Re: [asterisk-users] Oddball time problem in CID
Hi Al, That was it, Thank you!!! Al lists wrote: check tz option in your voicemail.conf On 10/5/07, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I have a really oddball time problem. When I check the server time using 'date' it is correct. When I review the time in Freepbx (under time conditions) it is correct. When I look at the time stamp in the CDR it is correct. When I review the time displayed for a voicemail in a web browser it is correct. When I hit *98 and then my extension the CID says a time that is some 6 hours off (early)??? I am really confused where could CID be getting this bogus info??? I am using Centos 4.5, Asterisk 1.2.7.1 http://1.2.7.1 and Freepbx version 2.3.0.3 http://2.3.0.3 Thanks Chuck Bunn ___ --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 No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.14.4/1055 - Release Date: 10/7/2007 10:24 AM ___ --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] Oddball time problem in CID
Hi, I have a really oddball time problem. When I check the server time using 'date' it is correct. When I review the time in Freepbx (under time conditions) it is correct. When I look at the time stamp in the CDR it is correct. When I review the time displayed for a voicemail in a web browser it is correct. When I hit *98 and then my extension the CID says a time that is some 6 hours off (early)??? I am really confused where could CID be getting this bogus info??? I am using Centos 4.5, Asterisk 1.2.7.1 and Freepbx version 2.3.0.3 Thanks Chuck Bunn ___ --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] Proximity detection versus GSM receiver
Hi, Can anyone tell me the pros and cons of Proximity Detection using bluetooth versus using GSM cell phone with receivers. I like the idea of calls be transferred to my cell phone when I am away from the office and I would like to implement such a system. Thanks Chuck Bunn ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Proximity Detection: Motorola Q + Bluetooth + Asterisk
Hi, Can anyone tell me if the Motorola Q has its Bluetooth always on like the IPhone? I want to use the Motorola Q in a Proximity Detection setup like that described on nerdvittles.com. I know the Treo 650 does not work well since the display must be on for the bluetooth to be on and this eats power. Thanks Chuck Bunn ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] One of my incomming lines is busy yet there is no indication in FOP.
Hi, I am in the US. It is only happening on on line out of 4. We also called Qwest (the local telco) and they said they could hear voices on the line - they are coming out next Thursday to check it. I will try moving the line to another card Would a defect cause cross talk? Thanks Lacy Moore - Aspendora wrote: On 1/12/07, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am using 2 TDM400P in a Centos 4.3 box. When we call from a cell phone to the line we get a busy signal... There was something similar to this posted a few months ago. What country is this in? I believe the similar problem was in the UK. Is this only happening on one line? What happens if you move that line to another part on the card, or what happens if moved to the other card? Try doing that, if you haven't, so we can eliminate a defect in the card. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.410 / Virus Database: 268.16.10/624 - Release Date: 1/12/2007 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One of my incomming lines is busy yet there is no indication in FOP.
Hi, I am having a weird problem with one of my incoming lines. After a reboot everything is fine if I disconnect the line from the wall and reconnect it. After an hour or so the lies goes busy but no indication of this shows up on the Flash Operator panel. I also do not see anything in the the CLI when I look at trunk or channel status. What am I missing? Is there someway to turn on extra debugging for this line so that I can see what is happening. Using Asterisk version 1.2.7.1 and FOP version .26 Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Zyxel P-2000W v2 not receiving calls
Hi, I am having an odd problem with one of our Zyxel P-2000W v2 wireless SIP phones. The phone can no longer receive calls. DND is not turned on and the phone has the exact same configuration as the other 2 phones (they each have unique extensions and such but all other settings are the same.) What can I do to debug this in the CLI. Nothing seems to shed any light on the problem. I am using the latest firmware on all of the phones 'wv0005'. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] One of my incomming lines is busy yet there is no indication in FOP.
Hi, I am having a weird problem with one of my incoming lines. After a reboot everything is fine if I disconnect the line from the wall and reconnect it. After an hour or so the lies goes busy but no indication of this shows up on the Flash Operator panel. I also do not see anything in the the CLI when I look at trunk or channel status. What am I missing? Is there someway to turn on extra debugging for this line so that I can see what is happening. Using Asterisk version 1.2.7.1 and FOP version .26 Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Zyxel P-2000W v2 not receiving calls
Hi, Sorry I forgot to mention that the phone is showing registered and 'sip show peers' shows that it is registered. Also the user can make outgoing calls without a problem. thanks Eric ManxPower Wieling wrote: Chuck Bunn wrote: Hi, I am having an odd problem with one of our Zyxel P-2000W v2 wireless SIP phones. The phone can no longer receive calls. DND is not turned on and the phone has the exact same configuration as the other 2 phones (they each have unique extensions and such but all other settings are the same.) What can I do to debug this in the CLI. Nothing seems to shed any light on the problem. I am using the latest firmware on all of the phones 'wv0005'. Problems receiving calls is frequently a registration problem. sip show peers will show the IP address of the phone if it is registered. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] One of my incomming lines is busy yet there is no indication in FOP.
Hi, I am using 2 TDM400P in a Centos 4.3 box. When we call from a cell phone to the line we get a busy signal... Thanks Lacy Moore - Aspendora wrote: On 1/12/07, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, I am having a weird problem with one of my incoming lines. After a reboot everything is fine if I disconnect the line from the wall and reconnect it. After an hour or so the lies goes busy but no indication of this shows up on the Flash Operator panel. I also do not see anything in the the CLI when I look at trunk or channel status. What am I missing? Is there someway to turn on extra debugging for this line so that I can see what is happening. Using Asterisk version 1.2.7.1 http://1.2.7.1 and FOP version .26 Disconnect what line? What kind of hardware are you using? How do you know the line goes busy? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.410 / Virus Database: 268.16.10/624 - Release Date: 1/12/2007 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Zyxel P-2000W v2 not receiving calls
Hi, Here the CLI output. SIP 498 is the calling phone and 411 is the phone that cannot receive a call: login as: root [EMAIL PROTECTED]'s password: Last login: Fri Jan 12 12:55:55 2007 from 10.0.0.72 [EMAIL PROTECTED] ~]# asterisk -r Asterisk 1.2.7.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Connected to Asterisk 1.2.7.1 currently running on turnip (pid = 2420) Verbosity is at least 3 -- Registered SIP '498' at 10.0.0.72 port 5060 expires 120 -- Executing Answer(SIP/498-e4d4, ) in new stack -- Executing Wait(SIP/498-e4d4, 1) in new stack -- Executing Goto(SIP/498-e4d4, pbdirectory|1) in new stack -- Goto (from-internal,pbdirectory,1) -- Executing AGI(SIP/498-e4d4, pbdirectory) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/pbdirectory -- AGI Script pbdirectory completed, returning 0 -- Executing GotoIf(SIP/498-e4d4, 1?hangup|1) in new stack -- Goto (from-internal,hangup,1) -- Executing Hangup(SIP/498-e4d4, ) in new stack == Spawn extension (from-internal, hangup, 1) exited non-zero on 'SIP/498-e4d4' -- Executing Macro(SIP/498-e4d4, hangupcall) in new stack -- Executing ResetCDR(SIP/498-e4d4, w) in new stack -- Executing NoCDR(SIP/498-e4d4, ) in new stack -- Executing GotoIf(SIP/498-e4d4, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing GotoIf(SIP/498-e4d4, 1?theend) in new stack -- Goto (macro-hangupcall,s,9) -- Executing Wait(SIP/498-e4d4, 5) in new stack -- Executing Hangup(SIP/498-e4d4, ) in new stack == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/498-e4d4' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 10) exited non-zero on 'SIP/498-e4d4' turnip*CLI Here is the CLI sip show peers: turnip*CLI sip show peers Name/username HostDyn Nat ACL Port Status 660/66010.0.0.76D 5060 Unmonitored 650/65010.0.0.73D 5060 Unmonitored 645(Unspecified)D 0Unmonitored 644/64410.0.0.74D 5060 Unmonitored 643/64310.0.0.82D 5060 Unmonitored 642(Unspecified)D 0Unmonitored 641(Unspecified)D 0Unmonitored 640(Unspecified)D 0Unmonitored 639/63910.0.0.83D 5060 Unmonitored 638/63810.0.0.75D 5060 Unmonitored 637(Unspecified)D 0Unmonitored 636(Unspecified)D 0Unmonitored 635(Unspecified)D 0Unmonitored 634/63410.0.0.84D 5060 Unmonitored 633(Unspecified)D 0Unmonitored 632(Unspecified)D 0Unmonitored 631/63110.0.0.85D 5060 Unmonitored 510/51010.0.0.33D 5060 OK (294 ms) 499(Unspecified)D 0Unmonitored 498/49810.0.0.72D 5060 OK (2 ms) 497/49710.0.0.70D 5060 Unmonitored 496(Unspecified)D 0Unmonitored 418/41810.0.0.80D 5060 Unmonitored 417(Unspecified)D 0Unmonitored 416(Unspecified)D 0Unmonitored 415/41510.0.0.77D 5060 Unmonitored 414/41410.0.0.69D 5060 Unmonitored 413/41310.0.0.34D 5060 OK (407 ms) 412/41210.0.0.67D 5060 OK (4 ms) 411/41110.0.0.35D 5060 OK (312 ms) 410(Unspecified)D 0UNKNOWN 31 sip peers [30 online , 1 offline] turnip*CLI Thanks Eric ManxPower Wieling wrote: Chuck Bunn wrote: Hi, Sorry I forgot to mention that the phone is showing registered and 'sip show peers' shows that it is registered. Also the user can make outgoing calls without a problem. A phone does NOT have to be registered in order to make outgoing calls. Registration
[asterisk-users] Problem with Zyxel P-2000W v2 not receiving calls
Hi, I am having an odd problem with one of our Zyxel P-2000W v2 wireless SIP phones. The phone can no longer receive calls. DND is not turned on and the phone has the exact same configuration as the other 2 phones (they each have unique extensions and such but all other settings are the same.) What can I do to debug this in the CLI. Nothing seems to shed any light on the problem. I am using the latest firmware on all of the phones 'wv0005'. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can anyone recommend a large button sip phone for the elderley.
Hi, Can anyone recommend a large button/type sip phone (VOIP) that an older person could use. I have a client that needs to have large button phones for elderly residents in her facility. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can anyone recommend a large button sip phone for the elderly.
Hi Cory, Do you know if any of the ATA to SIP converters that can handle visual indicators (flashing light in addition to ring sound) and for that matter can Asterisk handle visual indicators for ringing? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a Blue tooth wireless headset that will work with asterisk?
Hi, Does anyone know if there is a blue-tooth wireless headset that works with asterisk and/or a SIP software phone on the PC? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can anyone recommend a large button sip phone for the elderly.
Hi, So am I to understand that the visual indicator responds the same way a ring would and thus if Asterisk tells a phone to ring the visual indicator uses that signal and does not require a separate signal? I guess I am use to seeing visual indicators in hotels that blink when there is a message waiting and other stuff like that and in that case I would assume that the visual indicator has multiple uses and it some how addressable? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Examples of handeling input from phones with PHP
Hi, Can anyone direct me to where I might find examples of handling interactive input from a phone using PHP and AGI. I want to have someone dial an extension and then have the system request input from the user, take that input and put it into a database. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What does this error mean app.c: Huh....? no dial for indications?
Hi, What does the following error mean: Apr 5 12:39:40 NOTICE[22755] app.c: Huh? no dial for indications? Here is the 'full' log around the error: Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to agent '3002', on 'Local/[EMAIL PROTECTED],1' Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002 Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to agent '3001', on 'Local/[EMAIL PROTECTED],1' Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3001 Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|413|SIP/413) in new stack Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/413|20|Ttw) in new stack Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Called 413 Apr 5 12:38:24 VERBOSE[22758] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|510|SIP/510) in new stack Apr 5 12:38:24 VERBOSE[22758] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 12:38:24 VERBOSE[22758] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/510|20|Ttw) in new stack Apr 5 12:38:24 VERBOSE[22758] logger.c: -- Called 510 Apr 5 12:38:24 VERBOSE[22759] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|411|SIP/411) in new stack Apr 5 12:38:24 VERBOSE[22759] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 12:38:24 VERBOSE[22759] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/411|20|Ttw) in new stack Apr 5 12:38:24 VERBOSE[22759] logger.c: -- Called 411 Apr 5 12:38:24 VERBOSE[22758] logger.c: -- SIP/510-82b7 is ringing Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Agent/3002 is ringing Apr 5 12:38:25 VERBOSE[22759] logger.c: -- SIP/411-74d0 is ringing Apr 5 12:38:25 VERBOSE[22755] logger.c: -- Agent/3001 is ringing Apr 5 12:38:25 VERBOSE[22759] logger.c: -- SIP/411-74d0 is ringing Apr 5 12:38:25 VERBOSE[22757] logger.c: -- SIP/413-d3c8 is ringing Apr 5 12:38:25 VERBOSE[22755] logger.c: -- Agent/3005 is ringing Apr 5 12:38:29 VERBOSE[22758] logger.c: -- SIP/510-82b7 answered Local/[EMAIL PROTECTED],2 Apr 5 12:38:29 VERBOSE[22755] logger.c: -- Agent/3002 answered Zap/1-1 Apr 5 12:38:29 VERBOSE[22757] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 12:38:29 VERBOSE[22757] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 12:38:29 VERBOSE[22759] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 12:38:29 VERBOSE[22759] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 12:38:29 VERBOSE[22758] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 12:38:29 VERBOSE[22758] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 12:38:38 VERBOSE[22783] logger.c: -- Starting simple switch on 'Zap/3-1' Apr 5 12:39:39 VERBOSE[22755] logger.c: -- Started music on hold, class 'default', on Zap/1-1 Apr 5 12:39:39 VERBOSE[22755] logger.c: -- Playing 'pbx-transfer' (language 'en') Apr 5 12:39:40 NOTICE[22755] app.c: Huh? no dial for indications? Apr 5 12:39:42 VERBOSE[22755] logger.c: -- Stopped music on hold on Zap/1-1 Apr 5 12:39:42 VERBOSE[22755] logger.c: -- Executing Macro(Zap/1-1, stdexten|411|SIP/411) in new stack Apr 5 12:39:42 VERBOSE[22755] logger.c: -- Executing Set(Zap/1-1, DYNAMIC_FEATURES=automon) in new stack Apr 5 12:39:42 VERBOSE[22755] logger.c: -- Executing Dial(Zap/1-1, SIP/411|20|Ttw) in new stack Apr 5 12:39:42 VERBOSE[22755] logger.c: -- Called 411 Apr 5 12:39:44 VERBOSE[22755] logger.c: -- SIP/411-5f5d is ringing Apr 5 12:40:03 VERBOSE[22755] logger.c: -- Nobody picked up in 2 ms Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What causes deadlock?
Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing Answer(Zap/5-1, ) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing SetMusicOnHold(Zap/5-1, default) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing DigitTimeout(Zap/5-1, 5) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Set Digit Timeout to 5 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing ResponseTimeout(Zap/5-1, 30) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Set Response Timeout to 30 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing GotoIfTime(Zap/5-1, 8:00-21:00|*|*|*?default|s|7) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Goto (default,s,7) Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing Queue(Zap/5-1, extensions-home|tr|||25) in new stack Apr 5 14:02:42 VERBOSE[23363] logger.c: -- outgoing agentcall, to agent '3005', on 'Local/[EMAIL PROTECTED],1' Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Called Agent/3005 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- outgoing agentcall, to agent '3002', on 'Local/[EMAIL PROTECTED],1' Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Called Agent/3002 Apr 5 14:02:42 VERBOSE[23363] logger.c: -- outgoing agentcall, to agent '3001', on 'Local/[EMAIL PROTECTED],1' Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Called Agent/3001 Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|413|SIP/413) in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/413|20|Ttw) in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Called 413 Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|510|SIP/510) in new stack Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/510|20|Ttw) in new stack Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Called 510 Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Executing Macro(Local/[EMAIL PROTECTED],2, stdexten|411|SIP/411) in new stack Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Executing Set(Local/[EMAIL PROTECTED],2, DYNAMIC_FEATURES=automon) in new stack Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/411|20|Ttw) in new stack Apr 5 14:02:42 VERBOSE[23367] logger.c: -- Called 411 Apr 5 14:02:42 VERBOSE[23366] logger.c: -- SIP/510-1cb8 is ringing Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Agent/3002 is ringing Apr 5 14:02:43 VERBOSE[23365] logger.c: -- SIP/413-d49e is ringing Apr 5 14:02:43 VERBOSE[23363] logger.c: -- Agent/3005 is ringing Apr 5 14:02:43 VERBOSE[23365] logger.c: -- SIP/413-d49e is ringing Apr 5 14:02:43 VERBOSE[23367] logger.c: -- SIP/411-1a1a is ringing Apr 5 14:02:43 VERBOSE[23363] logger.c: -- Agent/3001 is ringing Apr 5 14:02:43 VERBOSE[23366] logger.c: -- SIP/510-1cb8 answered Local/[EMAIL PROTECTED],2 Apr 5 14:02:43 VERBOSE[23363] logger.c: -- Agent/3002 answered Zap/5-1 Apr 5 14:02:43 VERBOSE[23365] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 14:02:43 VERBOSE[23365] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 14:02:43 VERBOSE[23367] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 14:02:43 VERBOSE[23367] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 14:02:43 VERBOSE[23366] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' in macro 'stdexten' Apr 5 14:02:43 VERBOSE[23366] logger.c: == Spawn extension (macro-stdexten, s, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Apr 5 14:03:22 VERBOSE[2424] logger.c: -- Registered SIP '412' at 10.0.0.68 port 5060 expires 120 Apr 5 14:05:35 VERBOSE[23363] logger.c: == Spawn extension (default, s, 7) exited non-zero on 'Zap/5-1' Apr 5 14:05:35 VERBOSE[23363]
[Asterisk-Users] I have debug off why are the logs show debug info
Hi, I have debug off (debug level 0) why are the following lines showing up in '/var/log/asterisk/full' Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:26 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:35 DEBUG[2423] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Mar 31 06:57:12 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I have debug off why are the logs show debug info
Hi, Thank you that was it, I had 'debug' listed under 'full' in logger.conf. Not sure how I missed that... Thanks Again Filip Drągowski wrote: Look what you have in /etc/asterisk/logger.conf find: console = message = full = Hi, I have debug off (debug level 0) why are the following lines showing up in '/var/log/asterisk/full' Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:19 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:26 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Mar 31 06:56:35 DEBUG[2423] chan_sip.c: Auto destroying call '[EMAIL PROTECTED]' Mar 31 06:57:12 DEBUG[2423] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Wrong extension indicated when logging in as agent
Hi, I am not sure if this is a bug in FOP (Flash Operator Panel), a configuration error on my part or a bug in Asterisk. I am using Asterisk 1.2.5 and Zaptel 1.2.4 on a Centos 4.2 server with Linux version 2.6.9-22-EL-i686. Kernel updates are excluded and the server has been updated using 'yum -y update'. Okay here is the scenario: I am using AgentCallBackLogin as an extension in my extensions.conf. When an agent logs in from the extension they will be using to receive calls on everything is fine, so far so good. The problem occurred when an agent calls in on an extension and then registers an extension different from the extension they are using to call the AgentCallBackLogin. When an agent does this it shows the agent on the wrong extension (the one called from not the registered one). We sometimes will have one person log in all the agents from a single extension (of course when registering an agent they input the extension they will be at not the extension they are calling from). GRAPHICALLY: Extension 100Extension 101 Extension 801 ... ^ ^ ^ || || || Agent calls ext 801 from this Agent will be sitting here AgentCallBackLogin Extension desk. Agent enters user/passwd and 101 as new extension FOP shows agents logged in on ext 100 not 101 FILES: extensions.conf* [default] ;Operator queue, Operator Console, and Receptionist Phone exten = s,1,Answer() exten = s,2,SetMusicOnHold(default) exten = s,3,DigitTimeout(5) exten = s,4,ResponseTimeout(30) exten = s,5,GotoIfTime(8:00-21:00|*|*|*?default,s,7) exten = s,6,Goto(mainmenu,s,1) exten = s,7,Queue(extensions-home|tr|||25) exten = s,8,Goto(mainmenu,s,1) ... ... ;Voicemail Main exten = 800,1,Answer exten = 800,2,VoicemailMain(@default) ;Agent Login exten = 801,1,AgentCallbackLogin(||@default) ;Recording Interface exten = 820,1,Goto(phrase,s,1) ... queues.conf** [general] ;Operator Home [extensions-home] music=default strategy=ringall maxlen=0 leavewhenempty=yes context=mainmenu member = Agent/3000 member = Agent/3001 member = Agent/3002 member = Agent/3003 member = Agent/3004 member = Agent/3005 member = Agent/3006 member = Agent/3007 member = Agent/3008 member = Agent/3009 member = Agent/3010 member = Agent/3011 *** *agents.conf** [agents] wrapuptime=0 musiconhold = default updatecdr=yes recordagentcalls=no ;Operator - Ageless group=1 agent = 3000,3000,OP1 agent = 3001,3001,OP2 agent = 3002,3002,OP3 agent = 3003,3003,OP4 agent = 3004,3004,OP5 ... * Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?
Hi, What does your SIP config look like for the SJPhone? Also what operating system does this PC have and is it up to date with security and bug patches. Thanks Marco Mouta wrote: Hi all, I've my Server running well, then sometimes Sjphones looses registry and it only works well again if i restart the pc running sjphone. Has any one experience this? Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sjphone looses registry in my Asterisk Server then i need to restart Sjphone PC, why?
Hi, If your sip.conf is not setup properly SJPhone will not work. Here is my SJPHpne SIP config: sip.conf** ... ;SJphone [410] context=longdistance ;canreinvite=no type=friend username=410 secret=passwd410 callerid=410 qualify=yes nat=no host=dynamic [EMAIL PROTECTED] disallow=all ;allow=g729 allow=gsm allow=ilbc allow=ulaw allow=alaw dtmfmode=rfc2833 Callgroup=1 pickupgroup=1 ... * Thanks Marco Mouta wrote: Windows XP service Pack 2 What you mean with SIP config look like? I've everything by default, only config for Calls through SIP proxy Bug patches from sjphone? On 3/30/06, Chuck Bunn [EMAIL PROTECTED] wrote: Hi, What does your SIP config look like for the SJPhone? Also what operating system does this PC have and is it up to date with security and bug patches. Thanks Marco Mouta wrote: Hi all, I've my Server running well, then sometimes Sjphones looses registry and it only works well again if i restart the pc running sjphone. Has any one experience this? Best regards, Marco Mouta ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BUG: FOP reports incorrect (duplicate) IP address until restarted
Hi,
This problem may be related to a configuration problem but I believe it
is a bug in the FOP since restarting the FOP server clears the problem.
Here is the scenario: Using AgentCallBackLogin and have four agents
logged in a call is made to one of the agents directly from an internal
phone. Okay so far. Call is hung up and the same extension is used to
call another agent okay again, no problems. Now the problem. A call
coming from the outside, entering the queue and then being answered by
an agent - okay again. Same caller (I am running a test obviously with
same callerid) calls again and a different agent answers - problem
starts - now the first agent is showing busy with the same callerid as
the second agent, also after a few seconds the IP address of the second
agent shows as the same as the first (not possible since I can get into
each phone with there correct IP address). I can also call the first
agent without a problem from an internal phone even though FOP shows the
extension is busy (this is why I think the problem is with FOP and not
asterisk since asterisk can complete the call to a supposedly busy
extension). It also may be that asterisk is incorrectly reporting the
information and FOP is just reporting what it sees. (THUS THE REASON FOR
THE DOUBLE POST SO DON'T BAGGER ME ABOUT DOUBLE POSTING) The problem is
repeatable. I am using Asterisk 1.2.5 and Zaptel 1.2.4 on Centos 4.2
with kernel update disabled (kernel is 2.6.9-22.EL-i686) with all
updates applied through 'yum' and latest version of FOP (version 0.25).
We are using SJPhones and Zyxel P2000w-v2 wireless SIP phones with
latest firmware (wv0003 - Jan 6, 2006). Both SJPhones and Zyxel phones
are acting as agents. Please post a comment if you question anything in
my config files (if something look odd please say so!) or have seen
something similar. Here are some pieces of my files that might help:
*sip.conf
[general]
srvlookup=yes
Callgroup=1
pickupgroup=1
;SJphone
[410]
context=longdistance
;canreinvite=no
type=friend
username=410
secret=passwd410
callerid=410
qualify=yes
nat=no
host=dynamic
[EMAIL PROTECTED]
disallow=all
;allow=g729
allow=gsm
allow=ilbc
allow=ulaw
allow=alaw
dtmfmode=rfc2833
Callgroup=1
pickupgroup=1
;Zyxel - P2000WV2
[411]
context=longdistance
canreinvite=no
type=friend
username=411
secret=passwd411
callerid=411
nat=no
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=g729
allow=ulaw
allow=alaw
dtmfmode=rfc2833
Callgroup=1
pickupgroup=1
;SJphone
[412]
context=longdistance
;canreinvite=no
type=friend
username=412
secret=passwd412
callerid=412
qualify=yes
nat=no
host=dynamic
[EMAIL PROTECTED]
disallow=all
;allow=g729
allow=gsm
allow=ilbc
allow=ulaw
allow=alaw
dtmfmode=rfc2833
Callgroup=1
pickupgroup=1
;Zyxel - P2000WV2
[413]
context=longdistance
canreinvite=no
type=friend
username=413
secret=passwd413
callerid=413
nat=no
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=g729
allow=ulaw
allow=alaw
dtmfmode=rfc2833
Callgroup=1
pickupgroup=1
...
***
*extensions.conf*
[default]
;Operator queue, Operator Console, and Receptionist Phone
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(30)
exten = s,5,GotoIfTime(8:00-21:00|*|*|*?default,s,7)
exten = s,6,Goto(mainmenu,s,1)
exten = s,7,Queue(extensions-home|tr|||25)
exten = s,8,Goto(mainmenu,s,1)
include = mainmenu
;Ageless
exten = _400,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED])
exten = _405,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED])
exten = _41[0-3],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _49[6-9],1,Macro(novmail,${EXTEN},SIP/${EXTEN})
...
**
*queues.conf
[general]
;Operator Home
[extensions-home]
music=default
strategy=ringall
maxlen=0
leavewhenempty=yes
context=mainmenu
member = Agent/3000
member = Agent/3001
member = Agent/3002
member = Agent/3003
member = Agent/3004
member = Agent/3005
member = Agent/3006
member = Agent/3007
member = Agent/3008
member = Agent/3009
member = Agent/3010
member = Agent/3011
***
***agents.conf**
[agents]
wrapuptime=0
musiconhold = default
updatecdr=yes
recordagentcalls=no
;Operator - Ageless
group=1
agent = 3000,3000,Sxxx
agent = 3001,3001,Bxxx
agent = 3002,3002,E
agent = 3003,3003,As
agent = 3004,3004,Arxx
agent = 3005,3005,Lxxx
agent = 3006,3006,Loxxx
***
**modules.incl
[macro-stdexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten =
Re: [Asterisk-Users] Bluetooth headset in handsfree modewith SJPhoneorX-lite
Hi, This is interesting you used the ctp profile. Did you document how you changed it? Thanks wendell hamilton wrote: I am able to pickup, hangup, and flash, using the buttons on the phone with all of the soft clients and both of the headsets I mentioned below. I don't believe I had to change anything on the client side, just had to get the ctp (cordless telephone profile) working in the bluetooth stack, which was a pita. I struggled with the same issue...I could use the headset, but that's not very handy when you have to run to the pc to pickup or hangup a call! I'm working with doing a voice-recognition dial using the new voxeo prophecy server to add functionality to this, so that I can outgoing dial as well. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bluetooth stack for cordless telephone
Hi, Anyone know where I can download or purchase a Bluetooth stack that supports the CTP (Cordless Telephone Profile). Apparently this in the only way to get the answer/hangup button on a wireless headset to work with SIP soft phones. I have looked at the Widcomm site and they are not selling end products. Their development kit does not even support wireless headsets let alone cordless telephones. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bluetooth headset in handsfree mode with SJPhone or X-lite
Hi, After much searching I have found that it might be possible to get a bluetooth headset to answer/hangup with SJPhone or Xlite if the headset supports handsfree mode. My Toshiba bluetooth stack supports this but I have not been able to figure out how to enable it. Also Windows XP desktop bluetooth stack does not support handsfree but Windows CE does (go figure). Has anyone got handsfree mode working with a bluetooth headset? How about working with SJPhone or Xlite or some other SIP phone? For some reason the SJPhone when used with a bluetooth headset disconnects/reconnects bluetooth when the answer/hangup button is used on the headset (how the hell did that come about). Using a bluetooth headset with a SIP phone and asterisk would really help me by removing those pesky wires Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bluetooth headset in handsfree mode with SJPhoneor X-lite
Hi, I am not having trouble with the bluetooth stack since the Toshiba stack has the headset profile which supports a subset of AT commands http://en.wikipedia.org/wiki/AT_command from GSM 07.07 for minimal controls including the ability to ring, answer a call, hang up and adjust the volume. The problem is getting the softphone to work with these AT commands so that the answer/hangup function will work from the bluetooth headset. Thanks wendell hamilton wrote: Try replacing the XP Bluetooth stack with the widcomm drivers...google is your friend! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, March 27, 2006 6:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Bluetooth headset in handsfree mode with SJPhoneor X-lite Hi, After much searching I have found that it might be possible to get a bluetooth headset to answer/hangup with SJPhone or Xlite if the headset supports handsfree mode. My Toshiba bluetooth stack supports this but I have not been able to figure out how to enable it. Also Windows XP desktop bluetooth stack does not support handsfree but Windows CE does (go figure). Has anyone got handsfree mode working with a bluetooth headset? How about working with SJPhone or Xlite or some other SIP phone? For some reason the SJPhone when used with a bluetooth headset disconnects/reconnects bluetooth when the answer/hangup button is used on the headset (how the hell did that come about). Using a bluetooth headset with a SIP phone and asterisk would really help me by removing those pesky wires Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message is confidential. It may also be privileged or otherwise protected by work product immunity or other legal rules. If you have received it by mistake, please let us know by e-mail reply and delete it from your system; you may not copy this message or disclose its contents to anyone. Please send us by fax any message containing deadlines as incoming e-mails are not screened for response deadlines. The integrity and security of this message cannot be guaranteed on the Internet. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bluetooth headset in handsfree modewith SJPhoneor X-lite
Hi, Are you able to answer calls by pressing the answer/hangup button on your headset or are you using the computer to answer the calls with a SIP phone. I can get the head set to work fine with the PC and sip phones what I cannot do is get the answer/hangup button on the head set to actually answer a call (for some odd reason it turns the bluetooth on and off and this occurrs with several different head sets). Thanks for the info on the stack - good reference wendell hamilton wrote: Hi, You need to have completely replaced the Microsoft driver, because it doesn't support the headset or ctp Bluetooth profiles. This gave me fits! I followed the instructions at http://www.windowsdevcenter.com/pub/a/windows/2005/07/05/bluetooth.html and it works with both a Plantronics and a Motorola Headset, and I can answer calls with idefisk, eyebeam, x-lite, and kapanga. If you end up not having both of these in the Bluetooth service selection, you won't end up with the results you're looking for. HTH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, March 27, 2006 9:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bluetooth headset in handsfree modewith SJPhoneor X-lite ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bluetooth headset with SJPhone
Hi, Is it possible to get a blue tooth headset to work with SJPhone or any other SIP phone for that matter. Adapting the single button (answer/hangup) on the blue tooth headset to answer/accept a call has me stupefied. I must be missing something, it can not be that hard? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
Hi,
Without separate incoming and outgoing context you could not secure your
system from an outside caller using your system to dial a long distance
number.
