SPA3102 - PSTN call not transfering to VoIP
When I call my PSTN line, after the PSTN answer delay, I just hear dialtone. I have enabled debug and I do see an invite being sent to my SIP trunk.
Here is what I get in the debug:
01-10-2013 09:40:21 Local2.Debug 166.166.0.44 CC:Clean Up
01-10-2013 09:39:48 Local2.Debug 166.166.0.44 AUD:Stop PSTN Tone
01-10-2013 09:39:48 Local2.Debug 166.166.0.44 AUD:Stop PSTN Tone
01-10-2013 09:39:38 Local2.Debug 166.166.0.44 AUD:Play PSTN Tone 8
01-10-2013 09:39:28 Local2.Debug 166.166.0.44 AUD:Play PSTN Tone 1
01-10-2013 09:39:28 Local2.Debug 166.166.0.44 FXO:Stop CNDD
01-10-2013 09:39:28 Local2.Debug 166.166.0.44 AUD:Stop PSTN Tone
01-10-2013 09:39:20 Local2.Debug 166.166.0.44 FXO:Start CNDD
Any idea on what I might be missing?
Thanks
Scott
It is normal to get a dialtone when the PSTN-to-VoIP gateway answers the phone unless you have a dial plan setup to automatically dial a number. A dial plan like (S0<:1234567890>) would set the interdigit timeout to 0 and dial 1234567890 on the configuration you have setup on the PSTN Line Tab.
Similar Messages
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SPA3102 can you process incoming PSTN calls without translation into VOIP?
On the SPA3102 is it possible to avoid translation of incoming calls into VOIP and out again. If so does anyone know the SPA3102 setting?
I realise that this would probably lose me quite a lot on incoming call processing options, but the echo on the PSTN line - revealed by the delay caused by the translation - is terrible, and I have tried most things to resolve it!
Many thanks in anticipation for any help you can give.
Mousetry setting PSTN-To-VoIP Gateway Enable parameter under the PSTN tab to NO -- this should disable the gateway functionality for PSTN to VoIP
btw: try playing with Network Jitter Level under the PSTN tab to resolve the the echo
| isolate! isolate! isolate! | -
I have inserted an SPA3102 between my regular telephone and the PSTN line, and find that the calls are being dropped with regular monotony (and this doesnt happen when the SPA3102 is not in the equation). The calls tend to be conference calls where I am listening rather than speaking. The Info page shows that the last PSTN disconnect reason to be 'PSTN activity timeout' and the PSTN activity timer is set to '30000 (ms)', although I am not aware of any breaks in the conversation when the line gets cut off - it can even cut off when I am speaking? The line simply goes dead and then I get the UK 'unobtainable' tone.
Can anyone provide advice on how to rectify this?UKfarmer wrote:The last couple of disconnects the PSTN disconnect reason is CPC signal, so the nature of the disconnect seems to be varying. Is it safe to switch off both CPC and long silence detection?You should read about CPC as to what it is. This web site has a summary: http://www.sandman.com/cpcbull.html Switching it off affects the ability of the SPA3102 to detect when a called party has hung up. There is also a setting for the minimum duration of the CPC.>how would I switch on syslogging to track in detail what is happening?You download a syslog program and run it on your pc, you put your pc's ip address on the SPA's system tab, you set the Debug Level to 3 on the SPA's System tab, and you set the Sip Debug Option to FULL on the Line and on the PSTN Line tab (on a SPA3102). You can download a syslog program on this web site: http://www.sipura.net/Documents/faq/Section_2.html#9
Edit: Looks like Linksys abandoned the sipura web site, anyway you can't download the program right now. There is a free Kiwi syslog program that works OK. Search for it.
I have also tried a VOIP call into the SPA and out over the PSTN line. In this configuration I can hear the remote party but they cannot hear me.There is quite a bit written about that common problem. It is a NAT Travsversal problem caused by your router. Search the forums for one way audio.
