Re: [asterisk-users] G729 CPU Utilization
hello: it really depends on number of calls transcded. it the number is less 20 calls, the CPU should be ok. but it is very sure that software based take take much more than hardware based transcoder. ? 2013-9-9 15:53, Gopalakrishnan N ??: Hi, How much CPU utilization will it take when I use G729 transcoding via hardware based transcoder. Regards -- _ -- 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 -- Best regards! Connect with Sangoma in APAC www.hiastar.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using E1 PRI lines
hello:you can add T1_E1 by load card drivers Best regards, James.zhu website: www.hiastar.com From: akibsay...@gmail.com Date: Mon, 29 Jul 2013 21:48:19 +0530 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] using E1 PRI lines On Mon, Jul 29, 2013 at 9:20 PM, Gareth Blades mailinglist+aster...@dns99.co.uk wrote: On 29/07/13 16:28, Akib Sayyed wrote: Dear asterisk users I wanted to use E1 pri lines on my asterisk box but my provider support only 120ohm on E1 line. I dont know how to set those values. Please help me Its done on whatever interface cards you have. Some may have a jumper setting. I know Sangoma has it in their configuration file (wanpipe). I am using digium card TE410P. can anyone help me how to change jumper settings -- _ -- 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 -- Akib Sayyed Matrix-Shell akibsay...@gmail.com akibsay...@matrixshell.com Mob:- +91-966-514-2243 -- _ -- 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] CTI for asterisk?
hello: you can look for the queue. On 2013-7-15 10:01, bilal ghayyad wrote: Hello; Is there CTI module in asterisk with CTI client to login and logout and do ready and pause? Regards Bilal -- _ -- 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 -- Best regards! Connect with Sangoma in APAC www.hiastar.com -- Best regards! Connect with Sangoma in APAC www.hiastar.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] A quick question in terms of DAHDI channel
hi:you have to install libpri,dahdi and asterisk for E1 cards. Best regards, James.zhuVega VOIP gateways, Sangoma Asterisk cards, SBC, NetBorder VOIP Gateway, LYNC-TDM, Transcodingwebsite: www.hiastar.com From: longst...@gmail.com Date: Thu, 13 Jun 2013 10:31:28 +0200 To: asterisk-users@lists.digium.com Subject: [asterisk-users] A quick question in terms of DAHDI channel Hello, I have an Asterisk 1.8.11 installation. When I built up this Asterisk, I didn't install DAHDI channel, if I issue command connect*CLI core show channeltypes I would have response like: connect*CLI core show channeltypes TypeDescription Devicestate Indications Transfer -- --- --- --- USTMUNISTIM Channel Driver no yes no Phone Standard Linux Telephony API Driver no yes no Console OSS Console Channel Driver no yes no Skinny Skinny Client Control Protocol (Skinny) yes yes no Local Local Proxy Channel Driver yes yes no SIP Session Initiation Protocol (SIP)yes yes yes Agent Call Agent Proxy Channel yes yes no MGCPMedia Gateway Control Protocol (MGCP)yes yes no IAX2Inter Asterisk eXchange Driver (Ver 2) yes yes yes MulticastR Multicast RTP Paging Channel Driver no no no Bridge Bridge Interaction Channel no no no -- 11 channel drivers registered. But right now, I am planing to connect a PRI trunk to this Asterisk. so I put in a PRI card install dahdi-linux-complete-2.6.2 and libpri-1.4.14. Afterward, dahdi_tool is able to find PRI board, and all channels. But my question is when I try to send call to DAHDI channel in the dial plan, CLI print out a warning saying [Jun 13 07:48:21] WARNING[2393]: channel.c:5606 ast_request: No channel type registered for 'DAHDI' According to my description above, it make sense, since my Asterisk does not install DAHDI channel before. Therefore my question is in my case, it is required to re-intall whole Asterisk, or there is some other way that I just could only install DAHDI channel. I did some google search. but I didn't find a proper answer. Thanks for your help. longst -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] why number type always changed from subscriber user to national in libpri
hello:I used libpri 1.4.12 version with asterisk 1.8.7, after the pcap files, i found thatin wireshark setup message, the number type always changed from subscriber to national number.i have set pridialplan= local and prilocaldialplan=local in chan_dahdi.conf already. because that, the systemsometimes can not make outgoing calls. anyone can clarify that? Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk/sangoma cards, recording device, VOIP gateway. website: www.hiastar.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] question sangoma vs digium
