RE: [asterisk-users] Detecting no answers and/or disconnected numbers
Hi, I am interested about that, too. If someone have some more informations... Greg Hi, Using call files, is there a way to identify no answered calls from disconnected numbers (no longer in service). Both return the same value and so far I can not find a way to know one from the other. Thank you, Andre Courchesne ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ParkAndAnnounce + Paging
http://www.voip-info.org/wiki/index.php?page=Asterisk+local+channels On 12/6/06, Apesys [EMAIL PROTECTED] wrote: Hi everybody. It is possible to announce the parking position through a paging to a group of extensions? I would like that when someone parks a call, some phones will announce with the speaker the position. Something like: exten = s,1,ParkAndAnnounce(call-parked-at:PARKED|30|PAGE( LOCAL/[EMAIL PROTECTED]LOCAL/[EMAIL PROTECTED]|) Is there a way, maybe with a different approach? Thanks, Pol Po ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] calls not terminating (2nd posting)
(Posting 2nd time - I've had this problem on both my 1st and 2nd installation of asterisk) Hi, In short - Asterisk is not able to recognize that the 'other' person to whom call was made (using ZAP channel) has hung up - hence the channel stays busy and unusable. This is when zone is set to 'us' and my location in Pakistan. In long: I've been able to configure asterisk 1.1.12 with freepbx 2.1.3 and Digium TDM400 card (4 FXO ports). I've terminated 2 PSTN lines on these ports and making outbound calls successfully using the outbound rules. However, if the 2nd party hangs the call, this is not detected and the Flash panel (FOP) shows the ZAP channel as busy. This continues until the line is dropped on the XLITE too. The line is not available Ever weirder - once my PC rebooted in process of a call, so there was no way I could hang up using the PC. The other party hung up. Even then the zap channel stayed busy. The Xlite extension which made the original call appeared as busy on FOP, but was able to make and receive calls using line-2. I had to restart the server for channel to become free and FOP to reflect that. What could be the places I look at since I am using mostly original/default settings, making only dial-plan changes? I am residing in Pakistan, but since the Regards, Farzal -- Farzal Ali Dojki PK: 92-21-2635021-24 | US: 1-512-STAY-UBM Telecom :: Call Centre :: Security :: Computing http://www.ubm.com.pk [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?
On Wed, Dec 06, 2006 at 10:12:25PM -0600, Lacy Moore - Aspendora wrote: On 12/6/06, John Novack [EMAIL PROTECTED] wrote: Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run into some gotcha down the road where there is some missing file that needs to be put who knows where. Wow! Are you sure about that? That is a workaround for not using the package management system. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Requested transfer capability: 0x00 - SPEECH - How to change to 31KAUDIO?
Hi everyone! I'm having an issue calling the Numbers Information Service (similiar to 411 in the US) in my country. I use: TE110P, connected to a PRI line running on an E1. Besides that specific number, all calls pass through fine, in and out, no problems whatsoever. I called my Telco, and the guy did a comparison of my PRI call setup, and other calls that pass through and get fine to the Numbers Information Service. The only difference he could find, is that every other PBX in my country (mostly proprietary ones, I would assume...) - Request a transfer capability of 31KAUDIO, which I assume means 3.1KHz audio. I also assume SPEECH is actually 8KHz., which is more, and probably with higher quality. The guy at the telco said it may be the problem, maybe because the end system cannot reach the desired voice quality, or whatever (he never encountered that problem before...) I called Digium's hardware installation service, and they told me to change prilocaldialplan to unknown (I had it set to local before); I am not sure how this is related, because it seems to me not related to dialing at all - but Digium made the software and the hardware, so they must know :-) Anyways, that advice didn't really much help - my calls are still going out requesting SPEECH: -- Requested transfer capability: 0x00 - SPEECH Anybody has any advice on how to change this? Thanks! -- Shimi ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
Doug Crompton wrote: I remembered I had an x10 bottlerocket in my X10 junkbox so I connected it to a spare serial port on my linux server (asterisk resides there) and implemented with some mods the code mentioned earlier http://lorance.freeshell.org/asterisk/#asterisk-can-control-the-world and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. I would suggest that people who don't already have an investment in home automation equipment should look at Insteon rather than X10. Insteon is a next generation version of X10 that provides backwards compatibility with X10. The devices are a little more expensive, but not as expensive as some of the other alternatives. Insteon provides 2 way communication and is a lot more reliable than X10. If you already have an investment in X10 devices you can slowly convert to Insteon, since Insteon provides backwards compatibility, i.e. X10 controllers can control Insteon devices and Insteon controllers can control X10 devices, however you won't get all the advantages of Insteon until you have Insteon controllers controlling Insteon devices. For people with some soldering and basic circuit design skills, you may want to consider using ethernet as a home automation bus for some things. I love the Olimex PIC WEB and PIC Mini Web development boards (they cost $49.95 and $39.95 respectively). They have an ethernet port and an expansion connector for the available PIC I/O pins. Microchip provides a free C compiler for Pic processors, and they also have an open source networking stack that works on the Olimex boards. So with a ribbon cable connector and a small breadboard with a few IC's and/or driver transistors you can build a device that responds to commands via the network (or via a built in web server) from your Asterisk server that does about any task you can think of. Lots of fun ... I'm currently building a voicemail indicator (my wife didn't like me taking her answering machine away with the blinking lights when we switched to Asterisk voicemail) using a PIC Web board. Next project will be a web based sprinkler controller. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI across multiple servers
I decided to write my own simple voicemail application via AGI and store all voicemails in MySQL. The nice thing was the user can retrieve via phone (local and remote), via email attachment and also via web download. You can listen to old and new messages and change your outgoing message too. Regards Jon Jon Farmer Telford, Shropshire, UK - Original Message From: Porier, Jeremy M. [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 6 December, 2006 4:20:04 PM Subject: [asterisk-users] MWI across multiple servers We are about to deploy six Asterisk servers across the state with SIP phones at each site registering to their local server. However, we are centralizing voicemail at our main campus to enable the transfer of voicemails between users regardless of site. It also simplifies our backup procedures for voicemail. Any tips for distributing MWI messages amongst those separate servers that phones are registering to? I suppose I could script something on the voicemail server to put a file in the inbox on the distributed servers but perhaps there is something more elegant I'm unaware of? If not, has anyone scripted this before and willing to share? Would be much appreciated. Thanks, Jeremy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Send instant messages to your online friends http://uk.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to configure Asterisk to support SIP INFO method?
Hi all, I have a question: how to configure Asterisk to support SIP INFO method? I encountered this problem when I find my UA don't send INFO message to another UA, actually it should. Asterisk was used as a SIP proxy in this scenario (I know that SIP is not a SIP proxy:-)). Then I captured all the packets and found that my UA send REGISTER to Asterisk, and include Allow: ... INFO ... in its SIP message. But in the 200 OK message replied by Asterisk, there is no Allow fields included, or only Allow without INFO in it. So I guess that's why my UA don't send INFO SIP message. I think one way to solve the problem is to make Asterisk support INFO, but I don't know whether it's feasible or how to configure. Could anyone help me? Thanks. Jenny Cheung _ 率先尝试 Windows Live Mail。 http://ideas.live.com/programpage.aspx?versionId=5d21c51a-b161-4314-9b0e-4911fb2b2e6d___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] oh323.conf question
Hi all, I would like to know if it exists the possibility to send to different context according to the caller IP Addres I receive H323 calls, and I have to route this to different devices according to the caller ip. I tried to use the context=first-context alias=99 context=second-context alias=88 but I was not able to succed in this; Moreover, I think the keyword alias is related to the phone calling more than to the ip address, and it could be anything... In other terms, what I have to do is to send all the calls from one IP Address to a zap group, and all the calls coming from another IP to another zap group. Any help will be gratly appreciated, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip qualify unreachable/reachable - ci$co 7940
I have logs full with this messages... I must have qualify turned on, because phone is behind firewall, main problem si, that phone is each hour about one hour unavailable! :'( I tried to modify minexpiry/maxexpiry sip.conf timeouts, but nothing help me. I'm using latest firmware 8.4 in phone, will be better to downgrade? to what version? (latest asterisk 1.4branch) [Dec 7 00:36:56] NOTICE[19226] chan_sip.c: Peer '108' is now UNREACHABLE! Last qualify: 205 [Dec 7 01:36:23] NOTICE[19226] chan_sip.c: Peer '108' is now Reachable. (189ms / 2000ms) [Dec 7 02:36:41] NOTICE[19226] chan_sip.c: Peer '108' is now UNREACHABLE! Last qualify: 198 [Dec 7 03:36:22] NOTICE[19226] chan_sip.c: Peer '108' is now Reachable. (199ms / 2000ms) [Dec 7 04:36:38] NOTICE[19226] chan_sip.c: Peer '108' is now UNREACHABLE! Last qualify: 177 [Dec 7 05:36:05] NOTICE[19226] chan_sip.c: Peer '108' is now Reachable. (201ms / 2000ms) [Dec 7 06:36:21] NOTICE[19226] chan_sip.c: Peer '108' is now UNREACHABLE! Last qualify: 156 [Dec 7 07:36:02] NOTICE[19226] chan_sip.c: Peer '108' is now Reachable. (177ms / 2000ms) [Dec 7 08:36:19] NOTICE[19226] chan_sip.c: Peer '108' is now UNREACHABLE! Last qualify: 210 [Dec 7 09:35:46] NOTICE[19226] chan_sip.c: Peer '108' is now Reachable. (201ms / 2000ms) [Dec 7 10:36:02] NOTICE[19226] chan_sip.c: Peer '108' is now UNREACHABLE! Last qualify: 151 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] eicon diva BRI problems
Hi (Armin?) ! Today I had a problem with Diva Server 4BRI-8M 2.0. Asterisk 1.2.12.1 chan_capi-cm-0.6.5 divas4linux-melware-3.0.f-106.622-1 Asterisk could not receive and make calls on the BRI ports, although the ports looked fine within Asterisk. I usually use /usr/lib/divas/divactrl dchannel -c 1 to test line activity. This time there was no activity (cryptic log messages) (I waited for 10 minutes). Then I restarted Asterisk - but no improvement. Then I wanted to remove and reload the diva kernel modules, but /usr/lib/divas/divas_stop.rc could not remove the modules. Also manual remove did not worked (module still in use). Then I rebooted the server and also updated to chan_capi-0.7.1 divas4linux-melware-3.0.5-106.702-1 as there were problem with removing the kernel modules and the Asterisk restart did not helped I suspect there is a bug in the diva kernel modules. I searched for a changelog of the diva drivers, but couldn'T find them. Do you know of such a bug fixed in the newest version? Do you know of a changelog of the divas melware drivers? thanks klaus -- Klaus Darilion nic.at ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MWI across multiple servers
On Dec 7, 2006, at 4:14 AM, Jon Farmer wrote: I decided to write my own simple voicemail application via AGI and store all voicemails in MySQL. The nice thing was the user can retrieve via phone (local and remote), via email attachment and also via web download. You can listen to old and new messages and change your outgoing message too. Regards Jon Jon, Maybe you could post this application and a how-to to the wiki? Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] -- Called [EMAIL PROTECTED] Segmentation fault (core dumped)
OOH323 Debugging Enabled -- Executing Answer(SIP/3513-090f7d40, ) in new stack -- Executing Wait(SIP/3513-090f7d40, 1) in new stack -- Executing DeadAGI(SIP/3513-090f7d40, a2billing.php|1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1 a2billing.php|1: a2billing.php|1: line:67 - MODE : standard a2billing.php|1: -- AGI Script Executing Application: (Dial) Options: ( OOH323/[EMAIL PROTECTED]|60|HLxyz(540:31000:0)) --- ooh323_request - data [EMAIL PROTECTED] format 0x4 (ulaw) --- find_peer +++ find_peer +++ ooh323_request --- ooh323_call- [EMAIL PROTECTED] +++ ooh323_call -- Called [EMAIL PROTECTED] Segmentation fault (core dumped) [EMAIL PROTECTED] ~]# show version Asterisk 1.2.12.1 built by root @ localhost.localdomain on a i686 running Linux on 2006-10-18 18:35:57 UTC ; Objective System's H323 Configuration example for Asterisk ; ooh323c driver configuration ; ; [general] section defines global parameters ; ; This is followed by profiles which can be of three types - user/peer/friend ; Name of the user profile should match with the h323id of the user device. ; For peer/friend profiles, host ip address must be provided as dynamic is ; not supported as of now. ; ; Syntax for specifying a H323 device in extensions.conf is ; For Registered peers/friends profiles: ;OOH323/name where name is the name of the peer/friend profile. ; ; For unregistered H.323 phones: ;OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323 ; alias ; ; For dialing into another asterisk peer at a specific exten ; OOH323/exten/peer OR OOH323/[EMAIL PROTECTED] ; ; Domain name resolution is not yet supported. ; ; When a H.323 user calls into asterisk, his H323ID is matched with the profile ; name and context is determined to route the call ; ; The channel driver will register all global aliases and aliases defined in ; peer profiles with the gatekeeper, if one exists. So, that when someone ; outside our pbx (non-user) calls an extension, gatekeeper will route that ; call to our asterisk box, from where it will be routed as per dial plan. [general] ;Define the asetrisk server h323 endpoint ;The port asterisk should listen for incoming H323 connections. ;Default - 1720 port=1720 ;The dotted IP address asterisk should listen on for incoming H323 ;connections ;Default - tries to find out local ip address on it's own bindaddr=213.138.36.153 ;This parameter indicates whether channel driver should register with ;gatekeeper as a gateway or an endpoint. ;Default - no gateway=yes ;Whether asterisk should use fast-start and tunneling for H323 connections. ;Default - yes faststart=no ;h245tunneling=yes ;H323-ID to be used for asterisk server ;Default - Asterisk PBX h323id=ObjSysAsterisk ;e164=100 ;CallerID to use for calls ;Default - Same as h323id callerid=asterisk ;Whether this asterisk server will use gatekeeper. ;Default - DISABLE ;gatekeeper = DISCOVER gatekeeper = 213.138.36.153 ;gatekeeper = DISABLE ;Location for H323 log file ;Default - /var/log/asterisk/h323_log logfile=/var/log/asterisk/h323_log ;Following values apply to all users/peers/friends defined below, unless ;overridden within their client definition ;Sets default context all clients will be placed in. ;Default - default context=default ;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;Type of Service ;Default - none (lowdelay, thoughput, reliability, mincost, none) ;tos=none ;amaflags = default ;The account code used by default for all clients. ;accountcode=h3230101 ;The codecs to be used for all clients.Only ulaw and gsm supported as of now. ;Default - ulaw ; ONLY ulaw, gsm, g729 and g7231 supported as of now allow=all ;Note order of disallow/allow is important. ;allow=g729 ;allow=ulaw ; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad ; h245alphanumeric, h245signal. ;Default - rfc 2833 dtmfmode=rfc2833 ; User/peer/friend definitions: ; User config optionsPeer config options ; -- --- ; context ; disallow disallow ; allow allow ; accountcodeaccountcode ; amaflags amaflags ; dtmfmode dtmfmode ; rtptimeout ip ;port ;h323id ;email ;url ;e164 ;rtptimeout ; ;Define users here ;Section header is extension ;[myuser1] ;type=user
