similar to: Trouble Connecting Xlite to Asterisk

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 listed in the extensions.conf file. The phone is able to register

-- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 -- Attempting native bridge of SIP/1009-3df0 and SIP/101-f8f1 RFC3389: 5 bytes, level 0... Aug 27 08:32:16 NOTICE[23572]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible Killed Whenever I make a call between extension 101 and 1009 which are both Xten Xlite SIP clients, I get that error and

I have d/l the iso (*@home 0.9) , built the * box and followed the directions in the * handbook and http://www.geekgazette.com/index.php?option=com_content&task=view&id=2&Itemi d=26. I created extension 200 and verified that * was running fine. Loaded Xten lite, setup the proxy for local ip (10.0.0.201) per the handbook. After turning off the Norton Firewall protection, I am able to

Folks, I have made an inbound call. Now I want to make an out bound. I have a sip softphone hitting the pbx. On my motorola firewall I have allowed all traffic in and out to my address. There is not a firewall on the PBX yet. here is the log off the softphone..and the PBX logs look like this Registration from 'Tommy Denton <sip:201@209.101.93.30>' failed for '

I've google for hours trying to find a discussion of a similar problem as the one I'm having, so forgive me if this has come up before. If it has, please point me in the right direction! The problem occurs when a caller (A) is transferred by an intermediary party (B) to voicemail (Voicemail or VoicemailMain), either directly or by being taken to voicemail when the callee (C) doesn't

Hi, I have the following situation: I've installed Asterisk at Machine 1 (M1 - IP: 192.168.1.145) and X-Lite (X_lite-Xten-Win32-1103m.exe from www.xten.com) at Machine 2 (M2 - IP: 192.168.1.100) and Machine 3 (M3 - IP: 192.168.1.200). I need to catch the SIP and MGCP messages that will appear when M2 calls to M3 and vice versa. The SIP messages are working (I don't have problems with the

Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite

Hi I tried Xten, its very good, because it can stay in the taskbar (next to the clock) and start when windows starts, and is allways ready to receive calls. Maybe it s the best way to introduce VoIP to my company workers.... But theres a feature that s missing (or I couldnt find), there s no way to connect this softphone with the adress book. I think this feature is very important, because

hi there! I have a couple phones connected to a sipura ata and if I go into *- IVR, I press options on the regular phones and it all works fine and dandy. then I connect an xten softphone, a new extension in my dialplan, I dial the ivr, * asks me to dial something to go through it, I press keys on xten, but nothing happens, * just times out through as if I did not press anything! is there some

Which rpm has libstdc++.so.5? xten-xlite for linux says it needs this. Of course, there will be something else it will need after I get this....

Using the Xten X-Lite client (free) I am able to connect to a local asterisk@home server and when trying to connect to the remote server (a mirror of the local) I am unable to connect. The first server is a local lan, the remote is using microsofts pptp vpn client to connect. Looking at the diagnostics from X-Lite I see this: <begin log> SEND TIME: 341549001 SEND >>

Hi, IF i use a sip softphone or a iax softphone with asterisk, i get a lag of about 1 second. The two phones were on 2 different pc's near me. When I speak on one, i hear it on the other after about 1 second. I tried using iaxComm, Xten Xlite, etc. Same. FYI: The codec used was GSM. Using the fxo and fxs interfaces on the digium cards with POTS have no such issues. Any clue where the

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 Xlite ends up using the ulaw codec as

Hello list, I've been looking around to solve this problem both in the IRC, the wiki and Google without any luck. I have a box running * 1.07 (with zaptel 1.07 and libpri 1.07) and I'm having massive problems with random and increasing conversations cuts both in Zap and SIP channels. When one starts a call or attend any, sometimes it cuts-off at 20-25 seconds without any apparent

I'm new to Asterisk and have installed it on OS X. I am trying to use an X-Lite softphone to connect through Asterisk to FreeWorld Dialup and am not having any luck. I have looked at several settings pages at voip-info.com but have had no luck. I am on a school network which I assume has some security but the particulars are unknown as they are not very helpful. It is necessary to

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

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

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 local Asterisk server, AND be

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 these users have migrated to Skype ? much

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