similar to: Skype + Asterisk

Dear all, I've read some news about Sisky (http://www.yeastar.com/Products/SiSkyEE.asp), a service to interconnect Skype clients with SIP clients. Does anybody test Sisky and can tell me about his experience ??? (Sisky runs on Windows because Skype and its API are more stable on this OS). Regards, Alejandro

Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John

Hi Everyone, I'm planning my first asterisk box, and I'd like to know what SIP providers everyone likes. Voipjet? Gizmo? Somebody else? Thanks, Alex

Asterisk Users, I have 3 FXO modules with the TDM400P Digium Card. I can dial into the Asterisk rings my Sip phone, but dialing out with my SPA941 phone through the zap channel is a problem. I keep getting this message on the Asterisk CLI. What am I doing wrong? Thanks in advance. -- Executing [103 at default:1] Dial("SIP/200-006fa300", "{Zap/g0/{EXTEN:1}") in new

Asterisk Users, I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System 2.9.18-4-amd64. A TDM03B is installed on the Debian System. Every time, I try to change my voicemail pin via the Sip phone, the voicemail.conf does not get modify and I see this warning message on the Asterisk command line: [Aug 29 00:12:23] WARNING[19142]: app_voicemail.c:799 vm_change_password:

Hello all! Does anyone have a good howto to setup Asterisk and Skype. Thanks Gustavo A. Gonz?lez Dto. de Infraestructura Despegar.com, Inc. ggonzalez at despegar.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080526/831b3824/attachment.htm

Asterisk Users, I have never done a dial plan for this scenario before. Is it possible to have Sip Phones make outbound calls through the PSTN? What would the call routing/dial plan would look like? -John _________________________________________________________________ Messenger Caf? ? open for fun 24/7. Hot games, cool activities served daily. Visit now.

Hello, I recently installed chanskype on my asterisk box and it works like a dream, can phone out. But no idea how to setup the incoming calls, every time I phone my skype name it just connects and disconnect the call right away. I get the following on asterisk -rvvvvvvvvvvvvvv Verbosity was 1 and is now 14 == Sent cmd 'GET CALL 175 TYPE' to fd 18 on Skype dev 'skype1' ==

In particular, I'm comparing these two models: Yeastar NeoGate TG100 vs. Dinstar DWG2000-1G http://www.yeastar.com/products/NeoGate-TG100.asp http://www.dinstar.com/Product/Product_25.aspx?typeid=6 Wich model do you recommend me, Yeastar or Dinstar? Thanks in advance. -- Usuario Linux Registrado # 342019 --> http://linuxcounter.net/ <-- skype --> luedcortes gtalk -->

Hello, I bought paid for and used to use ChanSkype for skype connectivity. Now I tried to download the software again after the my old server got retired and someone else is squatting on their website. Can someone give me the latest ChanSkype binaries? Why do I have to rebuy the functionality I've already paid for too? Is digium going to feel for all the people who have to re-buy what they

Sometimes it happens to me that my remote SIP devices become incapable of receiving calls. This problem is easily fixed powering the hardware on and off, or reloading the application (when it is a softphone). I wonder if I can force that procedure from the SIP/Asterisk server Thanks in advance Alejandro Lengua -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, we have this problem with Trixbox 1.23 I have created an outgoing route where the 1st line has Voipjet and the 2nd an 3rd have voipcheap accounts. The problem is that at certain moments, when we call all the calls go through the voipcheap SIP accounts SIP, whose quality are not only not good enough but also consume a lot of bandwidth. The error message that returns Voipjet to Asterisk is

I am in the process of buying a TDM800 card from Yeastar ( http://www.yeastar.com/products.asp?TypeName=TDM800%20PCI%20Card&cTypeName=1 ) Any one has tested this cards? How reliable are them? I am specially interested in the FXO/FXS module. -- Gustavo Felisberto (HumpBack) Web: http://dev.gentoo.org/~humpback Blog: http://blog.felisberto.net/ ------------ It's most certainly GNU/Linux,

Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially.

Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk

Asterisk Users, I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian "Etch" system. On the recorded voice mail messages, the volume is really low when retrieving them with my cell phone. I tried with multiple cell phones with the volume level high and still, the same problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it had no affect on

Panasonic PBX KX-TDA600 it doesn't support SIP protocol for VoIP technology it only support H323 Trunk through 4 or 16 channels gateway card and TDM technology with ISDN BRI and PRI card. Mc GRATH Ricardo

Hola. Aunque es una noticia en inglés, supongo que será del interés de algunos... http://www.r-bloggers.com/backporting-r-3-0-0-to-quantal-precise-and-lucid/ Un Saludo, _____________________________ Miguel Ángel Rodríguez Muíños Dirección Xeral de Innovación e Xestión da Saúde Pública Consellería de Sanidade Xunta de Galicia http://dxsp.sergas.es Nota: A información contida

Asterisk Users, I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service. I have two Netgear switches on my T1 router, one for VOIP and another for data. I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for all data. This morning I saw this message a few times on the Asterisk command line. The lagged cause garbled phone calls. Is my network to

Hello We are developing an application to be deployed on E1 lines (inbound and outbound calls) What is the best way to fully test the application if we do not have E1 lines in the development environment? Is there some kind of software tester to test IVR/Callcenter applications virtually?? thanks and best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: