similar to: Voicemail Password Issue

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

Today I discovered that voicemail attachments are not working on our latest asterisk server (version 1.4.24.1). I have two other asterisk servers that I maintain but I didn?t do the configuration on these so this is my first time that I have done the voicemail.conf. I get an email but there is no attachment. Maybe there is something else I need to configure that I don?t know about? Here is my

Guys, I need you help on this one. Don't know when this broke but we have a custom gui that runs on top of Asterisk running a real-time static for configurations. Nothing has changed with the database other than upgrades of Asterisk 10. Customer complained that there password was not changing when they called into voicemail and changed it. Database is running standard ast_config with the

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!

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

Hi, I have some issue with voice mail with ODBC on asterisk 11.7 box. I may not understand database functionality on asterisk fully. The most suspected area is func_odbc. I already googled but not luck. Your guide is warmly welcomed *Error messages when I make call and leave message.* -- <SIP/1ffa9-00000007> Playing 'auth-thankyou.g722' (language 'en') [2014-05-28

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

Don't be afraid about the info that I'm going to post in this mail, but I want you to give as much info as possible. Also I want to show you what I've tried. What do I want When a voicemail-message is left via the Voicemail()-application, I want the .wav-file send to my mail-address as an attachment. My mail-setup I'm not using sendmail as MTA. I have msmtp as MTA and mutt as

I have a Asterisk server setup and I am able to connect to the server using a soft client 'x-lite' and call and leave a message on my second extension 102. I have setup a Vitelity account and add what I believe to be the correct information to my sip.conf and extension.conf. I would like to setup incoming and outgoing calls with voicemail support. I've searched all over but many of the

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.

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

Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-) My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL). Calls come in and are

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,

I've set voicemail almost successfully, only a minor detail remains :-) I can't get the dates in my local language (spanish). In sip.conf, zapata.conf and voicemail.conf, I've set: language=es and my locale is "es" also. However, the days and months names still appear in english in the emails!!! Thursday 11 de May de 2006, 18:49:34. instead of Martes 11 de mayo de

The advanced voice mail options in RC1 is not providing any functionality except "4 - place outgoing call" and "* - return to mail menu". Below is a sample line from voice mail.conf: 210 => 234,test1,test@dotality.com,,|attach=no|cid=yes|review=yes The is a rew install from the tgz file. No patched have been made to the source code. I did try to apply the advanced10

In Asterisk 1.2.13 in app/app_voicemail.c, line 4700 ext_pass_cmd is checked to decide whether to use vm_change_password or vm_change_password_shell to change a user's password for his voicemail account. I wonder, what is the difference between vm_change_password and vm_change_password_shell - what is that shell? The only reference I found on the Internet was the following bug report: