Re: [asterisk-users] Flashing Red Lights in TE420 (5th Gen) Card
Thanks Russ for your response. Finally found time to do more test on this thread. I uninstalled DAHDI-complete 2.9.1.1 and installed an older DAHDI version 2.4.1 It worked! Both READMEs said Digium TE420: PCI-Express quad-port T1/E1/J1 should work. But it seems that 5th gen TE420 (see below) only works with older DAHDI version. 04:08.0 Communication controller: Digium, Inc. Wildcard TE420 quad-span T1/E1/J1 card 3.3V (PCI-Express) (5th gen) (rev 02) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Russ Meyerriecks Sent: Saturday, 31 May 2014 3:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Flashing Red Lights in TE420 (5th Gen) Card On Fri, May 30, 2014 at 4:07 AM, Lee, John (Sydney) john@compuware.com wrote: Even without plugging in the ISDN into span 1, all 4 spans are flashing red. Blinking red led is normal for spans which have been configured, but are receiving no signal. I might try plugging up a physical loopback plug to the port to rule out a bad incoming signal. wct4xxp :04:08.0: TE4XXP: Span 1 configured for CCS/HDB3/CRC4 wct4xxp :04:08.0: Span 2 configured for ESF/B8ZS wct4xxp :04:08.0: All spans in alarm : No validspan to source RCLK from This looks like a normal startup for mixed-mode configuration with nothing connected to the ports. I might try setting all spans to T1 or all spans to E1 and plugging one or the other back up to test the connections. -- Russ Meyerriecks Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA direct: +1 256-428-6025 Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Flashing Red Lights in TE420 (5th Gen) Card
I have the following software installed in a Centos Box with a TE420 (5th Gen) card. . Centos 6.5 64-bit . Asterisk 1.4.22 . dahdi-linux-complete-2.9.1.1+2.9.1 . libpri-1.4.14. Even without plugging in the ISDN into span 1, all 4 spans are flashing red. Plugging an E1 into span 1 makes no difference. My system.conf is just simply: loadzone=au defaultzone=au span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-21 unused=22-31 dchan=16 span=2,0,0,esf,b8zs fxols=32-55 Below is the dmesg. dahdi: Version: 2.9.1.1 dahdi: Telephony Interface Registered on major 196 wct4xxp :04:08.0: PCI INT A - GSI 17 (level, low) - IRQ 17 wct4xxp :04:08.0: 5th gen card with initial latency of 2 and 1 ms per IRQ wct4xxp :04:08.0: Firmware Version: c01a016d wct4xxp :04:08.0: FALC Framer Version: 3.1 wct4xxp :04:08.0: Not prepped yet! wct4xxp :04:08.0: Found a Wildcard: Wildcard TE420 (5th Gen) wct4xxp :04:08.0: firmware: requesting dahdi-fw-oct6114-128.bin VPM450: echo cancellation for 128 channels wct4xxp :04:08.0: VPM450: hardware DTMF disabled. wct4xxp :04:08.0: VPM450: Present and operational servicing 4 span(s) p4p1: no IPv6 routers present dahdi_devices pci::04:08.0: local span 1 is already assigned span 1 dahdi_devices pci::04:08.0: local span 2 is already assigned span 2 dahdi_devices pci::04:08.0: local span 3 is already assigned span 3 dahdi_devices pci::04:08.0: local span 4 is already assigned span 4 wct4xxp :04:08.0: TE4XXP: Span 1 configured for CCS/HDB3/CRC4 wct4xxp :04:08.0: RCLK source set to span 1 wct4xxp :04:08.0: Recovered timing mode, RCLK set to span 1 wct4xxp :04:08.0: SPAN 1: Primary Sync Source wct4xxp :04:08.0: Span 2 configured for ESF/B8ZS wct4xxp :04:08.0: RCLK source set to span 1 wct4xxp :04:08.0: Recovered timing mode, RCLK set to span 1 wct4xxp :04:08.0: Span 3 configured for ESF/B8ZS wct4xxp :04:08.0: RCLK source set to span 1 wct4xxp :04:08.0: Recovered timing mode, RCLK set to span 1 wct4xxp :04:08.0: Span 4 configured for ESF/B8ZS wct4xxp :04:08.0: RCLK source set to span 1 wct4xxp :04:08.0: Recovered timing mode, RCLK set to span 1 wct4xxp :04:08.0: Setting yellow alarm span 1 wct4xxp :04:08.0: RCLK source set to span 2 wct4xxp :04:08.0: System timing mode, RCLK set to span 2 wct4xxp :04:08.0: Setting yellow alarm span 2 wct4xxp :04:08.0: RCLK source set to span 3 wct4xxp :04:08.0: System timing mode, RCLK set to span 3 wct4xxp :04:08.0: Setting yellow alarm span 3 wct4xxp :04:08.0: RCLK source set to span 4 wct4xxp :04:08.0: System timing mode, RCLK set to span 4 wct4xxp :04:08.0: Setting yellow alarm span 4 wct4xxp :04:08.0: All spans in alarm : No validspan to source RCLK from wct4xxp :04:08.0: RCLK source set to span 1 Any thoughts? The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Kernel and DAHDI
Hi, I have noticed it for a while but I just thought about confirming this with the Asterisk community. As the compilation of DAHDI will need to reference Kernel-devel, does it mean that after DAHDI is installed, we should not yum update kernel because it will affect the operation of DAHDI? Thanks. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Old Asterisk Release wanting to upgrade ...
Thanks Johan. I think I will stick with 1.4.x and DAHDI. Although it is a unsupported release, I never had any problems with them. Some machines have never been rebooted for 5+ years. I am a bit scared of going to 11. I have written a lot of AEL2 script in Asterisk 1.4.x and I am not sure if it will still run in 11. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Johan Wilfer Sent: Tuesday, 15 April 2014 7:03 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Old Asterisk Release wanting to upgrade ... 2014-04-15 10:37, Lee, John (Sydney) skrev: Hello, I have been running Asterisk for the past 5+ years on RedHat and I never upgraded it before. All my Asterisk software is of the following release: 1) Asterisk 1.4.21.2 2) Libpri-1.4.4 3) Zaptel-1.4.11 I would like to move the OS to CentOS and then I thought I can at the same time ponder about upgrading Asterisk releases. However, I am bewildered by the myriad of different releases like 1.6, 1.8, 10.x, 11.x, 12.x, 13.x Can someone please give me some advice as to what release I should upgrade? Or should I just stick to 1.4.x and just upgrade DAHDI? Thanks. Regards, John Lee The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. 1.4, and 1.6-series have no support anymore. 1.8 is an LTS and have support currently, but this is also true for 11 and asterisk 11 will be supported longer. You have the full list here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Versions I would go for Asterisk 11 in your case. You will have to think of it more like a migration than an upgrade thought, as a lot has happened since asterisk 1.4. On a side-note, I still run some old installations with a current Dahdi + Asterisk 1.4.44 and they work great together. There is the non-support catch however. Good luck! -- Johan Wilfer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Old Asterisk Release wanting to upgrade ...
Hello, I have been running Asterisk for the past 5+ years on RedHat and I never upgraded it before. All my Asterisk software is of the following release: 1) Asterisk 1.4.21.2 2) Libpri-1.4.4 3) Zaptel-1.4.11 I would like to move the OS to CentOS and then I thought I can at the same time ponder about upgrading Asterisk releases. However, I am bewildered by the myriad of different releases like 1.6, 1.8, 10.x, 11.x, 12.x, 13.x Can someone please give me some advice as to what release I should upgrade? Or should I just stick to 1.4.x and just upgrade DAHDI? Thanks. Regards, John Lee The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Function not Registered??
Thanks very much Mark for pointing that out to me. Before, I have always been coding using DEVSTATE until my colleague downloaded a new version of func_devstate.c which began to use DEVICE_STATE. So, my old AEL2 script will need to be changed. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark Wiater Sent: Saturday, 26 May 2012 5:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Function not Registered?? On 5/25/2012 3:18 AM, Lee, John (Sydney) said: -- Executing [*1223*1**1900@incoming:78] Set(SIP/1900-08ee1da8, DEVSTATE(Custom:cfalw1900)=INUSE) in new stack I use 'Set(DEVICE_STATE(Custom:var)=BUSY)' in my 1.4 dialplans to set device state. mark -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Function not Registered??
Hi all, I am running the same Asterisk 1.4.21.2 with the same configuration on all the servers in the region. I got this function called func_devstate which I use to control the lights of the Polycom phones. This module works well for all the Asterisk servers except this one. To get it to work, I basically compile this module together with the others and there is no need to explicitly load it in modules.conf. The problem is when my script uses function DEVSTATE, the Asterisk console shows that it is not registered. However, when I did a module show, it was there. I did restart Asterisk or include it in module.conf but it did not resolve the problem. Do you have any clues why this is happening? Thanks in advance. -- Executing [*1223*1**1900@incoming:78] Set(SIP/1900-08ee1da8, DEVSTATE(Custom:cfalw1900)=INUSE) in new stack [May 25 11:59:46] ERROR[8913]: pbx.c:1564 ast_func_write: Function DEVSTATE not registered /usr/lib/asterisk/modules/func_devstate.so /usr/src/asterisk-1.4.21.2/funcs/.func_devstate.makeopts /usr/src/asterisk-1.4.21.2/funcs/.func_devstate.moduleinfo /usr/src/asterisk-1.4.21.2/funcs/.func_devstate.o.d /usr/src/asterisk-1.4.21.2/funcs/func_devstate.c /usr/src/asterisk-1.4.21.2/funcs/func_devstate.o /usr/src/asterisk-1.4.21.2/funcs/func_devstate.so *CLI module show like func_devstate.so Module Description Use Count func_devstate.so Gets or sets a device state in the dialp 0 1 modules loaded -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inter-astersik dialling encounteres no audio
Thanks Sam, John and Justin for your wonderful advice. Yes, it was the sip.conf parameter reinvite= which was causing the problem. Setting it to NO will fix it. Thanks all in asterisk-users mailing list. The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it. inline: CPWRsig_04_11-03-2010.jpg-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inter-astersik dialling encounteres no audio
I have been deploying Asterisk (open source PABX) in the company which I work. So far, all the Asterisk servers do not really talk to each other. Recently, I am experimenting to dial from one Asterisk server to another through the WAN and I encountered a no-audio problem although the callee's phone can ring. I understand that the no-audio means that SIP traffic (TCP/UDP 5060) is allowed to go through but not RTP (UDP 16384-32767). Case A == This is a simplified diagram of how I am testing the dialling between 2 subnets. In this case, phone A is registered in Asterisk A and phone B is registered in Asterisk B. Phone A -- Asterisk A -- Router A == WAN == Router B -- Asterisk B -- Phone B Case B == However, before I have tested successfully using this kind of connection. In this case, phone B1 and B2 are registered in Asterisk B although they are on different subnets. Both phone B1 and B2 can ring and audio is allowed to pass through. Phone B1 -- Router A == WAN == Router B -- Asterisk B -- Phone B2 I am mystified why audio is allowed go through in case B but not case A. Can someone be kind enough to help me to understand why I have this problem? If the router is blocking RTP traffic, then why is that I have no audio problem in case B? Thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] secret=pw in sip.conf affecting inter-asterisk sip call
I was trying to do a SIP call between two Asterisk servers (1.4.21.2) 1) On the caller server, I coded the following in extensions.conf Dial(SIP/1166:password@asterisk-callee); 2) On the callee server, I coded the following in sip.conf [1166] type=friend; Friends place calls and receive calls context=incoming ; Context for incoming calls from this user host=dynamic ; This peer register with us dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info qualify=yes; Monitor latency between Asterisk server and phone call-limit=99 username=1166 ; Username to use in INVITE until peer registers secret=password; Normally you do NOT need to set this parameter mailbox=1166@default ; mailbox 5100 in voicemail context .default. callgroup=1 pickupgroup=1 The call was unsuccessful as follows. 1) On the caller machine, this is what we got from the console -- Executing [1166@incoming:1] Dial(SIP/1166-09d81668, SIP/1166:password@asterisk-callee) in new stack -- Called 1166:password@asterisk-callee -- SIP/asterisk-callee is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) 2) On the callee machine, this is what we got from the console, [Sep 14 14:34:12] NOTICE[11991]: chan_sip.c:14035 handle_request_invite: Call from '2765' to extension '1166:password' rejected because extension not found. However, I found out that if I remove secret=.. from the SIP entry and call without the password, then I will be able to call. Any thoughts? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] secret=pw in sip.conf affecting inter-asterisk sip call
I was trying to do a SIP call between two Asterisk servers (1.4.21.2) 1) On the caller server, I coded the following in extensions.conf Dial(SIP/1166:password@asterisk-callee); 2) On the callee server, I coded the following in sip.conf [1166] type=friend; Friends place calls and receive calls context=incoming ; Context for incoming calls from this user host=dynamic ; This peer register with us dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info qualify=yes; Monitor latency between Asterisk server and phone call-limit=99 username=1166 ; Username to use in INVITE until peer registers secret=password; Normally you do NOT need to set this parameter mailbox=1166@default ; mailbox 5100 in voicemail context .default. callgroup=1 pickupgroup=1 The call was unsuccessful as follows. 1) On the caller machine, this is what we got from the console -- Executing [1166@incoming:1] Dial(SIP/1166-09d81668, SIP/1166:password@asterisk-callee) in new stack -- Called 1166:password@asterisk-callee -- SIP/asterisk-callee is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) 2) On the callee machine, this is what we got from the console, [Sep 14 14:34:12] NOTICE[11991]: chan_sip.c:14035 handle_request_invite: Call from '2765' to extension '1166:password' rejected because extension not found. However, I found out that if I remove secret=.. from the SIP entry and call without the password, then I will be able to call. Any thoughts? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] secret=pw in sip.conf affecting inter-asterisk sip call
I was trying to do a SIP call between two Asterisk servers (1.4.21.2) 1) On the caller server, I coded the following in extensions.conf Dial(SIP/1166:password@asterisk-callee); 2) On the callee server, I coded the following in sip.conf [1166] type=friend; Friends place calls and receive calls context=incoming ; Context for incoming calls from this user host=dynamic ; This peer register with us dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info qualify=yes; Monitor latency between Asterisk server and phone call-limit=99 username=1166 ; Username to use in INVITE until peer registers secret=password; Normally you do NOT need to set this parameter mailbox=1166@default ; mailbox 5100 in voicemail context .default. callgroup=1 pickupgroup=1 The call was unsuccessful as follows. 1) On the caller machine, this is what we got from the console -- Executing [1166@incoming:1] Dial(SIP/1166-09d81668, SIP/1166:password@asterisk-callee) in new stack -- Called 1166:password@asterisk-callee -- SIP/asterisk-callee is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) 2) On the callee machine, this is what we got from the console, [Sep 14 14:34:12] NOTICE[11991]: chan_sip.c:14035 handle_request_invite: Call from '2765' to extension '1166:password' rejected because extension not found. However, I found out that if I remove secret=.. from the SIP entry and call without the password, then I will be able to call. Any thoughts? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] secret=pw in sip.conf affecting inter-asterisk sip call
chan_sip does not support specification of the password to be used for authentication in the dial string itself; your :password suffix is just being sent to the target system and it is trying to find a matching extension in the dialplan (and failing). Thanks Kevin. This is what I reckon from the tests that I did. I think I will have to remove all secret= from all my SIP entries. However, this is contrary to what the Asterisk books say. P.S. I have got problem receiving emails from asterisk-user mailing list. I could only see it from the web mail archive. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] No more ISDN in Malaysia Telekom???
