cancel
Showing results for 
Search instead for 
Did you mean: 
cancel
2386
Views
0
Helpful
3
Replies

One-way after a tranfer on Cisco Phone 88xx & 78xx

romuald.goux
Level 1
Level 1

Hi,

 

I have an one-way issue, when a call transfered on a Cisco Phone 88xx and 78xx, not on a Cisco Phone 79xx in SIP mode.

have you ever encountered this problem?

When we connect to the phone webpage, we can see the active RTP stream, and I see the incoming and and packets. But on the 88xx phone the user can't hear the caller.

The PSTN access is managed by a SIP trunk configured on a CUBE.

 

Regards

2 Accepted Solutions

Accepted Solutions

Sadav Ansari
VIP Alumni
VIP Alumni

This issue is for both incoming and outgoing call when you transferred?

 

Are you facing issue in both internal transfer and external transfer ?

 

did you check what codec is using by calling party and called Party ?

 

Did you make changes on region like both same region and different region can  use g711 for internal call ?

 

Try setting g711 codec for the region setting between sip trunk and ip phone as well. You can try it during off production hours and see if the issue goes away. Else, we can check the detailed ccm traces to see if the calling party is being updated about the final destination upon transfer.

 

Pls rate if its “Helpful”. If this answered your question please click “Accept as Solution”. 

 

View solution in original post

What happened for local transfer ? And can someone  from outside reach the final destination directly ? What about your region settings and what codec you use. are these all phones under same device pool ?

 

Refer below Thread.

 

https://community.cisco.com/t5/ip-telephony-and-phones/sip-inbound-one-way-audio-on-transfers/td-p/2396698

 

 

 

 

 



Response Signature


View solution in original post

3 Replies 3

Sadav Ansari
VIP Alumni
VIP Alumni

This issue is for both incoming and outgoing call when you transferred?

 

Are you facing issue in both internal transfer and external transfer ?

 

did you check what codec is using by calling party and called Party ?

 

Did you make changes on region like both same region and different region can  use g711 for internal call ?

 

Try setting g711 codec for the region setting between sip trunk and ip phone as well. You can try it during off production hours and see if the issue goes away. Else, we can check the detailed ccm traces to see if the calling party is being updated about the final destination upon transfer.

 

Pls rate if its “Helpful”. If this answered your question please click “Accept as Solution”. 

 

What happened for local transfer ? And can someone  from outside reach the final destination directly ? What about your region settings and what codec you use. are these all phones under same device pool ?

 

Refer below Thread.

 

https://community.cisco.com/t5/ip-telephony-and-phones/sip-inbound-one-way-audio-on-transfers/td-p/2396698

 

 

 

 

 



Response Signature


romuald.goux
Level 1
Level 1

Hi Sadav and Nithin,

 

Thanks for your quick answers.

I tested by disabling the OPUS codec, this solved the problem.

after I prioritized the G 711 codec and reactivate the OPUS codec, the transfer is operational.

 

thank you very much for your time

 

Romuald