search for: sjlabs

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

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

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 (call_reference=1349809548) at chan_h323.c:928 #2 0x41f8f34b in MyH323Connection::CreateRealTimeLogicalChannel(H323Capability const&, H323Channel::...

...************ > ************ > > > > [sip] > > ignorepat => 9 > > exten => _9NNNNNNNN,1,Dial(SIP/line1) > > exten => :9NNNNNNNN,2,Congestion > > > > But it just put the box in busy and interchange rtp G711 packets with my > client SJphone form sjlabs > > I would like a helping hand! > > -- > > _______________________________________________ > > Get your free email from http://www.hackermail.com > > > > Powered by Outblaze > > _______________________________________________ > > Asterisk-Users mail...

...t 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 useragent "SJLabs-SJphone/1.40.258" for peer 5000 [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes nat=yes canreinvite=no [5000] type=friend port=5060 canreinvite=no host=dynamic nat=yes insecure=yes auth=plaintext ____________________________________________________ Start your day wi...

...9EF@82.198.1.15 Content-Type: application/sdp From: "2000"<sip:2000@xxx.xxx.xxx.xxx>;tag=608598751280 CSeq: 1 INVITE Max-Forwards: 70 To: <sip:2002@xxx.xxx.xxx.xxx> Via: SIP/2.0/UDPyyy.yyy.yyy.yyy;rport;branch=z9hG4bK52c6010f0131c9b14237ee5f00007 9bb00000045 User-Agent: SJLabs-SJphone/1.40.258 ===== 12 headers, 16 lines Ignoring this request Transmitting (no NAT): SIP/2.0 503 Unavailable Via: SIP/2.0/UDPyyy.yyy.yyy.yyy;branch=z9hG4bK52c6010f000000244237ee60000052da000 00047 From: "2000"<sip:2000@xxx.xxx.xxx.xxx>;tag=608598751280 To: <sip:2002@x...

...xx.xxx.xxx.xxx SIP/2.0 l: 214 m: <sip:4802@192.168.1.111:5060> i: 08c50c2469d676562285f02f72e5f6be@xxx.xxx.xxx.xxx c: application/sdp Max-Forwards: 70 CSeq: 13 INVITE f: <sip:4802@xxx.xxx.xxx.xxx:2841>;tag=41280171719448 t: <sip:asterisk@xxx.xxx.xxx.xxx>;tag=as7cf27066 User-Agent: SJLabs-SJphone/1.30.252 v: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKc0a8016f0131c9b142227b690000594d00000078 v=0 o=- 3318544820 3318544833 IN IP4 192.168.1.111 s=SJphone c=IN IP4 0.0.0.0 t=0 0 a=direction:active m=audio 16394 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp...

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...

...f file to register a soft phone? I tried the following entry and the phone will not register: [xten1] type=friend username=xxx secret=ppp host=dynamic I am using the X-Ten Lite soft phone, but I understood that I didn't have the correct build. The next thing I did was download the SJLabs soft phone and tried it, and got the same thing. If I can get the Asterisk side (sip.conf) configured correctly, then I think configuring the soft phone will be a cakewalk. Can someone please help me with this? Thanks Steve Lane -------------- next part -------------- An HTML attachment was s...

Anyone have any suggestions on free sip phone software for windows?? Only have one IP phone and want to have one other computer hooked up to my Asterisk box for testing.

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, I was just wondering, is there any H.323 soft-phone that can be installed on a pocket PC (iPAQ). Walid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050116/f75bd2fe/attachment.htm

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

Hello! I'm searching for a way to call ATA (IAX or SIP) that is not registered with any server or proxy. Is it possible to make such a call from a softphone to an ATA just with IP? Something like (sip:// or iax://)1111@210.12.34.45 (where 210.12.34.45 is ATA's public ip)? Regards, CuPoTKa.

...). The TFTs they have are 1280x1024 and operators prefer the > larger font size. Many of the softphones I've tried end up with data > elements appearing in weird places (or not visibile at all) with the larger > font size. > Try to use SJphone. It's free and easy to use. http://sjlabs.com -- Sincerely Yours, Andrey Loginov Insource LLC. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users

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