Re: [asterisk-users] Autoanswer in Nokia SIP clients?
As a dual GSM/WiFi mode phone might be already engaged in a GSM conversion while a SIP call occurs, I think Symbian application dev rules would impose any application to centralize microphone and speaker allocation to a Symbian provided resource manager. So I think automatic answer of any kind are currently not supported by this Symbian provided resource manager. 2008/8/4 Stefan Gofferje [EMAIL PROTECTED] Hi, anybody knows if it is possible to make the Nokia SIP client in the phones autoanswer a call in speakerphone mode? --Stefan ___ -- 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
[asterisk-users] Asterisk dead but subsys locked
Hi Everyone, I am currently running Trixbox 2.6 and I have a problem with Asterisk. /etc/init.d/asterisk status Asterisk dead but subsys locked I deleted all files in /var/run/asterisk folder and asterisk restart... It's ok for a while. But some days after Asterisk again is dead. Can anybody help me? Rgs / budacsik ___ -- 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] Asterisk Realtime with MySQL Registration Failed
Dear Mr. Tilghman, Thank you for your attention. Actually it was NULL before when it was not working. I changed it to deny=no and permit=all after that thinking it could be the problem. Now I have changed it back to NULL using update sip_buddies set deny=NULL, permit=NULL where id=1. You can see the table below now. ++--+-+--+---+--+-+-+---+--+--++-+-+--+--+-+---+-+--++--+-+--+-+-+++++--+--+-+-+++--+++| id | name | accountcode | amaflags | callgroup | callerid | canreinvite | context | defaultip | dtmfmode | fromuser | fromdomain | fullcontact | host | insecure | language | mailbox | md5secret | nat | deny | permit | mask | pickupgroup | port | qualify | restrictcid | rtptimeout | rtpholdtimeout | secret | type | username | disallow | allow | musiconhold | regseconds | ipaddr | regexten | cancallforward | setvar |++--+-+--+---+--+-+-+---+--+--++-+-+--+--+-+---+-+--++--+-+--+-+-+++++--+--+-+-+++--+++| 1 | Abid | NULL| NULL | NULL | NULL | no | default | NULL | rfc2833 | NULL | NULL | NULL| dynamic | NULL | NULL | NULL| NULL | yes | NULL | NULL | NULL | NULL | yes | NULL| NULL| NULL | NULL | 1504 | friend | 1504 | all | g729;ilbc;gsm;ulaw;alaw | NULL| 0 || | yes| |++--+-+--+---+--+-+-+---+--+--++-+-+--+--+-+---+-+--++--+-+--+-+-+++++--+--+-+-+++--+++But it still has the same problem. Any further suggessions. Thanks. Abid Saleem From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Mon, 4 Aug 2008 10:15:06 -0500 Subject: Re: [asterisk-users] Asterisk Realtime with MySQL Registration Failed On Monday 04 August 2008 07:57:46 Abid Saleem wrote: | id | name | accountcode | amaflags | callgroup | callerid | canreinvite | | context | defaultip | dtmfmode | fromuser | fromdomain | fullcontact | | host| insecure | language | mailbox | md5secret | nat | deny | permit | | mask | pickupgroup | port | qualify | restrictcid | rtptimeout | | rtpholdtimeout | secret | type | username | disallow | allow| | musiconhold | regseconds | ipaddr | regexten | cancallforward | | setvar | |++--+-+--+---+--+--- |--+-+---+--+--++-+ |-+--+--+-+---+-+--+--- |-+--+-+--+-+-++--- |-+++--+--+ |-+-+++--++ |+| 1 | Abid | NULL| NULL | NULL | NULL | no | | default | NULL | rfc2833 | NULL | NULL | NULL | | dynamic | NULL | NULL | NULL| NULL | yes | no | all | | NULL | NULL| yes | NULL| NULL| NULL | NULL | | 1504 | friend | 1504 | all | | g729;ilbc;gsm;ulaw;alaw | NULL| 0 || | | yes|| You have deny=no, and permit=all, which are invalid values for permit and deny. Please set both of these fields to NULL. -- Tilghman ___ -- 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 _ News, entertainment and everything you care about at Live.com. Get it now!
Re: [asterisk-users] Asterisk dead but subsys locked
Steven Howes wrote: On 5 Aug 2008, at 09:16, Budacsik Attila wrote: Hi Everyone, I am currently running Trixbox 2.6 and I have a problem with Asterisk. /etc/init.d/asterisk status Asterisk dead but subsys locked I deleted all files in /var/run/asterisk folder and asterisk restart... It's ok for a while. But some days after Asterisk again is dead. Can anybody help me? Rgs / budacsik You could look in the log? see what happens before it dies.. Now dead Asterisk again: asterisk -rvvv Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl exist?) ls -l /var/run/asterisk/ total 4 srwxr-xr-x 1 asterisk asterisk 0 Aug 5 09:46 asterisk.ctl -rw-r--r-- 1 asterisk asterisk 6 Aug 5 09:46 asterisk.pid tail -f /var/log/asterisk/full [Aug 5 12:11:04] VERBOSE[13396] logger.c: [Aug 5 12:11:04] -- AGI Script dialparties.agi completed, returning 0 [Aug 5 12:11:04] DEBUG[13396] app_macro.c: Executed application: AGI [Aug 5 12:11:04] VERBOSE[13396] logger.c: [Aug 5 12:11:04] -- Executing [EMAIL PROTECTED]:7] Dial(mISDN/1-u10, SIP/200SIP/202SIP/211SIP/204|20|m(abba)tmwM(auto-blkvm)) in new stack [Aug 5 12:11:04] VERBOSE[13396] logger.c: [Aug 5 12:11:04] -- Called 200 [Aug 5 12:11:04] WARNING[13396] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Aug 5 12:11:04] VERBOSE[13396] logger.c: [Aug 5 12:11:04] -- Called 211 [Aug 5 12:11:04] WARNING[13396] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Aug 5 12:11:04] VERBOSE[13396] logger.c: [Aug 5 12:11:04] -- Started music on hold, class 'default', on mISDN/1-u10 [Aug 5 12:11:04] VERBOSE[13396] logger.c: [Aug 5 12:11:04] -- SIP/200-089c4088 is ringing [Aug 5 12:11:04] VERBOSE[13396] logger.c: [Aug 5 12:11:04] -- SIP/211-089c88c0 is ringing I think the problem is with asterisk.ctl. This is only a hint. ___ -- 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] Autoanswer in Nokia SIP clients?
Olivier schrieb: Well, if I'm in a GSM call, the Nokia SIP client anyway signals busy to a SIP call. Is it certain ? Yes, just tested it myself. Phone answers with Busy here if in a GSM call. From my understanding, Symbian applications MUST leave this decision type to an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. Well, there are Symbian application implementing a local answering machine on the phone. There even is a spy application which autoanswers a call from a specific number without any indication and rejects all other calls, so it must be possible for a Symbian app to autoanswer a call, even from GSM. I just don't see the point why it shouldn't be possible for a SIP client to autoanswer a call instead of waiting for the green button, given that the phone is not in a GSM call. -S ___ -- 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] Autoanswer in Nokia SIP clients?