Message Edited by hw on 08-14-2008 12:26 PM -
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 -
NOKIA 5800XM - Calls not transferred to headset
Hi, I am Austin, I purchased a nokia 5800xm 2 months back and it worked flawlessly. I updated the firmware to 30v and it was even better, the headset that was bundled with the handset also worked fine, Butsince past week even though all the function keys on the remote and earphones work properly on the headset [radio/music player] when i get a call i cant hear it on the earphones but instead on the handset earpiece. I did a soft reset a few days back but after i installed a few apps the problem has resurfaced!!!
I have only installed opera 5 beta, myphone 2.2, pictureflow and a few simple themes nothing else...
Please help me!!! Im really confused if i should take it to the service centre or there is something i must enable???
ThanksHi
Easiest way to get a demo going is to do this:
1) Create a 'skill' in the RmCm subsystem menu in appadmin, call it Test
2) Create a Contact Service Queeu in the RmRm subsystem menu, and call is TestCSQ
3) On the CSQ properties, configure it for skills-based routing and add the 'Test' skill on the second page.
4) On your application, configure the 'icd.aef' script. This is a simple ACD type demo. Set the CSQ property of the application to "TestCSQ" (this must match your CSQ name identically)
5) Assign the Test skill to your agent in the RmCm/resources menu
At this point you should be able to dial your trigger and hear some prompts. If you log in as your agent, they shoudl received the call.
Note that to transfer the call to the agent, the CSS of the CTI Ports (i.e. the Call Control Group) must have the agent DN partition in it.
Also if you have only standard, you will need to create a Resource Group instead of a skill, and set that on the agent and the CSQ.
Regards
Aaron HarrisonPrincipal Engineer at Logicalis UK
Please rate helpful posts... -
Spa3102 would not forward a voip call to pstn line
Good morning.
I've done the implementation provided here http://community.linksys.com/t5/VoIP-Adapters/SPA-3102-and-softphone-to-
make-calls-via-pstn-line/td-p/326390 .
It is a way to use for outgoing calls a given pstn line from anywhere I have internet (voip to pstn).
The spa3102 is connected to a router (with an active DHCP server and ip 192.168.1.1) from where it takes the internal
ip (192.168.1.3).On the same network is also a computer , connected to the router ( with ip 192.168.1.2). The spa3102
is set to bridge mode and thus inactivates the function of the router (on SPA3102), and it functions as a simple
network device . I have done port forwarding (from the router) to 192.168.1.3 (SPA3102) for the port 5061 (PSTN
LINE) ( but for 5060 for the LINE 1 also). I want to make calls from a voip softphone (x-lite 4) to the SPA 3102 and
this to forward the voip calls to PSTN line to which it is connected. In x-lite the SPA3102 is set as a proxy so that
i can type the phone number I want to call without being followed by the SPA3102's ip each time ( eg on x-lite I
give call number 2101111111 instead of 2101111111 @ wanip: 5061 where wanip is the external ip of the router).
When x-lite is running on the computer that is on the same network with the SPA3102 everything works as expected. A
voip call is made from x-lite ( using as a proxy the wanip everytime, or even for test purposes the dyndns domain
that i set up for this reason), this call is passese PSTN line and the phone of the called party rings . At x-lite
COMES indication "call established ".
The problem occurs when I do the same procedure from x-lite installed on a computer belonging to another network (
e.g. in another building with its own internet connection , own router, own computer , etc. ) . Always using the
wanip the x-lite makes the voip call to the SPA3102, writes "call established" ( meaning it connected to SPA3102) but
never routed the call to the called party ( the SPA3102 did not forward voip calls it receives to the PSTN line ) .
Trying to find what 's wrong I've tried to disable all firewalls (soft and hard from all involved machines ) . The
behavior is the same either the computer that makes the successful calls is connected to the network directly to the
router or through the port "ethernet" on the SPA3102.
What is the difference in these two voip calls to the SPA3102 and the one " triggers " it to forward the call to
PSTN line and the other does not ?
Thanks now for any ideas you give .The audio sound problem is more than likely also associated with the overall addressing problem initially encountered. As you may know, using the sip protocol the sip signalling exchanges ip addresses to be used for both the sip signalling and the exchange of rtp sound packets. In addition there is an exchange of port numbers to be used for the exchange of rtp sound packets. The sound is exchanged by two separate streams of packets, one stream in each direction. The result is an ip address and port number for the rtp packets flowing from the SPA3102 to the softphone and a different ip address and port number for the rtp packets flowing from the softphone to the SPA3102.