We are setting up an office in Malaysia. We contacted Telekom Malaysia and are surprised to be told that ISDN-30 is no longer available. They are yet to give us information of the replacement technology. Does anyone have any experience and information with this? Thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] No more ISDN in Malaysia Telekom???
Arstan, thank you for your response. Malaysia Telekom replied This service is limited to avaibility of ports and infra avaibility as we are now upgrading to NGN. You may use business broadband and PSTN lines to connect to your Digital PABX to replace this service. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arstan Jusupov Sent: Thursday, 20 January 2011 1:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] No more ISDN in Malaysia Telekom??? Hello Lee, Telekom Malaysia provide PRI lines. We've been actively using their services for the past few years with success. Let me know if you need contacts. Regards, Arstan On Thu, Jan 20, 2011 at 9:56 AM, Lee, John (Sydney) john@compuware.com wrote: We are setting up an office in Malaysia. We contacted Telekom Malaysia and are surprised to be told that ISDN-30 is no longer available. They are yet to give us information of the replacement technology. Does anyone have any experience and information with this? Thanks in advance. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use modprobe to find E1/T1 jumper setting onPRI card
Thanks Shaun. Unfortunately, I am still using zaptel. Is there a similar command in zaptel? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Thursday, 30 September 2010 1:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Use modprobe to find E1/T1 jumper setting onPRI card On 09/29/2010 02:52 AM, Lee, John (Sydney) wrote: Do you mean that if I could define 30 channels in span 1 for example, then span 1 is set to E1? If not, then it is T1. You could also see this information in the type output from dahdi_scan. For example before configuring a span: # ./dahdi_scan [1] active=yes alarms=UNCONFIGURED description=Wildcard TE122 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE122 location=PCI Bus 15 Slot 05 basechan=1 totchans=24 irq=90 type=digital-T1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=B8ZS,AMI framing_opts=ESF,D4 coding= framing= -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can't get libpri/PRI to work, missing PRI commands
In Asterisk, the funny thing is if a certain component is not installed properly or the config file has a typo or something, this will be shown up as a non-existent command in Asterisk command line interface. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of mis...@efro.us Sent: Thursday, 30 September 2010 6:57 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] can't get libpri/PRI to work, missing PRI commands I'm putting together a PBX using a TE420P card configured for E1s that is connected to an Errickson MTS. successfully compiled and installed libpri 1.4.11.3, DAHDI 2.3.0.1+2.3.0 and Asterisk 1.6.2.9. everythings seems to be working. SIP phone to SIP phone (POLYCOM) calls work fine however, network calls do not. When I went to debug PRI, the only command that showed up when I did CLI core show help pri was 'pri intense debug span' which seemed strange to begin with and when I did CLI pri intense debug span 1, I got something strange like 'pri set debug 2 span 1' is not a valid command. Tried reinstalling everything but keep getting the same result. Also, when I try to make call from SIP phone to a wireless phone and vise versa on our GSM network, I get something like Call to extension rejected because the extension is not found in context POLYCOM# but it is definitely in the extensions.conf with a line exten = 1,1,Dial(dahdi/g1/#xxx). Please help... Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Use modprobe to find E1/T1 jumper setting onPRI card
Do you mean that if I could define 30 channels in span 1 for example, then span 1 is set to E1? If not, then it is T1. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez Sent: Wednesday, 29 September 2010 4:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Use modprobe to find E1/T1 jumper setting onPRI card Simple enough. If your card says it has 30 ports then it is on, if it only says 24 it is off. On Wed, 29 Sep 2010 12:06:30 +1000, Lee, John (Sydney) wrote Does anyone know if I could use modprobe command to find out rather than set the jumper on a Digium PRI card? I want to find out the jumper settings on the card without opening the box which will cause down time. Thanks. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Use modprobe to find E1/T1 jumper setting on PRI card
Does anyone know if I could use modprobe command to find out rather than set the jumper on a Digium PRI card? I want to find out the jumper settings on the card without opening the box which will cause down time. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom not updating the directory list
The very obvious thing to check is the permission of the mac-addr-directory.cfg. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hin lee Sent: Thursday, 18 March 2010 4:56 PM To: Asterisk Users Subject: Re: [asterisk-users] Polycom not updating the directory list anyone? From: hin lee hi...@yahoo.com To: Asterisk Users asterisk-users@lists.digium.com Sent: Fri, March 12, 2010 10:08:53 AM Subject: Polycom not updating the directory list Hi, I have a strange problem with all of our Polycom 550 650 phones. I am running a TFTP server on my Asterisk server and option 66 Boot Host pointing to Asterisk on my DHCP server. The auto-provisioning is working because the phones are registering correctly with their extension. If I change the MAC.cfg file to another extension and reboot the phone, it will reflect the new ext. The part that doesn't work is the MAC-directory.cfg. If I make an update to this file and reboot the phones, they do not reflect the new directory list. The only way I was able to get the phone to see the new directory list was to Format the phone. Of course this is not the ideal way. Also to add, the MAC-directory.cfg files point to 0-directory.xml. This way I only have one file to maintain. Anyone knows why it's not pull the new MAC-directory.cfg file. Thank you! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
Yes, this is still one of the unsolved mysteries I wanted to find out about Polycom provisioning despite using it for a few years now. I used vsftpd and initially used boot server opt 66 and type string but could not get it to work. I asked our guru in DTW and he told me to use 129 and lo and behold, it worked but I was not told why when I asked and I never had time to find out why. Karl, if you could find the answer, please share it with us. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Thursday, 18 March 2010 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom DHCP Option 66 FTP provisioning Is anyone successfully using DHCP option 66 to specify an FTP [sic] provisioning of Polycom Sounpoint phones instead of TFTP? I know option 66 is typically used TFTP booting, but the Polycom doc doesn't appear to specify that option 66 implies TFTP instead of FTP (since you explicitly call out the protocol). TFTP option 66 booting was working fine. Does anyone know whether FTP provisioning of Polycom definitely requires a custom DHCP option like 160? Usually in a situation like this I'd just creatively try different things in a divide-and-conquer approach to find something that works. However in THIS case the phone tries to contact the boot server for SO LONG that the aforementioned 'brute-force' option would take me a decade. Therefore I'm trolling for tips, which would be very mcuh appreciated! Thanks! -Karl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning
I'll see if E4Strategies can open a support ticket at Polycom. They're really good about stuff like that. I'll let you know either way. What is E4Strategies? Polycom support is hopeless in Oz. They just shove you to some distributer who only knows to replace your hardware. What did you pass in option 129? Just an IP address? A fully qualified domain name? A whole URL? A whole URL including protocol and credentials? Just ftp://ftpuid:ftp...@fqdn; Out of curiousity, why did you choose option 129? I believe that's an undefined PiXiE boot option, but I'm curious because polycom seems to have made a de-facto convention of option 160. Unfortunately, our guru in DTW did not let me know why. Maybe when he sees this post, he might jump in to explain to us. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Thursday, 18 March 2010 1:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning Now that I know that I'm not the only person (i.e. it's less likely that I just made a careless mistake), I'll see if E4Strategies can open a support ticket at Polycom. They're really good about stuff like that. I'll let you know either way. What did you pass in option 129? Just an IP address? A fully qualified domain name? A whole URL? A whole URL including protocol and credentials? I'd love to see that portion of your dhcpd.conf file. Out of curiousity, why did you choose option 129? I believe that's an undefined PiXiE boot option, but I'm curious because polycom seems to have made a de-facto convention of option 160. In other words that's the preconfigured non-66 DHCP option in the bootrom. I try to do as little one-off 'setup' as possible hopefully in such a way as to never need to revisit the phone even if the bootserver address or even the subnet address were to change. It seems that if I can recycle the factory-assigned FTP username and DHCP option number, it would be a good idea all else being equal. Maybe 160 would give the same trouble as option 66 :-) Thanks for your post. -Karl - Original Message - From: Lee, John (Sydney) john@compuware.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, March 17, 2010 8:48 PM Subject: Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning Yes, this is still one of the unsolved mysteries I wanted to find out about Polycom provisioning despite using it for a few years now. I used vsftpd and initially used boot server opt 66 and type string but could not get it to work. I asked our guru in DTW and he told me to use 129 and lo and behold, it worked but I was not told why when I asked and I never had time to find out why. Karl, if you could find the answer, please share it with us. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Thursday, 18 March 2010 10:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Polycom DHCP Option 66 FTP provisioning Is anyone successfully using DHCP option 66 to specify an FTP [sic] provisioning of Polycom Sounpoint phones instead of TFTP? I know option 66 is typically used TFTP booting, but the Polycom doc doesn't appear to specify that option 66 implies TFTP instead of FTP (since you explicitly call out the protocol). TFTP option 66 booting was working fine. Does anyone know whether FTP provisioning of Polycom definitely requires a custom DHCP option like 160? Usually in a situation like this I'd just creatively try different things in a divide-and-conquer approach to find something that works. However in THIS case the phone tries to contact the boot server for SO LONG that the aforementioned 'brute-force' option would take me a decade. Therefore I'm trolling for tips, which would be very mcuh appreciated! Thanks! -Karl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com
Re: [asterisk-users] app_dial.c: Unable to create channel of type'Zap'(cause 34 - Circuit/channel congestion)
You didn't state what kind of computer the TE412P is in It was a DELL PE2950. the first thing to do if you have a hardware problem after a power bounce is to shutdown everything, power it off, wait 30 seconds, then turn it back on normally. You could be right. I think this is what someone told me before but I never took any notice because 99% of the time we don't need to do anything after a power shut down. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, 12 February 2010 3:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] app_dial.c: Unable to create channel of type'Zap'(cause 34 - Circuit/channel congestion) You didn't state what kind of computer the TE412P is in, but IME, the first thing to do if you have a hardware problem after a power bounce is to shutdown everything, power it off, wait 30 seconds, then turn it back on normally. Sorry you lost the day of usage. -- Danny Nicholas -- -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lee, John (Sydney) Sent: Thursday, February 11, 2010 1:21 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] app_dial.c: Unable to create channel of type 'Zap'(cause 34 - Circuit/channel congestion) Just to share some experience with everyone about what happened today to our Asterisk 1.4 box with Digium TE412P card. We had an unscheduled power outage which shut down the Asterisk box. When the power went up, Asterisk came back up okay but the ports on the card were all red. Zttool show red alarm and cat /proc/zaptel/1 show red alarm today. Both incoming and outgoing cannot be made. When a outgoing call was made, we got the following error message: app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) We suspect it was the ISDN line problem and so we waited a whole day for the engineer to arrive. He plugged an ISDN phone into the line and found it was working because he could call out. We are perplexed and thought about replacing the Digium card. We ended up just re-seating the card and lo and behold, everything was hunky dory after re-seating. Does anyone know why? -- _ -- 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 -- _ -- 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] app_dial.c: Unable to create channel oftype 'Zap' (cause 34 - Circuit/channel congestion)