hello: i think it can be done, please refer this link: http://wiki.sangoma.com/Asterisk-FAQ#Digium Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Wed, 4 Jan 2012 18:47:28 -0200 From: agustina.berre...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] question sangoma vs digium Hi! Hello! I wanted to know if you have experienced problems installing both a Sangoma and a Digium card in the same Server. Thnks a lot! -- _ -- 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] Asterisk 1.8 warns for lines starting with # in /etc/dahdi/system.conf
hello:it is an error. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: r...@linux-delhi.org To: asterisk-users@lists.digium.com Date: Fri, 23 Dec 2011 07:36:32 +0530 Subject: Re: [asterisk-users] Asterisk 1.8 warns for lines starting with # in /etc/dahdi/system.conf On Thursday 22 Dec 2011, Olivier wrote: Testing 1.8.8.0 (with Dahdi 2.5.0.2 and asterisk-gui 2.1.0-rc1), I'm seeing this on my console: WARNING[25363]: config.c:1208 process_text_line: Unknown directive '#' at line 1 of /etc/asterisk/../dahdi/system.conf This warning is repeated for every line starting with a # char. Shall I care ? Suggestions ? (To me, /etc/asterisk/../dahdi/system.conf belongs to Dahdi, not to Asterisk) Use ; instead of # to comment dahdi/system.conf. If it's auto- generated, perhaps this would help after the auto-generation: sed -i -e 's/^#/;/' /etc/dahdi/system.conf Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- 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] Set Caller Number in E1 PRI ISDN Lines
hello: please check callerid= in chan_dahdi.conf Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: kaushalshri...@gmail.com Date: Sat, 17 Dec 2011 06:08:05 +0530 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Set Caller Number in E1 PRI ISDN Lines Hi, I am having an E1 PRI ISDN Lines with 30 bearer channels and 1 D Channel with hundred DIDs (Direct Inward Dialing) numbers attached to a Sangoma PRI Card on the server, I am using asterisk 1.8.5 on CentOS 5.6. How can i configure DIDs so that if i make an outgoing call the DID number should go to the caller not the pilot number For example PRI Numbers Range - 31303000 - 31303099 Pilot Number - 31303000 So if i need to set caller number as 31303008 for example and not as 31303000, is there a way to set this in dial plan (extensions.conf) Please guide and let me know if anyone needs more information and have questions. Regards, Kaushal -- _ -- 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] Help needed for chan_ss7 for Digium device
hello: you can refer this link: http://mirror.su.lt/voip-info/wiki/view/Asterisk+ss7+channels.html Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Mon, 12 Dec 2011 20:21:36 +0530 From: max.aster...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Help needed for chan_ss7 for Digium device Hi All, I have installed centos 5.6 32 bit on xeon server and i have also installed latest version of asterisk 1.6 and dahdi as well. I want to install chan_ss7 for this server and I want to know about the following device. Digium TE420B I dont know much about the configuration files for Digium TE420B. Can anybody provide me required ss7.conf file and also provide dahdi configuration which is needed for this device. Thanks you so much in advance!! Thanks, Max Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Sporadic yellow alarms in dahdi_tool output
hi:please ask your provider to check the connection. yes, RAI means there is a problem with remote side. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Fri, 25 Nov 2011 18:48:33 +0530 From: ish...@exotel.in To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sporadic yellow alarms in dahdi_tool output Hi all, We have a telephony server in India which runs CentOS release 5.7 (Final) version with four-span Digium Card, one of which has an E1 PRI line terminating on the server.