AS5350 with SIP Server (Asterisk)

Having problem to route calls form sip server to E1 port of AS5350. can anybody help me with come sample configuration for AS5350 with SIP Server Peer.
Thanks

Well, for one is better not to write configuration out of memory, as you have some obvious mistake like "sipv3" and incoming "dialed" number.
Anyway. For the incoming pots DP, can you configure a better match than "incoming called-number .T" ? That would give a little more control and make sure the number is hitting the DP (however, it should already) eg, numbers starting with 0 are six digits, in accordance with your contry dialplan. You can classify numbers like local, long-distance, cellphone, etc.
The same pattern goes under dial-peer pots as destination-pattern as you did.
Then, please do "term mon" and "debug ccsip message" and "debug isdn q931". Post the result here. It is possbile the calls goes to PST but is refused from there.

Similar Messages

Registering with SIP server using advanced IP services

I have been able to make my lab router send registration messages to a SIP server. Is it a requirement to run from a CUBE IOS? Software info below. I am attempting to register using the information below.
sip-ua
authentication username 8459997008 password xxxxxxxxxxxx realm voip.sotel.systems.com
retry invite 3
retry register 10
registrar ipv4:4.28.93.140 expires 3600
voice service voip
redirect ip2ip
sip
Any help would be great....

Whats your current router model, IOS version? If you are on IOS 15.X you will need UC technology package installed (eval or permanent), CUBE needs licenses but its a Right To Use (RTU) license and will not stop you from configuring.
If you are on IOS 15.X and able to enter voice commands that means you already have proper IOS.
Still if you are unsure please post:
Sh version
-Terry
Edit: I overlooked the title of your post saying advanced services so appears you are on 12.X if thats the case see below:
Generally, for the Cisco 2800 and 3800 Series platforms, the Cisco Unified Border Element is supported on all Cisco IOS Software images of IP Voice and above. Specifically
1) INT VOICE/VIDEO, IPIP GW, TDMIP GW
2) INT VOICE/VIDEO, IPIPGW, TDMIP GW AES
3) INT VOICE/VIDEO GK, IPIPGW, TDMIP GW AES, LI

Link cocomo with a SIP VOIP-server (Asterisk)

any ideas/suggestions about communicating between cocomo and
a sip server (asterisk based) ?

Hi Leonard,
I was interested in doing something similar, and used Ribbit Flex SDK to make calls directly from my Flash app to a phone. It worked OK (although connecting the call was rather slow), but if the Flash app user did not wear a headset then the person on the phone would have an echo (as the sound from my laptop speakers was being fed back into the mic). This is because the Flash Player does not have Advanced Echo Cancellation built in (Skype, Adobe Connect and others have AEC built in).
If you'd like to vote for AEC to be built into Flash Player, the Adobe Bug is written up here (currently has 161 votes, open over a year but no comment from Adobe). This is a shame, as I think with AEC lots of people could write some killer apps that used the Flash Player.
thanks
Mark

Problems between an UC520 and Asterisk with sip trunk

I have an UC520 and Asterisk with a sip trunk created between them, the calls from the UC520 to the Asterisk are ok, but the calls form de Asterisk to the UC520 are always busy.
Logs from the asterisk show that the first part of the call is ok, but the call is not complete, this means that the part where the extensions are with @ipuc520 doesn't appear
I created a sip trunk from de CCA 1.9 and it puts this for incoming calls for the dial peer, if I compare with a CCME, there is no configuration for incoming call there
/* Style Definitions */
table.MsoNormalTable
{mso-style-name:"Tabla normal";
mso-tstyle-rowband-size:0;
mso-tstyle-colband-size:0;
mso-style-noshow:yes;
mso-style-priority:99;
mso-style-qformat:yes;
mso-style-parent:"";
mso-padding-alt:0cm 5.4pt 0cm 5.4pt;
mso-para-margin:0cm;
mso-para-margin-bottom:.0001pt;
mso-pagination:widow-orphan;
font-size:11.0pt;
font-family:"Calibri","sans-serif";
mso-ascii-font-family:Calibri;
mso-ascii-theme-font:minor-latin;
mso-fareast-font-family:Calibri;
mso-fareast-theme-font:minor-latin;
mso-hansi-font-family:Calibri;
mso-hansi-theme-font:minor-latin;
mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;
mso-fareast-language:EN-US;}
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target ipv4:x.y.z.w
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
And there is no configurarion at all that could block the calls
The x.y.z.w was the sip server ip (asterisk ip)
The comminication between sip and h323 are allowed in the four ways
The allowed codecs are   g711ulaw and g729r8
Asterisk is working now with other CCME and they are ok so I copied the configuration from those CCME to the UC520 and from the other sip trunks in asterisk the new trunk sip for uc520
The sip trunk created from the CCA was replaces for the one from the CCME that is working now
The routes are ok in Asterisk.
There is no translation profile in incoming calls.
There is no ACL applied in all configuration.
There is no log about callres incoming from the asterisk.
Could anyone halp me pls?

Hi Rina,
Help me to try and understand what you are trying to do.
In this code snippet i see the following:
001808: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Calling Number=7129, Called Number=7129, Peer Info Type=DIALPEER_INFO_SPEECH
001809: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Match Rule=DP_MATCH_DEST; Called Number=7129
001810: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersCore:
   Result=Success(0) after DP_MATCH_DEST
001811: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpMatchPeersMoreArg:
   Result=SUCCESS(0)
   List of Matched Outgoing Dial-peer(s):
     1: Dial-peer Tag=20036
This looks as though you have a call coming in from the Asterisk system to number 7129, which then leads to this according to the config file you provided.
number 7129
label 7129
description7129
name 7129
call-forward busy 6001
call-forward noan 6001 timeout 10
Which at this point I am going to assume this is ephone-dn 10 (Please confirm). If this is the case then the inbound call is being matched correctly to a DN (Which has its own dial-peer tag "Dial-peer Tag=20036".
But then i see this:
001817: 1w3d: //-1/55940098BA19/DPM/dpAssociateIncomingPeerCore:
   Result=Success(0) after DP_MATCH_INCOMING_DNIS; Incoming Dial-peer=1000
001818: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Calling Number=Unknown, Called Number=, Voice-Interface=0x0,
   Timeout=TRUE, Peer Encap Type=ENCAP_VOIP, Peer Search Type=PEER_TYPE_VOICE,
   Peer Info Type=DIALPEER_INFO_SPEECH
001819: 1w3d: //-1/xxxxxxxxxxxx/DPM/dpAssociateIncomingPeerCore:
   Result=NO_MATCH(-1) After All Match Rules Attempt
So the incoming call has been matched to Dial-peer 1000 which is an incoming VoIP dial-peer:
dial-peer voice 1000 voip
permission term
description ** Incoming call from SIP trunk (Generic SIP Trunk Provider) **
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
session protocol sipv2
session target sip-server
incoming called-number .%
dtmf-relay rtp-nte
ip qos dscp cs5 media
ip qos dscp cs4 signaling
no vad
But then can see it has no where to go. So either I am reading this all wrong and the 7129 number is a result of another call taking place whilst you were debugging the system, or it is part of the debug and I am missing something here.
Rina, just so I understand this all. Are you trying to do WAN type calling from one system UC-500 (System "A") to the Asterisk system ( System B) and same? And so far calls going from the UC-500 to the Asterisk system are fine, but calls coming in from the Asterisk system to the UC-500 are not?
What happens on the Asterisk side when you try to call an Extension on the UC-500, do you get any ringing? Or is it a fast busy tone?
I am going to look over your configuration and debug a little further when I get home, maybe I am missing something here and can identify it.
Cheers,
David.

P2P SPA3102 setup with no SIP server/service provider

Description:
- Peer-to-peer ATA connection with no service provider or SIP server
Location A (my location):
- Linksys SPA3102
- PABX analog device (Panasonic)
- Public and static IP
Location B (my partner location):
- D-Link DVG2001S
- Public IP
Questions:
- Can B ATA (D-Link) call to A ATA (Linksys) being different brands?
- Can I pick up the B ATA attached phone and directly to have PABX internal dial tone at A, for other extension or external calls?
- From PABX other extensions, can I call to SPA3102 extension number and directly redirect the call to B ATA?
I was read many .pdf documents about SPA setup, and try differents configuration, but I'cant make the communication with my partner location happens.
I think my problem is understanding and writing the correct PSTN LINE and LINE 1 Dial Plan parameters in my SPA3102.
Can I make it happens?
Thanks in advance,
Bitman
Message Edited by Bitman on 07-28-2007 05:37 AM

Good day! I just hope this is not one of those “.pdf documents” that you have already read but basically this should work if you’re setting up two SPA3000 or SPA3102 since they have exactly the same Voice configuration:
http://www.provu.co.uk/pdf/sipura/spa_backtoback_2x_spa3000.pdf
I’m guessing that IP dialing should work on your two units if they both support SIP, however, in this case, I haven’t tried setting up a D-Link before ( or you may try to wait for other users who happen to have a D-link also to post suggestions) , you may just want to try looking for documents on setting up IP dialing on D-link then try to match it’s IP dialing parameters with the SPA.

