During an attended transfer using the SIP phone feature buttons, Im getting a few complaints from recipients that they cant tell when the call they are receiving has been transferred. Is there any way (even if its complicated) to generate a beep t..
Im currently testing a so-called VQ RTCP-XR feature from a a SIP hardphone.When a phone has enabled this feature, it would send a SIP PUBLISH to its SIP Server letting this server dispatch to whatever is needs to.These messages are sent during ca..
2017-02-16 12:21 GMT+01:00 A J Stiles :Yes: I missed this Code tab completely. Using it, I could check out the code Using Download snapshot at the top of this Code tab also let you download a zip file (and avoid installing svn just for that).Thank ..
While googling, Ive just discovered Recqual. If Im not mistaken, projects sourceforge site  does not host any source or binary.Is there an alternative location to download this ?Suggestions ?Best regards http://blog.krisk.org/2008/12/introducing-recqual.html..
I have a user that prefers Soft SIP phone install on his laptop, for security reasons I have enable TLS on our Asterisk server to support TLSauthentication, It works well with hard phones. Has anybody in this forum use SIP Soft phones with TLS authenticat..
Ive read a couple of pages about VQ RTCP-XR (RFC 6035) which is implemented in some SIP hardphones.Do you have any experience to share about using VQ RTCP-XR ? Does it help to hightlight issues when those issues come from a provider of your ITSP ?Wh..
Sharing to the groupDigium Announced the Call for Speakers is Now Open for AstriCon 2017The 14th annual Asterisk user conference and expo to be held Oct. 3-5 atthe Omni Orlando Resort at ChampionsGate in Orlando, Fla. – Digium blog post: http://blogs.digium.com/2017/02/02/astricon-2017-october-orlan..
Asterisk Users.I have an issue with receiving fax on my Asterisk/SIP channel. I keep getting timeout under T.38 negotiation – see http://pastebin.com/6eCe26YMAny help would be greatly appreciated..
Ive got a 13.13.1 system using PJSIP stack on debian Jessie. It runs from 50 to 100 simultaneous calls (so 100 to 200PJSIP channels)all day long. From time to time, roughly meaning once a month, it segfaultswith lines(from dmesg -T output) like this:asterisk[116..
Asterisk Users,Hope you all doing fine!I am working with a quite complex dialplan, and Ive come to some situations where it makes some nasty use of pre-bridge handlers. The pre-bridge handlers wiki (https://wiki.asterisk.org/wiki/display/AST/Pre-Bridge+Handle..