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15
Helpful
7
Replies

Outgoing Call not allowed

josesantos2
Level 1
Level 1

‎07-03-2017 01:06 PM - edited ‎03-17-2019 10:41 AM

Hello All,

I'm trying to figure out why some extension is not able to place external calls if they are using the same level of access and configuration than others. Trying to find the problem I got the following logs:

008123: Jul 3 15:25:37.093 BRZ: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:0@10.80.18.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.80.17.153:5060;branch=z9hG4bK598e13f7
From: "VANESSA FONTES" <sip:8224@10.80.18.1>;tag=f0b2e577af13007376c1ab42-4bc50eaa
To: <sip:0@10.80.18.1>
Call-ID: f0b2e577-af130016-52f70779-11ef2524@10.80.17.153
Max-Forwards: 70
Date: Mon, 03 Jul 2017 18:25:39 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7821/10.3.1
Contact: <sip:9480-177A@10.80.17.153:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO
Remote-Party-ID: "VANESSA FONTES" <sip:8224@10.80.18.1>;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,sdp-anat,norefersub,resource-priority,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-7.0.0,X-cisco-xsi-8.5.1
Allow-Events: kpml,dialog
Authorization: Digest username="pool25",realm="all",uri="sip:0@10.80.18.1;user=phone",response="38ee8545942d3a0cabfaec1770780115",nonce="A8E6F64D05F8E820",cnonce="4384cb15",qop=auth,nc=00000002,algorithm=MD5
Content-Length: 324
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 17125 0 IN IP4 10.80.17.153
s=SIP Call
t=0 0
m=audio 24042 RTP/AVP 0 8 116 18 101
c=IN IP4 10.80.17.153
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

008124: Jul 3 15:25:37.097 BRZ: //17401/D81F2585B849/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.80.17.153:5060;branch=z9hG4bK598e13f7
From: "VANESSA FONTES" <sip:8224@10.80.18.1>;tag=f0b2e577af13007376c1ab42-4bc50eaa
To: <sip:0@10.80.18.1>
Date: Mon, 03 Jul 2017 18:25:37 GMT
Call-ID: f0b2e577-af130016-52f70779-11ef2524@10.80.17.153
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.6.2.T1
Session-ID: 00000000000000000000000000000000;remote=f7abfbb7d0005460a9f5d37ad8a3267b
Content-Length: 0


008125: Jul 3 15:25:37.141 BRZ: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
NOTIFY sip:0@10.80.18.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.80.17.153:5060;branch=z9hG4bK475bc759
To: <sip:0@10.80.18.1>;tag=3BBA6E30-502
From: <sip:8224@10.80.17.153>;tag=f0b2e577af1300742c2ec950-0c5387f6
Call-ID: D81FC245-5F5311E7-B84EC726-2F536D10@10.80.18.1
Date: Mon, 03 Jul 2017 18:25:39 GMT
CSeq: 1000 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:9480-177A@10.80.17.153:5060;transport=udp>
Authorization: Digest username="pool25",realm="all",uri="8224",response="6e9401c9916e8de9c7fa93c81fa18601",nonce="A8E6F64D05F8E820",cnonce="370ca064",qop=auth,nc=00000003,algorithm=MD5
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 0

008126: Jul 3 15:25:37.509 BRZ: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
NOTIFY sip:0@10.80.18.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.80.17.153:5060;branch=z9hG4bK30b8d3d0
To: <sip:0@10.80.18.1>;tag=3BBA6E30-502
From: <sip:8224@10.80.17.153>;tag=f0b2e577af1300742c2ec950-0c5387f6
Call-ID: D81FC245-5F5311E7-B84EC726-2F536D10@10.80.18.1
Date: Mon, 03 Jul 2017 18:25:40 GMT
CSeq: 1001 NOTIFY
Event: kpml
Subscription-State: active; expires=7200
Max-Forwards: 70
Contact: <sip:9480-177A@10.80.17.153:5060;transport=udp>
Authorization: Digest username="pool25",realm="all",uri="8224",response="f9f169d66ecd38193407fc7ffbb504cd",nonce="A8E6F64D05F8E820",cnonce="2dc0cb28",qop=auth,nc=00000004,algorithm=MD5
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE
Content-Length: 205
Content-Type: application/kpml-response+xml
Content-Disposition: session;handling=required

<?xml version="1.0" encoding="UTF-8"?>
<kpml-response xmlns="urn:ietf:params:xml:ns:kpml-response" version="1.0" code="200" text="OK" suppressed="false" forced_flush="false" digits="0" tag="Backspace OK"/>
008127: Jul 3 15:25:37.513 BRZ: %SIP_SUPPSERV-6-TRANSFER: Transfer from pool(25) to 00 is not allowed
008128: Jul 3 15:25:37.513 BRZ: //17401/D81F2585B849/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.80.17.153:5060;branch=z9hG4bK598e13f7
From: "VANESSA FONTES" <sip:8224@10.80.18.1>;tag=f0b2e577af13007376c1ab42-4bc50eaa
To: <sip:0@10.80.18.1>;tag=3BBA6FD0-1544
Date: Mon, 03 Jul 2017 18:25:37 GMT
Call-ID: f0b2e577-af130016-52f70779-11ef2524@10.80.17.153
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-15.6.2.T1
Reason: Q.850;cause=21
Session-ID: f7abfbb7d0005460a9f5d37ad8a3267b;remote=0ab26473e9815c67ae62c4c0ccf2b677
Content-Length: 0