What is the output of 'cat /proc/dahdi/1' ? I did not record it but it just shows every channel as 'red alarm'. What do you have in /etc/zaptel.conf ? loadzone=au defaultzone=au # # For OnRamp 10 # span=1,1,0,ccs,hdb3,crc4 bchan=1-10 unused=11-15,17-31 dchan=16 # # Rhino 24-port Channel Bank # span=2,0,0,esf,b8zs fxols=32-55 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Thursday, 11 February 2010 10:32 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] app_dial.c: Unable to create channel oftype 'Zap' (cause 34 - Circuit/channel congestion) On Thu, Feb 11, 2010 at 06:20:54PM +1100, Lee, John (Sydney) wrote: Just to share some experience with everyone about what happened today to our Asterisk 1.4 box with Digium TE412P card. We had an unscheduled power outage which shut down the Asterisk box. When the power went up, Asterisk came back up okay but the ports on the card were all red. Zttool show red alarm and cat /proc/zaptel/1 show red alarm today. What is the output of 'cat /proc/dahdi/1' ? What do you have in /etc/zaptel.conf ? Both incoming and outgoing cannot be made. When a outgoing call was made, we got the following error message: app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) We suspect it was the ISDN line problem and so we waited a whole day for the engineer to arrive. He plugged an ISDN phone into the line and found it was working because he could call out. We are perplexed and thought about replacing the Digium card. We ended up just re-seating the card and lo and behold, everything was hunky dory after re-seating. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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
[asterisk-users] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion)
Just to share some experience with everyone about what happened today to our Asterisk 1.4 box with Digium TE412P card. We had an unscheduled power outage which shut down the Asterisk box. When the power went up, Asterisk came back up okay but the ports on the card were all red. Zttool show red alarm and cat /proc/zaptel/1 show red alarm today. Both incoming and outgoing cannot be made. When a outgoing call was made, we got the following error message: app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) We suspect it was the ISDN line problem and so we waited a whole day for the engineer to arrive. He plugged an ISDN phone into the line and found it was working because he could call out. We are perplexed and thought about replacing the Digium card. We ended up just re-seating the card and lo and behold, everything was hunky dory after re-seating. Does anyone know why? -- _ -- 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] Polycom phone DND state
You may be right. Pressing DND will only return a BUSY dial status and so you really cannot distinguish whether it is a genuine DND. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Friday, 5 February 2010 12:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Polycom phone DND state Sorry it took awhile to answer. DND works flawlessly, but whenever using BLF I can only tell that a line is either in use (on a call) or not. I cannot tell a phone is on DND, or on hold for that matter. Would be extremely useful. Would be willing to pay for this developpement if it can be done as long as the feature makes it into trunk. Heck, I'll give 200$ for someone just to tell me how to configure it properly if it's a matter of just missing a config line. Mike From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stuart McQuade Sent: Wednesday, January 27, 2010 7:50 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom phone DND state Hi, At my previous company we ran 1.4.x.x (underneath DiVitas.com software) and our Polycom IP 550 would use DND without a problem, but the IP 331 (on exactly the same server) didn't work with DND. So it may be a model-specific problem rather than your Asterisk config. Stuart From: Lee, John (Sydney) john@compuware.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wed, 27 January, 2010 8:02:14 Subject: Re: [asterisk-users] Polycom phone DND state I am using 1.4.21.2 and DND is definitely working. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Saturday, 23 January 2010 2:50 AM To: ' Asterisk Users Mailing List - Non-Commercial Discussion ' Subject: [asterisk-users] Polycom phone DND state Hi, I know having Asterisk aware of Polycom Do No Disturb state wasn't working before (1.4), but is this working in any recent version? Is there any custom way of doing this? Regards, Mike -- _ -- 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] Use a BLF for monitoring
In your dialplan, you should put in...sth like exten = 1001,hint,Custom:virtext1001 In your script, you should put in...sth like Set(DEVSTATE(Custom:virtext001=INUSE); Set(DEVSTATE(Custom:virtext001=NOTINUSE); In the phone directory.xml, define an entry with ct=1001 and turn bw on. Reboot phone of course. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of jon pounder Sent: Tuesday, 2 February 2010 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Use a BLF for monitoring Richard Kenner wrote: Is there a way to make a virtual extension busy programmatically? I want to be able to turn lights on and off on a Polycom phone from a script. That's what the Custom device type is for. please elaborate I would like to know too -- _ -- 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
Re: [asterisk-users] Polycom phone DND state
I am using 1.4.21.2 and DND is definitely working. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Sent: Saturday, 23 January 2010 2:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Polycom phone DND state Hi, I know having Asterisk aware of Polycom Do No Disturb state wasn't working before (1.4), but is this working in any recent version? Is there any custom way of doing this? Regards, Mike -- _ -- 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] is there some Chinese version of sounds available?
when use the VoiceMail , all the directions all english. i want to know is there some Chinese version of sounds available now? or should i record it myself? http://www.voip-info.org/wiki/view/Asterisk+sound+files+international Look under Chinese (Mandarin) -- _ -- 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] Weird Polycom SP 650
Bon journo Aldo. I am having several issues with my first SP 650. * Assembly: 2345-12600-001 Rev.G I have deployed more than 200 IP650 with the same assembly as yours and so far there are no problems. The first thing I have noticed is that I was not able to upgrade the unit's firmware with the one currently available in the support area for this phone. The TFTP setup I used had worked for the upgrade of some additional SP's (SP 320/330); besides the fw files, that I got twice, even if I am not sure that this is necessary. I am using bootROM 4.2 and SIP 3.1.2 You may have a problem with the boot server or its permissions (just a guess). You have to go through your boot server and find out why. No easy way unfortunately. The second strange occurrence is the inability to change the unit's display language (to Italian settings). I just tried changing my phone to Italiano and was successful. The language files are in the SIP software and so maybe because you cannot upgrade your SIP release, that is why you cannot switch to Italiano. I was however able to activate the BLF function (through a customisation found online for the sip.cfg config file), joined with the activation of the 'Presence' setting for some custom created entries of the Directory of the phone. Well done. BLF is not that straightforward for Polycom phones. Some workaround is required. Needless to say that the other SP 330 have no similar issue, with 'copy cat' settings in the sip.conf file. The config of 650 is very similar to those of 330. By right, if it works for 330, it should also work for 650. Hope this helps! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Aldo Bergamini Sent: Monday, 11 January 2010 10:01 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Weird Polycom SP 650 Hi, I am seeking help with the installation of a Soundpoint 650 desk phone. Although I have some experience (and a good one! no single issue so far, besides the problem I am trying to solve...) installing a few SP 320/330 units, I am having several issues with my first SP 650. Polycom SP 650 Data: * P/N: 3150-11530-212 * SD Sound * FW: 2.1.2.0078 * Assembly: 2345-12600-001 Rev.G The first thing I have noticed is that I was not able to upgrade the unit's firmware with the one currently available in the support area for this phone. The TFTP setup I used had worked for the upgrade of some additional SP's (SP 320/330); besides the fw files, that I got twice, even if I am not sure that this is necessary. The second strange occurrence is the inability to change the unit's display language (to Italian settings). I was however able to activate the BLF function (through a customisation found online for the sip.cfg config file), joined with the activation of the 'Presence' setting for some custom created entries of the Directory of the phone. Furthermore, once installed at my customer's site I had to fiddle with problems related to DTMF tones. The customer reported that she could not link to voicemail, to get messages. And as a matter of fact when I checked there was no way to dial the password into Asterisk, until I changed the SIP settings for this extension to 'Inband'. Needless to say that the other SP 330 have no similar issue, with 'copy cat' settings in the sip.conf file. What is however a complete disaster is what happens when the user is talking on a call, and for any reason, a second calls is presented to the unit by the Asterisk 1.6 server. The user has its headset speaker muted (and therefore thinks that the call was lost/ended abruptly), yet the party at the other end of the call is still alive and well (aka connected) and has no idea we my user starts blabbering about problems to the call. Does anybody have similar experiences with the 650? There is very little I did differently on this unit than on the other SP 330s that are running without a problem, on the same Asterisk setup.. Any additional questions are more than welcome! Kind regards Aldo -- _ -- 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
[asterisk-users] MixMonitor stops audio in SIP to SIP call
Has anyone experienced this problem before? I am running Asterisk 1.4.21.2 If I run: MixMonitor(..) Dial(SIP/...) Both parties cannot hear each other. As soon as I comment out MixMonitor, the audio can be heard. I saw this issue on https://issues.asterisk.org/view.php?id=16256 It seems to match what I encountered. Any ideas? ___ -- 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] How to do a 3 party Warm Transfer in Asteriks 1.4
I don't think this can be done. In your scenario, B is effectively the host and if B drops the line, both A and C will be dropped off as well. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff Johnson Sent: Monday, 12 October 2009 2:25 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to do a 3 party Warm Transfer in Asteriks 1.4 We are running Asterisk 1.4 and need some help to determine how (if) * supports 3 party warm transfers. I've searched quite a bit and all I can find is information on attended transfers. What we are looking for is: (1) external inbound call A comes to * extension B, caller A is placed on hold and extension B calls external third party C. After explaining caller A issue to Party C, Ext B brings Caller A onto the call and introduces A to C. After the into, ext B then drops off the call while A C continue the call. Any help would be appreciated. Thanks Much, Jeff Johnson This email and any attached files are confidential and intended solely for the intended recipient(s). If you are not the named recipient you should not read, distribute, copy or alter this email. Any views or opinions expressed in this email are those of the author and do not represent those of NeturallySpeaking, LLC. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE121P Blue Alarm/Recovering
1) I have not seen a blue light (usually red/yellow) before on a Digium card and so don't really know what it means. 2) Try to see if you can see any messages coming up from the Asterisk box itself (not thru putty or other remote console). You should see a steady stream of error messages coming up. Also, look at /var/log/asterisk/messages or event_log. There may or may not be anything there. 3) I see that loadzone = cn which means the installation is in China. 4) My experience tells me that if it is in major China cities, the ISDN line would be E1 which is correct but from the installations I did, China's E1 does not like CRC4. 5) I think it is highly likely to be a ISDN line config (telco or Asterisk side) problem. 6) You can try contacting Digium support too. They provide prompt support but you have to register you serial no first. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaverstyn, David C Sent: Tuesday, 29 September 2009 10:07 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TE121P Blue Alarm/Recovering I'd appreciate it if someone was able to assist. Running the command: dahdi_hardware: pci::08:08.0 wcte12xp+ d161:8000 Wildcard TE121 dahdi_scan: [1] active=yes alarms=REC description=Wildcard TE121 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE121 with VPMADT032 location=PCI Bus 08 Slot 09 basechan=1 totchans=31 irq=169 type=digital-E1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS cat /etc/dahdi/system.conf: bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone = cn defaultzone = cn From: Klaverstyn, David C Sent: Monday, 28 September 2009 5:05 PM To: 'asterisk-users@lists.digium.com' Subject: TE121P Blue Alarm/Recovering Hi All, I have a TE121P card installed and since connected it to the PRI I keep getting the Current Alarm as continually changing from Blue Alarm/Recovering and Recovering. The config I have is: /etc/dahdi/system.conf bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone = cn defaultzone = cn /etc/asterisk/chan_dahdi.cfg [channels] Context=telco language=cn switchtype=euroisdn signalling=pri_cpe rxwink=300 usecallerid=yes . . echocancel=yes group=1 channel=1-15,17-31 I installed the following components Asterisk 1.4.26.2 DAHDI Linux 2.2.0.2 DAHDI Tools 2.2.0 Libpri 1.4.10.1 Addons 1.4.9 Any help would be greatly appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE121P Blue Alarm/Recovering
