[asterisk-users] Recording the follow-me calls
I want please to record the forwarded calls in the not working hours,so how can i record the follow-me calls to external numbers like mobile numbers?. Thank you -- _ -- 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] Recording the follow-me calls
I want please to record the forwarded calls in the not working hours,so how can i record the follow-me calls to external numbers like mobile numbers?. Thank you -- _ -- 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] Queue Setup
Hi All, I have Asterisk 1.6.2.13, I need to setup a queue of (6) agents, Ring All strategy, I need to set the maximum total time for the caller (Ringing/OR Waiting) on the queue is (2) minutes before going to a fail-over which is a Ring Group of external numbers. How the total max time is being calculated in terms of the number of agents, Ring Strategy, Agent Timeout, Retry, etc.. Can you please explain how it goes. Thank you.-- _ -- 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] Spy on Asterisk 1.2
someone has Asterisk 1.2 (upgrade is not possible), and wants to spy on specific extensions he can specify while dialing a code, could you please kindly tell us how to do this. Thanks _ Hotmail: Trusted email with Microsoft’s powerful SPAM protection. https://signup.live.com/signup.aspx?id=60969-- _ -- 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] PSTN Line Parameters Checking
I have Asterisk and Digium AEX808B What are please the commands that i can run on Asterisk to get the information about the connected lines from PSTN to see the parameters of them and as well the corresponding files in Asterisk that i can change into, to tune these parameters to be matched together. Thanks a lot. _ Keep your friends updated—even when you’re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PSTN Lines and AEX808B
I have a setup of Asterisk 1.2.28 and Digium AEX808B (8 FXOs), I connected the PSTN lines to the Digium card, everything is working fine but the issue is that when I make calls through the PSTN lines, some of calls get out successfully and the other give me a different dummy long ring back tone for about 12 seconds until it hangs up. I have taken the server with the card to another location in the same city, the same PSTN Company, but different exchange, everything worked fine, as well I tested it with a Prima Cell, and worked fine, I want to find the issue to try to sort it out. Any help will be highly appreciated. The following is zapata.conf ; ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn ;signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes busydetect=yes busycount=3 cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=no echotraining=800 rxgain=8.0 txgain=2.0 group=0 callgroup=1 pickupgroup=1 immediate=no ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 ;Include AMP configs #include zapata_additional.conf Thanks _ Windows Live: Keep your friends up to date with what you do online. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_1:092010___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Booting Error for /dev/kmem
Suddenly i found an error while booting, it says: Fuck: can't open /dev/kmem for read/write (2) So this is why, the Asterisk and Zaptel can not start. Any Suggestions Please Thanks a lot Torintino _ Windows Live: Make it easier for your friends to see what you’re up to on Facebook. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] interfacing asterisk with a legacy PBX
If I understood your question well,you can do this, if your legacy PBX has a feature of DISA, that's enable you after entering the PBX to dial your desired extension, you can setup each FXS port as an extension, then create a ring group containing group of extensions (with 1 digit for example) to ring with a hunt strategy, and if you want to call a one of 23 leagcy extensions, you can dial the Ring Group extension and then while you are on the DISA of the legacy PBX you can dial the desired extension. Hope this helps. Date: Sat, 24 Oct 2009 11:25:51 +1100 From: pdha...@optusnet.com.au To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] interfacing asterisk with a legacy PBX On 24/10/09 00:59, Lyle Giese wrote: PATRICK KANGETHE wrote: I want to interface asterisk with a legacy pbx that has around 23 extensions through my 8 fxs card, how do i work around this? Hint: I have already terminated 8 extensions from the legacy PBX, i was thinking whether i can peer the extensions from the PBX i.e like 5 extensions be peered to one extension connecting to the fxs? How can i do this? Thanks in advance, Are you planning to get rid of the legacy PBX completely? Or is Asterisk going to be a second PBX? I am going to assume you are replacing the legacy PBX. You can setup analog extensions so that you have multiple phones on each FXS channel. But they will be like a party line. If you put 6 phones on one FXS, all 6 ring at the same time, only one person can use that extension at a time. However you can add SIP phones to Asterisk and each can have their own extension instead. It just requires cat 5 cable back to a switch for each phone. Lyle Giese LCR Computer Services, Inc. Just to add my 5 cents - connecting too many phones to an FXS port can cause problems. The term is REN - ring equivalent number, and it's used to describe the maximum phones to attach to an FSX port (from memory) PaulH _ Windows Live: Keep your friends up to date with what you do online. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_1:092010___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7921
How can i please register Cisco 7921 (Skinny) wireless phone on Asterisk. Thanks _ Keep your friends updated—even when you’re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7921
Is available to setup chan_sccp on Asterisk 1.2.28 and register Cisco 7921 wireless phone on it? If yes, Can anybody please post his example. Thanks _ Keep your friends updated—even when you’re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ChanSpy
I am unsing Asterisk 1.2.28 I want please to use ChanSpy urgently my /etc/asterisk/extensions_additional.conf is as follow: [chanspy] include = chanspy-custom exten = 102**,1,Chanspy(102) exten = 102**,n,Hangup exten = 103**,1,Chanspy(103) exten = 103**,n,Hangup exten = 400**,1,Chanspy(400) exten = 400**,n,Hangup exten = 501**,1,Chanspy(501) exten = 501**,n,Hangup exten = 601**,1,Chanspy(601) exten = 601**,n,Hangup exten = 606**,1,Chanspy(606) exten = 606**,n,Hangup ; end of [chanspy] I created a Context to put my extension into it to be able to use ChanSpy. While there is a call with an extension 102 and my extension is 606 i call 102** to spy but i couldn't hear anything, all i hear is beep -- Executing ChanSpy(SIP/606-09430fd0, 102) in new stack -- Playing 'beep' (language 'en') -- Playing 'beep' (language 'en') Thanks Torintino _ Windows Live: Make it easier for your friends to see what you’re up to on Facebook. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ChanSpy
Thanks for your reply. Is ExtenSpy available in Asterisk 1.2? If yes, please how can i use it? and how can i cycle through the available channels by ChanSpy? Thanks. Torintino Date: Wed, 14 Oct 2009 18:15:29 +0300 From: rennes.n...@norby.ee To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] ChanSpy You must use extenspy if you want to spy on specific extension. Otherwise you can only cycle through available channels. Regards Rennes -Original Message- From: asterisk-users-boun...@lists.digium.com on behalf of Torintino T Sent: Wed 10/14/2009 17:46 To: asterisk-users@lists.digium.com Subject: [asterisk-users] ChanSpy I am unsing Asterisk 1.2.28 I want please to use ChanSpy urgently my /etc/asterisk/extensions_additional.conf is as follow: [chanspy] include = chanspy-custom exten = 102**,1,Chanspy(102) exten = 102**,n,Hangup exten = 103**,1,Chanspy(103) exten = 103**,n,Hangup exten = 400**,1,Chanspy(400) exten = 400**,n,Hangup exten = 501**,1,Chanspy(501) exten = 501**,n,Hangup exten = 601**,1,Chanspy(601) exten = 601**,n,Hangup exten = 606**,1,Chanspy(606) exten = 606**,n,Hangup ; end of [chanspy] I created a Context to put my extension into it to be able to use ChanSpy. While there is a call with an extension 102 and my extension is 606 i call 102** to spy but i couldn't hear anything, all i hear is beep -- Executing ChanSpy(SIP/606-09430fd0, 102) in new stack -- Playing 'beep' (language 'en') -- Playing 'beep' (language 'en') Thanks Torintino _ Windows Live: Make it easier for your friends to see what you're up to on Facebook. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_2:092009 No virus found in this incoming message. Checked by AVG - www.avg.com Version: 8.5.421 / Virus Database: 270.13.112/2391 - Release Date: 10/13/09 19:11:00 _ Keep your friends updated—even when you’re not signed in. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_5:092010___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Chanspy
How can i activate ChanSpy to spy on a dedicated extension? I see the following in /etc/asterisk/extensions_additional.conf [chanspy] include = chanspy-custom exten = 501**,1,Chanspy(801) exten = 501**,n,Hangup exten = 502**,1,Chanspy(802) exten = 502**,n,Hangup But when i try to call 501**, it doesn't give any response. Thanks. Torintino _ Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_4:092009___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Chanspy
Edited the post How can i activate ChanSpy to spy on a dedicated extension? I see the following in /etc/asterisk/extensions_additional.conf [chanspy] include = chanspy-custom exten = 501**,1,Chanspy(501) exten = 501**,n,Hangup exten = 502**,1,Chanspy(502) exten = 502**,n,Hangup But when i try to call 501**, it doesn't give any response. and is there any option for spying on a dedicated queue? Thanks. Torintino Windows Live Hotmail: Your friends can get your Facebook updates, right from Hotmail®. _ Windows Live: Friends get your Flickr, Yelp, and Digg updates when they e-mail you. http://www.microsoft.com/middleeast/windows/windowslive/see-it-in-action/social-network-basics.aspx?ocid=PID23461::T:WLMTAGL:ON:WL:en-xm:SI_SB_3:092010___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] E65 fails registration, soft phone works
Martin, Try to put qualify=yes. Torintino Date: Fri, 18 Sep 2009 22:45:05 -0700 From: lugos...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] E65 fails registration, soft phone works Martin, sounds like the hiccup my E71 had once. I think the symptoms were identical. Changing the transport type from Auto to UDP solved the problem for me. The Auto setting worked, but only sometimes. Maybe the E65 is similar... Luki 2009/9/12 martin f krafft madd...@madduck.net: Hey folks, I am trying to get an E65 to connect to asterisk, and I would really appreciate a second set of eyes. The SIP dialog completes fine, but the phone subsequently says Registration failed. I am in a network that has what seems to be a SIP-capable NAT gateway, but the asterisk is configured nat=yes anyway. Using a softphone (twinkle), I can connect just fine, SIP and RTP work. But when the E65 tries to connect, it seems to complete the SIP REGISTER dialog, but then it'll say Registration failed: ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Drag n’ drop—Get easy photo sharing with Windows Live™ Photos. http://www.microsoft.com/windows/windowslive/products/photos.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Call Disconnection
There is an environment Setup uses Asterisk 1.2 and doesn't want to upgrade. There is an issue while a call goes to any queue we create, the call is being disconnected after 20 seconds and it is hangup. The following is the configuration: - vi /etc/asterisk/queues_additional.conf [8] wrapuptime=0 timeout=30 strategy=ringall servicelevel=5 retry=4 reportholdtime=No queue-youarenext= queue-thereare= queue-callswaiting= periodic-announce-frequency=0 periodic-announce=periodic-announce music=default monitor-join=yes monitor-format= member=Local/8...@from-internal/n,0 member=Local/8...@from-internal/n,0 member=Local/8...@from-internal/n,0 member=Local/8...@from-internal/n,0 member=Local/8...@from-internal/n,0 member=Local/8...@from-internal/n,0 member=Local/8...@from-internal/n,0 member=Local/8...@from-internal/n,0 maxlen=0 leavewhenempty=no joinempty=Yes context= announce-holdtime=no announce-frequency=30 - The Verbosity logs: after ringing on the available extensions for 20 seconds, it goes to macro hangupcall -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(Local/8...@from-internal-80d5,2, SIP/808|30|Ttr) in new stack -- Called 808 -- Local/8...@from-internal-80d5,1 is ringing -- dialparties.agi: Checking CW and CFB status for extension 811 -- dialparties.agi: DbSet CALLTRACE/811 to 809 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(Local/8...@from-internal-3d60,2, SIP/811|30|Ttr) in new stack -- Called 811 -- Local/8...@from-internal-3d60,1 is ringing -- SIP/808-b7b038f8 is ringing -- SIP/811-0936e868 is ringing -- Stopped music on hold on SIP/809-09397eb0 == Spawn extension (from-internal, 8, 6) exited non-zero on 'SIP/809-09397eb0' -- Executing Macro(SIP/809-09397eb0, hangupcall) in new stack -- Executing ResetCDR(SIP/809-09397eb0, w) in new stack == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/8...@from-internal-3d60,2' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/8...@from-internal-3d60,2' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/8...@from-internal-3d60,2' == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/8...@from-internal-80d5,2' in macro 'dial' == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/8...@from-internal-80d5,2' in macro 'exten-vm' == Spawn extension (macro-dial, s, 10) exited non-zero on 'Local/8...@from-internal-80d5,2' -- Executing NoCDR(SIP/809-09397eb0, ) in new stack -- Executing Wait(SIP/809-09397eb0, 5) in new stack == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/809-09397eb0' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 3) exited non-zero on 'SIP/809-09397eb0' localhost*CLI - vi /etc/asterisk/extensions.conf [macro-hangupcall] exten = s,1,ResetCDR(w) exten = s,2,NoCDR() exten = s,3,Wait(5) exten = s,4,Hangup I tested multiple queues, and all of them are doing the same.. so please can anyone tell me how to resolve this issue. Thanks Torintino _ With Windows Live, you can organize, edit, and share your photos. http://www.microsoft.com/middleeast/windows/windowslive/products/photo-gallery-edit.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Event Log
I am using an IBM Server, after while in the MBR it said that Event logs are full, so after clearing it, the asterisk can't run. i think it deleted a file, so which file i have to create again. and what's its chmod. Thanks _ With Windows Live, you can organize, edit, and share your photos. http://www.microsoft.com/middleeast/windows/windowslive/products/photo-gallery-edit.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Event Log
No, this file is still existed, i think it's another file. Thanks From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 20 Jul 2009 13:20:23 -0500 Subject: Re: [asterisk-users] Event Log Probably /var/log/asterisk/messages 0644. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Sent: Monday, July 20, 2009 1:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Event Log I am using an IBM Server, after while in the MBR it said that Event logs are full, so after clearing it, the asterisk can't run. i think it deleted a file, so which file i have to create again. and what's its chmod. Thanks With Windows Live, you can organize, edit, and share your photos. _ More than messages–check out the rest of the Windows Live™. http://www.microsoft.com/windows/windowslive/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Clustering
Thanks Noah for your helpful reply. My setup will be 2 Asterisk (Trixbox) servers, Active/Passive, 2 PRIs through 2 Vega 400 gateways. i tried to follow some threats, like as the below: http://www.trixbox.org/forums/trixbox-forums/open-discussion/ha-cluster But i think something wrong in this or my setup, as from time to time the floating IP is timing out, and some services on both servers can't operate perfectly. As i was collecting different parts from different threats, And unfortunately i couldn't get a complete guide to follow to finalize my clustering setup successfully. So your help will be highly appreciated, if you can post your successful setup steps somewhere. Thanks a lot for your help and time. Torintino Date: Fri, 29 May 2009 22:52:06 -0400 From: noahisaacmil...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Clustering Please, does anybody have a good document describes well the optimum method to achieve Asterisk Redundancy/Clustering on 2 servers. Documentation?!... well... there's not much. It depends on what you're trying to achieve with your cluster. If you want a simple active/passive failover cluster, I'd suggest heartbeat/pacemaker for clusterizing the services coupled with drbd for replicating files. I recently set up a cluster like this that's now in production. This particular system connects to the PSTN via PRIs, and a specialized piece of hardware detects which system is the active node and physically routes the PRIs to that node. I should probably write something up and post it somewhere, but time is always an issue. If you need specific help with this kind of setup, though, feel free to ask, and I may be able to assist. If you want an active/active setup, I think you'll have to look into using dundi. - Noah ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Drag n’ drop—Get easy photo sharing with Windows Live™ Photos. http://www.microsoft.com/windows/windowslive/products/photos.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Clustering
Please, does anybody have a good document describes well the optimum method to achieve Asterisk Redundancy/Clustering on 2 servers. Thanks. _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk, SQL Database Update
Thanks for your helpful reply. I am not so good in coding. simply all i need is as follow When a call comes, goes into an IVR, and then depending on the entry option it will connect to a remote SQL Database, to check the pre-existed data, and in the end of the IVR the caller will enter an option that will need to be written in the SQL Database. Can you please give me a general scenrio how this will be achieved. and which files that i will need to modify. Thanks a lot. Date: Sun, 24 May 2009 22:15:31 +0200 From: philipp.kemp...@amooma.de To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk, SQL Database Update Torintino T schrieb: Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Depending on what you are trying to do there are various solutions: Channel Event Logging (CEL) - http://www.asterisk.org/node/48358 AGI System() ODBC_*() functions Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ More than messages–check out the rest of the Windows Live™. http://www.microsoft.com/windows/windowslive/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk, SQL Database Update
Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Thanks a lot. _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue Load, Asterisk Disconnected
Ok
Thanks a lot for your reply.
Date: Mon, 18 May 2009 14:51:58 +0300
From: a...@iq-labs.net
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Queue Load, Asterisk Disconnected
2009/5/17 Torintino T torinti...@hotmail.com:
I have Asterisk 1.2.29, Zaptel 1.2.24 , TE 121P Digium Card, and Freepbx
Setup for a queue up to 15 agents through a PRI line, it was working fine
for more than 1 year, suddenly, when there is a load on the queue, the
asterisk service disconnects and the calls are dropped. And the service
starts again after few seconds, and so on.
I am not using fax.
I checked PRI by zttool and there are no alarms.
The cdr logs
2195,2,,Busy,,2009-05-13 10:06:53,,2009-05-13 10:06:55,2,0,NO
ANSWER,DOCUMENTATION
,0225167604,237,from-internal,0225167604
0225167604,Local/2...@from-internal-a07f,2,,Busy,,2009-05-13
10:06:55,,2009-05-13 10:06:57,2,0,NO ANSWER,DOCUMENTATION
,0225167604,229,from-internal,0225167604
0225167604,Local/2...@from-internal-c662,2,,Busy,,2009-05-13
10:06:57,,2009-05-13 10:06:59,2,0,NO ANSWER,DOCUMENTATION
,0225167604,224,from-internal,0225167604
0225167604,Local/2...@from-internal-3a3a,2,,Busy,,2009-05-13
10:06:59,,2009-05-13 10:07:00,1,0,NO ANSWER,DOCUMENTATION
/usr/sbin/safe_asterisk: line 57: 14229 Segmentation fault (core dumped)
${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
cat: /var/run/asterisk.pid: No such file or directory
Automatically restarting Asterisk.
Verbosity logs:
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090513-092729|1242196049.184: Inbound recording enabled.
