Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon
No ideas? Sorry but I'm new to Linux and I am wondering why I can't stop or tart the deamon the way it works with 1.6.1.20. I installed Asterisk 1.8 with all defaults et. Maybe something is missing in any conf file? Make sure it starts without the daemon. Try asterisk -cvvv. Does it tart then? sean - es, without the daemon it starts and i don't see any errors. It also starts automatically after a system boot. But I am wondering why I can't stop|start|restart using /etc/init.d/asterisk start|stop|restart like in 1.6? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon
did you copied rc.redhat.asterisk script from contrib/init.d/ forlder to /etc/init.d/ folder? Regards, Faisal Hanif On 8/16/2010 2:28 PM, unsero...@aol.com wrote: No ideas? Sorry but I'm new to Linux and I am wondering why I can't stop or start the deamon the way it works with 1.6.1.20. I installed Asterisk 1.8 with all defaults set. Maybe something is missing in any conf file? Make sure it starts without the daemon. Try asterisk -cvvv. Does it start then? sean -- Yes, without the daemon it starts and i don't see any errors. It also starts automatically after a system boot. But I am wondering why I can't stop|start|restart using /etc/init.d/asterisk start|stop|restart like in 1.6? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon
I am using Debian Lenny, not RedHat. -Original Message- From: Faisal Hanif fai...@vopium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, Aug 16, 2010 11:33 am Subject: Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon did you copied rc.redhat.asterisk script from contrib/init.d/ forlder to /etc/init.d/ folder? Regards, Faisal Hanif On 8/16/2010 2:28 PM, unsero...@aol.com wrote: No ideas? Sorry but I'm new to Linux and I am wondering why I can't stop or tart the deamon the way it works with 1.6.1.20. I installed Asterisk 1.8 with all defaults et. Maybe something is missing in any conf file? Make sure it starts without the daemon. Try asterisk -cvvv. Does it tart then? sean - es, without the daemon it starts and i don't see any errors. It also starts automatically after a system boot. But I am wondering why I can't stop|start|restart using /etc/init.d/asterisk start|stop|restart like in 1.6? -- - Bandwidth and Colocation Provided by http://www.api-digital.com -- ew to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list o UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon
use the init script from debian http://svn.debian.org/viewsvn/pkg-voip/asterisk/trunk/debian/asterisk.init?revision=8502view=markup the one from the asterisk source seems to be broken, if have the same issue kristijan 2010/8/16 unsero...@aol.com: I am using Debian Lenny, not RedHat. -Original Message- From: Faisal Hanif fai...@vopium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, Aug 16, 2010 11:33 am Subject: Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon did you copied rc.redhat.asterisk script from contrib/init.d/ forlder to /etc/init.d/ folder? Regards, Faisal Hanif On 8/16/2010 2:28 PM, unsero...@aol.com wrote: No ideas? Sorry but I'm new to Linux and I am wondering why I can't stop or start the deamon the way it works with 1.6.1.20. I installed Asterisk 1.8 with all defaults set. Maybe something is missing in any conf file? Make sure it starts without the daemon. Try asterisk -cvvv. Does it start then? sean -- Yes, without the daemon it starts and i don't see any errors. It also starts automatically after a system boot. But I am wondering why I can't stop|start|restart using /etc/init.d/asterisk start|stop|restart like in 1.6? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Hardwares
Hello, Can antbody recommend devices that can be used along with my Asterisk server Paging Amplifier SIP enabled Paging Gateway VOIP SIP loudspeaker Also , please recommend video phone sets that suppot paging, intercom (autoanswer) Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on AMD
All, Thanks for confirming what I had already expected. I just wanted to verify my thoughts with a group of more experienced Asterisk users to ensure I didn't hit any hidden 'gotchas' along the way. Thanks, Lyle J. McKarns --- Networking/Linux Engineering Team n|m Nexus Management 4 Industrial Parkway Suite 101 Brunswick, Maine 04011 Tel (USA) : 1 207 319 1105 Tel (UK) : 0207 100 4968 Fax: 1 207 725 8552 Nexus Management, Inc.│ Registered Office: 4 Industrial Parkway, Suite 101, Brunswick, Maine. 04011│Company No. 19891257D, Registered in Maine│ A member of the Nexus Management Plc group of companies -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Saturday, August 14, 2010 2:08 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk on AMD On 08/14/2010 12:59 PM, Philipp von Klitzing wrote: Hi! By a mixed environment I mean some Asterisk servers running on AMD and some running on Intel If it was possible for that to matter, then the software would be very poorly written indeed. As another poster said, the only way that would have any effect is if you compiled binaries specifically for one family of processors and used them on the other. As far as how the software operates, by definition the processor type/family does not matter at all. Quite some time ago there was a difference in how the GSM codec was handled on AMD K6/Athlon systems, but that did not matter greatly, and it was just a tiny little optimisation setting in the Makefile so gain a little more speed. But it did not produce different output nor accept different input; it wouldn't have mattered if an Intel-based system was talking to an AMD-based system, because the data *outside* the system was the same. That was my point. There are many CPU family-specific optimizations that can be used for various parts of Asterisk, but in the end they don't affect how Asterisk operates, only the speed at which it does so. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] colored CLI with reattach
Using Asterisk 1.4.26.2 I can get a nice colored CLI if I run asterisk -c But I cannot achieve this when I reattach to an existing instance (as i want to do) with asterisk -r. Is there a way to reattach and have color? Thanks -- - Eric Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 4 Port FXO interface
Hi Eric, Wondering if this is something you would like to Try. V114 from Positron Telecom, which supports 4 FXO ports and 1 FXS port. It has asterisk on the card, which would mean you do not need a PC and can install this card as a PCI card on an existing system/server. They also offer an appliance option G124. Check out this website. www.positrontelecom.com Cheers Krishna On Fri, Aug 13, 2010 at 11:43 AM, Eric Merkel (Mail Lists) ejmerkel.li...@gmail.com wrote: I am looking to build a small PBX for an office that has 3 incoming analog lines and less than 10 extensions. For the Asterisk server I am going to use a small form factor PC with no-PCI slots so the FXO interface needs to be either FXO-SIP or USB. Can anyone make suggestions? I am looking at an AudioCodes MP114 FXO or possibly two Sangoma U100's but don't have experience with either. = Eric Merkel ejmerkel.li...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colored CLI with reattach
Try asterisk -rc. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Smith Sent: Monday, August 16, 2010 7:43 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] colored CLI with reattach Using Asterisk 1.4.26.2 I can get a nice colored CLI if I run asterisk -c But I cannot achieve this when I reattach to an existing instance (as i want to do) with asterisk -r. Is there a way to reattach and have color? Thanks -- - Eric Smith -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colored CLI with reattach
On Mon, Aug 16, 2010 at 08:12:28AM -0500, Danny Nicholas wrote: Try asterisk -rc. This is pointless. -c has no effect when you open a remote console. Also note that the colors are only set in the main Asterisk process and not in the remote console. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] parkcall: How to remove announcement.
Hello all, I want to park calls using the callpark application, but I don't want to hear the saydigit when the called is parked. To resolve this issue I use the following instruction in the dialplan: exten = _8XX,1,ParkAndAnnounce(|1000|local/1...@default|) Because local/1...@default is not defined to a peer I get a lot of warnings. :( Is there a better way to resolve this issue?? Thanks in advance. Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parkcall: How to remove announcement.
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alexandre Rodrigues Subject: [asterisk-users] parkcall: How to remove announcement. Hello all, I want to park calls using the callpark application, but I don't want to hear the saydigit when the called is parked. To resolve this issue I use the following instruction in the dialplan: exten = _8XX,1,ParkAndAnnounce(|1000|local/1...@default|) Because local/1...@default is not defined to a peer I get a lot of warnings. :( Is there a better way to resolve this issue?? Thanks in advance. Alex Why not exten = _8XX,1,Park() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parkcall: How to remove announcement.
Hi Danny, Thanks for your replay. When the call is parked using parkcall I get the following sequence of messages from asterisk console: Executing [...@internal:1] Park(SIP/test_peer-0004, ) in new stack == Parked SIP/test_peer-0004 on 7...@parkedcalls. Will timeout back to extension [internal] s, 1 in 1000 seconds -- Added extension '701' priority 1 to parkedcalls -- SIP/test_peer-0004 Playing 'digits/7' (language 'en') -- SIP/test_peer-0004 Playing 'digits/0' (language 'en') -- SIP/test_peer-0004 Playing 'digits/1' (language 'en') -- Started music on hold, class 'default', on SIP/test_peer-0004 == Spawn extension (internal, s, 1) exited non-zero on 'Parked/SIP/test_peer-0004ZOMBIE' How can I remove the Playing digits from parkcall application? The only way I found, as I said before, is to used application ParkAndAnnounce and send the announce to a dummy peer. Using this method I get a lot of warnings. Is there a better method??? Thanks in advance, Alex 2010/8/16 Danny Nicholas da...@debsinc.com *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Alexandre Rodrigues *Subject:* [asterisk-users] parkcall: How to remove announcement. Hello all, I want to park calls using the callpark application, but I don't want to hear the saydigit when the called is parked. To resolve this issue I use the following instruction in the dialplan: exten = _8XX,1,ParkAndAnnounce(|1000|local/1...@default|) Because local/1...@default is not defined to a peer I get a lot of warnings. :( Is there a better way to resolve this issue?? Thanks in advance. Alex Why not exten = _8XX,1,Park() -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor asterisk
Might be worth your time to check out: http://www.humbuglabs.org/ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Zulu Sent: Saturday, August 07, 2010 3:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Monitor asterisk Hey guys, I have my asterisk box running without a gui. I now need to monitor usage, calls, traffic of voice calls on this asterisk server. I cannot now install a gui because the configs will be wiped out, how can i go about monitoring all the above? -- Richard Zulu Managing Director Time Information Company P.O Box 31842 Clock Tower Kampala, Uganda www.time.co.ughttp://www.time.co.ug Mobile :+256752624006 Skype: zulu.richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parkcall: How to remove announcement.
