3CX SIP client with IP office registered but no audio

[mayubi]

I am using 3CX SIP client , got them registered with IP office and can make a call to a landline, Mobile phone, another 3CX SIP client, Nokia SIP client...
The phone rings, but once I answer the call, I dont hear any voice, and this happens irrespective of me calling a landline/fixed, mobile phone 3CX SIP client, Nokia SIP client.

Please suggest what am I missing.

Thank you,
Ayubi

 

[tlpeter]

Routing is your problem.

BAZINGA!

I'm not insane, my mother had me tested!

 

[mayubi]

Thank you Peter.

I am bit new to IP office, so I should be checking the routing on the netgear router or the LAN-1 settings on the IP office.

I am using only LAN-1 of the IP office and mentioned that IP in my 3CX SIP client.

Regards,
Ayubi

 

[mayubi]

I just installed X-lite SIP client( Extn 209) on the same PC where I had 3CX SIP client (Extn 210).

I can make calls between these extension, while doing so noticed that conenction type is shown as VCM and the Codec as G.711 A for one and for the other client it shows G.711 Mu, can that be a problem..

Regards,
Ayubi

 

[tlpeter]

That can be the problem.
You should use the same codec for all endpoints.
Codec mismatch can give speech problems too.


BAZINGA!

I'm not insane, my mother had me tested!

 

[mayubi]

Tried again ,and this time the connection type is : Direct Media and the codec is : G.711 Mu, still no audio, checking the routing.

However, the client on the local network as well have the same problem.

My customer says that when this Demo was given, the vendor got a card which he plugged into the IP office 500 , which we dont have currently. So am I missing some hardware required to get this worked.

Regards,
Ayubi

 

[tlpeter]

Then do you have a VCM card in the system?


BAZINGA!

I'm not insane, my mother had me tested!

 

[mayubi]

I dont have physical access to this IP office , however when I checked the System config/IP office Manager, under the control units I see device number 3 as : VCM32/ATM4
Attached the screen shot of the VCM config.
The Ip address of this card is all zeros.

 

[tlpeter]

Then you have it.
Then it is routing.
Add an iproute

0.0.0.0
0.0.0.0
gateway ip adress.
Lan1 or Lan2

BAZINGA!

I'm not insane, my mother had me tested!

 

[mayubi]

Did that already, no luck..

 

[hairlessupportmonkey]

so i suspect your issue is the RTP is being blocked somewhere.

what does a monitor trace show?

ACSS - SME
General Geek

 

[mayubi]

After the session/call gets established I see these messages :


59831176mS RES: Mon 13/6/2011 14:47:49 FreeMem=74386700(1) CMMsg=5 (6) Buff=200 968 1000 7454 5 Links=3707
59831176mS RES2: IP 500 5.0(8) Tasks=31 RTEngine=0 CMRTEngine=0 Timer=43 Poll=0 Ready=0 CMReady=0 CMQueue=0 VPNNQueue=0 Monitor=1
59836176mS RES: Mon 13/6/2011 14:47:54 FreeMem=74395876(1) CMMsg=5 (6) Buff=200 968 1000 7454 5 Links=3709
59836176mS RES2: IP 500 5.0(8) Tasks=31 RTEngine=0 CMRTEngine=0 Timer=42 Poll=0 Ready=0 CMReady=0 CMQueue=0 VPNNQueue=0 Monitor=1
59837471mS SIP Rx: UDP 192.168.0.14:49664 -> 192.168.0.133:5060

And on the System Monitor , under Status Tab , when I check RTP sessions, it shows no data .

 

[mayubi]

Also this :

61215307mS PRN: Monitor Status IP 500 5.0(8)
61215307mS PRN: LAW=A PRI=0, BRI=0, ALOG=4, ADSL=0 VCOMP=4, MDM=0, WAN=0, MODU=0 LANM=0 CkSRC=0 VMAIL=0(VER=0 TYP=0) CALLS=1(TOT=109)
61223847mS SIP Rx: UDP 192.168.0.14:49664 -> 192.168.0.133:5060

 

[hairlessupportmonkey]

I reckon your issue might lie in the alog lines

when you make a call, in SSA do you see a vcomp channel used? btw, you only have 4 voice compression channels licensed. also do you have a 3rd party endpoint licence?

ACSS - SME
General Geek

 

[mayubi]

Yes, one voice compression channel is used when I make a call .
If we get this working, planning to increase the channels.
We have the 3rd party endpoint license .

 

[mayubi]

Was observing the monitor for some time after the call established and then put one of the SIP client on Hold and saw these lines in RED
192.168.0.14 : SIP client in the private LAN
192.168.0.133 : DSL Router IP
78.100.190.131 : Nokia SIP client on Internet.

===========================

89682485mS H323Evt: AudioEndPoint::Tick not present 30000 rx_ok 0 tx_ok 1 send_only 0 last_rx 0 last_tx 89682469 last_icmp 0 start_icmp 0 start_operational 89382485
89694380mS SIP Rx: UDP 192.168.0.14:54362 -> 192.168.0.133:5060
89724404mS SIP Rx: UDP 192.168.0.14:54362 -> 192.168.0.133:5060
89742485mS H323Evt: AudioEndPoint::Tick not present 30000 rx_ok 0 tx_ok 1 send_only 0 last_rx 0 last_tx 89742471 last_icmp 0 start_icmp 0 start_operational 89382485
89754437mS SIP Rx: UDP 192.168.0.14:54362 -> 192.168.0.133:5060
89760952mS SIP Rx: UDP 78.100.190.131:51435 -> 192.168.0.133:5060
======================================================

Does this message suggest anything about why I am not able hear after call get established.

89742485mS H323Evt: AudioEndPoint::Tick not present 30000 rx_ok 0 tx_ok 1 send_only 0 last_rx 0 last_tx 89742471 last_icmp 0

==============================================================

 

[mayubi]

Got the SIP client to SIP client calls and Audio working both when behind a private LAN and over internet.

However, when calling an Analog phone ( internal or external), using the SIP client ( connected over 3G or private LAN) can only ring, but when answered no audio.

Is there additional configuration required to get audio between SIP phones and Analog phones.

Please suggest.

Thank you,
Ayubi