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Hold and Transfer doesnt work for specific type of call in a sip trunk.

Rafael Jimenez
Level 4
Level 4

I have a curios case with the hold and transfer features in cme and 3905 sip phones. There is not sccp phones.

The SIP SP give us a sip trunk with two headers, 3185840 and 3185860. I already setup the AA for both numbers.

Incoming Calls from SIP TRUNK.

When a called number is one of the headers, the transfer and hold works fine.

If the called number is another number using DID, 3185861 for example, the hold and transfer doesn’t work.

When the I pick up the phone, and press the hold (or transfer) , 3905 softkey the call is dropped. The calling party hears a busy tone.

The translation rule for the incoming dial-peer is simple, rule 1 /^.*\(....\)/ /\1/ , that rule just take the last four digits of the called number. The last four digits match with the phone extensions.

The attached debug has two scenarios:

  • •1.       Calling number ( xxx-4339866 )
  • •2.       Called number 3185861. (hold and transfer doesn’t work). Reason: Q.850;cause=65
  • •3.       Called number 3185860. Hold and transfer works fine.
  • •4.       The final number (transfer) extension 5863.

Please I need some help, thanks.

3 Replies 3

Gregory Garrian
Level 1
Level 1

Cause code 85 is a media negotiation failure.  Sound like a possible codec mismatch.  Do you have an MTP configured?

I have configured conference resources.

no xcode. May be the problem is a mistmatch between the codec listed in the voice class codec

and the dspfarm profile 1 conference:

!

voice class codec 1

  codec preference 1 g711ulaw bytes 160

  codec preference 2 g729br8 bytes 20

  codec preference 3 g711alaw bytes 160

!

dspfarm profile 1 conference

  codec g729br8

  codec g729r8

  codec g729abr8

  codec g729ar8

  codec g711alaw

  codec g711ulaw

  maximum conference-participants 8

  maximum sessions 1

!

I will change that config and I will let you know.

Thanks.

Solved with :

no supplementary-service sip moved-temporarily

no supplementary-service sip refer

sip

  g729 annexb-all