similar to: Dial() Command Parameter L Overflow?

On 30 April 2017 at 16:54, Tech Support <asterisk at voipbusiness.us> wrote: > I thought this was a non-commercial list. > > Yeah, I wouldn't mind so much if it had actually answered the original poster's query. "Switch to our proprietary solution and we can offer you this proprietary solution" isn't a contribution, it's an ad. -Barry > >

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

@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

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:

Thirdlane Connect can be used as a softphone. It works in modern browsers (no installation is required), on Mac, Windows and Linux desktops, 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,

The solution you choose should be based on many factors which should include your business requirements, team's experience, your budget, growth expectations and more. You can choose Asterisk or Freeswitch as a platform and start building on that - but it is not simple and being new to VoIP you are likely to make mistakes. The "do-it-yourself" approach will some money initially, but

Stopping and restarting Asterisk will lose the hints, then you will have to wait until the phone registers again. With 1.2.7.1 a reload shouldn't lose anything. Change the register time on the phones to something less that 60 minutes if it's a big problem. Instead of factory defaulting the phones you might find a simple restart will re-register the phone and BLF will work on the phone.

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 configured both asterisk and grandstream 2000 accourding to howtos on the web.. And everything seems working fin. But if i reload asterisk grandstream stops working with BLF. I need to restart the phone to enable BLF again. Any clues??

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

Hope someone can help. I have a situation where asterisk is sending a SIP CANCEL message before the Dial() timeout has hit. It doesn't always do it. Normally, we send an INVITE to the ITSP. They respond with a 100 Trying, then a 180 Ringing, or 183 Session Progress. It seems to be at this point that Asterisk starts the dial timer. Normally, when no more replies have been received by the dial

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

Hi all If I have the following in my dialplan: exten=>25001,hint,SIP/25001 Doing a core show hint 25001 results in 25001 at local : SIP/25001 State:Idle Watchers 0 1 hint matching extension 25001 in the Asterisk CLI. What does the Watchers 0 mean? I use the hints table output via core show hints for logic in my dialler application - but

One of our clients is using a Grandstream GXP2000 with an attendant console. We have used the same phone with past clients successfully however this particular operator processes around 200 calls a hours and the GXP2000 for sure does not like the quick line shuffling and call volume. We get the following problems randomly: 1. menu stops working 2. transfer key stops working 3. Line 1 LED gets

Version - Asterisk SVN-trunk-r12793M (1.2.4) I have 4 Grandstream GXP 2000 phones configured. However at the moment, I have had to disable BLF, Hints, and Call Limiting due to an extremely annoying bug which seems to make the phones channels "lock" in busy after a call has been hungup. If I do a show hints after say extension 200 has hung up I get the following -= Registered

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

Does anyone know the cause of these BAD BAD BAD messages? I think I lost all my calls when it happened too. We have nagios running against IAX and nagios reports that IAX is down. It would seem that the entire application locks up when this happens and calls are dropped. Connected to Asterisk 1.2.14 currently running on flexo (pid = 26846) Verbosity is at least 3 flexo*CLI> show channels

Ishfaq Malik <ish at pack-net.co.uk> schrieb: > The hints have to be in the same contexts in extensions.conf as defines in > the sip.conf subscribecontext which can be set per peer. Well, [anika_incoming] will be included in [default], of course... But I tried to define anika_incoming in subscribecontext, too. No changes... > Also, have you configured the phones as well? What do

Hi all, I am working on asterisk 1.2.18 zaptel 1.2.17 Polycom 650 polycom 430 SIP version 2.0.3.0131 for IP 650 SIP version for IP430 2.0.3.0127 freepbx 2.2.1 I am trying to configure BLF using asterisk but failed. I would be thankfull if somebody help me. Regards FArooq ********************************************** 1 ********************************************** in my

In testing Asterisk 11, I've found that Asterisk doesn't seem to be sending BLF updates to SIP peers that have subscribed to a hint looking at a Custom device if that Custom device state is RINGING or RING_INUSE. All other states seem to be working correctly. The hint section of the dialplan is: [hints] exten => _3XX,hint,Custom:${EXTEN} Console shows the following for core show