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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

Has anybody ever integrated Skype with Asterisk? If you have, which software would you recommend to accomplish such a task? ChanSkype? And how reliable are the calls? Did the DTMF tones work? Thanks in advance. _________________________________________________________________ Discover sweet stuff waiting for you at the Messenger Cafe.? Claim your treat today!

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 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.

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:

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 looking for Sip Providers for my Asterisk 1.2.13, running Debian Etch system with McLeodUSA's T1 service. Has anybody ever used Callcentric for their Sip Provider? Any service issues with Callcentric? Best Regards, John _________________________________________________________________ Messenger Caf? ? open for fun 24/7. Hot games, cool activities served daily.

Asterisk Users, I am running a Debian "Etch" system with Asterisk 1.4.11 with a TDM03B card. Once in awhile, I get this error on the Asterisk, which causes my channels to be busy/congested, leaving me with just one channel to recieve and make calls: NOTICE[31454]: chan_zap.c:6367 ss_thread: Got event 17 (Polarity Reversal)... WARNING[31454]: chan_zap.c:6499 ss_thread: CallerID

Does anybody know a "type 102 milliwatt test number" that I can dial in the USA? I need this in order to configure my "rxgain" and "txgain". My analog line provider, AT&T Repair Center was so confuse, when called them. Thanks in advance. -John _________________________________________________________________ Stay up to date on your PC, the Web, and your

Asterisk Users, I am running Asterisk-1.4.11 on a Debian "Etch" system. On an occasion, when customer calls into my Asterisk Box, I get this error messagefrom Asterisk "CallerID returned with error on channel Zap/3-1" , causing all my zap channels to be busy. So, I cannot make any calls in, nor out. I am located in the United States. Is there any other suggestions,

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

Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _________________________________________________________________ Gear up for Halo? 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1

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

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