Here is an example outgoing context that restricts who can call long
distance. If a SIP phone does not belong to the 'longdistance' context
they can only make 'local' calls through the ZAP trunk, likewise if a
outgoing ZAP channel is in the 'local' context it can only make local calls:
**Outgoing.incl
[local]
ignorepat = 9
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion(5)
exten = _9NXX,102,congestion(5)
exten = 911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911)
include = default
[longdistance]
ignorpat = 9
exten = _91NXXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91NXXNXX,2,Congestion(5)
exten = _91NXXNXX,102,congestion(5)
include = local
*
** SIP.conf**
...
[general]
srvlookup=yes
Callgroup=1
pickupgroup=1
;SJphone
[410]
context=longdistance
;canreinvite=no
type=friend
username=410
secret=passwd410
callerid=410
qualify=yes
nat=no
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=gsm
allow=ilbc
allow=ulaw
allow=alaw
dtmfmode=rfc2833
Callgroup=1
pickupgroup=1
;Zyxel - P2000WV2
[411]
context=longdistance
canreinvite=no
type=friend
username=411
secret=passwd411
callerid=411
nat=no
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
Callgroup=1
pickupgroup=1
*
and finally
***zapata.conf***
[trunkgroups]
[channels]
musiconhold=default
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
callwaitingcallerid=yes
transfer=yes
immediate=no
faxdetect=both
context=default
signalling=fxs_ks
group=1
channel = 1,5,6
context=default
signalling=fxs_ks
group=4
channel = 2
context=local
signalling=fxo_ks
group=2
channel = 3
context=longdistance
signalling=fxo_ks
group=3
channel = 4
As you can see above the outgoing context limits which phones have
access to longdistance lines. The incomming context cannot match the
outgoing or you will have on hell of a security problem...
I hope this example helps explain what I am talking about.
Thanks
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Re: [Asterisk-Users] Voice mail not working with Asteriks 1.2.5
Hi,
Check your context you need to specify voicemail as [EMAIL PROTECTED]
(context seems to have been more tightly enforced since version 1.2 came
out). Below is an example of one of the macro I use for extensions...
[macro-stdexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
Hope this helps
Thanks
Mazhar Hussain wrote:
Hi,
I have upgraded my PBX to Asterisk 1.2.5 , previously I was
using Asterisk 1.0.9, and Every thing was working fine ,But now
voice mail is not working. The error I am receiving in log files is
like following,
WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12'
I have searched for solution a lot can Any one of you let me know how
can I solve this issue do I need to apply any patch for asterisk
Here is voicemail.conf config file
[general]
format=wav49|gsm|wav
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
saycid=yes
[zonemessages]
eastern=America/New_York|'vm-received' Q 'digits/at' IMp
central=America/Chicago|'vm-received' Q 'digits/at' IMp
central24=America/Chicago|'vm-received' q 'digits/at' H
'digits/hundred' M 'hours'
[headoffice]
901=111, Arshed User, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
12 = 235, Mazhar User, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
6412 = 235, Mazhar User, [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
[other]
1234 = 5678,Company2 User,[EMAIL PROTECTED]
And here is modules.conf file
[modules]
autoload=yes
noload = app_intercom.so
noload = chan_modem.so
noload = chan_modem_aopen.so
noload = chan_modem_bestdata.so
noload = chan_modem_i4l.so
load = res_musiconhold.so
noload = chan_alsa.so
[global]
A quick response in this regard will be highly appreciated
Thanks,
Mazhar
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Re: [Asterisk-Users] How to make groups of extensions ???
Hi, I disagree that contexts are not for outgoing calls, how else do you restrict certain user to local calls only without using contexts?? On the subject of grouping extensions I use pickup groups so that any person can answer any phone in their immediate area by using a '*8' (as long as they belong to that group and they have the same context). Thanks Andrew Kohlsmith wrote: On Tuesday 21 March 2006 11:28, C F wrote: I disagree that it is very easy to miss, in fact even just mentioning it, makes it very easy to not miss, becuase it's a very understandable feature. Well when you know what to look for everything is easy to find. :-) How will groups without context help him? it's actualy both that he needs, however he will be able to get by without groups, but not without contexts. *nothing* works without contexts, which is why I said the answer doesn't help. Contexts are for incoming calls, not outgoing ones. How do contexts help him? -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi phones and WDS (Wireless Distribution System)
Hi, I use the Zyxel P-2000W v2 wireless VOIP phones with Zyxel G-1000 access points and the hand off calls fairly smoothly using a port for the hand off and using WEP security (the Zyxel is not capable of WPA security yet). I understand that people have problems with some manufactures access points not handling the hand off very well due to latency issues. I can not remember the article but I believe Network World or a similar rag did hand off tests and found the Zyxel to be one of the best at the time. Thanks [EMAIL PROTECTED] wrote: I'm about to start working with WiFi phones on my Asterisk installations. Can anyone tell me if they are using WiFi phones on wireless network that is extended with WDS and how well the phone handles jumping from access point to access point while on a call? Do any WiFi phones support WPA encryption or are they all still under the impression they are only being used on public hot spots? Thanks! Chip Schweiss ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to make groups of extensions ???
Hi,
Without separate incoming and outgoing context you could not secure your
system from an outside caller using your system to dial a long distance
number.
Here is an example outgoing context that restricts who can call long
distance. If a SIP phone does not belong to the 'longdistance' context
they can only make 'local' calls through the ZAP trunk, likewise if a
outgoing ZAP channel is in the 'local' context it can only make local calls:
**Outgoing.incl
[local]
ignorepat = 9
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion(5)
exten = _9NXX,102,congestion(5)
exten = 911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911)
include = default
[longdistance]
ignorpat = 9
exten = _91NXXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91NXXNXX,2,Congestion(5)
exten = _91NXXNXX,102,congestion(5)
include = local
*
** SIP.conf**
...
[general]
srvlookup=yes
Callgroup=1
pickupgroup=1
;SJphone
[410]
context=longdistance
;canreinvite=no
type=friend
username=410
secret=passwd410
callerid=410
qualify=yes
nat=no
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=gsm
allow=ilbc
allow=ulaw
allow=alaw
dtmfmode=rfc2833
Callgroup=1
pickupgroup=1
;Zyxel - P2000WV2
[411]
context=longdistance
canreinvite=no
type=friend
username=411
secret=passwd411
callerid=411
nat=no
host=dynamic
[EMAIL PROTECTED]
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
Callgroup=1
pickupgroup=1
*
and finally
***zapata.conf***
[trunkgroups]
[channels]
musiconhold=default
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
callwaitingcallerid=yes
transfer=yes
immediate=no
faxdetect=both
context=default
signalling=fxs_ks
group=1
channel = 1,5,6
context=default
signalling=fxs_ks
group=4
channel = 2
context=local
signalling=fxo_ks
group=2
channel = 3
context=longdistance
signalling=fxo_ks
group=3
channel = 4
As you can see above the outgoing context limits which phones have
access to longdistance lines. The incomming context cannot match the
outgoing or you will have on hell of a security problem...
I hope this example helps explain what I am talking about.
Thanks
Andrew Kohlsmith wrote:
On Tuesday 21 March 2006 12:25, Aaron Daniel wrote:
Yeah, I agree with Chuck. User's on our system are put into various
contexts depending on who they can call... local, long distance, or
internal only.
And *all* of those people are placing calls *in* to asterisk to get into those
contexts. :-)
When you pick up a telephone wired into an FXS port; asterisk sees an incoming
request for dialtone.
When you pick up your SIP phone and dial; it must match a friend or user entry
or you'll never get in.
When your IAX softphone client makes a call, again, it must match a friend or
user entry.
These are *all* incoming calls as far as Asterisk is concerned. You get
dumped into a specific part of the dialplan (the context specified) and you
tell Asterisk what they can dial. Internal extensions, external peers, Zap
channels or even applications... the second half of all of this is the
outgoing part, when Asterisk Dial()s.
-A.
-A.
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[Asterisk-Users] Is it possible to turn off password for transfers on FOP
Hi, Is it possible to turn off the request for a security code when transferring in FOP (Flash Operator Panel)? If not can the security code be set to use the SIP or voicemail passwords? I know there is a forum for FOP but no one seems to be answering there... so I thought I would see if anyone here might have experience with FOP. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to set priority for SJPhone.
Hi, Is there some way to set the priority for the poup window that SJPone displays when the phone is ringing (the popup that asks you to accept or ignore the ringing phone) so that it is always on top. We have users thas are using applications that will pop over the SJPhone pop up window and then the user cannot answer the phone (for some reason using ALT Tab to select the window is beyond their relm of understanding). Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Failed installing zaptel
Hi, I finally fixed this problem by reverting back to the 2.6.9-22.El kernel. Do the following if the 2.6.9-33.El kernel was installed: yum remove kernel-2.6.9-33.EL yum remove kernel-devel-2.6.9-33.EL vi /boot/grub/grub.conf and remove refences to new kernel... (this was the part I forgot to do the first time I tried this) reboot check to make sure the proper kernel is installed type 'uname -r' and you should get back 2.6.9-22.EL rerun make clean make linux26 make install and it should work fine. Make sure you have a link to the 2.6 kernel (ln -s /lib/modules/uname-r/build/ linux-2.6) as well. Thanks John covici wrote: Possibly you have a problem with your kernel sources -- either you did notdo a make oldconfig or some kind of make config against them,or its something wrong with the RedHat sources -- you might try a source from ftp.kernel.org and do make oldconfig with your .config and try again. on Monday 03/13/2006 Chuck Bunn([EMAIL PROTECTED]) wrote Hi, I am having the same exact problem. I am assuming that it was a problem with a kernel update I did. I am in the process of rolling back to an older kernel... I will let you let know if this works. There is also a patch for zaptel but I believe this is for going from 1.3 to 1.4? Thanks Hall, Eric M. wrote: Group Having trouble installing zaptel. Below is my server specs Intel Motherboard D101GGC TE405P CentOS-4.2-i386 Here is the output trying to do a 'make' === make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo rm -f *.ko *.mod.c .*o.cmd rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest [EMAIL PROTECTED] zaptel]# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.ELsmp/build make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:372: error: syntax error before zone_lock /usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel.c:372: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:372: error: initializer element is not constant /usr/src/zaptel/zaptel.c:372: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:373: error: syntax error before chan_lock /usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel.c:373: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:373: error: initializer element is not constant /usr/src/zaptel/zaptel.c:373: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c: In function `free_tone_zone': /usr/src/zaptel/zaptel.c:1022: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1025: warning: passing arg 1 of `_write_unlock' from
[Asterisk-Users] All calls in queue go to agent that is down??
