similar to: Softphone recommendation

Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Quick question from a newbie, I have asterisk configured to dial IAX extensions (which works). When dialing from one IAX extension (using Firefly) to another IAX extension (also using Firefly), the Firefly client rings on the receiving end and gives the option of accepting or denying the call. However, when I dial in to Asterix using a VoicePulse number and dial the same extension Firefly

Hi Timo, I have another problem, this time with user shared folders: User "markus" shared the folder "ForTest" to test: SETACL "INBOX/ForTest" test akxeilrwts dovecot-acl and shared-mailboxes.db have been successfully updated. As user "test" the folder (and "#User" namespace) is not visible. When I configure the shared namespace with

Hello I need to hook up someone's remote PC onto our Asterisk server over the Net. There are firewalls on each side, so I figured it's time to give IAX a try, and see if it's less of a pain to use than SIP. And since IAX hardphones are pretty are, I guess I'll go softphone. Apparently, the two most well-known IAX and SIP clients for Windows are ZoIPer and X-Lite, respectively.

I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with

I can either get the TE100P working or the 3 X100P clones working, but never both. I have the TE100P connected to a channel bank, and X100P clones to lines from the phone company. This is my zaptel.conf span=1,1,0,d4,ami fxsks=1-24 loadzone=us fxols=25-27 loadzone=us I then do [root@asterix root]# modprobe zaptel [root@asterix root]# modprobe wcte11xp ZT_CHANCONFIG failed on channel

Hi list, I have a asterisk box with an E400P that was running ok until last week. The machine just stop responding and after a reboot, the module (tor2) doesn't load anymore. anyone could help? regards Eduardo modprobe returns this: asterix:~# modprobe tor2 Zapata Telephony Interface Registered on major 196 Detected Tormenta 2 Quad T1/PRI or E1/PRA at 0xfe121800/0xfe121000 irq 7

Mostly to debug/test BLF, is there a softphone or another app. which can subscribe to hints on Asterisk? Heck, it doesn't even need to be able to do calls :-) Leif

i have a lot of questions, maybe you've got some answers. I've got the following setup : 1 central server ( linux ) , 17 servers ( LTSP ) on different locations, where each server has a number of thinclients connected to it ( max 6 ) The central server / 17 LTSP servers are on the internet and share 1 VPN ( freeswan ). It would be nice if the 17 LTSP servers could use asterix on one

Hi Guys, I know no doubt this has been covered on the list a zillion time before, but can anyone point me to some good resources on using Asterix as a VoIP gateway? I would like to get two TDM400P cards in two machines attached to separate adsl connections (in two different physical locations). I'd then like to be able to plug POTS telephones into each of them (I understand I will need FXS

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi List (Again) Thanks to some answers on this list and another I now have a MultiTenant system/setup working the way that I want it to, So now my next job is to find a SIP SoftPhone that I can brand to my own company images and so on. Again an OSS would be preferred, Even though X-Lite is bar far one of the best free SIP clients I have used and it

Imagine i want to create application like SMS Alert, however it's a call alert when something happened, for example server is crashed, i want to call 100 of my staff (administrator, manager, and others) using asterix, when they pick up their phone, my asterix will play an audio file Is it possible? what is the correct hardware for this?

Hi , I am a newbie with Asterix and not sure if Asterix is a right tool for my needs. Let's suppose this scenario : I have a telephone line in one office( all calls are paid to telephone operator). In other offices I have only internet connections. Is it possible to use Asterix so that I can make telephone calls from ALL offices( without direct telecom connection) ? if so, what telephone

Hello, I inherited an Asterix phone system. I am well versed in Windows based platforms but have zero experience in Linux and Asterix, no make matters worse I have no documentation on this system. I had to change the entire networks gateway address for various reasons but now the Asterix system will not send messages via email. I think it is because of the gateway change. How do I change the

Hello, Iam lookong for an Softphone for iPhor oder Android smartphone using togehter with an headset. I tried Zoiper and CSipSimple but quality was bad compared to an desktop SIP phone. Is there an better softphone? Or are there softphone solutions for PC desktop MAC or Android with an headset? I want to save cost for desktop phones. thanks Thomas

Folks, We have several people using SIP softphones in the office. When they leave for the day, they power down their workstations, causing their registration with Asterix to quickly timeout. Here's the entry for one such extension in extensions.conf: exten => 8102,1,Dial(SIP/someone,20) exten => 8102,2,Voicemail(u8102) exten => 8102,3,Hangup exten => 8102,102,Voicemail(b8102)

Hi, I have recently discovered the project along with the PhoneJacks produced by quicknet, they could be the answer to something I have been looking into. I would like to be able to test using a dial-in server & possibily also a Windows RAS server, however I only have 1 phone line. I was thinking that I create a setup like that illustrated below to solve the problem:- Ext 1000

I am use Fedora 3, and run into a 1.4 compile issue. When 'make install' I got this message. [root@asterix asterisk-1.4.1]# make install make: expand.c:489: allocated_variable_append: Assertion `current_variable_set_list->next != 0' failed. make: *** [utils] Aborted [root@asterix asterisk-1.4.1]#

--- Raja Chidambaram <raja_chidambaram82@yahoo.com> wrote: > Date: Wed, 3 Aug 2005 06:27:05 -0700 (PDT) > From: Raja Chidambaram > <raja_chidambaram82@yahoo.com> > Subject: Phone interface hardware > To: Asterisk-Users@lists.digium.com > > Hello All, > > We are in the process of putting together an asterix > based phone network in our office. The

Hello there, I'm a new Asterisk user and I'm having difficulties to connect to and from my Asterisk server. Can anybody give me a hand? Here's some background information: * I'm running RedHat Linux Enterprise 4.0 * When iptables is stopped, my server can register with IAX service providers and receive registrations from IAX softphones. However, it does not succeed in