warning: 392 212.97.59.76:5061 "Noisy feedback tells: pid=24845 req_src_ip=79.58.183.41 req_src_port=56043 in_uri=sip:9347xxxxx9@sip.messagenet.it:5061 out_uri=sip:9347xxxxx9@sip.messagenet.it:5061 via_cnt==1"

035193: May 25 10:34:41.163: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:9347xxxxx9@sip.messagenet.it:5061 SIP/2.0
Date: Wed, 25 May 2016 10:34:41 GMT
From: "05xxxxxxx2" <sip:112@sip.messagenet.it>;tag=EDCEE9AC-26E6
Allow-Events: telephone-event
Content-Length: 0
To: <sip:9347xxxxx9@sip.messagenet.it>;tag=98df93dff07c6bc0cd4e22344f9aa5a7.101e
Call-ID: 1F5A5458-219B11E6-ABB0876D-2EF134A6@192.168.2.254
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK3E33C2583
CSeq: 102 ACK
Max-Forwards: 70

Deepak Mehta · ‎05-26-2016

Hey Mike,

One thing i can see clearly in the logs is authentication is failing.

035192: May 25 10:34:41.159: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 403 Auth/From Username Matching Policy Failed>>>>>>>>>>>>

So may want to check with Service provider and confirm on the username.thanks

_Mike_ · ‎05-27-2016

Thank you Deepak,

but maybe it's not a credentials problems because I tried the same credential with a softphone like X-Lite and and the calls works fine.

Vivek Batra · ‎05-27-2016

Please check the FROM field of INVITE message;

From: "05xxxxxxx2" <sip:112@sip.messagenet.it>;tag=EDCEE9AC-26E6

So you're sending '112' as calling party number. I guess you need to send correct DDI number provided by your provider. Please apply desired translation rules and confirm.

- Vivek

_Mike_ · ‎05-27-2016

Thank you Vivek,

assuming the following info, will I need a translation rules for every internal extension?

Will I apply my rule under the dial-peer voip?

Are you able to provide me an example of trans.rule for 112?

Thanks again.

uc540#sh sip-ua register status
Line          peer           expires(sec) registered   P-Associated-URI
============ ============= ============ =========== ================
0*                                1           73            no
101                               20001       82            no
102                               20002       83            no
103                               20003       83            no
104                               20004       80            no
105                               20005       83            no
106                               20006       80            no
107                               20007       80            no
108                               20008       80            no
109                               20009       81            no
110                               20010       78            no
111                               20011       81            no
301                               20013       81            no
302                               20014       81            no
303                               20015       81            no
5xxxxxx2                      20019       2340          yes
A50...                            20016       82            no
A51...                            20017       82            no

Vivek Batra · ‎05-27-2016

Yes, you can apply translations to outbound voip dial peer.

Can you please share the DDI numbers range provided by your provider?

- Vivek

_Mike_ · ‎05-27-2016

Thank you, I have a single number (05431796482) and the URI is 5406042832

Vivek Batra · ‎05-27-2016

So if 05431796482 is the main number, apply translations to respective outbound dial peer which changes any number to 05431796482.

- Vivek

_Mike_ · ‎05-27-2016

Can you provide the cli commands Vivek?

Vivek Batra · ‎05-27-2016

voice translation-rule 10

rule 1 /.*/ /05431796482/

voice translation-profile 10

translate calling 10

Then apply the following command to outbound dial peer;

translation-profile outgoing 10

- Vivek

_Mike_ · ‎06-10-2016

Hi Vivek, sorry for the late, I've finally applied your suggestion configuring the translation rule but still not working, during a call always the fast busy tone is occuring. I share the new debug ccsip messages in attach,

have you any idea?

Thanks in advance.

_Mike_ · ‎06-13-2016

Anyone can help to fix this problem?

Thx

MOHIT SINGH · ‎06-14-2016

Hi Mike,

Can you compare the SIP REGISTER sent by X-lite with what CME is sending to PSTN? You can collect packet capture using wireshark on PC.

It will be nice if you can share the capture with us so that we can check it as well. Start the capture before you register X-lite with provider.

_Mike_ · ‎06-17-2016

Hi Mohit,

thank you for your help, in attach can find the SIP packet capture for X-Lite client.

Do you note something interesting comparing it with the cme logs?

Thanks again.

_Mike_ · ‎06-22-2016

Nobody can help??

Thx