Hi,
When an agent is logged in and his phone goes off the network (using an
SJPhone on a portable PC) and the user had forgotten to log out of the
queue all calls go to this agent that is no longer connected to the
system. I have tried training and retraining but I need some way to fix
this. This just seems like really odd behavior but I realize that SIP
has no way of telling if a connection is alive or not. I would think
that the unregistering (ie disconnection of the SIP device) of the SIP
device would be enough to fix this (why would asterisk send all calls to
a sip connection that was unregistered???) Perhaps I need to make the
SIP registration time shorter? I am using the AgentCallBackLogin command
in my configs. Here is a copy of some the files involved:
queues.conf
[general]
[default]
;Operator Home
[extensions-home]
music=default
strategy=ringall
maxlen=0
leavewhenempty=strict
context=mainmenu
member = Agent/3000
member = Agent/3001
member = Agent/3002
member = Agent/3003
member = Agent/3004
member = Agent/3005
member = Agent/3006
member = Agent/3007
member = Agent/3008
member = Agent/3009
member = Agent/3010
member = Agent/3011
*
agents.conf ***
[agents]
wrapuptime=0
musiconhold = default
updatecdr=yes
recordagentcalls=no
;Operator - Ageless
group=1
agent = 3000,3000,xxx
agent = 3001,3001,xxx
agent = 3002,3002,xxx...
extensions-home.incl included in extensions.conf
[default]
;Operator queue, Operator Console, and Receptionist Phone
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(30)
exten = s,5,GotoIfTime(8:00-21:00|*|*|*?default,s,7)
exten = s,6,Goto(mainmenu,s,1)
exten = s,7,Queue(extensions-home|tn|||25)
exten = s,8,Goto(mainmenu,s,1)
include = mainmenu
;Ageless
exten = _400,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED])
exten = _405,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED])
exten = _41[0-3],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(novmail,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED])
exten = _51[0],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _590,1,Macro(novmail,${EXTEN},ZAP/3)
;Chicken
;exten = _60[0],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Resedential
;exten = _70[0-3],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain(@default)
;Agent Login
exten = 801,1,AgentCallbackLogin(||@default)
;Recording Interface
exten = 820,1,Goto(phrase,s,1)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;Music on Hold
exten = 870,1,Answer
exten = 870,2,SetMusicOnHold(default)
exten = 870,3,WaitMusicOnHold(420)
exten = 870,4,Hangup
*
Thanks
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Re: [Asterisk-Users] Failed installing zaptel
Hi, I would rather not reinstall Centos. I have tried using 'yum remove kernel-2.6.9-34' and 'yum remove kernel-devel-2.6.9-34' and rebooting. When I run 'uname -r' I still get 2.6.9-34.ELsmp as the kernel so I am missing some steps here, I'll post more as soon as I figure out what I did wrong... Thanks Curt Shaffer wrote: I had the same kind of issue myself. The kernel was upgrading to 2.6.9-34 from 2.6.9-22 but for some reason it did not appear that way to the compiler. I reinstalled Cent OS 4.2 and updated everything except for the kernel and did a wget for the 2.6.9-22 source from the mirror and it worked like a charm! HTH Curt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, March 13, 2006 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Failed installing zaptel Hi, I am having the same exact problem. I am assuming that it was a problem with a kernel update I did. I am in the process of rolling back to an older kernel... I will let you let know if this works. There is also a patch for zaptel but I believe this is for going from 1.3 to 1.4? Thanks Hall, Eric M. wrote: Group Having trouble installing zaptel. Below is my server specs Intel Motherboard D101GGC TE405P CentOS-4.2-i386 Here is the output trying to do a 'make' === make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo rm -f *.ko *.mod.c .*o.cmd rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest [EMAIL PROTECTED] zaptel]# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.ELsmp/build make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:372: error: syntax error before zone_lock /usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel.c:372: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:372: error: initializer element is not constant /usr/src/zaptel/zaptel.c:372: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:373: error: syntax error before chan_lock /usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel.c:373: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:373: error: initializer element is not constant /usr/src/zaptel/zaptel.c:373: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c: In function `free_tone_zone': /usr/src/zaptel/zaptel.c:1022: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1025: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone': /usr/src/zaptel/zaptel.c:1035: warning: passing arg 1 of `_write_lock
Re: Spam? Re: [Asterisk-Users] Failed installing zaptel
Hi, I had asterisk running great on Centos 4.2 - kernel-2.6.9.22.EL, my problems did not start until I accidentially upgraded to kernel-2.6.9-34.EL. If you use Centos yum for updating then make sure you add 'exclude=kernel*' to the /etc/yum.conf file - my problems started when I acidentially forgot to do this. Thanks Hall, Eric M. wrote: Chuck, Thank You I'm also going to try CentOS 3 The problem is I have SATA HDD and running in to trouble getting Linux installed. Will update after I test Ver 3 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Monday, March 13, 2006 12:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Spam? Re: [Asterisk-Users] Failed installing zaptel Hi, I am having the same exact problem. I am assuming that it was a problem with a kernel update I did. I am in the process of rolling back to an older kernel... I will let you let know if this works. There is also a patch for zaptel but I believe this is for going from 1.3 to 1.4? Thanks Hall, Eric M. wrote: Group Having trouble installing zaptel. Below is my server specs Intel Motherboard D101GGC TE405P CentOS-4.2-i386 Here is the output trying to do a 'make' === make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo rm -f *.ko *.mod.c .*o.cmd rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest [EMAIL PROTECTED] zaptel]# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.ELsmp/build make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:372: error: syntax error before zone_lock /usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel.c:372: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:372: error: initializer element is not constant /usr/src/zaptel/zaptel.c:372: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:373: error: syntax error before chan_lock /usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel.c:373: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:373: error: initializer element is not constant /usr/src/zaptel/zaptel.c:373: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c: In function `free_tone_zone': /usr/src/zaptel/zaptel.c:1022: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1025: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone': /usr/src/zaptel/zaptel.c:1035: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1042: warning: passing
[Asterisk-Users] Zaptel not compiling on lastest Centos 4.2 kernel.
Hi, I made a big mistake on a Centos 4.2 box - I forgot to exclude the kernel from updating. Now zaptel will not do a make linux26 see below. Is there a way to roll this back or is there a patch to get Zaptel to compile? I have a link to the modules using 'ln -s /lib/modules/uname -r/build linux-2.6 so that I did not have to specifiy the kernel version directly. cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.ELsmp/build make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel-1.2.4 XPPMOD= modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' CC [M] /usr/src/zaptel-1.2.4/zaptel.o /usr/src/zaptel-1.2.4/zaptel.c:384: error: syntax error before zone_lock /usr/src/zaptel-1.2.4/zaptel.c:384: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel-1.2.4/zaptel.c:384: error: incompatible types in initialization /usr/src/zaptel-1.2.4/zaptel.c:384: error: initializer element is not constant /usr/src/zaptel-1.2.4/zaptel.c:384: warning: data definition has no type or storage class /usr/src/zaptel-1.2.4/zaptel.c:385: error: syntax error before chan_lock /usr/src/zaptel-1.2.4/zaptel.c:385: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel-1.2.4/zaptel.c:385: error: incompatible types in initialization /usr/src/zaptel-1.2.4/zaptel.c:385: error: initializer element is not constant /usr/src/zaptel-1.2.4/zaptel.c:385: warning: data definition has no type or storage class /usr/src/zaptel-1.2.4/zaptel.c: In function `free_tone_zone': /usr/src/zaptel-1.2.4/zaptel.c:1034: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:1037: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `zt_register_tone_zone': /usr/src/zaptel-1.2.4/zaptel.c:1047: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:1054: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `set_tone_zone': /usr/src/zaptel-1.2.4/zaptel.c:1095: warning: passing arg 1 of `_read_lock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:1107: warning: passing arg 1 of `_read_unlock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `zt_chan_reg': /usr/src/zaptel-1.2.4/zaptel.c:1188: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:1211: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel-1.2.4/zaptel.c:1584: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:1620: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `zt_ctl_ioctl': /usr/src/zaptel-1.2.4/zaptel.c:3343: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c:3345: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel-1.2.4/zaptel.c: In function `zt_init': /usr/src/zaptel-1.2.4/zaptel.c:6553: error: incompatible types in assignment /usr/src/zaptel-1.2.4/zaptel.c: At top level: /usr/src/zaptel-1.2.4/zaptel.c:188: warning: 'fcstab' defined but not used make[2]: *** [/usr/src/zaptel-1.2.4/zaptel.o] Error 1 make[1]: *** [_module_/usr/src/zaptel-1.2.4] Error 2 make[1]: Leaving directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' make: *** [linux26] Error 2 [EMAIL PROTECTED] zaptel-1.2.4]# Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Failed installing zaptel
Hi, I am having the same exact problem. I am assuming that it was a problem with a kernel update I did. I am in the process of rolling back to an older kernel... I will let you let know if this works. There is also a patch for zaptel but I believe this is for going from 1.3 to 1.4? Thanks Hall, Eric M. wrote: Group Having trouble installing zaptel. Below is my server specs Intel Motherboard D101GGC TE405P CentOS-4.2-i386 Here is the output trying to do a 'make' === make clean rm -f torisatool makefw tor2fw.h radfw.h rm -f ztcfg torisatool makefw ztmonitor ztspeed zttool zttest fxotune rm -f *.o ztcfg tzdriver sethdlc sethdlc-new rm -f zonedata.lo tonezone.lo libtonezone.so *.lo rm -f *.ko *.mod.c .*o.cmd rm -rf .tmp_versions rm -f gendigits tones.h rm -f libtonezone* rm -f tor2ee rm -f fxotune rm -f core rm -f ztcfg-shared fxstest [EMAIL PROTECTED] zaptel]# make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o zttool.o zttool.c cc -o zttool zttool.o -lnewt cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE-DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\/etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm /lib/modules/2.6.9-34.ELsmp/build make -C /lib/modules/2.6.9-34.ELsmp/build SUBDIRS=/usr/src/zaptel modules make[1]: Entering directory `/usr/src/kernels/2.6.9-34.EL-smp-i686' CC [M] /usr/src/zaptel/zaptel.o /usr/src/zaptel/zaptel.c:372: error: syntax error before zone_lock /usr/src/zaptel/zaptel.c:372: warning: type defaults to `int' in declaration of `zone_lock' /usr/src/zaptel/zaptel.c:372: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:372: error: initializer element is not constant /usr/src/zaptel/zaptel.c:372: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c:373: error: syntax error before chan_lock /usr/src/zaptel/zaptel.c:373: warning: type defaults to `int' in declaration of `chan_lock' /usr/src/zaptel/zaptel.c:373: error: incompatible types in initialization /usr/src/zaptel/zaptel.c:373: error: initializer element is not constant /usr/src/zaptel/zaptel.c:373: warning: data definition has no type or storage class /usr/src/zaptel/zaptel.c: In function `free_tone_zone': /usr/src/zaptel/zaptel.c:1022: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1025: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_register_tone_zone': /usr/src/zaptel/zaptel.c:1035: warning: passing arg 1 of `_write_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1042: warning: passing arg 1 of `_write_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `set_tone_zone': /usr/src/zaptel/zaptel.c:1083: warning: passing arg 1 of `_read_lock' from incompatible pointer type /usr/src/zaptel/zaptel.c:1095: warning: passing arg 1 of `_read_unlock' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_reg': /usr/src/zaptel/zaptel.c:1176: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1199: warning: passing arg 1 of `_write_unlock_irqrestore' from incompatible pointer type /usr/src/zaptel/zaptel.c: In function `zt_chan_unreg': /usr/src/zaptel/zaptel.c:1572: warning: passing arg 1 of `_write_lock_irqsave' from incompatible pointer type /usr/src/zaptel/zaptel.c:1608: warning: passing arg 1 of `_write_unlock_irqrestore' from
[Asterisk-Users] No ring when doing blind transfer.
Hi,
I have an odd problem when doing a blind transfer. The transfer is
intiated and the transferred caller hears nothing until the timeout. I
have tried setting the 'r' and the 'm' variables in the dial command.
Nothing happens when I use the 'r' variable when I use the 'm' variable
I briefly hear music on hold and then it stops until the timeout for no
answer is reached. When the timeout is reached and no on answers the
system does go to voice mail as expected. I have also tried it without
either the 'r' or 'm' variables and I get the same results no ring. I am
using asterisk 1.2.4 with zaptel 1.2.3.
Here are my files:
extensions.conf **
[general]
#include macros.incl
#include outgoing.incl
#include extensions-home.incl
#include menu.incl
[globals]
OUTBOUNDTRUNK=Zap/g1
PSTN1=Zap/1
PSTN2=Zap/2
PSTN3=Zap/5
PSTN4=Zap/6
PHONE1=Zap/3
PHONE2=Zap/4
***
macros.incl ***
[macro-stdexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
[macro-menuexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttmw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
[macro-novmail]
exten = s,1,Dial(${ARG2},20,Ttw)
exten = s,2,Playback(thank-you-for-callinggoodbye)
exten = s,3,Hangup
exten = s,102,Playback(thank-you-for-callinggoodbye)
exten = s,103,Hangup
**
extensions-home.incl ***
[default]
;Operator queue, Operator Console, and Receptionist Phone
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(30)
exten = s,5,GotoIfTime(8:00-21:00|*|*|*?default,s,7)
exten = s,6,Goto(mainmenu,s,1)
exten = s,7,Queue(extensions-home|tn|||25)
exten = s,8,Goto(mainmenu,s,1)
include = mainmenu
;Ageless
exten = _400,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED])
exten = _405,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED])
exten = _41[0-3],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(novmail,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED])
exten = _51[0],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _590,1,Macro(novmail,${EXTEN},ZAP/3)
;Chicken
;exten = _60[0],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Resedential
;exten = _70[0-3],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain(@default)
;Agent Login
exten = 801,1,AgentCallbackLogin(||@default)
;Recording Interface
exten = 820,1,Goto(phrase,s,1)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;Music on Hold
exten = 870,1,Answer
exten = 870,2,SetMusicOnHold(default)
exten = 870,3,WaitMusicOnHold(420)
exten = 870,4,Hangup
Thanks
___
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[Asterisk-Users] Music on hold volume too high - using built in music on hold.
Hi, I saw this problem mentioned before but the user appeared to be using the MP3 software with asterisk. I am using the native music on hold player in asterisk 1.2 and I too have a volume problem with music on hold. Is this controllable through the 'indications.conf'? I know this file controls frequency range for various sounds might it also control sound level or am I barking up the wrong tree? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?
Hi, Okay but then how do you transfer across contexts then? Thanks Moises Silva wrote: you need to set a TRANSFER_CONTEXT, either for the transferer or transferee channel. I dont know why, but res_features give priority to the transferee TRANSFER_CONTEXT, if not found, then use the transferer TRANSFER_CONTEXT. That context is used to match the extension to dial. So you can set this var to any context you want. Regards On 2/23/06, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Is setting the variable _TRANSFER_CONTEXT required in features.conf for Asterisk 1.2.4? It appears from the Wiki that transfers across contexts are not possible when this is set. If it is not set can one trasfer across contexts?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.1.0/269 - Release Date: 2/24/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?
Hi, I support multiple context on one asterisk server. I have a situation where there is a spa that has seperate voicemail and extensions and a resturant on the same campus that has different extensions and voicemail. They both use the same asterisk server but I do need the ability to transfer a caller from the spa to the resturant and vise versa. There are seperate phone lines comming in for the spa and resturant as well. Thanks Moises Silva wrote: it seems im not undestanding your question then. Could you provide a practical example? On 2/24/06, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Okay but then how do you transfer across contexts then? Thanks Moises Silva wrote: you need to set a TRANSFER_CONTEXT, either for the transferer or transferee channel. I dont know why, but res_features give priority to the transferee TRANSFER_CONTEXT, if not found, then use the transferer TRANSFER_CONTEXT. That context is used to match the extension to dial. So you can set this var to any context you want. Regards On 2/23/06, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, Is setting the variable _TRANSFER_CONTEXT required in features.conf for Asterisk 1.2.4? It appears from the Wiki that transfers across contexts are not possible when this is set. If it is not set can one trasfer across contexts?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.1.0/269 - Release Date: 2/24/2006 ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org http://www.gnu.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.1.0/269 - Release Date: 2/24/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Features set in the features.conf stopped working after upgrade.