2008/8/5 Stefan Gofferje [EMAIL PROTECTED] Olivier schrieb: As a dual GSM/WiFi mode phone might be already engaged in a GSM conversion while a SIP call occurs, I think Symbian application dev rules would impose any application to centralize microphone and speaker allocation to a Symbian provided resource manager. So I think automatic answer of any kind are currently not supported by this Symbian provided resource manager. Well, if I'm in a GSM call, the Nokia SIP client anyway signals busy to a SIP call. Is it certain ? Maybe, a TRYING message is replied to let user have enough time to put GSM call on hold ... It could do the same to a SIP call with autoanswer request... Absolutely It's just a question if you have to press the green button or not on an incoming SIP call... Agreed From my understanding, Symbian applications MUST leave this decision type to an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. PS: Please, note that I'm far from being an expert in GSM handsets but this is the conclusion I reached after reading Symbian forum and documentation. -S ___ -- 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] Asterisk dead but subsys locked
On 5 Aug 2008, at 09:16, Budacsik Attila wrote: Hi Everyone, I am currently running Trixbox 2.6 and I have a problem with Asterisk. /etc/init.d/asterisk status Asterisk dead but subsys locked I deleted all files in /var/run/asterisk folder and asterisk restart... It's ok for a while. But some days after Asterisk again is dead. Can anybody help me? Rgs / budacsik You could look in the log? see what happens before it dies.. ___ -- 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] Autoanswer in Nokia SIP clients?
Olivier schrieb: As a dual GSM/WiFi mode phone might be already engaged in a GSM conversion while a SIP call occurs, I think Symbian application dev rules would impose any application to centralize microphone and speaker allocation to a Symbian provided resource manager. So I think automatic answer of any kind are currently not supported by this Symbian provided resource manager. Well, if I'm in a GSM call, the Nokia SIP client anyway signals busy to a SIP call. It could do the same to a SIP call with autoanswer request... It's just a question if you have to press the green button or not on an incoming SIP call... -S ___ -- 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] Queue Penalties not working properly
Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr ___ -- 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] Asterisk dead but subsys locked
Steven Howes wrote: On 5 Aug 2008, at 09:16, Budacsik Attila wrote: Hi Everyone, I am currently running Trixbox 2.6 and I have a problem with Asterisk. /etc/init.d/asterisk status Asterisk dead but subsys locked I deleted all files in /var/run/asterisk folder and asterisk restart... It's ok for a while. But some days after Asterisk again is dead. Can anybody help me? Rgs / budacsik You could look in the log? see what happens before it dies.. I found this much [Aug 5 08:34:57] WARNING[11858] func_uri.c: Syntax: URIENCODE(data) - missing argument! [Aug 5 08:34:57] WARNING[11858] func_uri.c: Syntax: URIENCODE(data) - missing argument! [Aug 5 08:34:57] WARNING[11858] func_uri.c: Syntax: URIENCODE(data) - missing argument! [Aug 5 08:36:15] WARNING[12574] app_dial.c: Skipping dialing interface 'SIP/211' again since it has already been dialed [Aug 5 08:36:17] WARNING[12520] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Aug 5 08:36:17] WARNING[12520] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Aug 5 08:36:38] WARNING[12583] app_dial.c: Skipping dialing interface 'SIP/211' again since it has already been dialed [Aug 5 08:36:52] WARNING[12520] format_wav.c: Unexpected freqency 44100 [Aug 5 08:36:52] WARNING[12520] file.c: Unable to open format wav [Aug 5 08:36:52] WARNING[12520] file.c: Unable to open digits/2 (format 0x8 (alaw)): No such file or directory [Aug 5 08:38:44] WARNING[12691] app_dial.c: Skipping dialing interface 'SIP/211' again since it has already been dialed [Aug 5 08:38:44] WARNING[12639] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) [Aug 5 08:38:44] WARNING[12639] app_dial.c: Unable to create channel of type 'SIP' (cause 3 - No route to destination) I noticed error and restarted Asterisk at 8:43 ___ -- 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] Autoanswer in Nokia SIP clients?
You're certainly right : today, it's possible to insert a GSM PCI card in a PC, load it with Asterisk and do whatever you want with incoming and outgoing calls but the fact is, AFAIK, we can't do the same with a handset (you (mostly) can't edit Least Cost Routing rules, you can't autoanswer, etc ...) Which company publishes this Symbian application implementing a local answering machine on the phone, for instance ? Is it related with software extensions such as those published by Nokia for Cisco or Alcatel IPBX ? http://www.nokiaforbusiness.com/nfb/find_a_product/product_category.html?guid=f8219bb3c8ba6110VgnVCM10708ef393RCRD ___ -- 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] Autoanswer in Nokia SIP clients?
Olivier schrieb: Which company publishes this Symbian application implementing a local answering machine on the phone, for instance ? There are several. For instance, rock your mobile comes to my mind. http://www.rock-your-mobile.com/ http://www.rock-your-mobile.com/answering-machine.php -S ___ -- 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] Queue Penalties not working properly
Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn’t go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 ___ -- 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] RES: a simple Asterisk AMI interface with Delphi (or Lazarus+FreePascal)
Hello, Just wanted to let you know that the XP version works fine on vista. I was working on a similar program but didnt have enough time to finish, I was working on Delphi 7 btw. Thanks Marco. -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Gerald Harshany Enviada em: terça-feira, 5 de agosto de 2008 02:56 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [asterisk-users] a simple Asterisk AMI interface with Delphi (or Lazarus+FreePascal) Hi Everyone, Those of you who have a simple home-based Asterisk box might be interested in a simple Win32 (Win2K or WinXP) interface to the AMI manager. The quick-start versions merely require unzipping with NO Installation - hence, NO Uninstall (i.e., no registry writes at any time by the install nor by the program). (Unfortunately) the INSTALL version does write to the registry due to the database licensing requirements. Would suggest that you download the PDF and, if interested, (or if you hate to read manuals, just ), download the quick-start version which only requires 3 settings in Asterisk's manager.conf file (the user name, the password, and the read/write privileges - program defaults to the 5038 port). The program was really written as a nostalgic cruise down the old Pascal OOP thruway, and not as a contender to the likes of FOP, etc. Pascal has nice features such as declaring any of your functions inline; or for that matter writing inline Assembler code which was the language in the '70s (that is the 1900's, by the way). The Win2K version was compiled on an old Delphi 5 compiler (and for you young'uns, that was circa 1999 when Win2K was unveiled). However, fear not, the WinXP version was compiled with the latest Delphi 2007 R2. However, I did NOT insert some required Vista enabling statements (such as for the glass effect), since I have no interest in testing it (yet) in Vista; so, the XP version may or may not function well in XP compatibility mode within Vista. As for my Subject - Is anyone in this Asterisk group doing anything using Lazarus and FreePascal for the Asterisk box? The FreePascal compiler is a total (and, yes, an open source work in progress) cross-platform compiler. What I mean is, it can compile for Win, Mac, and Linux, but also for about half a dozen CPU's. The documentation for the compiler is an outstanding example for open-source projects. Downloads and info at: http://www.jerryh.us/Downloads/amifiles.htm Gerald Harshany, Ph.D. Professor Emeritus of Mathematics And again, for you young'uns, Emeritus simply means ancient :) ___ -- 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 Visioncom Tecnologia da Informação (www.visioncom.com.br) ___ -- 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] Queue Penalties not working properly
On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 Robin, round robin ___ -- 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] RES: Queue Penalties not working properly
You have to limit calls to these agents, use incomminglimit or call-limit on sip.conf to do that. That way, when the first agent answers a call, all the other calls directed to it will return with busy signal, and will be transferred to the other agent. __ Marco Eduardo Cordeiro Visioncom De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Syed Nasruddin Enviada em: terça-feira, 5 de agosto de 2008 09:40 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: [asterisk-users] Queue Penalties not working properly Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesnt go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr Visioncom Tecnologia da Informação (www.visioncom.com.br) ___ -- 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] Queue Penalties not working properly
