[Asterisk-Users] AGI - Freakin Lost
Ok, I have spent that past 4 - 5hrs working (trying to) figure some AGI syntax out in perl. Maybe I'm Looney / slow or what I don't know, but I am lost. I need to figure out how to send STDIN into the script. I understand the concept of it, but lost when it comes down to it. No matter what I try with standard stdin The app completes on me before I can ever enter in keys. I have tried to use the [WAIT FOR DIGIT] application, but that does not appear to get me anywhere. If I could get some pointers, just to get me going that would be great. I want to be able to have the keys entered on the phone and the ASCII text come back to me where I can manipulate that data. I'm sure it's simple, I just not getting the whole (big) picture Any help will be greatly appreciated -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] door phone
I have this http://www.doorbellfon.com installed on fxo ports supports 2 doors $300us for -1 door box, 1 ctrl panel, 130 -2 Outbound Relay Trigger Controller to open standard electric locks, 100 -1xtra door box, 50 http://www.marko.net/asterisk/archives/0209/0077.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI - Freakin Lost
On Thu, 27 Nov 2003, PBX wrote: I have spent that past 4 - 5hrs working (trying to) figure some AGI syntax out in perl. Maybe I'm Looney / slow or what I don't know, but I am lost. I need to figure out how to send STDIN into the script. I understand the concept of it, but lost when it comes down to it. No matter what I try with standard stdin The app completes on me before I can ever enter in keys. I have tried to use the [WAIT FOR DIGIT] application, but that does not appear to get me anywhere. If I could get some pointers, just to get me going that would be great. I want to be able to have the keys entered on the phone and the ASCII text come back to me where I can manipulate that data. I'm sure it's simple, I just not getting the whole (big) picture Any help will be greatly appreciated have you tried citats asterisk-perl package? http://asterisk.gnuinter.net - wasim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI - Freakin Lost
http://asterisk.gnuinter.net Jeremy McNamara PBX wrote: Ok, I have spent that past 4 - 5hrs working (trying to) figure some AGI syntax out in perl. Maybe I'm Looney / slow or what I don't know, but I am lost. I need to figure out how to send STDIN into the script. I understand the concept of it, but lost when it comes down to it. No matter what I try with standard stdin The app completes on me before I can ever enter in keys. I have tried to use the [WAIT FOR DIGIT] application, but that does not appear to get me anywhere. If I could get some pointers, just to get me going that would be great. I want to be able to have the keys entered on the phone and the ASCII text come back to me where I can manipulate that data. I'm sure it's simple, I just not getting the whole (big) picture Any help will be greatly appreciated -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancellation
Peter Zeltins wrote: I do not have a hardphone to play around with, but the echo is there both with built-in audio card (SigmaTel) and Bluetooth headset. There are no mixer settings than I can adjust as well. I'll try disabling AGC and/or lowering mike sensitivity. Peter According to the excellent Cisco paper Echo Analysis for Voice over IP, which the Wiki links to at: http://www.cisco.com/en/US/tech/tk652/tk701/technologies_white_paper09186a00800d6b68.shtml Headsets are particularly notorious for poor echo performance.. This is due to lack of acoustic isolation. Perhaps you could test using headphones and a mic. After reading the above paper, I was able to tune my setup and make a significant improvement. Given the high number of questions about echo on the list, it would almost be worth including the link to it as a file, README.echo in the source. Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Distinctive ring confusion
Thanks for all the help and I found the different cadences in chan_zap.c. Richard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Fields
Walker Haddock wrote: On Wed, Nov 26, 2003 at 08:33:13PM +0100, Olle E. Johansson wrote: Asterisk wrote: Hello! Does anyone know where I can find out about the CDR fields? I know most of them are self expiatory, but what is disposition for? I've done a search in Google, I even went to dictionary.com to check the meaning of the word, but I don't know why it always equals 4 in my CDR! http://www.voip-info.org/wiki-Asterisk+cdr+csv Explanations found in the source code, actually :-) /O Actually, I find that the ones on the wiki are not quite right for the Master.csv file. My fault, I documented the structure in memory. The actual printout to file was in another order. I've updated the page. Also, for text files, you have to enable uniqeid in the source code. It's disabled by default. I don't know how it works for database CDR logging. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIMPLE support in Asterisk?
Ok Yea, cause I used both Kphone and Windows messenger, and they successfully registered (and subscribed i think) towards asterisk. Using Kphone I even get a online status on all other users on the asterisk but no interaction with status or IM. So maybe there is some quasi presence avaible? I think it would be a great tool to support IM/Presence. There is so much that can be done with such implementations. rgds, /staffan kerker On Thu, 2003-11-27 at 04:53, John Todd wrote: Hi Is there any work being done on implementing IM/SIMPLE support for SIP on Asterisk? Like a presence server? rdgs, /Staffan Kerker No. There are currently requests in the system for that functionality (http://bugs.digium.com/bug_view_page.php?bug_id=134) but it's waiting for a White Knight to ride up and code a solution. However, there are some quasi-presence tools that appear to be built into Asterisk in ways that nobody has explained yet. I wouldn't use the term secret but the total lack of documentation and/or answers to the questions on how to use these features makes me wonder... JT Date: Thu, 16 Oct 2003 03:51:01 -0500 To: asterisk-users-lists.digium.com From: John Todd [EMAIL PROTECTED] Subject: Use of the hint modifiers - examples, anyone? Cc: Bcc: X-Attachments: I have found some references to the hint (or HINT?) variable and method in the source code, but quite a bit of Google-ing has not turned up any extensive answers as to some real-life examples of how to use this perhaps very useful tool. I understand the point of the tool, but I need to get some actual configs to look at before I think I'll figure it out. Even if my particular equipment doesn't support it, there may be other ideas I can get from it. (JerJer - maybe SCCP could use that data if there is an SCCP command of similar nature to the SIP SUBSCRIBE command - that would be pretty handy for those 7914 operator stations.) Searching through the source gives tantalizing hints (no pun intended) in pbx.c, but no actual real-life samples. Can someone who is familiar with it put some words to the features? I found this from March 20, 2003 from Andre Bierwirth: Subject: [Asterisk-Dev] Logged in users To: [EMAIL PROTECTED] I am currently work on it. If i am ready Asterisk have functions to get = device or extension state. int ast_extension_state(struct ast_channel *c, char *context, char = *exten) returns=20 -1 =3D error or no hint(device hint) for extension 0 =3D extension is free or unknown 1 =3D one device in extension is busy (have a call) 2 =3D all devices in extension unavailable(unregistered) ** You can give ast_device_state a Dialstring like SIP/mark or IAX/mark = ** int ast_device_state(char *device) returns -1 =3D error 0 =3D device is free or unknown 1 =3D device is busy (have a call) 2 =3D device is valid but unregistered So SIP can support SUBSCRIBE requests, and for Snom200 SUBSCIBE Dialogs = (Map a Key to an extension and see if the extension have a call (the LED = turned on)) Its easy to implement the device state support for IAX, i have talk with = mark about it. I implement only the PBX and Channel and SIP functions. With IAX you can poll the dialplan and get the extension states if its = implemented. Andre --- JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Staffan Kerker AerotechTelub AB, Communications [EMAIL PROTECTED] ph. +46(0)47042185 cell. +46(0)705391365 -- Don't get involved in politics man, just play the gig... /Sgt Floyd, Electric Mayhem Band ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI - Freakin Lost
