search for: epshteyn

...ur proprietary solution and we can offer you this proprietary solution" isn't a contribution, it's an ad. -Barry > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Alex > Epshteyn > Sent: Saturday, April 29, 2017 08:59 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] softphone instead of desktop phones > > Thirdlane Connect can be used as a softphone. It works in modern browsers > (no installation is required)...

...tops, and on mobile phones. Besides basic softphone functionality, it provides instant messaging, group chat (channels), voice and video conferencing, and screen sharing. It integrates with a variety of applications and CRMs such as Salesforce, Zoho, Zendesk, Redmine, etc. Try it out! -- Alex Epshteyn web: www.thirdlane.com ----- Original Message ----- > From: "Amit Patkar" <amit at avhan.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Saturday, April 29, 2017 9:16:05 AM > Subject: Re:...

Is anyone out there using Druid? After the switchbox announcement today I've been looking into some other gui's and as I'll probably do a trial install this weekend of the free switchvox iso but I thought I'd ask is there any other guis I should be burning trial ISO's of as well? Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at

Hello, almost any useful CLI command has an analogue on Asterisk Manager Interface, but I cannot find a way to get the list of subscriptions using AMI. Which is the command, if any? The CLI command is "sip show subscriptions" Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have two Asterisk Systems. One on of those, when I execute this: Dial("SIP/teleglobe-007931d0", "SIP/13033372500 at teleglobe|60|oL(4007520000:60000:30000)") ... It causes Asterisk to immediately read out the time limit of the call (66,792 minutes), as soon as the other end answers, even though we aren't down to 60s remaining yet. Asterisk then goes into an infinite

...nt PBX in 2005 - we do this as our core business and are still finding areas for improvement :). As your experience with VoIP is minimal I would side with your CTO - you should find a solution high enough in the stack to avoid the complexity of building it all yourself. Good luck, Alex -- Alex Epshteyn email: alex at thirdlane.com web: www.thirdlane.com phone +1 415.261.6601 ----- Original Message ----- > From: "J Montoya or A J Stiles" <asterisk_list at earthshod.co.uk> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.d...

You can use "Command" command, and "sip show subscriptions" as a parameter -- Alex Epshteyn email: alex at thirdlane.com web: www.thirdlane.com phone +1 415.261.6601 ----- Original Message ----- > From: "Leandro Dardini" <ldardini at gmail.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> >...

...es several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Fix Sorting Order For Parking Lots Stored In Static Realtime (Closes issue ASTERISK-21035. Reported by Alex Epshteyn) * --- Make ParkAndAnnounce return to priority + 1 when return context is not defined (Closes issue ASTERISK-20113. Reported by serginuez) * --- When a session timer expires during a T.38 call, re-invite with correct SDP (Closes issue ASTERISK-21232. Reported by Nitesh Bansal) *...

...es several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Fix Sorting Order For Parking Lots Stored In Static Realtime (Closes issue ASTERISK-21035. Reported by Alex Epshteyn) * --- Fix StopMixMonitor Hanging Up When Unable To Stop MixMonitor On A Channel (Closes issue ASTERISK-21294. Reported by daroz) * --- When a session timer expires during a T.38 call, re-invite with correct SDP (Closes issue ASTERISK-21232. Reported by Nitesh Bansal) * --- Fix w...

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of ?? Sent: Thursday, February 01, 2007 9:01 AM To: Asterisk Users Mailing List - No Subject:

Hello, I believe these descriptors are specific to flang, not to LLVM. You should probably ask your question on flang-dev list. Thank you, --Eugene From: llvm-dev [mailto:llvm-dev-bounces at lists.llvm.org] On Behalf Of Venkataramanan Kumar via llvm-dev Sent: Wednesday, April 25, 2018 8:44 AM To: llvm-dev at lists.llvm.org Subject: [llvm-dev] Help on understanding assume shape array

Hi, I am trying to understand how assume shaped arrays are received and processed in LLVM IR. I am using "flang" for my front end. There seems to be an array descriptor received as implicit argument for every assume shaped array. For my test routine: ---snip-- SUBROUTINE test(a,b,Li,Lj,Istr,Iend,Jstr,Jend) INTEGER, INTENT(IN) :: Li,Lj INTEGER, INTENT(IN) :: Istr, Iend,

I'm a network admin that maintains 3 commercial Asterisk servers for my employer. I am wanting to move away from the "pre-packaged" commercial PBXs to a more "pure" asterisk setup. The systems I have utilize a nice web GUI to make changes, but it really limits what I can do beyond what they have programmed into their GUI. Would I be better off starting with: a) Plain

Hello, Iam lookong for an Softphone for iPhor oder Android smartphone using togehter with an headset. I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP phone. Is there an better softphone? Or are there softphone solutions for PC desktop MAC or Android with an headset? I want to save cost for desktop phones. thanks Thomas

Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org) (with longest history that almost every one knows it, now the last version using the PJSIP stack) 2) FreeSwitch (http://www.freeswitch.org) (A lot people