$ dahdi_hardware pci::07:08.0 wct4xxp+ d161:1420 Wildcard TE420 (5th Gen) Recently we've been observing the status of Span 1 as observed from dahdi_tool output flaps between yellow and green. This happens a couple of few a day. While the span state is yellow, I observed that the modules dahdi_echocan_mg2 and wct4xxp are not loaded. After running sudo modprobe wct4xxp; sudo modprobe dahdi_echocan_mg2, the span state changes to green. The dahdi documentation has the following to say on yellow alarms: (RAI — Remote Alarm Indication)Your T1/E1 port will go into yellow alarm when it receives a signal from the remote switch that the port on that remote switch is in red alarm. This essentially means that the remote switch is not able to maintain sync with you, or is not receiving your transmission. As a newbie to asterisk, pointers will be very helpful. Here are some more details on the installed versions of various packages/modules. 1. Linux Kernel Version : $ uname -aLinux 022.exotel.in 2.6.18-274.3.1.el5 #1 SMP Tue Sep 6 20:13:52 EDT 2011 x86_64 x86_64 x86_64 GNU/Linux2. Dahdi verison: dahdi-linux-complete-2.5.0.1+2.5.0.1 3. Asterisk verison: asterisk-1.8.6.04. libpri version: libpri-1.4.12 Is this a configuration issue, package version issue or an operator issue? I'll be glad to share the necessary dahdi configuration files if needed. Thanks in advance. --Cheers,Ishwar. -- _ -- 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] Sporadic yellow alarms in dahdi_tool output
hi:you have to add that in system.conf, check the file, by default is is OSLEC, you can change to MG2.please refer this link fore the system.conf:http://www.voip-info.org/wiki/view/system.conf Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Fri, 25 Nov 2011 20:13:04 +0530 From: ish...@exotel.in To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sporadic yellow alarms in dahdi_tool output Thanks for the response James. I shall talk with the provider.If the problem resides with the remote side, how does loading dahdi_echocan_mg2 and wct4xxp modules fix the issue, atleast temporarily? --Regards,Ishwar. On Fri, Nov 25, 2011 at 8:04 PM, James zhu zhulizh...@live.com wrote: hi:please ask your provider to check the connection. yes, RAI means there is a problem with remote side. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Fri, 25 Nov 2011 18:48:33 +0530 From: ish...@exotel.in To: asterisk-users@lists.digium.com Subject: [asterisk-users] Sporadic yellow alarms in dahdi_tool output Hi all, We have a telephony server in India which runs CentOS release 5.7 (Final) version with four-span Digium Card, one of which has an E1 PRI line terminating on the server.$ dahdi_hardware pci::07:08.0 wct4xxp+ d161:1420 Wildcard TE420 (5th Gen) Recently we've been observing the status of Span 1 as observed from dahdi_tool output flaps between yellow and green. This happens a couple of few a day. While the span state is yellow, I observed that the modules dahdi_echocan_mg2 and wct4xxp are not loaded. After running sudo modprobe wct4xxp; sudo modprobe dahdi_echocan_mg2, the span state changes to green. The dahdi documentation has the following to say on yellow alarms: (RAI — Remote Alarm Indication)Your T1/E1 port will go into yellow alarm when it receives a signal from the remote switch that the port on that remote switch is in red alarm. This essentially means that the remote switch is not able to maintain sync with you, or is not receiving your transmission. As a newbie to asterisk, pointers will be very helpful. Here are some more details on the installed versions of various packages/modules. 1. Linux Kernel Version : $ uname -aLinux 022.exotel.in 2.6.18-274.3.1.el5 #1 SMP Tue Sep 6 20:13:52 EDT 2011 x86_64 x86_64 x86_64 GNU/Linux 2. Dahdi verison: dahdi-linux-complete-2.5.0.1+2.5.0.1 3. Asterisk verison: asterisk-1.8.6.04. libpri version: libpri-1.4.12 Is this a configuration issue, package version issue or an operator issue? I'll be glad to share the necessary dahdi configuration files if needed. Thanks in advance. --Cheers,Ishwar. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] 10.0.0-rc1: dahdi doesn't see card