[asterisk-users] Re: -- Called [EMAIL PROTECTED] Segmentation fault (core dumped)
Help me ooh323 core dumped. 2006/12/7, Ümit AYDINLI [EMAIL PROTECTED]: OOH323 Debugging Enabled -- Executing Answer(SIP/3513-090f7d40, ) in new stack -- Executing Wait(SIP/3513-090f7d40, 1) in new stack -- Executing DeadAGI(SIP/3513-090f7d40, a2billing.php|1) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php a2billing.php|1: line:58 - IDCONFIG : 1 a2billing.php|1: a2billing.php|1: line:67 - MODE : standard a2billing.php|1: -- AGI Script Executing Application: (Dial) Options: ( OOH323/[EMAIL PROTECTED]|60|HLxyz(540:31000:0OOH323/[EMAIL PROTECTED](540:31000:0 )) --- ooh323_request - data [EMAIL PROTECTED] format 0x4 (ulaw) --- find_peer +++ find_peer +++ ooh323_request --- ooh323_call- [EMAIL PROTECTED] +++ ooh323_call -- Called [EMAIL PROTECTED] Segmentation fault (core dumped) [EMAIL PROTECTED] ~]# show version Asterisk 1.2.12.1 built by root @ localhost.localdomain on a i686 running Linux on 2006-10-18 18:35:57 UTC ; Objective System's H323 Configuration example for Asterisk ; ooh323c driver configuration ; ; [general] section defines global parameters ; ; This is followed by profiles which can be of three types - user/peer/friend ; Name of the user profile should match with the h323id of the user device. ; For peer/friend profiles, host ip address must be provided as dynamic is ; not supported as of now. ; ; Syntax for specifying a H323 device in extensions.conf is ; For Registered peers/friends profiles: ;OOH323/name where name is the name of the peer/friend profile. ; ; For unregistered H.323 phones: ;OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323 ; alias ; ; For dialing into another asterisk peer at a specific exten ; OOH323/exten/peer OR OOH323/[EMAIL PROTECTED] ; ; Domain name resolution is not yet supported. ; ; When a H.323 user calls into asterisk, his H323ID is matched with the profile ; name and context is determined to route the call ; ; The channel driver will register all global aliases and aliases defined in ; peer profiles with the gatekeeper, if one exists. So, that when someone ; outside our pbx (non-user) calls an extension, gatekeeper will route that ; call to our asterisk box, from where it will be routed as per dial plan. [general] ;Define the asetrisk server h323 endpoint ;The port asterisk should listen for incoming H323 connections. ;Default - 1720 port=1720 ;The dotted IP address asterisk should listen on for incoming H323 ;connections ;Default - tries to find out local ip address on it's own bindaddr=213.138.36.153 ;This parameter indicates whether channel driver should register with ;gatekeeper as a gateway or an endpoint. ;Default - no gateway=yes ;Whether asterisk should use fast-start and tunneling for H323 connections. ;Default - yes faststart=no ;h245tunneling=yes ;H323-ID to be used for asterisk server ;Default - Asterisk PBX h323id=ObjSysAsterisk ;e164=100 ;CallerID to use for calls ;Default - Same as h323id callerid=asterisk ;Whether this asterisk server will use gatekeeper. ;Default - DISABLE ;gatekeeper = DISCOVER gatekeeper = 213.138.36.153 ;gatekeeper = DISABLE ;Location for H323 log file ;Default - /var/log/asterisk/h323_log logfile=/var/log/asterisk/h323_log ;Following values apply to all users/peers/friends defined below, unless ;overridden within their client definition ;Sets default context all clients will be placed in. ;Default - default context=default ;Sets rtptimeout for all clients, unless overridden ;Default - 60 seconds ;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity ; when we're not on hold ;Type of Service ;Default - none (lowdelay, thoughput, reliability, mincost, none) ;tos=none ;amaflags = default ;The account code used by default for all clients. ;accountcode=h3230101 ;The codecs to be used for all clients.Only ulaw and gsm supported as of now. ;Default - ulaw ; ONLY ulaw, gsm, g729 and g7231 supported as of now allow=all ;Note order of disallow/allow is important. ;allow=g729 ;allow=ulaw ; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad ; h245alphanumeric, h245signal. ;Default - rfc 2833 dtmfmode=rfc2833 ; User/peer/friend definitions: ; User config optionsPeer config options ; -- --- ; context ; disallow disallow ; allow allow ; accountcodeaccountcode ; amaflags amaflags ; dtmfmode dtmfmode ; rtptimeout ip ;port ;h323id ;email ;url ;e164
[asterisk-users] Plantronics and Snom RF feedback
Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360), I noticed my client is having some form of feed back on the phone. Because of Snom's inner oddities this is how I got it to work. Plantronic -- RJ11 -- SnomHandset Port (on Snom Base) Handset -- Plantronic jack (bottom base in the front) If I placed Plantronic(RJ11) -- Snom's Headset port, the auto lift on the Plantronic wouldn't work until the person pressed the headset key. Even by leaving the headset key on by default, Snom would revert to normal (non headset) mode whenever the headset piece was used. (Sort of defeats the purpose of walking away from your phone only to walk back to re-press the headset key)... How are others setting up these Plantronics... -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Running Asterisk on a Home rotuer
Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Standardized IVR UI Pattern (was: Re: [asterisk-users] Is there any Asterisk controllable thermostat?)
On Wed, 2006-12-06 at 23:51 -0700, [EMAIL PROTECTED] wrote: Date: Wed, 06 Dec 2006 22:37:01 -0500 From: Steve Prior [EMAIL PROTECTED] Subject: Re: [asterisk-users] Is there any Asterisk controllable thermostat? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Doug Crompton wrote: and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. Doug I've started to play with writing some code using the Java FastAGI interface to connect to my home automation system. The code is working and I could now write whatever I wanted, but I haven't figured out what would be a reasonable menu interface that wouldn't be very annoying to use. I'd be very interested to hear what menu structures and what actual capabilities people have found useful and nice to use. For example, has anyone come up with something less annoying than the following dialog: Press 1 for living room, press 2 for outside, press 3 for bedroom (I press 2) Press 1 for porch light, press 2 for garage light (I press 1) Press 1 to turn on, Press 2 to turn off, Press 3 to say current status (I press 1) congratulations, you just spent several minutes just to turn on a light! I don't know why IVR menus still include so much extra verbiage. They should act like numbered lists - everyone knows the stated number means the key to press, and the stated name means what you will get. So: (Listens for DTMF) Hello, this is home thermostat. 1 living room 2 outside 3 bedroom (waits for DTMF, maybe repeats after a 2 second pause) (I press 2) (Listens for DTMF) Outside 1 porch light 2 garage light (waits for DTMF, maybe repeats after a 2 second pause, offers to hangup after maybe 15 seconds) (I press 1) (Listens for DTMF) Outside Porch light 1 on 2 off 3 say current status (waits for DTMF, maybe repeats after a 2 second pause) (I press 1) (Listens for DTMF) Outside porch light status turned on star for options, hash to hangup (waits for DTMF, maybe repeats after a 2 second pause) That menu system would take about 10 seconds the first time through, listening to all prompts. Subsequent navigation could take 2-4 seconds. Subsequent shortcuts through a collapsed star-hash menu could take 1-2 seconds. Make the star key an escape key to the previous scope. Make the hash key an Enter key that terminates any multiple-key entry. Collapse all menu scopes/items into a single long list that can be reached at any time through star-hash. Introduce the whole menu system with press star for options, to the star-star menu. Make the 0 option in the star options menu the path to a human operator, if there is one. And always immediately feedback to any received key with at least a click. This simple UI should be common to every IVR app, so anyone can always use it without listening for a while to learn how to navigate the IVR. In fact, I call this system IKR (Interactive Key Response), and maybe every system should answer the call with first saying IKR. Then callers would immediately know when our skills on the common UI would work, without waiting to learn, or mistake it. If the server played a few touchtones, like 4-5-7 (keypad IKR) while saying IKR, smart automated clients could detect the system and use it. To complete the interactivity protocol, every spoken digit to be pressed in the numbered menus would also play the digits' DTMF. And the intro to the scope to which a client DTMF navigated would play the last digits that navigated there from the previous scope while saying the name of the new scope. This is the system that I used to use when I built dedicated IVR systems a dozen years ago (on Dialogic HW). Almost no IVR people were on the Internet then, before the Web. There was no community, and IVR vendors competed so harshly that they couldn't get such a standard interface going, even for mutual benefit. So now everyone hates using IVR, even when it's better than a human operator. And we still all roll our own from scratch. But with Asterisk, and web/maillists connecting a community, we can adopt a common system. If enough people like it, I will publish the spec, and maybe write the RFC. Or maybe there's a better one that will be adopted more widely more quickly, and we can get behind that. If you don't like it, you can still roll your own, just don't call it IKR when answering the call, and callers will be free to use your klugey, nonstandard UI, and hate it :). -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Plantronics and Snom RF feedback
JO == J Oquendo [EMAIL PROTECTED] writes: JO Plantronic -- RJ11 -- SnomHandset Port (on Snom Base) Handset JO -- Plantronic jack (bottom base in the front) If I placed JO Plantronic(RJ11) -- Snom's Headset port, the auto lift on the JO Plantronic wouldn't work until the person pressed the headset JO key. If auto lift means the mechanical lifter, then you should not use the head set jack on the Snom at all. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk accepting calls to fast
Hi, the german telco Colt Telekom has assigned the phone number block 56830-xxx to one of our customers. In the diaplan we have setup extensions like the following ones: exten = 56830910,1,Answer() exten = 56830910,2,Dial(SIP/bduerring,10,tr) exten = 56830910,3,VoiceMail,u20 exten = 56830910,4,hangup exten = 56830910,103,VoiceMail,b20 exten = 56830910,104,hangup exten = 5683091,1,Answer() exten = 5683091,2,DIAL(ZAP/g5/56830990,10,r) exten = 5683091,3,Hangup The problem now is, that sometimes (maybe when the caller doesn't hit the buttons fast enough) asterisk takes the extension for 5683091, although the 0 is still coming a little bit later. I'm not quite sure whether the delay in transferring the numbers is caused by the caller or by the telco. But is their a chance to tell asterisk to wait a little bit longer, before it starts searching the extensions.conf? Or do I have to tell the ISDN card to wait for the complete number, before it is forwarded to asterisk? Software hardware: SuSE 10.0 Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1p chan_capi-0.7.0 divas4linux_EICON-106.20-1 Eicon Networks Corporation Diva Server 4BRI Rev 2 Thanks for your help hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] eicon diva BRI problems
On Thu, 7 Dec 2006, Klaus Darilion wrote: Hi (Armin?) ! Today I had a problem with Diva Server 4BRI-8M 2.0. Asterisk 1.2.12.1 chan_capi-cm-0.6.5 divas4linux-melware-3.0.f-106.622-1 Asterisk could not receive and make calls on the BRI ports, although the ports looked fine within Asterisk. I usually use /usr/lib/divas/divactrl dchannel -c 1 to test line activity. This time there was no activity (cryptic log messages) (I waited for 10 minutes). Then I restarted Asterisk - but no improvement. Then I wanted to remove and reload the diva kernel modules, but /usr/lib/divas/divas_stop.rc could not remove the modules. Also manual remove did not worked (module still in use). Then I rebooted the server and also updated to chan_capi-0.7.1 divas4linux-melware-3.0.5-106.702-1 as there were problem with removing the kernel modules and the Asterisk restart did not helped I suspect there is a bug in the diva kernel modules. Yes, such an error is caused by kernel modules. I searched for a changelog of the diva drivers, but couldn'T find them. Do you know of such a bug fixed in the newest version? No, I never had such problems (or received a report about that). Do you know of a changelog of the divas melware drivers? I don't have a changelog. If this problem appears again, please create a memory dump of the cards memory (divactrl can do that). This will help to find the problem, but the latest driver/firmware should be used. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Backgroung usage
Hello, I try to use the background cmd for send incomings call on dial plan. I try in an internal number for resting: exten = 405,1,DigitTimeout,5 exten = 405,2,ResponseTimeout,10 exten = 405,3,Background(vm-accueilcreat) exten = 1,1,Goto(creat-in,s,1) exten = 2,1,Dial(IAX2/301,15,tr) exten = 3,1,Hangup But nothing happen when i hit 1, 2, or 3. Wher is the mistake?? Best regards, -- Olivier Saulnier STEGANUX 1er étage DIAMECANS BEL AIR 03410 St-Victor T: 04.70.02.27.62 F: 04.70.09.97.41 http://www.steganux.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: Standardized IVR UI Pattern (was: Re: [asterisk-users] Is thereany Asterisk controllable thermostat?)