Many thanks John of Sydney. My pleasure my fellow Asterisker! I removed the CRC4 and it worked straight away. Can you recommend a CRC type at all or would it be best to leave it as nothing? Just leave it blank would do. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Klaverstyn, David C Sent: Tuesday, 29 September 2009 10:49 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE121P Blue Alarm/Recovering Many thanks John of Sydney. I removed the CRC4 and it worked straight away. Can you recommend a CRC type at all or would it be best to leave it as nothing? David of Brisbane. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Lee, John (Sydney) Sent: Tuesday, 29 September 2009 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] TE121P Blue Alarm/Recovering 1) I have not seen a blue light (usually red/yellow) before on a Digium card and so don't really know what it means. 2) Try to see if you can see any messages coming up from the Asterisk box itself (not thru putty or other remote console). You should see a steady stream of error messages coming up. Also, look at /var/log/asterisk/messages or event_log. There may or may not be anything there. 3) I see that loadzone = cn which means the installation is in China. 4) My experience tells me that if it is in major China cities, the ISDN line would be E1 which is correct but from the installations I did, China's E1 does not like CRC4. 5) I think it is highly likely to be a ISDN line config (telco or Asterisk side) problem. 6) You can try contacting Digium support too. They provide prompt support but you have to register you serial no first. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Klaverstyn, David C Sent: Tuesday, 29 September 2009 10:07 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TE121P Blue Alarm/Recovering I'd appreciate it if someone was able to assist. Running the command: dahdi_hardware: pci::08:08.0 wcte12xp+ d161:8000 Wildcard TE121 dahdi_scan: [1] active=yes alarms=REC description=Wildcard TE121 Card 0 name=WCT1/0 manufacturer=Digium devicetype=Wildcard TE121 with VPMADT032 location=PCI Bus 08 Slot 09 basechan=1 totchans=31 irq=169 type=digital-E1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS cat /etc/dahdi/system.conf: bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone = cn defaultzone = cn From: Klaverstyn, David C Sent: Monday, 28 September 2009 5:05 PM To: 'asterisk-users@lists.digium.com' Subject: TE121P Blue Alarm/Recovering Hi All, I have a TE121P card installed and since connected it to the PRI I keep getting the Current Alarm as continually changing from Blue Alarm/Recovering and Recovering. The config I have is: /etc/dahdi/system.conf bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone = cn defaultzone = cn /etc/asterisk/chan_dahdi.cfg [channels] Context=telco language=cn switchtype=euroisdn signalling=pri_cpe rxwink=300 usecallerid=yes . . echocancel=yes group=1 channel=1-15,17-31 I installed the following components Asterisk 1.4.26.2 DAHDI Linux 2.2.0.2 DAHDI Tools 2.2.0 Libpri 1.4.10.1 Addons 1.4.9 Any help would be greatly appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International Numbering plan ?
I found that it was a bit incomplete for China. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Wednesday, 23 September 2009 3:35 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] International Numbering plan ? On 23/09/09 4:39 PM, Michael wrote: On Wed, 23 Sep 2009 16:19:26 Phibee Network Operation Center wrote: Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? thanks Jpc Country numbering plan can be easily found. Anything finer then that and you will need to pay. That link I provided is correct at least for New Zealand cities etc -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bringing people into a conference
BTW, I have been using the n-way conference feature from Polycom. By n-way, they mean only 4 parties (including the host) and the interface is quite neat because you can manage the conference from the display and you can mute, far-mute, hold and resume each parties. To use this Polycom nway conference, you need to purchase a productivity suite. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Wednesday, 23 September 2009 3:57 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Bringing people into a conference On 23/09/09 5:07 PM, Harley Holcombe wrote: 1. Internal person A calls person B 2. Person A presses *0, he is given a dial tone and person B is taken to a conference room 3. Person A calls person C and they can talk, and then person A presses **. 4. Person C is brought to the conference room, but person A is disconnected. Is there an extension: dynamic-nway,282,1 Oh, and please refrain from using HTML emails to lists. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] International Numbering plan ?
The URL is a good start but for some large countries which I have worked for, the list misses some important information like inter-city, inter-state, inter-city mobile and local mobile and IDD. To me, nothing can replace local intelligence. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Wednesday, 23 September 2009 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] International Numbering plan ? On 23/09/09 4:19 PM, Phibee Network Operation Center wrote: Hi anyone know where i can find all internatinal numbering plan in csv and for free or small price ? I'm not sure you understand the scale of what you're asking, but anyways. Here's a start: http://www.itu.int/oth/T0202.aspx?parent=T0202 Bear in mind that these numbers change reasonably regularly. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie: How to detect an * in Read()?
A user embedded an * in a Read command and it causes my AEL script to fail. Does anyone know how I could code to detect it? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk CLI commands not running !!!!!
I have a cron job that restarts Asterisk every night. This is supposed to be an old Asterisk best practice for 1.2.* but I think it does not harm. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mindaugas Kezys Sent: Tuesday, 8 September 2009 10:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk CLI commands not running ! Asterisk sometimes goes to sleep. (And never wakes-up). Restart it and all will be fine again. We have a watchdog which sends SIP OPTIONS packet to Asterisk and if it does not respond – restarts it. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of abdelkader Sent: 2009 m. rugsėjo 8 d. 10:40 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk CLI commands not running ! Hello, I am using Asterisk 1.4.22 in a debian 4.0 with a kernel version 2.6.18-6-amd64 (SMP). The processor type is Intel(R) Xeon(R) CPU E5420 @ 2.50GHz. Sometimes, I get a strange behavior from asterisk: The CLI commands does not work and Asterisk cannot receive calls. The output of every CLI command is that command is not known (no such command). Please help me resolve this problem: what can be the cause of it? is it Asterisk or my system? and what have I to do to eliminate this problem? Thks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium hardware support ?
does Digium provide a service support for a compatibility question about their PRI hardware ? Before you open a call with them, you will have to register your Digium card by entering the serial number. The serial number is printed on a sticker which is attached to the card. There is no way to find out the serial number from the software. I find the Digium support responsive and knowledgeable. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Prevent Agent Login from a second extension
I think you have to write your own agent login and logout so that you will not have this problem. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shanavaz E A Sent: Wednesday, 2 September 2009 4:27 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Prevent Agent Login from a second extension Hi friends, Is there any way to prevent an Agent from logging in from a second extension if he is already logged on from an extension. Right now, the scenario is if he login from a second extension, asterisk will automatically log him off from first extension. What I need is that asterisk should tell him that he is already logged on from an extension and should prevent him from logging in again from another extn. The problem with existing scenario is that, I am not getting CDR record for the automatic log out event. I need this for evaluation purposes. I am using asterisk 1.2.30. I have 1.4 also but that also is having the same behavior. Thanks in advance for any help. Regards Shanavaz. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Just a quick guess - is it because you did not program your Polycom digit plan properly in sip.cfg? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Tuesday, 1 September 2009 2:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Inquiry:Problem with Call Parking Dear All Can you please do me favor and let me know what is the problem with my Asterisk call parking as it is not functioning correctly on my Asterisk ? Please find attached my features.conf . According to my configuration , the subscriber needs to press hash (pound) key and dial 700 to initiate the transfer . We tried but it didn't get through on our Asterisk . Can you please let me know what extra config needs to be done for putting it into operation ? Regards H.Motamedi ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Problem with Call Parking
Please find attached my Asterisk sip.conf . Can you please let me know what modifications are needed ? Yes, I am referring to the Polycom sip.cfg and not the sip.cfg in Asterisk. Somethere down in sip.cfg, there is a line that looks like this: digitmap dialplan.digitmap=#700| ... Basically, Polycom will scan your input to see when it will pass all the keystrokes to Asterisk. In above, if it detects that you have entered #700, it will automatically send it to Asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s![swapper:0]
I'd contact Digium - they're really good with providing support - just add the following line and dial it: Thanks Matt for your suggestion. We despatched a new TE412P card to replace the existing card but the same problem occurred. So, I think it is not the Digium card problem. At the same time, we noticed that the 2nd port (which is configured as a T1 to connect to a Rhino Channel Bank) was reporting red/rec in zttool. So, we unplug the 2nd port and the soft lockup problem goes away. However, doing so means we cannot configure and use the analog channels from the E1 ISDN line which is connected to port 1 on the TE412P. I reported the problem to Rhino and the support confidently believed that the issues are related to the OS and platform and not the Rhino. Anyone has any suggestion? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ACD, call barge, recording
1) Can ACD (Automatic Call Distribution) service work with asterisk, and how to set up ACD in asterisk ? You can (and it is better to) write your own code in Asterisk. 2) How call barging can set up in asterisk ? There is a zap barge cmd - not sure if this is what you want. 3) How call recording can set up in asterisk? You can set up one-touch recording pretty easily. Please check voip-info.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot play soundfile, doesnt find it or wrong format? Weird, worked yesterday! :-)
It also means that unless your target cchannel is in gsm format How can I check what format my channels are using? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cannot play soundfile, doesnt find it or wrong format? Weird, worked yesterday! :-)
Is this the one you are talking about? Do you mean that if I play MOH using any of the formats below, then there will be no CPUs wasted for translation purposes? *CLI core show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) 65536 (1 16) (0x1) image jpeg (JPEG image) 131072 (1 17) (0x2) imagepng (PNG image) 262144 (1 18) (0x4) video h261 (H.261 Video) 524288 (1 19) (0x8) video h263 (H.263 Video) 1048576 (1 20) (0x10) video h263p (H.263+ Video) 2097152 (1 21) (0x20) video h264 (H.264 Video) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie: How to copy track from CD for MOH without getting Junk at beginning of frame ...
I was copying tracks from CD into mp3 files so that I could use it in Asterisk 1.4.21.2 MOH. (BTW, I have already secured proper license to play MOH to callers.) I used MS Media Player version 11 and rip it at 128kbps (smallest) but whenever I listen to MOH, I saw the following message on the Asterisk console. WARNING[20829]: mp3/interface.c:215 decodeMP3: Junk at the beginning of frame 49443303 I tried it with different bit rate (320 kbps) and the same error message appeared. I used the following musiconhold.conf [classical] mode=files directory=/var/lib/asterisk/moh/classical random=yes Are there any Asterisk+Audio expert that can offer me some advice? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to copy track from CD for MOH without getting Junk at beginning of frame ...
