RE: [Asterisk-Users] Record() problem
Kobus We have a similar requirement and our solution has been to minimise our dependence upon hardware wherever possible. We have two PSTN lines that are serviced by * and a pair of X101P cards but these are for emergency use, for example if our broadband connection were to be unavailable. For our main telephony needs we use www.voiptalk.com as a VoIP provider that supports IAX (by making sure we use IAX we know there will be no SIP/RPT/NAT issues). We get the benefit of fxo technology without having to buy and support it. We can make up to 25 outgoing calls and receive up to 5 calls simultaneously. Clearly not sustainable if the company were to grow significantly but effective and great value for money now and the immediate future. Out other investment has been in HandyTone ATAs into which we plug DECT phones so that they be used anywhere within the office. So far it has proved remarkably successfully. By the way, we started using Libretel and FWD but the FWD element has proved too unreliable. To be fair, they say that their IAX service is experimental. Moving to a commercial IAX provider has made a big difference. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kobus Wolvaardt Sent: December 24, 2004 4:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Record() problem Hi all, I want to install 4 to 6 lines for our new office and hook it into an * system. I want as little trouble as possible, what fxo hardware do you recommend? I see that poeple on the list are complaining about digium tdm400 cards...? Are grandstream phones stable and easy to setup? Any problems with *? Thanks, Kobus Wolvaardt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] where I can find some learning book about asterisk?
Hello, Take a look at http://www.signate.com You can also find various documentation resources at http://www.voip-info.org/tiki-index.php?page=Asterisk Regards Lamine - Original Message - From: FCG ZHAO Zigang [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Friday, December 24, 2004 2:05 AM Subject: [Asterisk-Users] where I can find some learning book about asterisk? Hello , I learn handbook-draft.but I think I don't understand asterisk. where I can find some learning book about asterisk? thank u. B.R. John. -- : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] : 20041224 7:51 : asterisk-users@lists.digium.com : Asterisk-Users Digest, Vol 5, Issue 350 Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than Re: Contents of Asterisk-Users digest... Today's Topics: 1. RE: rtp channels not through asterisk (Brian West) 2. Turning * Hangup off in queues ([EMAIL PROTECTED]) 3. Re: Voicemail email notification (Rich Adamson) 4. Can't Make Outgoing Call (Norman Zhang) 5. Re: Voicemail email notification (Dorn Hetzel) 6. Re: Asterisk in parallel with PSTN [OT] (Rich Adamson) 7. Re: rtp channels not through asterisk (Rich Adamson) 8. Re: Realtime sipbuddies table structure why? (Greg - Cirelle Enterprises) 9. RE: Polycom Buddies (Paul Hales) 10. Re: Queue - roundrobin member order (Adam Goryachev) 11. Re: Voicemail email notification (Rich Adamson) 12. Re: Can't Make Outgoing Call (Norman Zhang) 13. Re: Recommended IAX softphone. (Bruno Hertz) 14. Re: sip seeding vs registration (Greg - Cirelle Enterprises) 15. Asterisk 1.0.3 no RedHat zaptel script? (Jerry Geis) 16. Re: Recommended IAX softphone. (Erik Espinoza) -- Message: 1 Date: Thu, 23 Dec 2004 16:51:22 -0600 From: Brian West [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] rtp channels not through asterisk To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=US-ASCII canreinvite=yes Aterisk stays in the signaling path so unless you're running tcpdump or the like you'll never notice this. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of bijan Sent: Thursday, December 23, 2004 4:46 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] rtp channels not through asterisk In wiki pages it is stated that The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. Currently with my settings, I notice that all rtp's are passing through my asterisk. How could I achieve that they go directly from phone to phone? I assume this way, my machine will have less load and therefore could handle more calls. regards Bijan Karimi -- Message: 2 Date: Thu, 23 Dec 2004 19:16:19 -0600 (CST) From: [EMAIL PROTECTED] Subject: [Asterisk-Users] Turning * Hangup off in queues To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: TEXT/PLAIN; charset=US-ASCII Hi ! Can somebody tell me how to turn the * Hangup option utrned off in queues. I have not used any H option but still as an agent if I press * key the user gets disconnected. Somehow it is turned on by default. Can I turn this option off In my extensions.conf I have written : exten = 8000,3,Queue(supportq|t) plz help me inthis regard ... Thanks ! Usman. -- Message: 3 Date: Thu, 23 Dec 2004 16:51:34 -0600 From: Rich Adamson [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Voicemail email notification 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 Are there any common silent failure modes for email notification from the Voicemail module. I put the email and pager email addresses in my entry in voicemail.conf but no mail gets sent when I leave a voicemail. No obvious error messages either, unless I'm just not looking in the right place. Thanks for any clues :) Nop, that's it other then you have to have sendmail configured and running on the system (or have a substitute mail handler). Rich -- Message: 4 Date: Thu, 23 Dec 2004 14:58:04 -0800 From: Norman Zhang [EMAIL PROTECTED] Subject: [Asterisk-Users] Can't Make Outgoing Call To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users
[Asterisk-Users] asterisk at large
Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans kindly guides me. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] polycom and cdp
Ruchard, Richard wrote: I have a catalyst 3500xl and can't get it work. On the switch, I use int f0/xx switch access vlan 100 switch voice vlan 200 Do both vlan's exist (do they show up in your 'show vlan' list)? I've got it working flawlessly with both 7940's and 7960's on 3550's over here... Regards, Dirk-Jan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk at large
Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans Yes you can. Register your remote servers to your main server and choose different numbers for different Asterisk servers. Detailed informations are available at http://www.voip-info.org/wiki-Asterisk+-+dual+servers Regards Lamine kindly guides me. Thanks In Advance. Adnan Ahmed. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] * behaviour in agentcallbacklogin
when an agent logs in using AgentCallbackLogin(), during a call when agent presses * call is hanged up. how can I get rid of this behaviour. that nothing should happen by pressing *. thank you ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help on Register message with Proxy-Authorization
Can any one help me in understanding REGISTER message when i send REGISTER message to asterisk it is replying 407 with header Proxy-Authenticate: Digest realm=asterisk,nonce=1011592446 i want password for my user so i entered secret in sip.conf against userid can any one tell me how to handle Proxy-Authenticate and Proxy-Authorization i want to know where to enter the password in my second reply with Proxy-Authorization header and how to encrypt it __ Do you Yahoo!? Send a seasonal email greeting and help others. Do good. http://celebrity.mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P frame slips
