similar to: Extension Configuration Best Practice

Hi! Problem: I can't hear what the people at Location B i saying, they hear me but I do not hear them. They can call, I can call. Just no sound. My current setup is: Softphones/Hardphones(Location A) <-> Asterisk <-> Firewall/Nat <-> Internet <-> Firewall/Nat <-> Softphone/hardphone(Location B) I am having problems with sound, I have opened the

Hello, we want to setup the following scenario: - each user has a softphone AND a hardphone - the softphone is started with the operating system - the hardphone is connected all the time using SIP - only ONE extension for each user Both phones should ring when the user is called. We've setup an asterisk 1.4.18 and at the moment only the last registered client rings. In Asterisk 1.2 the

What's the proper syntax for implementing directed call pickup? Running cvs-head from today (9/24/05 including Mark's fixes), and tried: exten => *99,1,Pickup(${EXTEN:3}) but that does not seem to work, and there isn't an example in the configs directory. 'show application pickup' suggests the above should work with our sip phones, but apparently I'm missing

Hi, How can I retrieve those voicemails using my ip phone? and how will i confiugre it on asterisk? Please help I'm very new in asterisk. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Do?a Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

Hello, 1. Is it possible to install several SIP softphones on the same PC, have them registered to the same Asterisk server and attribute to each softphone a specific extension, ringtones or call forwarding rules ? 2. Is possible to do the same with SIP hardphones ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Does asterisk do transcoding when the call goes through the system, codecs are the same but signaling protocol is changed. example: SIP with GSM ---> IAX with GSM What quality destruction happen when I use transcoding? I know this is not a concrete/precise question, but I would like to know how is it in general. What CPU performance is needed for transcoding 30 channels e.g. from

Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -----????----- ???: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]?? asterisk-users-request@lists.digium.com ????: 2004?12?24? 7:51 ???: asterisk-users@lists.digium.com ??: Asterisk-Users Digest, Vol 5,

Good day, I have a puzzling issue that people in the IRC channel recommended I post to the list so here goes :) I am trying to call a SIP softphone from an H.323 hardphone. The hardphone is connected to a Definity Prologix R12 PBX with a MedPro card and a CLAN. The Avaya is setup to send any call to extension 1609 down an H.323 trunk group that is destined for the Asterisk server. When I call

Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich

I'm about to download Festival source, apply the astrisk diff's, and initiate basic testing. Thoughts are to download v1.4.3 (latest per the fesitval website. If anyone has an existing how-to, install notes, tips, or any suggestions I'd greatly appreciate it. Direct email is fine if you'd rather not post them. Thanks, Rich radamson@routers.com

Hey all, I've setup my asterisk install on a CentOS5 server, I've got a few IAX2 and SIP softphone clients connected on the same subnet and at least 1 external IAX2 softphone. However I'm having some difficulty getting the Polycom hardphone to function correctly. Watching the logs and debug trace it: - Registers correctly - Is able to make calls to other peers However it is not able

Slightly off topic, but maybe some can suggest something off list... Trying to upgrade a 7960 that was running skinny. I've got sip v3.2.0 installed and running, and am able to place calls via *, etc. However, when upgrading to v4.4.0 I can never get to the point of being able to place a call (eg, no dialtone, etc). I can ping the phone, look at the Network Config, etc, but I can't

Hi, I have configured a certified asterisk 13 server with chan_mobile and res_pjsip. I have a Cisco 7940 hardphone and I use ekiga as softphone client. Now the problem is, using the hardphone I'm able to call the softphone and hear everything properly. But when I call from the hardphone to some number that has to be dialed via chan_mobile, I'm not able to hear what the other side says (I

Hi, I know that SIP is using port 5060 for session initiation, but which port does it use for audio ? is it dynamically assigned ? Thanks, Adrien -- Adrien Laurent - CIO www.modulis.ca 514-284-2020 ext 202 adrien@modulis.ca

Hello, I am using asterisk 14.2 and PJSIP, with TLS transport. I?m sure I?m doing something wrong here .. In 2 distinct softphone clients (Bria and Groundwire), I am able to register successfully, and place a SIP call, with no certificate warnings. But shortly after I place that first call and hang up, I receive a certificate name mismatch error in the softphone, the error presenting me

(I'm new to Asterisk, after having started VOIP with vat on the mbone in the 90s.) I am setting up my first Asterisk system, and trying to read docs/guidance and follow best practices. I have read the 5th Edition of "Asterisk: The Definitive Guide" and like the 3rd Edition on the web it recommends that hardphones and softphones both have a unique name distinct from any concept of

Hi Rich, I had to change all my nat=yes to nat=route in the sip.conf. nat=yes seems to be ignored in today's CVS. Walter > > Message: 5 > Date: Fri, 27 Aug 2004 08:45:19 -0600 > From: Rich Adamson <radamson@routers.com> > Subject: Re: [Asterisk-Users] sip change? > To: Asterisk Users Mailing List - Non-Commercial Discussion >

Isn't it about time to lock down added functionality to v1.1 and fix the remaining bugs? There has been a significant amount of traffic on the cvs list, the irc and other channels with folks spending time adding new functionality to Head. Think its time to lock it down, fix the bugs that have been introduced, and get to "something" that the _majority_ can agree to call v1.1 Stable

Hello, We've released another update to our Asterisk GUI Client suite: http://astguiclient.sf.net/ Screen shots: http://astguiclient.sourceforge.net/screenshots.html The client suite runs on both Windows and UNIX and includes a dialer (the suite is not an asterisk configuration tool) In addition to the usual bug fixes, this is mostly an update for the VICIDIAL dialer application.

Hi there We have a Cisco VOIP environment here, with hard and softphones. I have a softphone account/etc, but I'm a Linux user and (as far as I'm aware) there is no Cisco softphone for Linux. However I can run Asterisk. So I was wondering if there is a way to "convert" a SIP phone transaction into a SKINNY transaction so that the Cisco environment thinks it is a Cisco