In your previous posting you mentioned that you "set the minimum EXTernal rtp port at the sip tab". Changing the "EXT RTP Port Min:" is an unusual change to make and in my opinion would only be made in special circumstances. Actually, I ran some tests and I'm not sure exactly what that setting does. In my tests it didn't appear to affect the rtp port number used in a predictable manner.
The common changes to make for audio problems typically would be to setup a STUN server. A STUN server is an external server that echos back to the initial sender the external ip address and port number that the STUN server received with the message received by the server. This allows the sender (SPA3102 or softphone) to determine its external ip address and external port numbers for both the sip signalling and rtp packets.
A STUN server is commonly recommended to be setup with the following settings in the SPA3102:
PSTN Line Tab:
NAT Mapping Enable: Yes
Sip Tab:
Handle VIA received: yes
Handle VIA rport: yes
Insert VIA received: yes
Insert VIA rport: yes
Substitute VIA Addr: yes
Send Resp To Src Port: yes
STUN Enable: yes
STUN Server:
The following web page has a list of "Public STUN Servers"
http://www.voip-info.org/wiki/view/STUN
You are using CounterPath's XLite softphone. stun.counterpath.net is a STUN server on the list.
I see XLite also has a setting to use a STUN server on the "Topology" tab. -
SPA3102 pstn-voip dial tone without entering pin
Hello,
Once spa3102 is authenticated by pin, the adapter gives dial tone without entering pin (just the # sign after the beep) even when a different phone is used to call the spa3102 pstn number.
Isn't this a security compromise? Any one can access my voip account if the pin does not work.
Any help will be greatly appreciated.that's weird...if PIN Authentication is enabled, it prompts the caller to enter the PIN number followed by a # key
how long is your PIN? i have the same set-up in my house and authentication works fine and i tried pressing # after the beeps but it didn't give me a dial tone....
| isolate! isolate! isolate! | -
AutoAttendant not transferring calls in CUE 8.6 to CME 9.1
**Issue is calls are not transferring once extension or directory is pressed on the phone.
I upgraded our 2811 CME 7.1/CUE 7.0 router to 2911 15.2 CME 9.1/CUE8.6. It has CME and V/k9 license enabled. I uploaded the configuration used on the 2811 to the 2911. I added the ip addresses in the ip trusted list. Calls come in and out if I point the translation list to the phone and not to the auto attendant pilot number. Incase if it was toll fraud issue I disabled it but still a no go.
Everything worked fine on the 2811. I can't understand why it would not work on the 2911 15.2. Has anyone had any issues when upgrading from 12.4 to 15.x with auto attendant not transferring calls?
I ran a trace on the Cue when placed a call to extension 114:
20.10.10.1- CME interface address
20.10.10.5- CUE interface address
659 09/12 11:43:39.418 ACCN SIPS 0 Call.transferFailed(114, RESOURCE_NOT_ACKNOWLEDGING) SIPCallContact[id=33,type=Cisco SIP Call,implId=5738D3Dse-20-10-10-5# [email protected],active=true,state=CALL_ANSWERED,inbound=true,handled=false,locale=en_USOk I'm starting to think its a dtmf issue. I checked to see if I call from internal to the AA and dial an extension if it works but the same issue. I ran the debug voip ccapi inout and I see consume mask is not set. What would that indicate? Could that be the problem?
