Outbound calls not working

Hello,
We have a E1 R2 and outbound calls are not being established. Inbound calls are ok.
ios:c2951-universalk9-mz.SPA.152-4.M7.bin
Call Flow: IP Phone -> CUCM -> SIP TRUNK -> Voice Gateway -> E1 R2-> TELCO
The debug vpm signal is attached.
What could be the issue?
I have already tested this E1 R2 link in two differents routers.
Regards
Leonardo Santana

Hi Dennis,
I agree with you, this is strange.
The number dialed was 2217-8631 from extension 3784
E_HTSP_SETUP_REQ DNIS=22178631 ANI=3784
Jan 19 00:18:07.900: r2_reg_generate_digits(0/0/0:0(1)): Tx digit '3'
Jan 19 00:18:08.000: htsp_digit_ready_up(0/0/0:0(1)): Rx digit='1'
Jan 19 00:18:08.000: R2 Outgoing Voice(0/0): DSX (E1 0/0/0:0): STATE: R2_OUT_PROCESS_A R2 Got Event 1
Jan 19 00:18:08.000: r2_reg_generate_digits(0/0/0:0(1)): Tx digit '#'
Jan 19 00:18:08.120: htsp_digit_ready(0/0/0:0(1)): Rx digit='#'
Jan 19 00:18:08.120: R2 Outgoing Voice(0/0): DSX (E1 0/0/0:0): STATE: R2_OUT_PROCESS_A R2 Got Event R2_TONE_OFF
Jan 19 00:18:08.120: r2_reg_generate_digits(0/0/0:0(1)): Tx digit '1'
Jan 19 00:18:08.220: htsp_digit_ready_up(0/0/0:0(1)): Rx digit='1'
Jan 19 00:18:08.220: R2 Outgoing Voice(0/0): DSX (E1 0/0/0:0): STATE: R2_OUT_PROCESS_A R2 Got Event 1
Jan 19 00:18:08.220: r2_reg_generate_digits(0/0/0:0(1)): Tx digit '#'
Jan 19 00:18:08.340: htsp_digit_ready(0/0/0:0(1)): Rx digit='#'
Jan 19 00:18:08.340: R2 Outgoing Voice(0/0): DSX (E1 0/0/0:0): STATE: R2_OUT_PROCESS_A R2 Got Event R2_TONE_OFF
Jan 19 00:18:08.340: r2_reg_generate_digits(0/0/0:0(1)): Tx digit '*'
Jan 19 00:18:08.448: htsp_process_event: [0/0/0:0(1), R2_Q421_OG_SEIZE_ACK, E_DSP_SIG_1000]
Jan 19 00:18:08.448: r2_q421_seize_to(0/0/0:0(1)) Tx CLEAR FWDvnm_dsp_set_sig_state:[R2 Q.421 0/0/0:0(1)] set signal state = 0x8
Jan 19 00:18:08.448: htsp_timer - 1000 msec
Jan 19 00:18:08.448: r2_reg_channel_disconnected(0/0/0:0(1))

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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
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FROM: "3158222726"<sip:[email protected]>;tag=ac3201ce-4d7
TO: <sip:[email protected]:5067>;epid=D2091CF753;tag=f373543c
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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
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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
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FROM: <sip:2138797082;[email protected];user=phone>;epid=DCFDB95F4C;tag=17d286a93
TO: <sip:2138797082;[email protected];user=phone>
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CALL-ID: de750f98bdd94e908be5f2f975228ff7
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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
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@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

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does the outbound call work through the same gateway?

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 {

Incoming call not working

Hello Guys,
I have an issue with the incoming calls. We I dial the 10 digitss number using my Cell phone, it doesn't go through. I receive a message stating "the call can't go through". But we can call from inside to outside and within the company.
Also when I try to add a dial-peer, an error message stating "Could not add peer" appears.
Could you please help me out?
The gateway is a 3640 H.323.
This is the output of the debug isdn q931. I am dialing 617-717-7982.
Oct 26 2006 02:23:56.129: ISDN Se1/0:23 Q931: RX <- SETUP pd = 8 callref = 0x00E1
Bearer Capability i = 0x8090
Standard = CCITT
Transer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xE1808386
Preferred, Interface 0, Channel 6
Net Specific Fac i = 0x00E7
Calling Party Number i = 0x2181, '6178692545'
Plan:ISDN, Type:National
Called Party Number i = 0xA1, '7982'
Plan:ISDN, Type:National
Oct 26 2006 02:23:56.169: ISDN Se1/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x80E1
Channel ID i = 0xA98386
Exclusive, Channel 6
Oct 26 2006 02:23:56.253: ISDN Se1/0:23 Q931: TX -> DISCONNECT pd = 8 callref = 0x80E1
Cause i = 0x80A6 - Network out of order
Oct 26 2006 02:23:56.301: ISDN Se1/0:23 Q931: RX <- RELEASE pd = 8 callref = 0x00E1
Oct 26 2006 02:23:56.309: ISDN Se1/0:23 Q931: TX -> RELEASE_COMP pd = 8 callref = 0x80E1
Thanks,

Thanks for the reply,
My question is why are we able to make outbound call if the session-targat address is wrong. Please consider the following two Dial-peer. For a call coming in from PSTN, which Dial-peer is going to be used? Accroding to me the Dial-peer voice 1001 will be used because the call is coming from PSTN therefore it going to use the PRI. Am I right?
dial-peer voice 7000 voip
preference 1
destination-pattern 7577
session target ipv4:172.16.165.10
incoming called-number .
dtmf-relay h245-alphanumeric
codec g711ulaw
no vad
dial-peer voice 1001 pots
preference 1
destination-pattern [2-9].........
direct-inward-dial
port 1/0:23
forward-digits all
Thanks,

Outbound calls not working

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