IPVC 3511 MCU with SIP : Regitration problem with LCS server

Hello,
We are trying to make visioConference with H323 and SIP client wuth IPVC MCU3511.
We use IPVC MCU 3511 version 4.0.31 (i've also tested 3.5.33 it's the same) and Microsoft Live Communication Server (LCS) 2005.
We are not able to register the 3511 MCU with the LCS server.
The LSC server refuse the registration because the "Epid" value is not set by the 3511 MCU : the 3511 MCU receive the message "SIP/2.0 400 Epid Not Present"
Have you any experience with the 3511 MCU and SIP ?
Thanks for your help,
Mickaël.
Below an extract of the 3511MCU logs :
Request from the MCU the the SIP server :
16425 INFO Adap Sip |DEBUG - MSGBUILD - TransportTCPSend - TCP message Sent on Connection 2fbe27c, 5060
16425 INFO Adap Sip |INFO - TRANSPOR - --> REGISTER sip:chu-grenoble.fr;transport=TCP SIP/2.0
16425 INFO Adap Sip |INFO - TRANSPOR - From: <sip:[email protected]> ;tag=ac1164bf-13c4-425b9e73-2819a-4a5a
16425 INFO Adap Sip |INFO - TRANSPOR - To: <sip:[email protected]>
16425 INFO Adap Sip |INFO - TRANSPOR - Call-ID: ac1164bf-13c4-425b9e73-2817c-1a05
16425 INFO Adap Sip |INFO - TRANSPOR - CSeq: 1 REGISTER
16425 INFO Adap Sip |INFO - TRANSPOR - Via: SIP/2.0/TCP 172.17.100.191:5060 ;branch=z9hG4bK-425b9e73-2819a-448b
16425 INFO Adap Sip |INFO - TRANSPOR - Max-Forwards: 70
16425 INFO Adap Sip |INFO - TRANSPOR - Contact: <sip:172.17.100.191:5060;transport=UDP>
16425 INFO Adap Sip |INFO - TRANSPOR - Expires: 3600
16425 INFO Adap Sip |INFO - TRANSPOR - User-Agent: RADVision ViaIp MCU Vers. 3.6
16425 INFO Adap Sip |INFO - TRANSPOR - Content-Length:0
Response from the SIP server to the MCU :
16426 INFO Adap Sip |DEBUG - MSGBUILD - TransportMsgBuilderTcpMakeMsg - TCP message Arrived on Connection 2fbe27c, port 5060
16426 INFO Adap Sip |INFO - TRANSPOR - <-- SIP/2.0 400 Epid Not Present
16426 INFO Adap Sip |INFO - TRANSPOR - Via: SIP/2.0/TCP 172.17.100.191:5060 ;branch=z9hG4bK-425b9e73-2819a-448b;ms-received-port=1027;ms-received-cid=c00
16426 INFO Adap Sip |INFO - TRANSPOR - From: <sip:[email protected]> ;tag=ac1164bf-13c4-425b9e73-2819a-4a5a
16426 INFO Adap Sip |INFO - TRANSPOR - To: <sip:[email protected]>;tag=BF8BA39117943996592C71550C3E1940
16426 INFO Adap Sip |INFO - TRANSPOR - Call-ID: ac1164bf-13c4-425b9e73-2817c-1a05
16426 INFO Adap Sip |INFO - TRANSPOR - CSeq: 1 REGISTER
16426 INFO Adap Sip |INFO - TRANSPOR - Content-Length: 0

Figure this out. Just need to put all the info (including SIP domain name, even if you are not running it) into via "Setup Wizard" and it will goes away.