hi:make sure chan_dahdi.conf with the data and stop now, restart asterisk. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: ewiel...@nyigc.com To: asterisk-users@lists.digium.com Date: Fri, 11 Nov 2011 19:38:56 -0500 Subject: Re: [asterisk-users] 10.0.0-rc1: dahdi doesn't see card Show us /etc/asterisk/chan_dahdi.conf (and any #include'd files) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy Sent: Friday, November 11, 2011 5:51 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] 10.0.0-rc1: dahdi doesn't see card From asterisk -cv == Parsing '/etc/asterisk/chan_dahdi.conf': == Found == Parsing '/etc/asterisk/users.conf': == Found -- Automatically generated pseudo channel [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: Ignoring any changes to 'userbase' (on reload) at line 23. [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: Ignoring any changes to 'hassip' (on reload) at line 35. [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: Ignoring any changes to 'hasiax' (on reload) at line 39. [Nov 11 17:43:28] WARNING[5756]: chan_dahdi.c:18155 process_dahdi: Ignoring any changes to 'hasmanager' (on reload) at line 47. == Registered channel type 'DAHDI' (DAHDI Telephony Driver) == Manager registered action DAHDITransfer == Manager registered action DAHDIHangup == Manager registered action DAHDIDialOffhook == Manager registered action DAHDIDNDon == Manager registered action DAHDIDNDoff == Manager registered action DAHDIShowChannels == Manager registered action DAHDIRestart chan_dahdi.so = (DAHDI Telephony Driver) .. But, no dahdi. dahdi show channels shows nothing: Chan Extension Context Language MOH Interpret BlockedState Description pseudodefaultdefault In Service dahdi_cfg -vvv DAHDI Tools Version - 2.4.1 DAHDI Version: 2.5.0.1 Echo Canceller(s): HWEC, OSLEC Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 02) Channel 04: FXS Kewlstart (Default) (Echo Canceler: oslec) (Slaves: 04) 3 channels to configure. Setting echocan for channel 1 to oslec Setting echocan for channel 2 to oslec Setting echocan for channel 4 to oslec From dmesg: dahdi: Telephony Interface Registered on major 196 [9.849036] dahdi: Version: 2.5.0.1 [9.899531] ACPI: PCI Interrupt Link [LNKC] enabled at IRQ 19 [9.899919] wctdm :01:0a.0: PCI INT A - Link[LNKC] - GSI 19 (level, low) - IRQ 19 [9.901845] Freshmaker version: 73 [9.902565] Freshmaker passed register test [ 11.287170] Module 0: Installed -- AUTO FXS/DPO [ 12.172362] Module 1: Installed -- AUTO FXS/DPO [ 12.172795] Module 2: Not installed [ 12.373273] Module 3: Installed -- AUTO FXO (FCC mode) [ 12.374071] Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) Any help appreciated. sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to add new Module in existed Asterisk
hi: please check the chan_gtalk.conf, add your account info. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Tue, 20 Sep 2011 22:37:00 +0700 From: tuant...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to add new Module in existed Asterisk Hi all, I have an Asterisk Server in version 1.4.36 , it runs stable. Now I want to add 2 new modules : jabber and chan_gtalk. How to add these modules and not change anything of configuration existed Asterisk ? Best regards, Ryan. -- _ -- 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] ISDN2 PCIe Card for Asterisk
yes, sangoma card is good. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Tue, 6 Sep 2011 10:38:44 +0200 From: th9...@googlemail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] ISDN2 PCIe Card for Asterisk what do you mean exactly?! One what?! What do you plan to accomplish?! Do you mean a 1 Port ISDN BRI Board?! Difficult to find, and thus boards are really expensive, not under 400.- � inkluding DSP Processors. I advise you taking Gentoo Linux, getting asterisk on it and put a single Port HFC-S PCI (not PCIe) Board in your CPU. If you need something really professional, for Serverside, I advise you sangoma. Tamer Am 06.09.2011 09:08, schrieb Arjan Kroon | Mobillion: Hi, I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk Could anybody give me an advise which card I can use? Regards, Arjan Kroon Mobillion. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginner Question: 4 fxo TDM410 setup
hi: yes, 4fxo is enough for that. four fox means that you have 4 PSTN line, do you really need 4 fxos? you have to use fxs or sip as extensions for pick up the call and make calls. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Tue, 6 Sep 2011 20:28:05 -0400 From: alsta...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Beginner Question: 4 fxo TDM410 setup Hello list. Just another beginner question. I am trying to setup a basic home phone system. I ordered a TDM410 card, which came with 4 fxo ports. I want the home phone system to be able to initiate and receive calls. Can it be done with this card with just one type(no fxs) of ports? If it can be done please help me with scenarios. Thanks for your guidance guys. It is highly appreciated. -- _ -- 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] Anyone using Asterisk on VirtualBox ?