Cannot i use ringall strategy with penalties??? Will rrmemory will fullfil my requirement?? My requirements: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 3. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Moreover why my queue status shows my agent as NOT IN USE while in fact it is busy answering the call?? Thanks Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, August 05, 2008 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 Robin, round robin ___ -- 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] a simple Asterisk AMI interface with Delphi (or Lazarus+FreePascal)
On Tue, Aug 5, 2008 at 1:55 AM, Gerald Harshany [EMAIL PROTECTED] wrote: Hi Everyone, Those of you who have a simple home-based Asterisk box might be interested in a simple Win32 (Win2K or WinXP) interface to the AMI manager. The quick-start versions merely require unzipping with NO Installation - hence, NO Uninstall (i.e., no registry writes at any time by the install nor by the program). (Unfortunately) the INSTALL version does write to the registry due to the database licensing requirements. Would suggest that you download the PDF and, if interested, (or if you hate to read manuals, just ), download the quick-start version which only requires 3 settings in Asterisk's manager.conf file (the user name, the password, and the read/write privileges - program defaults to the 5038 port). The program was really written as a nostalgic cruise down the old Pascal OOP thruway, and not as a contender to the likes of FOP, etc. Pascal has nice features such as declaring any of your functions inline; or for that matter writing inline Assembler code which was the language in the '70s (that is the 1900's, by the way). The Win2K version was compiled on an old Delphi 5 compiler (and for you young'uns, that was circa 1999 when Win2K was unveiled). However, fear not, the WinXP version was compiled with the latest Delphi 2007 R2. However, I did NOT insert some required Vista enabling statements (such as for the glass effect), since I have no interest in testing it (yet) in Vista; so, the XP version may or may not function well in XP compatibility mode within Vista. As for my Subject - Is anyone in this Asterisk group doing anything using Lazarus and FreePascal for the Asterisk box? The FreePascal compiler is a total (and, yes, an open source work in progress) cross-platform compiler. What I mean is, it can compile for Win, Mac, and Linux, but also for about half a dozen CPU's. The documentation for the compiler is an outstanding example for open-source projects. Downloads and info at: http://www.jerryh.us/Downloads/amifiles.htm Gerald Harshany, Ph.D. Professor Emeritus of Mathematics And again, for you young'uns, Emeritus simply means ancient :) I have to admit I am Win2k Server fan. I still have a few of these scattered in basements here and there, chugging away for years and years. Thanks, Steve T ___ -- 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] Queue Penalties not working properly
On Tue, Aug 5, 2008 at 5:27 PM, Syed Nasruddin [EMAIL PROTECTED] wrote: Cannot i use ringall strategy with penalties??? Will rrmemory will fullfil my requirement?? rrmemory isn't ringall, it won't ring all members. But yes - you can use ringall with penalties. My requirements: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 3. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Moreover why my queue status shows my agent as NOT IN USE while in fact it is busy answering the call?? What you are seeing is caused by status NOT IN USE. You have to set call-limit in sip.conf for all your phones, to any value, so that device states work correctly, and queue can know that those phones are busy. Now you probably can see in CLI that queue is sending second call to first agent(s). Regards, Atis Thanks Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, August 05, 2008 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 Robin, round robin ___ -- 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 -- 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 -- 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] Grandstream RS-232 config (slightly off-topic)
I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone unconfigurable via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the keypad update feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited and I can't change SIP configuration or reset to default values (and see if that helps to bring the HTTP back up). I did try to upgrade and downgrade the firmware with no change at all. The reset button reboots the device but doesn't restore default values! (I tried keeping it pressed for several minutes...) So my last chance before throwing it away is to administer it via serial port. The thing is that Grandstream's official manual says that there is an RS232 serial port for administration but it doesn't say anything else about it (how to connect, how to change config, how to reset the device, etc). There's absolutely nothing regarding RS-232. If someone has this or a similar device and accessed it via serial port then I'd greatly appreciate some quick tips. Thanks, Vieri ___ -- 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] Getting Asterisk out of the RTP media path
When calling from our SIP proxy through Asterisk to the PSTN provider, we support reINVITES which tend to work seamlessly. However, when creating a call file which essentially connects a call from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP media path. I understand that this is sort of the idea behind a bridged channel, but is there any way to avoid it? Is there any way to say Connect this number and this number and then get out of the way, or is this a design limitation? N. ___ -- 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] Grandstream RS-232 config (slightly off-topic)
Never mind. I set the wrong baud rate. The right values are 115200-8-N-1-No flow control. However, the serial connection is as good or as useless as the telent connection. I have no way to restore factory settings. --- On Tue, 8/5/08, Vieri [EMAIL PROTECTED] wrote: I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone unconfigurable via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the keypad update feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited and I can't change SIP configuration or reset to default values (and see if that helps to bring the HTTP back up). I did try to upgrade and downgrade the firmware with no change at all. The reset button reboots the device but doesn't restore default values! (I tried keeping it pressed for several minutes...) So my last chance before throwing it away is to administer it via serial port. The thing is that Grandstream's official manual says that there is an RS232 serial port for administration but it doesn't say anything else about it (how to connect, how to change config, how to reset the device, etc). There's absolutely nothing regarding RS-232. If someone has this or a similar device and accessed it via serial port then I'd greatly appreciate some quick tips. Thanks, Vieri ___ -- 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] vars in Macros called by DIAL with option M
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and would
like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in macros
called by DIAL so that I can use these vars in the Dialplan or in the h
extention.
best regards
Thomas
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[asterisk-users] Codec g729 issues
HI folks! my topology is: softswitch (BROADSOFT) -- [sip trunk] -- Asterisk I need to connect phone calls using g729 codec. Debugging some calls we found that calls cant connect because of codec incompatibility. Our Sip provider send us annexb=yes when a call is comming and our asterisk send annexb=no. Im running asterisk 1.4.21.1. Output debug shows: To: sip:[EMAIL PROTECTED];user=phone CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp Supported: 100rel User-Agent: Huawei SoftX3000 V300R006 Max-Forwards: 69 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,ME SSAGE,REFER Content-Length: 274 Content-Type: application/sdp v=0 o=HuaweiSoftX3000 194579 194579 IN IP4 189.8.113.170 s=Sip Call c=IN IP4 XXX.X.XXX.170 t=0 0 m=audio 49256 RTP/AVP 18 8 0 97 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-15 a=fmtp:18 annexb=yes Via: SIP/2.0/UDP XXX.X.XXX.170:5060;branch=z9hG4bKo2echm202o5g2dc38701.1;received=XXX.X.XXX.1 70 From: sip:000@ XXX.X.XXX.170;user=phone;tag=25f94692 To: sip:7002@ XXX.X.XXX.177;user=phone;tag=as0de67360 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Supported: replaces Contact: sip:7002@ XXX.X.XXX.177 Content-Type: application/sdp Content-Length: 262 v=0 o=root 10183 10183 IN IP4 XXX.X.XXX.177 s=session c=IN IP4 XXX.X.XXX.177 t=0 0 m=audio 10772 RTP/AVP 18 97 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:97 telephone-event/8000 a=fmtp:97 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Thanks for any help! Gustavo A. González Dto. de Infraestructura Despegar.com, Inc. [EMAIL PROTECTED] ___ -- 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] vars in Macros called by DIAL with option M
I don't think you can do that because, asterisk, in the caller thread
will only read MACRO_RESULT to know if he has to connect the call or not.
A workaround will be to :
1. before the dial, add a row in a database table and retrieve an id
2. pass the id to test_connect and test_connect will then write his
variable value into the database
3. after the dial,. use the id to retrieve the needed variable.
Hope this will help.
Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and would
like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in macros
called by DIAL so that I can use these vars in the Dialplan or in the h
extention.
best regards
Thomas
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Internal Virus Database is out of date.
Checked by AVG.
Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008
7:42 PM
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Re: [asterisk-users] vars in Macros called by DIAL with option M
On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and
would like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in macros
called by DIAL so that I can use these vars in the Dialplan or in the h
extention.
There isn't any good way to do that, period. When it comes to inheritance,
variables are only inherited from a master channel to a slave channel. In the
case of the Macro operating within the Dial, that Macro is occurring
exclusively on the slave channel. You cannot directly set variables on other
channels (for obvious race-condition reasons).
However, you could do this in a roundabout way, either by using a database or
by using shared variables in trunk. You'd need to first set (in the master
channel, before the Dial) an inherited variable containing the name of the
master channel, i.e. Set(_masterchan=${CHANNEL(name)}), then use that
inherited variable to set the shared variable in the master channel from the
slave channel, i.e. Set(SHARED(foo,${masterchan})=...). Finally, you would
be able to access the shared variable in the master channel with
${SHARED(foo)}. Again, the SHARED function is only available in trunk at this
time, although you could probably backport it to 1.4 with minimal trouble.
--
Tilghman
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[asterisk-users] ZRTP in Asterisk
Dear people, does anybody try the ZRTP patch for Asterisk in order to have ZRTP encrytion among SIP/RTP calls ??? In other words, did anybody succesfully implement ZRTP in Asterisk ??? Any documentation about it ??? Special thanks Alejandro ___ -- 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] vars in Macros called by DIAL with option M
And if you use DIALSTATUS and ANSWERTIME to check the last dial status,
you need to take care of the following bug
http://bugs.digium.com/view.php?id=13216
Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and would
like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in macros
called by DIAL so that I can use these vars in the Dialplan or in the h
extention.
best regards
Thomas
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Internal Virus Database is out of date.
Checked by AVG.
Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008
7:42 PM
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Re: [asterisk-users] Queue Penalties not working properly
Hi, Actully the way I want the penalties functionality to behave it is not doing it accordingly. I am right now using ringall. Set penalty 1 for one agent and 2 for secnd agent. All the calls come in and go to first agent#1 having penalty one. But the second call also go to agent#1 and start waiting for it to be free rather it should have gone to penalty two agent#2 I have added call-limit=1 for bot sip accounts. And started the services. Still find the status wrong. nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: Tuesday, August 05, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 5:27 PM, Syed Nasruddin [EMAIL PROTECTED] wrote: Cannot i use ringall strategy with penalties??? Will rrmemory will fullfil my requirement?? rrmemory isn't ringall, it won't ring all members. But yes - you can use ringall with penalties. My requirements: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 3. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Moreover why my queue status shows my agent as NOT IN USE while in fact it is busy answering the call?? What you are seeing is caused by status NOT IN USE. You have to set call-limit in sip.conf for all your phones, to any value, so that device states work correctly, and queue can know that those phones are busy. Now you probably can see in CLI that queue is sending second call to first agent(s). Regards, Atis Thanks Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, August 05, 2008 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 Robin, round robin ___ -- 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 -- 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 -- 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] vars in Macros called by DIAL with option M
I would suggest putting a NOOPIn the MACRO to ensure the variable IS
actually getting SET.
As I understand it VARIABLES are GLOBAL and what you are doing is
correct, BUT, This could be a learning opportunity for me too.
Be advised, there seems to be push by DIGIUM for folks to use the
subroutine rather than MACROS now
H, Can the Dial command CALL a SUBROTINE as it does a MACRO ???
Ruddy Gbaguidi wrote:
And if you use DIALSTATUS and ANSWERTIME to check the last dial status,
you need to take care of the following bug
http://bugs.digium.com/view.php?id=13216
Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and
would
like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in macros
called by DIAL so that I can use these vars in the Dialplan or in the h
extention.
best regards
Thomas
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Internal Virus Database is out of date.
Checked by AVG.
Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008
7:42 PM
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Re: [asterisk-users] Asterisk Realtime with MySQL Regi stration Failed
On Tuesday 05 August 2008 03:49:15 Abid Saleem wrote: Dear Mr. Tilghman, Thank you for your attention. Actually it was NULL before when it was not working. I changed it to deny=no and permit=all after that thinking it could be the problem. Now I have changed it back to NULL using update sip_buddies set deny=NULL, permit=NULL where id=1. You can see the table below now. It is likely that you got a different error this time, since it will no longer fail the permit/deny ACL checks. What is the error you got this time? Hopefully, you've done a 'sip reload' after the changing the database, or else you won't actually be testing the change. -- Tilghman ___ -- 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] vars in Macros called by DIAL with option M
Err - Ok - let me ask this in MUCH simpler way
1 - In dialplan , you set a Variable called MYVAR, to Apple
2 - You go into MACRO and NOOP the VALUE of MYVAR -What SHOULD it BE ?
3 - While IN MACRO you set VALUE of MYVAR = Pear
4 - You leave MACRO and get back to DIALPLAN and NOOP the value of MYVAR
- What should it be
=== 2nd Question ===
CAN the DIAL command call a SUBROUTINE instead of a MACRO ?
If so WOULD that help him out ?
Any clarification much apprecatted
Tilghman Lesher wrote:
On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and
would like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in macros
called by DIAL so that I can use these vars in the Dialplan or in the h
extention.
There isn't any good way to do that, period. When it comes to inheritance,
variables are only inherited from a master channel to a slave channel. In the
case of the Macro operating within the Dial, that Macro is occurring
exclusively on the slave channel. You cannot directly set variables on other
channels (for obvious race-condition reasons).
However, you could do this in a roundabout way, either by using a database or
by using shared variables in trunk. You'd need to first set (in the master
channel, before the Dial) an inherited variable containing the name of the
master channel, i.e. Set(_masterchan=${CHANNEL(name)}), then use that
inherited variable to set the shared variable in the master channel from the
slave channel, i.e. Set(SHARED(foo,${masterchan})=...). Finally, you would
be able to access the shared variable in the master channel with
${SHARED(foo)}. Again, the SHARED function is only available in trunk at this
time, although you could probably backport it to 1.4 with minimal trouble.
