search for: zoa

Zoa, I could see that you could do T38 passthrough with asterisk ans Zoiper. I have some problems to do it, can you help me? Cordialement, P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed.....

Steve Kann has developed a new jitterbuffer for IAX2, that hopefully will be integrated into Asterisk v1.1 soon, to be part of the 1.2 stable relase. Zoa and his bulgarian team is porting this buffer to SIP/RTP, but needs support in the form of funding in order to take the time to test this out and complete it in time. Please paypal your contribution to sponsor@astertest.com today. Every little dollar is worth quite a lot! I fully trust that Jo...

I'm coming to Paris Dec 19. I was wondering if there was any interest in having an Asterisk get together in Paris sometime near there. Any one out there interested? Anyone in Paris who could help organize something like that? :) Mark

...ww.asteriskguru.com/tools/idefisk_beta.php Changes since the last release include: - history panel is working - receiving messages and urls (sendtext command in asterisk) - some bugfixes (the annoying hangup bug is finally gone!). A big thanks to everybody who sent us comments and suggestions ! Zoa -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 254 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20050916/a90fc176/signature.pgp

...sed the talk that was given on wideband codecs @ astricon last week. I tried to lookup the speaker on astricon.net, but that website seems horribly broken at the moment, showing only a tmcnet video, whatever page i click on. Would somebody have the contact details for that speaker ? Greetings, Zoa

I have read that an IAX trunk requires a timing device. What wasn't clear to me was whether it is like TDM ie 1 timing device for the trunk, or if each end requires a timing device. I have a zaptel card in one server; do I have to have one in the second server in order to do an IAX trunk? I set up a trunk and so far calls can be made one way, but not the other. It is probably just not

> Zoa wrote: >> On friday we finally released Attrafax under a GPL2 license. >> It comes with its own set of modems and built in transparent gatewaying. >> The solution should be quite stable as long as the line quality is ok. >> (Some tools for measuring the line quality are incl...

I am having some issues when trying to connect with perl to the asterisk manager and doing an "IAX2 show channels". If i do that on a server that is heavily loaded, i sometimes get some events instead of the channels i asked for. Any suggestions how i could fix that ? zoa.

Hello, How many licenses to buy?? : From what we understood from digium website, we must buy as many licenses as the number of maximum simultaneous calls using G729 Codec we wish to make. For example, If we want to be able to make a maximum of 10 simultaneous calls using G729 Codec, we must buy 10 licenses. Is it right? Thanks you

...mber, the linux distro you tried it on, and the size of your shoes.) Detailed instructions on how to install it on different linux flavors will follow soon. (For now, trial and error will have to do just fine). If you are running MacOSX, stay tuned, that version is coming soon too. Greetings, Zoa. Asteriskguru.

...also put a newer version of the windows and linux versions online. Let us know how you feel about it, a more mac look (brushed metal) is coming. Please send us all problems you find, (support@asteriskguru.com), all stories about how it saved or ruined your life are also appreciated. Cheers, Zoa

List users, Over the last few days we have been working with MCI's development lab to test our Asterisk setup. We were using a piece of hardware called an Abacus 5000 that is capable of creating and terminating thousands of SIP calls. Initially, we could not get past 64 simultaneous digitally recorded calls without having call quality issues including dropped calls. We identified an

...at the > same (Ralf Mueller) > 10. RE : SPAM: [Asterisk-Users] fxo/fxs cards with 8 ports > (f6hqz-m@hamwlan.net) > 11. Re: Simple question about ringing multiple phones > (extensions)? (Ronald Wiplinger) > 12. Re: 5,000 concurrent calls system rollout question (Zoa) > 13. Re: Re: [Asterisk-Users] No IN and OUT on ISDN line at the > same (Armin Schindler) > 14. RE: Simple question about ringing multiple > phones(extensions)? (Henk Dick) > 15. Trunk is not released (temp@dreamx.org) > 16. RE: Lockups since upgrade 1.2.3 - anyone...

I am curious if anyone has had problems trunking iax2 with 100+ concurrent calls. I am planning on testing this out tomorrow, however I wanted to know if anyone else has had a problem with this prior to my test so I know what to look for if anything is known and what resolutions have been found if there are any known problems. Specifically I am doing this on fbsd 6 with asterisk 1.2.4 using

Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace:

Hi, During tests with a IAX2/PSTN gateway I've been getting strange results for processor idle time and load. I (re)search(ed) this issue for a while, but I didn't get any good explainations. Can somebody help me? I have several sites that rely on a central server for connection to the PSTN. Calls to the PSTN are routed over the Internet to this PSTN gateway using IAX2 in trunk mode. To

Digium's site now lists the Dell 1850 as a potential problem server, as it uses the intel ee1000 Ethernet chipset (as do a majority of servers in the market!). To my knowledge, ALL dell servers with Gigabit interfaces now use the same chipset. Does this mean the Digium cards can't be used in Dell servers unless you disable the onboard ethernet? I don't want to disable the onboard

Hello, any one advise a good, strong and free softphone that can work with SIP or/and IAX lines and supports attended transfer ? Thanks for help. Mit freundlichen Gr??en / best regards Andr? Herrlich IT-Operator / Developer ____________________________ LetMeRepair LMR Service and Consulting GmbH Fichtestr. 1A 02625 Bautzen Tel.: + 49 - (0)3591 - 2722 - 1451 Fax: + 49 - (0)3591 - 2722 -

Yes, try http://1ezphone.com its a browser softphone. ----- Original Message ----- From: Zoa To: "Asterisk Users Mailing List - Non-Commercial Discussion" Subject: Re: [asterisk-users] SIP Softphones and Citrix ? Date: Fri, 01 Feb 2008 23:09:56 +0200 I'm working for zoiper.com and i'm willing to help out with ours when needed. Zoa d4rk f1br wrote: &g...

Hello people, Went to IRC and zoa helped a bit by saying me to Do a insmod zaptel and insmod wcusb When I did the insmod zaptel it worked fine, but when I did the wcusb it said asterisk:/usr/src/zaptel# insmod wcusb Using /lib/modules/2.4.24/misc/wcusb.o /lib/modules/2.4.24/misc/wcusb.o: unresolved symbol usb_submit_...