similar to: Windows XP client?

I'm currently using the SJPhone softphone with asterisk for remote SIP. When I dial into the voicemail, and attempt to pass the extension, I "hear" the sounds, but asterisk is not receiving any DTMF signals. If I use the Estera softphone, asterisk does receive the DTMF signals. Normally, I'd just say "Use the Estera" softphone to myself, but that's not an option,

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?

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

We're on track for a release of Asterisk 0.4.0 soon. I'd like to try to see to it that we have squared away our SIP implementation by then, and after that point, try to keep it in tip top shape. In general, I find that SIP is extremely fragile, and every time I try to fix one bug, I end up creating another somewhere. What I need are strategies for verifying that the SIP implementation

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 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

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

This may be slight off topic, but perhaps it has relevance: My iconnecthere account no longer works for "inbound" calls through NAT using the standard configuration that they provide on their website. I have sent them a message, but I believe it will be flushed down the toilet by the first-tier support people. When I call my iconnect number, it goes directly to voicemail. There

Here are 2 different 4Gb USB drives. Flash memory: http://www.ecost.com/ecost/shop/detail~dpno~339754.asp $143 Mini Drive: http://www.ecost.com/ecost/shop/detail~dpno~306189.asp $80 How does one load Centos on these? Can you boot from CD, point to an FTP server for OS, and install on the USB drive, eventhough the system has its own harddrive? Does the install have to be for a specific

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

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

Hi All... I have been having a recurring problem with * for several weeks now. I am using it with two ATA186 devices configured for SIP. There is a USB zaptel device plugged in but to the PC but no phone is plugged into it. I have two X100P cards, each with a phone line. About once each day, * will just decide to stop answering calls from the phone lines or from an extension. The *

I played around tonight for a while trying to place a call to the answering machine at FWD. It didn't work. I sniffed the connection, and it looks like asterisk sends out an invite. The gateway at FWD then sends an Invite back, and then asterisk responds with a "482 Loop Detected" error message. I have attached the output of a sip debug for this session. Contents of the

Just change the install prefix directory in asterisk makefile to something like /usr/local/asterisk, and then you will see all the files needed for asterisk to run. > -----Original Message----- > From: Steven Critchfield [mailto:critch at basesys.com] > Sent: Saturday, 1 March 2003 22:03 > To: asterisk-users at lists.digium.com > Subject: Re: [Asterisk-Users] How could I install

Hi ;) I'm a french newbie and i installed asterisk 1 day ago. I've got an ATA186 and a computer with Sjphone installed. If i want to call the sjphone from the ata or call the ata from de sjphone everything is ok. My problem is ,that i can't call the voicemail or any other phone number ..as 600 for exemple from the ata or the jphone. I don't know why but i looked after a long

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;

Signed-off-by: ZhiHu <huzhi001 at 208suo.com> --- .../gpu/drm/nouveau/nvkm/engine/fifo/gk104.c | 40 ++++++++++++++----- 1 file changed, 29 insertions(+), 11 deletions(-) diff --git a/drivers/gpu/drm/nouveau/nvkm/engine/fifo/gk104.c b/drivers/gpu/drm/nouveau/nvkm/engine/fifo/gk104.c index d8a4d773a58c..b99e0a7c96bb 100644 --- a/drivers/gpu/drm/nouveau/nvkm/engine/fifo/gk104.c +++

Before you get too far.... The internet line jacks dont allow outbound calling. they cannot be used as trunk lines to the PSTN. the outbound code has not been written yet. I had to go buy FXO card from digium (that works much better than the Linejack) to get outbound calling to work Dave >>> fplandae@hotmail.com 4/11/2003 5:35:24 PM >>> Hi, I am a newbie. I have been