Yep, agreed. Convert the file to the native codec(s) in which it will be played. Alex, could you please elaborate on this? I am no audio guy. On Media player, I can rip it into mp3 or wav or windows media audio. Which one should I use? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie: How to find the serial number of Digium card?
Does anyone know how to find the serial number of Digium card without opening the machine? I was trying to call for support at Digium and they asked me for the serial number. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to find the serial number ofDigium card?
Thanks Tilghman. I learnt it the hard way - I never imagined I need to jot down the serial number of a PCI card :-( -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Monday, 17 August 2009 1:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie: How to find the serial number ofDigium card? On Sunday 16 August 2009 19:59:50 Lee, John (Sydney) wrote: Does anyone know how to find the serial number of Digium card without opening the machine? I was trying to call for support at Digium and they asked me for the serial number. You cannot. The serial number is not anywhere in the firmware, only on a sticker on the card itself. -- Tilghman Teryl with Peter, Cottontail, Midnight, Thumper, Johnny (bunnies) and Harry, BB, George (dogs) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]
I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no problems since Dec last year. We are using Digium TE412P to connect to an E1 ISDN line. Since Dec last year, we did not add or delete any software or hardware. We also did not do any yum update. The linux kernel is 2.6.18-92.1.22.el5 Last week, the users reported that people from outside could not dial in but users can dial out. We rebooted the box and everything was fine. Suddenly, starting this week, the box froze several times a day with a BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0] error message on the console. Before it freezes, I can see a continuous stream of error message ... timing source auto card 0! timing source auto card 0! timing source auto card 0! timing source auto card 0! ... coming up on the machine. We rebooted and it became okay for a few hours and we had to reboot it again in order to clear the problem. BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0] Pid: 0, comm: swapper EIP: 0060:[,C0417911.] CPU: 1 EIP is at smp_call_function+0x99/0xc3 EFLAGS: 0297 Tainted: G (2.6.10-92.1.22.e15 #1) EAX: 0002 EBX: ECX: 0001 EDX: 00fb ESI: 0003 EDI: EBP: c0417ae0 DS: 007B ES: 007b CR0: 8005003b CR2: b7fec780 CR3: 324B2000 CR4: 06d0 [c0417ae0] stop_this_cpu+0x0/0x33 [c041794e] smp_send_stop+0x13/0x1c [c0425bcf] panic+0x4c/0x16d [c040da17] intel_machine_check+0xf9/0x146 [c040d91e] intel_machine_check+0x0/0x146 [c0403ccf] error_code+0x39/0x40 [c0403ccf] mwait_idel+0x25/0x38 [c0522200] acpi_processor_idle+0x154/0x3b4 [c0403c90] cpu_idle+0x9f/0xb9 === Q1. A strange thing is I could not find this error message in /var/log/messages or dmesg. The soft lockup error message can only be found on the machine itself. Q2. Could it be kernel incompatibility problem? However, we did not ever change anything since it was installed. Q3. From the error message, how do I know it is a software (kernel?) or hardware problem? I would appreciate if someone could give me any suggestions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0]
I have this DELL PE2950 running Asterisk 1.4.21.2 on RHEL 5 with no problems since Dec last year. We are using Digium TE412P to connect to an E1 ISDN line. Since Dec last year, we did not add or delete any software or hardware. We also did not do any yum update. The linux kernel is 2.6.18-92.1.22.el5 Last week, the users reported that people from outside could not dial in but users can dial out. We rebooted the box and everything was fine. Suddenly, starting this week, the box froze several times a day with a BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0] error message on the console. Before it freezes, I can see a continuous stream of error message ... timing source auto card 0! timing source auto card 0! timing source auto card 0! timing source auto card 0! ... coming up on the machine. We rebooted and it became okay for a few hours and we had to reboot it again in order to clear the problem. BUG: soft lockup - CPU#1 stuck for 10s! [swapper:0] Pid: 0, comm: swapper EIP: 0060:[,C0417911.] CPU: 1 EIP is at smp_call_function+0x99/0xc3 EFLAGS: 0297 Tainted: G (2.6.10-92.1.22.e15 #1) EAX: 0002 EBX: ECX: 0001 EDX: 00fb ESI: 0003 EDI: EBP: c0417ae0 DS: 007B ES: 007b CR0: 8005003b CR2: b7fec780 CR3: 324B2000 CR4: 06d0 [c0417ae0] stop_this_cpu+0x0/0x33 [c041794e] smp_send_stop+0x13/0x1c [c0425bcf] panic+0x4c/0x16d [c040da17] intel_machine_check+0xf9/0x146 [c040d91e] intel_machine_check+0x0/0x146 [c0403ccf] error_code+0x39/0x40 [c0403ccf] mwait_idel+0x25/0x38 [c0522200] acpi_processor_idle+0x154/0x3b4 [c0403c90] cpu_idle+0x9f/0xb9 === Q1. A strange thing is I could not find this error message in /var/log/messages or dmesg. The soft lockup error message can only be found on the machine itself. Q2. Could it be kernel incompatibility problem? However, we did not ever change anything since it was installed. Q3. From the error message, how do I know it is a software (kernel?) or hardware problem? I would appreciate if someone could give me any suggestions. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Simple Queue Problem
I am running Asterisk 1.4.21.2 For reception, I defined a simple queue with one SIP phone as the only member. When I receive an incoming call, I test QUEUE_WAITING_COUNT to see if it is 0. If it is 0, then I will playback a message to tell the caller to be patient and then do a Queue(queue-name). If QUEUE_WAITING_COUNT is zero, then I will just Queue(queue-name, r) to ring the receptionist phone without playing any message. A problem arises if the receptionist is talking to someone on the phone. In this scenario, QUEUE_WAITING_COUNT is also zero but I will need to playback a pls-be-patient message as well. So, I need to find out whether the receptionist phone is busy even if QUEUE_WAITING_COUNT = 0. Do you know if there is anyway, without dialling a SIP channel, I can check if a SIP extension is engaged or not? ___ -- 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] Queue() Ignore Hangup Request
I saw a few posts of this problem before I could not figure out the reason I am getting it. I am running RHEL 5, Asterisk 1.4.21.2, zaptel 1.4.11 and libpri 1.4.4 Basically, if I dial into a queue and hang up the phone, Asterisk did not detect the hangup request and Asterisk will only hang up when the timer expires. There is no such problem if I do not use Queue(). Any thoughts? Here is my zaptel.conf loadzone=au defaultzone=au span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-21 unused=22-31 dchan=16 span=2,0,0,esf,b8zs fxols=32-55 Here is the log: -- Accepting call from '28835666' to '98857843' on channel 0/2, span 1 -- Executing [98857...@incoming:1] Answer(Zap/2-1, ) in new stack -- Executing [98857...@incoming:2] Goto(Zap/2-1, ael-queue-office-incoming|s|1) in new stack -- Goto (ael-queue-office-incoming,s,1) -- Executing [...@ael-queue-office-incoming:1] Answer(Zap/2-1, ) in new stack -- Executing [...@ael-queue-office-incoming:2] Set(Zap/2-1, quv_que_nam=office) in new stack -- Executing [...@ael-queue-office-incoming:3] Wait(Zap/2-1, 2) in new stack -- Executing [...@ael-queue-office-incoming:4] Set(Zap/2-1, cdv_sts_dbd=2) in new stack -- Executing [...@ael-queue-office-incoming:5] Set(Zap/2-1, ~~EXTEN~~=s) in new stack -- Executing [...@ael-queue-office-incoming:6] Goto(Zap/2-1, sw-104-2|10) in new stack -- Goto (ael-queue-office-incoming,sw-104-2,10) -- Executing [sw-10...@ael-queue-office-incoming:10] Set(Zap/2-1, nsv_sts_dbd=2) in new stack -- Executing [sw-10...@ael-queue-office-incoming:11] Set(Zap/2-1, nsv_div_exs=0) in new stack -- Executing [sw-10...@ael-queue-office-incoming:12] Set(Zap/2-1, ~~EXTEN~~=sw-104-2) in new stack -- Executing [sw-10...@ael-queue-office-incoming:13] Goto(Zap/2-1, sw-106-2|10) in new stack -- Goto (ael-queue-office-incoming,sw-106-2,10) -- Executing [sw-10...@ael-queue-office-incoming:10] Goto(Zap/2-1, ael-queue-office-au|s|1) in new stack -- Goto (ael-queue-office-au,s,1) -- Executing [...@ael-queue-office-au:1] SetMusicOnHold(Zap/2-1, cpwr) in new stack -- Executing [...@ael-queue-office-au:2] GotoIf(Zap/2-1, 1?3:5) in new stack -- Goto (ael-queue-office-au,s,3) -- Executing [...@ael-queue-office-au:3] Queue(Zap/2-1, office|r) in new stack -- SIP/343-098f5268 is ringing -- Channel 0/2, span 1 got hangup, cause 102 == Spawn extension (ael-queue-office-au, s, 3) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue() Ignore Hangup Request
Solution: http://bugs.digium.com/view.php?id=12655nbn=10 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Lee, John (Sydney) Sent: Wednesday, 22 April 2009 3:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Queue() Ignore Hangup Request I saw a few posts of this problem before I could not figure out the reason I am getting it. I am running RHEL 5, Asterisk 1.4.21.2, zaptel 1.4.11 and libpri 1.4.4 Basically, if I dial into a queue and hang up the phone, Asterisk did not detect the hangup request and Asterisk will only hang up when the timer expires. There is no such problem if I do not use Queue(). Any thoughts? Here is my zaptel.conf loadzone=au defaultzone=au span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-21 unused=22-31 dchan=16 span=2,0,0,esf,b8zs fxols=32-55 Here is the log: -- Accepting call from '28835666' to '98857843' on channel 0/2, span 1 -- Executing [98857...@incoming:1] Answer(Zap/2-1, ) in new stack -- Executing [98857...@incoming:2] Goto(Zap/2-1, ael-queue-office-incoming|s|1) in new stack -- Goto (ael-queue-office-incoming,s,1) -- Executing [...@ael-queue-office-incoming:1] Answer(Zap/2-1, ) in new stack -- Executing [...@ael-queue-office-incoming:2] Set(Zap/2-1, quv_que_nam=office) in new stack -- Executing [...@ael-queue-office-incoming:3] Wait(Zap/2-1, 2) in new stack -- Executing [...@ael-queue-office-incoming:4] Set(Zap/2-1, cdv_sts_dbd=2) in new stack -- Executing [...@ael-queue-office-incoming:5] Set(Zap/2-1, ~~EXTEN~~=s) in new stack -- Executing [...@ael-queue-office-incoming:6] Goto(Zap/2-1, sw-104-2|10) in new stack -- Goto (ael-queue-office-incoming,sw-104-2,10) -- Executing [sw-10...@ael-queue-office-incoming:10] Set(Zap/2-1, nsv_sts_dbd=2) in new stack -- Executing [sw-10...@ael-queue-office-incoming:11] Set(Zap/2-1, nsv_div_exs=0) in new stack -- Executing [sw-10...@ael-queue-office-incoming:12] Set(Zap/2-1, ~~EXTEN~~=sw-104-2) in new stack -- Executing [sw-10...@ael-queue-office-incoming:13] Goto(Zap/2-1, sw-106-2|10) in new stack -- Goto (ael-queue-office-incoming,sw-106-2,10) -- Executing [sw-10...@ael-queue-office-incoming:10] Goto(Zap/2-1, ael-queue-office-au|s|1) in new stack -- Goto (ael-queue-office-au,s,1) -- Executing [...@ael-queue-office-au:1] SetMusicOnHold(Zap/2-1, cpwr) in new stack -- Executing [...@ael-queue-office-au:2] GotoIf(Zap/2-1, 1?3:5) in new stack -- Goto (ael-queue-office-au,s,3) -- Executing [...@ael-queue-office-au:3] Queue(Zap/2-1, office|r) in new stack -- SIP/343-098f5268 is ringing -- Channel 0/2, span 1 got hangup, cause 102 == Spawn extension (ael-queue-office-au, s, 3) exited non-zero on 'Zap/2-1' -- Hungup 'Zap/2-1' ___ -- 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
Re: [asterisk-users] Asterisk routine maintenance activities
Daily Asterisk restart Daily log rotation Voicemail clean up for people leaving an organization. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Mutuku Sent: Wednesday, 22 April 2009 3:15 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk routine maintenance activities Hello(s), I know this might be test book question or one best suited for google but I will take the risk of asking. Here I go. What common routine maintenance tasks do you run on your asterisk box? Thanks James. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk routine maintenance activities
Daily Asterisk restart Do you think its mandatory in production env? It could be a pre-1.6 advice but I still stick to it. I did it to all my production Asterisk servers. ___ -- 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] AEL2: If-then-else not permitted in Switch-Case
Thanks guys.