Thank you.. I had found this package earlier in the evening... I guess I just need a Big Jolt of Caffine... I got it rollin now... Thanks again -gcc -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy McNamara Posted At: Thursday, November 27, 2003 2:06 AM Posted To: Asterisk User Group Conversation: [Asterisk-Users] AGI - Freakin Lost Subject: Re: [Asterisk-Users] AGI - Freakin Lost http://asterisk.gnuinter.net Jeremy McNamara PBX wrote: Ok, I have spent that past 4 - 5hrs working (trying to) figure some AGI syntax out in perl. Maybe I'm Looney / slow or what I don't know, but I am lost. I need to figure out how to send STDIN into the script. I understand the concept of it, but lost when it comes down to it. No matter what I try with standard stdin The app completes on me before I can ever enter in keys. I have tried to use the [WAIT FOR DIGIT] application, but that does not appear to get me anywhere. If I could get some pointers, just to get me going that would be great. I want to be able to have the keys entered on the phone and the ASCII text come back to me where I can manipulate that data. I'm sure it's simple, I just not getting the whole (big) picture Any help will be greatly appreciated -gcc ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unixodbc-vm-routines.h
Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=586 woop... Anyone wish to test and or make this better? (I know some of the code can be put into functions) bkw Do you think this will be merged into the CVS (seeing as its based on LGPL) or will it be an addon? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP problem
Hi all, I haveVOIP network built with MGCP endpoints.The manufacturer of endpoints is ASKEY. I downloaded latest Asterisk software and found it very useful for me. I configured it and it seems taht everything works OK when I am testing it with one or two endpoints. After that I tried to move Asterisk to working network and replace existing call manager. It starts working and calls are proceeding but after a while I could not hear a dialtone and saw in logs the following: Nov 27 11:40:57 WARNING[10251]: File chan_mgcp.c, Line 2127 (handle_hd_hf): Unable to create switch thread: Interrupted system call I looked in chan_mgcp.c file and saw that this error occures after pthread_create functions and it means that this "system call was interrupted permaturely with a signal before it was able to complete". Please, help me to resolve this problem. Best regards, Sergi Gabunia
Re: [Asterisk-Users] Crashed Asterisk
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 27 November 2003 05:57, Patrick Cantwell wrote: Just for the record, I'm using a slackware 9.0 and a slackware 9.1 box with asterisk, and I had no problems obtaining, building/compiling, or running asterisk with a fresh install. I'll back that up. It's all Slack here :) The only problem we've experienced is with the g729 codec which is currently still unusable (makes Asterisk segfault) - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/xbym2TEAILET3McRAtPZAJwJa3rrrsB0Z/2ESLL6lZgIKqjgCgCfTQ6a 08kczCt1NG+nypEfKiTRe2w= =dj01 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] App queue and all Agent busy
I have a queue defined as [blabla] member = SIP/101 member = SIP/102 and in extensions.conf this: exten = 101,1,Queue(blabla,t) exten = 101,102,Congestion but when both Agents are busy then still the called party does not get a busy signal. What I`d want is when both Agents are busy that the caller gets a busy, not the long tones, like the phone is ringing, but nobody answers it ( this is how this works now) Any Ideas? Thanks in advance -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP problem
Hi, I forgot tosay that I have about 300 MGCP endpoints in my real network. Best regards, Sergi Gabunia - Original Message - From: Sergi Gabunia To: [EMAIL PROTECTED] Sent: Thursday, November 27, 2003 12:05 PM Subject: [Asterisk-Users] MGCP problem Hi all, I haveVOIP network built with MGCP endpoints.The manufacturer of endpoints is ASKEY. I downloaded latest Asterisk software and found it very useful for me. I configured it and it seems taht everything works OK when I am testing it with one or two endpoints. After that I tried to move Asterisk to working network and replace existing call manager. It starts working and calls are proceeding but after a while I could not hear a dialtone and saw in logs the following: Nov 27 11:40:57 WARNING[10251]: File chan_mgcp.c, Line 2127 (handle_hd_hf): Unable to create switch thread: Interrupted system call I looked in chan_mgcp.c file and saw that this error occures after pthread_create functions and it means that this "system call was interrupted permaturely with a signal before it was able to complete". Please, help me to resolve this problem. Best regards, Sergi Gabunia
Re: [Asterisk-Users] Crashed Asterisk
slack here too - * is working STABLE Lubo Tais M. Hansen wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 27 November 2003 05:57, Patrick Cantwell wrote: Just for the record, I'm using a slackware 9.0 and a slackware 9.1 box with asterisk, and I had no problems obtaining, building/compiling, or running asterisk with a fresh install. I'll back that up. It's all Slack here :) The only problem we've experienced is with the g729 codec which is currently still unusable (makes Asterisk segfault) - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/xbym2TEAILET3McRAtPZAJwJa3rrrsB0Z/2ESLL6lZgIKqjgCgCfTQ6a 08kczCt1NG+nypEfKiTRe2w= =dj01 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and voice recog support ?
Hi, Calling the FWD, i see a feature a little different. I don't call any number, but TALK with the system and they go to others parts of the showed menu. There are any way to make the same with * ? Where are the link that i'm talking about. http://fwd.pulver.com/callme.php?userid=5 Thanks Carlos. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Express Router Asterisk
Hi, We will shortly launch a sip service. Architecture is: SER: for SIP registration and IP call routing, incoming number termination, STUN, Nat traversal etc. Asterisk: outgoing call routing, calling card platform, billing, extended facilities e.g. voicemail etc. Works well. Tan Telappliant.com Voiptalk.org -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Tinchev Sent: 27 November 2003 13:19 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SIP Express Router Asterisk There was some issues with Audiocodes MP10x - With both Asterisk and SER. It was fixed in last firmwire release. Hope it is fixed in Mediant too. It was general SIP issues... Ryan Tucker wrote: Greetings... We've been having some interoperability issues between Asterisk and an AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 somewhere. So, I've been pondering using iptel.org's SIP server (SIP Express Router) as a front end for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or anything at all involving SER and Asterisk? Thanks! -rt ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Crash - What is happening here???
