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I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a

> On Thu, May 28, 2009 at 02:00:15PM -0500, research at businesstz.com wrote: >> Hello >> >> May i please know if asterisk is now supporting sip call encryption. It >> has been a requirement from one of my client to ensure that all >> conversation is well secured from any potential sniffers or inside >> hackers >> >&gt...

What CRM your going to use? With regards N.Prakash From: Rusty Newton Sent: ?28-?06-?2014 01:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Popup URL for outgoing calls. On Sat, Jun 21, 2014 at 5:57 AM, Inventions <research at businesstz.com> wrote: > Can anyone tell me how to implement a popup URL native asterisk when > making outbound call? > > For example, a user (A Part) dial from a softphone number 07112233, when a > call is received (or even before) by B-Part, a CRM pops up with > information for user 071...

Hello May i please know if asterisk is now supporting sip call encryption. It has been a requirement from one of my client to ensure that all conversation is well secured from any potential sniffers or inside hackers Please help or suggest any solution that you feel may help Kind regards Sam

Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam -------------- next part

...discover how they work, then you can implement it on your asterisk. But it's going to take a lot of time. Sorry for my bad English Regards, Ikka Vertika Jakarta - Indonesia Rusty Newton <rnewton at digium.com> wrote: >On Sat, Jun 21, 2014 at 5:57 AM, Inventions <research at businesstz.com> wrote: >> Can anyone tell me how to implement a popup URL native asterisk when >> making outbound call? >> >> For example, a user (A Part) dial from a softphone number 07112233, when a >> call is received (or even before) by B-Part, a CRM pops up with >>...

Hello there I have been trying to install HP Storageworks DAT72 on CentOS 5 in vain. On system reboot, neither /dev/st not /dev/sg is available. May you please lead me through as this is my first time trying to do it lsmod Module Size Used by ipv6 267617 40 xfrm_nalgo 13381 1 ipv6 crypto_api 12609 1 xfrm_nalgo autofs4

Hi Team Can someone advice me on how i can lower the load average on my asterisk server? dahdi-linux-2.1.0.4 dahdi-tools-2.1.0.2 libpri-1.4.10.1 asterisk-1.4.25.1 2 X TE412P Digium cards on ISDN PRI Im using the system as an IVR without any transcoding or bridging ************************************** top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75 Tasks: 149

Hello Team I have installed the new DL580 and used the new TE420B to add capacity on our ivr. Before I put new E1?s I decided to first move the old e1 from the old system to this new one but it has errors which not only affect the audio quality, but also cause the asterisk to refuse any call after sometime even though the channels seems up and active {seems d-channel fails}.. When processing

Greetings List Im interested to know how long the setup time is for a particular call on asterisk. Is there any defined parameter that i can use to real this behavior? SETUP TIME = TIME BEFORE THE B-PART START RINGING Thank you in advance Sam

Has anyone tried to replace Witness or Nice recorder with asterisk. I saw a nice article on voip-info.org on how to replace voicemail server for Avaya Definity with asterisk. The idea behind is to record not only the external channels but also extension to extension (three way calling for which the third leg is asterisk PRI will do) Any suggestion will highly help Sam -------------- next part

Hi Shahnawaz Have you considered how you are going to address location issue for Mobile users calling 911. You should think of SS7 MAP/TCAP to atleast know their Cell ID Regards Sam > Thanks very much everybody who contributed their thoughts. I would try > to get some DID's so that each physical location can be identified by > 911 call Center. > > Regards > > Shahnawaz

Hi there I remember to ask this question in the past but now I have thought of something little bit difference. While I understand that asterisk dialplan accept the call to be answered[ Answer() ] in the dialplan, I wanna know if this is possible; i. A call on legacy PBX, extension to extension is made. ii. On call bridging, the legacy PBX initiate a third bridging to the recording system via an