SIP configuration trouble

Hi,
On my gateways 5400, SIP configuration is :
voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback none
h323
sip
bind control source-interface Loopback6
bind media source-interface Loopback6
But on one of my 5400, line "bind control source-interface Loopback6" doesn't appear. When i implement it, i have no error message.
SIP calls doesn't complete (dead air). Dial-peer is well configured (calls completed on another gw).
Thanks for your help.
Phil

Compare the IOSes running on these AS5400s.
Are they different? :)
If they are, choose one that you consider as being most stable and make the other running it :)

Similar Messages

Please help with SIP configuration on 2801 router

Hi All.
Please help me to setup a SIP account. I’m already struggling to do that for a few days, and can’t find out how to finish that. We have 2xISDN lines running, so I need to add a SIP trunk to existing config.
The information from our SIP provider:
We have issued the following DDI range: 018877000 – 99
There is no need to register the DDI’s as these will be offered to your PABX IP address provided to in the completed SIP trunking form.
Configuration details are as follows:
Our Primary Proxy:-        99.234.56.78
Codec supported:-             G711Alaw, G729 (G711Alaw is the preferred codec)
Fax Support:-                     T38 and G711Alaw
DTMF:-                                 RFC2833 and INFO
CLI Method:-                     Remote-Party-ID
Trunk doesn’t require registration; you just need to send Invite. In cisco this is done through Dial-peer session-target command. We are authenticating your IP address for outgoing calls and incoming calls we then forward to the IP mentioned in the sip form.
This is a SIP configuration on Cisco2801 router (I used outgoing calls only):
translation-rule 10
Rule 0 ^90 0
Rule 1 ^91 1
Rule 2 ^92 2
Rule 3 ^93 3
Rule 4 ^94 4
Rule 5 ^95 5
Rule 6 ^96 6
Rule 7 ^97 7
Rule 8 ^98 8
Rule 9 ^99 9
interface FastEthernet0/0.1
description ***DATA VLAN***
encapsulation dot1Q 1 native
ip address 10.1.1.101 255.255.255.0
interface FastEthernet0/0.2
description ***VOICE VLAN***
encapsulation dot1Q 2
ip address 192.168.22.1 255.255.255.0
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
h323
call start slow
sip
bind control source-interface FastEthernet0/0.2
bind media source-interface FastEthernet0/0.2
registrar server expires max 36000 min 600
voice class codec 1
codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw
dial-peer voice 1 pots
description ### External Dialling via BRI ###
preference 7
destination-pattern 9T
translate-outgoing called 10
direct-inward-dial
port 0/0/0
forward-digits all
dial-peer voice 2 pots
description ### External Dialling via BRI ###
preference 2
destination-pattern 9T
translate-outgoing called 10
direct-inward-dial
port 0/0/1
forward-digits all
dial-peer voice 9000 voip
description ** Outgoing calls to SIP **
preference 1
destination-pattern 9T
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:99.234.56.78:5060
dtmf-relay rtp-nte
codec g711alaw
no vad
sip-ua
timers connect 100
sip-server ipv4:99.234.56.78
I used debugging commands to troubleshoot the calls.
2801(config-dial-peer)#
094509: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=211, Called Number=, Voice-Interface=0x65FA35B4,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOICE, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
094510: Jan 24 09:27:06.204: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=20018
094511: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9, Peer Info Type=DIALPEER_INFO_SPEECH
094512: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9
094513: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094514: Jan 24 09:27:06.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094515: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90, Peer Info Type=DIALPEER_INFO_SPEECH
094516: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90
094517: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094518: Jan 24 09:27:06.816: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094519: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=908, Peer Info Type=DIALPEER_INFO_SPEECH
094520: Jan 24 09:27:06.912: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=908
094521: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094522: Jan 24 09:27:06.916: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094523: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9086, Peer Info Type=DIALPEER_INFO_SPEECH
094524: Jan 24 09:27:07.012: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9086
094525: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094526: Jan 24 09:27:07.016: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094527: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862, Peer Info Type=DIALPEER_INFO_SPEECH
094528: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862
094529: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094530: Jan 24 09:27:07.116: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094531: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=908621, Peer Info Type=DIALPEER_INFO_SPEECH
094532: Jan 24 09:27:07.212: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=908621
094533: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094534: Jan 24 09:27:07.216: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094535: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9086215, Peer Info Type=DIALPEER_INFO_SPEECH
094536: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9086215
094537: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094538: Jan 24 09:27:07.316: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094539: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157, Peer Info Type=DIALPEER_INFO_SPEECH
094540: Jan 24 09:27:07.412: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157
094541: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094542: Jan 24 09:27:07.416: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094543: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=908621577, Peer Info Type=DIALPEER_INFO_SPEECH
094544: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=908621577
094545: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094546: Jan 24 09:27:07.516: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094547: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=9086215777, Peer Info Type=DIALPEER_INFO_SPEECH
094548: Jan 24 09:27:07.612: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=9086215777
094549: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094550: Jan 24 09:27:07.616: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094551: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094552: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
