similar to: [Fwd: Re: asterisk hardware]

I've tried Idefisk and Cubix Softphones, and they both work fine, except for two issues: 1. Idefisk seems to have a longer delay between the time I can hit tones, and 2. Cubix, while can send DTMF faster, never actually connects to an Asterisk-dialed call -- I can't hear the party who answers. #2 has been asked but unanswered here:

Hi All Has anyone used Cubix / Firefly successfully with Asterisk? When someone calls a Cubix softphone, Cubix never seems to answer the call correctly. The other person just hears ringing even though it has been answered. I am using IAX as the SIP support doesn't seem to 100% either. Idefisk works 100% on the same setup. Kind Regards Garth

Hello, How to reproduce this bug (?) : 1. register sipura spa2 with 2 lines on asterisk. 2. use first line to call somewhere. 3. while using first line try to call from second line somewhere else in 3 step i hear fast busy tones on second line and asterisk console gives me this short error: Mar 27 07:01:43 NOTICE[29656]: chan_sip.c:3629 process_sdp: No compatible codecs! My sipura adapter

Hello folks, firstly, thank you for your useful and fast answers ! Is there anybody using D-Link SIP phones ? Are D-Link SIP phones ok to install in production environment ? Give your comments please. Tofik Suleymanov

Hello list, is there any open-source software that recodes g729 sound files to wav sound files ? The only way (at least) to do such transformation is with interactive form on: http://www.asteriskguru.com/audio_conversion.php Tofik Suleymanov

Hi, I have questions about the Polycom 601 and side card.... 1) In the side card the lights all time off... But all functions it's ok. I need help with extension module of polycom... All works fine... But lights not work.... So... I don't know when any person or extension is busy... Any ideas? , Olger On 3/27/06 11:34 PM, "asterisk-users-request@lists.digium.com"

Hello, We are creating a web-based conferencing application using Asterisk as the voice conferencing server. This as an open source project. We are trying to determine if there is interest of the community and perhaps work together to improve the application. Using the web application, you can upload your powerpoint presentation, manage the participants in the conference thru the web interface

Hello all, I am working on a creating some intelligent failover dial-plan logic and I'm running into something that I'd like some feedback on. Basically, it appears that if you place a call to an IAX2 peer that refuses the connection, or is unavailable, a NOANSWER dialstatus is returned. Example: -- Executing Macro("IAX2/cubix-19",

Hello, I just did a fresh svn install of 1.4 trunk everything. Everything compiles and installs just fine. When I get to asterisk-addons, I cannot select chan_mobile in menuselect. Chan_mobile is not even an option in menuselect for asterisk trunk. I tried the latest patch which failed in many places but did add an option for chan_mobile in menuselect for asterisk but it still cannot be

Before studying your configs, what have you tried so far? Did you change this? Also, try changing your second span timing from span=2,2,0,ccs,hdb3,crc4 to span=2,0,0,ccs,hdb3,crc4. Here is the documentation on voip-info for why it may be the cause of your issues http://www.voip-info.org/wiki/view/Zaptel.conf+span+syntax span definition format:

I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent On 8/20/07, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

It would be cool to get one of these and see if it can be hacked and loaded with your favorite SIP or IAX softphone. Looking at the pic, it looks like the dongle is both a soundcard and memory stick. Heck, I would be glad to have it if I could get the soundcard to work. Might as well since it is free after rebate. http://www.circuitcity.com/ssm/Accessories-for-Vonage-V-Phone-VPHONE/sem

Hi to all, I already setup asterisk on REDhat 9.0 linux machine. I will have 4 physical phone lines and 10 IP phones for it to use. I have a network setup already. Is getting TDM400P - 4port FXO from digium enough to start? Do I need anything else? Thank you

Please help me test my setup by dialing 800.564.0215 and listen to the queue for a bit. I have a quad port T1 with NFAS setup. I can dial-out but I cannot dial any 800 numbers (Global Crossing says I need LDS service and that will be a couple weeks) so I cant test it myself. I need at least 24 callers to feel comfortable enough that it is working properly. Thanks, Steve Totaro

Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on "Accept" button. Greg -------------- next part -------------- An

Any ideas? Sangoma support is closed for the evening. I have the latest Sangoma drivers and Asterisk 1.4 everything installed. When I fire up asterisk, I keep getting "Primary D-Channel on span 1 up" repeated over and over. The B channels never come up. There are no errors in any of the logs, zttool, or the wanpipe tools. Intense pri debug output: < Unnumbered frame: < SAPI:

Hello, I am wanting to close a specific channel for example; SofthangUp(SIP/EXTEN-UNIQUEID) but the problem is the channel is assigned a unique id as well. The need fits into the idea of receiving a call from a higher status user and thus closing a specific channel to allow the higher priority call to route through the dial plan to the freed extension. Any ideas welcome. Many thanks

Hello list, When new voicemail comes and i pick up the phone i hear special tones indicating that the new voicemail arrived.I've never had any problems with this feature, but several days ago it begin to behave strangely: 1. new voimcemail arrives, but i dont hear the special indicating tones when picking up the phone 2. there is no new voicemail (checked mailbox on filesystem), but when i

Hello guys, The so expected 2.0 release of Idefisk 2.0 softphone is a fact. Idefisk and Zoiper became one - Zoiper 2.06. Here are some of the features: SIP and IAX, TCP, TLS support, Multi-language support, Automatic provisioning (XML), URL handling, Outlook Integration, Native conferencing, API, Changeable number of lines.... You could read the complete Press Release here:

Hi Just moved offices in the UK and moved our Asterisk box from old one to new one. Using idefisk softphones, Junghanns quadbri card for ISDN 2e interfaces. At both offices we had one standard number and a DDI range, routed with Asterisk. We'd set up the configuration so each idefisk set its own caller ID which then got sent by the extensions.conf script. Worked fine at old place but in