similar to: direct ip-to-ip call

i've been having a problem getting two SIP phones to bridge running through asterisk, actually one is a SIP softphone, SJ Phone, and the other is the Go2Call calling gateway. Someone suggested that I don't have the right codecs. How do I find out which codecs are installed, and how can I install further codecs? Any suggestions which would be the right one? I think hte problem is from the

Is there a soft phone for PocketPC or iPaq? If not, is someone working on it? I will be more than willing to contribute my mite if needed. Thanks, -- sudhir

Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones?

Hi There, i think all VOIP clients for Linux are unusable! i got testet: Linphone + Linphonec all in version 12.2 Kphone gophone and other... the only programm that is usable is gnomemeeting... does anybody knew some other tools? Best Regards, Mark

I'm trying to use an hfc based pci card with asterisk but every call fails falling in the congestion extension. exten => _0.,1,Dial(${TRUNK}:${EXTEN:${TRUNKMSD}}||tr) exten => _0.,2,Congestion Looking in the syslog i can see: isdn: HiSax,ch0 cause: E001B it seems that this is a terrible error when arrives... hard to tell what is the cause. Also terrible is finding a lot of material

hello i am using asterisk-1.0.9. i have a NAT problem. without NAT registration is ok. and if user is bhind NAT it is registring on asterisk. but SJPhone is showing "not registered". i think asterisk is properly sending request to UA. any comments............this sip.conf setting was working previously -- Registered SIP '5000' at 0.0.0.0 port 5060 expires 120 -- Saved

Hello, We are getting error: Call rejected: 407 Proxy Authentication Required - if a user is trying to call using * over a long latency network (around 600 ms). There is no problem when the same user is trying to make a call with low latency network (around 300 ms). I have included the debug and log messages for Asterisk. This call is done with SJphone, the same problem exists with ATA;

Hi all: I've got some core dumps when I use chan_h323. I dial an extension using h323, routed thru an E100P (like a H323-ISDN_PRI gateway). Sometimes * hangs, sometimes not. The client used for test es SjPhone (http://www.sjlabs.com/). This is the data for one core dump: (gdb) bt #0 ast_rtp_get_us (rtp=0x0, us=0x5074759c) at rtp.c:790 #1 0x41f8879c in create_connection

Can anyone recommend a client / phone that runs on Windows XP, with either a sound card or some other hardware? Ideally free, but does not have to be. Thanks...

Hello! Can anyone recommend a good/handy/nice sip soft phone? I have already done some testing with kphone and gnome meeting (which cant do sip).Can you recommend a open source project? It should mainly be practial and have a address book. I found kphone quite unstable, the address book is designed quite poor, and if you would like to transfer a call with the transfer button you cant access the

Hi List, Is it possible to install a soft client on my Pocket Loox 610 (F.C.Siemens) an register it with asterisk? any suggestions? thx in advance. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, Does any one know of a way to make a three way call from Asterisk using X-Lite. I need the ability to be able to call someone on the PSTN using my IAX provider then bring another person from a local extension into the call if needs be? I believe most three way calling is done using a feature of the phone, and X-Lite doesn't look like it supports this. Can this be

I have a Mediatrix 1204 FXO gateway setup for SIP. I would like to know if anyone has gotten this item to work with Asterisk. I need to get a 2 or 4 port FX0 gateway working with asterisk. The Idea is the following. PBX at lets say any Hotel(Analog lines FXS) - FX0 Gateway(1204) -- {Internet} -- Asterisk - local IVR system. (IVR is not at present running Asterisk old dialogic system has FX0

Is it possible to set up Asterisk without any of the cards? I'm interested in setting it up for the company I work for, but I would like to set it up and see how difficult it will be before I start having the company spend a chunk on equipment. Additionally, what phones can be used with Asterisk? we currently use a NEC Nitsuko phone system with phones, but I have been confused as how to set

I just installed an Asterisknow server can someone suggest a software to be used for a PC - PC voice comunication to test in easy way the functionalities of my server. Thanks in advance for the help -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091027/bb5e7b1a/attachment.htm

Hi All I seem to have a small problem with the music on hold button on SJPhone. I have 2 asterisk installations one from the Rapid distribution and one from the latest CVS. On the rapid dist when I press the music on hold button on my SJPhone I get music on hold. When I do the same I get no music on hold just silence. I create extension like this exten => 1111,1,MusicOnHold(Default),

I've been working my way through the softphones listed on voip-info over the last few weeks and I've not really found anything to fit the bill. Has anyone had more luck? The environment is a small call centre of 5 users. Operators often need to be able to transfer calls to other operators with different specialties, so the softphone needs to be easy to use and quick to transfer calls.

Hi Folks, I am wanting advise on a good soft-phone on Linux. I have looked at Gnophone but cannot seem to get it to compile under debian sarge. I am now looing at sipXphone seem to be picking up that it is not that stable, but perhaps someone here can advise on what softphone I can use on Linux. Thanks in advance, Hamish ------------------------------------------------------------------- |

I added those lines to my configuration, and i just see with ethereal that my client dial and the 1204 led turn on and they started to interchange packets, im newbie with asterisk i have been trying another sip server with mediatrix that work so well, but i dont know how to set it up? could u send me all the configuration i need step by step? ----- Original Message ----- From: "Wojciech

Hi Guys, Wondering if anyone has tried to make a direct SIP peer to peer call using x-lite without any registrations of any sort. I can't seem to find the setting. Thanks, bruce -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100417/92f07927/attachment.htm