CME - outgoing calls did not work
woisch
Member Posts: 40 ■■□□□□□□□□
Hi,
ì`m a little bit new to the cme world. I have purchased a nice 2811 Router and want now to
test this in my lab. I have a external voice provider. This is what i have configured at the moment and incoming calls work but outgoing not:
voice translation-rule 3
rule 1 /^00/ /+/ type any international
!
!
voice translation-profile SIP-Outgoing
translate called 3
!
!
dial-peer voice 1 voip
description "INCOMING"
destination-pattern 004912345678912 <my international number>
voice-class codec 1
session protocol sipv2
session target dns:sip.provider.de
incoming called-number .T
no vad
!
dial-peer voice 4 voip
description "OUTGOING"
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target dns:sip.provider.de
dtmf-relay rtp-nte
no vad
!
voice call send-alert
voice call convert-discpi-to-prog
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol pass-through g711ulaw
h323
modem passthrough nse codec g711ulaw
sip
bind control source-interface FastEthernet0/0
session transport tcp
registrar server expires max 600 min 60
redirect contact order best-match
!
voice class codec 1
codec preference 1 g711ulaw
sip-ua
authentication username
<username>
password 7 <password>
no remote-party-id
retry invite 2
retry register 10
retry options 1
timers connect 100
registrar dns:<sip registrar> expires 3600
sip-server dns:<sip server>
host-registrar
dial-peer voice 1 voip
description SIP TRUNK TO PROVIDER
destination-pattern <my number in internataional format> extension-length 4 extension-pattern 1001
session protocol sipv2
session target dns:<sip server>
incoming called-number .T
voice-class codec 1
no vad
telephony-service
no auto-reg-ephone
max-ephones 20
max-dn 40
max-conferences 8 gain -6
transfer-system full-consult
---
incoming (works)
---
The Call Setup Information is:
Call Control Block (CCB) : 0x4A63D688
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : +49xxxxxxxxxxx (my mobile, 11 digits)
Called Number : 0049xxxxxxxxxxx (my home, 11 digits)
Source IP Address (Sig ): 192.168.5.99
Destn SIP Req Addr:Port : <sip provider>:5060
Destn SIP Resp Addr:Port : <sip provieder>:5060
Destination Name : <sip provider>
*Sep 30 22:01:34.571: //3/962E68E78003/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 : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 192.168.5.99
Source IP Port (Media): 18846
Destn IP Address (Media): <provider ip>
Destn IP Port (Media): 40064
Orig Destn IP Address:Port (Media): [ - ]:0
---
outgoing (works not)
---
The Call Setup Information is:
Call Control Block (CCB) : 0x4A63D688
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : +49xxxxxxxxxxx (my home, 11 digits)
Called Number : +49xxxxxxxxxxx (my mobile, 11 digits)
Source IP Address (Sig ): 192.168.5.99
Destn SIP Req Addr:Port : <sip provider>:5060
Destn SIP Resp Addr:Port : <sip provider>:5060
Destination Name : <sip provider>
Sep 30 22:13:21.864: //20/33C2F5638033/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 192.168.5.99
Source IP Port (Media): 18180
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
i have tried different outgoing formats (translation rules). I don`t know why i can`t dial out. Any suggestions?
ì`m a little bit new to the cme world. I have purchased a nice 2811 Router and want now to
test this in my lab. I have a external voice provider. This is what i have configured at the moment and incoming calls work but outgoing not:
voice translation-rule 3
rule 1 /^00/ /+/ type any international
!
!
voice translation-profile SIP-Outgoing
translate called 3
!
!
dial-peer voice 1 voip
description "INCOMING"
destination-pattern 004912345678912 <my international number>
voice-class codec 1
session protocol sipv2
session target dns:sip.provider.de
incoming called-number .T
no vad
!
dial-peer voice 4 voip
description "OUTGOING"
destination-pattern .T
voice-class codec 1
session protocol sipv2
session target dns:sip.provider.de
dtmf-relay rtp-nte
no vad
!
voice call send-alert
voice call convert-discpi-to-prog
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol pass-through g711ulaw
h323
modem passthrough nse codec g711ulaw
sip
bind control source-interface FastEthernet0/0
session transport tcp
registrar server expires max 600 min 60
redirect contact order best-match
!
voice class codec 1
codec preference 1 g711ulaw
sip-ua
authentication username
<username>
password 7 <password>
no remote-party-id
retry invite 2
retry register 10
retry options 1
timers connect 100
registrar dns:<sip registrar> expires 3600
sip-server dns:<sip server>
host-registrar
dial-peer voice 1 voip
description SIP TRUNK TO PROVIDER
destination-pattern <my number in internataional format> extension-length 4 extension-pattern 1001
session protocol sipv2
session target dns:<sip server>
incoming called-number .T
voice-class codec 1
no vad
telephony-service
no auto-reg-ephone
max-ephones 20
max-dn 40
max-conferences 8 gain -6
transfer-system full-consult
---
incoming (works)
---
The Call Setup Information is:
Call Control Block (CCB) : 0x4A63D688
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : +49xxxxxxxxxxx (my mobile, 11 digits)
Called Number : 0049xxxxxxxxxxx (my home, 11 digits)
Source IP Address (Sig ): 192.168.5.99
Destn SIP Req Addr:Port : <sip provider>:5060
Destn SIP Resp Addr:Port : <sip provieder>:5060
Destination Name : <sip provider>
*Sep 30 22:01:34.571: //3/962E68E78003/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 : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 192.168.5.99
Source IP Port (Media): 18846
Destn IP Address (Media): <provider ip>
Destn IP Port (Media): 40064
Orig Destn IP Address:Port (Media): [ - ]:0
---
outgoing (works not)
---
The Call Setup Information is:
Call Control Block (CCB) : 0x4A63D688
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : +49xxxxxxxxxxx (my home, 11 digits)
Called Number : +49xxxxxxxxxxx (my mobile, 11 digits)
Source IP Address (Sig ): 192.168.5.99
Destn SIP Req Addr:Port : <sip provider>:5060
Destn SIP Resp Addr:Port : <sip provider>:5060
Destination Name : <sip provider>
Sep 30 22:13:21.864: //20/33C2F5638033/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 192.168.5.99
Source IP Port (Media): 18180
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
i have tried different outgoing formats (translation rules). I don`t know why i can`t dial out. Any suggestions?
Comments
-
pitviper Member Posts: 1,376 ■■■■■■■□□□Try translating the calling party number to the exact number that is associated with your sip account for the outbound call (looks like 004912345678912). I know some providers (Callcentric is one) will reject a call if the caller-ID doesn’t match your account.CCNP:Collaboration, CCNP:R&S, CCNA:S, CCNA:V, CCNA, CCENT
-
woisch Member Posts: 40 ■■□□□□□□□□Hi Pitviper,
sorry for the delay. I was very busy, now i`m working straight on the problem. At the moment it looks like this. I think i missed a little think But i dont know what:
**CONFIG**
voice call send-alert
voice call convert-discpi-to-prog
voice call carrier capacity active
voice rtp send-recv
!
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
redirect ip2ip
fax protocol pass-through g711alaw
h323
modem passthrough nse codec g711alaw
sip
bind control source-interface FastEthernet0/0
session transport tcp
registrar server expires max 3600 min 3600
redirect contact order best-match
localhost dns:sipgate.de
!
voice class codec 1
codec preference 1 g711alaw
!
!
!
!
voice translation-rule 3
rule 1 /^0/ /+49/
!
voice translation-rule 4
rule 1 /^1001/ /004930129854765/
!
voice translation-rule 11
rule 1 /^/ /0/ type unknown unknown
rule 2 /^/ /00/ type national national
rule 3 /^/ /000/ type international international
!
voice translation-rule 12
rule 1 /^004930129854765/ /1001/
!
!
voice translation-profile SIP-Incoming
translate calling 11
translate called 12
!
voice translation-profile SIP-Outgoing
translate calling 4
translate called 3
!
!
!
dial-peer voice 11 voip
description **Incoming Call**
translation-profile incoming SIP-Incoming
session protocol sipv2
session target dns:sipgate.de
incoming called-number .%
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
dial-peer voice 10 voip
description **Outgoing Call**
translation-profile outgoing SIP-Outgoing
destination-pattern 0T
session protocol sipv2
session target dns:sipgate.de
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
!
!
sip-ua
credentials username 004930129854765 password 7 <password> realm sipgate.de
authentication username 004930129854765@sipgate.de password 7 <password>
no remote-party-id
retry invite 2
retry register 10
retry options 1
timers connect 100
registrar dns:sipgate.de expires 3600
sip-server dns:sipgate.de
host-registrar
!