If you want a standardized ivr ui pattern, wouldn't something like VoiceXML be interesting? That's a standard for use with IVR applications. Jon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Rubenstein Sent: 7. december 2006 15:53 To: Asterisk-Users Subject: Standardized IVR UI Pattern (was: Re: [asterisk-users] Is thereany Asterisk controllable thermostat?) On Wed, 2006-12-06 at 23:51 -0700, [EMAIL PROTECTED] wrote: Date: Wed, 06 Dec 2006 22:37:01 -0500 From: Steve Prior [EMAIL PROTECTED] Subject: Re: [asterisk-users] Is there any Asterisk controllable thermostat? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Doug Crompton wrote: and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. Doug I've started to play with writing some code using the Java FastAGI interface to connect to my home automation system. The code is working and I could now write whatever I wanted, but I haven't figured out what would be a reasonable menu interface that wouldn't be very annoying to use. I'd be very interested to hear what menu structures and what actual capabilities people have found useful and nice to use. For example, has anyone come up with something less annoying than the following dialog: Press 1 for living room, press 2 for outside, press 3 for bedroom (I press 2) Press 1 for porch light, press 2 for garage light (I press 1) Press 1 to turn on, Press 2 to turn off, Press 3 to say current status (I press 1) congratulations, you just spent several minutes just to turn on a light! I don't know why IVR menus still include so much extra verbiage. They should act like numbered lists - everyone knows the stated number means the key to press, and the stated name means what you will get. So: (Listens for DTMF) Hello, this is home thermostat. 1 living room 2 outside 3 bedroom (waits for DTMF, maybe repeats after a 2 second pause) (I press 2) (Listens for DTMF) Outside 1 porch light 2 garage light (waits for DTMF, maybe repeats after a 2 second pause, offers to hangup after maybe 15 seconds) (I press 1) (Listens for DTMF) Outside Porch light 1 on 2 off 3 say current status (waits for DTMF, maybe repeats after a 2 second pause) (I press 1) (Listens for DTMF) Outside porch light status turned on star for options, hash to hangup (waits for DTMF, maybe repeats after a 2 second pause) That menu system would take about 10 seconds the first time through, listening to all prompts. Subsequent navigation could take 2-4 seconds. Subsequent shortcuts through a collapsed star-hash menu could take 1-2 seconds. Make the star key an escape key to the previous scope. Make the hash key an Enter key that terminates any multiple-key entry. Collapse all menu scopes/items into a single long list that can be reached at any time through star-hash. Introduce the whole menu system with press star for options, to the star-star menu. Make the 0 option in the star options menu the path to a human operator, if there is one. And always immediately feedback to any received key with at least a click. This simple UI should be common to every IVR app, so anyone can always use it without listening for a while to learn how to navigate the IVR. In fact, I call this system IKR (Interactive Key Response), and maybe every system should answer the call with first saying IKR. Then callers would immediately know when our skills on the common UI would work, without waiting to learn, or mistake it. If the server played a few touchtones, like 4-5-7 (keypad IKR) while saying IKR, smart automated clients could detect the system and use it. To complete the interactivity protocol, every spoken digit to be pressed in the numbered menus would also play the digits' DTMF. And the intro to the scope to which a client DTMF navigated would play the last digits that navigated there from the previous scope while saying the name of the new scope. This is the system that I used to use when I built dedicated IVR systems a dozen years ago (on Dialogic HW). Almost no IVR people were on the Internet then, before the Web. There was no community, and IVR vendors competed so harshly that they couldn't get such a standard interface going, even for mutual benefit. So now everyone hates using IVR, even when it's better than a human operator. And we still all roll our own from scratch. But with Asterisk, and web/maillists connecting a community, we can adopt a common system. If enough people like it, I will publish the spec, and maybe write the RFC. Or maybe there's a better one that will be adopted more widely more quickly, and we can get behind that. If you don't like it, you can
Re: [asterisk-users] Running Asterisk on a Home rotuer
It may not be what you're thinking, but I use Astlinux on an older PIII. With a couple of options it has become my home router and works very well. On 12/7/06, Dovid B [EMAIL PROTECTED] wrote: Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
John, Two questions on your comments I have no seen an Insteon computer controller similiar to the old bottle rocket. Is there such a device? I am thinking of getting an Insteon starter kit bit I have so many X10 devices it will be awhie before, if ever, that I get it all changed over. Many items, like spotlights, are not available in Insteon. I would be interested in the Ethernet MWI. I am using many phones on an SPA3000 fxs and I can't seem to find an MWI on an analog phone that works with Asterisk and the SPA3000, although I have been told that there are some that do??? The quick answer would be to put a SIP phone with MWI where your wife wants to be able to see the light. I have a Budgtone 200 and MWI works fine on it. Of course then you have styling and color issues that might not past the muster. Doug On Thu, 7 Dec 2006, John Marvin wrote: I would suggest that people who don't already have an investment in home automation equipment should look at Insteon rather than X10. Insteon is a next generation version of X10 that provides backwards compatibility with X10. The devices are a little more expensive, but not as expensive as some of the other alternatives. Insteon provides 2 way communication and is a lot more reliable than X10. If you already have an investment in X10 devices you can slowly convert to Insteon, since Insteon provides backwards compatibility, i.e. X10 controllers can control Insteon devices and Insteon controllers can control X10 devices, however you won't get all the advantages of Insteon until you have Insteon controllers controlling Insteon devices. For people with some soldering and basic circuit design skills, you may want to consider using ethernet as a home automation bus for some things. I love the Olimex PIC WEB and PIC Mini Web development boards (they cost $49.95 and $39.95 respectively). They have an ethernet port and an expansion connector for the available PIC I/O pins. Microchip provides a free C compiler for Pic processors, and they also have an open source networking stack that works on the Olimex boards. So with a ribbon cable connector and a small breadboard with a few IC's and/or driver transistors you can build a device that responds to commands via the network (or via a built in web server) from your Asterisk server that does about any task you can think of. Lots of fun ... I'm currently building a voicemail indicator (my wife didn't like me taking her answering machine away with the blinking lights when we switched to Asterisk voicemail) using a PIC Web board. Next project will be a web based sprinkler controller. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954 * * 215-431-6307* * * * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Plantronics and Snom RF feedback
Benny Amorsen wrote: If auto lift means the mechanical lifter, then you should not use the head set jack on the Snom at all. /Benny I don't follow... Remove the mechanical lifter? Then do what, go from the Plantronic to the headset jack on the Snom, leave the receiver in its normal port? If I do this, the person has to hit the headset button on the Snom... PlantronicRJ11 -- Headset port on Snom -- User HAS TO hit the headset button on the Snom -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fax machine detect (akin to AMD)
Has anyone done any fax machine detection on outbound calls? I've heard of NV's fax detect app but I haven't seen any indications that it supports outbound fax machine detection. -MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue member refresh
I am experiencing this: 1 - A,B,C are SIP users logged on QUEUEA with ringall strategy 2 - I call QUEUEA 3 - A,B,C start ringing 4 - nobody answer 5 - D logs on the QUEUEA 6 - D doen's receive any call, but A,B,C are still ringing How can i avoid that? I'd like that when D joins the QUEUEA it will immediately receive the call that is still ringing on other users... Thanks in advance, nik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing caller id on a zap channel for one sip extension only
Thanks guys for all the help. For this setup I just did a GoToIf(), I will look into multiple context though, looks like thats whats needed for having alot of different outbound caller ids! Thanks again! On 12/6/06, C F [EMAIL PROTECTED] wrote: Asterisk supports whats called context, using a context just for that phone you can set a different callerid, then use a default context for all the other phones. On 12/6/06, Ron McCarthy [EMAIL PROTECTED] wrote: Hi List, Ive got one extension/login that when they call out from that it needs to show a different name/number, and then the rest of the phone will have a default one. Whats the best way to do this? I know it can be done, just cant figure out how! Ive looked around and seem to see no docs on it. Any help or examples would be great on this! Thanks! Ron ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Job Posting, Asterisk Engineer/Sales Engineer, Dallas TX Area
JOB DESCRIPTION: We are looking for an Engineer/Sales Engineer combination. Primary focus will be working with Asterisk/Linux and VoIP. Asterisk systems administrator and experience with Carrier/Service Provider Telecommunication experience. Talented individual with a thorough knowledge of VoIP, SIP, Cisco VoIP, Asterisk and Linux is needed. POSITION RESPONSIBILITIES: The qualified applicant will be an integral part of our engineering staff and will work together in a small team to create, administer, integrate, support, and maintain LINUX/Asterisk/Cisco VoIP systems which support business customers. Time will also be spent assisting Sales Professional with presentation materials, customer network designs and implementation to help close business. Proven experience-troubleshooting hardware, networking and Linux related problems as they arise. They will be required to be flexible, detail oriented and organized with the ability to multitask and work independently and also in a group. The qualified applicant must have the initiative to determine what types of tools or processes can be put in place in order to improve the VoIP products and services. This position is for full-time employment in the Dallas area, office is in Irving, TX. DESIRED SKILLS: Understanding of Asterisk and add-on modules is required Understanding of VoIP termination through SIP and IAX protocols Cisco Call Manager, Call manager Express, Cisco VoIP Experience Experienced VoIP network troubleshooting Willingness to handle urgent after hour support issues if they arise Solid web-related skills Familiarity with component and network monitoring systems MySQL experience is a plus. Comfortable on a bash shell Bash scripting Mid level networking Network security Network troubleshooting Routing and Subnetting Strong analytical and problem solving skills Demonstrated ability to work in a fast-moving dynamic team-oriented environment Documentation and Presentation VoIP Systems: Asterisk and Cisco Call Manager Server Systems: Linux and Widows Server 2000/2003 Desktop Environment: Windows XP, Microsoft Office, Visio Education: EE/BS/BE, relevant experience and certifications considered. Send resume to [EMAIL PROTECTED] -- JR Richardson Engineering for the Masses ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SER/OpenSER + Asterisk + Queue
- Original Message - From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, December 05, 2006 10:56 AM Subject: [asterisk-users] SER/OpenSER + Asterisk + Queue We are in the process of redesigning our single Asterisk server that handles several queues for our clients. We offer our clients hosted queueing/call center basic services. All the agents are in remote locations behind NATs using either softphones or PAP2-like devices. What we would like to accomplish is setup a SER or OpenSER (SER) server(s) in front of our Asterisk box such that all incoming and outgoing calls are handled by SER. The basic idea is to get set up for scaleability and redundancy. The goal is to be able to add additional Asterisk servers to spread our queue loads. Nothing fancy, maybe just separate clients on different boxes (not load balancing queues across multiple Asterisk boxes since that a totally different scope of project). We could then add additional SER boxes to protect our inbound and outbound SIP gateways to our SIP providers (all our calls are SIP-based - e.g. no TDM circuits). Lastly, all our agents would register against the SER server(s) instead of directly to the Asterisk boxes. Has anyone done this? Can anyone point me to some tips/documentation? Does anyone care to comment? If agents login using AgentCallBackLogin, will Asterisk know where the agents are and send the calls to them via SER? Thank you so much in advanced. - Daniel Yes, you can do this. We have our own SIP proxy server. We only use Asterisk as ACD. It works good. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
On Thu, 2006-12-07 at 07:20 -0700, [EMAIL PROTECTED] wrote: Date: Thu, 07 Dec 2006 02:11:59 -0700 From: John Marvin [EMAIL PROTECTED] Subject: Re: [asterisk-users] Is there any Asterisk controllable thermostat? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Doug Crompton wrote: I remembered I had an x10 bottlerocket in my X10 junkbox so I connected it to a spare serial port on my linux server (asterisk resides there) and implemented with some mods the code mentioned earlier http://lorance.freeshell.org/asterisk/#asterisk-can-control-the-world and it works great. Now I have one more way to control X10 devices. I can even call my VM on the way home and turn on my lights or whatever before I get home. I would suggest that people who don't already have an investment in home automation equipment should look at Insteon rather than X10. Insteon is a next generation version of X10 that provides backwards compatibility with X10. The devices are a little more expensive, but not as expensive as some of the other alternatives. Insteon provides 2 way communication and is a lot more reliable than X10. If you already have an investment in X10 devices you can slowly convert to Insteon, since Insteon provides backwards compatibility, i.e. X10 controllers can control Insteon devices and Insteon controllers can control X10 devices, however you won't get all the advantages of Insteon until you have Insteon controllers controlling Insteon devices. For people with some soldering and basic circuit design skills, you may want to consider using ethernet as a home automation bus for some things. I love the Olimex PIC WEB and PIC Mini Web development boards (they cost $49.95 and $39.95 respectively). They have an ethernet port and an expansion connector for the available PIC I/O pins. Microchip provides a free C compiler for Pic processors, and they also have an open source networking stack that works on the Olimex boards. So with a ribbon cable connector and a small breadboard with a few IC's and/or driver transistors you can build a device that responds to commands via the network (or via a built in web server) from your Asterisk server that does about any task you can think of. Lots of fun ... I'm currently building a voicemail indicator (my wife didn't like me taking her answering machine away with the blinking lights when we switched to Asterisk voicemail) using a PIC Web board. Next project will be a web based sprinkler controller. Are any of these home automation systems compatible with homeplug? Or WiFi, or BlueTooth? It seems to me that bundling a proprietary (or less popular) network protocol (and HW) with the device controller fragments the market, and prohibits reuse of the mass market network, which prevents economies of scale for consumers and developers. John -- (C) Matthew Rubenstein ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FXO USB that works with Asterisk?
Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Many thanks, Nathan -- www.nathanpralle.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] eicon diva BRI problems
Armin Schindler wrote: I don't have a changelog. If this problem appears again, please create a memory dump of the cards memory (divactrl can do that). This will help to find the problem, but the latest driver/firmware should be used. Hi Armin! Can you please tell me exactly the proper statement to make this dump? I guess I want have much time to read the docs when problem happens again. thanks klaus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UDP ports
Hello, Does anyone know what is this traffic from Polycom IP300 to asterisk server on RTP port range ? 17:49:35.355673 IP 192.168.2.215.2229 192.168.2.210.19615: UDP, length 72 17:49:42.372713 IP 192.168.2.214.2223 192.168.2.210.16487: UDP, length 72 17:49:44.353414 IP 192.168.2.216.2237 192.168.2.210.19915: UDP, length 72 17:49:45.353914 IP 192.168.2.215.2229 192.168.2.210.19615: UDP, length 72 Where 192.168.2.210 is the asterisk server and 211 to 219 the polycom phones this traffic emanates each 10 seconds from each phone. Thomas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CISCO 2600 - VWIC 1MFT-E1
Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any possibility of Vonage Integration
On Wed, 6 Dec 2006 12:58:32 -0800, Brad Templeton wrote: On Wed, Dec 06, 2006 at 08:13:00AM -0500, Paul wrote: Some companies offer PSTN failover on DIDs, which I think is a good idea. Works at least if your equipment, or their middle equipment is down but doesn't work if the PSTN failover equipment itself is down. Vonage does offer PSTN failover if your ATA is not responding. But having an FXO box talk to your Vonage ATA is just nuts. I wholeheartedly agree! This sort of setup becomes the defining hardware in your Asterisk experience...small FXOs pretty much suck, which is why they've been a recurring topic on this list for litterally years. I dropped Vonage specifically because it doesn't make sense for me to pay $40/mo/each for my home office lines during periods when I'm travelling and not able to use the service. Yes, I know that they eventually offered soft phones in support of travellers, but I'm not THAT much enamored of such things that I wouldn't just reach for my cell phone when out of office. Besides, hotels are so variable in their networking that SIP soft phones probably can't be relied upon. OTOH, using an IAX soft phone has worked just ahout everywhere I've tried it. I even tried Firefly over IAX2 using iLBC on a POTS dialup to Covad in a pinch one day. I simply used the same account info that my server uses to passs calls to ITSPs like VOIPJet, Nufone and Voxee. That was purely experimental, and not something I'd ever do for business. Paying by the minute, even a slightly higher rate, works out cheaper for me. Not theoretically cheaper, actually cheaper. Michael ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting outgoing caller id on a zap channel forone sip extension only
Is this for a pots line ? - Original Message - From: Ron McCarthy To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, December 06, 2006 9:10 PM Subject: [asterisk-users] Setting outgoing caller id on a zap channel forone sip extension only Hi List, Ive got one extension/login that when they call out from that it needs to show a different name/number, and then the rest of the phone will have a default one. Whats the best way to do this? I know it can be done, just cant figure out how! Ive looked around and seem to see no docs on it. Any help or examples would be great on this! Thanks! Ron -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk on a Home rotuer
I need a router for a reason. My client is in the middle east where they have lots of fun with tacking on money ;). A crappy router wont do much. - Original Message - From: Tom Lynn To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, December 07, 2006 5:11 PM Subject: Re: [asterisk-users] Running Asterisk on a Home rotuer It may not be what you're thinking, but I use Astlinux on an older PIII. With a couple of options it has become my home router and works very well. On 12/7/06, Dovid B [EMAIL PROTECTED] wrote: Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] eicon diva BRI problems
On Thu, 7 Dec 2006, Klaus Darilion wrote: Armin Schindler wrote: I don't have a changelog. If this problem appears again, please create a memory dump of the cards memory (divactrl can do that). This will help to find the problem, but the latest driver/firmware should be used. Hi Armin! Can you please tell me exactly the proper statement to make this dump? I guess I want have much time to read the docs when problem happens again. divactrl ctrl -c 1 -File divadump.mem -CoreDump Armin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO USB that works with Asterisk?
On Thu, 2006-12-07 at 10:16 -0600, Nathan E. Pralle wrote: Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Take a look: http://www.xorcom.com/astribank/features.html Many thanks, Nathan -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there any Asterisk controllable thermostat?
The Message Waiting Lamp (neon) on these phones requires a 90v signal which is generated and switched to the phone via a special station card on an analog PBX. This feature was developed mainly for Hotel and Motels but I doubt there are any manufacturers who would develop this functionality for any ATA's as this technology is very old. your best bet is to use the stuttered dial tone or buy (as a previous person has suggested) a cheapo Grandstream (you can re-spay them any colour) Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada John, Two questions on your comments I have no seen an Insteon computer controller similiar to the old bottle rocket. Is there such a device? I am thinking of getting an Insteon starter kit bit I have so many X10 devices it will be awhie before, if ever, that I get it all changed over. Many items, like spotlights, are not available in Insteon. I would be interested in the Ethernet MWI. I am using many phones on an SPA3000 fxs and I can't seem to find an MWI on an analog phone that works with Asterisk and the SPA3000, although I have been told that there are some that do??? The quick answer would be to put a SIP phone with MWI where your wife wants to be able to see the light. I have a Budgtone 200 and MWI works fine on it. Of course then you have styling and color issues that might not past the muster. Doug On Thu, 7 Dec 2006, John Marvin wrote: I would suggest that people who don't already have an investment in home automation equipment should look at Insteon rather than X10. Insteon is a next generation version of X10 that provides backwards compatibility with X10. The devices are a little more expensive, but not as expensive as some of the other alternatives. Insteon provides 2 way communication and is a lot more reliable than X10. If you already have an investment in X10 devices you can slowly convert to Insteon, since Insteon provides backwards compatibility, i.e. X10 controllers can control Insteon devices and Insteon controllers can control X10 devices, however you won't get all the advantages of Insteon until you have Insteon controllers controlling Insteon devices. For people with some soldering and basic circuit design skills, you may want to consider using ethernet as a home automation bus for some things. I love the Olimex PIC WEB and PIC Mini Web development boards (they cost $49.95 and $39.95 respectively). They have an ethernet port and an expansion connector for the available PIC I/O pins. Microchip provides a free C compiler for Pic processors, and they also have an open source networking stack that works on the Olimex boards. So with a ribbon cable connector and a small breadboard with a few IC's and/or driver transistors you can build a device that responds to commands via the network (or via a built in web server) from your Asterisk server that does about any task you can think of. Lots of fun ... I'm currently building a voicemail indicator (my wife didn't like me taking her answering machine away with the blinking lights when we switched to Asterisk voicemail) using a PIC Web board. Next project will be a web based sprinkler controller. John ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Those that sacrifice essential liberty to obtain a little temporary safety deserve neither liberty nor safety. -- Ben Franklin (1759) * Doug Crompton * * Richboro, PA 18954* * 215-431-6307 * ** * [EMAIL PROTECTED]* * http://www.crompton.com * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2600 - VWIC 1MFT-E1
On Thu, 2006-12-07 at 17:51 +0100, FaberK wrote: Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Your config says session transport tcp for SIP. Asterisk does not support SIP over TCP, only SIP over UDP so change that to UDP. Not sure about these: Your config says no dspfarm but you have specified g729br8 codecs. Is g729br8 supported on a Cisco 2600 without the PVDM2 or NM-HDV2 modules that have the dsp's on board to do the g729br8-alaw/ulaw transcoding? Your config says clock source internal. Why don't you use the clock of the telco that provides the E1? That would prevent clock slips as the telco's clock is bound to be more reliable and precise than the internal clock in the Cisco. Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2600 - VWIC 1MFT-E1
voice service voip sip session transport tcp Last I checked, asterisk doesn't support TCP SIP signaling (or RTP over TCP). See what happens if you change it back to the UDP default. On 12/7/06, FaberK [EMAIL PROTECTED] wrote: Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- I am Dyslexic of Borg. Fusistance is retile. Your ass will be laminated! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO USB that works with Asterisk?
Acabo de instalar la asterisk-1.2.11 , saben si se tiene q aplicar algun parche a esta version, tengo el siguiente error en la consola de asterisk cuando establesco llamada a traves del VICIDIAL,. WARNING[21235]: chan_sip.c:2561 *sip_write*: *Asked to transmit frame type 64*, *while* *native formats* is 8 (*read/write* = *64/64*) Alguna sugerencia From:"Guillermo Salas M." [EMAIL PROTECTED]Reply-To:[EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comSubject:Re: [asterisk-users] FXO USB that works with Asterisk?Date:Thu, 07 Dec 2006 12:46:44 -0500On Thu, 2006-12-07 at 10:16 -0600, Nathan E. Pralle wrote: Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Take a look:http://www.xorcom.com/astribank/features.html Many thanks, Nathan --Guillermo Salas M.Telconet S.A.Calle 15 y Avenida 24 EsqEdificio Barre #2 Primer PisoTelefono : +593 5 262 8071Celular: +593 9 985 5138e-mail : [EMAIL PROTECTED]www: http://www.manta.telconet.nethttp://www.telcocarrier.netLinux User: 255902Beat me, whip me, make me use Windows!Please avoid sending me Word or PowerPoint attachments.See http://www.fsf.org/philosophy/no-word-attachments.htmlPlease avoid the Top Posting, seehttp://es.wikipedia.org/wiki/Top-posting___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersÉxitos, grandes clásicos y novedades. Un millón de canciones en MSN Music. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] error en asterisk
Acabo de instalar la asterisk-1.2.11 , saben si se tiene q aplicar algun parche a esta version, tengo el siguiente error en la consola de asterisk cuando establesco llamada a traves del VICIDIAL,. WARNING[21235]: chan_sip.c:2561 *sip_write*: *Asked to transmit frame type 64*, *while* *native formats* is 8 (*read/write* = *64/64*) Alguna sugerencia From:"Guillermo Salas M." [EMAIL PROTECTED]Reply-To:[EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comSubject:Re: [asterisk-users] FXO USB that works with Asterisk?Date:Thu, 07 Dec 2006 12:46:44 -0500On Thu, 2006-12-07 at 10:16 -0600, Nathan E. Pralle wrote: Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Take a look:http://www.xorcom.com/astribank/features.html Many thanks, Nathan --Guillermo Salas M.Telconet S.A.Calle 15 y Avenida 24 EsqEdificio Barre #2 Primer PisoTelefono : +593 5 262 8071Celular: +593 9 985 5138e-mail : [EMAIL PROTECTED]www: http://www.manta.telconet.nethttp://www.telcocarrier.netLinux User: 255902Beat me, whip me, make me use Windows!Please avoid sending me Word or PowerPoint attachments.See http://www.fsf.org/philosophy/no-word-attachments.htmlPlease avoid the Top Posting, seehttp://es.wikipedia.org/wiki/Top-posting___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersÉxitos, grandes clásicos y novedades. Un millón de canciones en MSN Music. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ERROR EN ASTERISK
Acabo de instalar la asterisk-1.2.11 , saben si se tiene q aplicar algun parche a esta version, tengo el siguiente error en la consola de asterisk cuando establesco llamada a traves del VICIDIAL,. WARNING[21235]: chan_sip.c:2561 *sip_write*: *Asked to transmit frame type 64*, *while* *native formats* is 8 (*read/write* = *64/64*) Alguna sugerencia From:Armin Schindler [EMAIL PROTECTED]Reply-To:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo:Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comSubject:Re: [asterisk-users] eicon diva BRI problemsDate:Thu, 7 Dec 2006 18:30:46 +0100 (CET)On Thu, 7 Dec 2006, Klaus Darilion wrote: Armin Schindler wrote: I don't have a changelog. If this problem appears again, please create a memory dump of the cards memory (divactrl can do that). This will help to find the problem, but the latest driver/firmware should be used. Hi Armin! Can you please tell me exactly the proper statement to make this dump? I guess I want have much time to read the docs when problem happens again.divactrl ctrl -c 1 -File divadump.mem -CoreDumpArmin___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-usersOfertas y reservas para viajar por todo el mundo. Organiza y contrata tus viajes aquí. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any possibility of Vonage Integration
It's the tos that probably saves the. - Original Message - From: Paul [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 07, 2006 7:47 AM Subject: Re: [asterisk-users] any possibility of Vonage Integration Time Bandit wrote: $25/month will buy you close to 50 hours of urban SIP termination, it's down to half a cent in some of the big cities. Are you going to average 50 hours on the phone each month? Some people do, but most don't. (Otherwise Vonage could not even pretend it is going to make money.) You don't have a teenager in your home I guess ;) The teenager girl in my home can easily make more than 3000 minutes of call in a month ! The TV ads promote it as unlimited. If there are real cases where residential subscribers did not get unlimited residential service for the advertised price, why aren't any state attorney generals going after vonage? Maybe the answer is that no state is really liberal enough to protect the little guy from the big corporate fraudsters. Anyone here noticed how so many of the little voip companies are imitating vonage with the same false and deceptive advertising? How can we trust them if they can't take higher moral ground than vonage? The smaller company would be likely to invoke the fine print sooner if I had a few teenagers in the household using the unlimited service I bought for $25/month. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2600 - VWIC 1MFT-E1
Hi In dial-peer voice 697617664 voip your must specify into voip dial peer session protocol sipv2 and check if session target sip-server is corect doing a ping to sip-server . I think you must configure it with ipv4:ip_addres or map a host entry with ip host sip-server x.x.x.x in global configuration mode you have forgotten to configure a pots dial peer for your controler. put something like this dial-peer voice 10 pots destination-pattern 0T fax rate disable direct-inward-dial port 1/0:15 and try if you can write authentication username asterisk-uername password XX this last command should allow dial-peer voice 10 to register within asterisk I hope it will help you best regards 2006/12/7, FaberK [EMAIL PROTECTED]: Hi to all, I got a Cisco 2651XM wired to an E1 PRI. What I want to do is to pass all incoming calls to my asterisk. This is my actual conf: http://pastebin.ca/270677 with this I'm able to call my number from outside, but the call stop on the 2600, infact I can hear the tone, but I'm not able to forward calls to my asterisk. Anyone got an idea of my errors? Thanks to all. -- .:FaberK:. ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk accepting calls to fast
Have a look at TIMEOUT(digit) http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout On 12/7/06, Stefan-Michael. Guenther (in-put GbR) [EMAIL PROTECTED] wrote: Hi, the german telco Colt Telekom has assigned the phone number block 56830-xxx to one of our customers. In the diaplan we have setup extensions like the following ones: exten = 56830910,1,Answer() exten = 56830910,2,Dial(SIP/bduerring,10,tr) exten = 56830910,3,VoiceMail,u20 exten = 56830910,4,hangup exten = 56830910,103,VoiceMail,b20 exten = 56830910,104,hangup exten = 5683091,1,Answer() exten = 5683091,2,DIAL(ZAP/g5/56830990,10,r) exten = 5683091,3,Hangup The problem now is, that sometimes (maybe when the caller doesn't hit the buttons fast enough) asterisk takes the extension for 5683091, although the 0 is still coming a little bit later. I'm not quite sure whether the delay in transferring the numbers is caused by the caller or by the telco. But is their a chance to tell asterisk to wait a little bit longer, before it starts searching the extensions.conf? Or do I have to tell the ISDN card to wait for the complete number, before it is forwarded to asterisk? Software hardware: SuSE 10.0 Asterisk 1.2.7.1-BRIstuffed-0.3.0-PRE-1p chan_capi-0.7.0 divas4linux_EICON-106.20-1 Eicon Networks Corporation Diva Server 4BRI Rev 2 Thanks for your help hints, Stefan -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO USB that works with Asterisk?
Does anyone know if Xorcom's Astribank can work within a Xen VM ? Guillermo Salas M. wrote: On Thu, 2006-12-07 at 10:16 -0600, Nathan E. Pralle wrote: Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Take a look: http://www.xorcom.com/astribank/features.html Many thanks, Nathan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] illegal VoIP in India
I ran across this article today: http://economictimes.indiatimes.com/articleshow/726843.cms Anybody know what the implications are for asterisk servers in and out of the country used by people in the country? Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO USB that works with Asterisk?
On Thu, 2006-12-07 at 18:17 +, jose luis peche baldera wrote: Acabo de instalar la asterisk-1.2.11 , saben si se tiene q aplicar algun parche a esta version, tengo el siguiente error en la consola de asterisk cuando establesco llamada a traves del VICIDIAL,. WARNING[21235]: chan_sip.c:2561 *sip_write*: *Asked to transmit frame type 64*, *while* *native formats* is 8 (*read/write* = *64/64*) Alguna sugerencia Please, make a new message for a new question, do not reply a thread with a different topic, and finally, use english. Regards, __ From: Guillermo Salas M. [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] FXO USB that works with Asterisk? Date: Thu, 07 Dec 2006 12:46:44 -0500 On Thu, 2006-12-07 at 10:16 -0600, Nathan E. Pralle wrote: Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Take a look: http://www.xorcom.com/astribank/features.html Many thanks, Nathan -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Éxitos, grandes clásicos y novedades. Un millón de canciones en MSN Music. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 8071 Celular : +593 9 985 5138 e-mail : [EMAIL PROTECTED] www : http://www.manta.telconet.net http://www.telcocarrier.net Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid sending me Word or PowerPoint attachments. See http://www.fsf.org/philosophy/no-word-attachments.html Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk on a Home rotuer
On 7 Dec 2006, at 17:19, Dovid B wrote: I need a router for a reason. My client is in the middle east where they have lots of fun with tacking on money ;). A crappy router wont do much. It isn't a router, but the linksys NSLU2 runs asterisk quite nicely if you cut the config back. If you add a USB disk you can even build asterisk on it :-) Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Codec Selection in asterisk
I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and they all are able to register and make calls with no problem . My voip carrier supports gsm as well as ilbc .. Server takes calls from sip phones , does call recording in between and forwards to voip carrier . My problem is that half of my softphones use ilbc and rest use gsm and my provider supports both gsm as well as ilbc . Now when i put allow=gsmilbc in my voip carrier's extension then it uses gsm ( first preference ) to send calls but half of my softphones use ilbc so asterisk does codec transcoding in between using lot of cpu .. how ever my carrier does support ilbc tooo but when i put allow=ilbcgsm then it uses ilbc again and does codec transcoding from gsm to ilbc for rest of softphones . How can i make asterisk to be smart in choosing codec .. and use ilbc to voip carrier if softphone is using ilbc or use gsm when softphone is using gsm ( but still should do call recording in between ) .. I am using freepbx for most of configuration btw... Any suggestions ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk freezes when DNS not working: a BUG??
You can run dnsmasq on the machine for local caching of the dns names. (http://thekelleys.org.uk/dnsmasq/doc.html) and then apply this patch that will allow dnsmasq to set a minimum time to live (http://lists.thekelleys.org.uk/pipermail/dnsmasq-discuss/2005q2/000253.html). dnsmasq can be then configured to only allow localhost inquires On 12/6/06, Giorgio Incantalupo [EMAIL PROTECTED] wrote: Hi Bob, thanks for reply. The problem is all PBX are not in the same LAN and every customer wants his/her own DNS. I think I'll use /etc/hosts but the problem still remain: Asterisk shouldn't freeze during reloadthe registration should be located in another process but I think that such a change would modify too much Asterisk sip/iax applications and part of Asterisk architecture. So, I know it works that way, I accept it and I try to workaround it. Thanks Giorgio Incantalupo Bob Chiodini wrote: Giorgio, You could set up a caching name server in your local network, use it as your primary DNS server and your ISP's as a secondary. This would cache your ITSP's address(es) locally limiting your reliance on your ISP. Bob... On Wed, 2006-12-06 at 10:43 +0100, Giorgio Incantalupo wrote: Hi, I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers registrations: Asterisk freezes when it cannot (re-)register with VoIP provider (registration timeout). The problem is related to DNS names resolution: if DNS server is very slow to respond Asterisk stops every activity (no zap or restart commands on CLI). The bad news is VoIP providers usually do not give their IP so I cannot use it. Is there anybody who had a problem like this? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO USB that works with Asterisk?
Thursday, December 7, 2006, 7:31:42 PM, Tomer Horn wrote: Does anyone know if Xorcom's Astribank can work within a Xen VM ? Well, I think running asterisk in xen domain is very hardcore :) -- Best regards, Csibra Gergomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MWI across multiple servers
Jon: I will second that motion ... This is something I would be very interested in seeing as I have a similar requirement ... Have a number of folks on my system who work from home ... A number of them have Asterisk servers that register with the main office Asterisk server ... Right now I am handing off calls to each Asterisk server so the VM gets recorded locally and I can make the MWI light blink ... Only reason I did it that wasy is because I was not smart enough to figure out how to centralize VM at the main office and still turn on that darn MWI blinky light ... Are you able in your scenario to store all VM on a central server, but some how get the word back to the remote server that there is a message waiting ??? If so that is col !!! I spent days trying to figure out how to do that and finally just gave up ... PLEASE POST THAT ONE ON THE WIKI If you don't have time to write up a how-to, at least post your scripts with a quick and dirty of what it does ... Maybe make it searchable by remote MWI or something similar ... G.Hendershot ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk stopped Matching Defined Peer
HI All, Something weird has happened to my (*) setup. Setup: I'm using a Realtime-Driven (*) server for voicemail which has the knowledge of all mailbox users on the softswitch which is remote to this (*) box. Since that's all this box is used for, all I have in the sip.conf is the definition of a peer (tried friend as well) which is qualified by its IP address. This is where the calls come to the (*) box from when the call needs to access voicemail. Peer definition in sip.conf Looks something like this [POP] type=peer host=xxx.xxx.xxx.xxx -- I have the actual IP of the originating peer here context=to-voicemail insecure=very disallow=all allow=ulaw dtmfmode=rfc2833 general part of sip.conf itself looks like [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 10.0.3.53; Address to bind to (all addresses on machine) externip = 60.xxx.xxx.xxx localnet = 10.0.3.0/255.255.255.0 nat=route disallow=all allow=ulaw then in extensions.conf I have, the definitions of extensions under 'to-voicemail context. This was working like a champ but all of a sudden has stopped working. I basically just get back a 407 Proxy Authentication message on my softswitch/proxy servers which I would think I shouldn't when I have a defined peer. It was quite happily printing out SIP debug messages which clearly stated Found peer POP, now I don't see that. I didn't change anything so I'm not sure why this is happening. And even if it is, what I can do fix it? Thanks \R ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO USB that works with Asterisk?
