SIP to SIP Transfer - does it work for you? 1

[DavidCT]

I am having a problem with SIP to SIP external transfer. I have just one way audio. The person being transfered is fine, the person receiving the transfer hears nothing.

This only happens when both calls are using SIP, using the Transfer button, as well as both assisted and unassisted transfer from VMPro. DTMF breakout from voicemail for each user's mailbox also has the same problem.

System:

IPO 406v2 running admin 4.0.14 and VMpro 4.0.27.
5 SIP Licenses, 4 VCM Resources

 

[Bas1234]

I use SIP trunks one a IP500 V4.1(9) external tranfers works fine. My codec settings are set to G711 and also for the IPPhones one way audio is normaly a codec problem.
What settings did u use for the sip line?
And how is your ARS programmed?

Greetzzz....Bas

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It works! Now if only I could remember what I did...
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[DavidCT]

One way audio is happening on both Digital and IP Phones.

Codec is G729 per the SIP provider. Regular calls work fine (incoming and outgoing). Just transfering is a problem.

 

[Bas1234]

You could try to set the "Binding Refresh Time" on 600msec.
System > Lan Tab > Network Topology.

And make sure you've marked "RE-INVITE Supported"
Line > Sip line > Sip line tab.


Greetzz....Bas

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[DavidCT]

bas1234,

No luck...

My ITSP does not support Re-INVITE, but I enabled it anyway, along with the binding refresh at 600. Still only one way audio on transfers.

 

[tlpeter]

Maybe a part of your problem ?

CQ39141

Out of band DTMF fails on SIP trunks after upgrading to 4.0.7

4.0.7

MT_RELEASE_1Q08_4.1


ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[IPOfficeIreland]

hi DavidCT

I had the same issue but resolved it by doing the following:

1st check that when a call is transferred, if you press hold or park, then pick up the call again, can you now get full audio?

If your provider does not support Reinvites, turn it off. My provider does but this caused a problem when transferring from vmpro.

Switch off Direct Media Path also, this is where i think your problem will be. Switch it off at SIP trunk level and at IP phone extension level.


You will also need to match the codec your provider uses for proper use of VCM channels, if they use G729, then set G729 for all ip devices.

Finally, can you go to Monitor, System and enable Development tracing. You can now click on tab for Status, select Voice Compression (T1) and watch how the VCM resources are being used. A SIP call to an IP device should use a vcm resource correctly for a second then free the channel. I found in some cases 1 call held 2 vcm resoures, a transfer a 3rd, so if you had 4 and 5 needed, 1 party would get no voice.

Try the steps and let me know.

Cheers

 

[DavidCT]

Hi All,

It looks like I found the fix with bas1234's help... I set the binding refresh down to 30 (600 did not work).

I can transfer calls from VMPro, DTMF breakout works from a user's mailbox to a transfer, and transfering from a physical phone works too.

I left Re-INVITE off since my ITSP said they do not support it.

Does anyone know what a "normal" binding refresh time is? Should I leave mine at 30? I am thinking it is specific to my firewall and not my ITSP.

My only other question is now that transfer works, can I send the original calling party's CID instead of my office number? My ITSP does allow me to send whatever caller ID I want.

 

[Bas1234]

OK, with our provider a had to fill in 600msec that was on an Asterisk.

To sent the number CID goto; line > Sip line > Sip Uri tab.
Select the main number > edit > and change the "display name" from "Use Authentication Name" or "Use User Data" to the number you like. When you fill in Anonymous then there will be no number sent.

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[DavidCT]

bas1234,

That's how we have our sip lines configured now, so it always sends out OUR number on calls.

I want to send out the original caller's number on transfered calls, not our number. I thought I could use $CLI in the transfer field for VMPro using an assisted transfer. That did not work. But I still would not know how to do it if I did a regular transfer from a phone, or if someone did a DTMF breakout from a user's mailbox.

 

[Drew2400]

I tried transfer from SIP providers DIDWW, CallCentric and FWD. No problem. I even tried H.323 (from another IPO) to SIP with no problem. All settings were at their default.

I'd take a closer look at your Firewall....

 

[Bas1234]

If you want to sent per station a different outgoing CLI.

Goto; Line > SIP Line > SIP URI Tab > main number and put that on "Use user data"
Goto; Extension > Select Extension > and now you have an extra tab called "SIP", now you can fill in the outgoing number instead of the name.
Sometime the provider wants an international prefix code like 31 for the Netherlands.

Greetzzz...Bas

___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[Bas1234]

I just looked in a back-up to the "Binding Refresh Time" but i'm wrong with the 600msec i've put on 60msec, but if your 30msec works fine leave it there.


___________________________________________
It works! Now if only I could remember what I did...
___________________________________________

 

[DavidCT]

bas1234,

I know how to set the outbound caller ID on a per-trunk or per-user basis (exactly as you describe). Maybe my question was not clear, so here is an illustration:

- Incoming call from (555)555-5555 to the IPO.

- Receptionist answers the call and then transfers it externally to a mobile phone.

- We want the caller ID on the incoming call to the mobile phone to be (555)555-5555 (the original caller's #). Currently the number that shows is what we have programmed on the SIP user tab (which is the receptionist's DID).

Another scenario is similar, except the call will be transfered out of VMPro instead of manually by the receptionist. Each user has their voicemail greeting that says, "if this call is urgent, press 2 and you will be transfered to my mobile". When the incoming caller presses 2, the caller ID on the mobile phone is OUR SIP trunk, NOT the original caller.

 

[tlpeter]

David that is the way it works

You can't send the original number external !


ACA - Implement IP Office
ACA - Voice Services Management
______________
Women and cats can do as they please and men and dogs should relax and get used to the idea!

 

[Bas1234]

OK, i think that's not possible.
There is an option like this for ISDN but this will not work for every provider. I haven't seen this option for SIP trunking.



___________________________________________
It works! Now if only I could remember what I did...
___________________________________________