!
!
telephony-service
no auto-reg-ephone
max-ephones 20
max-dn 40
ip source-address 192.168.5.99 port 2000
load 7960-7940 P00308010200
max-conferences 8 gain -6
web admin system name cisco secret 5 <password>
dn-webedit
time-webedit
transfer-system full-consult
create cnf-files version-stamp Jan 01 2002 00:00:00
!
!
ephone-dn 1 dual-line
number 1001 secondary 004930129854765 no-reg primary
!
!
**DEBUG INCOMING**
*Nov 2 23:05:09.759: //8/xxxxxxxxxxxx/CCAPI/cc_api_caps_ind:
Call Entry Is Not Found
*Nov 2 23:05:09.763: //-1/F2512023800F/CCAPI/cc_api_display_ie_subfields:
cc_api_call_setup_ind_common:
cisco-username=+4917661022799
ccCallInfo IE subfields
cisco-ani=+4917975322866
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=004930129854765
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-lastrdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
*Nov 2 23:05:09.763: //-1/F2512023800F/CCAPI/cc_api_call_setup_ind_common:
Interface=0x49E2F7A4, Call Info(
Calling Number=+4917975322866,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=004930129854765(TON=Unknown, NPI=Unknown),
Calling Translated=FALSE, Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE,
Incoming Dial-peer=11, Progress Indication=NULL(0), Calling IE Present=TRUE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, CLID Transparent=FALSE), Call Id=8
Guid=F2512023-04DD-11E1-800F-807480744F27, Outgoing Dial-peer=20001
*Nov 2 23:05:09.775: //8/F2512023800F/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=+4917975322866
ccCallInfo IE subfields
cisco-ani=+4917975322866
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=1
dest=1001
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-lastrdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=-1
cisco-rdnsi=-1
cisco-redirectreason=-1 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
*Nov 2 23:05:09.779: //8/F2512023800F/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x4BAD9944, Interface Type=6, Destination=, Mode=0x0,
Call Params(Calling Number=+4917975322866,(Calling Name=+4917975322866 )(TON=Unknown, NPI=Unknown, Screening=User, Passed, Presentation=Allowed),
Called Number=1001(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=Unknown, FinalDestinationFlag=TRUE, Outgoing Dial-peer=20001, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
The Call Setup Information is:
Call Control Block (CCB) : 0x4AB78130
State of The Call : STATE_DEAD
TCP Sockets Used : NO
Calling Number : +4917975322866
Called Number : 004930129854765
Source IP Address (Sig ): 192.168.5.99
Destn SIP Req Addr:Port : 217.10.79.9:5060
Destn SIP Resp Addr:Port : 217.10.79.9:5060
Destination Name : 217.10.79.9
*Nov 2 23:05:14.575: //8/F2512023800F/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : g711alaw
Negotiated Codec Bytes : 160
Nego. Codec payload : 8 (tx), 8 (rx)
Negotiated Dtmf-relay : 6
Dtmf-relay Payload : 101 (tx), 101 (rx)
Source IP Address (Media): 192.168.5.99
Source IP Port (Media): 17066
Destn IP Address (Media): 217.10.79.9
Destn IP Port (Media): 40144
Orig Destn IP Address:Port (Media): [ - ]:0
*Nov 2 23:05:14.575: //8/F2512023800F/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 16
Disconnect Cause (SIP) : 487
**DEBUG OUTGOING**
Guid=A9C0A698-04DB-11E1-801B-F8B7FAD2A1EA, Outgoing Dial-peer=10
Nov 2 22:48:51.650: //12/A9C0A698801B/CCAPI/cc_api_display_ie_subfields:
ccCallSetupRequest:
cisco-username=
ccCallInfo IE subfields
cisco-ani=004930129854765
cisco-anitype=0
cisco-aniplan=0
cisco-anipi=0
cisco-anisi=0
dest=+4917975322866
cisco-desttype=0
cisco-destplan=0
cisco-rdie=FFFFFFFF
cisco-rdn=
cisco-lastrdn=
cisco-rdntype=0
cisco-rdnplan=0
cisco-rdnpi=0
cisco-rdnsi=0
cisco-redirectreason=0 fwd_final_type =0
final_redirectNumber =
hunt_group_timeout =0
Nov 2 22:48:51.650: //12/A9C0A698801B/CCAPI/ccIFCallSetupRequestPrivate:
Interface=0x4A07C064, Interface Type=3, Destination=, Mode=0x0,
Call Params(Calling Number=004930129854765,(Calling Name=)(TON=Unknown, NPI=Unknown, Screening=Not Screened, Presentation=Allowed),
