line is dead after transferring from Digital Receptionist

[keith.bucknall]

Dear All,

I have noticed any calls from our Digital Receptionist, which is played when a caller dials our SIP number, i.e. press 1 for sales or 2 for accounts - the call is then transferred to our Cisco 7940 phones and both caller and reciever can not hear anything.

Can someone help ASAP as this is causing us problems

I have 2 network cards in this server and would like 3CX to use on a perffered one - how do i do this

A sample of the log can be sent if needed

thanks

keith

 

[gbarnes]

No audio after transfer from DR...

Have you set firewall not to block your RTP ports?

RTP ports set the same, eg 80xx or 90xx etc.

Extensions/Options/Bind to media server?

Are the extensions registering correctly?

If not then a look at your logs?

Second question - No I dont think this is possible at present. It would be nice to seperate network cards. You can set the APACHE options to listen on a seperate address though.

Maybee in a later verssion.

 

[keith.bucknall]

Hi there,

What are the RTP ports, i have the 5060 and a few other SIP and UDP ports open the firewall. But these point to the specific network card - hence why i would like to specifiy it.

The line and ext register sucessfully.
What do you mean by - Extensions/Options/Bind to media server?

Thanks Keith

 

[keith.bucknall]

can someone please help still,

as there is no way to specify 1 network card i have now added the following rules to my firewall to allow the ports to my 3CX server on both network cards:

UDP - 9000 - 9004 / 7000 - 7500 / 5060 / 5004 / 8000 - 8012



This problem still happens

Keith

 

[gbarnes]

Logs

It looks like the RTP ports, eg the UDP ports used to transfer audio, are they set the same?, I use 8000 to 8050 here and opened the firewall for these ports. Maybee they are the 90xx ones on your system.

If you set logghing to verbose and post it Ill take a look at what is happening. (Ie logging/verbose) then restart.

Do your extensions work together, ie one can call the other with audio?

Ill await the logs, im sure someone will sport whats going on.

Graeme.

 

[keith.bucknall]

Graeme

Thanks for the quick reply and to be honest i have not changed the RTP ports these ports were always open on the firewall since i had A@H and Trixbox.

For reference the 2 NIC's are 10.99.99.21 (this is what i want 3CX to use) and 10.99.99.16 - this is what it is using.

10.99.99.23 is one of the test Cisco 7940's

3CX used to work fine without any issues since the last 2 upgrades the problems have appeared, my log is:

21:03:22.061 ServRegs::checkExpiration [CM113000] Registration for sip:[email protected] has expired
20:56:19.595 MediaServerReporting::RTPReceiver [MS105000] Call(2) Ext.301: No RTP packets were received on 00000004@:remoteAddr=10.99.99.23:28766,extAddr=0.0.0.0:0,localAddr=10.99.99.16:7006
20:56:19.564 StratLink:nHangUp [CM104001] Call(2): Ext.301 hung up call; cause: BYE; from IP:10.99.99.23
20:56:06.618 MediaServerReporting::SetRemoteParty [MS010001] Call(2) Ext.301: Failed to update remote party attributes. No codecs intersection found.
20:56:06.056 CallLegImpl:nConnected [CM103001] Call(2): Created audio channel for Ext.301 28766) with third party (10.99.99.16:7006)
20:56:06.024 StratInOut:nConnected [CM104005] Call(2): Setup completed for call from Ln:10000@Sipgate to Ext.301
20:56:03.213 CallConf:nProvisional [CM103003] Call(2): Ext.301 is ringing
20:55:57.107 MediaServerReporting:TMFhandler [MS211000] Call(2) Ln:10000@Sipgate: DTMF (RTP) from 217.10.68.72:52008 arrived. in-band DTMF tone detection is turned off.
20:55:53.328 CallLegImpl:nConnected [CM103001] Call(2): Created audio channel for Ln:10000@Sipgate (217.10.68.72:52008) with Media Server (10.99.99.16:7004)
20:55:53.250 CallConf:nIncoming [CM103002] Call(2): Incoming call from 07801397739 (Ln:10000@Sipgate) to sip:[email protected]
20:54:49.458 ServRegs:nAdd [CM113002] Registered: Ext.301
20:49:01.152 MediaServerReporting::RTPReceiver [MS105000] Call(1) Ext.301: No RTP packets were received on 00000002@:remoteAddr=10.99.99.23:28762,extAddr=0.0.0.0:0,localAddr=10.99.99.16:7002
20:49:01.090 StratLink:nHangUp [CM104001] Call(1): Ext.301 hung up call; cause: BYE; from IP:10.99.99.23
20:48:47.697 MediaServerReporting::SetRemoteParty [MS010001] Call(1) Ext.301: Failed to update remote party attributes. No codecs intersection found.
20:48:47.228 CallLegImpl:nConnected [CM103001] Call(1): Created audio channel for Ext.301 28762) with third party (10.99.99.16:7002)
20:48:47.197 StratInOut:nConnected [CM104005] Call(1): Setup completed for call from Ln:10000@Sipgate to Ext.301
20:48:41.165 CallConf:nProvisional [CM103003] Call(1): Ext.301 is ringing
20:48:35.539 MediaServerReporting:TMFhandler [MS211000] Call(1) Ln:10000@Sipgate: DTMF (RTP) from 217.10.68.75:54974 arrived. in-band DTMF tone detection is turned off.
20:48:30.866 CallLegImpl:nConnected [CM103001] Call(1): Created audio channel for Ln:10000@Sipgate (217.10.68.75:54974) with Media Server (10.99.99.16:7000)
20:48:29.960 CallConf:nIncoming [CM103002] Call(1): Incoming call from 07801 (Ln:10000@Sipgate) to sipsipgate.co.uk
20:48:11.192 ListenConnect [CM114000] SL: connected promo-vm3:0/PHPExtension_0 at [promo-vm3:0]/PHPExtension_0
20:35:49.625 ClientRegs:nSuccess [CM113005] Registration of sip:[email protected] is successful
20:35:49.203 ExtLine::Register [CM110004] Send registration for "08450045941"<sipsipgate.co.uk>
20:35:49.203 ExtLine::Register [CM110001] Use External IP for device line registration DN='10000' device='Sipgate'
20:35:47.953 StunClient:rocess [CM115002] STUN resolved external IP=:5060 by server 80.239.235.209
20:35:47.906 ListenConnect [CM114000] SL: connected promo-vm3:0/VoiceBoxManagerService at [promo-vm3:0]/VoiceBoxManagerService
20:35:47.906 StunClient:rocess [CM115001] Send initial STUN request to 80.239.235.209
20:35:47.796 CallMgr::Stack::thread [CM106001] ** Enter Stack Loop **
20:35:47.578 IVRConnected [CM111000] IVR Server is connected
20:35:47.578 ListenConnect [CM114000] SL: connected promo-vm3:5483/IVRServer at [promo-vm3:5483]/IVRServer
20:35:47.437 CallMgr:umThread::thread [CM100004] ** Enter DUM Thread **
20:35:46.906 DBA [CM109000] ** Database connection Ok **
20:35:45.953 CallMgr::Stack::Initialize [CM106000] ** Adding transports **
20:35:45.921 CallMgr::Initialize [CM100003] ** Initializing SIP stack **
20:35:45.843 CallMgr::Initialize [CM100002] Default Local IP address: 10.99.99.16:5060
20:35:45.765 MediaServerConnected [CM112000] Media Server is connected
20:35:45.765 ListenConnect [CM114000] SL: connected promo-vm3:0/MediaServer at [promo-vm3:0]/MediaServer
20:35:44.843 CallMgr::Initialize [CM100001] Version: 3.1.2434.0
20:35:44.843 CallMgr::Initialize [CM100000] Start 3CX PhoneSystem Call Manager
20:35:44.828 LoadLicenceInfo [CM100008] Licence loading error

 

[keith.bucknall]

tried making another call and:

18:56:55.613 MediaServerReporting::RTPReceiver [MS105000] Call(1) Ext.301: No RTP packets were received on 00000002@:remoteAddr=10.99.99.23:25694,extAddr=0.0.0.0:0,localAddr=10.99.99.16:7002
18:56:55.566 StratLink:nHangUp [CM104001] Call(1): Ln:10000@Sipgate hung up call; cause: BYE; from IP:217.10.79.23
18:56:42.471 MediaServerReporting::SetRemoteParty [MS010001] Call(1) Ext.301: Failed to update remote party attributes. No codecs intersection found.
18:56:41.926 CallLegImpl:nConnected [CM103001] Call(1): Created audio channel for Ext.301 25694) with third party (10.99.99.16:7002)
18:56:41.911 StratInOut:nConnected [CM104005] Call(1): Setup completed for call from Ln:10000@Sipgate to Ext.301
18:56:33.347 CallConf:nProvisional [CM103003] Call(1): Ext.301 is ringing
18:56:27.632 MediaServerReporting:TMFhandler [MS211000] Call(1) Ln:10000@Sipgate: DTMF (RTP) from 217.10.68.75:56838 arrived. in-band DTMF tone detection is turned off.
18:56:18.539 CallLegImpl:nConnected [CM103001] Call(1): Created audio channel for Ln:10000@Sipgate (217.10.68.75:56838) with Media Server (10.99.99.16:7000)
18:56:17.199 CallConf:nIncoming [CM103002] Call(1): Incoming call from (Ln:10000@Sipgate) to sip:

 

[keith.bucknall]

looking on sipgates web site they say the following - will this be anything to do with it:

News - 24/07
Modification: Adjustment DTMF standard at August 1st
sipgate changes the Dual Tone Multiple Frequency (DTMF) to the RfC2833 standard and so guarantees an overall hardware support. A change of the hardware configuration is normally not necessary.

Advice: If Dial-In and Call-Throug connections are actuated or an interactive voice response (IVR) is to be used, the settings have to be renewed. Therefor sipgate offers detailed instructions, which will be released at August 1st, 2007

 

[keith.bucknall]

can anyone still help and diagnose the logs please...

 

[Mirzab]

This is just off the top:

18:56:42.471 MediaServerReporting::SetRemoteParty [MS010001] Call(1) Ext.301: Failed to update remote party attributes. No codecs intersection found.

Are codecs configured properly? Sipgate seems to prefer G729 though it supports

sipgate uses and supports the following codecs:

# G.729

# G.711

# iLBC

# GSM

# G726

so there are common codecs with 3CX. Is it possible to control codecs used on the Sipgate side of configuration?

As far as RTP ports go I am learning as I go myself:

Are there specific ports assigned to RTP?
No, as explained in the section Port Assignment of the RTP profile:

As specified in the RTP protocol definition, RTP data is to be carried on an even UDP port number and the corresponding RTCP packets are to be carried on the next higher (odd) port number.

Applications operating under this profile may use any such UDP port pair. For example, the port pair may be allocated randomly by a session management program. A single fixed port number pair cannot be required because multiple applications using this profile are likely to run on the same host, and there are some operating systems that do not allow multiple processes to use the same UDP port with different multicast addresses.

So its a UDP port range an the end of all the reading.

I also remember reading in the Forum elsewhere Henk / ITFarmer commenting that he prefers G711 since the high compression codecs seem to disable DTMF in-band detection (if memory serves me right)

I am "sticking my nose in" solely because it seems that qualified members are a bit busy right now, and I readily admit that my knowledge is poor in this area

 

[keith.bucknall]

thank you very much for the reply on this, the codec's i have are just the default that 3CX uses which are:

G711 ULAW and ALAW

Shoould i change these?

Keith

 

[Mirzab]

I don't believe you can, unless you purchase one like G729 and I am unsure then of the process to integrate with 3CX (if even possible). For myself I selected G711a as the first or preferred codec for all software and devices (3CX server, PAP2s, SPA3102 and Eyebeam softphones) but did not check "use only" leaving the ability to use another if needed. If your loss of audio is due to a codec issue, ensuring such a common setup rule exists across your full installation may help. My inferiority complex :lol: requires me to say that this is just a Layman's opinion and I might be very wrong.

 

[SY]

can anyone still help and diagnose the logs please...

There is one problem, it seems logs are modified... At least there is some information which is absent. (otherwise, there is problem with broken configuration)

In your case, only full (and complete) verbose log for this call will help to fix your configuration (or PBX ) .

You may sent it to me, directly.

Regards,

 

[keith.bucknall]

Stephan

Please advise what you need and where to send it?

 

[SY]

Stephan

Please advise what you need and where to send it?

1. set verbose level of log. (if it already verbose, then switch if to medium press OK, then switch it back to verbose and press OK)
2. restart PBX services (PhoneSystem and MediaServer)
3. perform a problematic scenario
4. open support page and make support package
5. send package to me.

Regards

 

[keith.bucknall]

thanks i will do this - what is your email address?

 

[SY]

thanks i will do this - what is your email address?

check private