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Re: [asterisk-users] Queue Penalties not working properly
Syed Nasruddin wrote: Hi, Actully the way I want the penalties functionality to behave it is not doing it accordingly. I am right now using ringall. Set penalty 1 for one agent and 2 for secnd agent. All the calls come in and go to first agent#1 having penalty one. But the second call also go to agent#1 and start waiting for it to be free rather it should have gone to penalty two agent#2 I have added call-limit=1 for bot sip accounts. And started the services. Still find the status wrong. nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: Tuesday, August 05, 2008 7:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 5:27 PM, Syed Nasruddin [EMAIL PROTECTED] wrote: Cannot i use ringall strategy with penalties??? Will rrmemory will fullfil my requirement?? rrmemory isn't ringall, it won't ring all members. But yes - you can use ringall with penalties. My requirements: 1. 10 Call Center Agents. 2. All the calls coming in will ALWAYS be routed to specific 5 agents, firstly. 3. IF ALL the first 5 agents are busy then ONLY then the call will be routed to next 5 Agents. Moreover why my queue status shows my agent as NOT IN USE while in fact it is busy answering the call?? What you are seeing is caused by status NOT IN USE. You have to set call-limit in sip.conf for all your phones, to any value, so that device states work correctly, and queue can know that those phones are busy. Now you probably can see in CLI that queue is sending second call to first agent(s). Regards, Atis Thanks Syed nasr -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Tuesday, August 05, 2008 5:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue Penalties not working properly On Tue, Aug 5, 2008 at 8:47 AM, Robin Rodriguez [EMAIL PROTECTED] wrote: Syed Nasruddin wrote: Hi, I am using Asterisk 1.4.18. I am implementing Penalties for my agents. What is happening: two agents configuired one agent with penalty 1 and the other with penalty 2. All the calls must go first to Agent 1 and if his line is busy then only then agent 2 will get the call. However my queues are not behaving in this manner. I have impmemnted ringall strategy. Now when first call comes it ends up with agent 1, when secnd call comes it continue wait in queue and doesn't go to agent 2 and when agent one is free it goes to this agent. I have set penalties in queue.conf. I have monitered my queue and witnessed that my agent1 status shows Not In Use and Agent 2 also same status is this the reason behind this. I have copied my queue show results below.please help . how do I change this stauts problem callcenter*CLI queue show myqueue has 0 calls (max unlimited) in 'ringall' strategy (14s holdtime), W:0, C:2, A:0, SL:0.0% within 0s Members: SIP/1001 with penalty 1 (Not in use) has taken 2 calls (last was 2233 secs ago) SIP/1000 with penalty 2 (Not in use) has taken no calls yet No Callers Syed nasr You need to use the linear queue strategy, it is in 1.6 or there is a backport to 1.4 -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 Robin, round robin ___ -- 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 Very carefully reread the descriptions on penalties and queue strategies on voip-info.org, the first time I tried to do what you want I was confused too, but I assure you if you read about the linear strategy you will find what you need. Robin -- Robin Rodriguez VoIP/Telecom Engineer Atlantic.net 1-800-211-9496 ___ -- 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] Asterisk to Avaya
Hi,
Sorry this is so long, but I am reasonably desparate.
I am having real fun with hooking an Avaya system to Asterisk using
ISDN30. I have the ISDN signalling all sorted one way, and can pass
calls from the real world (ie. the telco and asterisk) TO the avaya
box, and it accepts that and sets up the call perfectly.
The problem is that the Avaya box is signalling outbound calls using
an odd method, which smacks of an analogue system with ISDN30 bolted
on for a bit of a laugh.
They send a q931 SETUP message. This contains the correct callerID,
but only the first 1 to 4 of the dialled number's digits - The
remainder of the number is I believe passed through using DTMF!!! From
the look of it they intentionally do not send an IE 161 Sending
Complete with the SETUP, so that the far end continues to listen for
these DTMF tones, until it resolves to a legal number.
My questions for some ISDN expert out there...
Part 1) I need to receive the number in the SETUP, which I guess will
be in ${EXTEN}, then I assume I can use Read() to collect DTMF digits,
and check the dialplan to see if it is a locally terminated number.
Once I am 100% sure it is not local, I can then dial the collected
number through the Telco ISDN channel. Make sense? I think I can
probably handle that. The problem is that I do not know whether I have
received all digits from the Avaya at that point, which leads to...
Part 2) Can I dial through Zaptel (via a Sangoma card if that makes a
difference) without sending the IE 161 call complete? I thought that
Dial(Zap/G1||D(${INITIAL}))
might send the initial digits using DTMF, and then leave the channel
open so that more DTMF could follow over the now bridged channel. In
fact I get an immediate failure as if the far end thinks I have
finished dialling. Can I assume that libpri does not currently support
this method of dialling? If not, how might it be added? I can hack the
code, I just need suggestions of where to look and how it might sanely
be added :)
Part 3) It is possible that the Avaya is not using DTMF at-all, and
that it will send more bits of the called-party number using the
D-Channel as you would expect, but Asterisk does not seem to be
waiting for them. Can this be changed in Zaptel/Asterisk. Does anyone
know the Avaya systems well enough to suggest how it might be working?
Many many thanks for any feedback.
Regards,
Steve
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Re: [asterisk-users] vars in Macros called by DIAL with option M
above the original post is very confusing. Please stop doing this.
The format of this post is in reverse, to demonstrate why posting a reply
option is only in trunk. So no, it would not help him out.
Yes, it works the same way, by using the U() option to Dial. However, this
Question 2:
prior to the origination of the slave channel.
channel. No inheritance is possible, because the master channel originated
channel, so any values set in the slave channel will not affect the master
Apple. The variable is only set in the slave channel, not in the master
Question 1:
On Tuesday 05 August 2008 11:50:28 Al Baker wrote:
Err - Ok - let me ask this in MUCH simpler way
1 - In dialplan , you set a Variable called MYVAR, to Apple
2 - You go into MACRO and NOOP the VALUE of MYVAR -What SHOULD it BE ?
3 - While IN MACRO you set VALUE of MYVAR = Pear
4 - You leave MACRO and get back to DIALPLAN and NOOP the value of MYVAR
- What should it be
=== 2nd Question ===
CAN the DIAL command call a SUBROUTINE instead of a MACRO ?
If so WOULD that help him out ?
Any clarification much apprecatted
Tilghman Lesher wrote:
On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and
would like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in
macros called by DIAL so that I can use these vars in the Dialplan or in
the h extention.
There isn't any good way to do that, period. When it comes to
inheritance, variables are only inherited from a master channel to a
slave channel. In the case of the Macro operating within the Dial, that
Macro is occurring exclusively on the slave channel. You cannot directly
set variables on other channels (for obvious race-condition reasons).
However, you could do this in a roundabout way, either by using a
database or by using shared variables in trunk. You'd need to first set
(in the master channel, before the Dial) an inherited variable containing
the name of the master channel, i.e. Set(_masterchan=${CHANNEL(name)}),
then use that inherited variable to set the shared variable in the master
channel from the slave channel, i.e. Set(SHARED(foo,${masterchan})=...).
Finally, you would be able to access the shared variable in the master
channel with ${SHARED(foo)}. Again, the SHARED function is only
available in trunk at this time, although you could probably backport it
to 1.4 with minimal trouble.
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Tilghman
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Re: [asterisk-users] vars in Macros called by DIAL with option M
On Tuesday 05 August 2008 18:04, Tilghman Lesher wrote:
On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and
would like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in
macros called by DIAL so that I can use these vars in the Dialplan or in
the h extention.
There isn't any good way to do that, period. When it comes to inheritance,
variables are only inherited from a master channel to a slave channel. In
the case of the Macro operating within the Dial, that Macro is occurring
exclusively on the slave channel. You cannot directly set variables on
other channels (for obvious race-condition reasons).
I see..
