search for: baldun

Hello, Most traditional PBX-es have the ability to use external audio source (e.g. radio tuner) for music on hold. This is also useful because you can let your users listen to radio by dialing some extension. I wanted to achieve the same on asterisk, and chan_alsa seemed the logical choice. I installed ALSA drivers, connected the radio to line-in and added the folowing to extensions.conf: exten

Hello, for a long time I've been having problems with analog modems and faxes communicationg over ISDN BRI lines. Now I began to suspect that this is due to * being out of sync with the PSTN. I have a quadBRI card with first two ports connected to PSTN, and defined as follows: span=1,1,0,ccs,ami span=2,2,0,ccs,ami This should mean that spans 1 and 2 are used as primary and secondary

Hello All I am wondering is someone knows how to configure the * to work with an Ericsson MD-110 with SL60 signaling?? through a TLU76 card. What is the right configuration in the zaptel.conf ? I currently have it configured as span=3.0.0,ccs,hdb3,crc4 but it doesn't detect anything when I connect it to the PBX and no activity can be seen either in the logs or in the asterisk console.

...isk-Users digest..." Today's Topics: 1. How to mark a user for a conference (Jagan Mohan) 2. Re: fax e-mail spandsp (Nils Segerdahl) 3. Re: Vmail.cgi - "Hrm, can't seem to open /var/spool/asterisk/voicemail .... (Frank Kostin) 4. Re: Analogue RAS Server (Niksa Baldun) 5. Re: Analogue RAS Server (Paradise Dove) 6. Zaptel config (ismaelg) 7. Re: Zhone channel bank issues (James Freire) 8. Re: Weir long distance behaviour... (Francois Meehan) 9. RE: Generic modem question (Rich Adamson) 10. Re: Zaptel config (Tzafrir Cohen) 11. RE: asterisk one...

...l_data is used (Reported by Stas Kobzar) * ASTERISK-27622 - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) * ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) * ASTERISK-21794 - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) * ASTERISK-277...

...l_data is used (Reported by Stas Kobzar) * ASTERISK-27622 - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) * ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) * ASTERISK-21794 - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) * ASTERISK-277...

...amp for bridge (Reported by sungtae kim) * ASTERISK-27483 - Allow wrapuptime to be set for each queue member (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-28055 - app_queue: Per-member wrapup time missing from AddQueueMember application (Reported by Niksa Baldun) * ASTERISK-28292 - Changed to show all channel stats including wrong media (Reported by sungtae kim) * ASTERISK-28253 - res_pjsip_session: Adding rtcp stats result into the session (Reported by sungtae kim) For a full list of changes in this release, please see the Chang...

...l_data is used (Reported by Stas Kobzar) * ASTERISK-27622 - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) * ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) * ASTERISK-21794 - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) * ASTERISK-277...

...l_data is used (Reported by Stas Kobzar) * ASTERISK-27622 - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) * ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) * ASTERISK-21794 - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) * ASTERISK-277...

...l_data is used (Reported by Stas Kobzar) * ASTERISK-27622 - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) * ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) * ASTERISK-21794 - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) * ASTERISK-277...

...l_data is used (Reported by Stas Kobzar) * ASTERISK-27622 - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) * ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) * ASTERISK-21794 - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) * ASTERISK-277...

.../lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ------------------------------ > > Message: 12 > Date: Fri, 11 Mar 2005 13:01:05 +0000 > From: Niksa Baldun <niksa.baldun@lumiss.hr> > Subject: Re: [Asterisk-Users] Unable to create Zap channel when > dialing using a bri cellular gateway > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <42319691.4050904@lumiss.hr...

Is there a way to execute specific applications at Asterisk startup (like "startup context" or something)?

Hi list! I'm trying to configure the Kirk IP 600 wireless (DECT) system. The wireless phones are regular DECT phones, the Kirk IP 600 is doing the voip part by registering/gatewaying the phones to a callmanager server. The phones do not work and I think the problem is that they do not register at asterisk but I'm not sure because I don't understand a thing of the sccp/skinny

...l_data is used (Reported by Stas Kobzar) * ASTERISK-27622 - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) * ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) * ASTERISK-21794 - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) * ASTERISK-277...

...l_data is used (Reported by Stas Kobzar) * ASTERISK-27622 - empty voicemail.conf required for ARA (realtime) voicemail to leave message (Reported by Jim Van Meggelen) * ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) * ASTERISK-21794 - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) * ASTERISK-277...

...amp for bridge (Reported by sungtae kim) * ASTERISK-27483 - Allow wrapuptime to be set for each queue member (Reported by Rodrigo Ramirez Norambuena) * ASTERISK-28055 - app_queue: Per-member wrapup time missing from AddQueueMember application (Reported by Niksa Baldun) * ASTERISK-28292 - Changed to show all channel stats including wrong media (Reported by sungtae kim) * ASTERISK-28253 - res_pjsip_session: Adding rtcp stats result into the session (Reported by sungtae kim) * ASTERISK-28246 - Support skipping on the g726 format...

...(Reported by Jim Van Meggelen) * ASTERISK-21794 - CLI command 'realtime update2' syntax failure when using according to usage help (Reported by Cedric BASSAGET) * ASTERISK-28349 - Pause reason not reported in QueueMember AMI event (Reported by Niksa Baldun) * ASTERISK-25429 - res_pjsip_endpoint_identifier_ip: Document support for hostnames (Reported by Joshua C. Colp) * ASTERISK-27775 - res_pjsip_notify: Multiple Event headers can be present instead of just one (Reported by AvayaXAsterisk) * ASTERISK-28682 - app_recor...

...e/ASTERISK-27483>] - Allow wrapuptime to be set for each queue member (Reported by Rodrigo Ramirez Norambuena) - [ASTERISK-28055 <https://issues.asterisk.org/jira/browse/ASTERISK-28055>] - app_queue: Per-member wrapup time missing from AddQueueMember application (Reported by Niksa Baldun) - [ASTERISK-28292 <https://issues.asterisk.org/jira/browse/ASTERISK-28292>] - Changed to show all channel stats including wrong media (Reported by sungtae kim) - [ASTERISK-28253 <https://issues.asterisk.org/jira/browse/ASTERISK-28253>] - res_pjsip_session: Adding rtcp s...

Hello, is it possible to wait indefinitely (i.e. until user hangs up) somewhere in the dialplan? I tried Wait(-1), but it doesn't work.