similar to: QUESTION: RINGING CONTINUES DURING CALL

Displaying 20 results from an estimated 10000 matches similar to: "QUESTION: RINGING CONTINUES DURING CALL"

2007 May 18
1
xten will not send tones to * and i from sip phone
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
2004 Jul 27
2
g729 + GSM + g723
Folks! We have purchased G729 and have been testing the codec on mUltiple Gateways. Here is what we have found. Here is the config I have used: ------------------------------- Asterisk Server On Dual Pentium Xeons with 6GB of RAM, running on Fedora Core 2 User1 is in USA on Broadband Cable User2 is in India on 64Kbps ISDN Line User1 using SIPURA SPA 2000 user2 using Xten professsional(X-pro)
2005 Jun 27
1
Polycom & VPN trouble
Hi All, I am a remote office that is connected to my office via openvpn on UDP. Voip has always worked well (after discovering g729). Initially I used a softphone, then an analog set on a sipura 2000, then a polycom IP500 (I still LOVE this phone). At that point, I started noticing that the polycom doesn't ring a lot of the time. Since I was desperate for a phone, I didn't upgrade
2005 Jun 23
1
*77 does not work ..
I have a SPA-2001 and I didn't realize I could use calling features on an analog handset. Does that mean you can dial *77 and use a VOIP feature? (like forward or hold)? Mike ________________________________ From: Jorge Carrasquillo [mailto:jorge.carrasquillo@gmail.com] Sent: Thursday, June 23, 2005 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:
2004 Apr 14
2
No ringing sound on GS phones
I've already installed * 0.9.0. All calls from my GS phone has no ringing sound but the other end rings! I also checked this with my CVS archive. The problem exists in CVSs from (Mar 6) up to now. and I have no problem with my Xten softphone when calling with it. anybody can help? -------------- next part -------------- An HTML attachment was scrubbed... URL:
2003 Apr 22
0
Xten - Free windows SIP client
Same here Michael and the PocketPC version seems unaudible with any codec; early days trying that though. Simon -----Original Message----- >From: "Michael Van Donselaar"<mvand@neb.rr.com> >Sent: 22/04/03 04:10:24 >To: "asterisk-users@lists.digium.com"<asterisk-users@lists.digium.com> >Subject: Re: [Asterisk-Users] Xten - Free windows SIP
2003 May 17
1
XTEN Lite TROUBLE
Dear Guys, I?ve test Xten Lite softphone to connect to my Asterisk Box but it registers all the three lines at the same time and if I try to dial an extension it tries to reach 3 Ext. at the same time, can somebody haved this trouble? and how can I fix it. Also, I ?ll like to have the Xten LITE or PRO Softphone (Lite is free and PRO about $50.00 USD) it can hanle 3 lines (lite) and 6 lines
2005 Sep 30
2
Asterisk and RTP streams
Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could. A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer will get one-way audio. I've read that this is commonly caused by the outgoing RTP port not
2006 Nov 12
2
same extension on softphones and hardphones
Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the
2005 Jan 07
0
Re: [Serusers] softphones
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
2007 Nov 23
1
AMI Newstate Ringing events -- Inconsistent caller id ?
Hello list, I'm observing what I believe to be inconsistent behaviour regarding "Newstate" AMI events for the "Ringing" state. As such I come to you asking for experience or advice: am I wrong or should I file a bug ? I present you a short introduction which I feel is relevant; however, if you want to go straight to my technical question, please scroll
2005 Mar 22
0
help with registration
I have a SIP account that I can successfully register with XTEN and a Sipura-2000. I have yet to be able to get it to authorize with *. My XTEN looks like: Username: 001234 Password: xxxx Authorization Username: 001234 Domain: domain.net Register with domain:
2004 May 20
2
Softphone lag
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
2005 May 25
1
Asterisk SIP cannot restrict call from softphone before registration
Hi all, I have problem with my Asterisk. I'm using the softphone Xten-Lite. I've removed the SIP client information in sip.conf. The softphone can't register to Asterisk, but it can make outgoing calls. I've tried to add back the SIP client information into the sip.conf, but make a wrong password in the softphone. The registration and outgoing calls are failed as expected.
2006 Feb 08
1
SPA-3000 VOIP-PSTN gateway - long delay between answering and ringing
Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000
2004 Nov 08
0
WINE and Sound - XTEN X-Pro Softphone
I've tried every softphone available for linux and I've found that Xten X-pro for Windows seems to load fine under the WINE API, the problem is - it wont work the sound... how do i get the sound to be emulated so I can place and receive calls using this software? the program's options seem to list my audio devices. keep in mind i have to use a microphone and sound output. is there a
2003 Aug 14
1
Re: The Almighty X-Lite DTMF Problem (patch tested)
Hi! I decided to apply Chris's patch for the rtp problem, it is working just fine now. Thanks Chris!. I think that Mark should submit it to the CVS. Ildefonso. icamarg@unet.edu.ve >Pete, > >Try this patch below... I noticed that eStara's softphone has the same >problem as xten's softphone when it comes to DTMF. Seems as though = >Asterisk >is not looking for
2007 Jan 17
4
windows mobile 5 softphone for square screen devices
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?
2003 May 11
3
Sound Quality
Hi All, I've just setup a test Asterisk system that allows incoming/outgoing calls via an ISDN card (l4i) and incoming/outgoing calls via SIP (iconnecthere). I have two SIP Softphones (Xten X-Lite) for making and receiving calls. When receiving an incoming call via the ISDN interface the sound quality is fine for the Softphone user (i can hear the caller perfectly), but the person
2006 Feb 09
1
SPA-3000 VOIP-PSTN gateway - longdelaybetweenanswering and ringing
You can even set it to zero. Mine works well when in zero. The line pick up immediately :> -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Stenton Sent: Thursday, February 09, 2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000 VOIP-PSTN