Hi, I recently moved all of my conf files over to a new Asterisk 1.2.4 server and every works except the features enabled in features.conf. Was there a syntax chnage in 1.2.4? Or is there something else... Here is my features.conf: [general] parkext = 880; What ext. to dial to park parkpos = 881-890; What extensions to park calls on context = parkedcalls; Which context parked calls are in parkingtime = 45; Number of seconds a call can be parked for ; (default is 45 seconds) transferdigittimeout = 3; Number of seconds to wait between digits when transfering a call courtesytone = beep; Sound file to play to the parked caller ; when someone dials a parked call xfersound = beep; to indicate an attended transfer is complete xferfailsound = beeperr; to indicate a failed transfer ;adsipark = yes; if you want ADSI parking announcements ;findslot = next; Continue to the 'next' parking space. Defaults to 'first' available pickupexten = *8; Configure the pickup extension. Default is *8 ;featuredigittimeout = 500; Max time (ms) between digits for ; feature activation. Default is 500 [featuremap] blindxfer = ##; Blind transfer ;disconnect = *0; Disconnect automon = *1; One Touch Record atxfer = *2; Attended transfer ** Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Keep getting message in logs that pbx.c cannot find extension context 'default'
Hi, I am getting repeated messages in my logs with the following: * Feb 23 07:56:11 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23 07:56:11 DEBUG[2470] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] Feb 23 07:56:12 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23 07:56:12 DEBUG[2470] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] Feb 23 07:56:14 NOTICE[2470] pbx.c: Cannot find extension context 'default' Feb 23 07:56:14 DEBUG[2470] chan_sip.c: SIP message could not be handled, bad request: [EMAIL PROTECTED] * I do not have a default context used in my extensions.conf - I use other names. Am I required to have a context named 'default'?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is setting the variable _TRANSFER_CONTEXT required in features.conf?
Hi, Is setting the variable _TRANSFER_CONTEXT required in features.conf for Asterisk 1.2.4? It appears from the Wiki that transfers across contexts are not possible when this is set. If it is not set can one trasfer across contexts?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Okay can somebody explain this...
Hi, I got rid of the messages I was getting in the CDR (pbx.c: Cannot find extension context 'default') by adding a blank 'default' context at the front of my extensions.conf (I use the context 'extensions-home') this also (well sort of ) fixed my problem with blind transfers. I can blind transfer using '##' (set in features.conf) once but I cannot transfer a second time (I was unable to transfer at all before adding the blank 'default' context)?? Now I am really confused. Do I have to have a context called 'default' for this to work properly? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...
Hi, I am throwing in the towel. I tried recompiling asterisk with changes to rights in app_voicemail.c (changed 0700 to 0770, 0600 to 0660) but I think the writing of voicemail is taking place somewhere else as this did fix new directories but files are still beining writen without group permissions. I also modified the call.monitor file in ARI so that permission were 0770 instaed of 0700 but this still did not work. I do have files being created with the asterisk group and user though! So I am ready to try the script for changing writes. Is there anything I need to do besides adding the chmod blah blah blah to a file and then making it executable?? Thanks Giorgio Incantalupo wrote: Hi Chuck. I had the same problem. I solved it using the externnotify parameter inside voicemail.conf. Just launch a script which changes the /var/spool/asterisk permissions. Giorgio Incantalupo Chuck Bunn wrote: Hi, I thought I had this problem licked but there still is a rights problem with ARI and Asterisk when using a non-root user (Following the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-rootdiff2=25). When I issue the following: chmod --recursive u=rwX,g=rX,o= /var/spool/asterisk The above command results in the following rights on messages: msg.gsmrwxr-x--- asterisk msg.txtrw-r- asterisk msg.wavrwxr-x--- asterisk I can transfer voicemails and play them but new messages comming in get the following: msg.gsmrwx-- asterisk msg.txtrw-r--r-- asterisk msg.wavrwx-- asterisk After changing the rights a transferred messages has the folowing rights: msg.gsmrw-r- apache msg.txtrw-r- apache msg.wavrw-r- apache New voicemail cannot be played, deleted or transferred by the ARI application. Apache is belongs to the Asterisk group. I thought I understood SUID, GUID and sticky bit now I am not so sure. What is really confussing to me is why the rights on the .txt file do not match the other 2 after running the 'chmod --recursive ...' command. Any help here would be greatly appreciated. I am using the lastest versions of Asterisk 1.2.4 and Zaptel 1.2.3, etc. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...
Hi, Just so I am clear this patch will work with 1.2.4 and requires manual updating to files and then a recomplie of Asterisk source correct?? Thanks Ben Klang wrote: Hello, I found the same problem very frustrating, mostly because it causes Asterisk to ignore ACLs and umask settings. If you are interested I have a patch submitted as Digium bug 6334 at http://bugs.digium.com/view.php?id=6334. This patch resolved the issue for me. Regards, /BAK/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...
Hi, Could you post the updated patch for 1.2.4 Thanks Ben Klang wrote: On Thursday 16 February 2006 11:47, you wrote: Just so I am clear this patch will work with 1.2.4 and requires manual updating to files and then a recomplie of Asterisk source correct?? This patch was written against trunk a couple weeks ago. Last night I applied it to 1.2.4 and there were only two small conflicts (easily resolved). Recompile and install Asterisk. You may need to manually poke existing files to get the perms the way you like but all new files should be created correctly. If you're having trouble getting it to apply to 1.2.4 let me know and I'll send you my rebuild patch. If you happen to be a SuSE user I've got Asterisk 1.2.4 RPMs built for SuSE 10.0. /BAK/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Install instructions for FOP Flash Operator Panel do not make sense...
Hi, Anyone got AFOP working. The install instructions tell you to copy all of the files extracted under the 'html' directory to a subdirectory under your main web directory (in my case this is /var/www/html/panel/) and then the instructions talk about modifying the 'op_server.cfg' file but they do not tell you were to put this file. There is something wrong with the instructions??? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...
Hi Giorgio: That seems like a kind of a kludge. I would rather have the program work right, than adding a work around. Dan of Littlejohnsconsulting has told me of one problem in ARI that he is fixing but I do not understand how it will fix the issue yet?? I will let you know as I find out more... Thanks Giorgio Incantalupo wrote: Hi Chuck. I had the same problem. I solved it using the externnotify parameter inside voicemail.conf. Just launch a script which changes the /var/spool/asterisk permissions. Giorgio Incantalupo Chuck Bunn wrote: Hi, I thought I had this problem licked but there still is a rights problem with ARI and Asterisk when using a non-root user (Following the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-rootdiff2=25). When I issue the following: chmod --recursive u=rwX,g=rX,o= /var/spool/asterisk The above command results in the following rights on messages: msg.gsmrwxr-x--- asterisk msg.txtrw-r- asterisk msg.wavrwxr-x--- asterisk I can transfer voicemails and play them but new messages comming in get the following: msg.gsmrwx-- asterisk msg.txtrw-r--r-- asterisk msg.wavrwx-- asterisk After changing the rights a transferred messages has the folowing rights: msg.gsmrw-r- apache msg.txtrw-r- apache msg.wavrw-r- apache New voicemail cannot be played, deleted or transferred by the ARI application. Apache is belongs to the Asterisk group. I thought I understood SUID, GUID and sticky bit now I am not so sure. What is really confussing to me is why the rights on the .txt file do not match the other 2 after running the 'chmod --recursive ...' command. Any help here would be greatly appreciated. I am using the lastest versions of Asterisk 1.2.4 and Zaptel 1.2.3, etc. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Rights problem with Voicemail and non-root user - yeah I know, I thought I had it fixed...
Hi, I thought I had this problem licked but there still is a rights problem with ARI and Asterisk when using a non-root user (Following the wiki at http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-rootdiff2=25). When I issue the following: chmod --recursive u=rwX,g=rX,o= /var/spool/asterisk The above command results in the following rights on messages: msg.gsmrwxr-x--- asterisk msg.txtrw-r- asterisk msg.wavrwxr-x--- asterisk I can transfer voicemails and play them but new messages comming in get the following: msg.gsmrwx-- asterisk msg.txtrw-r--r-- asterisk msg.wavrwx-- asterisk After changing the rights a transferred messages has the folowing rights: msg.gsmrw-r- apache msg.txtrw-r- apache msg.wavrw-r- apache New voicemail cannot be played, deleted or transferred by the ARI application. Apache is belongs to the Asterisk group. I thought I understood SUID, GUID and sticky bit now I am not so sure. What is really confussing to me is why the rights on the .txt file do not match the other 2 after running the 'chmod --recursive ...' command. Any help here would be greatly appreciated. I am using the lastest versions of Asterisk 1.2.4 and Zaptel 1.2.3, etc. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ARI - Voicemail not showing - Problem solved!
Hi, Just wanted to pass on a fix that I found with the ARI recordings interface (www.littlejohnconsulting.com) for using a browser to access voice mail. It turned out to be a rights issue and group membership issue. I was planning on moving Asterisk to a non-root (http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+non-rootdiff2=25) user but I had not done this prior to installing ARI. Once I setup Asterisk to use a non root user and added 'apache' to the 'asterisk' user group everything worked perfectly. I also want to thank Dan for his patience and help in solving my problem. If you have not tried the ARI interface you might look at it, my clients love it!! Great job DAN Thanks Chuck Bunn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with ARI and seeing voicemail...
Hi, I have tried both the stable version ARI-00.04.006 and the development version ARI-00.05.018 with the same results. I can see call detail records just fine but I cannot see any voicemail. I am using the voicemail extension and password to log in but I still do not see anything. If I log in as Admin with ari_password I see all of the call detail but still no voice mail. Any ideas where I might look for my problem. Voicemail is working since I can call the voicemail extension and retrieve messages. I am not using AMP and I have set the standalone flag to true. Thanks Chuck Bunn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SJPhone with external ringer
Hi, Does anyone know if it is possible to setup an SJPhone with an external ringer of some sort. One of the operators may not always be at her desk and when she is not wearing a headset she cannot hear the phone ring - is there some way to fix this? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automatic logoff of all agents at set time
Hi, I have not been able to find anything about persistent agents in any wiki? Where does this command go and what is its syntax? Thanks Michiel van Baak wrote: On 18:06, Tue 27 Dec 05, Bud Bach wrote: But, if the agents don't log out for some reason, they will still be logged in the next time the queue opens even if they aren't there right? yes. What you can do is 2 things: * you can set the autologoff time in agents.conf. This can give you some trouble when agents go to the toilet or grab a cup of coffee. * set persistant agents to off and restart asterisk at midnight. This will logoff the agents :) Hope this helps ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automatic logoff of all agents at set time
Hi, When I add 'persistentmembers=no' in queues.conf and reload I get a message in the message log file saying unknown keyword 'persistentmembers'. I got the syntax from http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue under the notes section. Thanks Alexander Lopez wrote: It is set in the queues.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Wednesday, December 28, 2005 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Automatic logoff of all agents at set time Hi, I have not been able to find anything about persistent agents in any wiki? Where does this command go and what is its syntax? Thanks Michiel van Baak wrote: On 18:06, Tue 27 Dec 05, Bud Bach wrote: But, if the agents don't log out for some reason, they will still be logged in the next time the queue opens even if they aren't there right? yes. What you can do is 2 things: * you can set the autologoff time in agents.conf. This can give you some trouble when agents go to the toilet or grab a cup of coffee. * set persistant agents to off and restart asterisk at midnight. This will logoff the agents :) Hope this helps ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automatic logoff of all agents at set time
Hi, Oh sorry I am using asterisk 1.2.1 Thanks Kevin P. Fleming wrote: Chuck Bunn wrote: When I add 'persistentmembers=no' in queues.conf and reload I get a message in the message log file saying unknown keyword 'persistentmembers'. I got the syntax from http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue under the notes section. You haven't told us what version of Asterisk you are using, but you are probably using 1.0.x, if it doesn't support that option. Regardless, that option won't do what you want anyway, since you are using agents and not dynamic queue members. The 'persistentagents' option in agents.conf could do it, but that's still an ugly way to handle it. Since agents can be logged off using CLI commands or manager interface actions, it would be quite simple to write a script to run via a cron job late at night to forcibly log off all your agents. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automatic logoff of all agents at set time
Hi, Is there a way to force the logoff of all agents at a set time say 8:00PM or do I have to do an agentcallbacklogin with each agents credentials? I am using Asterisk 1.2 The wiki shows an extension that the agent calls to preform the logoff - I need something that is completely automated as we need calls to stop going to a queue and to go to voice mail after hours. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automatic logoff of all agents at set time
Hi, I understand how GototIfTime works but that still leaves agents logged in and if an agent is absent the next day calls will go to an agent that is not there. Thanks Michiel van Baak wrote: On 16:22, Tue 27 Dec 05, Chuck Bunn wrote: Hi, Is there a way to force the logoff of all agents at a set time say 8:00PM or do I have to do an agentcallbacklogin with each agents credentials? I am using Asterisk 1.2 The wiki shows an extension that the agent calls to preform the logoff - I need something that is completely automated as we need calls to stop going to a queue and to go to voice mail after hours. Hi, You dont have to logoff your agents to do this. Have a look at the extensions.conf cmd GotoIfTime: http://www.voip-info.org/wiki-Asterisk+cmd+GotoIfTime Good luck ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1
Hi, What are you codec and dmtfmode settings in sip.conf and in the sip phone settings. If you dmtfmode is set to 'inband' and you are using anything other than ulaw or alaw codec it wont work. Also since your hear the phone sometimes you may be experiencing QOS issues on your network. Doe you have QOS set up on your switches in the points between the server running asterisk and the sip client? Hope this helps Evil Skymarshal wrote: Hi, I am using a sjphone to connect via SIP to an Asterisk 1.2.1. For testing reasons I but the following in extensions.conf ---cut--- [from-sip] exten = 2000,1,Answer() exten = 2000,2,Wait(1) exten = 2000,3,SayDigits(123) exten = 2000,4,Hangup() ---cut--- When ever I call the 2000 asterisk -vc says: ---cut--- -- Executing Answer(SIP/2303-1ae1, ) in new stack -- Executing Wait(SIP/2303-1ae1, 1) in new stack -- Executing SayDigits(SIP/2303-1ae1, 123) in new stack -- Playing 'digits/1' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/3' (language 'en') -- Executing Hangup(SIP/2303-1ae1, ) in new stack ---cut--- BUT I don't hear it everytime! Why? Sometimes I can hear it and most time I can not. I redial 20 times and I can hear it only 3-4 times. Does anybody have an idea what kind of strange problem that could be? Thanx ES ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1