On Thu, 2004-12-23 at 22:48 -0500, Andrew Kohlsmith wrote: On December 23, 2004 10:37 pm, James Sizemore wrote: Try commenting out ;echocancel=yes ;echotraining=yes I bet your faxs start working in both directions. But of course you will now have occasional echo problems. echocancel=no It's always disabled by * when it hears the fax tones anyway. I read somewhere that to be able to hear the fax tones you need to give Asterisk 1 or 2 seconds to be able to pick them up. So put a Wait(1) or Wait(2) in your dialplan (directly after Answer would make sense to me) so Asterisk can figure out it's a fax call and throw it to the fax extension. Merry Xmas! Regards, Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Switch polarity to disconnect a FXS channel
Hi friends, I´m trying to integrate Asterisk with another PBX (Nortel Meridian) I need to switch the polarity of a FXS port (ZAP channel) to inform the other PBX that the channel was released. Does anyone knows the way I can do it? Many thanks in advance. Merry Christmas to the * world ! ! !. Luis ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T100P frame slips
On December 24, 2004 08:48 am, Patrick wrote: I read somewhere that to be able to hear the fax tones you need to give Asterisk 1 or 2 seconds to be able to pick them up. So put a Wait(1) or Wait(2) in your dialplan (directly after Answer would make sense to me) so Asterisk can figure out it's a fax call and throw it to the fax extension. While this is true, it doesn't apply to my particular case -- I have a DID specific to faxes which is thrown to my faxsterisk box over IAX. Basically PRI - colo* dedicated IAX2 link faxsterisk - TDM430P - faxmachines faxmachines - world = good faxes world - faxmachines = 50+% failure rate world - rx_fax on faxsterisk = good faxes -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco, Codecs, Sip Phones et al
I am loving Asterisk! I have a Cisco 7960 (Sip) on which I want to try using g729 encoding. I cannot find a setting for this in the phone's interactive screen menu. Do I set it in the sip.conf file? I have also ordered 2 licenses from Digium. My understanding is that because this Cisco phone can handle the encoding, * just passes it thru. Is this correct? Also, I am using LiveVoip for my call termination via IAX (very happy with them, very unhappy with Gafachi). I cannot find any information from LiveVoip that indicates whether they accept G729. Is it likely or is it just dependant upon the provider? My interest is to improve voice quality over DSL and/or Cable Modem connections. I have QoS working (Sveasoft), and it has improved the situation, but the words are still bracketed with distortion. I am hoping the smaller Codec will mitigate this to an acceptable level. George Burt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE410P X100P Troubles
Do you get error messages when you do a ztcfg - after loading the modules? Are these two cards sharing IRQ's with any other cards/devices in your system? It sounds like it could be a resource conflict. Lyle - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 23, 2004 10:48 PM Subject: Re: [Asterisk-Users] TE410P X100P Troubles im doing modprobe wct4xxp and then modprobe wcfxo -jon - Original Message - From: Lyle Giese [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 23, 2004 7:59 PM Subject: Re: [Asterisk-Users] TE410P X100P Troubles What modules are you loading and in what order? Lyle - Original Message - From: [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 23, 2004 3:25 PM Subject: [Asterisk-Users] TE410P X100P Troubles All, I've got an asterisk box thats been running a TE410P without any problems. I recently added an X100P for our back office line, and now asterisk wont start. Any help is greatly appreciated. Zaptel.conf span=1,1,0,esf,b8zs span=2,2,0,esf,b8zs span=3,3,0,esf,b8zs span=4,4,0,esf,b8zs bchan=1-23 dchan=24 bchan=25-47 dchan=48 bchan=49-71 dchan=72 bchan=73-95 dchan=96 fxsks=97#This is for the X100P Zapata.conf [channels] context=inbound switchtype=national signalling=pri_cpe echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 immediate=no group=1 channel = 1-23 channel = 25-47 channel = 49-71 channel = 73-95 group=2 signalling=fxsks channel=97 context=inbound When I run asterisk -vvvgc I get the following output: Dec 23 16:51:08 ERROR[1420]: chan_zap.c:9422 setup_zap: Unknown signalling method 'fxsks' Dec 23 16:51:08 WARNING[1420]: chan_zap.c:765 zt_open: Unable to specify channel 97: No such device or address Dec 23 16:51:08 ERROR[1420]: chan_zap.c:6197 mkintf: Unable to open channel 97: No such device or address here = 0, tmp-channel = 97, channel = 97 Dec 23 16:51:08 ERROR[1420]: chan_zap.c:9141 setup_zap: Unable to register channel '97' Dec 23 16:51:08 WARNING[1420]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' -- Unregistered channel 1 -- Unregistered channel 2 -- Unregistered channel 3 -- Unregistered channel 4 -- Unregistered channel 5 -- Unregistered channel 6 -- Unregistered channel 7 -- Unregistered channel 8 -- Unregistered channel 9 -- Unregistered channel 10 -- Unregistered channel 11 -- Unregistered channel 12 -- Unregistered channel 13 -- Unregistered channel 14 -- Unregistered channel 15 -- Unregistered channel 16 -- Unregistered channel 17 -- Unregistered channel 18 -- Unregistered channel 19 -- Unregistered channel 20 -- Unregistered channel 21 -- Unregistered channel 22 -- Unregistered channel 23 -- Unregistered channel 24 -- Unregistered channel 25 -- Unregistered channel 26 -- Unregistered channel 27 -- Unregistered channel 28 -- Unregistered channel 29 -- Unregistered channel 30 -- Unregistered channel 31 -- Unregistered channel 32 -- Unregistered channel 33 -- Unregistered channel 34 -- Unregistered channel 35 -- Unregistered channel 36 -- Unregistered channel 37 -- Unregistered channel 38 -- Unregistered channel 39 -- Unregistered channel 40 -- Unregistered channel 41 -- Unregistered channel 42 -- Unregistered channel 43 -- Unregistered channel 44 -- Unregistered channel 45 -- Unregistered channel 46 -- Unregistered channel 47 -- Unregistered channel 48 -- Unregistered channel 49 -- Unregistered channel 50 -- Unregistered channel 51 -- Unregistered channel 52 -- Unregistered channel 53 -- Unregistered channel 54 -- Unregistered channel 55 -- Unregistered channel 56 -- Unregistered channel 57 -- Unregistered channel 58 -- Unregistered channel 59 -- Unregistered channel 60 -- Unregistered channel 61 -- Unregistered channel 62 -- Unregistered channel 63 -- Unregistered channel 64 -- Unregistered channel 65 -- Unregistered channel 66 -- Unregistered channel 67 -- Unregistered channel 68 -- Unregistered channel 69 -- Unregistered channel 70 -- Unregistered channel 71 -- Unregistered channel 72 -- Unregistered channel 73 --
[Asterisk-Users] SuperValetParkCall Application Unable to Re-Park Call
After retrieving a SuperValetParkCall using the SuperValetUnparkCall
command, I am unable to re- SuperValetParkCall the call again. Can
anyone confirm if this is a bug or my configs may be incorrect.