It was the If vs if that was causing the problem.
This is probably due to my good coding practice of other languages in
the past :-)
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Watkins, Bradley
Sent: Thursday, 5 March 2009 9:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] AEL2: If-then-else not permitted in
Switch-
Case
I just want to confirm but it seems that if-then-else is not
permitted
in case structure.
It was not really documented but it seems to be the case.
Can anyone confirm?
No, if-then-else works fine inside a case statement. See inline
comments.
switch(${DIALSTATUS})
{
case NOANSWER:
{
This brace, and its closing-brace mate, are superfluous though not
harmful.
// if-then-else not permitted
If (${ael-var} = 1)
Your primary problem is probably right here, the if needs to be all
lower-case ( If != if ).
{
Playback(beep);
return;
}
}
Again, unnecessary.
case BUSY:
{
return;
}
default:
{
Hangup();
};
}
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[asterisk-users] AEL2: If-then-else not permitted in Switch-Case
I just want to confirm but it seems that if-then-else is not permitted
in case structure.
It was not really documented but it seems to be the case.
Can anyone confirm?
switch(${DIALSTATUS})
{
case NOANSWER:
{
// if-then-else not permitted
If (${ael-var} = 1)
{
Playback(beep);
return;
}
}
case BUSY:
{
return;
}
default:
{
Hangup();
};
}
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Re: [asterisk-users] Problems with Outbound Calls
Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' We then Chmodded everything under /dev/zap/ , rebooted and almost fell off our chairs when it worked! By right, if the problem is due to this error, you should see a permission error message in /var/log/asterisk/messages. What it means is the directory permissions might be wrong somewhere in the beginning. This may not be related to your original warning. Warning [2630]: config.c:768 process_text_line: Unknown Directive at line 231 of /etc/asterisk/../zaptel.conf We were initially on the impression that Zaptel is only used with Analogue – can anyone verify this? No, it is responsible for PRI channels as well. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Wye-khe Kwok Sent: Friday, 27 February 2009 9:13 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problems with Outbound Calls Hey, thanks for the help David, Tzafrir. Lots of config tips there ☺ We managed to find a fix through the following (For anyone who’s interested): Running /sbin/ztcfg –vv to configure Zaptel initially resulted in an error of: Notice: Configuration file is /etc/zaptel.conf line 0: Unable to open master device '/dev/zap/ctl' We then Chmodded everything under /dev/zap/ , rebooted and almost fell off our chairs when it worked! We were initially on the impression that Zaptel is only used with Analogue – can anyone verify this? YK ___ -- 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] Call Fowarding and Polycom Phone
I did not really spend too much time on looking at call forwarding and wonder if someone could help me. It seems that for setting call forwarding on the Polycom phone itself, only forward all calls will work. The other call forward function like forward if no-answer for n rings or forward if busy does not work at all on the phone. If that is the case, it seems like Asterisk and Polycom do not talk to each other well and we will have to code a dialplan for that. Is this correct? ___ -- 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] IDAP T1
What is IDAP-T1? How different is it from normal T1? Any chance I can get it to work with Digium 412P and Asterisk 1.4.* ? If yes, what would zaptel.cof look like? Any difference from normal T1 config? Thanks. ___ -- 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] meetme application
My working meetme.conf is like below. [general] [rooms] conf = 101,, conf = 102,, Your original email says your meetme.conf is: [rooms] conf = 101; If you don’t want to use passwords, I think it is better to use: [general] [rooms] conf = 101 Hope this helps! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ?? Sent: Tuesday, 10 February 2009 6:13 PM To: Asterisk Users Mailing List - No; Asterisk Users Mailing List - No Subject: Re: [asterisk-users] meetme application yes，i conf the meetme.conf [rooms] conf = 1000 any other friends can give me some advices? 2009-02-10 邱磊 ___ -- 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] What do you use? .conf or AEL?
Of course you should be using AEL. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Alan Lord (News) Sent: Tuesday, 10 February 2009 6:24 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What do you use? .conf or AEL? Hi all, I built my first asterisk using the traditional (?) .conf files and constructs. I recall reading books at the time about AEL but it seemed new and untested so I left it alone. Now, I'm interested to poll the audience here to see if I should look into using AEL instead (or in addition to) for future work. TIA ___ -- 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
Re: [asterisk-users] Improving asterisk documentation - sources andwhat the community can do
www.voip-info.org [...] So, the easiest way that people could contribute to improving Asterisk documentation right now would appear to be by improving articles on www.voip-info.org... Absolutely. What I tend to do is the make contributions to a particular page whenever I encountered a problem that is not documented in voip-info or if that part is outdated. That gives me incentive to improve that part of the doc. ___ -- 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] No Ring on Analog Phone using Rhino ChannelBank in China
There's nothing special about analogue phones in China, they are fully interchangable with analogue phones elsewhere... Perhaps you have a configuration problem, or, hardware problem on the Rhino Channel Bank, perhaps the ports are wired the wrong way and the phones care, perhaps the phones have the ringers disabled... D, thanks for replying to my problem. I contacted Rhino and they told me to just reconfigure the T1 line and it appears to fix the problem. My question is as we made zero changes to the channel bank, why do we have to reconfigure it to get it to work? Do you need to reconfigure it every now and then? ___ -- 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] No Ring on Analog Phone using RhinoChannelBank in China
I've not used Rhino kit, but, that sounds like a firmware bug that they have a workaround for... With any luck it's very infrequent and they'll be releasing a fix once they've worked out the cause... Sorry I can't help, might be best to ask Rhino about the details of the problem... The reason I was suspecting it was a country-specific problem is because I have been using the Rhino in Oz for more than 1 year and I never need to reconfigure or reboot it. ___ -- 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] No Ring on Analog Phone using Rhino Channel Bank in China
I am testing analog phone and fax machine plugged into Rhino Channel Bank which is connected to TE412P card. This site is in China. I am running RHEL 5, Asterisk 1.4.21.2, Zaptel 1.4.11 and libpri 1.4.4 I ran into a problem which is analog phone can hear dial tone and can make outgoing calls. Another phone (ether internal or external) can call the analog phone ***but the phone does not ring***. However, if the person knows that someone is calling him and picks up the analog phone, he will be able to talk to the caller. This problem does not happen in other countries which I tested before. I have tried distinctive ring tones like [ Dial(Zap/32r5,20) ] but they don't seem to make the analog phone ring. I think it has to do with the analog phone doesn't recognize the ring voltage generated by Rhino. Does anyone have experience with this? Do we have to modify the output ring voltage from the channel bank to make it work? zaptel.conf === span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 span=2,0,0,esf,b8zs fxols=32-55 loadzone=cn defaultzone=cn zapata.conf === context=incoming switchtype=euroisdn signalling=pri_cpe channel = 1-15 channel = 17-31 signalling=fxo_ls channel = 32-55 Asterisk Log +=== -- Executing [...@incoming:1] Answer(SIP/251-086bfe48, ) in new stack -- Executing [...@incoming:2] Dial(SIP/251-086bfe48, Zap/32r5|20) in new stack -- Called 32r5 -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Nobody picked up in 2 ms -- Hungup 'Zap/32-1' -- Executing [...@incoming:3] Hangup(SIP/251-086bfe48, ) in new stack == Spawn extension (incoming, 299, 3) exited non-zero on 'SIP/251-086bfe48' ___ -- 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] Agents, Queues and logon/logoff
As the subject says, I need to implement on my call center the Agent functionality, son the agents could logon and logoff to the queue How can I do this configuration? Or where can I read some info about it Here is a few links I used when I developed mine. http://www.voip-info.org/wiki/view/Asterisk+config+queues.conf http://www.voip-info.org/wiki/view/Asterisk+cmd+Queue http://etel.wiki.oreilly.com/wiki/index.php?title=Asterisk_Queues_using_ AddQueueMemberprintable=yes http://www.voip-info.org/wiki/view/Queues+with+hotdesk+agents+login+voic email+AEL+1.4 Also, because cmd AgentCallBack() is deprecated, you will have to code your own agent logon and logoff. My experience is depending on how rigorous you want your code be, it could be quite involved and as always, please code in AEL2 because it is a much better script language when AEL. ___ -- 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] Newbie Polycom: Cannot conference with 10 digit 3rd party
Calling all Polycom gurus: I am using Polycom IP601 phones with Asterisk 1.4.21.2 In all Polycom phones, I set the following in sip.cfg. dialplan dialplan.impossibleMatchHandling=2 /dialplan (I leave the digitmap unchanged because I thought setting impossibleMatchHandling will ignore the bitmap) ...so that I could dial any number by entering a variable-size telephone number and then hit the send or dial key. This works quite well except when I am doing conferencing. It goes like this: I dialled the 1st party and was answered. Then I press conf key and then enter the 3rd party. I can keep entering until it reaches the 10th digit and then the 10-digit number is automatically dialled. Any thoughts? ___ -- 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] music on hold
The reason is your audio file is too high quality. Asterisk can only play back audio file of 4000Hz. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ?? Sent: Tuesday, 11 November 2008 5:35 PM To: asterisk-users Subject: [asterisk-users] music on hold hii guys: i get the message from the asterisk: Started music on hold, class 'default', on Local/[EMAIL PROTECTED],1 [2008-11-11 14:32:41] WARNING[1781]: format_wav.c:156 check_header: Unexpected freqency 11025 [2008-11-11 14:32:41] WARNING[1781]: file.c:322 fn_wrapper: Unable to open format wav [2008-11-11 14:32:41] WARNING[1781]: res_musiconhold.c:259 ast_moh_files_next: Unable to open file '/data/TOMSKYPEIVR/var/lib/asterisk/moh/callback1': No such file or directory -- Stopped music on hold on Local/[EMAIL PROTECTED],1 how can i solve the issue? thanks 2008-11-11 邱磊 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Machine Hang after calling in/out ISDN
I am testing Asterisk 1.4.22, zaptel 1.4.10.1 and libpri 1.4.4 on RHEL5 on DELL PE2950 and using ISDN-10. What device? I am using TE412P. No message on the console of the machine? Yes, nothing at all. The machine just froze and had to be rebooted. This probably means one of two things: 1. Bad kernel-level deadlock (maybe caused by Zaptel) I will upgrade zaptel to the latest version. 2. If asterisk is running with -p: it might be in a 100% CPU loop. I just use whatever it is in /etc/init.d/asterisk. I checked the file and it does not come with a -p option. I checked /var/log/asterisk but there is nothing unusual I can see. Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk/Machine Hang after calling in/out ISDN
I am testing Asterisk 1.4.22, zaptel 1.4.10.1 and libpri 1.4.4 on RHEL5 on DELL PE2950 and using ISDN-10. I thought about cutting over to production tonight when I noticed a serious problem. SIP calls are fine but if I dialed to outside (Dial(Zap/g1)) a few times or someone called in a few times, Asterisk just froze (cannot enter anything on the CLI console) and then even the machine had to be rebooted. I suspect there is a problem with zaptel. Is zaptel 1.4.10.1 a dodgy verions? Any suggestion on how to debug this problem? ___ -- 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] realtime queue_log to mySQL backport to 1.4
I did not know what I did but I bumped into something in the log that
says:
[Oct 16 ...] ERROR[24536] res_config_mysql.c: MySQL RealTime: Ping
failed
(2006). Trying an explicit reconnect.