The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting simple switch on 'Zap/32-2' -- Started three way call on channel 32 -- Started music on hold, class 'default', on Zap/6-1 -- Executing Macro(Zap/32-2, stdexten2|6112|Zap/34|20|tT) in new stack -- Executing SetLanguage(Zap/32-2, nl) in new stack -- Executing DBget(Zap/32-2, fwdexten=FEAT/6112/CFWD/CFU) in new stack -- DBget: varname=fwdexten, family=FEAT, key=6112/CFWD/CFU -- DBget: Value not found in database. -- Executing Goto(Zap/32-2, s|5) in new stack -- Goto (macro-stdexten2,s,5) -- Executing Dial(Zap/32-2, Zap/34|20|tT) in new stack -- Called 34 -- Zap/34-1 is ringing -- Zap/34-1 is ringing -- Stopped music on hold on Zap/6-1 -- Hungup 'Zap/6-1MASQ' -- Hungup 'Zap/32-1' == Spawn extension (netland_admin, s, 3) exited non-zero on 'Zap/32-2ZOMBIE' -- Executing Macro(Zap/32-2ZOMBIE, record-cleanup) in new stack -- Executing SetVar(Zap/32-2ZOMBIE, MONITORDIR=/var/spool/asterisk/monitor) in new stack -- Executing GotoIf(Zap/32-2ZOMBIE, 1?5:3) in new stack -- Goto (macro-record-cleanup,s,5) -- Executing NoOp(Zap/32-2ZOMBIE, ) in new stack -- Executing ChanIsAvail(Zap/32-2ZOMBIE, Zap/32) in new stack -- Hungup 'Zap/32-1' -- Executing Dial(Zap/32-2ZOMBIE, Zap/32|40|tm) in new stack -- Called 32 -- Started music on hold, class 'default', on Zap/32-2ZOMBIE -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/34-1 is ringing -- Zap/34-1 answered Zap/6-1 -- Attempting native bridge of Zap/6-1 and Zap/34-1 -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 answered Zap/32-2ZOMBIE -- Stopped music on hold on Zap/32-2ZOMBIE n010205*CLI Disconnected from Asterisk server ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Symmetric RTP
Yes. Jan. On 26-11 22:16, Olle E. Johansson wrote: Anyone that knows if the Asterisk SIP channel supports symmetric RTP? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] App queue and all Agent busy
Hi! but when both Agents are busy then still the called party does not get a busy signal. What I`d want is when both Agents are busy that the caller gets a busy, not the long tones, like the phone is ringing, but nobody answers it ( this is how this works now) Any Ideas? Method 1: Configure the queue to only take two callers Method 2: Don't use queues at all and instead use a simple Dial() in extensions.conf where you dial both agents extensions like exten = 1000,1,Dial(SIP/johnSIP/mary) exten = 1000,2, ... unavailable ... exten = 1000,102, ... busy ... Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] door phone
I posted my solution yesterday and it is $50 so not sure why people are still asking for a cheap solution. I have a cheap disposable walmart phone on the channel bank, in immediate mode so when it is picked up it jumps immediately to the default context. I can also dial it like any other extension to talk to people at the door. I have a $20 electric strike on the door which is coupled to my burglar alarm (that I can use a keypad outside to open right now) to interface to asterisk all I need is the following about $5 worth of stuff. 10k resistor from one of the data lines on the parallel port to base of a 2n transistor, emitter of to ground/earth, collector to a coil of a 12vdc pcb mount relay, 1n400x diode in inverse parallel with the coil to kill spikes, other end of the diode/coil to 12vdc from a disk drive connector in the pc. normally open contacts of the coil in parallel with whatever else drives the door strike (normally a 12-24AC/DC supply - AC if you like that Buzz, DC if a click is more your liking.) Make a simple agi script either using system command or in C - output a 1 to the bit of the parport, wait 5seconds, output a 0. Done. Much less than $50 - parts avail - Radio Shack or Dick Smith (are they still around in AU ?) Digikey is another fine source. [EMAIL PROTECTED] wrote: Hi, Anyone know anything about Asterisk's support for door phones? Receiving the call from the door intercom system, opening the door, etc? Any hardware recommendations? I understand that the equipment we have now is Panasonic proprietary and came with the currently deployed Panasonic TD12-32 pbx. We intend to deploy Asterisk in a 72 extensions + 16 trunks in a while, so any info will be great. thanks While this isn't an Asterisk question, I guess, I find it hard not to comment on a particularly excellent piece of equipment which which I've worked previously: John Todd said: Can be ordered from Grainger (at least, two years ago it could) here in North America under part number 4RR12 Emergency Access Phone for $623.50. I'm sure you can find it elsewhere, too. Ummm, you might go to that level for a large office/block of units or campus environment etc, but I was looking for something for home (mainly just for the coolness factor). Something where I can sit in my car in the pouring rain, dial into asterisk from my mobile, enter some pin code etc, have the door open, and then dash to the door with some bags/boxes etc and not have to fumble for keys. Another thought is the possibility of having small cameras mounted around the home linked to a linux box. Someone turns up and presses the intercom, since you are 'logged out' asterisk forwards the call to your mobile (or work phone etc). You answer the call, log onto your webcam, and let them in. You now watch what they are doing until they leave. Of course, you might not *really* want to do this in practice else you may end up with 'blind spots' and it will still take you a really long time to get there and try and stop them from doing whatever it is they are doing... Of course, it might be your girlfriend/mother/etc in which case you hopefully trust them a little. So, anyone got a solution for under AUD$100 ? Surely this is really just a bunch of cheap/commodity electronic components? Regards, Adam -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Express Router Asterisk
I just fixed a problem with Asterisk where it would not fill in all the headers correctly in the 407 - Proxy Authenication Required message that was causing some sip phones not to work with Asterisk. That fix might fix your problem as well. :) Have a good one. --Greg On Sun, 23 Nov 2003 20:47:05 -0500, Ryan Tucker wrote: Greetings... We've been having some interoperability issues between Asterisk and an AudioCodes Mediant 2000, and, well, I gotta use the Mediant 2000 somewhere. So, I've been pondering using iptel.org's SIP server (SIP Express Router) as a front end for PSTN calls going out to the Mediant, while using Asterisk for everything else. Has anyone done something similar, or anything at all involving SER and Asterisk? Thanks! -rt -- Ryan Tucker Network Engineer NetAccess, Inc. 1159 Pittsford-Victor Road Bldg. 5, Suite 140 Pittsford, New York 14534 585-419-8200 www.netacc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users - Greg Varga Author for RocketryNews http://www.rocketrynews.com CAR # 677 - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Symmetric RTP
Jan Janak wrote: On 26-11 22:16, Olle E. Johansson wrote: Anyone that knows if the Asterisk SIP channel supports symmetric RTP? Yes. Followup question: Both as a SIP UA (Client) and as a SIP proxy? /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Symmetric RTP
On 27-11 15:14, Olle E. Johansson wrote: Jan Janak wrote: On 26-11 22:16, Olle E. Johansson wrote: Anyone that knows if the Asterisk SIP channel supports symmetric RTP? Yes. Followup question: Both as a SIP UA (Client) and as a SIP proxy? I don't know, I tried asterisk as a SIP UA behind a NAT with SER in the public internet and it worked. So at least the SIP UA part works. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Symmetric RTP
Jan Janak wrote: On 27-11 15:14, Olle E. Johansson wrote: Jan Janak wrote: On 26-11 22:16, Olle E. Johansson wrote: Anyone that knows if the Asterisk SIP channel supports symmetric RTP? Yes. Followup question: Both as a SIP UA (Client) and as a SIP proxy? I don't know, I tried asterisk as a SIP UA behind a NAT with SER in the public internet and it worked. So at least the SIP UA part works. I remember that you called my inside Asterisk. But I can't find support for symmetric RDP in the SIP channel SDP parser. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] App queue and all Agent busy
Philipp von Klitzing wrote:
Hi!
but when both Agents are busy then still the called party does not get a
busy signal.
What I`d want is when both Agents are busy that the caller gets a busy,
not the long tones, like the phone is ringing, but nobody answers it (
this is how this works now)
Any Ideas?
Method 1: Configure the queue to only take two callers
what would callers get when the queue is full? that is still not such a
good solution as there may be a variable number of operators.
Method 2: Don't use queues at all and instead use a simple Dial() in
extensions.conf where you dial both agents extensions like
exten = 1000,1,Dial(SIP/johnSIP/mary)
exten = 1000,2, ... unavailable ...
exten = 1000,102, ... busy ...
Operators have a wrapup to do and such.
I`ve looked at the source of app_dial. and app_queue.c, and I found such
a piece of code in app_dial.c:
if (found 0) {
if (numlines == numbusies) {
if (option_verbose 2)
ast_verbose( VERBOSE_PREFIX_2
Everyone
is busy at this time\n);
/* See if there is a special busy message */
if (ast_exists_extension(in,
in-context, in-exten, in-priority + 101, in-callerid))
in-priority+=100;
} else {
if (option_verbose 2)
ast_verbose( VERBOSE_PREFIX_2
No one is available to answer at this time\n);
}
*to = 0;
notice the part about priority, there is an almost identical code in
app_queue.c:
if (found 0) {
if (numlines == numbusies) {
ast_log(LOG_DEBUG, Everyone is busy at
this time\n);
} else {
ast_log(LOG_NOTICE, No one is answered
queue %s\n, queue);
}
*to = 0;
almost the same without the part:
/* See if there is a special busy message */
if (ast_exists_extension(in,
in-context, in-exten, in-priority + 101, in-callerid))
in-priority+=100;
would it work if I merged them and code in app_queue.c looke like:
if (found 0) {
if (numlines == numbusies) {
ast_log(LOG_DEBUG, Everyone is busy at
this time\n);
/* See if there is a special busy message */
if (ast_exists_extension(in, in-context,
in-exten, in-priority + 101, in-callerid))
in-priority+=100;
} else {
ast_log(LOG_NOTICE, No one is answered
queue %s\n, queue);
}
*to = 0;
this is a production system and it better to know before I go in with
hacked code.
This would be a very good functionality, I can setup a prompt, like all
operators are busy and such.
thanks
Cheers, Philipp
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[Asterisk-Users] distinctive ring doesn't work
In my extensions.conf file I'm attempting to distinctively ring one of my zap channels with a different ring depending upon whether the call is received from the DID or inside extension. The DID extension looks like so: exten = 5551236543,1,Dial,Zap/28r1|20 However, when I dial in on the DID the phone is not ringing at all. The CLI shows that Zap/28 is ringing and you can hear it ringing on the outside line you call from but you can't hear it on the actual phone that is being dialed. It then goes to voicemail. Can anyone tell me what I am doing wrong here? Thanks AJ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attempting to get SJPhone configured for Asterisk- Help!
[EMAIL PROTECTED] wrote: My problem is the tutorial left out how to configure a SJPhone so that it connects to my asterisk server not directly FWD. I've tried everything I can think of, I must be missing something simple. I'm using SJPhone with the following config: sip.conf: [markspc] type=friend host=dynamic dtmfmode=inband callerid=Mark's PC 1 username=markspc SJPhone (SIP tab): Use local outbound proxy - checked. Proxy IP Address: 192.168.0.1 Caller ID: sip:[EMAIL PROTECTED] Register - checked. Account: markspc Password blank. Everything else is default (haven't changed the Advanced SIP Options or anything.) Of course, in production you'd want to add a secret line to sip.conf and the corresponding password to SJPhone. When it's working, SJPhone shows: Status: no active calls Default protocol: SIP SIP Proxy: registered with 192.168.0.1 Host address: 192.168.0.2 and Asterisk's console says: Registered SIP 'markspc' at 192.168.0.2 Asterisk also periodically reports: Got SIP response 481 Subscription does not exist back from 192.168.0.2 which seems harmless. HTH, Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Symmetric RTP
Have you tried SER to * in the same setup? David -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Jan Janak Verzonden: donderdag 27 november 2003 15:26 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] Symmetric RTP On 27-11 15:14, Olle E. Johansson wrote: Jan Janak wrote: On 26-11 22:16, Olle E. Johansson wrote: Anyone that knows if the Asterisk SIP channel supports symmetric RTP? Yes. Followup question: Both as a SIP UA (Client) and as a SIP proxy? I don't know, I tried asterisk as a SIP UA behind a NAT with SER in the public internet and it worked. So at least the SIP UA part works. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Symmetric RTP
I tested the following scenario: private network| public internet SIP Phone 1 --- Asterisk --- NAT --- SER --- SIP Phone 2 and it worked. I was able to make calls from phone 1 to phone 2 and vice versa. Jan. On 27-11 16:37, David Luyens wrote: Have you tried SER to * in the same setup? David -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Jan Janak Verzonden: donderdag 27 november 2003 15:26 Aan: [EMAIL PROTECTED] Onderwerp: Re: [Asterisk-Users] Symmetric RTP On 27-11 15:14, Olle E. Johansson wrote: Jan Janak wrote: On 26-11 22:16, Olle E. Johansson wrote: Anyone that knows if the Asterisk SIP channel supports symmetric RTP? Yes. Followup question: Both as a SIP UA (Client) and as a SIP proxy? I don't know, I tried asterisk as a SIP UA behind a NAT with SER in the public internet and it worked. So at least the SIP UA part works. Jan. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Crash - What is happening here???
Small tutorial: these errors are too generic to be solved in such way... hey my asterisk crashed, why it did?... there're many reasons... First: set ulimit -c unlimited on the console from which * starts, to let it dump cores. Then start it with 'g' in his parms , like asterisk -vvvgc, to enable debugging... then when it crashed, run gdb on the core and backtrace it also: try to find a way to reproduce the crash. random crashed aren't very useful... and... report also asterisk version, kernel, distro, blah blah blah Michiel, that message isn't only for you, but your post triggered my thoughts to how to report a crash, for anyone that just jump on th ML and say my asterisk crashed. please say me why... bye, matteo Scrive Michiel Betel [EMAIL PROTECTED]: The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting simple switch on 'Zap/32-2' -- Started three way call on channel 32 -- Started music on hold, class 'default', on Zap/6-1 -- Executing Macro(Zap/32-2, stdexten2|6112|Zap/34|20|tT) in new stack -- Executing SetLanguage(Zap/32-2, nl) in new stack -- Executing DBget(Zap/32-2, fwdexten=FEAT/6112/CFWD/CFU) in new stack -- DBget: varname=fwdexten, family=FEAT, key=6112/CFWD/CFU -- DBget: Value not found in database. -- Executing Goto(Zap/32-2, s|5) in new stack -- Goto (macro-stdexten2,s,5) -- Executing Dial(Zap/32-2, Zap/34|20|tT) in new stack -- Called 34 -- Zap/34-1 is ringing -- Zap/34-1 is ringing -- Stopped music on hold on Zap/6-1 -- Hungup 'Zap/6-1MASQ' -- Hungup 'Zap/32-1' == Spawn extension (netland_admin, s, 3) exited non-zero on 'Zap/32-2ZOMBIE' -- Executing Macro(Zap/32-2ZOMBIE, record-cleanup) in new stack -- Executing SetVar(Zap/32-2ZOMBIE, MONITORDIR=/var/spool/asterisk/monitor) in new stack -- Executing GotoIf(Zap/32-2ZOMBIE, 1?5:3) in new stack -- Goto (macro-record-cleanup,s,5) -- Executing NoOp(Zap/32-2ZOMBIE, ) in new stack -- Executing ChanIsAvail(Zap/32-2ZOMBIE, Zap/32) in new stack -- Hungup 'Zap/32-1' -- Executing Dial(Zap/32-2ZOMBIE, Zap/32|40|tm) in new stack -- Called 32 -- Started music on hold, class 'default', on Zap/32-2ZOMBIE -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/34-1 is ringing -- Zap/34-1 answered Zap/6-1 -- Attempting native bridge of Zap/6-1 and Zap/34-1 -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 answered Zap/32-2ZOMBIE -- Stopped music on hold on Zap/32-2ZOMBIE n010205*CLI Disconnected from Asterisk server ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Matteo Brancaleoni Espia System Administrator http://www.espia.it This message was sent using IMP, the Internet Messaging Program. Service is provided by Espia - Emmegi Srl - http://www.espia.it. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help for oh323
Hi Friends,
Hope you would help me out here, I have searched the asterisk
user list for hours and also read the readme and test files that
comes with the driver. I need a very simple scenario. I have SIP
clients and want to use oh323 to dial out to PSTN using a h323 gateway.
