Calls being dropped on transfer

[erikk]

We have Adtran IP 706 Series phones connected to an Adtran NetVanta 7100 (firmware:   NV7100A-A4-03-00-SC-E). We follow the instructions to Transfer a call from an outside caller:

1. During an active call, select the Transfer function key.

2. Dial the extension to which to transfer the call.

3. Press the Transfer key again when you hear the extension ring. This will disconnect you from the call.

In following this procedure, when the target phone is answered, the call immediately drops. This is happening consistently and can be repeated.

Any help in solving this problem would be appreciated.

-Erik

[Anonymous]

Erik,

To get to the bottom of this I would need to see a copy of your current configuration for the NetVanta 7100 and the output from a debug voice verbose and debug sip stack messages while reproducing the issue.  If you can attach those in a reply (remove any sensitive information first: passwords, public IPs, etc) I would be happy to look at it. 

I also had some follow-up questions for you:

• Are you using a SIP trunk to a provider for the PSTN connection?
• Is the phone that is attempting to transfer part of a "ring-group" or "operator-group"?
• Does a transfer work if they wait until the remote internal party answers and then press the transfer key (attended transfer)?

Thanks,
Matt

[erikk]

Matt,

My answers to your questions:

1) We ARE using A SIP trunk to our provider.

2) The phones that are attempting the transfer is NOT part of any group.

3) Attended transfers work, unattended do not.

On ANOTHER note, I was called from our provider that our SIP trunk does not support unattended transfers. There is a technology (I didn't catch its name) that the SIP trunk doesn't have that causes the termination on an unattended transfer. Does that sound right to you?

[Anonymous]

Erik,

I would need to see the output from the debugs listed above to tell you for sure what was going on.  If you can provide those I would be happy to help further.  Also, let me know if you can get the name of the unsupported feature on the SIP trunk that the provider said causes this issue.

Thanks,
Matt

[Anonymous]

We had a similar issue and we found that if the sip trunk codec was g729 we needed to make the MOH codec g729.  This is if we are using internal MOH.

[Anonymous]

erikk,

I went ahead and flagged this post as “Assumed Answered.”  If any of the responses on this thread assisted you, please mark them as either Correct or Helpful answers with the applicable buttons.  This will make them visible and help other members of the community find solutions more easily.  If you still need assistance, I would be more than happy to continue working with you on this - just let me know in a reply.

Thanks,

Matt

[erikk]

Matt, I definitely would like to still investigate this, I have attached the files you requested. I apologize for the delay.

Message was edited by: matt -removed attachments

[Anonymous]

Unfortunately the debug was clipped off and did not include debug sip stack messages.  It does look like you are using internal MOH, so it would be worth trying Brian’s suggestion of swapping over to the g729 codec for MOH. You can do this with the global command of voice music-on-hold preferred-codec g729.  If that causes any problems you can swap it back to voice music-on-hold preferred-codec g711ulaw as it is currently set.  If that does not resolve the issue I would need to see a complete debug of a call where this happens with both debug voice verbose and debug sip stack messages enabled.  You will likely need to adjust the scroll back of your terminal program from the default since there is so much output.   If you use Putty (recommended) you can set this from Window > Lines of scrollback. Setting it to 99999 will be enough to catch everything.

Thanks,
Matt

[Anonymous]

erikk,

Any update on this?  Did you ever try the suggestion of swapping the music-on-hold codec to g729?  If that does not solve the problem I would still need to see a complete debug.

Thanks,
Matt

[Anonymous]

erikk,

I went ahead and flagged this post as “Assumed Answered.”  Let us know if swapping the music-on-hold codec to g729 helped.  If you still need assistance I would be happy to help if you can supply a complete debug with both debug voice verbose and debug sip stack messages enabled.

Thanks,

Matt