SIP Call Not Working

Hello Everyone,
I have two CME sites connected via P2P. I have configured dial peer for calling between two sites. Calls from one site are working but from other site I am not able to make calls. Anyone can help me with this issue?
I am attaching debug ccsip all herewith.
Thanks

Hi Carlo,
This is the output of the debug from where calls are not working. There is no output on the other router.
Log Buffer (6000000 bytes):
001892: Mar 2 2015 11:44:57.370 IST: %SYS-5-CONFIG_I: Configured from console by TSAL.ADMIN on vty0 (192.168.100.12)
001893: Mar 2 2015 11:45:08.450 IST: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=, Called Number=1039, Peer Info Type=DIALPEER_INFO_SPEECH
001894: Mar 2 2015 11:45:08.450 IST: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=1039
001895: Mar 2 2015 11:45:08.450 IST: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
001896: Mar 2 2015 11:45:08.450 IST: //-1/xxxxxxxxxxxx/DPM/dpMatchSafModulePlugin:
   dialstring=NULL, saf_enabled=0, saf_dndb_lookup=0, dp_result=0
001897: Mar 2 2015 11:45:08.450 IST: //-1/xxxxxxxxxxxx/DPM/dpMatchPeers:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=6
Thanks

Similar Messages

Merge call not working in ios7 on Iphone5

Merge call not working in ios7 on Iphone5 ?

Same here. We have this call center in our company where after i have added a call to merge i have to push no. 4 and dial from the keypad. This has never been possible in ios but it have been possible to create a contact called "Merge" and add number 4 as it's phone number. Now by merging first my clients call to my collegegue and then to this "Merge" contact it has been dropping me out of line and my client and collegue are merged.
Now ios 7 wont let me call to this "Merge" contact's number "4"

Outgoing calls not working on iphone 4S after upgrading to ios 6.0.1

outgoing calls not working on iphone 4S after upgrading to ios 6.0.1.
It always says call back ...
incoming is working
i tried number of sim cards from different networks but same issue
tries resetting network settings but it also didnt worked

Just counter-check the following settings.
1. Settings>General>Set Side Swith To: Rotation Lock
2. Make sure side switch is not in lock position
3. If necessary do a reset a few more times

Calling not working

I continuously try calling and getting called by my friends who live across the state, only to have the call not work for me. I don't get any ringing or the pop-up window, it just shows me that they are in a call. When I do try to join, it sits there for approximately 30 seconds trying to connect and then kicks me out of the call. I have reset Skype, I have checked for updates; I have the most recent one, I'm up to date. They all have no problem calling with each other but I cannot join or make calls. Oddly enough, there is only one person I can call or get calls from out of our group, but I don't receive a notification unless it's from her. I have tried calling echo and it works every time but not with my friends. Please send help or something to try to fix this problem, it is making Skype unuseable.

Glad we found a workaround. On which network and in which country are you roaming? And who is your home network operator?
This would help Nokia to investigate. And it helps those to find the workaround who find this thread via an search on the Internet.

Click to call not working with Firefox 31 in Windo...

Hello All,
I have a problem with Click to Call in my new computer.
Windows 8.1
Skype 6.20.0.104
Click to call 7.3.16540.9015
Firefox 31
Click to Call not working with Firefox.
With IE11 it ok.
When I directly copy the URL "skype:+48xxxxxxxxx?call" into address field of my Firefox, then automatically opening Skype with only 1 window- contacts, but not opened window with call phone, and not started to call.
When I do this same in IE11 , Skype opened 2 windows ( contacts and call phone) and start call to phone number.
I don't know why. because in my second computer ( this same Firefox, Skype, click to call) but Windows 7, Click to Call working with Firefox.
Please help me.
Maybe someone have a solution ?
Best Regards
Jarek
ps. Sorry for my english

Sounds that you have installed a Firefox Beta release and thus are on the beta update channel, see Help > About.<br />
The Beta update channel receives an update twice a week.
You need to install the current release to switch the update channel to release.
Download a fresh Firefox copy and save the file to the desktop.
*Firefox 27: http://www.mozilla.org/en-US/firefox/all.html
If possible uninstall your current Firefox version to cleanup the Windows registry and settings in security software.
*Do NOT remove personal data when you uninstall your current Firefox version, because all profile folders will be removed and you lose personal data like bookmarks and passwords from profiles of other Firefox versions.
Your bookmarks and other personal data are stored in the Firefox profile folder and won't be affected by an uninstall and (re)install, but make sure that "remove personal data" is NOT selected when you uninstall Firefox.