a)If I set the extention.conf like this:
exten = _87.,1,Dial(OH323/16.52.153.206)
oh323 dials out (I can ring a netmeeting client at 16.52.153.206).
(b)But if I set it like this, oh323 will not dials out ?
exten = _87.,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED])
In summary what I am trying to achieve is the following;
Lets say Sip user dial 915105418168, then I want 9 to be dropped
and the extension information to be send to the g/w at
16.52.153.206. Isn't exten =
_9x,1,Dial(OH323/${EXTEN:[EMAIL PROTECTED]) is the right
way ?. Why is this not working ?
I must be doing a wrong syntaxt, but couldnt find where I go wrong.
I am attaching the trace for above two cases, please help ?
Cheers
Sathya
Traces for both cases are given below;
0:00.076 OpenH323 Wrapper OpenH323 WrapperVersion
0.0alpha
0 by inAccess Networks (www.inaccessnetworks.com) on Unix Linux
(2.4.20-8-i686)
at 2003/11/26 20:15:07.622
0:00.078 OpenH323 Wrapper H323Created endpoint.
0:00.078 H323 Cleaner H323Started cleaner thread
0:00.212 OpenH323 Wrapper H323Started listener
Listener[ip$*:1720]
0:00.214H323 Listener:8115c18 H323Awaiting TCP connections on
port 1720
0:00.214 OpenH323 Wrapper H323UDP Binding to interface:
0.0.0.0:1
0:00.243 OpenH323 Wrapper H323Added capability:
G.711-ALaw-64k{hw} 1
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/hookflash 2
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/basicString 3
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/dtmf 4
0:00.279 OpenH323 Wrapper H323Added capability:
UserInput/RFC2833 5
4:00.687 ThreadID=0x4a774440 H323Making call to:
[EMAIL PROTECTED]
4:00.688 ThreadID=0x4a774440 H323Attempt to use invalid URL
[EMAIL PROTECTED]
4:00.688 ThreadID=0x4a774440 H323Could not parse
[EMAIL PROTECTED]
4:00.757 ClearCallT...d:0812ab10 H323Attempt to clear unknown
call
8:14.840 ThreadID=0x4a774440 H323Making call to:
16.52.153.206
8:14.904 ThreadID=0x4a774440 H323Added capability:
G.711-ALaw-64k{hw} 1
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/hookflash 2
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/basicString 3
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/dtmf 4
8:14.905 ThreadID=0x4a774440 H323Added capability:
UserInput/RFC2833 5
8:14.905 ThreadID=0x4a774440 H323Found capability:
G.711-ALaw-64k{hw} 1
8:14.905 ThreadID=0x4a774440 H323Found capability:
UserInput/hookflash 2
8:14.905 ThreadID=0x4a774440 H323Found capability:
UserInput/basicString 3
8:14.906 ThreadID=0x4a774440 H323Found capability:
UserInput/dtmf 4
8:14.906 ThreadID=0x4a774440 H323Found capability:
UserInput/RFC2833 5
8:14.906 ThreadID=0x4a774440 RFC2833 Handler created
8:14.906 ThreadID=0x4a774440 H323Added capability:
G.711-ALaw-64k{hw} 1
8:14.907 ThreadID=0x4a774440 H323Created new connection:
ip$localhost/25259
8:14.908 H225 Caller:80f4688 H225Started call thread
8:15.064 H225 Caller:80f4688 H323TCP Could not connect to
16.52.153.2
06:1720 (local port=1) - No route to host(113)
8:15.065 H225 Caller:80f4688 H323Clearing connection
ip$localhost
/25259 reason=EndedByConnectFail
8:15.065 H225 Caller:80f4688 H323Call end reason for
ip$localhost
/25259 set to EndedByConnectFail
8:15.066 H225 Caller:80f4688 H225Sending release complete
PDU: ca
llRef=25259
8:15.200 H225 Caller:80f4688 H323Clearing connection
ip$localhost
/25259 reason=EndedByTransportFail
8:15.200 H323 Cleaner H323Cleaning up connections
8:15.201 H323 Cleaner H323Connection
ip$localhost/25259 cl
osing: connectionState=NoConnectionActive
8:15.201 H323 Cleaner H323H323Transport::Close
8:15.201 H323 Cleaner H323
H323Transport::CleanUpOnTerminat
ion for H225 Caller:80f4688
8:15.201 H323 Cleaner H323Connection
ip$localhost/25259 te
rminated.
8:15.203 H323 Cleaner H323Connection
ip$localhost/25259 de
leted.
8:15.204 H323 Cleaner H323Cleaning up connections
8:15.369 ClearCallT...d:080f1a48 H323Attempt to clear unknown
call ip
$localhost/25259
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Re: [Asterisk-Users] App queue and all Agent busy
Anton Yurchenko wrote:
I just wanted to say, that I patched the code like I wrote below, and it
works. When all the operators are busy, then it drops to priority + 101.
If that would break something please write me ASAP ;)
Philipp von Klitzing wrote:
Hi!
but when both Agents are busy then still the called party does not
get a busy signal.
What I`d want is when both Agents are busy that the caller gets a
busy, not the long tones, like the phone is ringing, but nobody
answers it ( this is how this works now)
Any Ideas?
Method 1: Configure the queue to only take two callers
what would callers get when the queue is full? that is still not such
a good solution as there may be a variable number of operators.
Method 2: Don't use queues at all and instead use a simple Dial() in
extensions.conf where you dial both agents extensions like
exten = 1000,1,Dial(SIP/johnSIP/mary)
exten = 1000,2, ... unavailable ...
exten = 1000,102, ... busy ...
Operators have a wrapup to do and such.