Configs:
exten =
3,1,SuperValetParkcall(${EXTEN:1}|mylot|500|${EXTEN:1}|1|local)
exten = _3,2,Hangup
exten = _*3,1,SuperValetUnParkCall(${EXTEN:2}|mylot)
exten = _*3,2,Hangup
The CLI:
-- Executing Dial(SIP/2205-d565, sip/2203|20|tr) in new stack
-- Called 2203
-- SIP/2203-1119 is ringing
-- SIP/2203-1119 answered SIP/2205-d565
-- Attempting native bridge of SIP/2205-d565 and SIP/2203-1119
-- Started music on hold, class 'default', on SIP/2205-d565
-- Executing SuperValetParkCall(SIP/2203-7a5d,
2204|mylot|500|2204|1|local) in new stack
-- Started music on hold, class 'default', on SIP/2203-7a5d
== Super Valet Parked SIP/2203-7a5d on slot 2204
-- Executing Hangup(SuperValetParked/SIP/2203-7a5d
ZOMBIE, ) in new stack
== Spawn extension (local, h, 1) exited non-zero on
'SuperValetParked/SIP/2203-7a5d
ZOMBIE'
-- Stopped music on hold on SIP/2203-7a5d
-- Stopped music on hold on SIP/2205-d565
-- Started music on hold, class 'default', on SIP/2203-7a5d
== Spawn extension (macro-stdexten, s, 2) exited non-zero on
'SIP/2203-7a5dZOMBIE' in macro 'stdexten'
== Spawn extension (local, 2203, 1) exited non-zero on
'SIP/2203-7a5dZOMBIE'
-- Executing Hangup(SIP/2203-7a5dZOMBIE, ) in new stack
== Spawn extension (local, h, 1) exited non-zero on
'SIP/2203-7a5dZOMBIE'
-- Executing SuperValetUnparkCall(SIP/2203-cbc3, 2204|mylot) in
new stack
-- Stopped music on hold on SIP/2205-d565
-- Channel SIP/2203-cbc3 connected to SuperValet Parked call 2204 in
lot mylot
-- Executing SuperValetParkCall(SIP/2203-376e,
2204|mylot|500|2204|1|local) in new stack
-- Started music on hold, class 'default', on SIP/2203-376e
== Super Valet Parked SIP/2203-376e on slot 2204
-- Executing Hangup(SuperValetParked/SIP/2203-376e
ZOMBIE, ) in new stack
== Spawn extension (local, h, 1) exited non-zero on
'SuperValetParked/SIP/2203-376e
ZOMBIE'
a
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there hardware to remote control
I found this interesting box at qkits.com QK108 It has 8 relay outputs and 4 inputs. It's controlled via RS-232c(serial port) instead of a printer port. I have an 8 port serial card in a linux server to control a bunch of stuff. I have apache on that server and can control the relays via a cgi script. I found it very easy to program a serial port via perl and with an 8 port serial card(from Perle). You can have a bunch of stuff hanging off it, like a 4 probe temp kit (QK 145) and a ups and an RS-232 voltmeter to monitor the commerical power coming in and I suspose it would be easy to take this even further to write AGI scripts and dial an extension and let * announce the temperature or status of those inputs and to control the outputs of the QK108. I also use it with their K2639 to monitor the sump pits, monitoring for sump pump failure.(therefore high water levels). Lyle - Original Message - From: David Cook [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, December 20, 2004 9:10 AM Subject: Re: [Asterisk-Users] Is there hardware to remote control From: Ronald Wiplinger [EMAIL PROTECTED] Subject: [Asterisk-Users] Is there hardware to remote control available? To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=us-ascii; format=flowed I am looking for a hardware, which can turn on / off (control) via the dial plan. Is something available? You can run an AGI from within your diaplan which can do anything available to the host machine. As for turning things on/off, you have several options. a) serial port control; b) parallel port control; c) attached microcontroller; d) X-10 signals. Please exuse this for going OT into home automation stuff, but in an effort to answer the original question, here goes ... a) I have often used a little program that flips the DTR RTS signals on a serial port (independently so you can control two things). You need to turn on/off a logic state or an LED that is fine. If you need to switch a larger electical load, put a solid state relay on that pin. I have my laser printer and my pool pump controlled that way. b) Parallel port works basically the same way with the 8 output pins on the connector that can be controlled. Haven't actually done this though. Lastly, connect a microcontroller like a Parallax Basic Stamp to your server where you can write code that runs on the microcontroller and does numerous things pseudo autonomously from c) Microcontroller like the Parallax Basic Stamp series. This allows you to run a program on this little computer device (100.00) that was made for I/O control. It can do all kinds of things pseudo autonomously and feed back the info to the PC. d) X10 have several interfaces for PC's. I like a little one called the Firecracker interface. It uses an RS232C line and can control devices by sending radio signals from it to a reciever module that is plugged into a wall socket. It then embeds the cammands you sent it into the electrical circuits in your home. Another module then plugs into the wall somewhere and you plug devices into it. The little wall modules recieves the signal coming along the electrical lines and turns the device on/off/dim, etc. The reason I like the Firecracker is that it is a dumb device. All program code must exist on the PC therefore I have more control. They have other devices which you download program code to then they are autonomous which I don't think is what you are looking for. I use a) d) extensively here. If anyone wants the code or more info, just ask. David Cook ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is there hardware to remote control
Lyle Giese wrote: I found this interesting box at qkits.com QK108 It has 8 relay outputs and 4 inputs. It's controlled via RS-232c(serial port) instead of a printer port. I have an 8 port serial card in a linux server to control a bunch of stuff. I have apache on that server and can control the relays via a cgi script. I found it very easy to program a serial port via perl and with an 8 port serial card(from Perle). You can have a bunch of stuff hanging off it, like a 4 probe temp kit (QK 145) and a ups and an RS-232 voltmeter to monitor the commerical power coming in and I suspose it would be easy to take this even further to write AGI scripts and dial an extension and let * announce the temperature or status of those inputs and to control the outputs of the QK108. I also use it with their K2639 to monitor the sump pits, monitoring for sump pump failure.(therefore high water levels). Cool, I've got to check that out. For the OP, check my 1st web page below for various odds and ends for the hardware and software that can be run on a *nix box. It's not directly * related but if you can run scripts you can take advantage of my collection of links. -- Linux Home Automation Neil Cherry [EMAIL PROTECTED] http://home.comcast.net/~ncherry/ (Text only) http://hcs.sourceforge.net/ (HCS II) http://linuxha.blogspot.com/My HA Blog ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SuperValetParkCall Application Unable to Re-Park Call
Kevin wrote:
After retrieving a SuperValetParkCall using the SuperValetUnparkCall
command, I am unable to re- SuperValetParkCall the call again. Can
anyone confirm if this is a bug or my configs may be incorrect.
Configs:
exten =
3,1,SuperValetParkcall(${EXTEN:1}|mylot|500|${EXTEN:1}|1|local)
exten = _3,2,Hangup
exten = _*3,1,SuperValetUnParkCall(${EXTEN:2}|mylot)
exten = _*3,2,Hangup
This is not a bug. I experienced the same problem, and took a peek at
the code.
Apparently app_supervaletparking uses it's own channel bridge function,
which
is very basic and doesn't process the # transfer command.
I need the ability to transfer a call after a SuperValetUnparkCall so I
went back to using app_valetparking. I wasn't using the new features in
app_supervaletparking, just the park and unpark functions.