Cucm , Cube via Sip and Sip Trunk to ISP , Outgoing calls not working

Hi
We have issue with the outgoing calls to sip trunk
Below is the config and the debugs
It will be great if you give your thoughts since we have stuck here
My thoughts are:
i see that for unknown reason the called number is going with 4 digits instead of 8 digits
i dont see any sip message comming from ISP
Maybe the call not going there ? to isp trunk? From the trace the call hit the correct dialpeer 888 but i see 4 digits as a called number , but i dodnt understant the reason to translated in 4 digits the called number.Not apply a translation rule for that
confused!!!
Calling Numbner:22324086
Called Number: 23823690
CUCM:192.168.1.241 and 242
CUBE:192.168.1.10
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*$....$/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^$9$$.......$$/ /99\2/
rule 3 /^$2$$.......$$/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
interface FastEthernet0/0
ip address 192.168.1.10 255.255.255.0
duplex auto
speed auto
interface FastEthernet0/1
description **SIP TRUNK WITH CYTA**
ip address 10.249.13.130 255.255.255.252
duplex auto
speed auto
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad
voice service voip
ip address trusted list
ipv4 0.0.0.0 0.0.0.0
dtmf-interworking rtp-nte
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
supplementary-service h450.12
no supplementary-service h225-notify cid-update
no supplementary-service sip moved-temporarily
no supplementary-service sip refer
fax protocol none
no fax-relay sg3-to-g3
h323
sip
registrar server
localhost dns:bbtb.cyta.com.cy
outbound-proxy dns:sbg.bbtb.cyta.com.cy
no update-callerid
early-offer forced
voice class codec 2
codec preference 1 g711alaw
codec preference 2 g711ulaw
codec preference 3 g729br8
codec preference 4 g729r8
voice translation-rule 1
rule 1 /.*$....$/ /\1/
voice translation-rule 3
rule 1 /^9/ //
voice translation-rule 4
rule 1 /\+/ /900/
rule 2 /^$9$$.......$$/ /99\2/
rule 3 /^$2$$.......$$/ /92\2/
rule 4 /^0/ /90/
rule 5 /^1/ /9001/
rule 6 /^3/ /9003/
rule 7 /^4/ /9004/
rule 8 /^5/ /9005/
rule 9 /^6/ /9006/
rule 10 /^7/ /9007/
rule 11 /^8/ /9008/
rule 12 /^9/ /9009/
rule 13 /^2/ /9002/
voice translation-rule 5
rule 1 // /2232/
rule 2 /^9/ //
voice translation-profile SIP_Incoming
translate calling 4
translate called 1
voice translation-profile SIP_Outgoing
translate calling 5
translate called 3
dial-peer voice 889 voip
description **SIP Trunk to CUCM**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.242:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 890 voip
description **SIP Trunk to CUCM2**
destination-pattern 4086
session protocol sipv2
session target ipv4:192.168.1.241:5060
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
no voice-class sip outbound-proxy
voice-class sip bind control source-interface FastEthernet0/0
voice-class sip bind media source-interface FastEthernet0/0
dtmf-relay sip-notify
no vad
dial-peer voice 888 voip
description **SIP Trunk to CYTA OUTGOING**
translation-profile incoming SIP_Incoming
translation-profile outgoing SIP_Outgoing
destination-pattern 9T
session protocol sipv2
session target sip-server
incoming called-number .
voice-class codec 2
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte
no vad

Hi Aok
I change the default value for IPVMS from g711ulaw to g711alaw but the results remained the same
Also i have restarted the IPVMS
SIP-GW#
SIP-GW#
*Mar 5 14:19:57.854: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 102 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Type: application/sdp
Content-Length: 244
v=0
o=CiscoSystemsCCM-SIP 38874 2 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 0.0.0.0
b=TIAS:64000
b=AS:64
t=0 0
m=audio 24784 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=ptime:20
a=inactive
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:57.878: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Supported: 100rel,timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Timestamp: 1362493197
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 262
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3807 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 101
c=IN IP4 10.249.13.130
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.878: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:57.926: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK355253C
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 102 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 213
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 2 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=inactive
*Mar 5 14:19:57.942: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK79816bbd4196
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5774 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 101
c=IN IP4 192.168.1.10
a=inactive
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
*Mar 5 14:19:57.946: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3562A4
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:57 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:57.946: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK798246ab3597
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 102 ACK
Allow-Events: presence
Content-Length: 0
*Mar 5 14:19:58.146: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
INVITE sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Supported: timer,resource-priority,replaces
Min-SE: 1800
User-Agent: Cisco-CUCM9.0
Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY
CSeq: 103 INVITE
Max-Forwards: 70
Expires: 180
Allow-Events: presence
Supported: X-cisco-srtp-fallback
Supported: Geolocation
Session-Expires: 1800;refresher=uac
P-Asserted-Identity:
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Contact:
Content-Length: 0
*Mar 5 14:19:58.158: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Supported: timer,resource-priority,replaces,sdp-anat
Min-SE: 1800
Cisco-Guid: 3698592896-0000065536-0000000107-4043417792
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Timestamp: 1362493198
Contact:
Expires: 60
Allow-Events: telephone-event
Content-Length: 0
*Mar 5 14:19:58.158: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
*Mar 5 14:19:58.218: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Received:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK3571933
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
From: [email protected]>;tag=125E594-5C7
Call-ID: [email protected]
CSeq: 103 INVITE
Contact:
Require: timer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Length: 216
Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, PRACK, REFER, NOTIFY, UPDATE, INFO
Accept: application/media_control+xml
Accept: application/sdp
Accept: application/x-broadworks-call-center+xml
v=0
o=BroadWorks 96335268 3 IN IP4 10.224.42.164
s=-
c=IN IP4 10.224.42.72
t=0 0
m=audio 54932 RTP/AVP 8 18 96 99
a=rtpmap:96 AMR/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
a=sendrecv
*Mar 5 14:19:58.234: //716/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7983739137ab
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
Allow-Events: telephone-event
Contact:
Supported: replaces
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Session-Expires: 1800;refresher=uac
Require: timer
Supported: timer
Content-Type: application/sdp
Content-Length: 283
v=0
o=CiscoSystemsSIP-GW-UserAgent 9410 5775 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 19314 RTP/AVP 8 18 101
c=IN IP4 192.168.1.10
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
*Mar 5 14:19:58.242: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
Received:
ACK sip:[email protected]:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.241:5060;branch=z9hG4bK7985648033f2
From: ;tag=38874~3aaec7ea-69ca-40db-8e65-34e9ff7aa74d-80213498
To: ;tag=125E62C-1354
Date: Tue, 05 Mar 2013 13:52:31 GMT
Call-ID: [email protected]
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: presence
Content-Type: application/sdp
Content-Length: 192
v=0
o=CiscoSystemsCCM-SIP 38874 3 IN IP4 192.168.1.241
s=SIP Call
c=IN IP4 192.168.1.241
t=0 0
m=audio 4000 RTP/AVP 8
a=X-cisco-media:umoh
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendonly
*Mar 5 14:19:58.262: //717/DC740C800000/SIP/Msg/ccsipDisplayMsg:
Sent:
ACK sip:[email protected]:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.249.13.130:5060;branch=z9hG4bK358582
From: [email protected]>;tag=125E594-5C7
To: [email protected]>;tag=h7g4Esbg_945723725-1362491526714
Date: Tue, 05 Mar 2013 14:19:58 GMT
Call-ID: [email protected]
Route:
Max-Forwards: 70
CSeq: 103 ACK
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 259
v=0
o=CiscoSystemsSIP-GW-UserAgent 6506 3808 IN IP4 10.249.13.130
s=SIP Call
c=IN IP4 10.249.13.130
t=0 0
m=audio 19234 RTP/AVP 8 99
c=IN IP4 10.249.13.130
a=sendonly
a=rtpmap:8 PCMA/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=ptime:20
SIP-GW#
SIP-GW#sh voip rtp connections
VoIP RTP active connections :
No. CallId     dstCallId LocalRTP RmtRTP     LocalIP                                RemoteIP
1     716        717        19314    4000     192.168.1.10                           192.168.1.241
2     717        716        19234    54932    10.249.13.130                          10.224.42.72
Found 2 active RTP connections