hi: please check the redfone solution. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: aster...@a-domani.nl To: asterisk-users@lists.digium.com Date: Thu, 1 Sep 2011 23:48:46 +0200 Subject: Re: [asterisk-users] Anyone using Asterisk on VirtualBox ? On Thu, 2011-09-01 at 21:32 +0530, RSCL Mumbai wrote: My main interest of being on Virtual platform is portability / Backup. In case of any h/w issues, or crashes, simply copy the VM on to another box and you are up in minutes. Sanjay -- Doing that right now, although in my case i use XEN. Besides being hw independant, it is easier to play with a different version for a while (1.4 / 1.6.0 / 1.6.1 / 1.6.2 / 1.8.0) and being able to switch back in minutes. hw -- _ -- 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] How to know how many user is connected
hi:
please refer this:
http://www.voip-info.org/wiki/view/Asterisk+manager+Example:+PHP
and check the manager.conf, make sure the accounts in managers.conf matchs the
managers displayed.
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
From: gohar.ah...@vopium.com
To: asterisk-users@lists.digium.com
Date: Thu, 25 Aug 2011 11:26:53 +0500
Subject: Re: [asterisk-users] How to know how many user is connected
I’m not a php expert, but seems your php script is incomplete/ you are sending
to socket (fputs) but note receiving anything(fgets) :See this page will help
you. From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Wednesday, August 24, 2011 6:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to know how many user is connected Hi List,
I want to know how many manager is connected into asterisk server. I have made
simple file but I don't have any idea how to get information back from Asterisk
CLI
?php
$socket = fsockopen(127.0.0.1,5038, $errno, $errstr, 30);
if (!$socket)
{
$done=0;
} else {
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: root\r\n);
fputs($socket, Secret: energy\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: manager show connected\r\n);
$done=1;
}
?
Now how to get information into this PHP file
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
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Re: [asterisk-users] Any Method for capturing ISUP packets in DAHDI/ASTERISK
hi: please show the config files. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Wed, 17 Aug 2011 10:51:48 +0530 From: dhaval.it01...@gmail.com To: asterisk-users@lists.digium.com; rmeyerrie...@digium.com Subject: Re: [asterisk-users] Any Method for capturing ISUP packets in DAHDI/ASTERISK Hi Russ, I have tried given patch and successfully compiled dahdi_pcap but when i try to run below command it gives me error. ./dahdi_pcap lapd 16 test.pcap error setting channel err=-1! error setting channel err=-1! error setting channel err=-1! error setting channel err=-1! Segmentation fault I have TE112 Card configured on my machine. Regards Dhaval On Thu, Aug 11, 2011 at 11:21 PM, Russ Meyerriecks rmeyerrie...@digium.com wrote: On Thu, Aug 11, 2011 at 12:43:43PM -0500, Russ Meyerriecks wrote: On Thu, Aug 11, 2011 at 11:57:46AM +0530, DHAVAL INDRODIYA wrote: Hi All, I want packets [request/response] capture for ISUP packets , i have E1 line terminated to my digium card i just want a packets flow between my machine and teleco side, is any tool or utility [command] availabele for observation this packets and data. This issue and patch added pcap support for a guy who wanted to monitor ss7 traffic. You'll need to use dahdi 2.5. You'll also need to compile the dahdi_pcap program on your own, or write a script to exercise the DAHDI_RXMIRROR and DAHDI_TXMIRROR ioctl's. This is an unsupported interface. Forgot to link to the feature request: https://issues.asterisk.org/view.php?id=16831 -- 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 -- _ -- 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] Interesting PRI issue
hi: Please check the status of PRI, i think the channels keeps up and down. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 8 Jun 2011 17:44:12 + Subject: [asterisk-users] Interesting PRI issue Hey Guys! Please help me to find out issue. I have two PRI ## Span 1: WPT1/0 wanpipe1 card 0 span=1,1,0,esf,b8zs bchan=1-23 hardhdlc=24 echocanceller=mg2,1-23 ## Span 2: WPT1/1 wanpipe2 card 1 span=2,2,0,esf,b8zs bchan=25-47 hardhdlc=48 echocanceller=mg2,25-47 Sometime my calls got through but some time i am getting pri cause 44 sebpbx1*CLI == Using SIP RTP CoS mark 5 -- Executing [6463279153@from-sip:1] Dial(SIP/8227-02b1, DAHDI/G1/16463279153) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/G1/16463279153 -- Span 2: Channel 0/23 got hangup, cause 44 -- Span 2: Forcing restart of channel 0/23 since channel reported in use -- Hungup 'DAHDI/i2/16463279153-fe' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/8227-02b1' status is 'CHANUNAVAIL' -- Span 2: Channel 0/23 successfully restarted -- _ -- 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] reload chan_dahdi.conf without disconnect active calls