Marv Horst
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: E1 card for Asterisk
Follow-up to the list: The boss decided to take over my private project (so the money's no longer coming from my own purse -- yeepee!) and, after doing some research of his own and talking to the Sangoma people, he decided to go the Sangoma way. I will keep the list posted on our success/failures, as I will be heavily involved with this on the technical side. Best Regards, -- Telmo. On Wed Dec 22 3:27 , [EMAIL PROTECTED] sent: Hello Folks, I'm trying to decide here between a few cards for connecting an Asterisk box to a single E1 channel (either PRI or R2 signaling): - Digium E100P: has been replaced by the TE110P below, but can still be had at places like digitnetworks.com for $475, and I guess there's always a place for good-olde-obsolete cards in the world as long as they work :-) - Digium TE110P: replacement for the above card. Costs $595 at the Digium webstore; aditionally can be configured to work with both E1 *and* T1. - Sangoma A101: billed as compatible with Asterisk on Sangoma's page at http://www.sangoma.com/products/p_a101-102-specs.htm; also for E1 and T1, the specs list only compatibility with CAS and PRI signaling, but a pal asked Sangoma directly and they said that the card is compatible with E1/R2 signaling. Cost is also $595 at voipstore.atacomm.com. On the side of the E100P/TE110P there's my wish to support Digium in any way I can for the nice work they are doing with Asterisk; But then, I've been using Sangoma cards for almost 9 years now on data (frame-relay, PPP, etc) applications, and the first card I bought from them is still working on a production machine(!); Also, the A101 works with Linux for data-only connections too (something I understand the Digium cards can't do at the moment) and can be upgraded to a dual-E1 A102 for an additional $300. So what I would like to ask the knowledgeable guys and gals here is: 1) Can anyone tell any good/bad/nice/ugly experiences with any of the above cards and Asterisk, specially in an E1/R2 configuration? 2) Can anyone point me to nice on-line sellers of Asterisk-compatible hardware, specially E1 cards? The only ones I know are the Digium webstore, and the above mentioned digitnetworks.com and voipstore.atacomm.com... I'm about to spend a fair chunk of money from my own pockets in this, friends, and all your comments/suggestions/advice is very, very welcome. Thanks in advance, Telmo. SeeqMail - the only email you'll ever need Starts Here Sign up for FREE personalized email today: http://www.seeqmail.com http://www.Grassroots.org/ - Make Change! SeeqMail - the only email you'll ever need Starts Here Sign up for FREE personalized email today: http://www.seeqmail.com http://www.Grassroots.org/ - Make Change! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: 'I'nvalid extension handling problems, even with workaround
Hi Rich,
Thanks for the further help you are extending me. I will be trying your
suggestions during the holidays, and when I get back on-line will post my
progress here.
Thanks again.
Merry Christmas to all,
--
Telmo.
On Thu Dec 23 5:06 , Rich Adamson [EMAIL PROTECTED] sent:
Hello Rich,
First of all, thank you very much for your help and patience.
I've just arrived home from work (yes, I'm one of the midnight oil burners
:-))
and implemented and tested your suggestions; unfortunatelly it didn't work,
the
same behaviour remains.
More details follow below, in-line:
On Wed Dec 22 4:14 , Rich Adamson [EMAIL PROTECTED] sent:
Inline...
Humrmrm... 2 days, no answers... :-/
Well, let me see if I can take a stab at this one.
You sure did. Your help was most appreciated, I learned a lot from thinking
about
your suggestions, even if they did not work as planned (see below) they made
a
lot of sense.
After working with * for about a year now, I'd suggest the toughest
part of the learning curve is truly understanding how to take advantage
of the various 'context' statements to accomplish an objective. Part of
reason for the steep learning curve appears to relate to the lack of
any reasonable form of tracing/debugging what the system is actually
using for a context at each step. (What I mean is that its not intutive
for the beginner.)
I agree. I'm working with 15 (!) -v on the command-line here and yet I can
only
perceive Asterisk has left or entered a context indirectly, by the commands
that
are executing.
It would really be nice if there was a 'debug context' type command
that would simply display each extensions.conf line as it is executed.
That can't be too hard to implement, when I'm a little more familiarized with
Asterisk I will certainly try to code that in.
Either I made a stupid question (I don't think so: I have *really* tried
to
solve
that on my own before asking the list) or this one's just something
nobody has
ever tried but me (I also find that unlikely: even the telco here plays a
message
when I dial a wrong number; also there's the wiki page I mentioned, which
indicates that someone in the past has had the same issue).
I'm having trouble configuring Asterisk to play an invalid extension
message to
anyone dialing an undefined extension.
I then did the separate context with _. trick the above wiki page
suggests; at
first it seemed to work: picking up an extension and dialing any invalid
extension would play the message (albeit it would play twice, can't
understand
why) and then hang up.
;;;extensions.conf
[internal] ;;; context used by our internal SIP-phon
include = voiptalk.org ;include context below
exten = 11,1,Dial(SIP/gsbt100,20,tr);calling : dial our office phone
include = invalid_calls;all ext numbers not handled above are
invalid
The 'separate context' approach _does_ work, but you've just confused
that approach by dropping the Dial statement in the middle. Change this
to something like:
[internal]
include = valid-extensions
include = voiptalk
include = any-other-context-that-you-need
include = invalid-calls
Makes a lot of sense, and also leads to much more intelligible structure. I
implemented it almost literally as you suggest (see below for my
extensions.conf
file after the modifications).
The above _sequence_ of include statements is maintained for each call.
In other words, if a call does _not_ match entries in 'valid-extensions',
then it proceeds to the next include. However, if a match is found
(including
special cases such as 't', etc) then the call processing may _not_ step
through the remaining includes.
OK. I understand what you are stating, it makes a lot of sense.
Unfortunatelly it
seems Asterisk disagrees with us... :-) please see below.
I'm not sure at all about using context names with a period in it, so
just to ensure that isn't causing an issue, stick to context names with
alphanumeric characters. (At least eliminate that uncertainty.)
OK. Let's play it _really_ safe: I've removed . from context names, and
replaced _ (underline) for - (dashes), so right now I'm only using
characters
from the set [a-z0-9-] in context names.
[voiptalk.org]
;forwards any calls starting with an 8 thru voiptalk.org
exten = _8.,1,Answer
exten = _8.,3,SetCIDNum()
exten = _8.,4,SetCIDName(My Name And Surname)
exten = _8.,5,Dial(SIP/${EXTEN:[EMAIL PROTECTED],,g)
exten = _8.,6,HangUp
[invalid_calls] ;;; default context for invalid calls
exten = _.,1,Wait(1)
exten = _.,2,Answer
exten = _.,3,Playback(invalid)
exten = _.,4,Hangup
;;;end of extensions.conf
In the above [invalid_calls] context, change the order to:
exten = _.,1,Answer
exten = _.,2,Wait(1)
exten = _.,3,Playback(invalid)
exten = _.,4,Hangup
Also makes a lot of sense. Done it that
Re: [Asterisk-Users] Record() problem
It was executed from the dial plan within extensions.conf and I did not hard
code the IAX2/[EMAIL PROTECTED]/5 in there. I will paste the exact text below
from my extensions.conf which I really should have done the first time :)
sorry..
I didn't include the Macro but that's not where it's blowing up. Any help
would be appreciated.
Happy Holidays to all!
*From extensions.conf*
; 1100 - Test call whisper type thing
;exten = 1100,1,Wait(0.2)
;exten = 1100,2,Playback(say-name)
;exten = 1100,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
;exten = 1100,4,Record(${SCREEN_FILE}.gsm,6,25)
;exten = 1100,5,Dial(SIP/1100,60,gM(screen^${SCREEN_FILE}))
;exten = 1100,6,Voicemail([EMAIL PROTECTED])
***End
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Bill Seddon [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, December 24, 2004 3:00 AM
Subject: RE: [Asterisk-Users] Record() problem
You syntax for the command is incorrect. See
http://www.voip-info.org/wiki-Asterisk+cmd+record.
