similar to: Callerid Error- Causing All Zap Channels Busy

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, 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 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 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.

pls kindly respond to this email thx ! _________________________________________________________________ Connect and share in new ways with Windows Live. http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008

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

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

I'm trying to find a way to get remote connections from my Linux Desktop machine to a remote Linux box which is hosting a MySQL database, via SSH port 22, and then once connection is established, log in to the database port 3306 and have either an SSH and or SCP connection established so I can securely do edits and queries. I am able to connect this way from my Windows machine to remote

So, version .16 has come out and I wouldn't mind upgrading. The thing is, I have to patch my Wine source code for a particular game. I guess I have to remove the Wine version I have now, patch the .16 and compile the source code? If there is another way, please let me know. Though, my main question is if I uninstall Wine, will it take thing installed with it; I mean, does .wine still exist

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

Nmhxc Sent from my BlackBerry? wireless handheld -----Original Message----- From: centos-request at centos.org Date: Sat, 15 Mar 2008 12:00:07 To:centos at centos.org Subject: CentOS Digest, Vol 38, Issue 15 Send CentOS mailing list submissions to centos at centos.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.centos.org/mailman/listinfo/centos or, via email,

Hi, all. When I get calls on my SoundPoints, I only see the number -- is there a way to get the alpha portion of the CID, as well? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hey all... I am trying to connect to a Epson FX-85 24-pin printer on my file/print/samba server. I am running SuSE 9.1/Samba 3.0.2a with CUPS as the print spool. When I try to connect to the printer from W2K land, I get this error: "Permission denied" I am at a loss as to where to look (sorry I dont have my smb.conf file posted. I am away from home for a couple of days). ALl else

Hi, I'm currently running asterisk 1.0.9-r1 and zaptel 1.0.9_p1-r1 (the current gentoo ~x86 versions), with the UK CallerID patches from http://www.lusyn.com/asterisk/patches.html applied. The Zap interface itself seems to work fairly well - although it's a little quiet, there is no echo. Unfortunately, there's also no CallerID. My zapata.conf is as follows: [channels]

Hi, I just upgraded by 1.0.x home server to 1.2.2. Overall the upgrade went fine, but a strange problem has cropped up with the CALLERID name value of incoming calls from the X101P card. When an incoming call is presented (via Vonage ATA), the calledid value getting double quotes up from: -- Executing NoOp("Zap/1-1", """WIRELESS CALLE" <1404xxxxxxx>") in

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! I'm testing a telephone connected to FXS port of a Sangoma A200 card. But I'm observing that callerid is not working. The configuration that I'm using in chan_dahdi.conf is the following one: - --------------------------------------------------------------------- ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit