similar to: Outcall 1.40 released

Bicom Systems releases outCALL, an Asterisk open source Outlook integration LONDON, UK (11th April 2007) - Bicom Systems announced today it has released outCALL, an open source desktop application allowing integration Microsoft Outlook. OutCALL allows users an easy way for placing and receiving phone calls integrated with users Outlook contacts. "The open source PBX market needed

Hello, Don't know if this is related but I just got a segmentation fault today while trying to register my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from

Hi, all-- What do you make of this? Here's my call log--looks like there are a lot of calls going in and out of the server that are not real incoming or outgoing calls. Does anybody have any clue what is happening? 2006-11-14 16:41:00 Local/8183... 8183461773 "8183461773" <8183461773> 8183461773 NO ANSWER 1 47. 2006-11-14 16:40:59 IAX2/Voice... 8183461773

Use oh323. Download the openh323 and pwlib tarballs from openh323.org Follow Jeremy's instructions in the /asterisk/channels/h323/ directory EXACTLY! good luck Regards, Sean Langley, P.Eng Firmware Engineer General Dynamics Canada (403)730-1482 sean.langley@gdcanada.com > -----Original Message----- > From: Senad Jordanovic [mailto:senad@cwcom.net] > Sent: Friday, September 12,

I assume that from your previous post that you are using iconnect Is your register line in the format: Register => 18005551212:1234@213.137.73.178/18005551212 I've had good luck using the IP address vs. the fully qualified hostname. Remember that the register line goes in the [general] section of sip.conf. Also, are you using the latest CVS release of *? -----Original Message-----

Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a

Hi everyone, Does anyone have the documentation for OutCall? http://code.google.com/p/outcall/ The link isn't working. Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150

Hi! I am new to Asterisk too, I got the similar problem and I would like to know how to get * to work behind NAT. When I have the SIP Debug turn on, I got the error 479 from FWD when * try to register with FWD, it looks like * is using the local IP (192.168.x.x) in the Contact field. I have put the nat=yes in the [FWD.Pulver.com] content, but it does not seems to make Asterisk aware the

Has anyone implemented an outcall notification when there is a voice message waiting? I would like to have the system notify me of awaiting voice messages by a telephone call rather than an email notification. I would imagine that a call could be dumped into the asterisk spool directory, but I'm not sure how I would monitor for messages waiting. Has anyone implemented such a feature for

Resent to the list since I didn't think you would mind. Kevin wrote: > Eric, > > I have been using your vm outcall script for some time and it has worked > well. Thanks for your efforts. > > I am trying to re-install and I can't seem to get a call file generated. > I have set up postfix and in the log it appears that it pipes the > message to the vmoutcall

asterisk-dev-bounces@lists.digium.com wrote: > I have a DSP based system that is working on a four port FXS system > using a 200MHz arm processor. Well.. since we are talking about this topic I owe you guys notes of my experience with SC1100 CPU used by various boards (www.soekris.com , www.pcengines.ch etc.). We made a Linux distro and compacted it into 32MB flash. Installed asterisk and

Hi list; I am looking for an application that can be used with call center, in this application we can integrate the telephony part of the call center (like CTI Client ad so on), any one can advise for a good application to be used with Asterisk Call Center? - Note: The application to be customized easy, to be able to use it with Banking, Telecom, Oil, .. etc. Regards Bilal

Sherwood, Your intentions are noble and your desire to build this, fullfills an immediate need for business. If your intention is just to build a GUI for Asterisk, read no further. If your desire is to build something more purposeful, your best bet would be to see the existing commercial GUI/HostedPBX offerings like Pbxware and Switchware from bicomsystems.com ( http://www.bicomsystems.com)

hi all we want to use Two single port Bri cards in Trixbox. Any idea which card is having good support and performance repotation especially when using two or more in Trixbox. Regards farooq --

Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark

Hello! We are located in the UK, and we are planning to replace our old pbx with an asterisk based pbx. For outgoing calls our present pbx is connected to three PSTN lines which all have the same number. Internally, the pbx caters for quite a few extensions, and each extension can make outbound phone calls. Our telecom provider (your communications) gives us monthly itemized bills that list

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Has anyone done any integration with this? All I know so far is that it appears to use some non standard form of SIP. Any pointers? - -- Kind Regards, Matt Riddell Director _______________________________________________ http://www.venturevoip.com (Great new VoIP end to end solution) http://www.venturevoip.com/news.php (Daily Asterisk News -

Hi, I have enabled ztdummy in order to have * compile it. I can modprobe ztdummy with no problems. The sole reason, i need ztdummy is to heve musiconhold and meetme working. However when I start *, it says this and does not start. ---------------------------------------------------------------------------- ---------------------- == Parsing '/etc/asterisk/zapata.conf': Found

Hi List; Where I can download Asterisk GUI and what I can have benifit from it? Regards Bilal ____________________________________________________________________________________ Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545469

Hi All, Has anyone passed the tests using Asterisk: http://www.btwholesale.com/pages/static/Products/Converged_Voice/IP_Exchange.html I presume the same rules apply for scaling and possibly have OpenSIPS/Kamailio on the front? Thanks. -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com