recordingcheck|20090513-092729|1242196049.184: CALLFILENAME=1242196049.184
-- AGI Script recordingcheck completed, returning 0
-- Executing Monitor(Local/2...@from-internal-b759,2,
wav49|1242196049.184| mb) in new stack
-- Executing Macro(Local/2...@from-internal-b759,2, dial|30|Ttr|211) in
new stack
-- Executing AGI(Local/2...@from-internal-b759,2, dialparties.agi) in
new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
-- dialparties.agi: priority is 1
dialparties.agi: Caller ID name is '0227559600' number is '0227559600'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 211 to extension map
-- dialparties.agi: Extension 211 cf is disabled
dialparties.agi: Extension 211 has do not disturb enabled
-- AGI Script dialparties.agi completed, returning 0
-- Executing NoOp(Local/2...@from-internal-b759,2, Returned from
dialparties with no extensions to call) in new stack
-- Executing Set(Local/2...@from-internal-b759,2, DIALSTATUS=BUSY) in
new
stack
-- Executing GotoIf(Local/2...@from-internal-b759,2, 1?s-BUSY|1) in new
stack
-- Goto (macro-exten-vm,s-BUSY,1)
-- Executing NoOp(Local/2...@from-internal-b759,2, Extension is reporting
BUSY and has no Voicemail) in new stack
-- Executing Busy(Local/2...@from-internal-b759,2, ) in new stack
-- Local/2...@from-internal-b759,1 is busy
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on
'Local/2...@from-internal-b759,2' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on
'Local/2...@from-internal-b759,2'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2'
== Spawn extension (ext-queues, 100, 7) exited non-zero on 'Zap/25-1'
-- Hungup 'Zap/25-1'
-- Stopped music on hold on Zap/27-1
-- Playing periodic announcement
-- Playing 'custom/Busy' (language 'en')
-- Called Local/2...@from-internal/n
-- Executing Macro(Local/2...@from-internal-e5d7,2, exten-vm|novm|221)
in
new stack
-- Executing Macro(Local/2...@from-internal-e5d7,2, user-callerid) in
new
stack
-- Executing Set(Local/2...@from-internal-e5d7,2, AMPUSER=) in new stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 0?109) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2, EMERGENCYCID=) in new
stack
-- Executing Set(Local/2...@from-internal-e5d7,2, AMPUSERCIDNAME=) in
new
stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 1?7) in new stack
-- Goto (macro-user-callerid,s,7)
-- Executing NoOp(Local/2...@from-internal-e5d7,2, Using CallerID
0227559600 0227559600) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2, FROMCONTEXT=exten-vm)
in new stack
-- Executing Macro(Local/2...@from-internal-e5d7,2,
record-enable|221|IN)
in new stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 0 0
[asterisk-users] Queue Load, Asterisk Disconnected
I have Asterisk 1.2.29, Zaptel 1.2.24 , TE 121P Digium Card, and Freepbx Setup
for a queue up to 15 agents through a PRI line, it was working fine for more
than 1 year, suddenly, when there is a load on the queue, the asterisk service
disconnects and the calls are dropped. And the service starts again after few
seconds, and so on.
I am not using fax.
I checked PRI by zttool and there are no alarms.
The cdr logs
2195,2,,Busy,,2009-05-13 10:06:53,,2009-05-13 10:06:55,2,0,NO
ANSWER,DOCUMENTATION
,0225167604,237,from-internal,0225167604
0225167604,Local/2...@from-internal-a07f,2,,Busy,,2009-05-13
10:06:55,,2009-05-13 10:06:57,2,0,NO ANSWER,DOCUMENTATION
,0225167604,229,from-internal,0225167604
0225167604,Local/2...@from-internal-c662,2,,Busy,,2009-05-13
10:06:57,,2009-05-13 10:06:59,2,0,NO ANSWER,DOCUMENTATION
,0225167604,224,from-internal,0225167604
0225167604,Local/2...@from-internal-3a3a,2,,Busy,,2009-05-13
10:06:59,,2009-05-13 10:07:00,1,0,NO ANSWER,DOCUMENTATION
/usr/sbin/safe_asterisk: line 57: 14229 Segmentation fault (core dumped)
${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
cat: /var/run/asterisk.pid: No such file or directory
Automatically restarting Asterisk.
Verbosity logs:
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090513-092729|1242196049.184: Inbound recording enabled.
recordingcheck|20090513-092729|1242196049.184: CALLFILENAME=1242196049.184
-- AGI Script recordingcheck completed, returning 0
-- Executing Monitor(Local/2...@from-internal-b759,2, wav49|1242196049.184|
mb) in new stack
-- Executing Macro(Local/2...@from-internal-b759,2, dial|30|Ttr|211) in new
stack
-- Executing AGI(Local/2...@from-internal-b759,2, dialparties.agi) in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
-- dialparties.agi: priority is 1
dialparties.agi: Caller ID name is '0227559600' number is '0227559600'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 211 to extension map
-- dialparties.agi: Extension 211 cf is disabled
dialparties.agi: Extension 211 has do not disturb enabled
-- AGI Script dialparties.agi completed, returning 0
-- Executing NoOp(Local/2...@from-internal-b759,2, Returned from dialparties
with no extensions to call) in new stack
-- Executing Set(Local/2...@from-internal-b759,2, DIALSTATUS=BUSY) in new
stack
-- Executing GotoIf(Local/2...@from-internal-b759,2, 1?s-BUSY|1) in new
stack
-- Goto (macro-exten-vm,s-BUSY,1)
-- Executing NoOp(Local/2...@from-internal-b759,2, Extension is reporting
BUSY and has no Voicemail) in new stack
-- Executing Busy(Local/2...@from-internal-b759,2, ) in new stack
-- Local/2...@from-internal-b759,1 is busy
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on
'Local/2...@from-internal-b759,2' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on
'Local/2...@from-internal-b759,2'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2'
== Spawn extension (ext-queues, 100, 7) exited non-zero on 'Zap/25-1'
-- Hungup 'Zap/25-1'
-- Stopped music on hold on Zap/27-1
-- Playing periodic announcement
-- Playing 'custom/Busy' (language 'en')
-- Called Local/2...@from-internal/n
-- Executing Macro(Local/2...@from-internal-e5d7,2, exten-vm|novm|221) in
new stack
-- Executing Macro(Local/2...@from-internal-e5d7,2, user-callerid) in new
stack
-- Executing Set(Local/2...@from-internal-e5d7,2, AMPUSER=) in new stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 0?109) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2, EMERGENCYCID=) in new
stack
-- Executing Set(Local/2...@from-internal-e5d7,2, AMPUSERCIDNAME=) in new
stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 1?7) in new stack
-- Goto (macro-user-callerid,s,7)
-- Executing NoOp(Local/2...@from-internal-e5d7,2, Using CallerID
0227559600 0227559600) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2, FROMCONTEXT=exten-vm) in
new stack
-- Executing Macro(Local/2...@from-internal-e5d7,2, record-enable|221|IN)
in new stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(Local/2...@from-internal-e5d7,2,
recordingcheck|20090513-092731|1242196051.186) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090513-092731|1242196051.186: Inbound recording enabled.
recordingcheck|20090513-092731|1242196051.186: CALLFILENAME=1242196051.186
-- AGI Script recordingcheck completed, returning 0
-- Executing
[asterisk-users] Queue Load, Asterisk Disconnected
I have Asterisk 1.2.29, Zaptel 1.2.24 and Freepbx Setup for
a queue up to 15 agents through a PRI line, it was working fine for more than 1
year, suddenly, when there is a load on the queue, the asterisk service
disconnects and the calls are dropped. And the service starts again after few
seconds, and so on.
I am not using fax.
I checked PRI by zttool and there are no alarms.
The cdr logs
2195,2,,Busy,,2009-05-13
10:06:53,,2009-05-13 10:06:55,2,0,NO
ANSWER,DOCUMENTATION
,0225167604,237,from-internal,0225167604
0225167604,Local/2...@from-internal-a07f,2,,Busy,,2009-05-13
10:06:55,,2009-05-13 10:06:57,2,0,NO
ANSWER,DOCUMENTATION
,0225167604,229,from-internal,0225167604
0225167604,Local/2...@from-internal-c662,2,,Busy,,2009-05-13
10:06:57,,2009-05-13 10:06:59,2,0,NO
ANSWER,DOCUMENTATION
,0225167604,224,from-internal,0225167604
0225167604,Local/2...@from-internal-3a3a,2,,Busy,,2009-05-13
10:06:59,,2009-05-13 10:07:00,1,0,NO
ANSWER,DOCUMENTATION
/usr/sbin/safe_asterisk: line 57: 14229 Segmentation fault (core dumped)
${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
cat: /var/run/asterisk.pid: No such file or directory
Automatically restarting Asterisk.
Verbosity logs:
-- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090513-092729|1242196049.184: Inbound recording enabled.
recordingcheck|20090513-092729|1242196049.184: CALLFILENAME=1242196049.184
-- AGI Script recordingcheck completed, returning 0
-- Executing Monitor(Local/2...@from-internal-b759,2,
wav49|1242196049.184| mb) in new stack
-- Executing Macro(Local/2...@from-internal-b759,2,
dial|30|Ttr|211) in new stack
-- Executing AGI(Local/2...@from-internal-b759,2,
dialparties.agi) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
-- dialparties.agi: priority is 1
dialparties.agi: Caller ID name is '0227559600' number is '0227559600'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 211 to extension map
-- dialparties.agi: Extension 211 cf is disabled
dialparties.agi: Extension 211 has do not disturb enabled
-- AGI Script dialparties.agi completed, returning 0
-- Executing NoOp(Local/2...@from-internal-b759,2, Returned
from dialparties with no extensions to call) in new stack
-- Executing Set(Local/2...@from-internal-b759,2,
DIALSTATUS=BUSY) in new stack
-- Executing GotoIf(Local/2...@from-internal-b759,2,
1?s-BUSY|1) in new stack
-- Goto (macro-exten-vm,s-BUSY,1)
-- Executing NoOp(Local/2...@from-internal-b759,2, Extension
is reporting BUSY and has no Voicemail) in new stack
-- Executing Busy(Local/2...@from-internal-b759,2, ) in
new stack
-- Local/2...@from-internal-b759,1 is busy
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on
'Local/2...@from-internal-b759,2' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on
'Local/2...@from-internal-b759,2'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2'
== Spawn extension (ext-queues, 100, 7) exited non-zero on 'Zap/25-1'
-- Hungup 'Zap/25-1'
-- Stopped music on hold on Zap/27-1
-- Playing periodic announcement
-- Playing 'custom/Busy' (language 'en')
-- Called Local/2...@from-internal/n
-- Executing Macro(Local/2...@from-internal-e5d7,2,
exten-vm|novm|221) in new stack
-- Executing Macro(Local/2...@from-internal-e5d7,2,
user-callerid) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2,
AMPUSER=) in new stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2,
0?109) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2,
EMERGENCYCID=) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2,
AMPUSERCIDNAME=) in new stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2,
1?7) in new stack
-- Goto (macro-user-callerid,s,7)
-- Executing NoOp(Local/2...@from-internal-e5d7,2, Using
CallerID 0227559600 0227559600) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2,
FROMCONTEXT=exten-vm) in new stack
-- Executing Macro(Local/2...@from-internal-e5d7,2,
record-enable|221|IN) in new stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 0
0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(Local/2...@from-internal-e5d7,2,
recordingcheck|20090513-092731|1242196051.186) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090513-092731|1242196051.186: Inbound recording enabled.
recordingcheck|20090513-092731|1242196051.186: CALLFILENAME=1242196051.186
-- AGI Script recordingcheck completed, returning 0
-- Executing
[asterisk-users] Cleared Event Log
I am using IBM Server I cleared the event log from BIOS and asterisk couldn't run which file i have to create ? and what is its permission? thanks a lot _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/products/events.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel Config
Is there different points in the zaptel configuration according to each country? Thanks. _ Drag n’ drop—Get easy photo sharing with Windows Live™ Photos. http://www.microsoft.com/windows/windowslive/products/photos.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel Config
so they are only. loazone and defaultzone thanks. From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 6 Apr 2009 16:01:25 -0500 Subject: Re: [asterisk-users] Zaptel Config I’d read this article (http://www.voip-info.org/wiki-Asterisk+config+zaptel.conf) but as I see it, you only have 2 lines in zaptel.conf for country specification; the rest of the lifting is done in Zapata.conf. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Torintino T Sent: Monday, April 06, 2009 3:55 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Zaptel Config Is there different points in the zaptel configuration according to each country? Thanks. What can you do with the new Windows Live? Find out _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/products/events.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Upgrade
after doing that (erasing Asterisk 1.4 completely and installing Asterisk 1.2) will this impact all of the trunks configurations that are existed in FreePBX that i made before i mean, will i need to make something to operate all these trunks configurations as before?. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 16:11:39 +0200 Subject: Re: [asterisk-users] Asterisk Upgrade Thanks to you. Date: Fri, 16 Jan 2009 13:24:16 + From: gordon+aster...@drogon.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Upgrade On Fri, 16 Jan 2009, Alex Balashov wrote: 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk I'd suggest not removing /etc/asterisk if that's the only source of your config files... If you (re)generate them from elsewhere, it's probably OK. and the important one, I'd have thought is /usr/lib/asterisk/modules Gordon 2. Install 1.2.29. Torintino T wrote: How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again in steps please. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 03:25:33 +0200 Subject: [asterisk-users] Asterisk Upgrade I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us See all the ways you can stay connected to friends and family http://www.microsoft.com/windows/windowslive/default.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/events.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Upgrade
How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again in steps please. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 03:25:33 +0200 Subject: [asterisk-users] Asterisk Upgrade I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/events.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Upgrade
Thanks. Date: Fri, 16 Jan 2009 07:15:29 -0500 From: abalas...@evaristesys.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Upgrade 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk 2. Install 1.2.29. Torintino T wrote: How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again in steps please. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 03:25:33 +0200 Subject: [asterisk-users] Asterisk Upgrade I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us See all the ways you can stay connected to friends and family http://www.microsoft.com/windows/windowslive/default.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/events.aspx___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Upgrade
Thanks to you. Date: Fri, 16 Jan 2009 13:24:16 + From: gordon+aster...@drogon.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Upgrade On Fri, 16 Jan 2009, Alex Balashov wrote: 1. rm -rf /var/lib/asterisk /var/spool/asterisk /etc/asterisk I'd suggest not removing /etc/asterisk if that's the only source of your config files... If you (re)generate them from elsewhere, it's probably OK. and the important one, I'd have thought is /usr/lib/asterisk/modules Gordon 2. Install 1.2.29. Torintino T wrote: How can i erase asterisk 1.4 completely to reinstall 1.2.29 successfully again in steps please. From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jan 2009 03:25:33 +0200 Subject: [asterisk-users] Asterisk Upgrade I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! Try it! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us See all the ways you can stay connected to friends and family http://www.microsoft.com/windows/windowslive/default.aspx ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (678) 237-1775 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Upgrade
I was Using freepbx-2.1.1, Asterisk 1.2.29 and Asterisk Addons 1.2.9 i tried to upgrade to Asterisk 1.4.22.1 and Addons 1.4.7 all of the IAX trunks got not working at all. I tried to downgrade by make clean; make; make install in Atserisk 1.2.29 directory.but make gives errors in the end. How can i downgrade asterisk again and undo all changes i made?. (in steps please). and can Backup and Restore return all the previous asterisk configurations?. Thanks. _ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy! http://spaces.live.com/spacesapi.aspx?wx_action=createwx_url=/friends.aspxmkt=en-us___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fresh installed box