However, you could do this in a roundabout way, either by using a database
or by using shared variables in trunk. You'd need to first set (in the
master channel, before the Dial) an inherited variable containing the name
of the master channel, i.e. Set(_masterchan=${CHANNEL(name)}), then use
that inherited variable to set the shared variable in the master channel
from the slave channel, i.e. Set(SHARED(foo,${masterchan})=...). Finally,
you would be able to access the shared variable in the master channel with
${SHARED(foo)}. Again, the SHARED function is only available in trunk at
this time, although you could probably backport it to 1.4 with minimal
trouble.
I dont want to use trunk, but thanks for info...
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Re: [asterisk-users] vars in Macros called by DIAL with option M
On Tuesday 05 August 2008 18:02, Ruddy Gbaguidi wrote: I don't think you can do that because, asterisk, in the caller thread will only read MACRO_RESULT to know if he has to connect the call or not. A workaround will be to : 1. before the dial, add a row in a database table and retrieve an id 2. pass the id to test_connect and test_connect will then write his variable value into the database 3. after the dial,. use the id to retrieve the needed variable. thanks, but Iam using additional redirect though AMI, so it could be that the channel is redirected to another context and never see exten after DIAL or h extension in that context. It seemed to be that I have to add additional programming outside the dialplan if doing redirect. best regards Thomas ___ -- 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] vars in Macros called by DIAL with option M
On Tuesday 05 August 2008 18:50, Al Baker wrote:
Err - Ok - let me ask this in MUCH simpler way
1 - In dialplan , you set a Variable called MYVAR, to Apple
2 - You go into MACRO and NOOP the VALUE of MYVAR -What SHOULD it BE ?
You should better use
M(x[^arg]) - Execute the Macro for the *called* channel before connecting
to the calling channel. Arguments can be specified to the Macro
using '^' as a delimeter.
Its not a problem to get vars in the MACRO
3 - While IN MACRO you set VALUE of MYVAR = Pear
4 - You leave MACRO and get back to DIALPLAN and NOOP the value of MYVAR
- What should it be
regardless if you set MYVAR, _MYVAR or __MYVAR in the MACRO, it is not
working.
=== 2nd Question ===
CAN the DIAL command call a SUBROUTINE instead of a MACRO ?
If so WOULD that help him out ?
Any clarification much apprecatted
Tilghman Lesher wrote:
On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and
would like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in
macros called by DIAL so that I can use these vars in the Dialplan or in
the h extention.
There isn't any good way to do that, period. When it comes to
inheritance, variables are only inherited from a master channel to a
slave channel. In the case of the Macro operating within the Dial, that
Macro is occurring exclusively on the slave channel. You cannot directly
set variables on other channels (for obvious race-condition reasons).
However, you could do this in a roundabout way, either by using a
database or by using shared variables in trunk. You'd need to first set
(in the master channel, before the Dial) an inherited variable containing
the name of the master channel, i.e. Set(_masterchan=${CHANNEL(name)}),
then use that inherited variable to set the shared variable in the master
channel from the slave channel, i.e. Set(SHARED(foo,${masterchan})=...).
Finally, you would be able to access the shared variable in the master
channel with ${SHARED(foo)}. Again, the SHARED function is only
available in trunk at this time, although you could probably backport it
to 1.4 with minimal trouble.
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Re: [asterisk-users] vars in Macros called by DIAL with option M
On Tuesday 05 August 2008 14:19:44 Thomas Winter wrote:
On Tuesday 05 August 2008 18:04, Tilghman Lesher wrote:
On Tuesday 05 August 2008 10:47:34 Thomas Winter wrote:
Hi all,
Iam using an DIAL Command wird Macro if callee is answer the call.
exten = 123,n,DIAL(SIP/[EMAIL PROTECTED],180,gM(test_connect))
exten = 123,n,NoOp( ${var_from_macro})
In Macro test_connect Iam generating an new variable var_from_macro and
would like to use this var in the original dialplan.
I tried also __var_from_macro but didnt work. How can I set vars in
macros called by DIAL so that I can use these vars in the Dialplan or
in the h extention.
There isn't any good way to do that, period. When it comes to
inheritance, variables are only inherited from a master channel to a
slave channel. In the case of the Macro operating within the Dial, that
Macro is occurring exclusively on the slave channel. You cannot directly
set variables on other channels (for obvious race-condition reasons).
I see..
However, you could do this in a roundabout way, either by using a
database or by using shared variables in trunk. You'd need to first set
(in the master channel, before the Dial) an inherited variable containing
the name of the master channel, i.e. Set(_masterchan=${CHANNEL(name)}),
then use that inherited variable to set the shared variable in the master
channel from the slave channel, i.e. Set(SHARED(foo,${masterchan})=...).
Finally, you would be able to access the shared variable in the master
channel with ${SHARED(foo)}. Again, the SHARED function is only
available in trunk at this time, although you could probably backport it
to 1.4 with minimal trouble.
I dont want to use trunk, but thanks for info...
Just for the sake of showing how easy it was, I just backported func_shared
to 1.4. Took all of 5 minutes:
http://svncommunity.digium.com/view/tilghman/branches/1.4/
--
Tilghman
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[asterisk-users] When shall SIP phone reply 480 Temporarily Unavailable
Hello, When sending this AMI request ... 192.168.64.5 - Action: Originate 192.168.64.5 - Channel: SIP/9122 192.168.64.5 - Async: True 192.168.64.5 - Callerid: 9122 Guest2 9122 192.168.64.5 - Exten: 9123 192.168.64.5 - Context: local 192.168.64.5 - Priority: 1 ... I've got this INVITE from Asterisk INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK690b7694;rport From: 9121 Guest1 sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=as237a9159 To: sip:[EMAIL PROTECTED]:5060;user=phone Contact: sip:[EMAIL PROTECTED] [EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX As you can see, SIP From and To headers are different but both somehow refer to the peer. When receiving such INVITE, my SIP hardphone (a Thomson ST2030) replies with : SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK690b7694;rport From: 9121 Guest1sip:[EMAIL PROTECTED] [EMAIL PROTECTED] ;tag=as237a9159 To: sip:[EMAIL PROTECTED]:5060;user=phone;tag=c0a80101-a611e Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Content-Length: 0 Is it normal to reply this way ? I tried with another SIP phone (a Siemens Gigaset) and it accepted the INVITE (and started to ring). regards ___ -- 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] vars in Macros called by DIAL with option M
Thomas Winter wrote: On Tuesday 05 August 2008 18:02, Ruddy Gbaguidi wrote: I don't think you can do that because, asterisk, in the caller thread will only read MACRO_RESULT to know if he has to connect the call or not. A workaround will be to : 1. before the dial, add a row in a database table and retrieve an id 2. pass the id to test_connect and test_connect will then write his variable value into the database 3. after the dial,. use the id to retrieve the needed variable. Yes - this is possible, but the enormous ovehead to accomplish somthing that otherwise would seem to be very straightforward, i.e , get the value BACK from a very basic programming constract, call it a MACRO or Subrotine, just seems starteling and excessive. It seems to necessitate writing spagetti code. Or am I missing soimething ? ___ -- 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] email notification to external email address
All, I have a problem. The company I work for has been bought out by a bigger company. The employees are in the process of changing all their email addresses to the new company name. I have my voicemail.conf file setup to email users when they have a voicemail message. The mail server that was used to notify everybody is on the same network as the Asterisk PBX. Now I have to change all the emails in the voicemail.conf file to the new company's email addresses. The email server for them are external of the network that the Asterisk sits on. I have change a couple to test but the email notification is not happening. Any idea what is going on and how to resolve. Is there something else that I need to do to get the emails to work? I am new to the Asterisk and have been forced to take over for someone that has left the company. I do have telephony experience with Legacy systems. Any help is appreciated. -- Brian Simpson Tech Support Engineer Intertouch USA (614)856-1100 Support hotline (614)751-2018 Fax Intertouch USA is an entity of InterTouch Group of Companies; a NTT DoCoMo Group company Enhancing Guest Experience www.inter-touch.com www.maginet.net www.percipia.com www.nomadix.com www.azure.com.au NOTICE: The information and attachment(s) contained in this communication are intended for the addressee only, and may be confidential and/or legally privileged. If you have received this communication in error, please contact the sender immediately, and delete this communication from any computer or network system. Any interception, review, printing, copying, re-transmission, dissemination, or other use of, or taking of any action upon this information by persons or entities other than the intended recipient is strictly prohibited by law and may subject them to criminal or civil liability. None of the InterTouch Group of Companies shall be liable for the improper and/or incomplete transmission of the information contained in this communication or for the delay in its receipt. -- DOCOMO interTouch provides a full suite of integrated solutions to the hospitality industry. With over 1000 employees operating in 63 countries, DOCOMO interTouch is one of the world's largest hotel technology service providers, backed by mobile communications leader NTT DOCOMO. Email disclaimer - www.docomointertouch.com/Email_Disclaimer.aspx ___ -- 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] email notification to external email address