Hi, If you do not have QOS assigned to the SIP protocol it is quite possible that there are packets time outs and the packets are discarded. Is it possible to test the network during the evening or at a time when traffic is at it lowest? Also try several traceroutes and see if there is a wide variation in return times (widely varying treceroutes could indicate network saturation). You are using gsm are you using dmtfmode=rfc2833 or something else (this must be set in the sip.conf and on the sip soft phone and they must match!) Thanks Evil Skymarshal wrote: Hi Chuck, 2005/12/17, Chuck Bunn [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: What are you codec and dmtfmode settings in sip.conf and in the sip phone settings. I use gsm. If you dmtfmode is set to 'inband' and you are using anything other than ulaw or alaw codec it wont work. I changed the settings and tried: ---cut--- exten = 2000,1,Answer() exten = 2000,2,Wait(1) exten = 2000,3,Playback(hello-world) exten = 2000,4,Hangup() ---cut--- Same problem. Sometimes it works but most of the times it doesn't. Also since your hear the phone sometimes you may be experiencing QOS issues on your network. Of course it could be a QOS problem. But should I hear at least something? cu ES ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1
Hi, Something else I should mention. Sip uses UDP and TCP packets. TCP packets are used if there is congestion on the network. I am unclear about what mechanism causes sip to switch between UDP and TCP but I believe it is controllable - I believe It would be easier to use QOS though. If UDP is used that packets could time out and you would never know it since UDP is dumb and has no packet loss recovery mechanism. What is the topology of your network. Is the Asterisk box and the client on the same backbone and switch? Thanks Evil Skymarshal wrote: Hi Chuck, 2005/12/17, Chuck Bunn [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: What are you codec and dmtfmode settings in sip.conf and in the sip phone settings. I use gsm. If you dmtfmode is set to 'inband' and you are using anything other than ulaw or alaw codec it wont work. I changed the settings and tried: ---cut--- exten = 2000,1,Answer() exten = 2000,2,Wait(1) exten = 2000,3,Playback(hello-world) exten = 2000,4,Hangup() ---cut--- Same problem. Sometimes it works but most of the times it doesn't. Also since your hear the phone sometimes you may be experiencing QOS issues on your network. Of course it could be a QOS problem. But should I hear at least something? cu ES ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Strange problem with sjphone and 1.2.1
Hi, Rich I stand corrected you are absolutely right - see http://www.voip-info.org/wiki-Asterisk+config+sip.conf The following appears on the page: Please note * Asterisk does not yet support SIP over TCP. It only supports SIP http://www.voip-info.org/wiki/view/SIP over UDP. * For Grandstream http://www.voip-info.org/wiki/view/Grandstream phones: set *dtmfmode=info* * Asterisk uses the incoming RTP http://www.voip-info.org/wiki/view/RTP Stream as a timing source for sending its outgoing Stream. If the incoming stream is interrupted due to silence suppression then musiconhold will be choppy. So in conclusion, you cannot use silence suppression. *Make sure ALL SIP phones have disabled silence suppression.* There is a solution for the silence suppression problem, see bug 5374 http://bugs.digium.com/view.php?id=5374 for details. Thanks Rich Adamson wrote: I don't believe asterisk has any sip tcp support. Its all udp. Hi, Something else I should mention. Sip uses UDP and TCP packets. TCP packets are used if there is congestion on the network. I am unclear about what mechanism causes sip to switch between UDP and TCP but I believe it is controllable - I believe It would be easier to use QOS though. If UDP is used that packets could time out and you would never know it since UDP is dumb and has no packet loss recovery mechanism. What is the topology of your network. Is the Asterisk box and the client on the same backbone and switch? Thanks Evil Skymarshal wrote: Hi Chuck, 2005/12/17, Chuck Bunn [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]: What are you codec and dmtfmode settings in sip.conf and in the sip phone settings. I use gsm. If you dmtfmode is set to 'inband' and you are using anything other than ulaw or alaw codec it wont work. I changed the settings and tried: ---cut--- exten = 2000,1,Answer() exten = 2000,2,Wait(1) exten = 2000,3,Playback(hello-world) exten = 2000,4,Hangup() ---cut--- Same problem. Sometimes it works but most of the times it doesn't. Also since your hear the phone sometimes you may be experiencing QOS issues on your network. Of course it could be a QOS problem. But should I hear at least something? cu ES ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.1/206 - Release Date: 12/16/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best way to automatically stop and start Asterisk nightly
Hi, Thanks for the input. I will try your suggestions. By slowing down the server takes longer and longer to respond to prompts such as retrieving voice mail. I am recompiling my install this weekend as I have had a continued problem with logs (see other post) and this might be related to the problem. I will use your command to see if 'asterisk.pid' inflates over time... Thanks Again Tzafrir Cohen wrote: On Wed, Dec 14, 2005 at 08:49:06AM -0700, Chuck Bunn wrote: Hi, I am planning on restarting asterisk nightly as I seem to be experiencing some sort of memory leak (Asterisk slows down over time). This is not an indication of a memory leak. The size of the asterisk process: ps `cat /var/run/asterisk/asterisk.pid` -o vsize -o rss Do those inflate over time? I have reviewed the Asterisk suggestions for management and one item is the routine rebooting of Asterisk. Since I have Asterisk 1.2.1 what is the recommend way to implement an automatic stop and start of asterisk asterisk -rx 'restart now' from a daily cron job? Mind you, this is a bad patch and *NOT A FIX*. (there are changes in 1.2 with reload and restart) and is this enough or should I restart the hardware as well?? If you suspect a user-space memory leak than restarting the application should free that memory. BTW: what do you mean by slow down? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] '#' (fast foward) and '*' (Rewind) not working in VoicemailMain
Hi, '#' (fast forward) and '*' (Rewind) not working in VoicemailMain with Asterisk 1.2.1 Do I have to do something in the dialplan to make this work? I have '##' set as a blind transfer and '*2' set as a attended transfer in features.conf. Per the Wiki Voicemailmain has the following settings: * *1* Read voicemail messages o *3* Advanced options (with option to reply; introduced in Asterisk CVS Head April 28, 2004 with 'enhanced voicemail') + *1* Reply + *2* Call back(1) + *3* Envelope + *4* Outgoing call(1) o *4* Play previous message o *5* Repeat current message o *6* Play next message o *7* Delete current message o *8* Forward message to another mailbox o *9* Save message in a folder o *** Help; during msg playback: Rewind o *#* Exit; during msg playback: Skip forward * *2* Change folders * *0* Mailbox options o *1* Record your unavailable message o *2* Record your busy message o *3* Record your name o *4* Record your temporary message (new in Asterisk v1.2) o *5* Change your password o *** Return to the main menu * *** Help * *#* Exit * After recording a message (incoming message, busy/unavail greeting, or name) o 1 - Accept o 2 - Review o 3 - Re-record o 0 - Reach operator(1) (not available when recording greetings/name) Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Blind transferred user does not hear phone ring while waiting for phone to be picked up.
Hi,
Please excuse the double post but I am about to report this as a bug and
I want to verify that others are having the same problem. Also I have
seen numerous bugs reported that are not bugs but just misconfiguration,
etc. and I do not want to burden the developers with a frivolus bug
report if the problem is mine. I have found several postings addressing
this issue but no solution. I have done a partial work around but I do
not like the results. Here is the problem - when I blind transfer a user
the transferred user does not here the phone ringing despite adding the
'r' option to the Dial statement (I will provide all of my files in a
moment..). I have also tried the dial statement without the 'r' option
and I get the same results. If I place a the 'm' option in the dial
statement the transferred user does here musiconhold but this also means
that users doing inter office calls hear musiconhold when calling one
another user instead of ringing (thus my work around that is not
desirable). I also am using a macro to handle dialing and voicemail and
perhaps there is a problem here. In my menus I created a separate macro
that does use the 'm' option as it does seem appropriate here. There is
nothing in the CLI output that appears to show a problem so that further
confuses the issue. Here are my files:
extensions.conf
[general]
#include macros.incl
#include incoming-home.incl
#include extensions-home.incl
#include phrase.incl
#include menu.incl
#include outgoing.incl
[globals]
OUTBOUNDTRUNK=Zap/g1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
*extensions-hone.incl
[extensions-home]
;Operator queue, Operator Console, and Receptionist Phone
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15)
exten = s,5,Queue(extensions-home|tr|||20)
exten = s,6,Goto(mainmenu,s,1)
include = mainmenu
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _590,1,Macro(novmail,${EXTEN},ZAP/3)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain(@extensions-home)
;Agent Login
exten = 801,1,AgentCallbackLogin(||@extensions-home)
;Recording Interface
exten = 820,1,Goto(phrase-menu,s,1)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;Music on Hold
exten = 870,1,Answer
exten = 870,2,SetMusicOnHold(default)
exten = 870,3,WaitMusicOnHold(420)
exten = 870,4,Hangup
macros.incl
[macro-stdexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttrw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
[macro-menuexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttmw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
[macro-novmail]
exten = s,1,Dial(${ARG2},20,Ttrw)
exten = s,2,Playback(thank-you-for-callinggoodbye)
exten = s,3,Hangup
exten = s,102,Playback(thank-you-for-callinggoodbye)
exten = s,103,Hangup
menu.incl
[mainmenu]
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15)
exten = s,5,Background(custom/welcome-main)
exten = 2,1,Goto(spa,s,1)
exten = 3,1,Goto(ageless,s,1)
exten = 4,1,Directory(extensions-home,extensions-home,f)
exten = 5,1,Directory(extensions-home,extensions-home)
exten = t,1,Playback(please-try-again)
exten = t,2,Goto(mainmenu,s,1)
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(mainmenu,s,1)
exten = 0,1,Goto(operator,s,1)
[operator]
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15)
exten = s,5,Background(custom/operator)
exten = s,6,Macro(menuexten,300,SIP/300)
exten = t,1,Playback(please-try-again)
exten
[Asterisk-Users] Best way to automatically stop and start Asterisk nightly
Hi, I am planning on restarting asterisk nightly as I seem to be experiencing some sort of memory leak (Asterisk slows down over time). I have reviewed the Asterisk suggestions for management and one item is the routine rebooting of Asterisk. Since I have Asterisk 1.2.1 what is the recommend way to implement an automatic stop and start of asterisk (there are changes in 1.2 with reload and restart) and is this enough or should I restart the hardware as well?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blind transferred user does not hear phone ring while waiting for phone to be picked up.
OOPs I forgot to mention I am using Asterisk 1.2.1 and I had the same
problem with 1.0.9 and 1.2.0
Chuck Bunn wrote:
Hi,
Please excuse the double post but I am about to report this as a bug
and I want to verify that others are having the same problem. Also I
have seen numerous bugs reported that are not bugs but just
misconfiguration, etc. and I do not want to burden the developers with
a frivolus bug report if the problem is mine. I have found several
postings addressing this issue but no solution. I have done a partial
work around but I do not like the results. Here is the problem - when
I blind transfer a user the transferred user does not here the phone
ringing despite adding the 'r' option to the Dial statement (I will
provide all of my files in a moment..). I have also tried the dial
statement without the 'r' option and I get the same results. If I
place a the 'm' option in the dial statement the transferred user does
here musiconhold but this also means that users doing inter office
calls hear musiconhold when calling one another user instead of
ringing (thus my work around that is not desirable). I also am using a
macro to handle dialing and voicemail and perhaps there is a problem
here. In my menus I created a separate macro that does use the 'm'
option as it does seem appropriate here. There is nothing in the CLI
output that appears to show a problem so that further confuses the
issue. Here are my files:
extensions.conf
[general]
#include macros.incl
#include incoming-home.incl
#include extensions-home.incl
#include phrase.incl
#include menu.incl
#include outgoing.incl
[globals]
OUTBOUNDTRUNK=Zap/g1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
*extensions-hone.incl
[extensions-home]
;Operator queue, Operator Console, and Receptionist Phone
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15)
exten = s,5,Queue(extensions-home|tr|||20)
exten = s,6,Goto(mainmenu,s,1)
include = mainmenu
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _590,1,Macro(novmail,${EXTEN},ZAP/3)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain(@extensions-home)
;Agent Login
exten = 801,1,AgentCallbackLogin(||@extensions-home)
;Recording Interface
exten = 820,1,Goto(phrase-menu,s,1)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;Music on Hold
exten = 870,1,Answer
exten = 870,2,SetMusicOnHold(default)
exten = 870,3,WaitMusicOnHold(420)
exten = 870,4,Hangup
macros.incl
[macro-stdexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttrw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
[macro-menuexten]
exten = s,1,Set(DYNAMIC_FEATURES=automon)
exten = s,2,Dial(${ARG2},20,Ttmw)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail([EMAIL PROTECTED])
exten = s-NOANSWER,2,Playback(thank-you-for-callinggoodbye)
exten = s-NOANSWER,3,Hangup
exten = s-BUSY,1,Voicemail([EMAIL PROTECTED])
exten = s-BUSY,2,Playback(thank-you-for-callinggoodbye)
exten = s-BUSY,3,Hangup
exten = s-CHANUNAVAIL,1,Voicemail([EMAIL PROTECTED])
exten = s-CHANUNAVAIL,2,Playback(thank-you-for-callinggoodbye)
exten = s-CHANUNAVAIL,3,Hangup
exten = _s-.,1,Goto(s-NOANSWER,1)
[macro-novmail]
exten = s,1,Dial(${ARG2},20,Ttrw)
exten = s,2,Playback(thank-you-for-callinggoodbye)
exten = s,3,Hangup
exten = s,102,Playback(thank-you-for-callinggoodbye)
exten = s,103,Hangup
menu.incl
[mainmenu]
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15)
exten = s,5,Background(custom/welcome-main)
exten = 2,1,Goto(spa,s,1)
exten = 3,1,Goto(ageless,s,1)
exten = 4,1,Directory(extensions-home,extensions-home,f)
exten = 5,1,Directory(extensions-home,extensions-home)
exten = t,1,Playback(please-try-again)
exten = t,2,Goto(mainmenu,s,1)
exten = i,1,Playback(pbx-invalid)
exten = i,2,Goto(mainmenu,s,1)
exten = 0,1,Goto(operator,s,1)
[operator]
exten = s,1,Answer()
exten = s,2,SetMusicOnHold(default)
exten = s,3,DigitTimeout(5)
exten = s,4,ResponseTimeout(15
[Asterisk-Users] What would prevent logs from being recreated if they are deleted?
Hi,
Please excuse the cross post but these seems to be one of those issues
that may be answered by a developer or someone with direct
administrative knowledge of the deep workings of Asterisk. I have
deleted my log files expecting them to be recreated by Asterisk 1.2 but
nothing happens after a reboot or any of the log commands in the CLI
interface ('logger restart', 'logger rotate', 'logger show channels').
What would prevent these files from being recreated and what mechanism
recreates these logs after being deleted?
Thanks
___
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Re: [Asterisk-Users] Re: sound problem in X-Lite phone with asterisk server
Hi, Two things does your codec set in X-lite match what is set in the sip file and have you rebooted since setting up music on hold. I should also ask if ran a make and make install in the asterisk-addons directory, this installs a mp3 player (among other things) in Asterisk 1.2? Vipul Patel wrote: Hi all I am a newbie to the asterisk. I just installed asterisk server and two X-Lite softphones. I allready configured sip.conf and extension.conf. Now when i call from one softphone to other , sip signaling is going perfect. Both phone are in ringing mode. But i can't able to hear ring. When i pickup call, there is not any sound at all. The asterisk server give following output during call: Dec 5 12:49:57 NOTICE[1931]: res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?! Dec 5 12:49:57 WARNING[1931]: res_musiconhold.c:205 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3' Dec 5 12:49:57 WARNING[1931]: res_musiconhold.c:278 monmp3thread: unable to spawn mp3player Can any one pls tell me where i am going wrong. Thanks Vipul ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.11/191 - Release Date: 12/2/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer/take call to/from other phone
Hi, Push the '#' key followed by the extension for a blind transfer. Thanks Denny Schierz wrote: hi, my asterisk 1.2 works very well with ISDN and SIP, but how can i transfer calls from my phone (for example ISDN, MSN 400) to the another phone (ISDN, MSN 401)? Or when the phone 401 rings, but my boss is not there, how can i take the phonecall from 401 to 400? Do i need special options in my extensions.conf or is that feature from the isdn phone? cu denny This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Restore logging functionality...