Called Number=+4917975322866(TON=Unknown, NPI=Unknown), Calling Translated=FALSE,
Subscriber Type Str=RegularLine, FinalDestinationFlag=FALSE, Outgoing Dial-peer=10, Call Count On=FALSE,
Source Trkgrp Route Label=, Target Trkgrp Route Label=, tg_label_flag=0, Application Call Id=)
The Call Setup Information is:
Call Control Block (CCB) : 0x4ADCE9F0
State of The Call : STATE_DEAD
TCP Sockets Used : YES
Calling Number : 004930129854765
Called Number : +4917975322866
Source IP Address (Sig ): 192.168.5.99
Destn SIP Req Addr:Port : 217.10.79.9:5060
Destn SIP Resp Addr:Port : 217.10.79.9:5060
Destination Name : sipgate.de
Nov 2 22:48:56.826: //13/A9C0A698801B/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream : 1
Negotiated Codec : No Codec
Negotiated Codec Bytes : 0
Nego. Codec payload : 255 (tx), 255 (rx)
Negotiated Dtmf-relay : 0
Dtmf-relay Payload : 0 (tx), 0 (rx)
Source IP Address (Media): 192.168.5.99
Source IP Port (Media): 18588
Destn IP Address (Media): -
Destn IP Port (Media): 0
Orig Destn IP Address:Port (Media): [ - ]:0
Nov 2 22:48:56.826: //13/A9C0A698801B/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC) : 38
Disconnect Cause (SIP) : 503
Hope you see a misconfig. :-/ -
woisch Member Posts: 40 ■■□□□□□□□□I have tried to debug this with ccsip messages. For incoming i get the foloowing bu t for outgoing no debug appears. Why? Any idea?
CME#
*Nov 4 12:33:48.639: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:004930129854765@192.168.5.99:5060 SIP/2.0
Record-Route: <sip:217.10.79.9;lr;ftag=156cabf2;n=4>
Record-Route: <sip:217.10.79.9;lr;ftag=156cabf2;rqu=WkNGWS19QD9UTEBaMHpKNS8xBANoZRB2Gw4SEmslU10/>
Record-Route: <sip:217.10.79.9;lr;ftag=156cabf2>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK6b7d.dbf19946.1
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.c6c281b3.0
Via: SIP/2.0/UDP 213.20.246.108:5060;received=213.20.246.108;branch=z9hG4bK608d35d8;rport=5060
From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
To: <sip:004930129854765@217.10.79.9>
Contact: <sip:+41796035695@213.20.246.108>
Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
CSeq: 2 INVITE
Date: Fri, 04 Nov 2011 12:31:03 GMT
P-Asserted-Identity: +4xxxxxxxxx <sip:+4xxxxxxxxxx@carpo-2.sip.mgc.voip.telefonica.de;user=phone>
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,OPTIONS,INFO
Max-Forwards: 49
Content-Type: application/sdp
Content-Length: 223
X-Remote-IP: 217.10.79.9:5060
v=0
o=root 19413 19413 IN IP4 213.20.246.108
s=-
c=IN IP4 213.20.246.108
t=0 0
m=audio 40124 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
*Nov 4 12:33:48.667: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0,SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK6b7d.dbf19946.1,SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.c6c281b3.0,SIP/2.0/UDP 213.20.246.108:5060;received=213.20.246.108;branch=z9hG4bK608d35d8;rport=5060
From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
To: <sip:004930129854765@217.10.79.9>
Date: Fri, 04 Nov 2011 12:33:48 GMT
Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
CSeq: 2 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Nov 4 12:33:48.711: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0,SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK6b7d.dbf19946.1,SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.c6c281b3.0,SIP/2.0/UDP 213.20.246.108:5060;received=213.20.246.108;branch=z9hG4bK608d35d8;rport=5060
From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
To: <sip:004930129854765@217.10.79.9>;tag=404C4-25F3
Date: Fri, 04 Nov 2011 12:33:48 GMT
Call-ID: 1474b05c1ef48d165419e8
CME#072b6826d3@213.20.246.108
CSeq: 2 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact: <sip:004930129854765@192.168.5.99:5060>
Record-Route: <sip:217.10.79.9;lr;ftag=156cabf2;n=4>,<sip:217.10.79.9;lr;ftag=156cabf2;rqu=WkNGWS19QD9UTEBaMHpKNS8xBANoZRB2Gw4SEmslU10/>,<sip:217.10.79.9;lr;ftag=156cabf2>
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CME#
*Nov 4 12:33:52.327: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