Brian Simpson wrote: All, I have a problem. The company I work for has been bought out by a bigger company. The employees are in the process of changing all their email addresses to the new company name. I have my voicemail.conf file setup to email users when they have a voicemail message. The mail server that was used to notify everybody is on the same network as the Asterisk PBX. Now I have to change all the emails in the voicemail.conf file to the new company's email addresses. The email server for them are external of the network that the Asterisk sits on. I have change a couple to test but the email notification is not happening. Any idea what is going on and how to resolve. Is there something else that I need to do to get the emails to work? I am new to the Asterisk and have been forced to take over for someone that has left the company. I do have telephony experience with Legacy systems. Any help is appreciated. Most likely the box is using sendmail or postfix to send those emails out. You need to setup sendmail/postfix to use a smarthost using smtp auth to allow relaying from this box. Lyle Giese LCR Computer Services, Inc. ___ -- 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] email notification to external email address
On Tue, Aug 5, 2008 at 3:10 PM, Brian Simpson [EMAIL PROTECTED] wrote: the network that the Asterisk sits on. I have change a couple to test but the email notification is not happening. Any idea what is going on Is sendmail installed and running? ___ -- 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] email notification to external email address
Hi Brian, Interesting company you work for :) - would be good to see some Asterisk integration into the azure network. Are you sure that the Asterisk server has SMTP access to the outside world? Just because you are changing the ip address of the server it was being sent to originally to the new server apart from changing the ip address etc the rest will be firewall/access issues. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian Simpson Sent: Tuesday, 5 August 2008 6:10 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] email notification to external email address All, I have a problem. The company I work for has been bought out by a bigger company. The employees are in the process of changing all their email addresses to the new company name. I have my voicemail.conf file setup to email users when they have a voicemail message. The mail server that was used to notify everybody is on the same network as the Asterisk PBX. Now I have to change all the emails in the voicemail.conf file to the new company's email addresses. The email server for them are external of the network that the Asterisk sits on. I have change a couple to test but the email notification is not happening. Any idea what is going on and how to resolve. Is there something else that I need to do to get the emails to work? I am new to the Asterisk and have been forced to take over for someone that has left the company. I do have telephony experience with Legacy systems. Any help is appreciated. -- Brian Simpson Tech Support Engineer Intertouch USA (614)856-1100 Support hotline (614)751-2018 Fax Intertouch USA is an entity of InterTouch Group of Companies; a NTT DoCoMo Group company Enhancing Guest Experience www.inter-touch.com www.maginet.net www.percipia.com www.nomadix.com www.azure.com.au NOTICE: The information and attachment(s) contained in this communication are intended for the addressee only, and may be confidential and/or legally privileged. If you have received this communication in error, please contact the sender immediately, and delete this communication from any computer or network system. Any interception, review, printing, copying, re-transmission, dissemination, or other use of, or taking of any action upon this information by persons or entities other than the intended recipient is strictly prohibited by law and may subject them to criminal or civil liability. None of the InterTouch Group of Companies shall be liable for the improper and/or incomplete transmission of the information contained in this communication or for the delay in its receipt. -- DOCOMO interTouch provides a full suite of integrated solutions to the hospitality industry. With over 1000 employees operating in 63 countries, DOCOMO interTouch is one of the world's largest hotel technology service providers, backed by mobile communications leader NTT DOCOMO. Email disclaimer - www.docomointertouch.com/Email_Disclaimer.aspx ___ -- 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] email notification to external email address
On Tuesday 05 August 2008 17:10:17 Brian Simpson wrote: I have a problem. The company I work for has been bought out by a bigger company. The employees are in the process of changing all their email addresses to the new company name. I have my voicemail.conf file setup to email users when they have a voicemail message. The mail server that was used to notify everybody is on the same network as the Asterisk PBX. Now I have to change all the emails in the voicemail.conf file to the new company's email addresses. The email server for them are external of the network that the Asterisk sits on. I have change a couple to test but the email notification is not happening. Any idea what is going on and how to resolve. Is there something else that I need to do to get the emails to work? I am new to the Asterisk and have been forced to take over for someone that has left the company. I do have telephony experience with Legacy systems. The most likely issue is that after changing voicemail.conf, you've failed to do 'voicemail reload', or, more generically, module reload app_voicemail.so which causes the configuration file to be re-parsed. -- Tilghman ___ -- 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] email notification to external email address
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lyle Giese Sent: Tuesday, August 05, 2008 6:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] email notification to external email address Brian Simpson wrote: All, I have a problem. The company I work for has been bought out by a bigger company. The employees are in the process of changing all their email addresses to the new company name. I have my voicemail.conf file setup to email users when they have a voicemail message. The mail server that was used to notify everybody is on the same network as the Asterisk PBX. Now I have to change all the emails in the voicemail.conf file to the new company's email addresses. The email server for them are external of the network that the Asterisk sits on. I have change a couple to test but the email notification is not happening. Any idea what is going on and how to resolve. Is there something else that I need to do to get the emails to work? I am new to the Asterisk and have been forced to take over for someone that has left the company. I do have telephony experience with Legacy systems. Any help is appreciated. As a temporary work around while you resolve the larger issue, you could install something like mailx or ssmtp to relay over any smtp server you currently have functioning access to. Thanks, Matt ___ -- 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] libpri versions 1.2.8 and 1.4.7, and libss7 version 1.0.1 released
The Asterisk development team has released new versions of three libraries used with Asterisk. They are: libpri-1.2.8: This release contains a number of bugfixes that had been unreleased for months, along with clarification of the licensing of the source code. The change log is here: http://downloads.digium.com/pub/telephony/libpri/ChangeLog-1.2.8 libpri-1.4.7: This release contains primarily only clarification of the licensing of the source code and some minor build system fixes. There is no need for users of version 1.4.6 to upgrade. The change log is here: http://downloads.digium.com/pub/telephony/libpri/ChangeLog-1.4.7 libss7-1.0.1: This release contains a number of bugfixes, along with clarification of the licensing of the source code. The change log is here: http://downloads.digium.com/pub/telephony/libss7/ChangeLog-1.0.1 Thanks for using Asterisk! -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- 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] email notification to external email address
Brian Simpson wrote: All, I have a problem. The company I work for has been bought out by a bigger company. The employees are in the process of changing all their email addresses to the new company name. I have my voicemail.conf file setup to email users when they have a voicemail message. The mail server that was used to notify everybody is on the same network as the Asterisk PBX. Now I have to change all the emails in the voicemail.conf file to the new company's email addresses. The email server for them are external of the network that the Asterisk sits on. I have change a couple to test but the email notification is not happening. Any idea what is going on and how to resolve. Is there something else that I need to do to get the emails to work? I am new to the Asterisk and have been forced to take over for someone that has left the company. I do have telephony experience with Legacy systems. Any help is appreciated. Your 1st step should be to test smtp funtionality to the outside world. I usually do that from the command line by typing 'mail [EMAIL PROTECTED]', for example: # mail [EMAIL PROTECTED] Subject: Test Email Body of Test . Cc: # You should do that while you have another console open doing a capture on port 25 (I use 'ngrep port 25'). That should give you a clue as to what is happening. Andres http://www.neuroredes.com ___ -- 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] Queue Penalties not working properly
Syed Nasruddin wrote: Actully the way I want the penalties functionality to behave it is not doing it accordingly. I am right now using ringall. Set penalty 1 for one agent and 2 for secnd agent. All the calls come in and go to first agent#1 having penalty one. But the second call also go to agent#1 and start waiting for it to be free rather it should have gone to penalty two agent#2 I have added call-limit=1 for bot sip accounts. And started the services. Still find the status wrong. nasr To do this the sip device must return a sip busy message from the device already on a call from the queue. Make sure that you disable call waiting on this line appearance. Anthony ___ -- 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] G722 capable soft phone?
