Call on hold (SIP Trunk)

[onlysim]

Hi Guys,

I am facing some issue on NetVanta 71000 IP-PBX , let me describer my setup and scenario below

The Adtran IP Phone 712 is connected with PBX 7100

The PBX is configured with a trunk account.

Scenarios Call on Hold.

When we put call on hold , the IP phone sends send a new invite of in dialog with SDP parameter sendonly to PBX , but PBX does not forward same SDP parameter to trunk.Instead PBX originate new invite (in dialog with sendrecv).

This new invite is strange to our Application and more like PBX doing RTP manipulation instead of originating SDP sendonly parameter.

Regards,

A Sidd

[Anonymous]

A Sidd,

Thank you for posting your question.  Can you upload your current configuration along with the output from a debug voice verbose and debug sip stack messages (both enabled at the same time) while you recreate the issue to our FTP server with the instructions below?

Open Internet Explorer web browser on their PC
Type the following URL:  ftp://ftp.adtran.com

Press the Alt key, click View, and then click Open FTP Site in Windows Explorer

Double-click the "Incoming" folder
Drag and drop files from PC into the Internet Explorer window

Reply to this post with the exact filenames used so we can retrieve the files

Thanks,

Matt

[jayh]

onlysim wrote:

Scenarios Call on Hold.

When we put call on hold , the IP phone sends send a new invite of in dialog with SDP parameter sendonly to PBX , but PBX does not forward same SDP parameter to trunk.Instead PBX originate new invite (in dialog with sendrecv).

This new invite is strange to our Application and more like PBX doing RTP manipulation instead of originating SDP sendonly parameter.

This sounds like normal behavior depending on how (and if) you are sourcing music-on-hold. 

The IP phone is sending an in-dialog reinvite telling the PBX to stop the outbound RTP stream from the phone and invite the MOH source to now send audio to the remote party.  It's sendonly because the audio coming from the remote party isn't going anywhere useful as far as the IP phone is concerned. 

If the external trunk is SIP, the PBX will then reinvite the remote party away from the phone and to the MOH source which may be internal to the PBX.  It's possible that the sendrecv is used to allow the remote party to escape from hold by connecting the receive stream to a DTMF receiver, or it may just be convenient to do so in all cases regardless if there is anything actually listening to the audio coming from the on-hold party. 

[onlysim]

Hi Jayh,

Thanks for details , it is possible I avoid this normal behavior.

In our setup we have IMS Application server and we want call hold and call transfer should be handled by that server.(PBX transparently forward SIP to SBC and ).

Regards,

Sim

[jayh · Accepted solution]

onlysim wrote:

In our setup we have IMS Application server and we want call hold and call transfer should be handled by that server.(PBX transparently forward SIP to SBC and ).

If you have another server handling call hold and transfer, it sounds like you don't need an IP PBX at all, as those are PBX functions.  Probably a SIP proxy between the phones and your IMS server is what you want to have your IP phones communicate with IMS. 

[onlysim]

Yes Agree you are right but is it possible to setup PBX to appear as a proxy .

Regards,

Sim

[Anonymous]

Sim,

There is not a way to configure a NetVanta 7100 as a sip proxy, as it is an IP PBX.  There is a way to convert it into a NetVanta 6355, which does support SIP Proxy.  The details on that process are covered in this post: .

Thanks,
Matt