Thanks Matt, I will check them. From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Sat, 25 Oct 2008 14:28:27 -0400 Subject: Re: [asterisk-users] Fresh installed box Hi Torintino, 1. Login to FreePBX, Go to extensions, Select the extension you want to configure, Scroll down to the bottom under the voicemail setup section, and check the “Attach to Email” checkbox and then save the extension and reload freepbx. Now your emails will be sent including the voicemail. Note that mail has to be setup on the box for it to work (ssmtp or local mta). 2. Here are some tutorials - http://www.voip-info.org/wiki-Asterisk+fax, http://www.voip-info.org/wiki/view/T.38 http://nerdvittles.com/index.php?p=88 http://asterfax.sourceforge.net/ 3. Ah, I’m not positive on what would work for this – sounds like some modifications to FOP may be in need. Maybe someone else on the list has ideas. Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Torintino T Sent: Saturday, October 25, 2008 5:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fresh installed box Thanks Matt, would you please tell me in details about the following 1- the Linux mail configuration steps to enable it to send voicemail to email. 2- the steps to use T.38 and pass thru...or Fax detection...and fax to email. 3- for the live monitoring.i wanna a software to monitor and to make spying on the calls, etc... if you will send me helpful documents , your help will be appreciated Thanks, Torintino From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 24 Oct 2008 21:45:22 -0400 Subject: Re: [asterisk-users] Fresh installed box http://www.trixbox.org/forums/trixbox-forums/open-discussion/asterisk-guru-queuestats-install-guide-video Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 24, 2008 8:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fresh installed box queuestats? Original Message Subject: Re: [asterisk-users] Fresh installed box From: Matt Gibson [EMAIL PROTECTED] Date: Fri, October 24, 2008 6:16 pm To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com after a fresh installation of Freepbx 1- How can i make Freepbx send voicemail to Email. (the Linux mail configuration steps) 2- How can i operate Fax machine and How it will be able to send the FAX to email. 3- Is there any software application i can run to monitor live the calls and to see precise reports of the recorded calls, queue, time conditions and all the details that are necessary for the Call Center. Hello, 1. This is an option when you setup the voicemail accounts. Go down and select the attach voicemail option. 2. You would attach via either T38 ATA and enable pass thru, or you would setup fax detection and forward it to an analogue port with the fax machine attached. Converting to PDF/etc is beyond the scope of FreePBX. 3. Yes, Freepbx comes with flash operator panel - and you could install something like the queuestats to compliment the information you receive from FOP. Thanks, Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Discover the new Windows Vista Learn more! _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fresh installed box
Thanks Matt, would you please tell me in details about the following 1- the Linux mail configuration steps to enable it to send voicemail to email. 2- the steps to use T.38 and pass thru...or Fax detection...and fax to email. 3- for the live monitoring.i wanna a software to monitor and to make spying on the calls, etc... if you will send me helpful documents , your help will be appreciated Thanks, Torintino From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 24 Oct 2008 21:45:22 -0400 Subject: Re: [asterisk-users] Fresh installed box http://www.trixbox.org/forums/trixbox-forums/open-discussion/asterisk-guru-queuestats-install-guide-video Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com : http://www.asterisk-jobs.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton Sent: Friday, October 24, 2008 8:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Fresh installed box queuestats? Original Message Subject: Re: [asterisk-users] Fresh installed box From: Matt Gibson [EMAIL PROTECTED] Date: Fri, October 24, 2008 6:16 pm To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com after a fresh installation of Freepbx 1- How can i make Freepbx send voicemail to Email. (the Linux mail configuration steps) 2- How can i operate Fax machine and How it will be able to send the FAX to email. 3- Is there any software application i can run to monitor live the calls and to see precise reports of the recorded calls, queue, time conditions and all the details that are necessary for the Call Center. Hello, 1. This is an option when you setup the voicemail accounts. Go down and select the attach voicemail option. 2. You would attach via either T38 ATA and enable pass thru, or you would setup fax detection and forward it to an analogue port with the fax machine attached. Converting to PDF/etc is beyond the scope of FreePBX. 3. Yes, Freepbx comes with flash operator panel - and you could install something like the queuestats to compliment the information you receive from FOP. Thanks, Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Discover the new Windows Vista http://search.msn.com/results.aspx?q=windows+vistamkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Freepbx or Trixbox Presentation
Please does anyone have Freepbx or Trixbox Powerpoint Presentation? Thanks _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Panasonic x Asterisk ... NO PROBLEM!
Can i install Asterisk beside Nortel PCM, just for recording all calls on E1 (incoming and outgoing calls) I want to get the E1 into Asterisk (Digium) how can this scenario be achieved in details please ? Date: Sat, 25 Oct 2008 07:42:09 +1300 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Panasonic x Asterisk ... NO PROBLEM! Jonn R Taylor wrote: Install a T1 between the Panasonic and Asterisk and program the T1 in the Panasonic as a other custom PBX. VOIP card would be the best. Jonn One thing to beware of with the Panasonic VoIP card, is that I have found no way of getting it to pass out of band DTMF, possibly because it handles this in a proprietary way. This has been my experience with a TDA100 and VoIP card. Regards, Richard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+worldmkt=en-USform=QBRE___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fresh installed box
after a fresh installation of Freepbx 1- How can i make Freepbx send voicemail to Email. (the Linux mail configuration steps) 2- How can i operate Fax machine and How it will be able to send the FAX to email. 3- Is there any software application i can run to monitor live the calls and to see precise reports of the recorded calls, queue, time conditions and all the details that are necessary for the Call Center. Thanks _ Connect to the next generation of MSN Messenger http://imagine-msn.com/messenger/launch80/default.aspx?locale=en-ussource=wlmailtagline___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