Lync 2013 PSTN calling not working with Sonus SBC 1000 over TLS and SRTP

Dear All,
We have recently installed Lync 2013 Enterprise Edition with a Pool of 3 FE Servers (MEDIATION COLLOCATED).
We need to implement TLS and SRTP with Sonus SBC 1000. However calls are not routing b/w SBC and Lync.
We are using wild card certificate with multiple SIP Domains as SAN(s), for internal FE servers as well SBC.
Also i would like to mentioned here that inbound and outbound calls are routing properly when we tested it over TCP.
When I move to TLS Only calls from Lync to SBC (outgoing) are working without encryption.
Here are the OCS Logger traces for incoming calls which are not landing on lync:
TL_INFO(TF_PROTOCOL) [1]2C5C.0D30::04/30/2014-14:35:18.020.00026518.020.00026518.020.00026518.020.00026518.020.00026518.020.00026518.020.00026518.020.000265d2
(S4,SipMessage.DataLoggingHelper:sipmessage.cs(774))[3491463749]
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_AE0419>], 10.10.0.11:5067->10.10.7.50:25678
SIP/2.0 400 Bad Request
FROM: "3158222726"<sip:[email protected]>;tag=ac3201ce-4d7
TO: <sip:[email protected]:5067>;epid=D2091CF753;tag=f373543c
CSEQ: 2 INVITE
CALL-ID: [email protected]
VIA: SIP/2.0/TLS 10.10.7.50:5067;branch=z9hG4bK-UX-ac32-01ce-0b14
CONTENT-LENGTH: 0
SERVER: RTCC/5.0.0.0 MediationServer
------------EndOfOutgoing SipMessage
TL_INFO(TF_PROTOCOL) [1]2C5C.0D30::04/30/2014-14:35:18.027.00026518.027.00026518.027.00026518.027.00026518.027.00026518.027.00026518.027.00026518.027.000265d7
(S4,SipMessage.DataLoggingHelper:sipmessage.cs(774))[2666394843]
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_370F030>], 10.10.0.11:58059->10.10.0.13:5061
SERVICE sip:2138797082;[email protected];user=phone SIP/2.0
FROM: <sip:2138797082;[email protected];user=phone>;epid=DCFDB95F4C;tag=17d286a93
TO: <sip:2138797082;[email protected];user=phone>
CSEQ: 3 SERVICE
CALL-ID: de750f98bdd94e908be5f2f975228ff7
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 10.10.0.11:58059;branch=z9hG4bKd47f1d3c
CONTACT: <sip:[email protected];gruu;opaque=srvr:MediationServer:CiGdW3iH5FiI3Qvr3PIKGQAA>
CONTENT-LENGTH: 630
SUPPORTED: gruu-10
USER-AGENT: RTCC/5.0.0.0 MediationServer
CONTENT-TYPE: application/msrtc-reporterror+xml
<?xml version="1.0" encoding="us-ascii"?>
<reportError xmlns="http://schemas.microsoft.com/2006/09/sip/error-reporting">
<error callId="[email protected]" fromUri="sip:3158222726;[email protected];user=phone" toUri="sip:2138797082;[email protected];user=phone" fromTag="ac3201ce-4d7"
toTag="" requestType="INVITE" contentType="application/sdp;call-type=audio" responseCode="400"><diagHeader>10013;reason="Gateway peer in inbound call is not found in topology document or does not depend
on this Mediation Server"</diagHeader><progressReports /></error></reportError>------------EndOfOutgoing SipMessage
Call
Send SMS
Add to Skype
You'll need Skype CreditFree via Skype

@Paul, Thanks for you response.
All ports / IP Add / DNS are defined properly. Telenet on listening port is working.
We are using Public Certificate for 3 Domains (wild card) and same is loaded and verified in SBC
I've not reviewed the OCS logs properly posted above.
What i've found or seems to me is that in a TLS Calls:
After receiving SIP Invite from SBC, mediation server started TLS Negotiation Process b/w Lync 2013 Server Pool and it fails.
SIP Domains:
contoso.com (default)
fabrikam.com
Lync FE Pool (lync.contoso.com
Here are the some more logs.
TL_INFO(TF_PROTOCOL) [0]2DF8.2930::05/01/2014-11:50:31.612.00025e49 (S4,SipMessage.DataLoggingHelper:sipmessage.cs(774))[2716989131]
<<<<<<<<<<<<Incoming SipMessage c=[<SipTlsConnection_103DFE0>], 10.10.0.11:5067<-10.10.7.50:24591
INVITE sip:[email protected]:5067 SIP/2.0
FROM: "3158222726" <sip:[email protected]>;tag=ac3201ce-ae
TO: <sip:[email protected]:5067>
CSEQ: 2 INVITE
CALL-ID: [email protected]
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 10.10.7.50:5067;branch=z9hG4bK-UX-ac32-01ce-010c
CONTACT: <sip:[email protected]:5067;transport=TLS>
CONTENT-LENGTH: 406
SUPPORTED: replaces,update,100rel
USER-AGENT: SONUS SBC1000 3.1.2v293 Sonus SBC
CONTENT-TYPE: application/sdp
ALLOW: INVITE, ACK, CANCEL, BYE, NOTIFY, OPTIONS, REFER, REGISTER, UPDATE, PRACK
P-ASSERTED-IDENTITY: "3158222726" <sip:[email protected]>
v=0
o=SBC 9 1001 IN IP4 10.10.7.50
s=VoipCall
c=IN IP4 10.10.7.50
t=0 0
m=audio 16418 RTP/AVP 8 0 101 13
c=IN IP4 10.10.7.50
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtpmap:13 CN/8000
a=ptime:20
a=tcap:1 RTP/SAVP
a=pcfg:1 t=1
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:pqL6Tke8pVmXPuplJ1G3+Sr9jM97H8R7iBagWzzh|2^31|1:1
a=sendrecv
------------EndOfIncoming SipMessage
TL_INFO(TF_PROTOCOL) [1]2DF8.0E04::05/01/2014-11:50:31.665.00025e8e (S4,SipMessage.DataLoggingHelper:sipmessage.cs(774))[2716989131]
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_103DFE0>], 10.10.0.11:5067->10.10.7.50:24591
SIP/2.0 100 Trying
FROM: "3158222726"<sip:[email protected]>;tag=ac3201ce-ae
TO: <sip:[email protected]:5067>
CSEQ: 2 INVITE
CALL-ID: [email protected]
VIA: SIP/2.0/TLS 10.10.7.50:5067;branch=z9hG4bK-UX-ac32-01ce-010c
CONTENT-LENGTH: 0
------------EndOfOutgoing SipMessage
TL_INFO(TF_CONNECTION) [1]184C.0EFC::05/01/2014-11:50:32.652.00025f32 (SIPStack,SIPAdminLog::WriteConnectionEvent:SIPAdminLog.cpp(454))[946832530] $$begin_record
Severity: information
Text: TLS negotiation started
Local-IP: 10.10.0.11:5061
Peer-IP: 10.10.0.11:52529
Connection-ID: 0x10BE00
Transport: TLS
$$end_record
TL_INFO(TF_PROTOCOL) [1]184C.0EFC::05/01/2014-11:50:32.669.00026236 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[1853494582] $$begin_record
Trace-Correlation-Id: 1853494582
Instance-Id: 425D
Direction: incoming
Peer: 10.10.0.11:52529
Message-Type: request
Start-Line: NEGOTIATE sip:127.0.0.1:5061 SIP/2.0
FROM: <sip:contoso.com>;ms-fe=LYNCFE1.fabrikam.com
TO: <sip:contoso.com>
CALL-ID: aa53739ef9b34b93ba9c97d3ee56cb99
CSEQ: 1 NEGOTIATE
VIA: SIP/2.0/TLS 10.10.0.11:52529
MAX-FORWARDS: 0
CONTENT-LENGTH: 0
SUPPORTED: NewNegotiate
SUPPORTED: ECC
REQUIRE: ms-feature-info
SERVER: RTC/5.0
$$end_record
TL_INFO(TF_CONNECTION) [1]184C.0EFC::05/01/2014-11:50:32.669.0002636e (SIPStack,SIPAdminLog::WriteConnectionEvent:SIPAdminLog.cpp(383))[946832530] $$begin_record