Record is an application to be executed from within the dialplan. So the
channel it will record is implicit and cannot be explicitly stated as one
of
the parameters.
If you want to originate and record a call automatically, you will have to
do this via AGI.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Me
Sent: December 24, 2004 6:38 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Record() problem
Any idea why this:
Record(IAX2/[EMAIL PROTECTED]/5, /tmp/whatever.gsm|6|25)
Would result in this:
WARNING[3293201]: app_record.c:117 record_exec: No extension found
Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.0.3 no RedHat zaptel script?
Remco... Look in your Zaptel source directory, you'll find the following related to config scripts... zaptel.sysconf zaptel.init zaptel.conf.sample Happy Holidays! Grady -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Thursday, December 23, 2004 3:17 PM To: Asterisk Users List Subject: [Asterisk-Users] Asterisk 1.0.3 no RedHat zaptel script? Just out of interest, why is there no zaptel script included in the tarballs of 1.0.3? I used to use the RPMS but they haven't been updated for some time but now I'm missing the zaptel init.d script. Or should I have the modules loaded another way? Cheers! Remco ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help on Register message with Proxy-Authorization
Sounds like you are developing an application. You should read RFC-3261 and RFC-2617 Kamran Ahmad wrote: Can any one help me in understanding REGISTER message when i send REGISTER message to asterisk it is replying 407 with header Proxy-Authenticate: Digest realm=asterisk,nonce=1011592446 i want password for my user so i entered secret in sip.conf against userid can any one tell me how to handle Proxy-Authenticate and Proxy-Authorization i want to know where to enter the password in my second reply with Proxy-Authorization header and how to encrypt it __ Do you Yahoo!? Send a seasonal email greeting and help others. Do good. http://celebrity.mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Registration failure with debug
can anybody identify why the CLI is issuing a failure message while debug shows everything is fine this makes no sense to me. also, why is the username being updated? this has got to be wrong from CLI -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]: chan_sip.c:7742 handle_request: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 -- Saved useragent Grandstream BT100 1.0.5.20 for peer 52221 -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 from debug /var/log/messages Dec 24 12:16:34 DEBUG[15776]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '52221' Dec 24 12:16:34 DEBUG[15776]: MySQL RealTime: Everything is fine. Dec 24 12:16:34 VERBOSE[15776]: -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '52221' Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Everything is fine. Dec 24 12:16:35 VERBOSE[15776]: -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 Dec 24 12:16:35 NOTICE[15776]: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.70.26' Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '52221' Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Everything is fine. Dec 24 12:16:35 VERBOSE[15776]: -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 Dec 24 12:16:35 VERBOSE[15776]: -- Saved useragent Grandstream BT100 1.0.5.20 for peer 52221 Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Update SQL: UPDATE sip_buddies SET ipaddr = '192.168.70.26', port = '5060', regseconds = '1103912195', username = '52221' WHERE name = '52221' Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Everything is fine. Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Updated 1 rows on table: sip_buddies Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '52221' Dec 24 12:16:35 DEBUG[15776]: MySQL RealTime: Everything is fine. Dec 24 12:16:35 VERBOSE[15776]: -- SIP Seeding '52221' at [EMAIL PROTECTED]:5060 for 3600 Dec 24 12:16:35 DEBUG[16042]: Device 'SIP/52221' changed to state '0' Dec 24 12:16:50 DEBUG[15776]: Auto destroying call '[EMAIL PROTECTED]' Regards Greg Cirino ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream 1.0.5.20 firmware?
Greg - Cirelle Enterprises wrote: -- Saved useragent Grandstream BT100 1.0.5.20 for peer 52221 Greg, Completely unrelated to your current query. Your logs show that your BT100 is running 1.0.5.20 firmware. Is this correct? The last I knew, they were at 1.0.5.18 Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Preventing Asterisk from sending 'h' across to SIP Provider
Hi,
I want to prevent Asterisk from sending the h extension across to the SIP
provider or to prevent it from hitting the script at all. The SIP Provider
does not know what to do with the h extensions once it receives it. My SIP
Provider takes all digits and forwards them off to a softswitch for
processing. Everytime a call hangs up, it complains about running AGI scripts
on hungup channels and to use DeadAGI. I want it to not send the hangup to
the provider at all. Taking out the Hangup line does not help nor does
autofallthrough=no. I have posted my extensions.conf below. Thanks -
[general]
autofallthrough=yes
context=default
[default]
;exten = _.,1,Dial(SIP/[EMAIL PROTECTED],70,t)
exten = _.,1,AGI(mta_auth.agi,${EXTEN})
exten = _.,2,Hangup
--
Brian Wilkins
Software Engineer
[EMAIL PROTECTED]
Heritage Communications Corporation
Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Preventing Asterisk from sending 'h' across to SIP Provider
On Dec 24, 2004, at 5:44 AM, Brian Wilkins wrote:
[default]
;exten = _.,1,Dial(SIP/[EMAIL PROTECTED],70,t)
exten = _.,1,AGI(mta_auth.agi,${EXTEN})
exten = _.,2,Hangup
Don't use _., it matches s, h, i, and all of the other 1-letter
extensions. Use _X. instead.
Scott
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with DISA application
I am testing DISA but can not dial out after getting local dialtone. The * server takes accepts my password, gives me dialtone and allows me to dial the digits then promply hangs up on me. CLI log shows -- Executing DISA(Zap/1-1, 1234|contextname) in new stack -- Accepting call from '9104025011' to '9105551212' on channel 0/1, span 1 Dec 24 14:50:01 WARNING[1989798704]: app_disa.c:290 disa_exec: DISA on chan Zap/1-1 password is good Dec 24 14:50:06 WARNING[1989798704]: cdr.c:286 ast_cdr_init: CDR already initialized on 'Zap/1-1' -- Executing Dial(Zap/1-1, Zap/g2/9104041000) in new stack -- Called g2/9104041000 -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Hangup(Zap/1-1, ) in new stack == Spawn extension (contextname, 9104041000, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' My extensions.conf reads: ; Disa test exten = 9105551212,1,DISA,1234|contextname Any ideas? Thanks, Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream 1.0.5.20 firmware?
