search for: xlite

Hi, I've got an Asterisk box with grandstream and xlite clients on it. No here's the thing: - I grey out all the codecs on the Xlite except for GSM - I call the Grandstream from the Xlite, the Xlite uses the GSM codec and the Grandstream uses ulaw, with Asterisk doing the conversion, everything fine - I call the Xlite from the Grandstrea, the...

Hello I have an Asterisk 1.4 server and two XLite softphones, where Asterisk and the local XLite phone are located in a LAN behind a NAT router, and the remote XLite phone is located elsewhere on the Net behind its own NAT router: http://img252.imageshack.us/img252/3749/asterisknat.png I'm having the following issue: When the _local_ XLite c...

Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work. I have traces of xlite for the invite and register this done to see if someone can hel...

Asterisk version 1.2.14 I use snom190 and xliteV3 as sip phones. asterisk send the rtp stream never to the xlite softphone. Any hits for me? *CLI> rtp debug RTP Debugging Enabled -- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack -- Called snom -- SIP/snom-00797110 is ringing -- SIP/snom-007...

Dear all, I have tried to run an asterisk instance together with XLite on a single machine (a PowerBook). The intent is to take advantage of IAX connections to easily cross NATs while traveling. While the IAX setup proved 'easy', just having to fiddle a little with working configs at both sides, I did not succeed so far in getting XLite to connect to the loca...

I have a bunch of road warriors who I've set up with Xlite clients. Unfortunately the sound quality has been intermittent at best. Sometimes it's great other times completely unusable. When it's bad one usually hears harsh static when the other party speaks or their voice gets "clipped" to static if they speak too loudly. Many of thes...

Hi, I'm wondering if anyone has experienced an issue with the XLite softphone and asterisk accepting dtmf? I can listen to my voicemail perfectly from my hardphone. However when I dial the voicemail number from my XLite softphone and enter the password at the voicemail prompt, an error appears vm-incorrect and I get an "Unable to read password" message on...

...ed baseline by calling ext. 1000 and now I have no sound from the demo extension voice prompts or music on hold. I can still call and talk between the two soft-phones. When I call Extension 1000 the console reports it is playing the 'demo-congrats' audio: -- Executing Goto("SIP/xlite-7fe7", "default|s|1") in new stack -- Goto (default,s,1) -- Executing Wait("SIP/xlite-7fe7", "1") in new stack -- Executing Answer("SIP/xlite-7fe7", "") in new stack -- Executing Set("SIP/xlite-7fe7", "TIMEOUT...

...ntax [compat] pbx_realtime=1.6 res_agi=1.6 app_set=1.6 ============ cat /etc/asterisk/chan_dahdi.conf [channels] language=fr signalling = fxs_ks ... ============ cat /etc/asterisk/sip.conf [general] language=fr port = 5060 ... ============ CLI> -- Executing [8888 at internal:1] Set("SIP/xlite-00000002", "CHANNEL(language)=fr") in new stack -- Executing [8888 at internal:2] Wait("SIP/xlite-00000002", "2") in new stack -- Executing [8888 at internal:3] Answer("SIP/xlite-00000002", "") in new stack -- Executing [8888 at internal:4] Pla...

Dear All, I have downloaded the xlite version 2.0 for windows and I made the following conf in the xlite itself as the document suggested in order to make it work with Asterisk but still it doesn't work as a matter of fact when I tried to make a tcp dump I can see no packets going between the windows client and the Asterisk server...

Hello Asteriskians! I have an Asterisk box with a simple HFC card in it and a bunch of people using the Xlite software to connect. The HFC card is connected to an internal extension on our legacy PBX. So far so good. The Xlite clients can call each other, and the internal extensions on the PBX and the Xlites can call each other, no problem. The problem is when using an Xlite to dial an external number th...

hi all, was wondering if someone could assist with a slight problem i'm having. I have asterisk setup with extensions 101 to 109 and am using xlite, grandstream budgetone, polycom ip500 and a couple of other phones. the problem is: 1. only the xlite extension (107) can receive calls. 2. all extensions can dial into voicemail and get mwi when msgs are received. 3. when dialing a non-xlite extension (from any ext), it goes straight to voicemail...

Lets presume that my both software are open. Xlute and Eyebeam But I want my calls from Asterisk to land only on Eyebeam and Not on xlite. How to set it ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090129/4011be6c/attachment.htm

Hello I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite on XP as an SIP client: http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png The problem is that by default, Asterisk doesn't rewrite the CID name + number in incoming calls, so that XLite displays whatever name I used in the 3102 and the extension the 3102 uses to register with A...

What's the secret to getting sound through Xlite? The SIP messages all look OK to me, but the sound isn't coming through. It was trying to use GSM, so I searched the archive and tried: disallow=gsm allow=ulaw Now it says that it's using ULAW but I still get no sound in either direction. Phil Skuse (MBJEJPIEUI) <phil.skuse@vicorp.co...

I am using the latest xlite phone to connect to the latest version of asterisk (20040727). When I try to make a call the xlite phone tells me "Call not approved". I used the configuration options that were listed on the wiki. The context in the sip.conf file is "from-sip". I have a matching context lis...

I have Asterisk configured and can place calls from XLite. But when I call my Asterisk box and try the extension where I'm logged in via my XLite, it doesnt ring and goes immediately to vm. I'm using AMP. Any ideas?

Has anyone managed to get these two phones to make a video call to each other ? If so, care to share how the hell you managed ? the GXV is at the latest firmware, and xlite the latest download Asterisk 1.4 trunk TIA Julian

Beginner to Asterisk, but not beginner to VoIP FreePBX front end running on a dell 1550 and XLite running on a different Woindows XP box Both boxes connected via switch on same subnet. No NAT involved On FreePBX I created a new extension 1001 with a SIP password of 1001 On Xlite, username is 1001, password is 1001, authorization user name is 1001, and domain is IP of Free PBX XLite tries to...

...ernal IP (192.168.1.115) I have put 35000 and 37000 into the rtp.conf as the start/end ports extracts of sip.conf: externip = 60.234.129.154 localnet = 192.168.1.115 localmask = 255.255.255.0 [88] type=friend secret=********** dtmfmode=rfc2833 nat=yes host=dynamic canreinvite=no Trying with xlite at the other end Registered ok, can dial both ways, just no audio at all. In the log of xlite (cant see it at the moment as im not vnc'd in at the moment) it showed the xlite machines private IP address on some of the transactions that were logged. The client has a dynamic IP address so ca...