This is right when I dialed extension 114
2811-TEST#
001288: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001289: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
001290: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001291: Sep 16 14:59:22.870: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
001292: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001293: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
001294: Sep 16 14:59:23.194: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 1 to dstCallId 0x38
001295: Sep 16 14:59:23.198: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
001296: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_begin:
Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
2811-TEST#
001297: Sep 16 14:59:23.614: //55/D62BA3D180C8/CCAPI/cc_relay_digit_begin_for_3way_conference:
Check DTMF relay digit begin for 3way conf
001298: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_api_call_digit_end:
Consume mask is not set. Relaying Digit 4 to dstCallId 0x38
001299: Sep 16 14:59:23.618: //55/D62BA3D180C8/CCAPI/cc_relay_digit_end_for_3way_conference:
Check DTMF relay digit end for 3way conf
2811-TEST#
At this point theres silence on the phone I see this message:
2811-TEST#
001300: Sep 16 14:59:28.914: //55/D62BA3D180C8/CCAPI/ccGenerateToneInfo:
Stop Tone On Digit=FALSE, Tone=Null,
Tone Direction=Sum Network, Params=0x0, Call Id=55
001301: Sep 16 14:59:28.918: //56/D62BA3D180C8/CCAPI/cc_api_call_feature:
Feature Type=50, Interface=0x49FC2B80, Call Id=56
2811-TEST#
2811-TEST#
At this point I get the message "the phone number you are trying to reach" then the call disconnects
2811-TEST#
001242: Sep 16 14:48:57.719: %VOICE_IEC-3-GW: SIP: Internal Error (ACK wait timeout): IEC=1.1.129.7.67.0 on callID 52 GUID=5D21AC143CE711E480BCEEC6B624839
Also I checked show voice iec description:
2811-TEST#show voice iec description 1.1.129.7.67.0
IEC Version: 1
Entity: 1 (Gateway)
Category: 129 (Call setup timeout)
Subsystem: 7 (SIP)
Error: 67 (ACK wait timeout)
Diagnostic Code: 0
2811-TEST# -
External PSTN calls are not making it to CUE voicemail?
External PSTN calls are not being forwarded to my Unity Express Voicemail. Internall calling and forwarding to Voicemail is working like a charm. However, calls from the outside just busyout after so many rings and never make it to voicemail.
Can some one help point me in the right direction. I think it might be as simple as dial-peer problem.
Thanks in advance,
CurtI'm assuming this is CME-integrated. Basically, if you have a PRI connection, the easiest thing to do is to do a 'debug isdn q931' and check out the called number. Or if you have an FXO connection to the PSTN, then you've got some kind of 'connection plar' under the voice port to give you the called number. Basically the called number must match a destination pattern under a voip dial-peer pointing to CUE (you can use 'debug ccsip message' and look for an INVITE message to see where it's going to). Next, it has to match a trigger ('show ccn trigger' in the CUE CLI) to envoke a particular application, such as voicemail. If you have trouble, what we'll need to see is the exact called number (either via the isdn q931 debug or connection plar configuration statement) and the configuration (show run) from the CME and the CUE.
Lastly, take a look at the following notice:
http://www.cisco.com/en/US/partner/products/sw/voicesw/ps5520/products_field_notice09186a008023cfe2.shtml
which might apply to your situation. -
PSTN call transferring between sites
We are planning on having 2 sites
Site A: Main Lync server with mediation/frontend roles. Edge server. SIP trunk
Site B: SBA with PRI
Sites are to be connected with a VPN
We will have users at Site A that will take a call on the PSTN and need to be able to transfer that call to users at Site B. Site B has no receptionist, their receptionist is at Site A.
My initial plan was to have every users line uri set up with the phone number and extension, no DIDs. Site A could be tel:+19997654321;ext=1000 and Site B could be tel:+19991234567;ext=2000.
My question is what is the call path for transferring when a call comes in on Site A PSTN -> Site A User -> Site B User?
Will this call flow through the VPN to Site B from Site A and be using up one of the SIP trunks for the duration?
What happens if I make users at Site B have DIDs instead of extensions for their line uri? Could the call be transferred to the PRI and not have to flow through the VPN?Hi kuptzlocher,
Have a look at the following picture.
Based on my understanding, if the number was already in a globally unique format (E.164), then Reverse number lookup will query it :
Is this number assigned to anything in Lync? If it is, then Lync will be able to translate the number to a SIP URI and route the call directly to inbound routing based on said URI where the process will then end.(In other words, it
will through the VPN)
Best regards,
Eric -
Call on PRI Line are not transfered to IP-AA on CCM
Greetings,
Having an issue with H.323 Voice gateway. Incoming calls are not transferred to Auto-Attendant. This happens randomly but I have noticed that the frequency is higher when there are no incoming call no PRI lines.