hi: there is no way to do that. why do you do that? Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 7 Jun 2011 21:12:37 + Subject: [asterisk-users] reload chan_dahdi.conf without disconnect active calls Hi ALL, Is there any way i can reload chan_dahdi.conf without disconnecting active PRI calls ? I want to change pridialplan= option -S -- _ -- 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] PRI issue its BUSY
hi: make sure your pri is up and active. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: ca...@usawide.net To: asterisk-users@lists.digium.com Date: Mon, 6 Jun 2011 20:24:06 -0500 Subject: Re: [asterisk-users] PRI issue its BUSY From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, June 06, 2011 8:20 PM To: asterisk-users Subject: [asterisk-users] PRI issue its BUSY Hi all, I just configures my PRI and incoming calls are working fine but outside calling giving error PRI is BUSY :( any idea ? I have same setup on other box and that boxes works perfect. -- DAHDI/i1/6463279153-2 is proceeding passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is making progress passing it to SIP/7328-0002 -- DAHDI/i1/6463279153-2 is busy -- Hungup 'DAHDI/i1/6463279153-2' == Everyone is busy/congested at this time (1:1/0/0) -- Auto fallthrough, channel 'SIP/7328-0002' status is 'BUSY' Maybe the problem is external to the box. Try swapping PRIs briefly for testing. C. -- _ -- 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 I use phone line to recive faxes?
hi: yes, make sure you also have a fxs to connect your fax if you want to receive fax by fax Mac. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: asterisk_l...@earthshod.co.uk To: asterisk-users@lists.digium.com Date: Fri, 3 Jun 2011 09:30:49 +0100 Subject: Re: [asterisk-users] Can I use phone line to recive faxes? On Thursday 02 Jun 2011, khalid touati wrote: Hi Guys, Actually My question is as in the subject, may I use a regular phone line to receive faxes with FFA (Fax For Asterisk), I am using asterisk 1.6.2.8. Yes, you can. BUT, you will need some sort of FXO interface (allows the computer to connect to the telephone socket on the wall), which is supported by DAHDI (or its predecesor, Zaptel). -- AJS Answers come *after* questions. -- _ -- 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] To know if the ISDN PRI E1 is UP?
hello: Make sure the setting is correct and no Irq LOCKED. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Tue, 31 May 2011 05:32:53 -0700 From: bilmar...@yahoo.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] To know if the ISDN PRI E1 is UP? Hi All; This is the output of the pri show status, so I appreciate if to know if that means the E1 is UP? What does it means that the status us (Status: In Alarm, Down, Active)? What in the below result give an indication that it is UP? CC*CLI pri show span 1 Primary D-channel: 16 Status: In Alarm, Down, Active Switchtype: EuroISDN Type: Network Overlap Dial: 0 Logical Channel Mapping: 0 Timer and counter settings: N200: 3 N202: 3 K: 7 T200: 1000 T202: 1 T203: 1 T303: 4000 T305: 3 T308: 4000 T309: 6000 T313: 4000 T-HOLD: 4000 T-RETRIEVE: 4000 T-RESPONSE: 4000 Overlap Recv: No Regards Bilal -- _ -- 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] About Redfone Configuration
hi: please refer this: http://support.red-fone.com/index.php?_m=knowledgebase_a=viewarticlekbarticleid=20 Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Fri, 27 May 2011 15:58:52 +0530 From: maheshka...@flexydial.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] About Redfone Configuration Dear sir, Please have you any document of Redfone or links. I need to learn this about redfone. i am totally confusing in this. -- Best Regards, Mahesh Katta BUZZWORKS Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E) Mumbai 400069 GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is this Asterisk issue of feature
hi:
i think the wait is used for answer command. please show any debug info?