CANCEL sip:004930129854765@192.168.5.99:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0
From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
To: <sip:004930129854765@217.10.79.9>
CSeq: 2 CANCEL
Max-Forwards: 70
User-Agent: MCCS Dispatcher 1.7.0
Content-Length: 0
*Nov 4 12:33:52.347: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0
From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
To: <sip:004930129854765@217.10.79.9>
Date: Fri, 04 Nov 2011 12:33:52 GMT
Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
CSeq: 2 CANCEL
Content-Length: 0
*Nov 4 12:33:52.351: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0,SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK6b7d.dbf19946.1,SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.c6c281b3.0,SIP/2.0/UDP 213.20.246.108:5060;received=213.20.246.108;branch=z9hG4bK608d35d8;rport=5060
From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
To: <sip:004930129854765@217.10.79.9>;tag=404C4-25F3
Date: Fri, 04 Nov 2011 12:33:52 GMT
Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
CSeq: 2 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0
*Nov 4 12:33:52.435: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:004930129854765@192.168.5.99:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0
From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
To: <sip:004930129854765@217.10.79.9>;tag=404C4-25F3
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: MCCS Dispatcher 1.7.0
Content-Length: 0
*Nov 4 12:33:52.443: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 487 Request Cancelled
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0,SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK6b7d.dbf19946.1,SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.c6c281b3.0,SIP/2.0/UDP 213.20.246.108:5060;received=213.20.246.108;branch=z9hG4bK608d35d8;rport=5060
From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
To: <sip:004930129854765@217.10.79.9>;tag=404C4-25F3
Date: Fri, 04 Nov 2011 12:33:52 GMT
Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
CSeq: 2 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Reason: Q.850;cause=16
Content-Length: 0
CME#
*Nov 4 12:33:52.503: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:004930129854765@192.168.5.99:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6b7d.e6c281b3.0
From: "+4917975322866" <sip:+4917975322866@sipgate.de>;tag=156cabf2
Call-ID: 1474b05c1ef48d165419e8072b6826d3@213.20.246.108
To: <sip:004930129854765@217.10.79.9>;tag=404C4-25F3
CSeq: 2 ACK
Max-Forwards: 70
User-Agent: MCCS Dispatcher 1.7.0
Content-Length: 0
*Nov 4 12:33:52.543: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
...
*Nov 4 12:33:52.543: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
...
CME#
*Nov 4 12:34:02.835: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
...
*Nov 4 12:34:02.835: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
...
CME#
*Nov 4 12:34:13.015: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
...
*Nov 4 12:34:13.015: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
...
CME# -
woisch Member Posts: 40 ■■□□□□□□□□Ok. it worked.
I have added "session transport udp" to the outgoing (and incoming) dial-peer. That was it. -
pitviper Member Posts: 1,376 ■■■■■■■□□□Hmm, that’s odd – never had to use that command before. I wonder if it’s ITSP specific.CCNP:Collaboration, CCNP:R&S, CCNA:S, CCNA:V, CCNA, CCENT
-
networker050184 Mod Posts: 11,962 ModHi Pitviper,
sip
bind control source-interface FastEthernet0/0
session transport tcp
registrar server expires max 3600 min 3600
redirect contact order best-match
localhost dns:sipgate.de
!
There's your issue. SIP defaults to UDP so you shouldn't need to set to UDP on your dial-peers. Looks like that gloabal SIP setting was fudging you up though.An expert is a man who has made all the mistakes which can be made. -
pitviper Member Posts: 1,376 ■■■■■■■□□□That would explain it! Good eye - I didn't even notice it in the config.CCNP:Collaboration, CCNP:R&S, CCNA:S, CCNA:V, CCNA, CCENT