094553: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Partial Matches(1) after DP_MATCH_DEST
094554: Jan 24 09:27:07.716: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=MORE_DIGITS_NEEDED(1)
094555: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
094556: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774T
094557: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
094558: Jan 24 09:27:10.711: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
094559: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094560: Jan 24 09:27:10.711: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
094561: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
094562: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
094563: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
094564: Jan 24 09:27:10.715: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
094565: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
094566: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
094567: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
094568: Jan 24 09:27:10.715: //-1/6A877F6F9054/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
094569: Jan 24 09:27:10.719: fb_get_reject_cause_code: ERROR cause_code NULL
094570: Jan 24 09:27:10.727: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:10 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397230
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094571: Jan 24 09:27:11.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:11 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397231
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094572: Jan 24 09:27:12.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam " <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam " <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:12 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397232
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
094573: Jan 24 09:27:14.227: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.22.1:5060;branch=z9hG4bK47D116D3
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Sam" <sip:[email protected]>;tag=CDCFB8AC-F98
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 09:27:14 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 1787264879-1168380385-2421457215-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327397234
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 244
v=0
o=CiscoSystemsSIP-GW-UserAgent 3237 2021 IN IP4 192.168.22.1
s=SIP Call
c=IN IP4 192.168.22.1
t=0 0
m=audio 18258 RTP/AVP 8 101
c=IN IP4 192.168.22.1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
I made some changes in the router configuration.
I removed FA0/0.2 Voice interface from Voice service voip configuration (bind control source-interface FastEthernet0/0.2 and bind media source-interface FastEthernet0/0.2). And now it’s using ip address 10.1.1.101 (data ip).
The debugging is changed now. I can send and receive a respond from SIP server. But It shows an error: SIP/2.0 404 Not Found
Then it moves to ISDN line, and use this line to make a call.
102988: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774T, Peer Info Type=DIALPEER_INFO_SPEECH
102989: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774T
102990: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
102991: Jan 24 14:45:47.290: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
102992: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=90862157774, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
102993: Jan 24 14:45:47.290: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
102994: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
102995: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
102996: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
102997: Jan 24 14:45:47.294: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
102998: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
102999: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
103000: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
103001: Jan 24 14:45:47.294: //-1/EDCA21089304/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
103002: Jan 24 14:45:47.298: fb_get_reject_cause_code: ERROR cause_code NULL
103003: Jan 24 14:45:47.310: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
Remote-Party-ID: "Sam" <sip:[email protected]>;party=calling;screen=no;privacy=off
From: "Seam" <sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>
Date: Tue, 24 Jan 2012 14:45:47 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 3989446920-1171263969-2466545983-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327416347
Contact: <sip:[email protected]:5060>
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 2438 9821 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 19412 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
103004: Jan 24 14:45:47.354: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 404 Not Found
From: "Sam "<sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
Call-ID: [email protected]
CSeq: 101 INVITE
Via: SIP/2.0/UDP 10.1.1.101:5060;received=88.99.77.44;branch=z9hG4bK4875CB9
Content-Length: 0
103005: Jan 24 14:45:47.362: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK4875CB9
From: "Sam " <sip:[email protected]>;tag=CEF37490-172C
To: <sip:[email protected]>;tag=7fad61f03708-100007f-13c4-55013-a0142-10fd12c8-a0142
Date: Tue, 24 Jan 2012 14:45:47 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0
103006: Jan 24 14:45:47.374: %ISDN-6-LAYER2UP: Layer 2 for Interface BR0/0/1, TEI 96 changed to up
103007: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=211, Peer Info Type=DIALPEER_INFO_SPEECH
103008: Jan 24 14:45:51.313: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=211
103009: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
103010: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=20018
103011: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=0862157774, Peer Info Type=DIALPEER_INFO_SPEECH
103012: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=0862157774
103013: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   No Outgoing Dial-peer Is Matched; Result=NO_MATCH(-1)
103014: Jan 24 14:45:51.317: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
   Result=NO_MATCH(-1)
103015: Jan 24 14:46:08.815: %ISDN-6-LAYER2DOWN: Layer 2 for Interface BR0/0/1, TEI 96 changed to down
2801(config-dial-peer)#
Then I removed SIP-UA as I was told there is no registration necessary, only Dial-peer configuration.
But it didn’t affect anything.
Then I add translate-outgoing called 10 command to dial-peer 9000, nothing happened.
Really stuck and don't know where to look at.
Any help will be highly appreciated.
Thanks.

Hi Dan.
Yes, I saw that RTP debugging, but what can I change there? Maybe I need to open more ports on ASA for RTP like 19412?
I use Cisco ASDM for ASA to make changes.
There are static NAT rules for: Server source IPs(10.1.1.100) to Outside(translated IPs, 88.99.77.44) for a few ports.
Also I added Security policy access rules for LAN: Any to SIP, and Outside: SIP to any.
For NAT:
I can't add this: for LAN: STATIC ROUTER IP 10.1.1.101 (AS SOURCE) UDP 5060 TO OUTSIDE IP 88.99.77.44
(AS TRANSLATED) UDP 5060
Because there is already translation for the Server.
Debugging looks like that now. There is no Received: SIP/2.0, but I can make an outside call with no audio.
116013: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=90862157774, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
116014: Jan 25 15:28:25.584: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_ORIGINATE; Incoming Dial-peer=9000
116015: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=90862157774, Peer Info Type=DIALPEER_INFO_SPEECH
116016: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=90862157774
116017: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
116018: Jan 25 15:28:25.584: //-1/0D0EB9CE9708/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=9000
     2: Dial-peer Tag=2
     3: Dial-peer Tag=1
116019: Jan 25 15:28:25.588: fb_get_reject_cause_code: ERROR cause_code NULL
116020: Jan 25 15:28:25.600: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:25 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505305
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116021: Jan 25 15:28:26.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:26 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505306
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116022: Jan 25 15:28:27.096: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam " ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:27 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505307
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
116026: Jan 25 15:28:57.092: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK491484D
Remote-Party-ID: "Sam" ;party=calling;screen=no;privacy=off
From: "Sam " ;tag=D4410748-1C9D
To:
Date: Wed, 25 Jan 2012 15:28:57 GMT
Call-ID: [email protected]
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 219068878-1184895457-2533916991-1958389preference 1771
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1327505337
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1984 5803 IN IP4 10.1.1.101
s=SIP Call
c=IN IP4 10.1.1.101
t=0 0
m=audio 18782 RTP/AVP 8 101
c=IN IP4 10.1.1.101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
I'll add Incoming dial-peer now.
Not sure what kind of NAT rule should I put into ASA to allow in and out sip traffic.
Appretiate your help.
Thanks a mill.