[Oct 16 ...] DEBUG[24536] res_config_mysql.c: MySQL RealTime: Server
Error
(2006): MySQL server has gone away
[Oct 16 ...] DEBUG[24536] res_config_mysql.c: MySQL RealTime:
Successfully
connected to database.
However, I believe the problem has something to do with MySQL refusing
to
talk to Asterisk.
That was my wrong assumption.
I checked res_config_mysql.c and the comments says:
/* MySQL likes to return an error, even if it reconnects successfully.
* So the postman pings twice. */
if (mysql_ping(mysql) != 0 mysql_ping(mysql) != 0) {...}
So, at this stage, my res_config_mysql.c is still not writing anything
into table queue_log despite having: a) correct res_mysql.conf b)
extconfig.conf c) mysql up and running d) res_config_mysql.c start up
okay
I believe that it is because the following if condition in logger.c is
never met:
***if (ast_check_realtime(queue_log))***
{
va_start(ap, fmt);
vsnprintf(qlog_msg, sizeof(qlog_msg), fmt, ap);
va_end(ap);
snprintf(time_str, sizeof(time_str), %ld, (long)time(NULL));
ast_store_realtime(queue_log, time, time_str, callid, callid,
queuename, queuename, agent, agent, event,
event, data, qlog_msg, NULL);
Does anyone know what does ast_check_realtime do?
Is there a developer mailing list I can try?
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Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4
Also i would suggest enabling full log, as it's one place you can see everything. Then use grep to search for realtime messages. Your logger.conf should already have commented line: full = notice,warning,error,debug,verbose Yes, I did that. # tail -fn0 /var/log/asterisk/full | grep -F res_config_mysql to see every message about realtime driver. Still, I did not get any hit on res_config_mysql when I do an AddQueueMember or RemoveQueueMember. When I restarted Asterisk, I saw: [Oct 16 ...] DEBUG[24880] res_config_mysql.c: MySQL RealTime Host: localhost [Oct 16 ...] DEBUG[24880] res_config_mysql.c: MySQL RealTime Port: 3306 [Oct 16 ...] DEBUG[24880] res_config_mysql.c: MySQL RealTime User: uid [Oct 16 ...] DEBUG[24880] res_config_mysql.c: MySQL RealTime Password: pwd [Oct 16 ...] DEBUG[24880] res_config_mysql.c: MySQL RealTime: Successfully connected to database. [Oct 16 ...] NOTICE[24880] config.c: Registered Config Engine mysql [Oct 16 ...] VERBOSE[24880] logger.c: MySQL RealTime driver loaded. [Oct 16 ...] VERBOSE[24880] logger.c: res_config_mysql.so = (MySQL RealTime Configuration Driver) They all looked fine on startup. I did not know what I did but I bumped into something in the log that says: [Oct 16 ...] ERROR[24536] res_config_mysql.c: MySQL RealTime: Ping failed (2006). Trying an explicit reconnect. [Oct 16 ...] DEBUG[24536] res_config_mysql.c: MySQL RealTime: Server Error (2006): MySQL server has gone away [Oct 16 ...] DEBUG[24536] res_config_mysql.c: MySQL RealTime: Successfully connected to database. Then I recalled seeing something on the Internet about MySQL timing out on connection from Asterisk. I created a /etc/my.cnf and bung in something like below and restarted both mysql and Asterisk. [mysqld] ... wait_timeout=60 connect_timeout=10 interactive_timeout=120 but it still does not work. However, I believe the problem has something to do with MySQL refusing to talk to Asterisk. The funny thing is I never had any problem with cdr_addon_mysql.so Maybe they work using different methodologies I reckon? Any more thoughts? ___ -- 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] realtime queue_log to mySQL backport to 1.4
Hi Atis, queue_log = mysql,asteriskcdrdb,queue_log that is engine,database,table If it's wrong, you should see some warnings when asterisk is starting up. Thanks for the suggestion. I did not put in queue_log for table and it has just taken the default which is queue_log. In the console startup, you can see below that it has successfully bound queue_log to /mysql/db1/queue_log. # asterisk -rvvv Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. [...] == Parsing '/etc/asterisk/extconfig.conf': Found == Binding queue_log to mysql/db1/queue_log Connected to Asterisk 1.4.21.2 currently running on machine Verbosity is at least 3 In /var/log/asterisk/messages, I saw: [Oct 15 15:31:48] NOTICE[20941] config.c: Registered Config Engine mysql Another idea that came into my mind is, that (if this config doesn't still work) you might have to do make dist-clean within asterisk-addons after reinstalling asterisk, and then configure, make, make install. It's because addons do use headers from installed version of asterisk, and they might not have correct declarations. Basically, I did: - Asterisk-1.4.21.2 make clean ./configure make make install - Asterisk-addons-1.4.7 make dist-clean ./configure make make install Also, you mentioned that you checked /var/log/asterisk/messages, however i think debug is written into file called debug. Anyway you can enable full in logger.conf and get everything there. To debug this you shouldn't need more than core set verbose 3 and core set debug 1. I turned on debug mode and tried an agent login and logoff. However, when I looked into debug and messages, there are lots of chan_sip.c and a few cdr_addon_mysql.c but no occurrence at all of res_config_mysql.c What is happening? Do I have to explicitly load it? *CLI module show like res_config_mysql Module Description Use Count res_config_mysql.so MySQL RealTime Configuration Driver 0 1 modules loaded ___ -- 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] realtime queue_log to mySQL backport to 1.4
You might want to double check the socket path. Some distributions use /var/run/mysqld/mysqld.sock as the socket file. Thanks for the suggestion Tilghman. I am using Redhat and the socket file is indeed /var/run/mysql/mysqld.sock. Actually, if you specify the wrong socket file, you will see an mySQL Realtime error message in /var/log/asterisk/messages. ___ -- 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] realtime queue_log to mySQL backport to 1.4
http://ftp.iq-labs.net/queue_log- 1.4/asterisk_queue_log_realtime_1.4.19.patch This uses standardized realtime/mysql library from asterisk addons. For it to support SQL inserts in 1.4, you would also need to apply both patches from (1 for asterisk, another for asterisk-addons) http://ftp.iq-labs.net/realtime_store_destroy-1.4/ This will later allow you to upgrade to 1.6 and having everything working without patching. I have patched in asterisk 1.4 . main/logger.c . include/asterisk/config.h . main/config.c I have patched in asterisk-addons 1.4 . res/res_config_mysql.c I have re-installed asterisk and asterisk-addons. I created a database called db1 and in there created a table called queue_log as per instruction http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL I changed /etc/asterisk/extconfig.conf to add the following line: [settings] queue_log = mysql,db1 I changed /etc/asterisk/res_mysql.conf to add the following: [general] dbhost = localhost dbname = db1 dbuser = user dbpass = password dbport = 3306 dbsock = /var/lib/mysql/mysql.sock 1) However, whenever I perform an agent login, no row is written to table queue_log. I checked /var/log/asterisk/queue_log and a new entry is written there. 2) I set debug to 10 on the console in asterisk and re-did the test but there were no error messages in /var/log/asterisk/messages. 3) I set debug on in mysqld and there are no information for inserting into table queue_log, except the cdr logging as below. Tcp port: 0 Unix socket: (null) Time Id CommandArgument 081013 15:59:36 1 Connect [EMAIL PROTECTED] on db1 2 Connect [EMAIL PROTECTED] on db1 081013 16:00:32 1 Query INSERT INTO cdr_log ... 081013 16:01:42 1 Query INSERT INTO cdr_log ... Is there anyone who can help me? ___ -- 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] realtime queue_log to mySQL backport to 1.4
Yes, I certainly applied the patch in http://ftp.iq-labs.net/queue_log-1.4/asterisk_queue_log_realtime_1.4.19. patch Just to double-check, there is only one patch in this URL which is main/logger.c By the way, did you see anything wrong with my config files? /etc/asterisk/extconfig.conf [settings] queue_log = mysql,db1 /etc/asterisk/res_mysql.conf [general] dbhost = localhost dbname = db1 dbuser = user dbpass = password dbport = 3306 dbsock = /var/lib/mysql/mysql.sock -Original Message- From: Atis Lezdins [mailto:[EMAIL PROTECTED] Sent: Monday, 13 October 2008 8:02 PM To: Lee, John (Sydney) Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] realtime queue_log to mySQL backport to 1.4 Hi John, On Mon, Oct 13, 2008 at 9:51 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: http://ftp.iq-labs.net/queue_log- 1.4/asterisk_queue_log_realtime_1.4.19.patch Haven't you forgotten this one? ;) if you have applied everything correctly - queue_log file shoudln't have any more lines (except init when restarting asterisk). Regards, Atis This uses standardized realtime/mysql library from asterisk addons. For it to support SQL inserts in 1.4, you would also need to apply both patches from (1 for asterisk, another for asterisk-addons) http://ftp.iq-labs.net/realtime_store_destroy-1.4/ This will later allow you to upgrade to 1.6 and having everything working without patching. I have patched in asterisk 1.4 . main/logger.c . include/asterisk/config.h . main/config.c I have patched in asterisk-addons 1.4 . res/res_config_mysql.c I have re-installed asterisk and asterisk-addons. I created a database called db1 and in there created a table called queue_log as per instruction http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL I changed /etc/asterisk/extconfig.conf to add the following line: [settings] queue_log = mysql,db1 I changed /etc/asterisk/res_mysql.conf to add the following: [general] dbhost = localhost dbname = db1 dbuser = user dbpass = password dbport = 3306 dbsock = /var/lib/mysql/mysql.sock 1) However, whenever I perform an agent login, no row is written to table queue_log. I checked /var/log/asterisk/queue_log and a new entry is written there. 2) I set debug to 10 on the console in asterisk and re-did the test but there were no error messages in /var/log/asterisk/messages. 3) I set debug on in mysqld and there are no information for inserting into table queue_log, except the cdr logging as below. Tcp port: 0 Unix socket: (null) Time Id CommandArgument 081013 15:59:36 1 Connect [EMAIL PROTECTED] on db1 2 Connect [EMAIL PROTECTED] on db1 081013 16:00:32 1 Query INSERT INTO cdr_log ... 081013 16:01:42 1 Query INSERT INTO cdr_log ... Is there anyone who can help me? -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- 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] realtime queue_log to mySQL backport to 1.4
if you have applied everything correctly - queue_log file shoudln't
have any more lines (except init when restarting asterisk).
Thanks Atis.
I see what you are saying. In the patch for logger.c,
The code to write to mysql is there except that we need to perform
ast_check_realtime(queue_log).
I guess ast_check_realtime() is looking into extconfig.conf and
searching for
queue_log = mysql,db1
which is there in my extconfig.conf already.
Can any Asterisk developers enlighten me on this?
void ast_queue_log(const char ...)
{
+ char qlog_msg[8192];
+ char time_str[16];
+
+ if (ast_check_realtime(queue_log)) {
va_start(ap, fmt);
+ vsnprintf(qlog_msg, sizeof(qlog_msg), fmt, ap);
va_end(ap);
+
+ snprintf(time_str, sizeof(time_str), %ld,
(long)time(NULL));
+ ast_store_realtime(queue_log, time, time_str,
+ callid, callid,
+ queuename, queuename,
+ agent, agent,
+ event, event,
+ data, qlog_msg,
+ NULL);
+ } else {
+ if (qlog) {
+ AST_LIST_LOCK(logchannels);
+ va_start(ap, fmt);
+ fprintf(qlog, %ld|%s|%s|%s|%s|,
(long)time(NULL), callid, queuename, agent, event);
[...]
+ }
}
-Original Message-
From: Atis Lezdins [mailto:[EMAIL PROTECTED]
Sent: Monday, 13 October 2008 8:02 PM
To: Lee, John (Sydney)
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] realtime queue_log to mySQL backport to
1.4
Hi John,
On Mon, Oct 13, 2008 at 9:51 AM, Lee, John (Sydney)
[EMAIL PROTECTED] wrote:
http://ftp.iq-labs.net/queue_log-
1.4/asterisk_queue_log_realtime_1.4.19.patch
Haven't you forgotten this one? ;)
if you have applied everything correctly - queue_log file shoudln't
have any more lines (except init when restarting asterisk).
