login | register
Sat 17 of May, 2008 [07:41 UTC]

voip-info.org

Navigation
Today's Deals
Recently Visited Pages
Search with Google
Search this site with Google. Results may not include recent changes.
 
Google Ads
Last Changes
Shoutbox
  • Juan Ortega, Thu 15 of May, 2008 [10:33 UTC]: Hi everybody, I'm Juan, an ITCom student, and I need to know what basic elements I need to create a VoIP network. Can anybody helpme, please?,Thank you very much
  • gineta, Wed 14 of May, 2008 [03:58 UTC]: any here not fine the configuration of firewall juniper -screem for VOIP asterisk????
  • Anoop Prabhakaran, Tue 13 of May, 2008 [12:16 UTC]: I am developing Asterisk IVR, Whenever i make a internation call to the IVR system, the DTMF is not getting detected properly, this happens only for the first time, second call onwards system works fine. why this is happening
  • joe, Mon 12 of May, 2008 [04:27 UTC]: Is there an opensource browser based softphone, or a system like Busta where everything is not manages through their website?
  • Nick Barnes, Fri 09 of May, 2008 [11:36 UTC]: Christopher - yesterday I tried an Asterisk install on a CentOS 5.1 box with stock GUI and it all worked fine. Sorry I can't help.
  • aero, Fri 09 of May, 2008 [08:20 UTC]: can someone help me out on this, i tried to play some sound files on my asterisk box and this is the error message i got. WARNING[4429]: format_wav.c:169 check_header: Unexpected freqency 22050 May 8 11:17:39 WARNING[4433]: codec_gsm.c:194 gsmtolin_fra
  • Christopher Faust, Thu 08 of May, 2008 [14:15 UTC]: I beleive that I may have to change something in the xserver configuration. Please advise
  • Christopher Faust, Thu 08 of May, 2008 [14:14 UTC]: Everything was perfect. In the bios I have increased the memory allocated Still receive input not supported on my display.
  • Christopher Faust, Thu 08 of May, 2008 [14:13 UTC]: This would not be my main box. I am doing some testing to see if I can install zaptel and asterisk 1.4 on a full centos 5.1 box with development software Its bizzare, because before I went through the asterisk and zaptel installation everything was perfe
  • Nick Barnes, Thu 08 of May, 2008 [13:44 UTC]: Christopher - I can't see any way in which an Asterisk installation would muck your GUI, but remember that it is advised not to use a GUI on an Asterisk box anyway.
Server Stats
  • Execution time: 0.38s
  • Memory usage: 2.22MB
  • Database queries: 32
  • GZIP: Disabled
  • Server load: 0.38

SIP COMMUNICATOR

https://sip-communicator.org/

SIP Communicator is an audio/video Internet phone and instant messenger that supports some of the most popular instant messaging and telephony protocols such as SIP, Jabber, AIM/ICQ, MSN and soon others like Yahoo and IRC

SIP Communicator is completely Open Source / Free Software, and is freely available under the terms of the GNU Lesser General Public License
Created by jht2, Last modification by Emil Ivov on Tue 27 of Feb, 2007 [11:16 UTC]

Comments

Comments Filter

by Rick Neubauer on Thursday 31 of May, 2007 [23:47:47 UTC]
the reason this is happening is because you have qualify = yes for the extensions you are trying to use. This makes the asterisk box send out the "OPTIONS" to the extension. Basically it is a form of keepalive. If the softphone dioesn't know how to answer this, then asterisk doesn;t get a reply, and assumes that the extension is down. So you can only make outbound calls from it. If you look at tthe asterisk info page (Trixbox) you will see "UNREACHABLE" for the status of the extensions. Just simply turn i t off and you will be all set...

by Rick Neubauer on Thursday 31 of May, 2007 [23:47:27 UTC]
the reason this is happening is because you have qualify = yes for the extensions you are trying to use. This makes the asterisk box send out the "OPTIONS" to the extension. Basically it is a form of keepalive. If the softphone dioesn't know how to answer this, then asterisk doesn;t get a reply, and assumes that the extension is down. So you can only make outbound calls from it. If you look at tthe asterisk info page (Trixbox) you will see "UNREACHABLE" for the status of the extensions. Just simply turn i t off and you will be all set...

conference call

by John Reyes on Tuesday 17 of October, 2006 [01:58:45 UTC]
Can i use sip communicator for a conference call? ( one to many or many to many ) client?
or i need a server(?) to make a conference call?
Does S.I.P. provides interface for conference call?

:)

Re: SIP Communicator and Asterisk

by carlos on Wednesday 28 of June, 2006 [16:43:59 UTC]
Hello. I have the same problem.
I can establish calls in both directions but when SIP Communicator is conneced to asterisk, it immediately brings up a strange error dialog box message which looks like this:


SipCommunicator's SipManager didn't know how to handle the message OPTIONS in the current c ontext! (See Details)

Has anyone solved this?
Thanks

Re: SIP Communicator and Asterisk

by carlos on Wednesday 28 of June, 2006 [16:43:40 UTC]
Hello. I have the same problem.
I can establish calls in both directions but when SIP Communicator is conneced to asterisk, it immediately brings up a strange error dialog box message which looks like this:


SipCommunicator's SipManager didn't know how to handle the message OPTIONS in the current c ontext! (See Details)

Has anyone solved this?
Thanks

by Cheukh Tidiane DIENG on Friday 03 of February, 2006 [12:49:34 UTC]
beccon do you fix your problem?

Re: help me

by Ulises Estecche on Wednesday 21 of December, 2005 [06:35:21 UTC]
Hi..

I could get it to work but I'm only listening one-way voice. I mean, others can hear me, but I can't hear them. Is this have to do with asterisk?

Cheers

SIP Communicator and Asterisk

by beccon on Tuesday 04 of October, 2005 [10:23:32 UTC]
Has anyone successfully run SIP-Communicator at an Asterisk system? I can establish calls in both directions but when SIP Communicator is conneced to asterisk, it immediately brings up a strange error dialog box message which looks like this:<br><br>

<i>SipCommunicator's SipManager didn't know how to handle the message OPTIONS in the current c ontext! (See Details)<br><br>

OPTIONS sip:192.168.0.1:24075;transport=udp SIP/2.0 <br>
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK02215330 <br>
From: "asterisk" <sip:asterisk@192.168.0.1>;tag=as6faf003b <br>
To: <sip:192.168.0.1:24075;transport=udp>;tag=-1239673351 <br>
Contact: <sip:asterisk@192.168.0.1> <br>
Call-ID: 710f3ebd16d4fa970235f07a530eacdc@192.168.0.1 <br>
CSeq: 102 OPTIONS <br>
User-Agent: Asterisk PBX <br>
Date: Mon, 03 Oct 2005 10:52:05 GMT <br>
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER <br>
Content-Length: 0 </i><br><br>


Unfortunately this box keeps returning annoyingly every 20 seconds, making the use of
sip-communicator with almost impossible.<br>

Conrad

help me

by tarek007 on Sunday 17 of April, 2005 [13:47:48 UTC]
i want the package voip.router for java

Please update this page with new information, just login and click on the "Edit" or "Add Comment" button above. Get a free login here: Register Thanks! - support@voip-info.org

Page Changes | Comments

Sponsored by:
Terms of Service Privacy Policy
© 2003-2008 VOIP-Info.org LLC

Powered by bitweaver