similar to: Multi-line phones - Asterisk uses wrong callerid

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:

Please remove this email from your mailing list. UNSUBSCRIBE Thank you. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Thursday, March 29, 2007 9:14 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 32, Issue 118 Send asterisk-users

Hi, Our operator frequently gets requests to transfer a call directly to voicemail in order for the caller to leave a message without disturbing the callee. Basicly, I'm looking for a blindxfer to vm. My first thought was to prepend a digit (eg 7) to the extension but this does not fit well with our dialplan. According to an article on voip-info.org Asterisk@Home appears to implement

Grandstream GXP-2000 with firmware 1.1.4.18 (beta) fixes an issue where the phone did not send rtp keepalives when on mute (resulting in disconnect from tech support hold and concalls) A side effect seems to be that Asterisk pops the following warning on the console... Jul 26 14:06:35 WARNING[31654]: rtp.c:463 ast_rtp_read: RTP Read too short Grandstream say they are not sure what it is but

Hi, We have a few Aastra 480ci phones and we've noticed that in order to get the phone to receive a call, qualify must be = no. Apparently the Aastras do not respond to the qualify message (or respond in a way Asterisk doesn't understand) and Asterisk thinks the phone is unreachable. However, this now prevents MWI from working properly on the phones. Does anyone know how to get MWI

Hi, we are searching for wireless IP Phones (DECT preferred) with have an solution for an external telephone book. We don't want to enter all of our numbers into every telephone, but have one location for all the numbers and every phone looks them up there, e.g. an ldap server. We have tried Kirk but they are working on an solution without any information when it will be available. Does

> > Allen Casteran wrote: > > Anyone know what the POE draw is for the Aastra 480i phones? > > We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. > > A Cisco 3560 switch will do 15.6 watts on all 24 ports. > > Just trying to find out if we need that much power. > Drew wrote: > According to Aastra tech support, 5 watts

Some light relief .... SNOM say "Please note that you will not be able to reach us by phone." http://www.theregister.co.uk/2008/03/13/dont_call_us/ regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com

I need to buy some new phones for our own offices. I've used only Polycom phones until now, but I'd like to broaden my experience. I'm trying to decide which phones to experiment with. I have these options: - A combination of Polycom, Aastra and Snom - Just Polycom One the one hand, I'd like to keep things uniform, since it greatly simplifies provisioning. On the other hand, I

Is anyone having problems with Cisco's 2960/3560 LAN switch? Problems causing "retries exceeded" in Asterisk? Thanks

On Fri, 2007-07-20 at 02:08 -0400, BSumrall wrote: > My dial plan of issues?.. > exten => s,1,Answer(60) > exten => s,2,Background(otherwise-press) > exten => s,1,Playback(digits/1) > exten => s,2,Goto(default,s,1) > exten => s,1,Playback(digits/2) > exten => s,2,Goto(default,s,1) I'm not sure why you have three different sets of priorities one and two

Hello, I feel like I understand how the dial plan works pretty well with one exception. It seems like queues are using the stdexen macro to ring the agents/extensions. Is this normal? Is there anyway to configure this differently? I realize this is a newbie question, but I have searched google/archives and haven't been able to find the answer. Thanks, Elliot --------------

Hi, I am having some issues with a new server install in Singapore. Outbound calls work fine. Inbound calls are not picked up by Asterisk. Zaptel 1.2.25 and Asterisk 1.2.28 both built from source. libpri installed wctdm and zaptel load without error Jun 6 23:34:03 fs01 kernel: [211138.372933] Zapata Telephony Interface Registered on major 196 Jun 6 23:34:03 fs01 kernel: [211138.372937]

Hello, i have a SPA3102 and asterisk v. 1.2.18. I also hev a mysql database with cdr. Well all I want is to receive incoming calls from pstn on specified sip account (suppose 8000), and to initiate outgoing calls from all my asterisk sip accounts through SPA3102 device. Someone can explain me what may i set on SPA and asterisk to do this thing. Thank you for your support. -------------- next part

Hi, We recently updated from an early Asterisk 1.2 SVN to 1.2.15 (on Debian Sarge) and the behaviour of our Call Centre queues has changed slightly. Before the upgrade, when a caller was waiting in the queue, the estimated hold time was announced as expected ("estimated hold time is less than 2 minutes ..."). Now the caller gets an announcement of their sequence in the queue

My mission is to get one * box to dial another * box' extensions. I have set this up previously without any issues by making a simple IAX trunk/extension pair on the two boxes and create a dial plan with a prefix like 9|XXX to select an extension on the other box. My problem is that I now have to do this with extremely restrictive firewalls thrown into the mix - firewalls I have no control

Hi all, We have a client that needs to setup about 80 desk phones (about 50 in one location and about another 30 in 5 different locations). Which brand/model would you recommend. We were personally thinking in recommending either Cisco, Aastra, Polycom, or Snom, for we've heard great things about them. However, having no real experience with them makes it hard in recommending one to

For some unfathomable reason, Siemens USA doesn't offer the Gigaset IP range in the U.S. I'm particularly interested in the Gigaset S685 IP. Since it's DECT 6.0, and there's an English (UK) version, I'm thinking it should work just fine, after dealing with the walwart issue (and maybe caller ID signalling). Anyone imported one from the UK and using it in the US? for how

Anyone know what the POE draw is for the Aastra 480i phones? We have switches that will do 15 watts on 12 ports but only do 7.7 watts on all 24 ports. A Cisco 3560 switch will do 15.6 watts on all 24 ports. Just trying to find out if we need that much power. Can't seem to find any info on the Aastra site. Comments?

Hi, Has somebody experiences with the Grandstream GXP2020 / 2000 phones in a business graded installation (with really traffic on .... not 3 calls a day ;-) ) Of course with Asterisk PBX (1.4.1 or 1.4.11 or 1.4 in generall) Thanks! Kind Regards, Erik