008129: Jul 3 15:25:37.569 BRZ: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:0@10.80.18.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.80.17.153:5060;branch=z9hG4bK598e13f7
From: "VANESSA FONTES" <sip:8224@10.80.18.1>;tag=f0b2e577af13007376c1ab42-4bc50eaa
To: <sip:0@10.80.18.1>;tag=3BBA6FD0-1544
Call-ID: f0b2e577-af130016-52f70779-11ef2524@10.80.17.153
Max-Forwards: 70
Date: Mon, 03 Jul 2017 18:25:40 GMT
CSeq: 101 ACK
Content-Length: 0


008130: Jul 3 15:25:37.569 BRZ: //17401/D81F2585B849/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x0x23850488
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : 8224
Called Number : 00
Source IP Address (Sig ): 10.80.18.1
Destn SIP Req Addr:Port : 10.80.17.153:5060
Destn SIP Resp Addr:Port : 10.80.17.153:5060
Destination Name : 10.80.17.153

008131: Jul 3 15:25:37.569 BRZ: //17401/D81F2585B849/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711ulaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 0 (tx), 0 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 10.80.18.1
Source IP Port (Media): 30970
Destn IP Address (Media): 10.80.17.153
Destn IP Port (Media): 24042
Orig Destn IP Address:Port (Media): [ - ]:0

008132: Jul 3 15:25:37.569 BRZ: //17401/D81F2585B849/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 21
Disconnect Cause (SIP) : 403

Labels:
0 Helpful
7 Replies 7

Slavik Bialik
Level 7
Level 7

‎07-03-2017 01:12 PM

Hi,

What is the device type of this extension? And is it registered to CUCM or CME with CUBE? You didn't mention anything about it.

Because it looks like that this device type is configured to dial only with kpml, so for every digit you dial it sends a NOTIFY SIP message with the digit. But the SIP trunk expects to get all the digits at once in order to send it to the PSTN, but it only gets the first digit, therefore you won't be able to dial out. This is my first impression at least. But if you can also post your GW configuration, it can be nice and answer my questions above it can help more.

josesantos2
Level 1
Level 1
In response to Slavik Bialik

‎07-03-2017 01:38 PM

Hi,

Sorry but I'm new user of Cisco CME, I was used to work with Avaya. I think we are using CME without Cube.

Please see attached the config file from the router.

Preview file
18 KB
0 Helpful

Slavik Bialik
Level 7
Level 7
In response to josesantos2

‎07-03-2017 02:01 PM

I miss Avaya ;)

Anyway, I'm not working a lot with CME, but it is very strange for me to see that a Cisco 7821 over SIP that is dialing with KPML and NOTIFY's.

The other 7821 phones you have don't make this issue? If not, can you verify that a phone that is working fine, and the one that isn't are running the same firmware (you can see it in the 'About phone' option in the settings menu of the phone). Also you can try to factory reset it, by doing the following:

http://www.ipwithease.com/how-to-reset-the-cisco-7821ip-phone-to-the-factory-default/

josesantos2
Level 1
Level 1
In response to Slavik Bialik

‎07-03-2017 02:04 PM

All other extension, the same model, are working fine.  I tried to remove and create again these extensions and after that they are able to place outgoing calls for a while. After few hours or minutes, the problem returns.

0 Helpful

Dennis Mink
VIP Alumni
VIP Alumni
In response to josesantos2

‎07-04-2017 03:24 AM

Can you add the debug ccsip and debug dial peer output of a failed call and add to the post, thanks

Please rate if helpful

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josesantos2
Level 1
Level 1
In response to Dennis Mink

‎07-04-2017 06:27 AM

Hello,


Please find attached the log file. The extension I had problem is 8224.

Preview file
87 KB
0 Helpful

Dennis Mink
VIP Alumni
VIP Alumni
In response to josesantos2

‎07-04-2017 03:50 PM

Jose,

your call fails because of a 403 FORBIDDEN, so what Slavik is saying about digit by digit analysis is true.

Can you confirm the firmware version after a factory reset. Thanks

015322: Jul  4 10:13:48.432 BRZ: //-1/734C9E7A82E7/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
From: "VANESSA FONTES" <sip:8224@10.80.18.1>;tag=f0b2e577af1300914db46d48-72e36814
To: <sip:0@10.80.18.1>

Please remember to rate useful posts, by clicking on the stars below.

0 Helpful
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