Regards,
Atis
This uses standardized realtime/mysql library from asterisk addons.
For it to support SQL inserts in 1.4, you would also need to apply
both patches from (1 for asterisk, another for asterisk-addons)
http://ftp.iq-labs.net/realtime_store_destroy-1.4/
This will later allow you to upgrade to 1.6 and having everything
working without patching.
I have patched in asterisk 1.4
. main/logger.c
. include/asterisk/config.h
. main/config.c
I have patched in asterisk-addons 1.4
. res/res_config_mysql.c
I have re-installed asterisk and asterisk-addons.
I created a database called db1 and in there created a table called
queue_log as per instruction
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
I changed /etc/asterisk/extconfig.conf to add the following line:
[settings]
queue_log = mysql,db1
I changed /etc/asterisk/res_mysql.conf to add the following:
[general]
dbhost = localhost
dbname = db1
dbuser = user
dbpass = password
dbport = 3306
dbsock = /var/lib/mysql/mysql.sock
1) However, whenever I perform an agent login, no row is written to
table queue_log. I checked /var/log/asterisk/queue_log and a new
entry
is written there.
2) I set debug to 10 on the console in asterisk and re-did the test
but
there were no error messages in /var/log/asterisk/messages.
3) I set debug on in mysqld and there are no information for
inserting
into table queue_log, except the cdr logging as below.
Tcp port: 0 Unix socket: (null)
Time Id CommandArgument
081013 15:59:36 1 Connect [EMAIL PROTECTED] on db1
2 Connect [EMAIL PROTECTED] on db1
081013 16:00:32 1 Query INSERT INTO cdr_log ...
081013 16:01:42 1 Query INSERT INTO cdr_log ...
Is there anyone who can help me?
--
Atis Lezdins,
VoIP Project Manager / Developer,
[EMAIL PROTECTED]
Skype: atis.lezdins
Cell Phone: +371 28806004
Cell Phone: +1 800 7300689
Work phone: +1 800 7502835
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[asterisk-users] Compile logger-mysql.c with UNDEFINED REF to `mysql_error'
Sorry to post a C compile error on this mailing list but this is Asterisk related. Basically, I was following http://www.plack.net/index.php/2007/01/07/asterisk_modification_for_queu e_logging to patch logger.c and Makefile in Asterisk 1.4.* in order to write queue_log to mySQL database. When I ran make, it complained: In function `write_mysql_logger': [...] /usr/src/asterisk-1.4.21.2/main/logger-mysql.c:98: undefined reference to `mysql_error' [...] collect2: ld returned 1 exit status make[1]: *** [asterisk] Error 1 make: *** [main] Error 2 In my modified Makefile, I already had the line: ASTCFLAGS+=-I/usr/include/mysql and I found that mysql.h is already in /usr/include/mysql. I also already had mysql-client installed. In logger-mysql.c, there is already a line at the front of the program: #include mysql.h Any thoughts? ___ -- 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] Compile logger-mysql.c with UNDEFINED REF to`mysql_error'
This looks really old and weird. I could suggest using realtime queue_log backport from 1.6 which i'm currently using. That's good info, Atis. I will definitely give it a go. winmail.dat___ -- 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] can't find mysqlclient : asterisk-addons-1.6.0
Yes, unfortunately, VOIP wiki did not mention about installing mysql-client which it should have been. Without mysql-client, you cannot change passwords, grants, etc. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Stefan Schmidt Sent: Tuesday, 7 October 2008 6:14 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] can't find mysqlclient : asterisk-addons- 1.6.0 Klaverstyn, David C schrieb: Hi All, I can not install the asterisk-addons as it thinks there is no mysqlclient installed. I have installed mysql, mysql-server and mysql-devel and I am still unable to install the addons. I am running CentOS 5.2 i386. Please somebody help. Hello, maybe you should install mysql-client too ;) best regards steve smith ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie AEL2: Syntax for Hint
Steve, I downloaded the latest Asterisk version (see below). *CLI core show version Asterisk 1.4.21.2 built by root @ machine1 on a i686 running Linux on 2008-09-11 06:10:06 UTC If I code: Hint(Custom:light1) It will pass aelparse but when it runs, it says Hint is an unknown application on the console. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Murphy Sent: Thursday, 11 September 2008 2:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie AEL2: Syntax for Hint On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote: I am struggling to find out how to code hint in AEL2. I did hint(Custom:light1) and it keeps complaining about the : (colon). It works fine for SIP device like hint(SIP/439). Anyone who has tried it before? Yes, a while back I upgraded AEL to handle both ':' and '' inside the hint parens. This should work on 1.4 on up. What version of asterisk are you using? 1.2? murf -- Steve Murphy Software Developer Digium ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie AEL2: Syntax for Hint
context BLF {
hint(Sip/1000) 1000 = NoOp();
};
Works for me
Thanks Eric.
I did not experience any problem in hint with SIP. The problem is if you use
it with Custom.
winmail.dat___
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[asterisk-users] Newbie AEL2: Syntax for Hint
I am struggling to find out how to code hint in AEL2. I did hint(Custom:light1) and it keeps complaining about the : (colon). It works fine for SIP device like hint(SIP/439). Anyone who has tried it before? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie AEL2: Syntax for Hint
*CLI core show version Asterisk 1.4.13 built by root @ machine1 on a i686 running Linux on 2008-09-10 06:46:17 UTC Thanks Steve. What syntax should I use then? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Murphy Sent: Thursday, 11 September 2008 2:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Newbie AEL2: Syntax for Hint On Wed, 2008-09-10 at 18:10 +1000, Lee, John (Sydney) wrote: I am struggling to find out how to code hint in AEL2. I did hint(Custom:light1) and it keeps complaining about the : (colon). It works fine for SIP device like hint(SIP/439). Anyone who has tried it before? Yes, a while back I upgraded AEL to handle both ':' and '' inside the hint parens. This should work on 1.4 on up. What version of asterisk are you using? 1.2? murf -- Steve Murphy Software Developer Digium ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] extensions.conf programming?
A cheaper alternative would be the voip wiki. http://www.voip-info.org/tiki- index.php?page=Asterisk%20config%20extensions.conf Unfortunately, as advised by other asterisk users, http://www.voip-info.org is sometimes really not that up-to-date. However, that does not mean that we should give up on using and updating http://www.voip-info.org because I think it is still the best voip resource. The best way is still to double check with the asterisk version that you have installed by running CLI like below: *CLI core show function AGENTARRAYBASE64_DECODE BASE64_ENCODEBLACKLISTCALLERID [...] VMCOUNT *CLI core show application AddQueueMember ADSIProg AgentCallbackLogin [...] ZapScan ZapSendKeypadFacility *CLI -= Info about application 'WaitExten' =- [Synopsis] Waits for an extension to be entered [Description] WaitExten([seconds][|options]): This application waits for the user to enter a new extension for a specified number of seconds. Note that the seconds can be passed with fractions of a second. For example, '1.5' will ask the application to wait for 1.5 seconds. Options: m[(x)] - Provide music on hold to the caller while waiting for an extension. Optionally, specify the class for music on hold within parenthesis. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF - multiple buddies
I believe that this is what I need to enable more than one buddy icon? Can you please point me in the right direction. Only the polycom screen, I can only see 1 buddy icon despite having 2 speed dial entries. I have been able to successfully turned on presence (which is the term used outside Polycom) on IP601. As I can recall, you need to a) configure sip.conf in the [general] and per [extn] context; b) code hint extn in extensions.conf c) turned on presence on the phone which will be buddy watching others d) turn on bw on the phone which I saw you did. However, I have never set what you did as in below and have no idea what they are. In the phone1.cfg file I set: attendant attendant.uri=4158149992 attendant.reg=1/ Just check out voip wiki and there are useful information over there about presence (but may not be that much about Polycom phones sadly :-(. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF - multiple buddies
It's not perfect, because it doesn't display DND or queue login/pause status, but it's better than nothing. James, on a different note, is it true that at this stage, we can never get any queue login status/light on Polycom phone? I posted a query a few days ago but I have got 0 reply. Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom BLF - multiple buddies
Sorry, needed to add one more note. To clarify, my agent phones have a speed dial assigned for their login, and another to pause/unpause. I could then use DEVSTATE to enable or disable the light next to their speed dial button based on their status. I can't use it to update anything on the LCD screen. James, very useful info especially about enable/disable the light next to the speed dial button which is exactly what I am after. I am currently using 1.4.x and would be interested to know how this can be achieved. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie Polycom: ACD AgentLogin display on phone
I have been coding my own IVR for ACD (aka queue) using Polycom phones using AEL2. In particular, I have coded my own AgentCallbackLogin because a) cmd AgentCallbackLogin() is buggy and will not be supported by dev anymore b) I can put in features like hotdesking and additional validation like prohibiting repeated logins and current phone already logged on by other agent and so forth. Having said that, that still leaves one feature not available which is a visible display on the Polycom phone that an agent has already logged on to the phone. I searched the mailing list up and low and there were some sketchy notes about bweschke had developed a patch which could understand the acd-login-logout of Polycom phones. However, I hope someone can answer the following questions for me. a) Is bweschke's patch available in the current version or do we have to download and install it separately? b) Does bweschke's patch only interface with the AgentLogin() command? In other words, after we enabled the acd-login-logout parameters in the Polycom config files and we pressed the key on the phone, will the phone then basically initiate an AgentLogin() command to the Asterisk server? And does the light beside the key shows red to signify that an agent has logged on successfully. c) I have coded my own Agent Login and Logout extension and it would be great if the softkey could call my own agent login and logout extension (this bit is easy) and then showing the red light if it is a successful login (hard?). Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallbackLogin AddQueueMember
Just out of curiosity, where do you get this AddQueueMember syntax from? Here: http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.c om /books/9780596510480.pdf page: 367 Oh so the VOIP Wiki is out of date! Now, where should we go to for reliable Asterisk info then? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallbackLogin AddQueueMember
I need login Agent(Member) in asterisk. use this option: for example: AddQueueMember(queuetest,SIP/ekiga,10,,Agent/13) Just out of curiosity, where do you get this AddQueueMember syntax from? http://www.voip-info.org/wiki/view/Asterisk+cmd+AddQueueMember Description: AddQueueMember(queuename[|interface][|penalty]): ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] music on hold is not working
I have made class for MOH uploaded a mp3 file to the folder. Now I am using this class for music on hold during dialing. Now when call has been established, I put the other end on hold. So from that end I should listen uploaded file. But I am not getting audio. From memory, you need to install asterisk-addons in order to play mp3 file. The default audio file is .gsm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Console softphone
Hello all! Is there a way to (mis)use asterisk itself as a softphone? Can I make a call from within the CLI? Can asterisk from itself produce a ringtone? I Or can bind a system-command to incoming calls? Any help is sincerely appreciated! You can install a browser softphone on the same server and make calls from any browser. Better still - is it possible to SSH (or some sort of connection method) from a remote PC to the Asterisk server and make a call using CLI? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Console softphone
Better still - is it possible to SSH (or some sort of connection method) from a remote PC to the Asterisk server and make a call using CLI? Sure, you can use the CLI 'console dial' command. Do you mean that I will be able to hear the call from my PC if I do 'console dial' on the remote Asterisk server provided that I install browser softphone on the server? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: Queue and CDR Reporter and Analyser
Doesn't Queuemetrics run on a license basis? Anything else that's probably open source and free? Does anyone have any comments/experience about using asteriskguru queue statistics? http://www.asteriskguru.com/tutorials/installation_guide.html ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] remove queue call
2700 has 1 calls (max unlimited) in 'rrmemory' strategy (35s holdtime), W:0, C:134, A:48, SL:88.8% within 120s Members: Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Local/[EMAIL PROTECTED] (dynamic) (Not in use) has taken no calls yet Callers: 1. Local/[EMAIL PROTECTED],2 (wait: 113:19, prio: 0) Can you try ... CLI module reload app_queue.so CLI reload CLI restart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