I`ve looked at the source of app_dial. and app_queue.c, and I found
such a piece of code in app_dial.c:
if (found 0) {
if (numlines == numbusies) {
if (option_verbose 2)
ast_verbose( VERBOSE_PREFIX_2
Everyone
is busy at this time\n);
/* See if there is a special busy
message */
if (ast_exists_extension(in,
in-context, in-exten, in-priority + 101, in-callerid))
in-priority+=100;
} else {
if (option_verbose 2)
ast_verbose( VERBOSE_PREFIX_2
No one is available to answer at this time\n);
}
*to = 0;
notice the part about priority, there is an almost identical code in
app_queue.c:
if (found 0) {
if (numlines == numbusies) {
ast_log(LOG_DEBUG, Everyone is busy at
this time\n);
} else {
ast_log(LOG_NOTICE, No one is answered
queue %s\n, queue);
}
*to = 0;
almost the same without the part:
/* See if there is a special busy
message */
if (ast_exists_extension(in,
in-context, in-exten, in-priority + 101, in-callerid))
in-priority+=100;
would it work if I merged them and code in app_queue.c looke like:
if (found 0) {
if (numlines == numbusies) {
ast_log(LOG_DEBUG, Everyone is busy at
this time\n);
/* See if there is a special busy message */
if (ast_exists_extension(in, in-context,
in-exten, in-priority + 101, in-callerid))
in-priority+=100;
} else {
ast_log(LOG_NOTICE, No one is answered
queue %s\n, queue);
}
*to = 0;
this is a production system and it better to know before I go in with
hacked code.
This would be a very good functionality, I can setup a prompt, like
all operators are busy and such.
thanks
Cheers, Philipp
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Re: [Asterisk-Users] Crash - What is happening here???
Matteo, I AM running -gc and ulimit -c unlimited (from safe_asterisk) on RH7.2 Thats the weird thing... it crashed without any message. And looking through the source I still don't see how the Dial could start on a Zombie channel... But you are right, I'll try to reproduce it tomorrow morning (Its a production system) Michiel Matteo Brancaleoni wrote: Small tutorial: these errors are too generic to be solved in such way... hey my asterisk crashed, why it did?... there're many reasons... First: set ulimit -c unlimited on the console from which * starts, to let it dump cores. Then start it with 'g' in his parms , like asterisk -vvvgc, to enable debugging... then when it crashed, run gdb on the core and backtrace it also: try to find a way to reproduce the crash. random crashed aren't very useful... and... report also asterisk version, kernel, distro, blah blah blah Michiel, that message isn't only for you, but your post triggered my thoughts to how to report a crash, for anyone that just jump on th ML and say my asterisk crashed. please say me why... bye, matteo Scrive Michiel Betel [EMAIL PROTECTED]: The following transfer led to a crash of asterisk, without leaving a core or any utterances in messages or debug file. It looks like the zombie which was created during the MASQ-transfer was not cleaned up... But why did it start a Dial??? And... why does Asterisk die when this happens?? Thanks!!! Michiel -- Zap/32-1 answered Zap/6-1 -- Stopped music on hold on Zap/6-1 -- Starting simple switch on 'Zap/32-2' -- Started three way call on channel 32 -- Started music on hold, class 'default', on Zap/6-1 -- Executing Macro(Zap/32-2, stdexten2|6112|Zap/34|20|tT) in new stack -- Executing SetLanguage(Zap/32-2, nl) in new stack -- Executing DBget(Zap/32-2, fwdexten=FEAT/6112/CFWD/CFU) in new stack -- DBget: varname=fwdexten, family=FEAT, key=6112/CFWD/CFU -- DBget: Value not found in database. -- Executing Goto(Zap/32-2, s|5) in new stack -- Goto (macro-stdexten2,s,5) -- Executing Dial(Zap/32-2, Zap/34|20|tT) in new stack -- Called 34 -- Zap/34-1 is ringing -- Zap/34-1 is ringing -- Stopped music on hold on Zap/6-1 -- Hungup 'Zap/6-1MASQ' -- Hungup 'Zap/32-1' == Spawn extension (netland_admin, s, 3) exited non-zero on 'Zap/32-2ZOMBIE' -- Executing Macro(Zap/32-2ZOMBIE, record-cleanup) in new stack -- Executing SetVar(Zap/32-2ZOMBIE, MONITORDIR=/var/spool/asterisk/monitor) in new stack -- Executing GotoIf(Zap/32-2ZOMBIE, 1?5:3) in new stack -- Goto (macro-record-cleanup,s,5) -- Executing NoOp(Zap/32-2ZOMBIE, ) in new stack -- Executing ChanIsAvail(Zap/32-2ZOMBIE, Zap/32) in new stack -- Hungup 'Zap/32-1' -- Executing Dial(Zap/32-2ZOMBIE, Zap/32|40|tm) in new stack -- Called 32 -- Started music on hold, class 'default', on Zap/32-2ZOMBIE -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/34-1 is ringing -- Zap/34-1 answered Zap/6-1 -- Attempting native bridge of Zap/6-1 and Zap/34-1 -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 is ringing -- Zap/32-1 answered Zap/32-2ZOMBIE -- Stopped music on hold on Zap/32-2ZOMBIE n010205*CLI Disconnected from Asterisk server ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agent Logoff inability when calls are being received from queue
Hello everybody, I have started using Asterisk in a call center with ACD. I have noticed something and I wonder if anyone knows whether it is a bug or a feature! I am using Queue application to ring a number of agents that have logged on using AgentCallbackLogin. Now, while an agent receives a call from the Queue they cannot logoff using AgentCallbackLogin. Instead the Agent is asked for their agent no and their password and after that they get the agenty-alreadyon message!!! When the call goes away they are able to logoff!! It sounds trivial but in a busy call center where lines are coming all the time it gives the agent a hard time and at the same time the line is not being answered since the agent's extension is considered available and Asterisk keeps sending the call to the agent. By the way I am using Asterisk from CVS of August 17. Does anyone knows whether the above is meant to work like that? If it is a bug does anyone know whether it is being corrected in later CVS? Thanks in advance for any answer. Anna ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Attempting to get SJPhone configured for Asterisk- Help!
Olle E. Johansson [EMAIL PROTECTED] wrote: And now, Marks information on SJphone and Asterisk is appended to the Wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+phone+sjphone Thanks for posting that for me - I'm honored! :) I've touched it up a bit to improve my writing. I'm no Wiki pro, so maybe you could double-check it to make sure it's OK. Reddog4891, you may want to check the Wiki page above for more detail on the sample config and SJPhone. Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIMPLE support in Asterisk?