"Server Header" with Weblogic SIP Server 2.2

Hi All,
I'm using Weblogic SIP Server 2.2, based on Weblogic Server 8.1 SP5, and I want to desactivate the "Send Server Header" option.
I try to do it by using the console like describe in http://e-docs.bea.com/wls/docs81/ConsoleHelp/domain_server_protocols_http.html page. The config.xml file is modified ( <i><WebServer Name="myserver" SendServerHeaderEnabled="false"/></i>), but all response messages still contain the server header.
This header adds information about the version of WebLogic Server that is running. Which is a security leak.
It seem's that this problem is solved since Weblogic Server 8.1 SP4. But I don't succeed in turning the option off.
Any idea about this issue?
Thanks,
Karine.

Hi,
This problem is resolved by installing the new version of WebLogic SIP server (3.0 prÃ©views).
Thanks,
Karine.

Configurating SPA509G with SIP

Hi,
I just bought a new SPA509G phone. I updated to the newest software (7.5.5) and I have it configurated the phone to work with our "local" Asterisk system.
Requirement however is to work with some external sip accounts also, like my bosses home.
We do have this working with a Siemens Gigaset, so I am trying to port these settings to the SPA509G. I have tried for hours but I can't get it to work.
Even worse, when enable this extension, the phone on rebooting always gets stuck at "Checking DNS". When I take it out again, everything works.
The settings that we have on the Gigaset are:
Account Name: 622
Password: ***
User Name: 622
Domain: fritz.box
proxy-server: ***.dyndns.org
Server-port: 5060
registrar-server: ***.dyndns.org
registrar-server-port: 5060
registration-refresh-time: 180
use stun: yes
stun-server: stunserver.org
outbound-proxy-mode: auto
If I get this correct, Stun settings are not on an per extension basis on the SPA500 but global. That might be one problem.
Also, when I put the ***.dyndns.org:5060 in the proxy or Outbound Proxy fiel, I get this “DNS” hangup.
Any idea how to port the above settings to the SPA509G ?
THANK YOU VERY MUCH
Stephan

Any idea how I could get help for this?

What is the problem between NAT/PAT-ed network with SIP?

Hi guys,
I'm not really good at voice - so please bare with me :)
I have a situation where I cant make a voip call via SIP using class4/5 softswitch behind NAT/PAT network.
The diagram :
NAT/PAT --- cloud/MPLS --- softswitch.
the softswitch provides IP centrex service - so there will be caller-group. the 2nd problem was that in a caller-group It cant establish a call origin from ip 1.1 back to ip 1.1. And i cant touch that softswitch (its xener - i dont exactly know what type). I'm wondering this softswitch capability - anyone using it?.
We have tested using other SIP server (using asterisk-based softswitch) and sniffed all SIP-related traffic - we have 403 error and the like - but my opinion its the PEs NAT router that dropped the SIP handshake - so the RTP wont pass-thru both caller/called party.
Modifying a single PE probably easy - but my catch is that - as long as I have some NAT router/firewall along the PE and softswitch path it will not work, correct?
Before i go further with Cisco Unified Border Element and Session Border Controller proposal - anyone would like to give me a comment about my understanding from above scenario?
any help would be appreciated,
thanks.

The NAT Support for SIP feature allows SIP embedded messages passing through a router configured with Network Address Translation (NAT) to be translated and encoded back to the packet. An application layer gateway (ALG) is used with NAT to translate the SIP or SDP messages.
See the following url for more details about NAT support for SIP:
http://www.cisco.com/en/US/docs/ios/12_2t/12_2t8/feature/guide/ftnatsip.html

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.

Multiple registration with sip-ua

Hi,
someone know a way to do multiple registration with a single 2811 using sip-ua configuration with multiple accounts??
thnx
s.

Hi, I have got to authenticate more than one account in the SIP provider with the hidden command "credentials" the problem that I have now is how to route all the calls done to the second account to the extension 101.
I want that incoming calls from 964812530 goes to extension 100 and incoming calls from 965072519 goes to extension 101
How can I do it?
I have tried this but it's not running:
sip-ua
credentials username 965072510 password 115849534F43415C557B7967 realm beta.awa
voz.com
authentication username 964812539 password 13544744535D4E7A7A757A70
no remote-party-id
retry invite 4
retry response 3
retry bye 2
retry cancel 2
retry register 5
timers register 250
registrar ipv4:213.162.201.146 expires 60
sip-server ipv4:213.162.201.146
voice service voip
sip
voice translation-rule 1
rule 1 /1../ /964812539/
voice translation-rule 3
rule 1 /964812539/ /100/
voice translation-rule 4
rule 1 /965072510/ /101/
voice translation-profile SIPout
translate calling 1
voice translation-profile incoming
translate called 3
voice translation-profile incoming2
translate called 4
voip-incoming translation-profile incoming
All the incoming calls are going to extension 100
regards

Deploying war file in weblogic SIP server 2.2

Hi,
I trying to deploy a "sar" file in weblogic SIP server,but it is not allowing me to do that.So one of my collegue told me to change the "sar" to "war" and try to deploy.
When i am trying to deploy that it is saying that unable to find deployment descripter WEB-INF/web.xml in the war file.Actually it is there.But i don't know why it is giving that error.
Actually i need to deploy "sar" file and i changed that file into "war" file
What are all the trail and error to solve this problem
(1)I made the "war" file from file folders.
suppose my application directory name is "myapp"
then i am going into the "myapp" folder and from there i am trying to make war file and it is having all the related class files , sip.xml file , web.xml file.
Then i tryed to deploy that file from weblogic SIP server 2.2.console.
(2) Next I included the weblogic.xml file also into the war file.
(3)The "sar" file working fine in SDS 3.1,but not in we blogic SIP server.
(4)The SDS3.1 is working with JRE 5 so i changed that JRE from 1.5 to jre 1.4. and again prepared the sar file and changed that SAR file to WAR file.
I created a domain in SIP server which was created in "user_projects."
For e.g., the domain name is "mydomain" and
the directory structure is
mydomain
|
|
|----------applications(dir)
|
|----------myserver(dir)
|
|----------rmfilestore(dir)
|
|----------sipserver(dir)
|
|-----------boot (file)
|
|-----------config.xml
|
|-----------startWeblogic
|
|-----------stopWebLogic
|
|----------upload-------xxxxmanager----META-INF---WEB-INF----log4j.properties
--------------sip.xml
-----------------web.xml
-------------weblogic.xml
web.xml, sip.xml,log4j.properties are under WEB-INF folder
Do i need to change any path setting in startWebLogic.cmd file.I didn't change any thing in the startWeblogic.cmd.
Since my war file is having all the related files like all the related "jar" files in that.
Any body's help will be appriciated.
Thanks
vish

Can any body please reply for above question.
Hi BEA support please flash some light on my above question
vkviswanadh

How to reset password of weblogic sip server 3.1

Hi ,
Can anybody help me in resetting the username/password of weblogic sip server 3.1
Thanks a lot
Regards
Saurabh

Hi,
If we have only one admin account available, I am afraid there is no means to reset the password other than reinstalling Windows. See this article as a reference:
What to do if you forget your Windows password
Besides, we may try to create the Windows 8.1 install media with the following article:
Create installation media for
Windows 8.1
How to perform a clean installation of Windows
And please note, if we do a clean install, we need to activate Windows again.
If Windows 8 came preinstalled on your computer, your product key should be on a sticker on your computer or with your documentation. Or we may contact the manufacturer for further help.
Best regards
Michael Shao
TechNet Community Support

