similar to: IAX2 wait on channel

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Hi, VoiceGlo is comercial version of Asterisk? :))) loooooooooollllllllllllllllllll Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031201/307c10e9/attachment.htm

Hi, I have a deltathree account and I can place calls but I can't receive calls. I use Grandstram sip phones. When I call my deltathree phone # the voicemail is answer :(( I need some help and solutions from the guys who allready are using deltathree. I search on Internet and I try all types of configurations... :( This is my configurations files: - sip.conf - [general] port = 5060

I have a working system with two fxo and two fxs channels. I recenlty got an IAX2 account I would like to use also. While I have gotten the IAX2 channel to "register", it remains non functional, as the incoming calls, go nowhere and the outgoing calls attempt to go out over the ZAP channel. I can see this, via the CLI, with debugs on. I strongly suspect this is a dial plan/config

i cannot see caller ID of the call originated from outside zap channel. i hv configured both zapata.conf and extensions.conf. i m right now in india i think asterisk only supports Bellcore enable caller ID. so is it the same bug of BT caller ID problem in UK? or it is the bug of my asterisk configuration? i hv enabled callerID from my TELCO. -------------- next part -------------- An HTML

What is the no authority found problem? And how can I register with * on IAX. It keeps rejecting the request telling that XXX not dynamic host. rejected any idea THX -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030520/a8a5907d/attachment.htm

Which CVS branch is the correct official Asterisk -STABLE? cvs up -r v1-0-1 asterisk zaptel libpri asterisk-sounds asterisk-addons OR cvs up -r v1-0 asterisk zaptel libpri asterisk-sounds asterisk-addons -------------- next part -------------- A non-text attachment was scrubbed... Name: eric.vcf Type: text/x-vcard Size: 146 bytes Desc: not available Url :

I've downloaded diax-0.9.6b and configured for IAX2. Calls from Diax to * are perfect. However, when calling from * to Diax, I get the following: channel.c:1097 ast_read: Dropping incompatible voice frame on IAX2[mike]/3 of format GSM since our native format has changed to ULAW In iax.conf I have: allow=all disallow=g723.1 disallow=lpc10 allow gsm Has anyone else seen this? Thanks,

Trying to use IAX2 extension as call center agent but getting this on the CLI May 24 20:34:20 WARNING[1209214400]: channel.c:1783 ast_request: No channel type registered for 'IAX2[2001@2001]' using AddQueueMember as the login mechanism and that seems to work but * will not deliver to the IAX2 extension. any ideas? Jason Kawakami -------------- next part -------------- An HTML

Hi All, I'm using DISA on my * server to avoid overseas toll charges when making calls to Western Europe from my cell phone. I have DISA working with a DID from a VoicePulse Connect account. The outgoing call to Europe is also made via Voicepulse Connect. I see that the IAX media path is bridging the inbound call to the outbound call so that the media stream entirely bypasses my server once

Hi guys, I'm running Asterisk-0.5.0 and accidentally stumbled on this problem while in the VoicemailMain2 application: If you login to it, or even if you call it w/ 's<extension>' to skip the login and press an '8' near the beginning (and possibly at any point, I'm not sure), the channel seems to lockup, even if the handset is hungup, the channel remains frozen

Everyone, I am trying to set up a hosted PBX. I want to set up call parking with in each context. Does anyone have any suggestions? Gene -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041007/1da29250/attachment.htm

I'm trying to switch from one call to another incoming call from PSTN. When I'm getting a "beep" I press flash but instead of swithing to the second call, I'm getting a dial tone. even if I press *0, I cannot connect to the second call. Anybody had this problem? Tx, Bogdan

Hi, Can U tell me the Vonage ATA 186 settings? I would like to try to have a web interface on my adapter :-)) Best regards, Chris Hariga

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I've just installed an X100P, built the kernel module, and tried to use it to make an outgoing call (via a phone connected to an ATA-186). However, I just get a reorder tone and see this on the console: -- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to create channel of type

Hi everyone! I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf ***** [ip-incoming]

Greetings everyone, I'm still a newbie, so please indulge me. I have set up an asterisk server on my RH9 boxes for testing. Instead of immediately launching into the hardware aspect of it I decided to go with 2 IAX clients (gnophone) which I placed on 2 other strategic machines within my LAN. Here's my problem, on one of the IAX clients (gnophone) I am able to do the asterisk demo

Hi, si posible on SIP phones to have the dial tone after 9 like on the FXS card? I set ignorepat => 9 on my extensions.conf... Best regards, Chris HARIGA

Hi, I have Fedora Core 2 running with a T1 card. I try to put the log on db but I get the error: Aug 20 15:17:47 ERROR[262160]: cdr_addon_mysql.c:378 my_load_module: Failed to connect to mysql database asteriskcdrdb on localhost. The database exists and I try with "mysqlaccess localhost asteriskcdrdb" and I get: Access-rights for USER 'localhost', from HOST