(Check the attachments for debugs)
The calls seems to be answered by gateway but ISDN-Connect message is never displayed. I tried to call from my IP Phone to but no ISDN-Connect message for this call, however, the call is landed on gateway. Running CCM 4.2(3), IPCC Express 4.0(4),IOS version 12.4(8) Advance IP Services, tried IOS 12.4(5a) Advance Enterprise Services as well but in vain.
Another questions is that: we can run both MGCP and H.323 on voice gateway but how will it register with CallManager. I guess we can use some loopback technique for this purpose. Need suggestions/comments!!!
Please find the attachments of Q921 and Q931 debugs. Took debugs for 2-scenarios
1- Called from and outside number to PRI line
2- Called from IP-phone to PRI line
Regards,
Syed Khalid AliYou can refer this guide for more information on configuring loopback for PRI interface:
http://www.cisco.com/en/US/docs/ios/12_2/dial/configuration/guide/dafchant.html#wp1002838 -
Hi there,
I have several SPA3102s in my enviroment that I am using to connect legacy PSTN lines to the VoIP network. Calls come in on the PSTN lines and are routed to our Asterisk server by the gateways. The Asterisk server connects the calls to the correct IP phone extensions. When calls come in this way (or go out using the same line) the inside party hears an echo of themselves. During an individual call, quality fluctuates but the echo is present during every call. Calls between IP extensions have no echo. Calls connected to the PSTN network through a different gateway from the same IP extension have no echo.
So far I have upgraded all gateways to the current firmware and followed the troubleshooting steps outlined here; http://www.cisco.com/en/US/products/ps10024/products_qanda_item09186a0080a359cc.shtml
with no luck. Turning off the built in echo reducing features makes the calls consistent, with bad echo constant throughout all PSTN calls but none of the other steps reduce the echo as much as the built in features. I have searched the forums and not found anyone else have this problem with just the PSTN port. Does anyone have any ideas?Hey guys, have you tried the following to reduce echo on the SPA3102.
Q. How can I reduce the PSTN echo on SPA3102?
A. Experiencing echoes in the PSTN line is a common problem. This is because the SPA3102 passes calls from the PSTN to LINE1 by converting it to VoIP internally then converts it back to analog. This process does not produce any echo, however, it can add about 30ms of latency to the call which later produces the echo.
Reducing the Echo on the PSTN Line
Make sure you are running the latest firmware. Everything should be set to factory defaults, or at least undo all the previous tweaking.
Disable all the echo cancellation functions of your SPA3102. These settings can be found on line 1 and PSTN line tabs of your SPA3102. Echo Canc Enable = No
Echo Canc Adaptive Enable = No
Echo Supp Enable = No
Remove devices connected to your phone line except the SPA3000. This includes all the extension cables and splitters. These can cause impedance problems which lead to echoes.
Set the FXO port Impedance on the PSTN tab to 220+820||120nF, and set FXS Port Impedance to 220+820||115nF as a starting point.
Look for Network Jitter Level on the PSTN Line tab and set it to low. Then, look for Jitter Buffer Adjustment and set it to disable. This reduces the delay across your SPA3000. Note: If you are using a poor quality VoIP service, go to the Line 1 tab and look for Network Jitter Level. Set it to low and set the Jitter Buffer Adjustment from up to down. However, if you are using a poor quality PSTN, set the Network Jitter Level to medium.
Go to the PSTN Line under Audio Configuration. Look for Preferred Codec and set it to your preferred settings, then lock it in by setting the Use Pref Codec Only to yes. Adjust these settings if you are accessing your PSTN line via VoIP from a remote network. Then, go to Line 1 and set Preferred Codec with the same settings you set with PSTN Line. Under Preferred Codec Only set it to no. These settings reduce your latency and can make the echo less obvious or easier to catch with the echo canceller.
Power cycle the SPA3000 by powering down the device. This sometimes fixes the problem especially after changing the physical phone wiring.
Make some test calls and observe if you can hear an echo. If yes, the problem might be that you are sending too much power down the line and it gets reflected back somewhere as an echo. Even if you have good wiring but you are too close to the mouthpiece, you will still hear an echo. To resolve this, you need to increase the level of Gain by going to PSTN and look for SPA To PSTN Gain, then slowly adjust the level until you can clearly hear the person on the other line. Note: If you enable Echo Supp Enable, you will negate these parameters. The echo suppression is just an automatic gain control. It is recommended to keep it disabled.
Make a test call to someone with a phone that works via the SPA's PSTN line, or call in to the PSTN line. If the remote party is hearing an echo, your phone might be loud and is experiencing feedback in the microphone. Lower the PSTN to SPA Gain until you are comfortable hearing the person on the other line. If the remote user can still hear an echo, try using a different phone plugged into the SPA. If this solves the problem, your phone might not be working properly, or there is an impedance mismatch between your phone and the SPA. Try changing the FXS Port Impedance to 600 on the Regional tab and change the FXO port impedance to 600 or global. If this does not help, change it back. The impedance will only affect what the remote party hears and will not help remove the echo you are hearing.
After lowering the echo to a tolerable level, go back to the PSTN tab and enable Echo Can Enable by selecting Yes, OK. Check if the echo has improved. If the echo is tolerable at this level, leave the adaptive echo canceller off. You should have the echo level down to a level that can be filtered by the echo canceller. If you are using a sip device to talk through your PSTN line, you should probably do all the echo cancellation at that device and leave it switched off in the SPA. -
I am using spa3102 which have Line port and PSTN port
I have no problem with line.... But I configured the PSTN port with 8 accounts and I enabled
The PSTN Caller Auth Method: PIN under PSTN-To-VoIP Gateway Setup and I configured
8 pin for each account, Normally when I call PSTN and I hear the beep tone I enter My Pin followed by # sign and it work fine...But if I called again and I put only # sign without any pin
Spa3102 open the last pin number was making a call
So maybe there is an option I must do or it is some bugs with the firmware
Note that my spa3102 have the latest upgrade
Regards
Wael Saad
00965-9618253It's not broken. The problem is was with the cable used to connect it to the UK British Telecom wall socket. I did not use the grey cable that came with it, but used a UK phone cord I had lying around - bad move.
When talking to Linuxemporium, from where I bought the SPA3102, I was made aware that the cable might the problem. So I went over the road to a small phone shop to get a US-UK adapter (the cable packed with the SPA3102 is still a US cable), which didn't solve it. I got another adapter from a well known UK electronics retailer, and that one solved the problem. The SPA3102 is now happily working as my PSTN gateway.
I hope this can help others having similar problems. -
Hi,
I'm running a new SRP547W running the latest firmware.
Firmware Version:
1.2.4 (003) Jan 11 2012
I have 3 phones connected to the unit. Ports 1 & 3 are used for outbound calls using VOIP and are registered to the VOIP provider.
The third phone is connected to phone port 4 and has NO VOIP settings and is not registered.
When an incoming PSTN call is received the phone on port 4 rings likes it should. Phones on ports 1 & 3 don't but that's due to an acknowledged bug.
When I answer the PSTN call by picking up the phone on port 4 the call is not answered. All I get is silence and the person calling me continues to get the phone ringing signal until they hang up.
Any ideas why I cannot answer the call? And what I can do to fix the problem?
Thanks
GavinI know its bad form to answer your own questions but I think I've solved it.
I reset the voice settings on the unit back to factory defaults. Put my voip settings back in, changed region to Australia, updated dialplan and now, when I get an incoming call on my PSTN it rings *ALL* the phones (even ones that are registered for voip) and I can answer the call. So I must have mucked up a setting somewhere while I was playing.
If all else fails do a factory defaults and try again :-)
g -
SPA3102 - PSTN Distinctive Ring Detection
Hi,
I have SPA3102 connected to Asterisk. I have a single PSTN line with two numbers, each with a distinctive ring pattern. Is SPA3102 capable to detect the ring pattern of an inbound PSTN call and provide this info to the SIP gateway (for example, in the Alert-Info SIP header).
Thanks!This may not be possible as the this device I think cannot recognize the Ring pattern being to the FXO port for the two numbers.
Some features are disabled or does not work when connected to the SPA3102 compared to when connected to a regular telephone.
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