VoIP (SIP) configuration guide for Nokia 5800xm

hi.everybody i would like to ask how to configure this
VoIP (SIP) configuration for my n5800, because when i saw and follow this link http://www.elisanet.fi/craig/sipvoip/nokia_n97.html ,i only configure my SIP setting and not my net setting which i tried to find in my phone. can someone help me find this Net setting in my handset and can you put a link on how to call from phone to phone using this "VoIP" and is this free of charge. thank..more power

Hi, don't have an answer to your question, but Skype for S60.5 was announced today, works fine, will give Skype to Skype calls free, and low cost calls to non Skype numbers, but both Skype or Voip will incur data charges if you don;t have a data plan !
Good Luck
Skype.com/m with your phones browser will take you to Skype
If I have helped at all, a click on the White Star is always appreciated :
you can also help others by marking 'accept as solution'

Sip configuration for Vyke

Hi
Can someone tell me where to find the Sip configuration for VYKE system?

Profile name: Arbitrary
Service profile: IETF
ACCESS POINT : [your wifi access point]
PUBLIC USER NAME: sip:[username]@sip.vyke.com
use compression: no
registration : “always on” or “when need”
SECURITY: NO
PROXY SERVER
PROXY SERVER: sip:sip.vyke.com
REALM: sip.vyke.com
USER NAME: :[sip_user]
password: [sip_pass]
allow loose routing: yes
transport type: UDP
PORT: 5060
REGISTRATION SERVER
REGISTRATION SERVER: sip:sip.vyke.com
realm: sip.vyke.com
user name: [sip_user]
password: [sip_pass]
transport type: UDP
PORT: 5060
N86 8mp: RM-485,0590552; Version 20.115 - 6/29/2009. - -- E71-2; RM-357; 0569371; Version 100.07.76 - 6/08/2008.
N8; N900

Logging API configuration trouble

Here is my Java code:
private static final String LOGGER_CONFIG_FILE="java.util.logging.config.file";
private static final String CONFIG_FILE="c:/logging.properties";
public static synchronized void init() {
System.setProperty(LOGGER_CONFIG_FILE, CONFIG_FILE);
try {
LogManager.getLogManager().readConfiguration();
} catch (IOException E) {
System.err.println("Error while reading logging configuration file \""+CONFIG_FILE+"\":"+E);
Here is my logging.properties:
handlers=java.util.logging.ConsoleHandler, java.util.logging.FileHandler
.level=INFO
com.mypackage.level=ALL
java.util.logging.ConsoleHandler.level = WARNING
java.util.logging.ConsoleHandler.formatter = java.util.logging.SimpleFormatter
java.util.logging.FileHandler.level = ALL
java.util.logging.FileHandler.formatter = java.util.logging.SimpleFormatter
java.util.logging.FileHandler.pattern=c:/java%u.log
java.util.logging.FileHandler.limit=50000
java.util.logging.FileHandler.count=1
Now here is my trouble:
when I'm running init() method all is ok !
when I'm trying to do something like this MyClass.getLogger().severe(msg); then there is an error message in my console:
Can't set level for java.util.logging.ConsoleHandler
Can't set level for java.util.logging.FileHandler
But the fact is, that my servere message is actually displayed in console and in my java0.log file :(
And the second trouble: all FileHanler's output is in XMLFormatter, but in config file there is SimpleFormatter set.
What are the problems, guys, please help !
Thanks

Help somebody, pleaseeee !

CME Extension Mobility, SIP configuration

Hi,
Need help with CME Extension Mobility with SIP Phones (7841). I'm using CME 10.5 and I configured the parameters below for extension mobility but the phones won't register right after I put the logout profile in the voice register pool.
They work normally when not in Extension Mobility though. Please help I need to deploy this to my customer soon.
hostname Router
boot-start-marker
boot system flash:c3900-universalk9-mz.SPA.154-3.M2.bin
boot-end-marker
no aaa new-model
no authentication logging verbose
ip dhcp excluded-address 192.168.1.1 192.168.1.20
ip dhcp excluded-address 192.168.1.254
ip dhcp pool Phones
network 192.168.1.0 255.255.255.0
default-router 192.168.1.254
option 150 ip 192.168.1.254
no ip domain lookup
ip cef
no ipv6 cef
multilink bundle-name authenticated
cts logging verbose
voice-card 0
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/1.10
bind media source-interface GigabitEthernet0/1.10
registrar server expires max 600 min 60
voice register global
mode cme
source-address 192.168.1.254 port 5060
max-dn 110
max-pool 110
load 7841 sip78xx.10-1-1SR1-4
time-format 24
date-format D/M/Y
service https
url authentication http://192.168.1.254/CCMCIP/authenticate.asp
tftp-path flash:
create profile sync 0002641841434163
voice register dn 1
number 6001
name Poh Huat - 6001
label Poh Huat - 6001
voice register dn 4
number 6005
name Coordinator - 6005
label Coordinator - 6005
voice register pool 1
logout-profile 100
busy-trigger-per-button 2
id mac 547C.69D6.1AB6
type 7841
voice register pool 4
logout-profile 100
busy-trigger-per-button 2
id mac 547C.69D6.1A2F
type 7841
voice logout-profile 100
pin 1234
user 6000 password 12345
number 6000 type normal
speed-dial 1 999 label "EMERGENCY"
voice user-profile 1
pin 12345
user richard password richard
number 6001 type normal
speed-dial 1 996506901 label "Richard"
voice user-profile 2
pin 12345
user 6005 password 12345
number 6005 type normal
license udi pid C3900-SPE100/K9 sn FOC16145MQA
license boot module c3900 technology-package uck9
username xtra privilege 15 secret 5 $1$STRs$Qsuesm8dF23Okof.vRyf5.
redundancy
ip ftp username xtra
ip ftp password xtra2006admin
interface Embedded-Service-Engine0/0
no ip address
shutdown
interface GigabitEthernet0/0
ip address dhcp
duplex auto
speed auto
interface GigabitEthernet0/1
no ip address
duplex auto
speed auto
interface GigabitEthernet0/1.10
encapsulation dot1Q 10 native
ip address 192.168.1.254 255.255.255.0
interface GigabitEthernet0/2
no ip address
shutdown
duplex auto
speed auto
ip forward-protocol nd
ip http server
no ip http secure-server
ip http path flash:
nls resp-timeout 1
cpd cr-id 1
tftp-server flash:PHONES/sip78xx.10-1-1SR1-4.loads alias sip78xx.10-1-1SR1-4.loads
control-plane
mgcp behavior rsip-range tgcp-only
mgcp behavior comedia-role none
mgcp behavior comedia-check-media-src disable
mgcp behavior comedia-sdp-force disable
mgcp profile default
gatekeeper
shutdown
telephony-service
authentication credential 6000 12345
em keep-history
max-ephones 110
max-dn 110
service phone webAccess 0
max-conferences 8 gain -6
transfer-system full-consult
create cnf-files version-stamp 7960 Mar 05 2015 15:50:52
I have turned on debug ip http all and debug voice em-profile on and right after I entered the logout profile 100 under pool i get the following logs.
Router(config-register-pool)#
Mar 5 16:15:28.299: Thu, 05 Mar 2015 16:15:28 GMT 192.168.1.21 /CMEserverForPhone/serviceurl ok
Protocol = HTTP/1.1 Method = GET Query = locale=English_United_States&name=SEP547C69D61A2F
Mar 5 16:15:28.299:
Mar 5 16:15:28.299: Getting SIP phone index by IP address 192.168.1.21
Mar 5 16:15:28.299: SIP phone 4 found with contact IP address 192.168.1.21
Mar 5 16:15:33.363: Thu, 05 Mar 2015 16:15:33 GMT 192.168.1.21 /CMEserverForPhone/serviceurl ok
Protocol = HTTP/1.1 Method = GET Query = locale=English_United_States&name=SEP547C69D61A2F
Mar 5 16:15:33.363:
Mar 5 16:15:33.363: Getting SIP phone index by IP address 192.168.1.21
Mar 5 16:15:33.363: SIP phone 4 found with contact IP address 192.168.1.21
Mar 5 16:15:37.539: Thu, 05 Mar 2015 16:15:37 GMT 192.168.1.21 /CMEserverForPhone/extensionmobility ok
Protocol = HTTP/1.1 Method = GET
Mar 5 16:15:37.539:
Mar 5 16:15:37.539: Getting SIP phone index by IP address 192.168.1.21
Mar 5 16:15:37.539: SIP phone 4 found with contact IP address 192.168.1.21
After this the phones are still not registering, I'm suspecting it is the url authentication command, as i can't put the application-name and password after the command, any suggestions would be appreciated. THanks in advance.
-richard

Try adding:
voice register global
url authentication http://192.168.1.254/CCMCIP/authenticate.asp secretname psswrd
if still doesn't work try to compare your config with the reference guide here:
http://www.cisco.com/c/en/us/td/docs/voice_ip_comm/cucme/admin/configuration/guide/cmeadm/cmemobl.html#pgfId-1163414
-Terry
Please rate all helpful posts

SIP configuration for SX10

Hello,
I just received a cisco telepresence SX10 device which uses SIP protocol only and I have no experience at all with this. Older devices that I have worked with used H323 protocol which is very easy to configure. Looking for some assistance on how to setup the SIP. Your help will be greatly appreciated
Thank you

For sip you need to configure the following under sip profile
1. sip uri...This is usually the extension of your sx10 ex..[email protected]
2.Display name: used for identifying this device when making a call
3.Authentication: login name and password if you have one set up on your vcs
4. Default Transport: TCP/UDP/TLS..Change this to whatever protocol you are using on your proxy (VCS)
5. Outbound..You can enable your sx10 to use multiple connection to your proxy servers by specifying the proxy addresses here. or You can turn this off and use the proxy option to configure which proxy servers you want your device to connect to
These are the basic that should get your device to register on your proxy server. Check more details here..
http://www.cisco.com/c/dam/en/us/td/docs/telepresence/endpoint/sx-series/tc7/administration-guide/sx10-administrator-guide-tc71.pdf

Need help on SIP configuration

hi guys
i m a starter with the SIP technology ,i m planning to use SIP as session protocol instead of cisco properitry.can anyone pls tell me do i need to configure a SIP proxy or anything that kind to use this.
i have total 6 locations and i want all the 6 locations to communicate between each other.so i planned for SIP in this scenario.this is a IP based network
all u r comments r welcome
regds
prem

hi
pls go thru the details of the H/W ,IOS ver and about the interface
System image file is "flash:c1700-sv8y-mz.122-8.T1.bin"
cisco 1751 (MPC860P) processor (revision 0x200) with 55706K/9830K bytes of memory.
Processor board ID JAD06200B7K (241514676), with hardware revision 0000
MPC860P processor: part number 5, mask 2
Bridging software.
X.25 software, Version 3.0.0.
1 FastEthernet/IEEE 802.3 interface(s)
2 Serial(sync/async) network interface(s)
2 Voice FXS interface(s)
my doubt is shall i give this command Router(config-dial-peer)# session protocol {cisco | sipv2} and go ahead with the config or do i need to configure the SIP proxy server.??
if i use session protocol as Cisco it will take as H.323 is it equivalent to not putting this entry since its going to take h.323 only as its default ..
regds
prem