Severity: information
Text: Connection established
Peer-IP: 10.10.0.11:52529
Peer: lync.contoso.com:52529;ms-fe=LYNCFE1.fabrikam.com
Peer-Cert: contoso.com(LYNCFE1.fabrikam.com)
Transport: M-TLS
Data: alertable="yes"
$$end_record
TL_WARN(TF_CONNECTION) [1]184C.0EFC::05/01/2014-11:50:32.669.00026387 (SIPStack,SIPAdminLog::WriteConnectionEvent:SIPAdminLog.cpp(386))[946832530] $$begin_record
Severity: warning
Text: The pool FQDN provided by the peer in its NEGOTIATE feature information does not match the pool configured in CMS for the server FQDN that it provided
Peer-IP: 10.10.0.11:52529
Peer: lync.contoso.com:52529;ms-fe=LYNCFE1.fabrikam.com
Peer-Cert: contoso.com(LYNCFE1.fabrikam.com)
Transport: M-TLS
Data: fqdn="LYNCFE1.fabrikam.com";pool="contoso.com";expected-fqdn="lync.contoso.com";info="Possible server configuration issue"
$$end_record
TL_INFO(TF_DIAG) [1]184C.0EFC::05/01/2014-11:50:32.670.000265be (SIPStack,SIPAdminLog::WriteDiagnosticEvent:SIPAdminLog.cpp(802))[1853494582] $$begin_record
Severity: information
Text: Routed a locally generated response
SIP-Start-Line: SIP/2.0 200 OK
SIP-Call-ID: aa53739ef9b34b93ba9c97d3ee56cb99
SIP-CSeq: 1 NEGOTIATE
Peer: lync.contoso.com:52529;ms-fe=LYNCFE1.fabrikam.com
$$end_record
TL_INFO(TF_PROTOCOL) [1]184C.0EFC::05/01/2014-11:50:32.670.00026615 (SIPStack,SIPAdminLog::ProtocolRecord::Flush:ProtocolRecord.cpp(265))[1853494582] $$begin_record
Trace-Correlation-Id: 1853494582
Instance-Id: 425E
Direction: outgoing;source="local"
Peer: lync.contoso.com:52529;ms-fe=LYNCFE1.fabrikam.com
Message-Type: response
Start-Line: SIP/2.0 200 OK
FROM: <sip:contoso.com>;ms-fe=LYNCFE1.fabrikam.com
To: <sip:contoso.com>;tag=C3A751556F332F7265E9BA2517C878D4
CALL-ID: aa53739ef9b34b93ba9c97d3ee56cb99
CSEQ: 1 NEGOTIATE
Via: SIP/2.0/TLS 10.10.0.11:52529;ms-received-port=52529;ms-received-cid=10BE00
Content-Length: 0
Require: ms-feature-info
Supported: NewNegotiate,OCSNative,ECC,IPv6,TlsRecordSplit
Server: RTC/5.0
$$end_record
TL_INFO(TF_PROTOCOL) [1]2DF8.1078::05/01/2014-11:50:32.671.000266da (S4,SipMessage.DataLoggingHelper:sipmessage.cs(774))[720988281]
>>>>>>>>>>>>Outgoing SipMessage c=[<SipTlsConnection_F8A09B>], 10.10.0.11:52529->10.10.0.11:5061
SERVICE sip:2138797082;[email protected];user=phone SIP/2.0
FROM: <sip:2138797082;[email protected];user=phone>;epid=16FEE4A02E;tag=22fd877f3a
TO: <sip:2138797082;[email protected];user=phone>
CSEQ: 3 SERVICE
CALL-ID: ac0f7bc4cdc94c1dbd0bb51c7c02c890
MAX-FORWARDS: 70
VIA: SIP/2.0/TLS 10.10.0.11:52529;branch=z9hG4bK67a4c9d1
CONTACT: <sip:[email protected];gruu;opaque=srvr:MediationServer:CiGdW3iH5FiI3Qvr3PIKGQAA>
CONTENT-LENGTH: 628
SUPPORTED: gruu-10
USER-AGENT: RTCC/5.0.0.0 MediationServer
CONTENT-TYPE: application/msrtc-reporterror+xml
- <reportError xmlns="http://schemas.microsoft.com/2006/09/sip/error-reporting">
- <error callId="[email protected]"
fromUri="sip:3158222726;[email protected];user=phone"
toUri="sip:2138797082;[email protected];user=phone"
fromTag="ac3201ce-ae"
toTag=""
requestType="INVITE"
contentType="application/sdp;call-type=audio"
responseCode="400">
<diagHeader>10013;reason="Gateway peer in inbound call is not found in topology document or does not depend on this Mediation Server"</diagHeader>
<progressReports/>
- </error>
------------EndOfOutgoing SipMessage