Does anyone know where I might purchase a G.722 capable SIP soft phone? Counterpath say that they offer one, but only in the OEM versions do they support G.722. I need only a couple of licenses. Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- 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] Asterisk to Avaya
Steve, what kind of Avaya system is this? They make several.
On Tue, Aug 5, 2008 at 11:36 AM, Steve Davies [EMAIL PROTECTED] wrote:
Hi,
Sorry this is so long, but I am reasonably desparate.
I am having real fun with hooking an Avaya system to Asterisk using
ISDN30. I have the ISDN signalling all sorted one way, and can pass
calls from the real world (ie. the telco and asterisk) TO the avaya
box, and it accepts that and sets up the call perfectly.
The problem is that the Avaya box is signalling outbound calls using
an odd method, which smacks of an analogue system with ISDN30 bolted
on for a bit of a laugh.
They send a q931 SETUP message. This contains the correct callerID,
but only the first 1 to 4 of the dialled number's digits - The
remainder of the number is I believe passed through using DTMF!!! From
the look of it they intentionally do not send an IE 161 Sending
Complete with the SETUP, so that the far end continues to listen for
these DTMF tones, until it resolves to a legal number.
My questions for some ISDN expert out there...
Part 1) I need to receive the number in the SETUP, which I guess will
be in ${EXTEN}, then I assume I can use Read() to collect DTMF digits,
and check the dialplan to see if it is a locally terminated number.
Once I am 100% sure it is not local, I can then dial the collected
number through the Telco ISDN channel. Make sense? I think I can
probably handle that. The problem is that I do not know whether I have
received all digits from the Avaya at that point, which leads to...
Part 2) Can I dial through Zaptel (via a Sangoma card if that makes a
difference) without sending the IE 161 call complete? I thought that
Dial(Zap/G1||D(${INITIAL}))
might send the initial digits using DTMF, and then leave the channel
open so that more DTMF could follow over the now bridged channel. In
fact I get an immediate failure as if the far end thinks I have
finished dialling. Can I assume that libpri does not currently support
this method of dialling? If not, how might it be added? I can hack the
code, I just need suggestions of where to look and how it might sanely
be added :)
Part 3) It is possible that the Avaya is not using DTMF at-all, and
that it will send more bits of the called-party number using the
D-Channel as you would expect, but Asterisk does not seem to be
waiting for them. Can this be changed in Zaptel/Asterisk. Does anyone
know the Avaya systems well enough to suggest how it might be working?
Many many thanks for any feedback.
Regards,
Steve
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Re: [asterisk-users] Grandstream RS-232 config (slightly off-topic)
On Tue, Aug 5, 2008 at 6:29 PM, Vieri [EMAIL PROTECTED] wrote: Never mind. I set the wrong baud rate. The right values are 115200-8-N-1-No flow control. However, the serial connection is as good or as useless as the telent connection. I have no way to restore factory settings. --- On Tue, 8/5/08, Vieri [EMAIL PROTECTED] wrote: I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone unconfigurable via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the keypad update feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited and I can't change SIP configuration or reset to default values (and see if that helps to bring the HTTP back up). I did try to upgrade and downgrade the firmware with no change at all. The reset button reboots the device but doesn't restore default values! (I tried keeping it pressed for several minutes...) So my last chance before throwing it away is to administer it via serial port. The thing is that Grandstream's official manual says that there is an RS232 serial port for administration but it doesn't say anything else about it (how to connect, how to change config, how to reset the device, etc). There's absolutely nothing regarding RS-232. If someone has this or a similar device and accessed it via serial port then I'd greatly appreciate some quick tips. Thanks, Vieri Have you tried powering it on, while holding reset button? Additionally you can try to leave it for week powered off and hope that there's some old battery keeping up settings. Are you sure that there isn't some enable admin mode command in telnet? It should allow you everything that's available from web. Regards, Atis -- 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 -- 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] Least Cost Routing
ASTPP (www.astpp.org) will do calling cards / prepaids as well as lcr. Darren Wiebe [EMAIL PROTECTED] emist wrote: Hello, does anyone know of a good calling card solution for asterisk that is able to do lcr? Does astcc do this? I've been searching around and I can find some lcr modules/apps but none that incorporate prepaid card functionality. Regards, Igor H. ___ -- 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] email notification to external email address
On Tue, Aug 05, 2008 at 07:41:05PM -0400, Andres wrote: Brian Simpson wrote: All, I have a problem. The company I work for has been bought out by a bigger company. The employees are in the process of changing all their email addresses to the new company name. I have my voicemail.conf file setup to email users when they have a voicemail message. The mail server that was used to notify everybody is on the same network as the Asterisk PBX. Now I have to change all the emails in the voicemail.conf file to the new company's email addresses. The email server for them are external of the network that the Asterisk sits on. I have change a couple to test but the email notification is not happening. Any idea what is going on and how to resolve. Is there something else that I need to do to get the emails to work? I am new to the Asterisk and have been forced to take over for someone that has left the company. I do have telephony experience with Legacy systems. Any help is appreciated. Your 1st step should be to test smtp funtionality to the outside world. I usually do that from the command line by typing 'mail [EMAIL PROTECTED]', for example: # mail [EMAIL PROTECTED] You're confusing mail/mailx and sendmail. mail/mailx generates the Subject and such headers on its own. I think the intended command here was sendmail. Subject: Test Email Body of Test . Cc: # You should do that while you have another console open doing a capture on port 25 (I use 'ngrep port 25'). That should give you a clue as to what is happening. Or better: the logs of the mail server. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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