the latest firware for Grandstream devices can be found at http://fm.grandstream.com/gs/ - Original Message - From: Doug Lytle [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, December 24, 2004 10:47 AM Subject: [Asterisk-Users] Grandstream 1.0.5.20 firmware? Greg - Cirelle Enterprises wrote: -- Saved useragent Grandstream BT100 1.0.5.20 for peer 52221 Greg, Completely unrelated to your current query. Your logs show that your BT100 is running 1.0.5.20 firmware. Is this correct? The last I knew, they were at 1.0.5.18 Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Switch polarity to disconnect a FXS channel
On Fri, 24 Dec 2004, Luis Czop wrote: Hi friends, I´m trying to integrate Asterisk with another PBX (Nortel Meridian) I need to switch the polarity of a FXS port (ZAP channel) to inform the other PBX that the channel was released. Does anyone knows the way I can do it? There has been some talk lately regarding both detcting reversals and generating them. The former is mostly on the bug tracker and the latter (which you are interesetd in) can be found on asterisk-users. Search for TDM400P FXS polarity reversal. I don't think either is implemented yet in the standard distribution/cvs tree. Peter ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Preventing Asterisk from sending 'h' across to SIP Provider
Brian Wilkins wrote:
Hi,
I want to prevent Asterisk from sending the h extension across to the SIP
provider or to prevent it from hitting the script at all. The SIP Provider
does not know what to do with the h extensions once it receives it. My SIP
Provider takes all digits and forwards them off to a softswitch for
processing. Everytime a call hangs up, it complains about running AGI scripts
on hungup channels and to use DeadAGI. I want it to not send the hangup to
the provider at all. Taking out the Hangup line does not help nor does
autofallthrough=no. I have posted my extensions.conf below. Thanks -
[general]
autofallthrough=yes
context=default
[default]
;exten = _.,1,Dial(SIP/[EMAIL PROTECTED],70,t)
exten = _.,1,AGI(mta_auth.agi,${EXTEN})
exten = _.,2,Hangup
Don't use _. pattern. . means match anything. Use _X. or something
like that.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Uniden UIP200 firmware v4.63
I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in the sip.conf). Does the UIP 200 work across a nat yet? If it does, care to share your config for it? Thanks, Lyle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream 1.0.5.20 firmware?
Got that from grandstream, and testing it for a couple of things Greg At 12:47 PM 12/24/04, you wrote: Greg - Cirelle Enterprises wrote: -- Saved useragent Grandstream BT100 1.0.5.20 for peer 52221 Greg, Completely unrelated to your current query. Your logs show that your BT100 is running 1.0.5.20 firmware. Is this correct? The last I knew, they were at 1.0.5.18 Doug ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Regards Greg Cirino ___ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Web Application Development Design www.cirelle.net ProSpeed High Speed Dial-up - 6 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com NNEREN MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a Rhyme... There's a Reason! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tie web application to VOIP
I want to tie my web application (built using .NET + MS SQL Server) into a VOIP service so that users can call each other. I want them to interface with my application's username system. On the receiving user's end, he can either receive the call using a VOIP phone, or windows application (like skype). I would use Skype's API, but it appears I need to use their username system, and not my own. My question is, what software/hardware solutions would I need to do this? Any suggestions/feedback would be greatly appreciated. Btw, I was told that Asterisk + SER would do the trick. However, I'm a newbie to the world of VOIP. If someone can give me some tips/hints, it would be great. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tie web application to VOIP
Any particular reason you want to do it in .NET and MS SQL? I personally would write applications in something a bit more portable. Just curious. And I wouldn't use skype for anything, but then again, I am a bit anti-skype as well. -mishehu I want to tie my web application (built using .NET + MS SQL Server) into a VOIP service so that users can call each other. I want them to interface with my application's username system. On the receiving user's end, he can either receive the call using a VOIP phone, or windows application (like skype). I would use Skype's API, but it appears I need to use their username system, and not my own. My question is, what software/hardware solutions would I need to do this? Any suggestions/feedback would be greatly appreciated. Btw, I was told that Asterisk + SER would do the trick. However, I'm a newbie to the world of VOIP. If someone can give me some tips/hints, it would be great. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users !DSPAM:41cc921930314968028340! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SuperValetParkCall Application Unable to Re-ParkCall
Thanks, for the feedback. It would be nice if the Valetpark application
didn't have that limitation. The park to a specific park number is a
nice feature.
-Original Message-
From: Paul Zimm [mailto:[EMAIL PROTECTED]
Sent: Friday, December 24, 2004 11:40 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] SuperValetParkCall Application Unable to
Re-ParkCall
Kevin wrote:
After retrieving a SuperValetParkCall using the SuperValetUnparkCall
command, I am unable to re- SuperValetParkCall the call again. Can
anyone confirm if this is a bug or my configs may be incorrect.
Configs:
exten =
3,1,SuperValetParkcall(${EXTEN:1}|mylot|500|${EXTEN:1}|1|local)
exten = _3,2,Hangup
exten = _*3,1,SuperValetUnParkCall(${EXTEN:2}|mylot)
exten = _*3,2,Hangup
This is not a bug. I experienced the same problem, and took a peek at
the code.
Apparently app_supervaletparking uses it's own channel bridge function,
which
is very basic and doesn't process the # transfer command.
I need the ability to transfer a call after a SuperValetUnparkCall so I
went back to using app_valetparking. I wasn't using the new features
in
app_supervaletparking, just the park and unpark functions.
Marv Horst
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
No virus found in this incoming message.
Checked by AVG Anti-Virus.
Version: 7.0.296 / Virus Database: 265.6.4 - Release Date: 12/22/2004
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uniden UIP200 firmware v4.63
I had th same problem, had to finally connect the phone directly to a tftp server to get it to work. -Original Message- From: Lyle Giese [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 24 Dec 2004 15:36:44 -0600 Subject: [Asterisk-Users] Uniden UIP200 firmware v4.63 I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in the sip.conf). Does the UIP 200 work across a nat yet? If it does, care to share your config for it? Thanks, Lyle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FC3, TDM11B (DEVPCI) and asterisk
Hey folks; I am a old Dialogic telephony hack and have tried asterisk a while back on a laptop with the OSS module and I liked what I saw. I recently came across some info regarding asterisk 1.0 and thought I should give it a go. I used a AMD K7 MB, compiled the binaries without a real problem and realized that I might have a sound card issue because I couldn't hear the messages but the demo worked otherwise. I noticed that my sound card was having a problem so I abandoned that approach but I'm not finished just yet. I then decided to get a DEV kit and give it a real go. I compiled the binaries (zaptel and asterisk) without incident and I was able to start the drivers. I have green lights on port 1 and 4 and I understand that port 1 is the station port (for a 2500 set regular phone set?) and that port 4 is able to connect to a regular 1FL (simple pots line). I used some 'examples' from the Digium site and when that didn't work I used a few from a link I found (http://iheavy.com/modules.php?op=modloadname=Newsfile=articlesid=35) but I have no dialtone on the phone and it won't pickup (or write ANY logs) when I call the line. My question(s) for the list is/are; Does anyone have a working asterisk system using zaptel TDM cards on FC3 with kernel 2.6.9-1.681_FC3 and if so, is there something that I am missing? Any tips would be appreciated and thanx in advance. Jeff ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Uniden UIP200 firmware v4.63
I have a tftp server on the local subnet and it's picking up the config files just fine. It can call out and I have two-way audio. Asterisk cann't seem to talk to the phone and therefore you cann't call the phone. That's as far as I could get with it. I played with the proxy and registrar settings on the phone and the nat= settings in sip.conf as well as the port parameter with no success. Lyle - Original Message - From: Charles S. Antrim [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Lyle Giese [EMAIL PROTECTED] Sent: Friday, December 24, 2004 5:25 PM Subject: Re: [Asterisk-Users] Uniden UIP200 firmware v4.63 I had th same problem, had to finally connect the phone directly to a tftp server to get it to work. -Original Message- From: Lyle Giese [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Fri, 24 Dec 2004 15:36:44 -0600 Subject: [Asterisk-Users] Uniden UIP200 firmware v4.63 I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in the sip.conf). Does the UIP 200 work across a nat yet? If it does, care to share your config for it? Thanks, Lyle ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoiceConduits?