Hi, A while back I made the stupid mistake of deleting my log files 'full' and 'messages' for asterisk. I recreated the files by 'touch' filename and I have gone into the Asterisk CLI and tried both 'logger restart' and 'logger rotate' but the logs still show nothing. I run 'logger show channels' and the output below shows up. I have recompiled Asterisk 1.2 and still the logs do not show up. I am getting data into the 'queue_log' and the 'events' logs however so I know logger is running. Any suggestions to fix this??? CLI output tomato*CLI logger show channels Channel Type StatusConfiguration --- --- tomato*CLI tomato*CLI Output from /var/log/asterisk directory [EMAIL PROTECTED] asterisk]# ls -la total 140 drwxr-xr-x 4 asterisk asterisk 4096 Dec 5 08:22 . drwxr-xr-x 11 root root 4096 Dec 4 04:03 .. drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-csv drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-custom -rw-r--r-- 1 root root 0 Dec 5 08:22 event_log -rw-r--r-- 1 asterisk asterisk 1186 Nov 12 07:43 event_log.0 -rw-r--r-- 1 root root 0 Nov 17 06:41 event_log.1 -rw-r--r-- 1 root root 0 Nov 17 06:45 event_log.2 -rw-r--r-- 1 root root 0 Nov 18 06:38 event_log.3 -rw-r--r-- 1 root root 0 Nov 18 06:37 full -rw-r--r-- 1 root root 0 Nov 18 06:37 messages -rw-r--r-- 1 asterisk asterisk 111711 Dec 5 08:23 queue_log *** Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer/take call to/from other phone
Hi, To pick up another persons phone that is ringing dial '*8' followed by their extension. To do an attended transfer dial '*2' followed by the extension... Hope that helps Denny Schierz wrote: hi, Quoting Chuck Bunn [EMAIL PROTECTED]: Push the '#' key followed by the extension for a blind transfer. absolut perfect, thanks :-) .Is there also a shortcut, to take a phone call from other phones to me? cu denny This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Restore logging functionality...
Hi, I deleted the files and ran 'logger restart' - no dice, 'logger rotate' - no dice, 'reload' - no dice, 'restart gracefully' - no dice. Logs are not recreated??? Any other ideas Thanks Marco Supino wrote: The user running asterisk doesnt have permission to write on the files, delete them , and asterisk will recreate them as user asterisk, or chown them, or change them to 777 best of all, delete them! Marco. Chuck Bunn wrote: Hi, A while back I made the stupid mistake of deleting my log files 'full' and 'messages' for asterisk. I recreated the files by 'touch' filename and I have gone into the Asterisk CLI and tried both 'logger restart' and 'logger rotate' but the logs still show nothing. I run 'logger show channels' and the output below shows up. I have recompiled Asterisk 1.2 and still the logs do not show up. I am getting data into the 'queue_log' and the 'events' logs however so I know logger is running. Any suggestions to fix this??? CLI output tomato*CLI logger show channels Channel Type StatusConfiguration --- --- tomato*CLI tomato*CLI Output from /var/log/asterisk directory [EMAIL PROTECTED] asterisk]# ls -la total 140 drwxr-xr-x 4 asterisk asterisk 4096 Dec 5 08:22 . drwxr-xr-x 11 root root 4096 Dec 4 04:03 .. drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-csv drwxr-xr-x 2 asterisk asterisk 4096 Nov 8 21:39 cdr-custom -rw-r--r-- 1 root root 0 Dec 5 08:22 event_log -rw-r--r-- 1 asterisk asterisk 1186 Nov 12 07:43 event_log.0 -rw-r--r-- 1 root root 0 Nov 17 06:41 event_log.1 -rw-r--r-- 1 root root 0 Nov 17 06:45 event_log.2 -rw-r--r-- 1 root root 0 Nov 18 06:38 event_log.3 -rw-r--r-- 1 root root 0 Nov 18 06:37 full -rw-r--r-- 1 root root 0 Nov 18 06:37 messages -rw-r--r-- 1 asterisk asterisk 111711 Dec 5 08:23 queue_log *** Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Anyone know anything about the new Linksys One product - does it use Asterisk?
Hi, Does anyone have any details about the Linksys one product that was just announced? Does it use Asterisk? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why does musiconhold.conf changes require a reboot?
Hi, Why do changes to musiconhold.conf require a reboot. Also if I put mp3's into the /var/lib/asterisk/mohmp3 directory will the be played if I use the -r option? Using Asterisk 1.2 and have run the make config in the /usr/src/asterisk-addons directory. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I escape queue with a '*'?
Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Extension Manual
Hi, I understand that I must set up extensions myself but in the release notes for Asterisk 1.2 it specifically states that multiple digit extensions can be used in the exit context of a queue. Prior to version 1.2 only a single digit would would when exiting from a queue. I have tried setting up the '*' in a exit context for a queue but it does not seem to work. I can get a single digit to work without a problem. I have not tried multiple digits. I figured since multiple digits worked that perhaps the *' or '#' might work as well... Thanks Yair Hakak wrote: what is your question? you must set up extensions yourself in extensions.conf...you can set the extensions to whatever you want (say, if you are replacing an existing PBX and want the users to have the same extensions). -yair On 12/3/05, Vladimir Montealegre [EMAIL PROTECTED] wrote: *411 Directory *43 Echo Test *60 Time *61 Weather *62 Schedule wakeup call *65 festival test (your extension is XXX) *70 Activate Call Waiting (deactivated by default) *71 Deactivate Call Waiting *72 Call Forwarding System *73 Disable Call Forwarding *77 IVR Recording *78 Enable Do-Not-Disturb *79 Disable Do-Not-Disturb *90 Call Forward on Busy *91 Disable Call Forward on Busy *97 Message Center (does no ask for extension) *98 Enter Message Center *99 Playback IVR Recording *666 Test Fax Simulate incoming call - Original Message - From: Vladimir Montealegre [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 03, 2005 2:34 PM Subject: [Asterisk-Users] Extension Manual in wath link or page is the * commands for the phone extensions?? example *79 is for on or off the extension ?? Thanks again in advance - Original Message - From: Chuck Bunn [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, December 03, 2005 2:06 PM Subject: [Asterisk-Users] Can I escape queue with a '*'? Hi, Can I escape a call queue by pressing a '*' or do I have to use a digit?? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Converted mp3 files to raw for musiconhold and still does not work...
Hi, I setup music on hold as directed for Asterisk 1.2 but still no music on hold. Any ideas what I did wrong. I see it start in the CLI but then it immediately stops?? I also see the Hangup occur 20 seconds later as it should according to WitMusicOnHold(20). I used a test setup suggested in the wiki... ** CLI Output Spawn extension (longdistance, 870, 4) exited non-zero on 'SIP/499-39ff' -- Executing Answer(SIP/499-206c, ) in new stack -- Executing SetMusicOnHold(SIP/499-206c, default) in new stack -- Executing WaitMusicOnHold(SIP/499-206c, 20) in new stack -- Started music on hold, class 'default', on channel 'SIP/499-206c' -- Stopped music on hold on SIP/499-206c == Spawn extension (longdistance, 870, 3) exited non-zero on 'SIP/499-206c' *** ;Music on Hold exten = 870,1,Answer exten = 870,2,SetMusicOnHold(default) exten = 870,3,WaitMusicOnHold(20) exten = 870,4,Hangup * musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/mohmp3 * zapata.conf [trunkgroups] [channels] musiconhold=default echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=10.0 txgain=3.0 usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=no transfer=no immediate=no faxdetect=both context=incoming-home signalling=fxs_ks group=1 channel = 1,2 context=local signalling=fxo_ks group=2 channel = 3 context=longdistance signalling=fxo_ks group=3 channel = 4 ** Output of /var/lib/asterisk/mohmp3 directory [EMAIL PROTECTED] mohmp3]# ls -la total 107772 drwxr-xr-x 2 asterisk asterisk 4096 Dec 3 20:33 . drwxr-xr-x 9 asterisk asterisk 4096 Nov 11 10:18 .. -rw-r--r-- 1 root root 1939812 Nov 11 10:24 fpm-calm-river.mp3 -rw-r--r-- 1 root root 2582496 Nov 11 10:24 fpm-sunshine.mp3 -rw-r--r-- 1 root root 2217563 Nov 11 10:24 fpm-world-mix.mp3 -rw-r--r-- 1 asterisk asterisk 884864 Oct 29 12:39 QuajiroPromo.mp3 -rw-r--r-- 1 asterisk asterisk 835712 Oct 29 12:39 TristeAlegriaPromo.mp3 Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can I get to a menu system while in a queue??
Hi,
Is it possible to get to a menu system while in a call queue. I want
users to be able to hit the '*' and be able to goto a menu system from a
queue if they so desire. I thought the following would do this but no
dice...
*
extension.conf
[general]
#include macros.incl
#include incoming-home.incl
#include extensions-home.incl
#include phrase.incl
#include menu.incl
#include outgoing.incl
[globals]
OUTBOUNDTRUNK=Zap/g1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
*
incoming-home.incl
[incoming-home]
exten = s,1,Goto(extensions-home,s,1)
exten = t,1,Goto(extensions-home,s,1)
exten = i,1,Goto(extensions-home,s,1)
**
extensions-home.incl
[extensions-home]
;Operator queue, Operator Console, and Receptionist Phone
exten = s,1,Answer()
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(15)
exten = s,4,Queue(extensions-home|tr|||20)
exten = s,5,Goto(mainmenu,s,1)
exten = 0,1,Goto(operator,s,1)
exten = *,1,Goto(mainmenu,s,1)
exten = i,1,Goto(mainmenu,s,1)
exten = t,1,Goto(mainmenu,s,1)
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _590,1,Macro(novmail,${EXTEN},ZAP/3)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain
;Agent Login
exten = 801,1,AgentCallbackLogin(||@extensions-home)
;Recording Interface
exten = 820,1,Goto(phrase-menu,s,1)
;Voice Conferencing;
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
menu.incl
[mainmenu]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(15)
exten = s,4,Background(custom/welcome-main)
exten = 1,1,Goto(spa,s,1)
exten = 2,1,Goto(ageless,s,1)
exten = 3,1,Goto(dialbyext,s,1)
exten = 4,1,Directory(extensions-home,extensions-home,f) ;search by
first name
exten = 5,1,Directory(extensions-home,extensions-home) ;search by last name
exten = 0,1,Goto(operator,s,1)
exten = *,1,Goto(mainmenu,s,1)
exten = t,1,Goto(s,1,1)
exten = i,1,Goto(s,1,1)
[operator]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(15)
exten = s,4,Macro(stdexten,300,SIP/300)
exten = 0,1,Goto(operator,s,1)
exten = *,1,Goto(mainmenu,s,1)
exten = t,1,Goto(operator,s,1)
exten = i,1,Goto(operator,s,1)
[spa]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(15)
exten = s,4,Macro(stdexten,500,SIP/500)
exten = 0,1,Goto(operator,s,1)
exten = *,1,Goto(mainmenu,s,1)
exten = t,1,Goto(spa,s,1)
exten = i,1,Goto(spa,s,1)
[ageless]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(15)
exten = s,4,Macro(stdexten,300,SIP/300)
exten = 0,1,Goto(operator,s,1)
exten = *,1,Goto(mainmenu,s,1)
exten = t,1,Goto(ageless,s,1)
exten = i,1,Goto(ageless,s,1)
[dialbyext]
exten = s,1,Answer
exten = s,2,DigitTimeout(5)
exten = s,3,ResponseTimeout(15)
exten = s,4,Background(ext-or-zero)
exten = 0,1,Goto(operator,s,1)
exten = *,1,Goto(mainmenu,s,1)
exten = t,1,Playback(please-try-again)
exten = t,2,Goto(dialbyext,s,1)
exten = i,1,Playback(num-not-in-db)
exten = i,1,Goto(dialbyext,s,1)
include = extensions-home
Thanks
___
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[Asterisk-Users] Context confict question??
Hi, If I have an extension in a context and I have another context with the same extension and I include the second context in the first does this cause a conflict or does Asterisk know that there is a 600 extension in each context [big-business] exten = 600,1,Dial(ZAP/1,20) include = small-business [small-business] exten = 600,1,Dial(ZAP/2,15) Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What kind of extension numbers can be used in the exit context of a queue?
Hi, In Asterisk 1.2 according to the wiki and I quote: It is now possible to use multi-digit extensions in the exit context for a queue (although you should not have overlapping extensions, as there is no digit timeout). This means that the EXITWITHKEY event in queue_log can now contain a key field with more than a single character in it. What is considered a legal extension number in the exit context? In other words can I use a '0' or a '*' as an extension in the exit context for a Queue?? Example exit context given in wiki: extensions.conf: [queue] exten = 129,1,Playback(some_announce) ; Important, see notes exten = 129,2,Queue(example_queue|tT|||300) ;dont set n option until really needed exten = 129,3,Playback(some_announce_after_leaving_queue) exten = 129,4,Voicemail(s1234) * queues.conf: [example_queue] music = default strategy = ringall context = queue-out ; Here we go when the caller presses a single digit, while in the queue timeout = 15 wrapuptime=10 announce-frequency = 30 announce-holdtime = yes joinempty = yes member = Agent/1234 member = Agent/1235 agents.conf: [agents] ackcall=no ; Agent don't has to press # to answer the call musiconhold = default agent = 1234,,Agent1_Name agent = 1235,, Agent2_Name Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Context confict question??
Hi, When you say it has a higher priority what does that mean?? Does that mean that a call to extension 600 always goes to the higher priority unless it is busy? Thanks Andy Kuo wrote: Hi, The one in [big-business] has higher priority than the one in [small-business] Included context has lower priority. Hope this helps. Andy On 12/2/05, *Chuck Bunn* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi, If I have an extension in a context and I have another context with the same extension and I include the second context in the first does this cause a conflict or does Asterisk know that there is a 600 extension in each context [big-business] exten = 600,1,Dial(ZAP/1,20) include = small-business [small-business] exten = 600,1,Dial(ZAP/2,15) Thanks ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.1.362 / Virus Database: 267.13.11/191 - Release Date: 12/2/2005 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] All agent calls going to powered down agent extension?
