[asterisk-users] Multiple Servers: Multiple Peers: call-limit

I would like to enforce call-limit across multiple servers. Is there any way to pass a call-limit variable between servers 01 02, as shown below? Use a global call-limit between multiple servers and peer connections. A -- 01 -- Z A -- 02 -- Z A is using round-robin to reach Z, but in the

Re: [asterisk-users] T1 Card RED ALARM

Since there are a number of setting that could be causing the alarm, AMI/B8ZS, SF/ESF, etc... Start with a loop-test, make sure the card can communicate with itself (using the current settings). Connect the following pins: 01 (RX-) -- 04 (TX+) 02 (RX+) -- 05 (TX-) Sincerely, Brian LaVallee

Re: [asterisk-users] Multiple Servers: Multiple Peers: call-limit

Store the call count in a shared SQL db. Sent from my Verizon Wireless 4G LTE DROID Brian LaVallee b.laval...@globaltank.jp wrote: I would like to enforce call-limit across multiple servers. Is there any way to pass a call-limit variable between servers 01 02, as shown below? Use a global

Re: [asterisk-users] OT - How to configure Apache2 server to receive Polycom log files ?

Thanks for sharing this. I'll give it a try ASAP and post my comments here. Thanks again. 2014-06-17 14:51 GMT+02:00 Stepan Hradsky stepan.hrad...@ha-vel.cz: Hi, I have this configuration in apache site configuration Directory /home/prov/polycom RewriteEngine On

[asterisk-users] Play announcement only once in a Call Queue after 10 seconds

Hello, how can I create the following scenario : I have a Call Queue and I want to play an announcement, but only once after about 10 seconds. The current option |periodic| |-| |announce| |-| |frequency| keeps on playing the announcement indefinitely. (it should have an option 'once' like

Re: [asterisk-users] Multiple Servers: Multiple Peers: call-limit

Something like memcachedb is also an option. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gregory Malsack Sent: Wednesday, June 25, 2014 5:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

[asterisk-users] OPTIONS Request without username - Forbidden

Hi gurus!!! I have a Freepbx with Asterisk 1.8.25.0 with a sip trunk on the pstn Every minute asterisk sends an OPTION Request, i beleived that it's related to qualify functions. The every minute annoyng answer of the pstn is 403 Forbidden. Some people told that asterisk is not sending the

Re: [asterisk-users] RPM updates

On Mon, Jan 28, 2013 at 05:21:10PM +0200, Tzafrir Cohen wrote: On Mon, Jan 28, 2013 at 01:55:09PM +, Steven Howes wrote: Hi All, Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's

[asterisk-users] Echo Cancellation when calling from softphone to mobile.

Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana

Re: [asterisk-users] Echo Cancellation when calling from softphone to mobile.

Put line side echo cancelation chip on ur PRI card. On 25-Jun-2014 10:35 PM, Anurag Rana anuragrana31...@gmail.com wrote: Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end

Re: [asterisk-users] Echo Cancellation when calling from softphone to mobile.

Is there any Software solution? On Wed, Jun 25, 2014 at 11:38 PM, Mitul Limbani mi...@enterux.in wrote: Put line side echo cancelation chip on ur PRI card. On 25-Jun-2014 10:35 PM, Anurag Rana anuragrana31...@gmail.com wrote: Hi, I am using Twinkle to call mobile phone but there is too

Re: [asterisk-users] Echo Cancellation when calling from softphone to mobile.

There are two common types of echo. Accoustic Echo: This is caused by microphone picking up audio from the speaker. This echo cannot generally be removed by echo cancelers. The solution to accoustic echo is to prevent the microphone from picking up audio from the speaker (or handset or

[asterisk-users] Asterisk 12 and chan_local

I am migrating my app to Asterisk12 and pjsip, but I cannot find chan_local, what happened? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] Asterisk 12 and chan_local

On Wed, Jun 25, 2014 at 4:38 PM, CDR vene...@gmail.com wrote: I am migrating my app to Asterisk12 and pjsip, but I cannot find chan_local, what happened? from https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12 chan_local: The /b option has been removed. chan_local

Re: [asterisk-users] Play announcement only once in a Call Queue after 10 seconds

Hi Jonas, While I don't work with queues, but you could playback announce-holdtime before putting the caller into the queue. exten = _X.,1,NoOp(Post Queue Announcement) same = n,Answer() same = n,Wait(10) same = n,Playback(announce-holdtime) same = n,Queue(real_queue) Brian On 6/25/14,

[asterisk-users] PJSIP Dial via IP fails

Dear friends This is my simple dialplan [demopjsip] exten = _X.,1,Dial(PJSIP/${EXTEN}@10.10.10.2) exten = _X.,n,Hangup() I need to dial out via an IP address, not using an endpoint, as shown above. It fails with Executing [1957408@demopjsip:3] Dial(PJSIP/federico-0002,