Thursday, December 7, 2006, 7:40:09 PM, Guillermo Salas M. wrote: Please, make a new message for a new question, do not reply a thread with a different topic, and finally, use english. Quoting a whole mail with headers and footers is bad too... -- Best regards, Csibra Gergomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom buddies question
I know this is not asterisk specific but we all know this group is used for Polycom issues as well... I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip version: 1.6.6.0036 for all platforms except 11402_001 1.6.6.0042 for 11402_001 Any way around this? The big issue is moving from a key system to this is the ability to use the phone to see who is on the phone so you know if you can transfer a call. Obviously web based interfaces work but that draws your attention from the phone to the computer reducing effectiveness. Buddies half solve this... Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RE: Polycom buddies question
Figures I email this and realized I can hit Menu 1 (Features) 4 (Presence) 2 (Buddy Status) Wow that's a lot of key strokes. Anyway to reduce that to a one button touch? I don't mind doing that but I guess I should think of the users :-) Bill From: Bill Gibbs Sent: Thursday, December 07, 2006 2:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Polycom buddies question I know this is not asterisk specific but we all know this group is used for Polycom issues as well... I have hints working ok on Asterisk. However the Polycom phone will only show the buddies key if there is not a call. This defeats the purpose of using the buddies to see if you can transfer a call to another extension (using the Buddy key to see if they are on the phone). Polycom sip version: 1.6.6.0036 for all platforms except 11402_001 1.6.6.0042 for 11402_001 Any way around this? The big issue is moving from a key system to this is the ability to use the phone to see who is on the phone so you know if you can transfer a call. Obviously web based interfaces work but that draws your attention from the phone to the computer reducing effectiveness. Buddies half solve this... Bill ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] illegal VoIP in India
I ran across this article today: http://economictimes.indiatimes.com/articleshow/726843.cms Anybody know what the implications are for asterisk servers in and out of the country used by people in the country? Ummm Anybody offering VPN IAX services yet? -- Henry J. Cobb http://www.io.com/~hcobb/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2600 - VWIC 1MFT-E1
http://pastebin.ca/270840 This is the newone with some changements. Unfortunately, always the same problem. Fran, if I add the dial-peer voice 10 pots, I receive the advise that the number does not exist. Also, I do not find the way to add authentication username asterisk-uername password XX. The story continues... Thanks F. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO USB that works with Asterisk?
going to depend on how the driver is provided. If its a binary driver, very unlikely that it'll work. If you get the source and can compile it, you can usually hack it into submission. I got my PCI FXO cards working in xen this way. Tomer Horn wrote: Does anyone know if Xorcom's Astribank can work within a Xen VM ? Guillermo Salas M. wrote: On Thu, 2006-12-07 at 10:16 -0600, Nathan E. Pralle wrote: Hi all. Done some research, Googled a lot, but can't find out if there is a USB FXO adapter that works well with Asterisk. If someone knows of one or has used one, I'd be very interested to hear about it. Take a look: http://www.xorcom.com/astribank/features.html Many thanks, Nathan ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jason The place where you made your stand never mattered, only that you were there... and still on your feet ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] how to configure Asterisk to support SIP INFO method?
In sip.conf set dtmfmode=INFO On 12/7/06, CheungJenny [EMAIL PROTECTED] wrote: Hi all, I have a question: how to configure Asterisk to support SIP INFO method? I encountered this problem when I find my UA don't send INFO message to another UA, actually it should. Asterisk was used as a SIP proxy in this scenario (I know that SIP is not a SIP proxy:-)). Then I captured all the packets and found that my UA send REGISTER to Asterisk, and include Allow: ... INFO ... in its SIP message. But in the 200 OK message replied by Asterisk, there is no Allow fields included, or only Allow without INFO in it. So I guess that's why my UA don't send INFO SIP message. I think one way to solve the problem is to make Asterisk support INFO, but I don't know whether it's feasible or how to configure. Could anyone help me? Thanks. Jenny Cheung -- 通过 Windows Live Messenger 表达您自己！ Windows Live Messenger！http://get.live.com/messenger/overview ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO USB that works with Asterisk?
you would be surprised. I run it at home in a xen vm with one of those cheapy FXO cards and have had great luck. CPU usage isn't too outrageous and the zttesst gives me the same results in a VM as I get on hardware. The only trouble I've had is that I need to wakeup the FXO card if I do a reboot of the VM by doing a ztcfg -a. Csibra Gergo wrote: Thursday, December 7, 2006, 7:31:42 PM, Tomer Horn wrote: Does anyone know if Xorcom's Astribank can work within a Xen VM ? Well, I think running asterisk in xen domain is very hardcore :) -- Jason The place where you made your stand never mattered, only that you were there... and still on your feet ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] queue agent Monitor
Hello list.
Does anyone know if and how I can use in my context the following variable
found in the CDR field?
DSTCHANNEL
I am trying to make the answering agent part of the monitor file name, but it
is not working.
exten= 0072,1,Answer
exten= 0072,2,Ringing
exten= 0072,3,Wait(2)
exten= 0072,4,set(MONITORFILENAME=${DST_CHANNEL}${CALLERID}-${TIMESTAMP})
exten= 0072,5,Queue(NOC)
exten= 0072,6,Hangup
include = parkedcalls
#include users.conf
This is what I am getting for a file name.
4072493400-20061207-160632.wav
Caller - timestamp.wav
But I want to see
Agent(1656)-caller-timestamp.wav
Thank you
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
image001.gif
Description: image001.gif
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Re: [asterisk-users] calls not terminating
Hello, In short – Asterisk is not able to recognize that the 'other' person to whom call was made has hung up – hence the channel stays busy. http://kb.digium.com/entry/1/6/ I would try with busydetect and busycount.. Best regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Session Progress Transmission to Phone
Asterisk doesn't seem to be relaying 183, Session Progress SIP messages received from an upstream host back to the phone. Anyone know why? Here's the SIP message that Asterisk receives, and it does nothing with it. It doesn't pass it back to the phone. -- SIP read from xxx.yyy.142.234:5060: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP xxx.yyy.142.232:5060;branch=z9hG4bK675a9b71;rport=5060 From: OneEighty Communications sip:[EMAIL PROTECTED];tag=as520d008c To: sip:[EMAIL PROTECTED];tag=1c282937849 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE Contact: sip:[EMAIL PROTECTED] Record-Route: sip:xxx.yyy.142.234;lr=on;ftag=as520d008c Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Server: Audiocodes-Sip-Gateway-Mediant 2000/v.4.80A.027.002 Content-Type: application/sdp Content-Length: 237 v=0 o=AudiocodesGW 283013199 283012901 IN IP4 216.187.142.190 s=Phone-Call c=IN IP4 216.187.142.190 t=0 0 m=audio 6350 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv Doug. -Original Message- From: Csibra Gergo [mailto:[EMAIL PROTECTED] Sent: Thursday, December 07, 2006 12:17 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FXO USB that works with Asterisk? Thursday, December 7, 2006, 7:40:09 PM, Guillermo Salas M. wrote: Please, make a new message for a new question, do not reply a thread with a different topic, and finally, use english. Quoting a whole mail with headers and footers is bad too... -- Best regards, Csibra Gergomailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Codec Selection in asterisk
you can try this patch, 0004825: [patch][post 1.4] New codec negotiation algorithm http://bugs.digium.com/view.php?id=4825 I'm think, this is one of the most wanted feature, but unfortunately will not be in asterisk 1.4 and we must wait for 1.6 to be officially supported feature :'( PJ Vicky wrote: I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and they all are able to register and make calls with no problem . My voip carrier supports gsm as well as ilbc .. Server takes calls from sip phones , does call recording in between and forwards to voip carrier . My problem is that half of my softphones use ilbc and rest use gsm and my provider supports both gsm as well as ilbc . Now when i put allow=gsmilbc in my voip carrier's extension then it uses gsm ( first preference ) to send calls but half of my softphones use ilbc so asterisk does codec transcoding in between using lot of cpu .. how ever my carrier does support ilbc tooo but when i put allow=ilbcgsm then it uses ilbc again and does codec transcoding from gsm to ilbc for rest of softphones . How can i make asterisk to be smart in choosing codec .. and use ilbc to voip carrier if softphone is using ilbc or use gsm when softphone is using gsm ( but still should do call recording in between ) .. I am using freepbx for most of configuration btw... Any suggestions ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CISCO 2600 - VWIC 1MFT-E1
what about to try mgcp to control gateway? I haven't try this yet, but mgcp is standard signaling protocol supported by asterisk for controling voip gateways, advantage of mgcp is centralized configuration/dialplan/call processing in asterisk. PJ FaberK wrote: http://pastebin.ca/270840 This is the newone with some changements. Unfortunately, always the same problem. Fran, if I add the dial-peer voice 10 pots, I receive the advise that the number does not exist. Also, I do not find the way to add authentication username asterisk-uername password XX. The story continues... Thanks F. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue agent Monitor
The queue application sends the call to an agent. Use the agent
extension's dialplan to set up the monitor, that way you will have the
actual agent extension.
On Thu, 2006-12-07 at 14:18 -0600, Ed Nuñez wrote:
Hello list.
Does anyone know if and how I can use in my context the following
variable found in the CDR field?
DSTCHANNEL
I am trying to make the answering agent part of the monitor file name,
but it is not working.
exten= 0072,1,Answer
exten= 0072,2,Ringing
exten= 0072,3,Wait(2)
exten= 0072,4,set(MONITORFILENAME=
${DST_CHANNEL}${CALLERID}-${TIMESTAMP})
exten= 0072,5,Queue(NOC)
exten= 0072,6,Hangup
include = parkedcalls
#include users.conf
This is what I am getting for a file name.
4072493400-20061207-160632.wav
Caller - timestamp.wav
But I want to see
Agent(1656)-caller-timestamp.wav
Thank you
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
___
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http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Codec Selection in asterisk
I am still on asterisk 1.2 branch svn ( afraid of word beta on server :( ) . I will try out that patch. On 08/12/06, Pavel Jezek [EMAIL PROTECTED] wrote: you can try this patch, 0004825: [patch][post 1.4] New codec negotiation algorithm http://bugs.digium.com/view.php?id=4825 I'm think, this is one of the most wanted feature, but unfortunately will not be in asterisk 1.4 and we must wait for 1.6 to be officially supported feature :'( PJ Vicky wrote: I have around 20-30 softphones behind NAT .. My sip.conf has nat=yes and they all are able to register and make calls with no problem . My voip carrier supports gsm as well as ilbc .. Server takes calls from sip phones , does call recording in between and forwards to voip carrier . My problem is that half of my softphones use ilbc and rest use gsm and my provider supports both gsm as well as ilbc . Now when i put allow=gsmilbc in my voip carrier's extension then it uses gsm ( first preference ) to send calls but half of my softphones use ilbc so asterisk does codec transcoding in between using lot of cpu .. how ever my carrier does support ilbc tooo but when i put allow=ilbcgsm then it uses ilbc again and does codec transcoding from gsm to ilbc for rest of softphones . How can i make asterisk to be smart in choosing codec .. and use ilbc to voip carrier if softphone is using ilbc or use gsm when softphone is using gsm ( but still should do call recording in between ) .. I am using freepbx for most of configuration btw... Any suggestions ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4 + Cisco 7970
Hi All, I recently received my Cisco 7970 and have it up and running with 8.0.4 firmware, with asterisk 1.4. Seems to function pretty great so far, aside from a few issues. Here is what I have noticed so far, anybody have any fixes for these issues? 1. Contrary to the forums and lists I've been reading - the phone indeed does register. The little registration animation stays on the screen for 10 minutes, but eventually goes away. I am able to make and receive phone calls just fine, and asterisk sees the phone as registered. 2. The little phone on the extension icon continually shows a big red X on it - it this because the phone does not think it has registered itself? (because of MWI?) 3. RingList.xml doesn't seem to be fetched from the tftp server, or even requested. Has anyone successfully gotten new ringtones onto the phone? I have tried ringlist.xml and 'RingList.xml as well as the old documentation recommending RINGLIST.DAT. Background images are working fine. 4. MWI doesn't work - This is abundantly documented, but just to re-iterate, I'm receiveing the following on MWI: Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: REGISTER) Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: NOTIFY) Reliably Transmitting (no NAT) to 10.0.1.37:5060: NOTIFY sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.0.1.24:5060;branch=z9hG4bK0f72f41b;rport From: asterisk sip:[EMAIL PROTECTED];tag=as4aee27ee To: sip:[EMAIL PROTECTED]:5060;transport=udp Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 90 Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 1/1 (0/0) --- --- SIP read from 10.0.1.37:49430 --- SIP/2.0 400 Bad Request Via: SIP/2.0/UDP 10.0.1.24:5060;branch=z9hG4bK0f72f41b;rport From: asterisk sip:[EMAIL PROTECTED];tag=as4aee27ee To: sip:[EMAIL PROTECTED]:5060;transport=udp Call-ID: [EMAIL PROTECTED] Date: Fri, 25 Aug 2006 GMT Warning: 399 Bad MWI NOTIFY CSeq: 102 NOTIFY Content-Length: 0 - --- (9 headers 0 lines) --- -- Got SIP response 400 Bad Request back from 10.0.1.37 Thanks, Diwelf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [asterisk-users] Asterisk accepting calls to fast
Hi, Am Donnerstag, 7. Dezember 2006 19:31 schrieb Forrest Beck: Have a look at TIMEOUT(digit) http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+DigitTimeout I don't see how this function could help me. If I change exten = 5683091,1,Answer() exten = 5683091,2,DIAL(ZAP/g5/56830990,10,r) exten = 5683091,3,Hangup to exten = 5683091,2,Set(TIMEOUT(digit)=10) exten = 5683091,2,Answer() exten = 5683091,3,DIAL(ZAP/g5/56830990,10,r) exten = 5683091,4,Hangup() Asterisk has already chosen the wrong extension. The server has to wait for the complete number before it starts to look for any extension.I guess I need some kind of global timeout value. Stefan the german telco Colt Telekom has assigned the phone number block 56830-xxx to one of our customers. In the diaplan we have setup extensions like the following ones: exten = 56830910,1,Answer() exten = 56830910,2,Dial(SIP/bduerring,10,tr) exten = 56830910,3,VoiceMail,u20 exten = 56830910,4,hangup exten = 56830910,103,VoiceMail,b20 exten = 56830910,104,hangup exten = 5683091,1,Answer() exten = 5683091,2,DIAL(ZAP/g5/56830990,10,r) exten = 5683091,3,Hangup The problem now is, that sometimes (maybe when the caller doesn't hit the buttons fast enough) asterisk takes the extension for 5683091, although the 0 is still coming a little bit later. I'm not quite sure whether the delay in transferring the numbers is caused by the caller or by the telco. But is their a chance to tell asterisk to wait a little bit longer, before it starts searching the extensions.conf? Or do I have to tell the ISDN card to wait for the complete number, before it is forwarded to asterisk? -- in-put GbR - Das Linux-Systemhaus Stefan-Michael Guenther Moltkestrasse 49 D-76133 Karlsruhe Tel./Fax : +49 (0)721 / 83044 - 98/93 http://www.in-put.de Schulungen Installationen Beratung Support Voice over IP - Lösungen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help on AgentCallbackLogin()
When I use AgentCallbackLogin() to logout an agent, it always ask for new extension. I can press # to logout. But I'd like the remove this new extension prompt so when agents are trying to logout, they do not have to press #. Does anybody know how to do this? I am using Asterisk 1.2.12.1 Gary___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] illegal VoIP in India
On 12/7/06, Henry J. Cobb [EMAIL PROTECTED] wrote: Anybody offering VPN IAX services yet? I'm not sure, but does this only apply to VoIP service providers? What about self run asterisk servers? Tom ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Plantronics and Snom RF feedback
JO == J Oquendo [EMAIL PROTECTED] writes: JO I don't follow... Remove the mechanical lifter? Then do what, go JO from the Plantronic to the headset jack on the Snom, leave the JO receiver in its normal port? If I do this, the person has to hit JO the headset button on the Snom... JO PlantronicRJ11 -- Headset port on Snom -- User HAS TO hit the JO headset button on the Snom Yes, I'm saying plantronics - handset port (not headset). Then tell the Snom that no headset is connected. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk on a Home rotuer
I have been using the Linksys BEFRS81 Version 2 8 port router for some time now, using IAX to and from other Asterisk boxes, and before that the 4 port version, but discovered that after 18 minutes or so, SIP traffic ( Vonage or Stanaphone through Asterisk ) would hose the router, and all traffic would stop. At least 2 different 4 port devices. The 8 port version has been fine. John Novack Dovid B wrote: Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] queue agent Monitor
I just tried that and it doesn't work. This may be perhaps because the file
name needs to be defined before the call is sent to the queue.