Mac 10.6.8's video calls not working! Please help!

Reposting the text below because there was no reply to the last message. Ack! Please help. Skype, is there a solution yet? Or even just an explanation and a time frame so we know when we can expect video calls to work again? Thanks! ---------------I have a Mac laptop, version 10.6.8. I have the most updated version of Skype available for my computer, Skype version 6.15. For the past few months, I've been unable to use video chat or file sharing. I can receive some files, but cannot send files. Other people can video chat with me and I can see them, but they can't see me. My built-in camera works just fine for other purposes, so I know it's not an issue with my camera. I've also noticed that a lot of Mac users seem to be having trouble with Skype lately. I've checked my audio/video preferences for Skype and my computer's camera preferences, and everything is as it should be. Does anyone know how to fix this?
Thanks in advance for any help!

See: How to perform a "clean install" of Flash Player in Mac OS X
In addition to these steps, I recommend, BEFORE emptying the trash:
Go to: [User]/Library/Preferences (hold the "Option" key when clicking "Go" from the Finder menu to reveal the hidden Library folder)
Trash the ENTIRE Macromedia folder from there
Aside from that additional step, follow the rest of the instructions to the letter and you should be back up and running.

Skype Click to Call Not Working for Landlines and ...

Hello,
I am trying to use Skype Click to Call, but it does not seem to be working. I was able to call contacts using callto://CONTACT_NAME, but the second I try to call any mobile or landline number (I.E. - callto://9731234567), it does not work. Skype will come up, but it will not make the call. I am running on:
Windows XP - 2002 Version
4 core Pentium processor (1.7 GHz)
256 MB of RAM
Google Chrome
If anyone could provide any insight or ideas, I would really appreciate it. I have already tried a large number of things to fix this issue. Thank you so much for any help!

Sounds that you have installed a Firefox Beta release and thus are on the beta update channel, see Help > About.<br />
The Beta update channel receives an update twice a week.
You need to install the current release to switch the update channel to release.
Download a fresh Firefox copy and save the file to the desktop.
*Firefox 27: http://www.mozilla.org/en-US/firefox/all.html
If possible uninstall your current Firefox version to cleanup the Windows registry and settings in security software.
*Do NOT remove personal data when you uninstall your current Firefox version, because all profile folders will be removed and you lose personal data like bookmarks and passwords from profiles of other Firefox versions.
Your bookmarks and other personal data are stored in the Firefox profile folder and won't be affected by an uninstall and (re)install, but make sure that "remove personal data" is NOT selected when you uninstall Firefox.

Function call not working - why?

Hmmmm. I just don't see why this is not working. I have an application with a new class I just created. The class loads, but will not call it's own internal function.
package com.parkerandkent.components.classic.photogallery {
     import caurina.transitions.Tweener;
     import flash.display.MovieClip;
     import flash.events.Event;
     import flash.events.MouseEvent;
     public class CallTag extends MovieClip {
          trace ("test1");
          init();
          private function init():void {
               trace ("test2");
               b_arrow.buttonMode = true;
               b_arrow.addEventListener(MouseEvent.CLICK, arrowMenuCLICK);
          private function arrowMenuCLICK():void {
"Test 2" will not fire here. And I get this error message:
CallTag.as , Line 10        1180: Call to a possibly undefined method init.

package com.parkerandkent.components.classic.photogallery {
     import caurina.transitions.Tweener;
     import flash.display.MovieClip;
     import flash.events.Event;
     import flash.events.MouseEvent;
     public class CallTag extends MovieClip {
function CallTag(){
          trace ("test1");
          init();
          private function init():void {
               trace ("test2");
               b_arrow.buttonMode = true;
               b_arrow.addEventListener(MouseEvent.CLICK, arrowMenuCLICK);
          private function arrowMenuCLICK():void {

Merge call not working

I tried to initiate a 3 way call using an Apple 5s, and it did not work. I am traveling in Toronto, could that have something to do with it?

@DanPetrie
Let's see what 3 way / Conference calling options are available to you. Please share more details of what you experienced. Were you on an existing call, trying to place another call to conference or were you attempting to accept a second incoming call to merge? Please see more details about Call Conferencing in the Device Manual here: http://bit.ly/1ruLn6H on page 52. Thanks!
AnthonyTa_VZW
Follow us on Twitter @VZWSupport

Skype Calls Not Working

I always used skype and it worked perfectly fine. But about a month ago, calls stopped working. When I tried calling someone it would say Connecting, but the timer would stay at 00:00. Then after the call ending/going to voice mail, it would tell the person that there was a missed call. However, they wouldn't get a notification popping up saying that I was trying to call. Same for me when they tried calling it wouldn't work and I will see that there was a missed call even though I was looking at my screen when they were trying to call. There would be no window popping up asking to accept the call as well. I tried doing stuff like deleting the Db temp files and renaming the skype program folder and it did not help. I need help with this because it is so frustrating. I even tried making new account and it would keep doing this.