Hi, Not sure if this is by design or my error... Several agents are logged in. One of the phones was turned off without the agent logging off first. After the phone was powered down all calls routed to the powered off agent and no other phones rang. Is there a way to turn this behavior off. (I want the other agents to still ring when an agent is not logged off and powered down). We have have the Queue set to ringall. queues.conf [general] ;Operator Home [extensions-home] music=default strategy=ringall maxlen=0 context=extensions-home member = Agent/300 member = Agent/301 member = Agent/310 member = Agent/311 member = Agent/312 member = Agent/313 member = Agent/314 member = Agent/499 member = Agent/500 member = Agent/510 member = Agent/511 member = Agent/512 ** agents.conf [agents] wrapuptime=0 musiconhold = default updatecdr=yes recordagentcalls=no ;Operator - Home group=1 agent = 300,300,name1 agent = 301,301,name2 agent = 310,310,name3 agent = 311,311,name4 agent = 312,312,name5 agent = 313,313,name6 agent = 314,314,name7 agent = 399,399,Test agent = 499,499,name8 ** Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comas versus pipe command in AgetCallBackLogin
H, Not sure if this is normal but I thought the coma ',' was replaceable by the pipe command '|' and vice versa? When I used comas instead of the pipe command in AgentCallbackLogin certain SIP phones do not here the operator prompts when calling the agent extension. Is this normal - I thought the pipe command and coma were interchangeable??? Oh and I am using Asterisk 1.2. Works: exten = 801,1,AgentCallbackLogin(||@extensions-home) Does not work: exten = 801,1,AgentCallbackLogin(,,@extensions-home) Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Pasting phrases together....
Hi, Is it possible to paste phrases together and if so how do I separate each phrase. exten = s,4,BackGround(to-compose-a-message,press-1) and exten = s,4,BackGround(to-compose-a-message|press-1) do not work... Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pasting phrases together....
Hi, Yes that does work but in some cases it just seems it would be clearer (also less code) to be able to have them on one line... Thanks Anthony Rodgers wrote: exten = s,4,BackGround(to-compose-a-message) exten = s,5,BackGround(press-1) doesn't work? On Nov 29, 2005, at 3:41 PM, [EMAIL PROTECTED] wrote: Hi, Is it possible to paste phrases together and if so how do I separate each phrase. exten = s,4,BackGround(to-compose-a-message,press-1) and exten = s,4,BackGround(to-compose-a-message|press-1) do not work... Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can 'spandsp' ack as an intermediary between a fax machine and a TDM400P?
Hi, I understand that a fax machine cannot connect through a Digium TDM400p card (FXS connected to fax and FXO connected to a pots line) but can spandsp send and receive faxes as an intermediary between the pots line and the fax machine. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zyxel P2000Wv2 cannot do agent login, SJPhone work just fine?
Hi,
Problem solved sort of. For some reason I cannot get the Zyxel to work
with agentcallbackLogin when the codec is alaw, ulaw or g729 and DTMF is
rfc2833. I had to change the codec to ulaw and DTMF to inband to get it
to work. Which means the voice quality dropped some and I noticed the
echo and jitter control did not work as well, but at least now the
phones can be used to ack as an agent.
Thanks
Chuck Bunn wrote:
Hi,
Okay we have agents logging in to receive calls from a queue. Agents
logging in from a SJPhone (SIP Phone) can dial the login extension and
are asked for their 'username followed by #' and then they are asked
for their 'password followed by #' and then the system asks them what
'extension they are at followed by #'. This works perfectly. When
someone calls in the agents extensions that have logged in ring. When
someone using the Zyxel phone (by the way the latest version is a
great little phone with great clarity) calls into the agent extension
it asks for their extension as before but as soon as the user enters
the extension followed by a # the system hangs up on them, go
figure Here are my files. Oh and logging out of the agent
application works fine from SJPhone.
extensions.conf
[general]
#include macros.incl
[incoming-home]
exten = s,1,Goto(extensions-home,100,1)
exten = t,1,Goto(extensions-home,100,1)
exten = i,1,Goto(extensions-home,100,1)
[extensions-home]
include = parkedcalls
;Operator queue, Operator Console, and Receptionist Phone
exten = 100,1,Answer()
exten = 100,2,Queue(extensions-home|trn|||120)
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = 590,1,Dial(ZAP/3,20)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain
;Agent Login
exten = 801,1,AgentCallbackLogin(,,@extensions-home)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;exten = i,1,Voicemail(s300)
;exten = t,1,Voicemail(s300)
exten = fax,1,Dial(ZAP/4,20)
exten = fax,2,Congestion
exten = fax,102,Congestion
[local]
ignorepat = 9
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion(5)
exten = _9NXX,102,congestion(5)
exten = 911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911
include = extensions-home
[longdistance]
ignorpat = 9
exten = _91NXXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91NXXNXX,2,Congestion(5)
exten = _91NXXNXX,102,congestion(5)
include = local
[globals]
OUTBOUNDTRUNK=Zap/G1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
zapata.conf
[trunkgroups]
[channels]
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
rxgain=14.0
txgain=4.0
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
immediate=no
faxdetect=both
context=incoming-home
signalling=fxs_ks
group=1
channel = 1,2
context=local
signalling=fxo_ks
group=2
channel = 3
context=longdistance
signalling=fxo_ks
group=3
channel = 4
***
queues.conf
[general]
;Operator Home
[extensions-home]
music=default
strategy=ringall
maxlen=0
member = Agent/@1
*
sip.conf
[general]
context=default
srvlookup=yes
;Zyxel - P2000WV2
[300]
context=longdistance
type=friend
username=300
secret=x
callerid=300
nat=no
host=dynamic
mailbox=300
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
;Zyxel - P2000WV2
[301]
context=longdistance
type=friend
username=301
secret=x
callerid=301
nat=no
host=dynamic
mailbox=301
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
.
.
.
.
;SJphone
[310]
context=longdistance
type=friend
username=310
secret=x
callerid=310
qualify=yes
nat=no
host=dynamic
mailbox=310
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
;SJphone
[311]
context=longdistance
type=friend
username=311
secret=x
callerid=311
qualify=yes
nat=no
host=dynamic
mailbox=311
disallow=all
allow=alaw
allow=ulaw
allow=gsm
dtmfmode=rfc2833
...
***
agents.conf
[agents]
wrapuptime=0
musiconhold = default
updatecdr=yes
;Operator - Home
group=1
agent = 300,300,name
agent = 301,301,nam2
agent = 310,310,name3
agent = 311,311,name4
...
***
Zyxel Phone settings
*PHONE SETTINGS*
Default Voice Codec
Speaking Volume(-14~14) Listening Volume(-14~14) RTP Port
Jitter Buffer Small Medium Large Voice Frames per Packet
Small Medium Large DTMF Relay
DTMF Payload(0~127)
**
CLS Output
WHEN IT WORKS
-- Executing AgentCallbackLogin
Re: [Asterisk-Users] Strategy=ringall does not ring all agents.
Hi,
I have now tried other strategies including random and round robin. I am
beginning to think there is some sort of bug with Agent groups? I will
try assigning members to a queue not by their group but individually.
Thanks
Chuck Bunn wrote:
Hi,
In the queue.conf I have set the strategy set to ringall but only the
lowest
agent number ever rings??? A show agents at the CLI shows three agents
logged
in yet only the first agent ever rings. I have my agents in a group,
group 1.
queue.conf
[general]
;Operator Home
[extensions-home]
music=default
strategy=ringall
maxlen=0
context=extensions-home
member = Agent/@1
**
agents.conf
[agents]
wrapuptime=0
musiconhold = default
updatecdr=yes
;Operator - Home
group=1
agent = 300,300,name1
agent = 301,301,name2
agent = 310,310,name3
agent = 311,311,name4
agent = 312,312,name5
agent = 313,313,name6
agent = 314,314,name7
agent = 499,499,name8
;Operator - Spa
agent = 500,500,name9
agent = 510,510,name10
agent = 511,511,name11
agent = 512,512,name12
;Operator - Rest
group=2
agent = 600,600,name13
extensions.conf
[general]
#include macros.incl
[incoming-home]
exten = s,1,Goto(extensions-home,100,1)
exten = t,1,Goto(extensions-home,100,1)
exten = i,1,Goto(extensions-home,100,1)
[extensions-home]
include = parkedcalls
;Operator queue, Operator Console, and Receptionist Phone
exten = 100,1,Answer()
exten = 100,2,Queue(extensions-home|trn|||120)
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = 590,1,Dial(ZAP/3,20)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain
;Agent Login
exten = 801,1,AgentCallbackLogin(,,@extensions-home)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;exten = i,1,Voicemail(s300)
;exten = t,1,Voicemail(s300)
exten = fax,1,Dial(ZAP/4,20)
exten = fax,2,Congestion
exten = fax,102,Congestion
[local]
ignorepat = 9
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion(5)
exten = _9NXX,102,congestion(5)
exten = 911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911
include = extensions-home
[longdistance]
ignorpat = 9
exten = _91NXXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91NXXNXX,2,Congestion(5)
exten = _91NXXNXX,102,congestion(5)
include = local
[globals]
OUTBOUNDTRUNK=Zap/G1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
CLI Output
Starting simple switch on 'Zap/1-1'
-- Executing Goto(Zap/1-1, extensions-home|100|1) in new stack
-- Goto (extensions-home,100,1)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Queue(Zap/1-1, extensions-home|trn|||120) in new
stack
-- outgoing agentcall, to agent '300', on
'Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,1'
-- Called Agent/@1
-- Executing Macro(Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2,
stdexten|300|SIP/300) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2,
SIP/300|20) in new
stack
-- Called 300
-- SIP/300-00ed is ringing
-- Agent/300 is ringing
-- SIP/300-00ed answered Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2
-- Agent/300 answered Zap/1-1
...
Thanks
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Re: [Asterisk-Users] Strategy=ringall does not ring all agents.
Hi,
Okay I ran a test and if I define each member in a queue individually it
works. It looks like there is a bug with Agent grouping, but before I
report this as a bug I would like to know if anyone has queues working
with agent groups with Asterisk 1.2.
new queues.conf file
[general]
;Operator Home
[extensions-home]
music=default
strategy=ringall
maxlen=0
context=extensions-home
member = Agent/300
member = Agent/301
member = Agent/310
member = Agent/311
member = Agent/312
member = Agent/313
member = Agent/314
member = Agent/499
member = Agent/500
member = Agent/510
member = Agent/511
member = Agent/512
***
Thanks
Chuck Bunn wrote:
Hi,
I have now tried other strategies including random and round robin. I
am beginning to think there is some sort of bug with Agent groups? I
will try assigning members to a queue not by their group but
individually.
Thanks
Chuck Bunn wrote:
Hi,
In the queue.conf I have set the strategy set to ringall but only the
lowest
agent number ever rings??? A show agents at the CLI shows three
agents logged
in yet only the first agent ever rings. I have my agents in a group,
group 1.
queue.conf
[general]
;Operator Home
[extensions-home]
music=default
strategy=ringall
maxlen=0
context=extensions-home
member = Agent/@1
**
agents.conf
[agents]
wrapuptime=0
musiconhold = default
updatecdr=yes
;Operator - Home
group=1
agent = 300,300,name1
agent = 301,301,name2
agent = 310,310,name3
agent = 311,311,name4
agent = 312,312,name5
agent = 313,313,name6
agent = 314,314,name7
agent = 499,499,name8
;Operator - Spa
agent = 500,500,name9
agent = 510,510,name10
agent = 511,511,name11
agent = 512,512,name12
;Operator - Rest
group=2
agent = 600,600,name13
extensions.conf
[general]
#include macros.incl
[incoming-home]
exten = s,1,Goto(extensions-home,100,1)
exten = t,1,Goto(extensions-home,100,1)
exten = i,1,Goto(extensions-home,100,1)
[extensions-home]
include = parkedcalls
;Operator queue, Operator Console, and Receptionist Phone
exten = 100,1,Answer()
exten = 100,2,Queue(extensions-home|trn|||120)
;Office Personnel
exten = _30[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _31[0-4],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _399,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _40[0-1],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _499,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Spa Personnel
exten = _500,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = _51[0-2],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
exten = 590,1,Dial(ZAP/3,20)
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain
;Agent Login
exten = 801,1,AgentCallbackLogin(,,@extensions-home)
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
;exten = i,1,Voicemail(s300)
;exten = t,1,Voicemail(s300)
exten = fax,1,Dial(ZAP/4,20)
exten = fax,2,Congestion
exten = fax,102,Congestion
[local]
ignorepat = 9
exten = _9NXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _9NXX,2,Congestion(5)
exten = _9NXX,102,congestion(5)
exten = 911,1,Dial(${OUTBOUNDTRUNK}/911)
exten = 9911,1,Dial(${OUTBOUNDTRUNK}/911
include = extensions-home
[longdistance]
ignorpat = 9
exten = _91NXXNXX,1,Dial(${OUTBOUNDTRUNK}/${EXTEN:1})
exten = _91NXXNXX,2,Congestion(5)
exten = _91NXXNXX,102,congestion(5)
include = local
[globals]
OUTBOUNDTRUNK=Zap/G1
PSTN1=Zap/1
PSTN2=Zap/2
PHONE1=Zap/3
PHONE2=Zap/4
CLI Output
Starting simple switch on 'Zap/1-1'
-- Executing Goto(Zap/1-1, extensions-home|100|1) in new stack
-- Goto (extensions-home,100,1)
-- Executing Answer(Zap/1-1, ) in new stack
-- Executing Queue(Zap/1-1, extensions-home|trn|||120) in new
stack
-- outgoing agentcall, to agent '300', on
'Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,1'
-- Called Agent/@1
-- Executing Macro(Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2,
stdexten|300|SIP/300) in new stack
-- Executing Dial(Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2,
SIP/300|20) in new
stack
-- Called 300
-- SIP/300-00ed is ringing
-- Agent/300 is ringing
-- SIP/300-00ed answered Local/[EMAIL PROTECTED]
javascript:open_compose_win('to=300%40extensions-home-b560thismailbox=sent-mail');,2
-- Agent/300 answered Zap/1-1
...
Thanks
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[Asterisk-Users] Using transfer button in SJPhone
Hi, Does anyone know how to implement the tranfer feature (button) on the SJPhone in extension.conf Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call transfer with phones that cannot handle more than one line
Hi, Does anyone have a sample config for phones (like the Zyxel P2000wv2) that cannot handle more than one line. I have tried using # followed by the extension and nothing happens??? I have parking setup but for some reason we cannot retrieve the parked call. I call the user who the call is transfered to and they dial the parked extension in this case between 701 and 710 and nothing happens. I am just using the default feature file. *** features.conf [general] parkext = 700 parkpos = 701-720 context = parkedcalls parkingtime = 45 transferdigittimeout = 3 courtesytone = beep xfersound = beep xferfailsound = beeperr ;adsipark = yes findslot = next pickupexten = *8 featuredigittimeout = 500 [featuremap] blindxfer = #1 disconnect = *0 ;automon = *1 atxfer = *2 ** Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