When I saw you answer I thought it would work because it sounded very logical.
:-)
This is the macro I use to send the call to the extension
Just in case I put the line before and after the extension.
[macro-extensions]
exten = s,1,set(MONITOR_FILENAME=${EXTEN}-${CALLERID}-${TIMESTAMP})
exten = s,2,Dial(${ARG1}|30|t,,wW)
exten = s,3,set(MONITOR_FILENAME=${EXTEN}-${CALLERID}-${TIMESTAMP})
exten = s,4,Voicemail(u${ARG2})
exten = s,104,Voicemail(b${ARG2})
Ed Nuñez
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Thursday, December 07, 2006 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue agent Monitor
The queue application sends the call to an agent. Use the agent
extension's dialplan to set up the monitor, that way you will have the
actual agent extension.
On Thu, 2006-12-07 at 14:18 -0600, Ed Nuñez wrote:
Hello list.
Does anyone know if and how I can use in my context the following
variable found in the CDR field?
DSTCHANNEL
I am trying to make the answering agent part of the monitor file name,
but it is not working.
exten= 0072,1,Answer
exten= 0072,2,Ringing
exten= 0072,3,Wait(2)
exten= 0072,4,set(MONITORFILENAME=
${DST_CHANNEL}${CALLERID}-${TIMESTAMP})
exten= 0072,5,Queue(NOC)
exten= 0072,6,Hangup
include = parkedcalls
#include users.conf
This is what I am getting for a file name.
4072493400-20061207-160632.wav
Caller - timestamp.wav
But I want to see
Agent(1656)-caller-timestamp.wav
Thank you
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
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Re: [asterisk-users] Re: Plantronics and Snom RF feedback
For what its worth, We use plantronics headsets on snom360. Plantronics headset gets connected to the headset jack, then you use the headset button to activate it. The feedback is due to grounding issues. The phone MUST be connected to a hub/switch using a STP cable. UTP cables will NOT provide a ground and you will end up with noise on the headset. The external power adaptor also does not provide ground. We went through this with plantronics and with Snom. We finally connected to phone to a PC's unused ethernet port using a STP cable and problem went away. The current problem we have now is that the headset connection audio takes forever to cut through when answering a call. It seems to happen 1 out of every 5 calls or so. Most noticeably after the phone sits idle for a while. If anyone has a suggestion for that problem it would be greatly appreciated. Jason Bachman ASON, Inc. Benny Amorsen wrote: JO == J Oquendo [EMAIL PROTECTED] writes: JO I don't follow... Remove the mechanical lifter? Then do what, go JO from the Plantronic to the headset jack on the Snom, leave the JO receiver in its normal port? If I do this, the person has to hit JO the headset button on the Snom... JO PlantronicRJ11 -- Headset port on Snom -- User HAS TO hit the JO headset button on the Snom Yes, I'm saying plantronics - handset port (not headset). Then tell the Snom that no headset is connected. /Benny ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Audio Convert Module
I know this has been added to SVN but I'm looking for the source for the original module. It used to be at http://redice.krisk.org/ but this page no longer seems to display anything. I'd like to add it to my 1.2.13 stable install. Does anyone have a copy of the original? I used to have this somewhere but have been unable to locate it. Jeremiah ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk on a Home rotuer
Just out of interest are they openwrt compatible? Bails John Novack wrote: I have been using the Linksys BEFRS81 Version 2 8 port router for some time now, using IAX to and from other Asterisk boxes, and before that the 4 port version, but discovered that after 18 minutes or so, SIP traffic ( Vonage or Stanaphone through Asterisk ) would hose the router, and all traffic would stop. At least 2 different 4 port devices. The 8 port version has been fine. John Novack Dovid B wrote: Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: What's up with the Manager Interface?!?!
Doug, Everyone: I'll make you an offer you (hopefully) can't refuse: I've been fixing manager bugs here and there, and am willing to take on any manager issues out there, for 1.4, and trunk, especially, so as to have things nice and solid for 1.4 before it gets out of beta. So, give me some details. I will file the bug, if you don't. I will reproduce(if I can), and debug, and fix 'em. Just tell me (as explicitly as possible, please!) what the problems are-- especially you, Doug-- where are those inconsistencies, exactly? Richard-- I'll lab up 1.4 and see if I can get the hiccups you mention. murf Douglas Garstang wrote: The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- And, Richard Lyman wrote: just wait till you get a 'hiccup' that causes a line to get cut off, drop a char, and continue on next line. G (examples below) this is an older manager.c there have been a lot of mods to the manager interface in the 1.4 tree, but there is no way i would put that into a production envir. - Event: OriginateFailure Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reason: 3 Uniqueid: null CallerID: xx CallerIDName: ~308C D13-47426-true~ - Event: OriginateFailure Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reason: 5 (rest was gone) - Event: OriginateFailure Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reason: 0 Uniqueid: null CallerID: xx Ca lerIDName: ~308CLD14-40566-true~ - Event: OriginateSuccess Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reaso : 4 Uniqueid: 1163128185.2006 CallerID: xx CallerIDName: ~308CLD13-50454-true~ -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] queue agent Monitor
We had to deal with something similar to that a long time a go. I don't
know if things have changed since, but here is what we did. We had to set
the filename before sending the call to the queue. Obviously, this won't
tell you the DSTCHANNEL because the queue hasn't distributed the call.
exten =
s,n,Set(MONITOR_FILENAME=${TIMESTAMP}_QUEUE_${CALLERIDNUM}_${UNIQUEID})
exten = s,n,Queue(queue_name)
As you can see, we store the UNIQUEID as part of the file name. Then we
had a process that monitored the call recordings and once the call had
ended, it would actually lookup the UNIQUEID in the CDR table and then
rename the file accordingly.
We did this for a client of ours a long time ago and I don't have the
source of that perl script to share, but it should be fairly trivial.
Hope this helps,
Daniel
-Original Message-
From: Ed Nuñez [EMAIL PROTECTED]
Sent: Thu, December 7, 2006 5:34 pm
To: [EMAIL PROTECTED], Asterisk Users Mailing List - Non-Commercial
Discussion asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] queue agent Monitor
I just tried that and it doesn't work. This may be perhaps because the
file name needs to be defined before the call is sent to the queue.
When I saw you answer I thought it would work because it sounded very
logical. :-)
This is the macro I use to send the call to the extension
Just in case I put the line before and after the extension.
[macro-extensions]
exten = s,1,set(MONITOR_FILENAME=${EXTEN}-${CALLERID}-${TIMESTAMP})
exten = s,2,Dial(${ARG1}|30|t,,wW)
exten = s,3,set(MONITOR_FILENAME=${EXTEN}-${CALLERID}-${TIMESTAMP})
exten = s,4,Voicemail(u${ARG2})
exten = s,104,Voicemail(b${ARG2})
Ed Nuñez
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Thursday, December 07, 2006 3:48 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue agent Monitor
The queue application sends the call to an agent. Use the agent
extension's dialplan to set up the monitor, that way you will have the
actual agent extension.
On Thu, 2006-12-07 at 14:18 -0600, Ed Nuñez wrote:
Hello list.
Does anyone know if and how I can use in my context the following
variable found in the CDR field?
DSTCHANNEL
I am trying to make the answering agent part of the monitor file name,
but it is not working.
exten= 0072,1,Answer
exten= 0072,2,Ringing
exten= 0072,3,Wait(2)
exten= 0072,4,set(MONITORFILENAME=
${DST_CHANNEL}${CALLERID}-${TIMESTAMP})
exten= 0072,5,Queue(NOC)
exten= 0072,6,Hangup
include = parkedcalls
#include users.conf
This is what I am getting for a file name.