Yea, cause I used both Kphone and Windows messenger, and they successfully registered (and subscribed i think) towards asterisk. Using Kphone I even get a online status on all other users on the asterisk but no interaction with status or IM. So maybe there is some quasi presence avaible? I think it would be a great tool to support IM/Presence. There is so much that can be done with such implementations. SIMPLE could be added within chan_sip, but there is no mechanism within Asterisk to move text from one channel to another *without* the context of a call. *With* the context of a call, we definitely have such a thing (TEXT frames) Mark ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Crashed Asterisk
Assuming you haven't cvs updated yet I can look at this problem but I need matching sources/binaries/cores. If you've cvs updated, there isn't much I can do. Mark On Thu, 27 Nov 2003, Tais M. Hansen wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Thursday 27 November 2003 05:57, Patrick Cantwell wrote: Just for the record, I'm using a slackware 9.0 and a slackware 9.1 box with asterisk, and I had no problems obtaining, building/compiling, or running asterisk with a fresh install. I'll back that up. It's all Slack here :) The only problem we've experienced is with the g729 codec which is currently still unusable (makes Asterisk segfault) - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQE/xbym2TEAILET3McRAtPZAJwJa3rrrsB0Z/2ESLL6lZgIKqjgCgCfTQ6a 08kczCt1NG+nypEfKiTRe2w= =dj01 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Timeout feature in queues.conf does not seem to work
Hello again, I have noticed with Queues and roundrobin policy that if even if a timeout is set for a queue, Asterisk keeps ringing an available member of the queue after the timeout expires. This continues a few times before the next available agent is tried. I am using CVS of August 17 but I have read in the list that roundrobin worked fine since earlier in August. Does anyone know if this has been fixed in earlier CVS or is it supposed to work like this? outgoing agentcall, to agent '8601', on 'Local/[EMAIL PROTECTED],1' -- Executing Dial(Local/[EMAIL PROTECTED],2, OH323/h323:[EMAIL PROTECTED]) in new stack -- Called 8601 -- Called h323:[EMAIL PROTECTED] -- Nobody picked up in 15000 ms -- Hungup 'H323:17087' == Spawn extension (default, 4302, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' -- outgoing agentcall, to agent '8601', on 'Local/[EMAIL PROTECTED],1' -- Executing Dial(Local/[EMAIL PROTECTED],2, OH323/h323:[EMAIL PROTECTED]) in new stack -- Called 8601 -- Called h323:[EMAIL PROTECTED] -- Nobody picked up in 15000 ms -- Hungup 'H323:17088' == Spawn extension (default, 4302, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' -- outgoing agentcall, to agent '8601', on 'Local/[EMAIL PROTECTED],1' -- Executing Dial(Local/[EMAIL PROTECTED],2, OH323/h323:[EMAIL PROTECTED]) in new stack -- Called 8601 -- Called h323:[EMAIL PROTECTED] -- Stopped music on hold on H323:5404 == Spawn extension (vservices, 5512, 5) exited non-zero on 'H323:5404' -- Hungup 'H323:17089' == Spawn extension (default, 4302, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' -- Hungup 'H323:5404' Thanks in advance, Anna Anna Panagidou Technology Department Hellas On Line Agiou Konstantinou 59-61 15124, Maroussi Tel. no: (+30210) 8762309 E-mail address: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unixodbc-vm-routines.h
That was the whole reason I did this. Since the unixODBC stuff is LGPL we can side step all the drama. :P I still wanna clean it up a bit more bkw On Thu, 27 Nov 2003, WipeOut wrote: Brian West wrote: http://bugs.digium.com/bug_view_page.php?bug_id=586 woop... Anyone wish to test and or make this better? (I know some of the code can be put into functions) bkw Do you think this will be merged into the CVS (seeing as its based on LGPL) or will it be an addon? Later.. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Larger SIP packets
Hi all, We would like to increase the size (sample length) of RTP packets sent by Asterisk to SIP phones. I gather that Asterisk currently always uses 20ms packets for RTP, although I can't find in the source where that's defined, unless it's in chan_zap.c. I'm guessing from [http://lists.digium.com/pipermail/asterisk-users/2003-September/019490.html] that it's not possible right now (or at least wasn't in September). Does anyone know if and when it's coming as an option, or if we can modify the source to always use, say, 40ms packets instead of 20? Thanks in advance for any advice. Please CC me since I'm not subscribed to the list anymore (too much traffic). Cheers, Chris. -- _ __ __ _ / __/ / ,__(_)_ | Chris Wilson -- UNIX Firewall Lead Developer | / (_ ,\/ _/ /_ \ | NetServers.co.uk http://www.netservers.co.uk | \__/_/_/_//_/___/ | 21 Signet Court, Cambridge, UK. 01223 576516 | ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agent Logoff inability when calls are being received from queue
Hi! Now, while an agent receives a call from the Queue they cannot logoff using AgentCallbackLogin. Instead the Agent is asked for their agent no and their password and after that they get the agenty-alreadyon message!!! When the call goes away they are able to logoff!! I remember having read something about a patch/feature that would allow agent to go for a break/smoke/coffee, but I couldn't find it under ACD nor under Experimental features... you might want to search Mantis yourself: http://bugs.digium.com/ Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] door phone
who says they have a channel bank, or a zaptel card for that matter. your $50 solution figure would then be a bit skewed, eh? Jon Pounder wrote: I posted my solution yesterday and it is $50 so not sure why people are still asking for a cheap solution. I have a cheap disposable walmart phone on the channel bank, in immediate mode so when it is picked up it jumps immediately to the default context. *snipped ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] door phone
who says they have a channel bank, or a zaptel card for that matter. your $50 solution figure would then be a bit skewed, eh? so use a port on one of the 4port fxs cards then. no matter what solution you use, you still need a port on the pbx to connect to. or pickup one of the $10 x100p clones, and another $10 for the fxo to fxs convertor - You can hard code the caller id, not like it has to work through that mess, since there is only one phone on the end of it. bottom line is anyone can afford it. Jon Pounder wrote: I posted my solution yesterday and it is $50 so not sure why people are still asking for a cheap solution. I have a cheap disposable walmart phone on the channel bank, in immediate mode so when it is picked up it jumps immediately to the default context. *snipped ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Agent Logoff inability when calls are being received from queue
Now, while an agent receives a call from the Queue they cannot logoff using AgentCallbackLogin. Instead the Agent is asked for their agent no and their password and after that they get the agenty-alreadyon message!!! When the call goes away they are able to logoff!! when the nice lady asks for a extension just use '#' to log off extension must look like exten 1,1 agentcallbacklogin(someagent#|@somecontext) note i did not use the optional [EMAIL PROTECTED] synatx ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multi-line TTS Outbound Dialer
Hello, I've been lurking around the mailing list and browsing around on Asterisk-related links while I wait for my X100P to come in the mail. So far I haven't found very much information related to what I want to do with Asterisk. I was wondering if someone could point me in the direction of any work that may already have been done on a project similar to the one I'm trying to do. I'm interested in creating an outbound dialer application that will leave voice alerts with our customers. I want it to select a list of phone numbers and accompanying text messages from a PostgreSQL database, use a TTS engine to convert the text messages into audio files, and use all available lines to send out the messages as quickly as possible. I also want to make sure that it works with an answering machine. After looking into a number of cheap dialer apps that only support half-duplex voice modems, I found Asterisk and the accompanying hardware at the Asterisk store, which I'm told supports full-duplex audio. As far as I understand it, full-duplex audio is necessary to detect answering machines well. This is what I have in mind for the answering machine detection algorithm: 1) Dial the number. 2) Wait until line is picked up. 3) Wait until 1-2 seconds of relative silence (silence threshold will require some calibration). 4) Begin leaving message. 5) If during message, noise is heard coming from the other end, stop sending message and loop back to step number 3. 6) After leaving message successfully, hang up. Anyway, if anyone could point out any work that has already been done in this regard, I would really appreciate it. Thanks, Carl Youngblood ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Has anyone else had problems with Chagres?