Does anyone have any experience with http://www.voiceconduits.com ? I decided to try them out because the administrative interface on their service looked promising but so far I can't even get my IAX.conf entries generated and their support hasn't responded to a single issue yet. I've tried them on AIM and via e-mail and I've never heard anything back. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-AS5350 misplaced RTP to 127.0.0.1 (AS5350 party don't hear)
Try to explicitly bind it to eth0... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firefly Transfer call ?
Does anybody know how to transfer a call from firefly ? Is it possible ? Will it be included in a new version ? Thanks and Merry Christmas ___Jean-François Rousseauwww.sys-tech.net[EMAIL PROTECTED]Tél. 24h (418) 520-0739 Télec.(418) 520-45541-877-969-tech Ouverture Technologique ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Tie web application to VOIP
Any particular reason you want to do it in .NET and MS SQL? I personally would write applications in something a bit more portable. Just curious. MS SQL 2005 Express is probably the best free DB out there? And I run lots of Mono code just fine... I want to tie my web application (built using .NET + MS SQL Server) into a VOIP service so that users can call each other. I want them to interface with my application's username system. Here's what you can do: Write a web service (ASMX) that writes .call files and drops them in the outgoing spool. Load Mono on the Asterisk box, and host the web service with XSP (or Apache with mod_mono). Then you can call that web service from your .NET web app, and away you go. I do this, (as well as vice versa, having Mono on Linux connect back to my Windows .NET apps) without any problems. Also, I'm still working on ast_mono, which will allow you to write native Asterisk apps in C# or C++, or whatever you please (I won't mention VB :P). -Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What do I need to build up DID services?
To complete my project, I would like to setup DIDs in several areas. What do I need to do that? Another Asterisk box or can I use gateways instead? Which hardware can I use? Who has experience? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Preventing Asterisk from sending 'h' across to SIPProvider
Don't use _. Use _X.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Brian Wilkins
Sent: Friday, December 24, 2004 7:45 AM
To: Asterisk-users
Subject: [Asterisk-Users] Preventing Asterisk from sending 'h' across to
SIPProvider
Hi,
I want to prevent Asterisk from sending the h extension across to the
SIP
provider or to prevent it from hitting the script at all. The SIP Provider
does not know what to do with the h extensions once it receives it. My SIP
Provider takes all digits and forwards them off to a softswitch for
processing. Everytime a call hangs up, it complains about running AGI
scripts
on hungup channels and to use DeadAGI. I want it to not send the hangup to
the provider at all. Taking out the Hangup line does not help nor does
autofallthrough=no. I have posted my extensions.conf below. Thanks -
[general]
autofallthrough=yes
context=default
[default]
;exten = _.,1,Dial(SIP/[EMAIL PROTECTED],70,t)
exten = _.,1,AGI(mta_auth.agi,${EXTEN})
exten = _.,2,Hangup
--
Brian Wilkins
Software Engineer
[EMAIL PROTECTED]
Heritage Communications Corporation
Melbourne, FL USA 32935
321.308.4000 x33
http://www.hcc.net
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Record() problem
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
Refer to my example on that bug note.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Me
Sent: Friday, December 24, 2004 11:06 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-
Commercial Discussion
Subject: Re: [Asterisk-Users] Record() problem
It was executed from the dial plan within extensions.conf and I did not
hard
code the IAX2/[EMAIL PROTECTED]/5 in there. I will paste the exact text
below
from my extensions.conf which I really should have done the first time :)
sorry..
I didn't include the Macro but that's not where it's blowing up. Any help
would be appreciated.
Happy Holidays to all!
*From extensions.conf*
; 1100 - Test call whisper type thing
;exten = 1100,1,Wait(0.2)
;exten = 1100,2,Playback(say-name)
;exten = 1100,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
;exten = 1100,4,Record(${SCREEN_FILE}.gsm,6,25)
;exten = 1100,5,Dial(SIP/1100,60,gM(screen^${SCREEN_FILE}))
;exten = 1100,6,Voicemail([EMAIL PROTECTED])
***End
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: Bill Seddon [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Friday, December 24, 2004 3:00 AM
Subject: RE: [Asterisk-Users] Record() problem
You syntax for the command is incorrect. See
http://www.voip-info.org/wiki-Asterisk+cmd+record.
Record is an application to be executed from within the dialplan. So
the
channel it will record is implicit and cannot be explicitly stated as
one
of
the parameters.
If you want to originate and record a call automatically, you will have
to
do this via AGI.
Bill Seddon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Me
Sent: December 24, 2004 6:38 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Record() problem
Any idea why this:
Record(IAX2/[EMAIL PROTECTED]/5, /tmp/whatever.gsm|6|25)
Would result in this:
WARNING[3293201]: app_record.c:117 record_exec: No extension found
Thanks!
--
Start Your Own ISP!