SIP configuration in Nokia N 97

I am unable to configure SIP in my N 97, whereas E 71 phone was ver easy to configure. Anyone can suggest any steps to help me. the configuration is as below:
SIP Server Or SIP Proxy Server: ata.nymgo.com
SIP User ID: nymgo username
Authenticate ID: nymgo username
SIP password: nymgo password
SIP Port: 5060
Codec Used: g729
STUN Server: stun.nymgo.com
STUN port: 80

Try search there are many many posts on this here.
I love my purple screen, 1/2 day battery, sketchy touch screen and $800 price tag - I call it my Lumia 900

SIP configuration on N95 problem

Hi,
I'm trying to set a SIP profile on my N95 using Nokia native SIP. My goal is to run a VoIP service named Spikko.
I followed the Skippo instruction sheet (to be enclosed below), but fail to establish a connection.
I checked and rechecked the SIP parameters and found a difference between my inputted paramters and the ones which my N95 uses. Whenever I define an address, in my case SIP.SPIKKO.COM, the parameter shown is SIPIP.SPIKKO.COM (SIP: added).
Can this explain my failure? If so, how do I solve this problem? If that's a harmless change, what can explain and solve my problem?
Here's the configuration instruction sheet:
Connecting to Spikko using Nokia SIP:
Nokia N95, E71, E61, and several other models include a Nokia “Native SIP” application. The configuration differs slightly between models. The following is the N95 configuration:
• Go to “Menu » Tools » Settings » Connection » SIP settings » Options » New SIP profile and use the following settings:
--Profile name: Spikko
--Service profile: IETF
--Default Access Point: Choose “WiFi” or “3G internet access point” - לא קיים
--Public username [email protected] (for instance: [email protected])
--Use compression: No
--Registration: When needed
--Use security: no
--proxy server:
--Proxy Server Address: sip.spikko.com
--Realm: spikko
--Username: your Spikko username
--Password: your Spikko password
--Allow loose routing: yes
--Transport type: Automatic
--Port: 5090
--Registration server: Same as proxy server
• Now, go to “Menu » Tools » Settings » Connection » Internet-Tel” and choose the Spikko profile. Your Nokia now is ready to make and receive calls.
• See additional information at http://www.spikko.com/Nokia/Movie.aspx.

Hi,
I'm trying to set a SIP profile on my N95 using Nokia native SIP. My goal is to run a VoIP service named Spikko.
I followed the Skippo instruction sheet (to be enclosed below), but fail to establish a connection.
I checked and rechecked the SIP parameters and found a difference between my inputted paramters and the ones which my N95 uses. Whenever I define an address, in my case SIP.SPIKKO.COM, the parameter shown is SIPIP.SPIKKO.COM (SIP: added).
Can this explain my failure? If so, how do I solve this problem? If that's a harmless change, what can explain and solve my problem?
Here's the configuration instruction sheet:
Connecting to Spikko using Nokia SIP:
Nokia N95, E71, E61, and several other models include a Nokia “Native SIP” application. The configuration differs slightly between models. The following is the N95 configuration:
• Go to “Menu » Tools » Settings » Connection » SIP settings » Options » New SIP profile and use the following settings:
--Profile name: Spikko
--Service profile: IETF
--Default Access Point: Choose “WiFi” or “3G internet access point” - לא קיים
--Public username [email protected] (for instance: [email protected])
--Use compression: No
--Registration: When needed
--Use security: no
--proxy server:
--Proxy Server Address: sip.spikko.com
--Realm: spikko
--Username: your Spikko username
--Password: your Spikko password
--Allow loose routing: yes
--Transport type: Automatic
--Port: 5090
--Registration server: Same as proxy server
• Now, go to “Menu » Tools » Settings » Connection » Internet-Tel” and choose the Spikko profile. Your Nokia now is ready to make and receive calls.
• See additional information at http://www.spikko.com/Nokia/Movie.aspx.