Best regards,
James.zhu
Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP).
website: www.voipviews.com
Date: Thu, 26 May 2011 16:15:31 +0530
From: virbh...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Is this Asterisk issue of feature
Hi List,
I am confuse about this feature.
When we use Wait(20) in active call session then it's work. But when we use
after hangup the call then Asterisk don't wait from define time.
Ex:-
[call_log]
exten = 4368,1,Answer()
exten = 4368,n,Flite(Welcome)
exten = 4368,n,Set(__StartTime=${STRFTIME(${EPOCH},Asia/Calcutta,%Y-%m-%d
%H:%M:%S)})
exten = 4368,n,Set(__uniqueId=${UNIQUEID})
exten = 4368,n,Wait(20) ; At this moment it's work.
exten = 4368,n,Hangup()
exten = h,1,NoOp(***Now wait(20) wouldn't work
* )
exten = h,n,Wait(20) ; At this moment it's not work.
exten = h,n,NoOp(***never come into execution
**)
-
Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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Re: [asterisk-users] synway
hi: yes, Synway's asterisk cards work with zaptel and dahdi. it is a good product. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: l...@lopl.net Date: Wed, 25 May 2011 09:15:47 +0430 To: asterisk-users@lists.digium.com Subject: [asterisk-users] synway Dear,do you have any successful experience for installing SHT-8C/PCI/FAX (synway) with asterisk ?is it compatibe with asterisk (dahdi/zaptel)? best -- Pezhman Lali -- _ -- 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] chan_zap
hi: yes, it is true, the channel is locked by some tasks. you can please restart asterisk and zatpel. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Tue, 24 May 2011 00:08:48 +0300 From: tzafrir.co...@xorcom.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] chan_zap On Mon, May 23, 2011 at 10:36:09AM -0700, bilal ghayyad wrote: Hi All; Suddenly the zaptel channel look like stop working and it is giving me this error when I do zap restart: [May 24 19:30:21] ERROR[2772]: chan_zap.c:7219 mkintf: Unable to open channel 1: Device or resource busy It is already open by something else? Another program? A different instance of Asterisk? here = 0, tmp-channel = 1, channel = 1 [May 24 19:30:21] ERROR[2772]: chan_zap.c:10582 build_channels: Unable to register channel '1,2' -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] difference between SIP peer and SIP user ?
hello: please refer this link: http://www.voip-info.org/wiki/view/Asterisk+SIP+user+vs+peer Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com Date: Sat, 21 May 2011 17:49:37 +0530 From: virbh...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] difference between SIP peer and SIP user ? Hi list, I am confuse about these CLI commands sip show users sip show peers Can someone clear my doubt . what are the difference between them? - Thanks and regards Virendra Bhati +91-9172341457 -- _ -- 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] Automatic dialing + SMS
hello: i think you can use php and get message from GUI and send by php AGI. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com To: asterisk-users@lists.digium.com From: gadgetron...@gmail.com Date: Thu, 19 May 2011 09:57:23 + Subject: Re: [asterisk-users] Automatic dialing + SMS Hi Guys Using call files might be easiest. But I d also try out AGI scripting too. I ll be sure to call back if I require any help. For the sms bit,...let's say I want to send bulk sms to multiple mobile devices. Thanks a lot Regards Sent from my BlackBerry® smartphone from Vodafone -Original Message- From: Steve Edwards asterisk@sedwards.com Sender: asterisk-users-boun...@lists.digium.com Date: Wed, 18 May 2011 06:46:25 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Automatic dialing + SMS On Wed, 18 May 2011, gadgetron...@gmail.com wrote: Does it mean Asterisk has no in-built applications for auto dialing. Asterisk is a telephony Erector Set*. You get to build what you want. All the pieces are there. What scripting language can easily and best be used for the AGI. Easy may not be best. 'Easiest' is the language you know best. Best depends on your needs. A scripting language like PHP may be easiest for you if you know that language. A compiled language like C may be best if you want to run a bazillion calls per second. You can execute xxx AGIs written in C in the time it takes to load the Perl or PHP interpreter and parse your script. *) http://en.wikipedia.org/wiki/Erector_set -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