4072493400-20061207-160632.wav
Caller - timestamp.wav
But I want to see
Agent(1656)-caller-timestamp.wav
Thank you
Ed Nuñez
IT/Telecom Engineer
4037 Metric Drive
Winter Park, FL
(o) 407-384-4200 x 1656
(f) 407-384-4222
(c) 732-925-0730
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Re: [asterisk-users] Running Asterisk on a Home rotuer
tacking pn = adding on - sorry for not being more specific. I have seen that people in the past have used a linksys router to run asterisk. It would be to expensive to bring in a PC for every location. So we want to import cheap home routers put asterisk on them as use them as the go in between the IP phones and the asterisk server. - Original Message - From: Howard Lowndes [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 07, 2006 8:10 PM Subject: Re: [asterisk-users] Running Asterisk on a Home rotuer So, what is your reason? Are you looking for a brand name router to run Linux on, or are you wanting to build a Linux box to run Asterisk. What are you meaning by router? Also: tacking on == adding on ? tacking on == choking on? Please don't use slang, it can get mis-interpreted depending open culture. Dovid B wrote: I need a router for a reason. My client is in the middle east where they have lots of fun with tacking on money ;). A crappy router wont do much. - Original Message - *From:* Tom Lynn mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Thursday, December 07, 2006 5:11 PM *Subject:* Re: [asterisk-users] Running Asterisk on a Home rotuer It may not be what you're thinking, but I use Astlinux on an older PIII. With a couple of options it has become my home router and works very well. On 12/7/06, *Dovid B* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Howard. LANNet Computing Associates - Your Linux people http://lannetlinux.com When you want a computer system that works, just choose Linux; When you want a computer system that works, just, choose Microsoft. -- Flatter government, not fatter government; abolish the Australian states. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.0-beta3 spandsp rxfax woes (or me being hard of thinking)
Hi, I have installed the latest version of asterisk(1.4.0-beta3), and built app_rxfax/txfax. I'm using spandsp from here, http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20061207.tar.gz Everything builds ok. I had to manually apply the patch from the site so configure would spot spandsp libraries. However, when I try dialing my virtual fax extension (either from a phone or fax machine) Asterisk bombs out with the following message... Executing [EMAIL PROTECTED]:1] RxFAX(SIP/101-081d63d0, /tmp/test.tif) in new stack asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: span_set_message_handler This was me dialing from a normal sip extension, hoping to hear fax tones. I did try the latest ebuild Gentoo have (1.2.13), this gave me perfect fax tones, but completely refused to include chan_zap, so I can't win :-) Please somebody tell me where I'm going wrong, been trying to get this to work for hours. I've got rid of all the old libraries, recompiled... my next step is to sacrifice a goat! Any help greatly appreciated. Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Running Asterisk on a Home rotuer
On 12/7/06, Dovid B [EMAIL PROTECTED] wrote: Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid Sure. I have 5 units out there on Linksys WRT54GS v1.1 through v4 units. The software is OpenWRT.org. Asterisk is simply an available package to load once you have replace the original firmware with OpenWRT. There are several models that can run the software. Check the HW compat list on the site. They go right down to revision numbers identified by serial # patterns. Be careful of the amount of RAM they have. You will be storing voicemail in RAM unless you put it off-device like an NFS mount, etc. (Some mfg/models have USB2 ports and you can put a USB stick on them and basically forget about the problem). -- David Cook (Canada) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4.0-beta3 spandsp rxfax woes (or me being hard of thinking)
Same thing occuring here, on gentoo as well :( On 07/12/06, Chris Glover [EMAIL PROTECTED] wrote: Hi, I have installed the latest version of asterisk(1.4.0-beta3), and built app_rxfax/txfax. I'm using spandsp from here, http://www.soft-switch.org/downloads/snapshots/spandsp/spandsp-20061207.tar.gz Everything builds ok. I had to manually apply the patch from the site so configure would spot spandsp libraries. However, when I try dialing my virtual fax extension (either from a phone or fax machine) Asterisk bombs out with the following message... Executing [EMAIL PROTECTED]:1] RxFAX(SIP/101-081d63d0, /tmp/test.tif) in new stack asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_rxfax.so: undefined symbol: span_set_message_handler This was me dialing from a normal sip extension, hoping to hear fax tones. I did try the latest ebuild Gentoo have (1.2.13), this gave me perfect fax tones, but completely refused to include chan_zap, so I can't win :-) Please somebody tell me where I'm going wrong, been trying to get this to work for hours. I've got rid of all the old libraries, recompiled... my next step is to sacrifice a goat! Any help greatly appreciated. Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?
On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote: On 12/6/06, John Novack [EMAIL PROTECTED] wrote: Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run into some gotcha down the road where there is some missing file that needs to be put who knows where. Wow! Are you sure about that? Doesn't seem like an issue to me. yum install foo is easy, and I've always preferred servers that are as lean as possible, rather than all porky with unnecessary packages and services. Someone else mentioned AstLinux, and it is very nice. About 40 megabytes. No lard at all. -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power users ~ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Running Asterisk on a Home rotuer
Not listed as one that is. JN bails wrote: Just out of interest are they openwrt compatible? Bails John Novack wrote: I have been using the Linksys BEFRS81 Version 2 8 port router for some time now, using IAX to and from other Asterisk boxes, and before that the 4 port version, but discovered that after 18 minutes or so, SIP traffic ( Vonage or Stanaphone through Asterisk ) would hose the router, and all traffic would stop. At least 2 different 4 port devices. The 8 port version has been fine. John Novack Dovid B wrote: Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?
Carla Schroder wrote: On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote: On 12/6/06, John Novack [EMAIL PROTECTED] wrote: Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run into some gotcha down the road where there is some missing file that needs to be put who knows where. Wow! Are you sure about that? Doesn't seem like an issue to me. yum install foo is easy, and I've always preferred servers that are as lean as possible, rather than all porky with unnecessary packages and services. Someone else mentioned AstLinux, and it is very nice. About 40 megabytes. No lard at all. That may be true for you and those that know Linux and how to respond to a missing file because it wasn't initially installed. For those who don't practice Linux as a religion but simply want to use a telephony application, it works to install everything, and move on to learning Asterisk and all IT'S warts and gotchas, John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: What's up with the Manager Interface?!?!
Hi Steve. Thanks, but unfortunately, I can't be involved in that. We are running Asterisk in a production environment and we're using 1.2, not 1.4. I don't have the resources to work with 1.4. Last time I filed a bug against 1.2 I got told off. Here's an example of that cruddy output. hestia*CLI dundi show peer 00:0e:0c:a1:92:4d Peer:00:0e:0c:a1:92:4d Model: Symmetric Host:xxx.187.142.203 Dynamic: no KeyPend: no Reg: No In Key: dundikey Out Key: dundikey Include logic: -- include all Query logic: -- permit all hestia*CLI The delimiter should not be the colon, as the data may also contain a colon (in this case the MAC address). That makes it really difficult to split the data into fields. Also, the apparent key:value rule gets broken when you get down to the Include Logic line. The '--include all' should be on the same line! Just about every single Asterisk command has screwed up output like this. Fixing all this would be a LOT of work. Doug. -Original Message- From: Steve Murphy [mailto:[EMAIL PROTECTED] Sent: Thursday, December 07, 2006 4:26 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: What's up with the Manager Interface?!?! Doug, Everyone: I'll make you an offer you (hopefully) can't refuse: I've been fixing manager bugs here and there, and am willing to take on any manager issues out there, for 1.4, and trunk, especially, so as to have things nice and solid for 1.4 before it gets out of beta. So, give me some details. I will file the bug, if you don't. I will reproduce(if I can), and debug, and fix 'em. Just tell me (as explicitly as possible, please!) what the problems are-- especially you, Doug-- where are those inconsistencies, exactly? Richard-- I'll lab up 1.4 and see if I can get the hiccups you mention. murf Douglas Garstang wrote: The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- And, Richard Lyman wrote: just wait till you get a 'hiccup' that causes a line to get cut off, drop a char, and continue on next line. G (examples below) this is an older manager.c there have been a lot of mods to the manager interface in the 1.4 tree, but there is no way i would put that into a production envir. - Event: OriginateFailure Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reason: 3 Uniqueid: null CallerID: xx CallerIDName: ~308C D13-47426-true~ - Event: OriginateFailure Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reason: 5 (rest was gone) - Event: OriginateFailure Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reason: 0 Uniqueid: null CallerID: xx Ca lerIDName: ~308CLD14-40566-true~ - Event: OriginateSuccess Privilege: call,all Channel: Zap/g1/xx Context: gdincoming Exten: Reaso : 4 Uniqueid: 1163128185.2006 CallerID: xx CallerIDName: ~308CLD13-50454-true~ -- Steve Murphy Software Developer Digium ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switching from FreeBSD to Linux - which distro?
On Thursday 07 December 2006 17:42, John Novack wrote: Carla Schroder wrote: On Wednesday 06 December 2006 20:12, Lacy Moore - Aspendora wrote: On 12/6/06, John Novack [EMAIL PROTECTED] wrote: Go get the ISO's, and remember to INSTALL EVERYTHING, then you won't run into some gotcha down the road where there is some missing file that needs to be put who knows where. Wow! Are you sure about that? Doesn't seem like an issue to me. yum install foo is easy, and I've always preferred servers that are as lean as possible, rather than all porky with unnecessary packages and services. Someone else mentioned AstLinux, and it is very nice. About 40 megabytes. No lard at all. That may be true for you and those that know Linux and how to respond to a missing file because it wasn't initially installed. For those who don't practice Linux as a religion but simply want to use a telephony application, it works to install everything, and move on to learning Asterisk and all IT'S warts and gotchas, You're saying it's a religion to understand how to administer the operating system on a server? What a novel concept. -- ~ Carla Schroder Linux geek and random computer tamer check out my Linux Cookbook! http://www.oreilly.com/catalog/linuxckbk/ best book for sysadmins and power users ~ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] wierd callerid problem
I have a site running asterisk 1.2.8 with a hand full of polycoms and grandstream 2Kxp's. When a call is missed and you look at the missed call logs on either, its has the persons exten, not the incoming caller id. Any ideas? \\\|///\\ ~ ~ // ( @ @ )--oOOo-(_)-oOOo—___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AMI - Originate Action and Busy, NoAnswer calls - CDR
Gang, I'm wondering if anyone has run into this problem and found a solution. When I use the manager interface to generate a call, I don't get very much information in my CDR records when the dial status is BUSY, FAILED, NOANSWER, etc. I am putting the dialed number into the CDR Userfield in my dialplan, but the field doesn't populate the CDR record unless the Originate action is successful and the dialed party answers the call. I need to postprocess the CDR records and I absolutely have to have the phone number in the CDR. Ideally I'd like to populate the CDR Userfield with several pieces of information, which I am able to do only if the Dial() or Originate operation results in a connect. I've tried numerous variations of context/extension wrangling to no avail. I can supply examples of what didn't work but I'm really interested in hearing about examples that do work. Has anyone found a workaround or a best practice that allows CDR records to contain the dialed phone number for every Dial() or Originate that Asterisk processes? Thanks, MC ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Basic question regarding re-INVITE
All, This basic question might have been asked thousands of timesbut anyways: when can Asterisk send out an re-INVITE to the line/trunk side? It seems that the canreinvite does NOT matter for calls toward the trunk. E.g. When I put a phone on hold, the re-INVITE is sent from phone to the Asterisk, but then that's it. The Asterisk never sends it out. It seems to work for extension to extention, but not extension to line. What am I missing? Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: AMI - Originate Action and Busy, NoAnswer calls - CDR
Michael Collins wrote:
Has anyone found a workaround or a best practice that allows CDR records
to contain the dialed phone number for every Dial() or Originate that
Asterisk processes?
I got around this by generating a call to a Local channel which is
always (well...nearly always) successful. The Local channel then issues
the Dial command and the dialplan captures the ${DIALSTATUS} via AGI.
Messy but works.
___
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[asterisk-users] ASTERISK y AGC
Buenas noches Tengo montado el sistema de marcador predictivo ASTGUICLIENT en 2 Servidores, en uno de ellos que es un Server HP Proliant G3 350 3GB RAM con Slackware 11 y Asterisk 1.2.12.1 , dicho server solo se encarga de la voz. Luego otro server un poco mas modesto (HP ML110 3.2GB), que tiene Apache y la BD MySQL con el ASTGUICLIENT 2.0.1, mis agentes se conectan a este ultimo ya q el sistema corree alli, los softphone Eyebeam de mis agentes estan conectados al server VOIP, Tengo ademas en el server voip una tarjeta digium de 2 puertos instalados 2 lineas ISDN, En el ASTGUICLIENT tengo varias campañas ya trabajando, las cuales sacan llamadas a traves de las lineas ISDN, solo una de estas campañas Saca las llamadas a traves de una troncal IAX o SIP que tengo con mi filial de otro pais o mi proveedor de voz. Tengo un pico maximo de 25 agentes conectados, el problema esta en que llega un momento en que el marcador ya no pasa llamadas a los agentes, mejor dicho se demora demasiado en marcar y pasar la llamada.,Si uno marca desde un softphone (eyebeam) en la pantallla de mismo obtengo el mensaje TRYNG ..Y despues de varios segundos logra sacar la llamada, en el caso de mi agentes q trabajan con el marcador, esperan por un largo tiempo q le pasen las llamadas. A que se debe este problema, la carga de mis servidores no es mucha no pasar de 5% en su load average? He actualizado la version del asterisk que tenia y del SO pensando q lo podria solucionar pero aun con las nuevas versiones el problema se presenta? Sera q aun con 2 servidores compartiendo la carga no pueda meter mas de 20 agentes al sistema? Anteriormente tuve 20 agentes todo en un servidor, no tenia este problema pero el sistema estaba muy lento y hacia muy dificil el trabajo por lo cual decidi dividirle la carga al servidor quitandole la BD y el WEB para pasarle a otro servidor. Alguno de ustedes ha tenido este problema, de q forma lo han solucionado Gracias de antemano por sus respuestas Saludos Aldo Leyva ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] nordx designator labels ?
its a bit off topic for asterisk but not for a bunch of telcom guys : does anyone have a word or other wordprocessor or spreadsheet template they use (and are willing to share) to create labels for nordx IBDN punchdown designator strip labels ? (something with a box line you can cut on to the right size, and spots to label each cable or pair as the need may be) Thanks in advance if you share it. Jon Pounder _/_/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/ _/ _/_/_/ _/ _/_/ _/_/ _/_/ _/ _/_/ _/_/ _/ _/_/_/ _/_/ _/_/_/_/ _/_/_/ _/_/ _/_/_/_/ Inline Internet Systems Inc. Thorold, Ontario, Canada Tools to Power Your e-Business Solutions www.inline.net www.ihtml.com www.ihtmlmerchant.com www.opayc.com This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