Conky configuration troubles

Hello, everyone. I'm trying to set up conky on my laptop, but I'm having some trouble configuring it the way I want. Now, what I want is to have four separate windows, one in each corner of my screen. To do so, I have created four different conkyrc files and I plan on adding four instances of conky to startup, with each one launching a different file.
The trouble is with the alignment parameter: The alignments are set to tl, tr, bl, and br. The tl and tr ones are working properly. However, the bl and br ones display only slightly below where the tl and tr windows start, causing them to overlap and become unreadable. I have also tried setting them to ml and mr instead of bl and br, and that causes them to display even higher, going partially off-screen. This really makes no sense and I'd really appreciate some help with this issue.
I can post a screenshot or the source of my conkyrc files if necessary. Also, if there's a better way to do this four-window setup than having four instances of conky, that'd be great.

Have you tried the unabreviated option (bottom_left, bottom_right)? If that doesn't work, you could always mess with gap_x and gap_y to adjust the windows to where you want them.
Though, are you sure you want four instances running? That's going to take a drain on your cpu, especially if they're all updating frequently. It might be better to set one instance up as full screen with a transparent bg and offset the text to where you want it.
Last edited by TheHebes (2011-12-05 03:38:16)

UC560 SIP Configuration via CCA

We are trying to configure SIP trunks VIA CCA (Latest version) on a UC560 latest version.
However the username and password field doesn't save. When you open CCA SIP trunk configuration the fields are blank, fill them in and save, exist and go back in they are still blank. If you ftp the config off the unit and search for the field "username" only the phones and admins come up - no mention of SIP.
Are we missing something?

Hello,
This sounds like you may have Windows firewall or Anti-virus software running. Also make sure you are using Windows 7 or XP. Windows 8 will not work properly with CCA.
Also if you are using SSL VPN, make sure that only version Anyconnect 2.5.6005 is loaded on the UC. Anything higher will use up the memory and you will not be able to write to the config.
I hope this helps.
Regards,
Chris

Sip configuration

Hi to all,
i have a 1751v and i want to configure for my isp voip.
I have configured registrar and sip server but on the isp router i have a parameter that i don't know how to configure.
This parameter is sip proxy and i don't know hot to configure it on cisco router.
Someone can help me?
Thanks and best regards,
Carlo S.

Thanks for the reply,
in the sip-ua i have insterted the realm because in the isp router i have the sip domain parameter, and i think is the same(is this right?).
I don't have finished the configuration i think i missed the dial-peer for the fxs interface.

SIP Configuration for W995

How can i configure and use SIP in sonny W995.
Basmaa

the standard software by W995 don't have any option to call via SIP!! I don't understand why this phone has such a Otption that you can not use

Sip configuration needed

hi plese any body give me thread that show sip extention adding configuration in uc5400 please help me thank u.

voice register dn 23
number 29
name Lab 1
no-reg
label Lab 1
mwi
voice register dn 24
number 30
name Lab 2
preference 1
no-reg
label Lab 2
mwi
voice register dn 25
number 31
name Lab 3
no-reg
label Lab 3
mwi
voice register pool 23
id mac
type
number 1 dn 23
dtmf-relay rtp-nte
description lab1
call-forward b2bua busy 99
call-forward b2bua noan 99 timeout 10
codec g711ulaw
voice register pool 24
id mac
type
number 1 dn 24 dtmf-relay rtp-nte description lab2 call-forward b2bua busy 99 call-forward b2bua noan 99 timeout 10 codec g711ulaw ! voice register pool 25
id mac
type
number 1 dn 25 dtmf-relay rtp-nte description Lab3 call-forward b2bua busy 99 call-forward b2bua noan 99 timeout 10 codec g711ulaw
Hope this information helps you configure your phones.
Cheers,
Farbod
PS. Please rate this post if you find it helpful. This way we will be more motivated to answer questions.

SIP configuration trouble

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