search for: jordanov

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 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

...ster => 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 progress...

...23 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. > Is h3...

In United Kingdom, we have time based dialling pricing from most of Telco's based on time the call is placed! It is called PEAK (08.00- 18.00 Mon-Fri), OFF PEAK(18.00-08.00 Mon-Fri) and WEEKEND (all other times! Could someone from any of other countries let me know if time based charging exists in your country? Also, what numbers (up to 999) are commonly used for emergency, police or other

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

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

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

...ion, 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... * hangs. It i...

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

Hello! How do I handle calls when they reach a queue that has no members? Currently, the callers are thrown out, because of the autofallthrough. The message is app_queue.c:2094 queue_exec: Unable to join queue 'queue-name' == Auto fallthrough, channel 'Zap/3-1' status is 'UNKNOWN' It seems that Queue() won't continue at a specific priority - like n+101 - if

...ot 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

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 all, Let me be the first to wish everyone, especially the Digium team, an awesome year in 2004.. Later..

hi all anyone that've heard of any working IAX hardphones yet? roy

Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ

Dunno if anyone else has seen this yet: http://www.scmagazine.com/us/news/article/563800/vulnerabilities+put+asterisk+telephone+systems+risk/ -- Aaron Daniel Computer Systems Technician Sam Houston State University amdtech@shsu.edu (936) 294-4198

-----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 -

I have a cisco 7910 phone, I'm trying to get it to connect to asterisk, But it seems like it needs either a SEPDefault.cnf file or a SEPMACADDR.cnf file to Continue, I created empty ones but it's still sitting there saying "opening" Does anyone have examples of the SEPDefault.cnf file? Kevin,

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