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alexandre Rodrigues Subject: Re: [asterisk-users] parkcall: How to remove announcement. snip You could try exten = _8XX,1,ParkAndAnnounce(|1000|console/dsp|) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] colored CLI with reattach
On Monday 16 August 2010 07:42:36 Eric Smith wrote: Using Asterisk 1.4.26.2 I can get a nice colored CLI if I run asterisk -c But I cannot achieve this when I reattach to an existing instance (as i want to do) with asterisk -r. Is there a way to reattach and have color? Yes, but you'll need to upgrade to the latest 1.4 release. This also only works if you do not explicitly disable colors (-n) in the main daemon. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.8 beta3 - Unable to stop/start/restart deamon
On Mon, Aug 16, 2010 at 6:07 AM, Kristijan Vrban vrban.l...@googlemail.com wrote: the one from the asterisk source seems to be broken, if have the same issue If it is broken, open a new issue at https://issues.asterisk.org -- Paul Belanger | dCAP Polybeacon | Consultant Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) blog.polybeacon.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom 331 freezes connecting to FreePBX
We deployed a single phone handset (Polycom 331) at a remote site. We have a IPSEC VPN running between the firewall at the remote site and the firewall at the site where our Asterisk/FreePBX box lives. We have used a similar configuration for this site before and it worked fine. We gave the phone a static IP address and pointed it to the configuration server on the remote end that has the CFG files for it. The phone starts up, downloads SIP and the new application and otherwise seems to be booting normally. Then it gets to the LAN Properties screen that shows the phone's IP address, MAC address and firmware version and then...nothing. It just sits there frozen. I assume it's trying to register with the Asterisk server but for some reason that seems to be failing. I've swapped in a different, brand new, Polycom 331 on that spot and it does the exact same thing. From my laptop I can ping the Asterisk server across the VPN just fine. All of the network connectivity looks good, as far as I can tell. Anybody have a hint for what we should be looking at? I don't see any obviously blocked ports and the VPN should take care of that anyhow. I've looked in Polycom's KB but it didn't seem to offer any explanation for what it means when the phone freezes on the LAN properties screen. Any suggestions welcomed. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower 1155 Fort Street Mall Honolulu, Hawaii 96813 Mobile: 808-782-6306 Fax: 808-533-3677 www.rolandschorr.com http://www.rolandschorr.com/ b...@rolandschorr.com mailto:b...@rolandschorr.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932
After chasing this some more, I decided to do the following: 1. Change the pickup code on the phone to *8# 2. Add an extension as follows: exten = _*8XXX,1,Pickup($EXTEN:2}) This worked. When I first tried it, I included a context but that didn't work for me (could be my dialplan context includes). Cassius -Original Message- From: Cassius Smith cass...@cassius.org Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932 Date: Sat, 14 Aug 2010 23:02:06 -0500 Yes, all set to same pickup group. Here is sip.conf setup (all ext's are similarly configured): [600] type=friend mailbox=...@default context=users pickupgroup=1 host=dynamic secret=*** -Original Message- From: Ron nha...@gmail.com Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] BLF/Call Pickup using SPA942, SPA962, SPA932 Date: Sun, 15 Aug 2010 07:29:11 +0800 hi, just taking a wild guess here, are the extensions set to be in the same pickupgroup? regards ron On 8/15/10 7:01 AM, Cassius Smith wrote: Hi all, There are a lot of posts around the web about my question; unfortunately I have not been able to get any of the solutions to work. I'm using Asterisk 1.6.2.8 under CentOS 5.5. I'm trying to get call pickup working for the secretaries that monitor their bosses' phones. The BLF and the speed dial works great on the Linksys phones. Call pickup is the problem. My features.conf has *8 as the pickupexten in features.conf. On the SPA's the extended function is: fnc=blf+sd+cp;sub=...@$proxy;ext=...@$proxy the SPA932 Call Pickup Code: field is set to *8. I ring the extension; the lamp flashes on the shared line on the SPA, just like it should. When I press the flashing lamp, the CLI gives me: Notice [1328] Nothing to pick up for baf8bc-e23bc...@192.168.1.39 note: (this is the ip address of the SPA-942 in this case) then Got SIP response 603 Decline back from 192.168.1.47 note: (this is the ringing extension, in this case a Polycom 330). I have tried different pickup codes, and some web pages say to add a # at the end of the call pickup code. When I do that, the CLI says Notice [1328] Call from '602' to extension '**600' rejected because extension not found So - how to resolve this? Do I need dialplan code to handle this? I get the clue from nothing to pickup for blah blah that I'm close but may be missing something simple. Thanks all Cassius -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parkcall: How to remove announcement.
