[asterisk-users] Asterisk PJSIP enforce Transport

2018-11-26 Thread Benjamin Marty
Hello, I have an Asterisk 15.6.0 installation with PJSIP SIP Driver and Sorcery for Realtime. My Goal is to enforce endpoints to UDP, TCP or TLS. For that I set the 'transport' column in the endpoint to the corresponding transport in pjsip.conf. But if I e.g. set the transport to my 'transport-tls

Re: [asterisk-users] Send QueueMemberAdded Event via AMI

2018-11-26 Thread Systemmanagement
Hi Joshua, thank you for the quick answer. We will take a look at the code and maybe patch it for this purpose for us. best regards Am 26.11.18 um 17:34 schrieb Joshua C. Colp: On Mon, Nov 26, 2018, at 12:30 PM, Systemmanagement wrote: Hello everybody, we are using asterisk 16 with a realt

[asterisk-users] Xorcom PRI

2018-11-26 Thread Leonid Fainshtein
Steve, Try to uncomment "dchan=24" and remove "hardhdlc=24" in the /etc/dahdi/system.conf Best regards, Leonid On Tue, Nov 13, 2018 at 8:01 PM wrote: > Send asterisk-users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide W

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

2018-11-26 Thread John Kiniston
So, LOCAL in this context is a 'Technology' or 'Channel Driver' , Instead of PJSIP, SIP, IAX, it's sending a call to a dialplan target. Your entry in queues.conf with LOCAL/105@internal would send the call to the context 'internal' extension '105' and execute whatever that dialplan does. The para

Re: [asterisk-users] Queue not dialing out to cell phone for some reason

2018-11-26 Thread Ivan Demkovitch
Got it working! Thanks a lot again. As a bonus, is there a background on why SIP/ did not work with a sip trunk provider? :) From: John Kiniston To: Ivan Demkovitch Cc: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, November 16, 2018 3:08 PM Subject: Re: [as

Re: [asterisk-users] asterisk-users Digest, Vol 171, Issue 9

2018-11-26 Thread Ivan Demkovitch
Sebastian, Well, this can't be problem with trunk because:1. Call coming from outside, so trunk works2. sip show registry shows it registered. Trunk allows for 2 channels which is not a problem here either It's just weird that out of 4 queue member only 2 being called and log doesn't show anythin

Re: [asterisk-users] Send QueueMemberAdded Event via AMI

2018-11-26 Thread Joshua C. Colp
On Mon, Nov 26, 2018, at 12:30 PM, Systemmanagement wrote: > Hello everybody, > > we are using asterisk 16 with a realtime config and have a problem with > FOP2. We have developed a webinterface for managing the queues. If we > add a member to a queue, everything works fine but the user is not s

[asterisk-users] Send QueueMemberAdded Event via AMI

2018-11-26 Thread Systemmanagement
Hello everybody, we are using asterisk 16 with a realtime config and have a problem with FOP2. We have developed a webinterface for managing the queues. If we add a member to a queue, everything works fine but the user is not shown in the queue in FOP2 Panel. The problem is that the FOP2 Panel