We have a customer on HPBX who is using an Adtran TA924 to connect quite a few of their existing analog phones into our HPBX service (analog FXS sub facing, SIP facing towards us). We have the TA924 set up so that the lines register to our Metaswitch. One thing I cannot figure out how to do is to initiate a blind (or even an assisted) transfer. The way I envision doing it is to have the customer flash hook, obtain dialtone, call phone number they want to transfer to then once it starts ringing hang up (blind transfer) OR talk the the second party then hang up to complete the transfer (assisted). No matter which combination of settings I've tried the only thing I've accomplished so far is tearing down both sides of the transfer when I hang up. I can initiate a three-way call by flashhooking again once the second party starts ringing but once I hang up, both parties get hung up on instead of joined together. I've tried transfer mode in both network and local, flashhook mode in both transparent and interpreted, and both options on unattended transfer mode in addition to having the flashook mode on the Metaswitch line in both Metasphere CFS and endpoint modes. No matter what I do I can't get the TA924 to send a SIP REFER packet to accomplish the transfer. Even if the TA924 remained latched onto the call and mixed the audio locally I'd be okay with that, but thus far I haven't seemed to be able to accomplish anything. Metaswitch's interop report shows that this is a supported feature with a TA924 and their softswitch and I also started a discussion over on the Metaswitch Communities site as well to see if anybody else has run into this issue.
I've attached a sanitized copy of the config as it is right now. Is there some sort of option I need to enable in the CLI to allow transferring?
Thanks for posting. I believe you have been working this issue with an Adtran Technical Support ticket, but if I am mistaken on that, please let me know. Given your attached configuration, there are a few changes I would recommend before getting any debug output. Below are the changes I would suggest.
voice call-appearance-mode multiple
voice flashhook mode interpreted
Once we have made those changes, we would need to see complete debug output of the call. Below are the commands we typically run.
debug sip stack message
debug voice verbose
debug interface fxs
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