Hi! How can I remove the Playing digits from parkcall application? In general you can address problems like this by creating your own set of sounds files where the obstructing files are either simply missing or replaced by silence. Use Set(LANGUAGE) right before the action (here: parking the call) and create your own imaginary language strucutre below /var/lib/asterisk/sounds/. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX
On Mon, Aug 16, 2010 at 4:21 PM, Ben Schorr b...@rolandschorr.com wrote: We gave the phone a static IP address and pointed it to the configuration server on the remote end that has the CFG files for it. The phone starts up, downloads SIP and the “new application” and otherwise seems to be booting normally. Then it gets to the “LAN Properties” screen that shows the phone’s IP address, MAC address and firmware version and then…nothing. It just sits there frozen. I have a suggestion... Put back the 'old application', and determine whether the 'new application' broke your phone boot. Since you don't mention changing anything else, survey says it's probably the last thing you changed that broke things. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parkcall: How to remove announcement.
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von Klitzing Subject: Re: [asterisk-users] parkcall: How to remove announcement. Hi! How can I remove the Playing digits from parkcall application? In general you can address problems like this by creating your own set of sounds files where the obstructing files are either simply missing or replaced by silence. Use Set(LANGUAGE) right before the action (here: parking the call) and create your own imaginary language strucutre below /var/lib/asterisk/sounds/. Philipp Not a bad suggestion Phillipp, but you lose points for suggesting missing files as OP wanted a way to reduce/eliminate warning messages. But to elaborate on this, OP could set up the imaginary language as any two letter code that asterisk recognizes and just copy 0.gsm thru 9.gsm from /var/lib/asterisk/sounds/digits/en to /var/lib/asterisk/sounds/digits/xx where xx is the imaginary language (fr - French, gr - german, es - Spanish for starters). Of course you would want to overlay these 10 files with a silence file. Then in the dialplan Exten = _8XX,1,Set(CHANNEL(language)=xx) exten = _8XX,n,ParkAndAnnounce(|1000|local/1...@default|) Exten = _8XX,n,Set(CHANNEL(language)=es) - without this, any further sounds in the call would be in xx language. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Subject: Re: [asterisk-users] Polycom 331 freezes connecting to FreePBX snip I have a suggestion... Put back the 'old application', and determine whether the 'new application' broke your phone boot. Since you don't mention changing anything else, survey says it's probably the last thing you changed that broke things. Also quite possible that permissions are changed on the new application; the connection is quite picky about those things. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] parkcall: How to remove announcement.
Thanks very much for your help! :) 2010/8/16 Danny Nicholas da...@debsinc.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp von Klitzing Subject: Re: [asterisk-users] parkcall: How to remove announcement. Hi! How can I remove the Playing digits from parkcall application? In general you can address problems like this by creating your own set of sounds files where the obstructing files are either simply missing or replaced by silence. Use Set(LANGUAGE) right before the action (here: parking the call) and create your own imaginary language strucutre below /var/lib/asterisk/sounds/. Philipp Not a bad suggestion Phillipp, but you lose points for suggesting missing files as OP wanted a way to reduce/eliminate warning messages. But to elaborate on this, OP could set up the imaginary language as any two letter code that asterisk recognizes and just copy 0.gsm thru 9.gsm from /var/lib/asterisk/sounds/digits/en to /var/lib/asterisk/sounds/digits/xx where xx is the imaginary language (fr - French, gr - german, es - Spanish for starters). Of course you would want to overlay these 10 files with a silence file. Then in the dialplan Exten = _8XX,1,Set(CHANNEL(language)=xx) exten = _8XX,n,ParkAndAnnounce(|1000|local/1...@default|) Exten = _8XX,n,Set(CHANNEL(language)=es) - without this, any further sounds in the call would be in xx language. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