Lync HP 4120 Sign in problems with Lync Server 2013

Hi, this is my second request for help, this with more information...
I`ll ready install the follow infrastructure: (I change the name of my organization for contoso)
Lync Server 2013 Installation with Enterprise mode with 1 front end : Pool: lync.contoso.com Front End: lyncfe01.contoso.com Back End: lyncsql01.contoso.com
I`ll ready install a PKI infrastructure with two tiers, the root offline and the subordnate ac.contoso.com
This with the defailt algorithm configuration RSA SHA1
My phones are HP 4120
In the Front End Server i configured the SCHANNEL registers:
EnableSessionTicket in 2
Send..etc in 0
Ok, the installation is ok, services are OK, Client login trough PC its OK, PSTN Configuration... (I can make a phonecall with the lync client of Office 365)
Commnd Checks:
When i run the command Test-CsPhoneBootstrap -PhoneOrExt 12345 -PIN 123456 -TargetFqdn lync.contoso.com
The result is:
Target Fqdn : lync.contoso.com
Target Uri : https://lync.contoso.com:443/CertProv/CertProvisioningService.svc
Result : Success
Latency : 00:00:09.0559615
Error Message :
Diagnosis :
When i run the command Test-CsPhoneBootstrap -PhoneOrExt 12345 -PIN 123456 for
check the DHCP the result is:
Target Fqdn : lync.contoso.com
Target Uri : https://lync.contoso.com:443/CertProv/CertProvisioningService.svc
Result : Success
Latency : 00:00:09.0559615
Error Message :
Diagnosis :
When i run the follow command
PS C:\Users\Administrator> $cred = Get-Credential
cmdlet Get-Credential at command pipeline position 1
Supply values for the following parameters:
Credential
PS C:\Users\Administrator> Test-CsClientAuth -TargetFqdn lync.contoso.com -UserSipAddress "sip:[email protected]" -UserCredential $cred
I got this:
Target Fqdn : lync.contoso.com
Target Uri : https://lync.contoso.com:443/CertProv/CertProvisioningService.svc
Result : Success
Latency : 00:00:00.3431783
Error Message :
Diagnosis :
But.. when i use the same command but i remove the -targetFqdn for check the Dhcp i got this:
VERBOSE: Workflow Instance Id 'bca95636-af7b-4b0a-b43d-dba259294b2d', started.
VERBOSE: Command line executed is 'Test-CsClientAuth -UserSipAddress "sip:[email protected]" -UserCredential $cred
-Verbose'.
Target Fqdn :
Target Uri :
Result : Failure
Latency : 00:00:00
Error Message : 10060, A connection attempt failed because the connected party did not properly respond after a period
of time, or established connection failed because connected host has failed to respond 194.90.8.20:5061
Inner Exception:A connection attempt failed because the connected party did not properly respond after
a period of time, or established connection failed because connected host has failed to respond
194.90.8.20:5061
Diagnosis :
VERBOSE: Workflow 'Microsoft.Rtc.SyntheticTransactions.Workflows.STClientAuthWorkflow' started.
Workflow 'Microsoft.Rtc.SyntheticTransactions.Workflows.STClientAuthWorkflow' completed in '5.62E-05' seconds.
Target web service Url not provided. Will have to extract it from authentication challenge.
An exception 'Unable to establish a connection.' occurred during Workflow
Microsoft.Rtc.SyntheticTransactions.Workflows.STClientAuthWorkflow execution.
Exception Call Stack: at Microsoft.Rtc.Signaling.SipAsyncResult`1.ThrowIfFailed()
at Microsoft.Rtc.Signaling.Helper.EndAsyncOperation[T](Object owner, IAsyncResult result)
at Microsoft.Rtc.SyntheticTransactions.Activities.GetSTSUriActivity.InternalExecute(ActivityExecutionContext
executionContext)
at Microsoft.Rtc.SyntheticTransactions.Activities.SyntheticTransactionsActivity.Execute(ActivityExecutionContext
executionContext)
at System.Workflow.ComponentModel.ActivityExecutor`1.Execute(T activity, ActivityExecutionContext executionContext)
at System.Workflow.ComponentModel.ActivityExecutorOperation.Run(IWorkflowCoreRuntime workflowCoreRuntime)
at System.Workflow.Runtime.Scheduler.Run()
at System.Net.Sockets.Socket.EndConnect(IAsyncResult asyncResult)
at Microsoft.Rtc.Internal.Sip.TcpTransport.OnConnected(Object arg)
'GetSTSUri' activity started.
Starting STS Uri Discovery...
ERROR getting STS Uri.
'UnRegister' activity started.
'UnRegister' activity completed in '3.12E-05' seconds.
VERBOSE: Workflow Instance ID 'bca95636-af7b-4b0a-b43d-dba259294b2d' completed.
VERBOSE: Workflow run-time (sec): 126.0548512.
The Real Problem is that my Lync HP 4120 Phone can't make a sign in, not from USB cable loging, nor with PIN authentification
When I try to make a login with the USB cable, I set the user and password and the phone says "Connecting to Lync".. "Downloading a certificate" ... "Installing certificate"... "Downloading Certificate"...
"Installing Certificate".. forever
When I try to make a login with PIN Authentification, the phone first displays the following:
Account used is not authorized, Please Contact your support team and then shows this:
An Account matching this phone number cannot be found. Please contact your support team.
The Pin authentification is enable
In the Lync Server Enable Kerberos Authentification, Enable Integrated Windows Authentification and Enable Certificate Authentification are enable
This is the configuration from DHCP
Starting Discovery ...
Sending Packet (Size: 284, Network Adapter: xx.xx.xx.xx, Attempt Type: Broadcast only)
--Begin Packet--
DHCP: INFORM (xid=130EA7FA)
DHCP: Op Code (op) = 1
DHCP: Hardware Type (htype) = 6
DHCP: Hops (hops) = 0
DHCP: Transaction ID (xid) = 319727610
DHCP: Seconds (secs) = 0
DHCP: Flags (flags) = 0000
DHCP: Client IP Address (ciaddr) = Xx.xx.xx.xx
DHCP: Your IP Address (yiaddr) = 0.0.0.0
DHCP: Server IP Address (siaddr) = 0.0.0.0
DHCP: Relay IP Address (giaddr) = 0.0.0.0
DHCP: Client HW Address (chaddr) = FC15B4###--End Packet--
Received Packet
Sender:xx.xx.xx.xx:67, Size:363
--Begin Packet--
DHCP: ACK (xid=130EA7FA)
DHCP: Op Code (op) = 1
DHCP: Hardware Type (htype) = 6
DHCP: Hops (hops) = 0
DHCP: Transaction ID (xid) = 319727610
DHCP: Seconds (secs) = 0
DHCP: Flags (flags) = 0000
DHCP: Client IP Address (ciaddr) = xx.xx.xx.xx
DHCP: Your IP Address (yiaddr) = 0.0.0.0
DHCP: Server IP Address (siaddr) = 0.0.0.0
DHCP: Relay IP Address (giaddr) = 0.0.0.0
DHCP: Client HW Address (chaddr) = FC15B4100289
DHCP: Server Host Name (sname) =
DHCP: Boot File Name (file) =
DHCP: Magic Cookie = 99.130.83.99
DHCP: Option Field
DHCP: DHCP MESSAGE TYPE( 53) = (Length: 1) DHCP ACK
DHCP: Server Identifier( 54) = (Length: 4) XX.XX.XX.XX
DHCP: Client Identifier( 61) = (Length: 0) ()
DHCP: SIP Server( 120) = (Length: 17) enc:0 lync.contoso.com (00046C796E6306756E69736F6E026D7800)
DHCP: Host Name( 12) = (Length: 0)
DHCP: Vendor Identifier( 60) = (Length: 0)
DHCP: Param Req List( 55) = (Length: 0) 0 0
DHCP: Vendor Info( 43) = (Length: 86) MS-UC-Clienthttpslync.contoso.com443%/CertProv/CertProvisioningService.svcÜNAP (010C4D532D55432D436C69656E7402056874747073030E6C796E632E756E69736F6E2E6D78040334343305252F4365727450726F762F4365727450726F766973696F6E696E67536572766963652E737663DC034E4150)
DHCP: End of this option field
--End Packet--
Result: Success
DHCP Server : xx.xx.x.xx.
SIP Server FQDN : lync.contoso.com
Certificate Provisioning Service URL : https://lync.contoso.com:443/CertProv/CertProvisioningService.svc
thanks for all, hope somebody can help me with this problem.. i am going crazy...

