search for: senad

...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----- From: Senad Jordanovic [mailto:senad@cwcom.net] Sent: Thursday, September 11, 2003 1:52 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 I am really desperate to have any help on this problem below as it prevents us from making any further...

...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, 2003 8:18 AM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] h323 v oh323 > > > I have been reading archive post in regards to h323 support, > and I am not > clear on this: > > 1. &...

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

...r session, start asterisk and let it sit there hanging for a few minutes. Stop Ctrl-C in the terminal session running tcpdump and send me the file "foo" that was created by tcpdump. You might want to gzip it if it's large and send it to me off list. -----Original Message----- From: Senad Jordanovic [mailto:senad@cwcom.net] Sent: Thursday, September 11, 2003 2:48 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Segmentation fault due to SIP registration NUMBER 2 Tim, thanks for your answer. I tried, all of the options you suggested, and still the same... * ha...

..., Line 301 (ast_load_resource): chan_zap.so: load_module failed, returning -1 WARNING[16384]: File loader.c, Line 396 (load_modules): Loading module chan_zap.so failed! ---------------------------------------------------------------------------- -------------------------------------------------- Senad -------------- next part -------------- A non-text attachment was scrubbed... Name: winmail.dat Type: application/ms-tnef Size: 1844 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030925/2dcfb920/winmail.bin

Hi all, Anyone has experience using * through 128 kbs (or bigger) satelite link? In particular I am interested to hear how many calls could be put through 128Kbs satelite link simultaneously? Ta SJ

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

Hi I have been playing with trying to get voicemail.conf from DB, I am using cvs-head, but when I start asterisk, it dies a horrible death, because it cant load any voicemil setting. I looked at my mysql logs to see what query was being sent, and I get SELECT category, var_name, var_val, cat_metric FROM voicemail_users WHERE filename='voicemail.conf' and commented=0 ORDER BY

I have recently noticed that druid @ http://www.voiceroute.org has created an open source edition of their platform. I downloaded it today and installed it on a play system where I have about 20 ip phones ranging from cisco, polycom and aastra phones. I didn't even have to configure them as the system automatically did it for me. I have been using trixbox/freepbx combination for over that last

-----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 -

...E0F5F3@192.168.1.104 for > seqno 43 (Response) > WARNING[81926]: File chan_sip.c, Line 2002 (__transmit_response): Unable to > determine sequence number from '' > ------------------------------------------------- > > > > Has anyone had the same problem? > > Senad > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for...

Hi, Just received recently released Grandstream handytone 286 ATA for testing. I can call ATA from any other extensions and conversations seems to be of quite good quality. However placing calls from ATA is not possible at all to any extensions. After dialing there no indications of any kind from ATA at all. It just "hangs in there". ATA is behind NAT, registers to an * with public IP

Hello all, I am sure that this is possible - for helpdesk envioroments i.e. when you hear an announcment "Your call may be recorded for quality purposes" can asterisk record all calls onto disk or similar - hopefully as MP3? Thanks Nick

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

...ould not be loaded if Contact's info contains some escaping characters) - Settings and other dialogs cannot be opened twice - Settings and other dialogs can now be accessed from the taskbar - Added all DLL dependencies into the setup Available at: http://outcall.sourceforge.net/ Regards, Senad Jordanovic www.bicomsystems.com senad at bicomsystems.com +1 (212) 400 7921 +44 (20) 7043 3488 Regards, Senad Jordanovic www.bicomsystems.com senad at bicomsystems.com +1 (212) 400 7921 +44 (20) 7043 3488

Hi all I think this could be en old question. I would like to do a realtime billing prepaid system, mainly using asterisk. I have found few things; I can not get CDR function into agi because asterisk set them once the call is absolutely finish (at least main values for the main porpouse, billsec,duration, etc..) There is a patch that allow you to use CDR

iaxclient.sourceforge.net is the home of Steve Kann's crossplatform port of the iax library. iaxComm is a client written in c++ using wxWindows. There is a Win32 binary on the site. I think that it should be compilable on Linux and MacOSX, but can't test it. Feedback is welcome.

Hi, all This is certainly very good news! http://www.neowin.net/comments.php?id=21119&category=main

Hi, Do the callers in USA dialling from USA Telco lines always have to prefix the CITY/AREA code with "1" in order To successfully make a call to other USA destinations? ---- I have not been to USA (yet) :) Ta SJ