The BCM50 sends SIP INFO messages for DTMF. using Asterisk’s SIPDtmfMode() application, the DTMF send mode can be forced to in-band. Since the call has been set up as G711, in-band DTMF will still be intelligible by the far end. I have tested this in our lab set-up, and it works fine, so that is issue 1 out of the way. As for issue 2, I have placed a call to a number which I know will not answer (Looks like I’m the only mug that works on a Saturday 
). The call rang out for over 2 minutes before receiving a SIP 486 (user busy) from Voiceflex, with their Asterisk server setting X-Asterisk-HangupCause as “Network out of order”. 2 minutes seems like plenty of time to be ringing to me, most likely this is limited by a max ringing timeout timer somewhere along the line, either at Voiceflex or on the PSTN.
Scott Donnelly’s Blog - A Personal blog, plus my VoIP, Web Dev, and Astronomy musings.
9th
Aug
08
Hi, I found your blog on this new directory of WordPress Blogs at blackhatbootcamp.com/listofwordpressblogs. I dont know how your blog came up, must have been a typo, i duno. Anyways, I just clicked it and here I am. Your blog looks good. Have a nice day. James.
Hello cousin Scotty!
Hey,
Noticed through a google search you use asterisk and voiceflex as a sip provider…. I’m having registration issues with voiceflex, and was wondering if you could give me a hand… currently the SIP accounts show as reachable, but voiceflex and it’s portal is sure that we aren’t registered. ’sip show registry’ appears to be blank…
any inspirational thoughts?
Many thanks!
Hi Steve,
I have only just seen your comment, sorry for the delayed reply, I have not visited here for a few days! Anyway, I have emailed a copy of the config files that I used for Asterisk. These are running on a small embedded Linux device, the Linksys NSLU2, but the relevant SIP config should work with a few minor alterations. They work fine on Voiceflex’s partner company Frontier’s VxDSL broadband offering.
I have found the most efficient way to troubleshoot is to use Ethereal to sniff the SIP packets in either direction, tweaking the asterisk configs and reloading them as I go. If you have any further problems, let me know.
Scotty
Scotty
I have been playing with AsteriskNow X-Lite and Voiceflex to swap out my Avaya IPOffice – I would appreciate a copy of your Voiceflex config files as I don’t seem to be getting anywhere with it !!
Many thanks
Matthew
Hi Matthew,
If you are configuring Asterisk to trunk to Voiceflex, then yeah, sure, I can help with that. I am out of the office until Monday – I will post the details when I am back from the weekend. As for X-Lite to Voiceflex – here are the settings.
Scotty
Great thanks for that – I’d appreciate the Asterisk Voiceflex config when you get a moment – then I’m done.
Matthew
Scotty
I have solved the voiceflex config problesm I had – well I didn’t have a problem – there was an outbound call bar in place so I couldn’t make any calls but inbound was OK. ANyway the call bar has been lifted so everything is working.
Thanks for your help.
Matt
Perhaps you could post your working Asterisk & Voiceflex config. I cant get mine to work!
Hi Steven,
I have published a new post, detailing my configs for Asterisk on SIP Trunks. hope this helps.