Hi, i connected the Lync Phone to another switch and i update the firmware to the newest firmware and i got the same problem..
The lync phone download the certificate but cant install it and the still the same error with the SIP login
An Account matching this phone number cannot be found. Please contact your support team.

NullPointer when invoke InitialContext in standard application sip server

I am unable to do a new InitialContext() in an application that does not have any sipservlets , but is deployed on sip server engine tier
When an application is deployed in weblogic engine server (sip server 3.1 ) without a sip.xml as the application does
Not have any sip servlets , a new InitialContext() throws a NullpointerException . The stack trace is below .
This error is not encountered in weblogic 9.2 server (non sip server)
InitialContext ic = new InitialContext();
javax.naming.NamingException [Root exception is java.lang.NullPointerException: SIP Servlet context not found]
at javax.naming.spi.NamingManager.getURLObject(NamingManager.java:588)
at javax.naming.spi.NamingManager.getURLContext(NamingManager.java:533)
at javax.naming.InitialContext.getURLOrDefaultInitCtx(InitialContext.java:279)
at javax.naming.InitialContext.lookup(InitialContext.java:351)
at

Hi B.,
have you managed to find the solution for this problem? I'm facing same issue.
I also tried to publish the WS in standalone app (via javax.xml.ws.Endpoint.publish()) and it worked with no problem that way. However, when I moved this to servlet definition, it failed.
Thanks a lot.
Regards,
F.

AS5350 with SIP Server (Asterisk)

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