Issue with transfer to specific Calling number

[schoerlin]

Hi,

we have a strange issue.

We are company, which are located in several countries. For all outgoing calls, we are using one Sip-Gateway which is located in Israel.

Now we have the following issue.

1. We call a mobile number which is located in UK (the codec that is used is G711alaw)

2. We do an internal transfer to another phone in the company. The call can be picked up, and we can speak which each other. To transfer the call I press the button "Transfer" on the phone

3. The call is still alive, but on both ends you are not able to hear each other. The signalling is working, the Voice stream not. At that moment the codec is G711ulaw

We only have this problem for mobile numbers in UK. Mobile phones in Europe are working. Landline in UK are working as well.

If I do a conference call, instead of Transfer, it is working as well

If I use the local breakout from UK or from Switzerland it is also working

Does anyone has an idea??

Thanks in advance

Frank

[Manish Gogna]

Ideally we need to compare the signaling for a successful transfer to a mobile no in other European country with the transfer to a mobile in UK. The IP addr / codec / port info in the SDP needs to be looked at to see if there is a problem with that. if the signaling part comes out to be good then the network hops in the call flow ( routers/firewalls ) need to be investigated for any routes/ports being blocked.

HTH

Manish

[Chavdar_Baramov]

Seems like codec mismatch to me.

I can think of 2 solutions:

1) reorder codecs on your low loss preff list. and put a-law before u-law

2) Use MTP on the trunk (if yes - check do you have free resources)

[Saurabh]

Based on your scenario, when you  finally transfer the call to connect 2 parties, the call is active but not able to hear each other

To Verify this, you can check the codec's negotiated in the SDP header for transfer call. For the gateway, you should also verify what channel is getting negotiated with PSTN as in the transfer scenario PSTN number will be hearing MoH and ones you connect the mobile to next party. there could be chances that PSTN gateway doesn't resume with the channel.

If still, you are facing this issue, provide me with the logs of this call from call processing node and outbound gateway

[Dennis Mink]

Can you add the debug ccsip messages output for a successful transfer and a failing one on yout SIP gateway (assuming its a Cisco Box) so we can compare the two, also state the DN's used.

Please rate if useful