Have you got this question resolved ever since? I've got the same problem with my iphone 5S. I've bought it a month ago as part of a 2 year contract with Orange France. My plan has 2Gb data on 4G. But only when I'm on wifi does any of those work (viber, facetime, skype). Once on data plan, no matter where I am in Paris, I can't hear people answering my facetime/skype/viber calls and they can't hear me either. Even though the status shows 4G (or sometimes 3G). I've tried disabling 4G and forcing 3G, the same. I've got the iphone replaced, then the SIM card replaced, the problem stays. I still have more than 1Gb left from my data so that's certainly not the problem. All softwares are updated (certainly downloaded, re-installed everything several times, phone factory reset, etc). I pay like 45 euros a month so I can't imagine this would be a 'simple plan' or maybe my VoIP got somehow disabled by mistake? I've been to 5 Orange stores and they sent me away that it must be application problem they don't know what's wrong. As I said, Apple got my iphone even replaced but didn't have more suggestions when it still didn't work. I'm kinda hopeless, doomed to go for 2 years with an iPhone 5S that can't use any audio communication applications from its data plan. Which is ridiculous cause that's one of the main reason I've bought a 5S, I was told this one can do Facetime on data plan too, not just on wifi.

Messaging, FaceTime and calling not working

I am having trouble sending iMessages, using FaceTime and calling only one contact. Each time I send an iMessage it says not delivered, when I try to call using FaceTime it pops up that the contact is not available for FaceTime and when I call the contact it goes to voicemail. Could they have me blocked?

Hi,
Have you allowed either app through the MAc Firewall ?
(System, Preferences > Security > Firewall)
Are either app allowed though your Router ?
In fact what exactly do you mean by "Not Working" ?
For Instance
Face Time Launched
Go to the Face Time Menu > Preferecnes
Is it set to ON ?
9:29 PM Sunday; August 21, 2011
Please, if posting Logs, do not post any Log info after the line "Binary Images for iChat"
 G4/1GhzDual MDD (Leopard 10.5.8)
 MacBookPro 2Gb( 10.6.8)
 Mac OS X (10.6.8),
"Limit the Logs to the Bits above Binary Images."  No, Seriously

JTOpen: Program Call Not Working

I can't seem to get the program sample from the following site working from my project within Creator.
http://www.itjungle.com/mpo/mpo011702-story04.html
The code works as a stand alone program called from command line. However when I add the same code to my java class it will not work. The code is being accessed from the Application Server. This is the exception that is returned;
java.security.AccessControlException: access denied (java.lang.RuntimePermission createClassLoader)

The permission needs to be given to the code that is trying to create a class loader. Permissions are set in the server.policy file for the appserver.
- Edit <creatordir>//SunAppServer8>//domains/creator/config/server.policy file.
- Add the line "permission java.lang.RuntimePermission "createClassLoader";" before the line containing " permission java.lang.RuntimePermission "loadLibrary.*";".
The above will give the permission to all code; you can restrict the premission by granting permission to only some apps by using 'grant codebase' sections instead of the global grant section...

Video Group call not working

Hello! I have recently purchased a laptop with Windows 8.1 with pre-installed Skype application. When When I video-call a single person, everything is working normally, and I am able to see the other person. However, when making a group call, I am the only one that cannot see neither of the parties, though I am able to hear them. I do have the latest Skype application. Also, I cannot recieve pictures. Where is the problem?
Solved!
Go to Solution.

Hi there, I am having a similar problem. When I make group video calls (on my PC with Windows 8.1), the people on macs or iphones can only be heard but not seen. And I get a message on my screen that says something like their version doesn't support group video? But I've done everything the forums says:
1) they have the latest iPhone and just downloaded Skype which is the newest version (no older versions here)
2) I'm the one hosting the call, so they're just joining in (because I know iPhone can't host, but they're supposed to be able to join group video chats).
3) I've added their name to my contacts list and dragged their contact into the group call box to initiate the call (so they're not joining in later, they're there at the start of the call).
4) And we tested prior to the group call to make sure that one-on-one video calling is working.
5) Everyone else on the call who are on a PC has the group video chat working fine.
6) Only calling with 3-4 people in a group.
What am I missing? Please help.

SIP Call Not Working

Similar Messages

Maybe you are looking for