http://www.YourOwnISP.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Tie web application to VOIP
On Fri, 2004-12-24 at 20:17 -0600, Michael Giagnocavo wrote: Any particular reason you want to do it in .NET and MS SQL? I personally would write applications in something a bit more portable. Just curious. MS SQL 2005 Express is probably the best free DB out there? And I run lots of Mono code just fine... Thats highly debateable as you have a insane prerequisite of running windows first. So you start with an insecure crap software as the base. How can you have anything that is really worth the time of running when it is first built on top of crap? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Calling Party ringing indicator
I am new to Asterisk and I'm setting up my first Asterisk box. When recieving inbound SIP calls from sipphone.com my friends would not get the ringing indicator after I did an answer command. I found that after playing a gsm file first the ringing started to work like it did for my pots incoming calls. I created a short virtually silent gsm file and that worked as well if played before the dial command. Just wondering why SIP works this way and if there is a better solution. BTW, I'm not using the r parameter to the dial command. John ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Tie web application to VOIP
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steven Critchfield Sent: Friday, December 24, 2004 9:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Tie web application to VOIP On Fri, 2004-12-24 at 20:17 -0600, Michael Giagnocavo wrote: Any particular reason you want to do it in .NET and MS SQL? I personally would write applications in something a bit more portable. Just curious. MS SQL 2005 Express is probably the best free DB out there? And I run lots of Mono code just fine... Thats highly debateable as you have a insane prerequisite of running windows first. So you start with an insecure crap software as the base. How can you have anything that is really worth the time of running when it is first built on top of crap? -- Steven Critchfield [EMAIL PROTECTED] Steven, Just a quick reminder, MS SQL on Windows is hands down the best performing transact SQL database on the planet, and Oracle on Windows is a close #2. Some might argue that Oracle is #1 and MS is #2. Anyone that argues any Linux SQL db even comes close in performance better provide some evidence to back their argument. The asp.net SQL database providers for MS SQL and Oracle SQL are highly optimized direct socket interfaces to the SQL server. (no odbc crap!) You would be hard pressed to build a faster web app than asp.net talking to MS or Oracle SQL servers with the native .net provider. It will not surprise me a bit to see the better of the asterisk web front ends on MS platforms, primarily because the larger companies that might use * in the future already use MS SQL and Oracle for their customer databases. Don't get me wrong, I have nothing against UNIX/Linux, I just know that both windows and Linux have their place in the world. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ALERT_INFO issue CVS-HEAD-12/24/04
Anyone having any problems with CVS-HEAD-12/24/04-15:59:15 and ALERT_INFO I have a system setup with polycom phones configured to auto answer on internal calls. When we upgraded to the latest CVS the auto answer stopped working. My dialplan has not changed. I did a sip debug and I dont see the alert-info tag in any of the sip traces. Any help would be appreciated. Thanks John Bittner Simlab.net This is a what I have in my dialplan. exten = 207,1,SetVar(ALERT_INFO=Ring Answer) exten = 207,2,Dial(SIP/207) exten = 207,3,Hangup ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tie web application to VOIP
On December 24, 2004 09:17 pm, Michael Giagnocavo wrote: MS SQL 2005 Express is probably the best free DB out there? And I run lots of Mono code just fine... *cough* okay. Sure. Whatever you say... -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tie web application to VOIP
Damon Estep wrote: Steven, Just a quick reminder, MS SQL on Windows is hands down the best performing transact SQL database on the planet, and Oracle on Windows is a close #2. Some might argue that Oracle is #1 and MS is #2. Anyone that argues any Linux SQL db even comes close in performance better provide some evidence to back their argument. I think I would cite Oracle. They say their database runs much better on Linux than on Windows. If Oracle on Windows is arguably better than MS SQL, Oracle on linux must be a hands down winner. The last time I used MS SQL was about 5 years ago, but it was hands down the biggest heap of crap in the database world back then. I guess it has improved, but if you think its the best out there I guess you haven't tried very much stuff. The asp.net SQL database providers for MS SQL and Oracle SQL are highly optimized direct socket interfaces to the SQL server. (no odbc crap!) Its only the Windows crap which ever really used ODBC. Most people have always used direct operations. You would be hard pressed to build a faster web app than asp.net talking to MS or Oracle SQL servers with the native .net provider. It will not surprise me a bit to see the better of the asterisk web front ends on MS platforms, primarily because the larger companies that might use * in the future already use MS SQL and Oracle for their customer databases. Don't get me wrong, I have nothing against UNIX/Linux, I just know that both windows and Linux have their place in the world. I think you've been drinking the Koolade. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What do I need to build up DID services?
On Sat, 25 Dec 2004, Ronald Wiplinger wrote: To complete my project, I would like to setup DIDs in several areas. What do I need to do that? Another Asterisk box or can I use gateways instead? Which hardware can I use? Who has experience? You either set up your own points of presence, or buy service from somebody who already has them set up. So you could install an * with a zap card, a stand-alone spa-3000, or some other device to establish the POP where you want it (and you'll need broadband there too), or you could just buy the service from a VOIP carrier. The latter, IMHO, is better because you won't have to deploy hardware all over. Deployed hardware requires a place to live, somebody to feed it and provide TLC whenever necessary. Usually it's better to keep hardware in as few places as possible to simplify that task.. If you don't like the VOIP delivery, telcos for years have offered market expansion lines which give you a number in a remote rate center which automatically forwards to your real number. (funny thing is, when I talked to Qwest about that a month ago, this forwarding service cost more than regular residential phone service!) Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Linux Distribution
What are the known distributions of Linux with which Asterisk is known not to work? Seth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help:could asterisk work with other sip proxy?
Help:could asterisk work with other sip proxy? Different Enterprise asterisk PBX want to contact each other. could I use other manufacturer sip proxy contact different enterprise asterisk PBX ? B.R. Zhao Zigang. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Multicast Support desperately needed :: Mission critical bug in Asterisk
Friends! I have recently discovered that chan_sip, chan_sip2 and chan_sipx all lack support of SIP multicast. This has a major impact on my network, since I haven't got the bandwidth needed to call all of you and send you this message. With that feature missing, I have to go back to old Internet communication methods and use e-mail instead: *---* * I wish all of you a Merry Christmas and a Happy New Year! * *---* In the Stockholm area of Sweden we almost have no snow. I have no problem with that, but the kids miss the snow a bit. Real Christmas spirit seems to require snow. For me, it requires a lot of good food :-) We celebrate Christmas today. Seems like we belong to the Santa Platinum Membership Card Club and get priority over the countries that is included in the gift distribution activities tomorrow. Have a nice Christmas! Send your business phone into an endless Christmas torture loop and go offline, enjoy an Asterisk-hacking-free holiday and enjoy the holiday with friends and family! Best regards, /Olle PS: No, I don't believe IAX2 has the multicast support either... :-) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Record() problem
You syntax for the command is incorrect. See http://www.voip-info.org/wiki-Asterisk+cmd+record. Record is an application to be executed from within the dialplan. So the channel it will record is implicit and cannot be explicitly stated as one of the parameters. If you want to originate and record a call automatically, you will have to do this via AGI. Bill Seddon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Me Sent: December 24, 2004 6:38 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Record() problem Any idea why this: Record(IAX2/[EMAIL PROTECTED]/5, /tmp/whatever.gsm|6|25) Would result in this: WARNING[3293201]: app_record.c:117 record_exec: No extension found Thanks! -- Start Your Own ISP! http://www.YourOwnISP.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Record() problem
Hi all, I want to install 4 to 6 lines for our new office and hook it into an * system. I want as little trouble as possible, what fxo hardware do you recommend? I see that poeple on the list are complaining about digium tdm400 cards...? Are grandstream phones stable and easy to setup? Any problems with *? Thanks, Kobus Wolvaardt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Record() problem
Kobus Wolvaardt wrote: Hi all, I want to install 4 to 6 lines for our new office and hook it into an * system. I want as little trouble as possible, what fxo hardware do you recommend? I see that poeple on the list are complaining about digium tdm400 cards...? Are grandstream phones stable and easy to setup? Any problems with *? As was mentioned in an earlier post on a different topic, the squeaking wheels are easier to hear on the list than the satisfied users are. I have five installations using various combinations of iaxys, TDMX00s, and X101s, every day, and everyone is happy. Grandstreams are a love them or hate them proposition. I am in an odd middle: I know they sound a bit shitty, but I am looking to live in a future where audio phone conversations don't hog bandwidth the way that TDM does now. So I use the BT101 with iLBC for all of my calls, and while none of my co-callers think it sounds like I'm in the room with them, all of them say my speech is perfectly intelligible. I'm placing all my calls on an instance of asterisk running on a $70 Linksys WRT54GS, too, for the same reason. There are many, many choices. Go to www.voip-info.org and read up. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Xmas ;-)
Hello list, the guys here at Xcept wanted to say Merry Christmas to the list and the people who make Asterisk possible, so we put together this little Asterisk-based Christmas app... Justr dial IAXtel 1-700-444-6295 and be happy :-) Yours, lenz, laura and marco the crew at Xcept -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
