similar to: Problems connecting xlite phone

I just installed Xorcom Rapid and I'm trying to connect with Xlite. In my SIP Proxy I have set the Domain/Realm and SIP Proxy as the IP Address of the new install. I can ping that box. When I try to connect I get hung on the "Awaiting Proxy login information" and the log reads: ======================================================================== ? 2004 Xten Networks, Inc. All

Helo List, First I would like to apologize for my bad spelling as well as that I did not search the wiki first. I only have email access at the moment. I am having trouble setting both variables and global variables thru an extension. I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4 with an Xlite softphone. I have two xlite phones on diffent computers. One logs in as xlite1 and the other as

Hi guys, I do have some weird situation. I do have an * box, and I want to connect to that box from my Windows box by SIP via X-Ten Lite. I made configuration of that soft phone as it was suggested by lots of tutorials I found by Google. But... it doesn't work! I don't know what is wrong there, but I have unobstructed access to my asterisk box, created user in sip.conf, enabled

Hello, These are error messages I get when I try to call a number over CAPI channel. -- Executing SetCallerID("SIP/xlite1-3b80", "0") in new stack -- Executing Dial("SIP/xlite1-3b80", "CAPI/hfcpci/b17") in new stack > data = hfcpci/b17 > capi request for interface 'hfcpci' == hfcpci: Call CAPI/hfcpci/b17-1 (pres=0x00,

I'm trying to setup a demo where my Asterisk box with a TDM01B (FXO) answers an incoming call and forwards that call to a SIP softphone (X-lite.) Seems all is built/installed okay: # ztcfg -vv Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. I'm pretty new at this and the extensions.conf file is eating my

Hello guys. I am a fairly new user to Asterisk, and I'm just having a tough time. My goal is to set up a VOIP PBX. I have signed up with a BroadVoice number, and I have three systems with SIP phones. The PBX and the SIP phones are all behind a Cisco PIX running NAT. I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with little luck. The SIP phones are two X-Lites on

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

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

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

Hi, I installed asterisk server in my linux box. I configured a user 1000 using xlite and registered with asterisk server in the same linux box. I configured one more user 1001 in other box and this user also got registered with asterisk. But i am facing two issues here. 1. When a call is made from 1001 to 1000 i could see an incoming call blinking but no audio flow is observed. 2. When i made a

Hi, I have an Asterisk box behind a NAT address and also a Xlite 4 soft phone behind a different NAT network. Asterisk -> Nat -> Internet -> Nat -> Softphone. I can register my softphone to the asterisk box ok via SIP but the RTP stream from the asterisk box is addressed to the private non-routeable address of the softphone when I turn on rtp debuging. How can I configure the rtp

-- 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

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.com>

Hi, I have install asterisk and it works fine. But when I use Sjphone and I use Ethereal a Client send "Request:OPTIONS sip:obelix.foo" and Server answer "Status: 404 Not found". But i can talk with two client and asterisk. When I use Xlite i don't have this request it's clean. I don't understand??????????????

I could use a few pointers to get this working please? I have asterisk installed on my linux server. It is setup to register with sipgate and works for incoming calls. I have xlite installed on my windows pc and this connects fine with the asterisk server and can get the incoming calls fine. Now I want to be able to make outboun calls from xlite via sipgate. I also need to be able to dial