similar to: VOIP phone, how to use with asterisk ??

Hi, I tested asterisk server with Xpro program, and all the function working well ( like 3 way calling, transfer.... ). But on the VOIP phone, I don't know press which key for 3 way calling function and transfer function... Can anybody teach me ? thanks

We're having a strange problem with our receptionist. She runs an xpro softphone and we're using a queue to handle incoming calls. It seems nearly all of the calls that come in through the queue get dropped. At first we thought it might have been human error (clicking the wrong button in xpro or something) or that the person waiting in the queue just gave up and hungup, however it

Hello All, Anyone here has experience of accepting a ilbc call and sending it on g711 or g729 I am having problem in VOICE , call goes though but there is no voice. Senario: Call is coming in from Machine A to Machine B, sending to Machine C Machine B is an asterisk box, transcoding it from IBLC to G711 and g729. Problem: Voice is not appearing on the sip user sitting on machine A Already

Olly Betts <olly at survex.com> wrote: > On Fri, Mar 30, 2018 at 05:21:43PM +0000, Eric Wong wrote: > > Hello, is there a way to optimize sorting by certain values > > for queries which return a huge amount of results? > [...] > > $enquire->set_sort_by_value_then_relevance(0, 1); > > If you're just wanting the 200 newest, it'll be faster not to

Hello all, I am trying to complete my dial plan and have come up with an interesting situation. My configuration is set up with 12 xlite SIP clients on SUSE linux workstation. They are calling out via 10 analog lines, TE110P->rhino 24 fxo. It all works and dials out great ... but ... this unit was brought in to handle the "global" office. So the help desk support on the Suse

-----Original Message----- I'm crazy here trying to make X-Pro use ONLY g729, and you're struggling to make it not to use it :)... Can you please indicate what's your config for X-Pro and sip.conf? sip.conf: [12345] type=user username=12345 secret=12345 nat=no host=dynamic reinvite=no canreinvite=no disallow=all allow=g729 allow=g729a allow=g723.1 allow=g726 allow=ulaw allow=alaw

Anyone have recommendations for a gateway / ATA for business that can do GroundStart? Preferably with an rj-21 - but okay if not.. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110614/42cbd90d/attachment.htm>

Hi, We have a page on the wiki just for phone reviews, but I think it needs a bit of format change. Instead of individual reviews for each phone, I think each person should review all phones they have worked with and list the phones they have had access to and rank them in relation to each other. Also each review should have a date so the reader can see how fresh the data is to current.

Greetings, asterisk list and community, I have a problem in how our telefon switch (Siemens HiCOM) "talks" with my new configured Asterisk server (V.11.18.0) without my Asterisks server in the middle.... <phone> <--> Siemens HiCOM <-ISDN-> NTBA <-...-> PBX Telekom A phone connected to the switch requests an "Outgoing" line by dialing "0".

Hi everybody. I know maybe this question has been posted some time ago, but I need your updated opinion on the subject. I'm replacing our old pbx with asterisk. I have two TE207 dual pri (e1) cards on a clustered system (one on each node). I absolutely need to connect 4/5 analog extensions with modems, they're being used for remote assistance on very old systems which cannot be upgraded

Does anyone have any views on the best software base SIP client to use that normal users could use with Asterisk without being too techie ? I have tried the X-Lite client with varying success. The first version worked OK but music on hold broke the voice paths and the slightly newer version initiated the call but failed to make the voice connect in both directions. The SJphone client works but

Polycom 601, hands down. - Brad -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Warren Sent: Thursday, June 15, 2006 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Best $300 VoIP phone for asterisk? If you had approx $300 per phone as a budget and needed to buy

Hi everyone, I compiled parts of LLVM to JavaScript using emscripten and made a demo of parsing and executing LLVM assembly, http://kripken.github.com/llvm.js/demo.html Basically you enter some LLVM IR, press a button and see the output of compiling and running it, directly in the browser. This was done mainly as a fun hacking project over the holidays, but I'm posting it here in case

This seems to have been reported before, but I've seen no resolution: http://lists.digium.com/pipermail/asterisk-dev/2003-August/001470.html http://lists.digium.com/pipermail/asterisk-users/2003-August/019091.html http://lists.digium.com/pipermail/asterisk-users/2003-October/024472.html When forcing use of Speex, no sound comes out at all (Speex-1.0.3 on the Asterisk server) When forcing

Hi all, I just checked out that Asterisk which is a platform I am interested of. I would like to install it to the Linux box for a trial. I have some legacy Dialogic hardware on hand, don't know they will work with Asterisk or not. For analog loop start interface I have Dialogic D/41 E which is of ISA bus with four Telco interfaces. Will it work on Asterisk? Best regards, Patrick.

Hi All, I have setup asterisk 1.4.22; so far everything good. Except, I am still searching for voIP phones. Which grandstream phone should I buy, this is going to be for small office for testing purposes. I am on a budget, hoping to find someone here who has some used to sell or point me in the direction of a seller. I am in the US. thanks, Mike

Hello, a user outside the office regularly gets a call from ext. 101 but that extension does not exist in my extensions.conf. when the user pickup the phone no one answers. Any Idea how to fix this issue? that user uses Polycom SP 450, Thanks in advance, Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear all, Due to the power failure, so i need to restart my redhat linux server. but i got below error messages, pls everybody can help/teach me fix the problem by return mail. ------------------------------------------------------------------------------------------------------------------------------------------------ Yours system appears to have shut down uncleanly Press Y within 1 seconds

Hello, All I'm looking for is a yes/no answer here. I have heard that the following scenario is possible (reasonably easy to implement as well) . but I just don't get it :-) . if it is possible I'll go ahead and learn on my own, I just don't want to waste time on something that will not work. Scenario: 2x VoIP phones - Each phone is configured to register

> > Olly Betts <olly at survex.com> wrote: > > > > > > The reverse order (ENQ_ASCENDING) is really fast - about 0.0001 seconds. > > > This is because in that case we can just stop once we've found 200 > > > matches. With a few million documents, that ENQ_ASCENDING sounds promising :) So, it looks like if I had ideal ordering, I could do