I have an order for an SPA-2000 through them, and they won't respond to any email I send them. I've also tried calling them, but I can never get a human. I've left voice messages, but they haven't responded. Does anyone know any other way I can get in contact with them? Thanks! Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modem cards??
--On Wednesday, November 26, 2003 9:03 PM +0500 [EMAIL PROTECTED] wrote: On Wed, 26 Nov 2003, Angel Gabriel wrote: I've just been informed, in an IRC room, that it is possible to use a modem card with *. Can someon please confirm this for me? Thanks in advance. why don't you also mention that after being informed of this, about 10 different people also told you it wouldn't work and to just go ahead and by an X100P ? Ok if standard voice modmes do not work with Asterisk as an FXO then why does the X100P look an awful lot like the AMI-IA-96 modem? In fact why do they even have the same FCC ID numbers? I have no problems with supporting Digium by purchasing their hardware, but what makes this modem more equal than others? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Modem cards??
Is it just me or do we have this same modem x100p clone conversation on here at least once in two weeks literally ? For anyone who doesn't know the facts, look at the past emails on this subject on google. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Nelson Sent: Thursday, November 27, 2003 9:39 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Modem cards?? Ok if standard voice modmes do not work with Asterisk as an FXO then why does the X100P look an awful lot like the AMI-IA-96 modem? AMI-IA-96's are all of ten bucks, buy one and find out. In fact why do they even have the same FCC ID numbers? I have no problems with supporting Digium by purchasing their hardware, but what makes this modem more equal than others? -Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ENUM regexp replacements
Anyone succeeded in using regexp replacements in ENUM, like !\\+421257296(.*)$!sip:[EMAIL PROTECTED] I can't get it to work in ASterisk. I've added '\\1' and Debug echos 1 I've added '1' and debug echoes \1, but regexp fails to work. The example above is from the nic.at presentation, I can't get it to work. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ENUM regexp replacements
Thats because thats not correct. show me your full NAPTR record. bkw On Thu, 27 Nov 2003, Olle E. Johansson wrote: Olle E. Johansson wrote: Anyone succeeded in using regexp replacements in ENUM, like !\\+421257296(.*)$!sip:[EMAIL PROTECTED] I can't get it to work in ASterisk. I've added '\\1' and Debug echos 1 I've added '1' and debug echoes \1, but regexp fails to work. The example above is from the nic.at presentation, I can't get it to work. Enabled the ENUM debug stub in enum.c strcpy(regexp, !^\\+43(.*)[EMAIL PROTECTED]); ...and it fails... I'll file a bug report. /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ENUM regexp replacements
A working ENUM record: ; Mecosta Test routing *.2.7.9.1.3.2.1 IN NAPTR 100 10 u E2U+X-IAX2 !^\\+(.*)$!iax2:gw-mecosta/\\1! . Jeremy McNamara Olle E. Johansson wrote: Anyone succeeded in using regexp replacements in ENUM, like !\\+421257296(.*)$!sip:[EMAIL PROTECTED] I can't get it to work in ASterisk. I've added '\\1' and Debug echos 1 I've added '1' and debug echoes \1, but regexp fails to work. The example above is from the nic.at presentation, I can't get it to work. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: ENUM regexp replacements
Brian West wrote: On Thu, 27 Nov 2003, Olle E. Johansson wrote: Olle E. Johansson wrote: Anyone succeeded in using regexp replacements in ENUM, like !\\+421257296(.*)$!sip:[EMAIL PROTECTED] I can't get it to work in ASterisk. I've added '\\1' and Debug echos 1 I've added '1' and debug echoes \1, but regexp fails to work. The example above is from the nic.at presentation, I can't get it to work. Enabled the ENUM debug stub in enum.c strcpy(regexp, !^\\+43(.*)[EMAIL PROTECTED]); ...and it fails... I'll file a bug report. Thats because thats not correct. Aha, I thought the stub was correct.. show me your full NAPTR record. I try all versions: 0.5.2.3.4.4.4.8.6.4 IN NAPTR 100 10 u E2U+IAX2 !^\\+*(.*)$!iax2:webwayvx/3000! . *.7.6.7.0.0.0 IN NAPTR 100 10 u E2U+SIP !^\\+000767(.*)$!sip:[EMAIL PROTECTED] . *.1.6.4.0.0.0 IN NAPTR 100 10 u E2U+SIP !^+000461(.*)$!sip:[EMAIL PROTECTED] . Thank you for looking into this. /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK WITHOUT ANY CARD
Hcqm wrote: CAN I USE/COMPILE ASTERISK without any telephone/sound card? I only want to use it as a IP PBX. Yes, please go ahead. I'm running Asterisk on both LInux and FreeBSD servers without any PSTN or ISDN hardware. Have fun! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK WITHOUT ANY CARD
CAN I USE/COMPILE ASTERISK without any telephone/sound card? I only want to use it as a IP PBX. Yes. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN, BRI, PRI, voice over IP, and more...
Hi, I am thinking of starting a project at Work to pilot to use of Astrisk and VOIP, and would like to tie in several projects together. Currently we are looking at purchasing an ISDN and H.323 based video conferencing system from Polycom. It suggests the use of 3-4 BRI lines (ie 6-8 B channels, but in the form of individual BRIs). We need ISDN for the video conferencing to communicate with the Department of Interior conferencing system. Also we are looking at setting up a US Robotics Total Control V.90 modem rack for field users to be able to dial into the network. We can feed it with a few analog lines and run it at V.34, but would prefer to use the PRI card and feed with a few channels. We are only looking at using about 4 modems at once, at the absolute most. I can bring in ISDN BRI lines from Qwest through University Telcom, or I can bring in a single PRI via any number of CLECs, and have the University transport it to us. The PRI would be far cheaper in the long run. The problem is finding a way to split the PRI, and Astrisk looks like it might be it. I also want to start deploying VOIP phones in a pilot project, and need circuits for it. From reading the archives I have learned that a 4 port T-1 card can accept an incoming PRI and give an outgoing fractional PRI for a modem rack, but I have not been able to find any information about serving ISDN BRI circuits. If the Polycom unit would accept a fractional PRI I would be set, but it does not appear to have that option. Does anyone have any information they can share on this? Harry -- Harry McGregor, Computing Manager Tucson Support Group - U.S. Geological Survey University of Arizona - Environment and Natural Resource Building 520-670-5574 (office) - [EMAIL PROTECTED] 520-661-7875 (Cell) - [EMAIL PROTECTED] The opinions/statements expressed herein are my own and should not be taken as a position, opinion, or endorsement of the University of Arizona or the U.S. Geological Survey. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Ethereal plugin v0.3 is out
Hi people. The latest version of my Ethereal plugin for IAX2 is now available here: - http://almaw.com/ethereal-iax2-plugin-0.3.zip A screenshot showing what you're missing is here: - http://almaw.com/ethereal.png The new version adds the following features/bugfixes: - Decomposes the CODEC fields for supported CODECs, complete with nice English descriptions. This gives you a list of both supported and unsupported CODECs during initial negotiation (see screenshot). - Understands CONTROL packets better. - Decomposes mini-voice packets properly. - Now uses the INFO column to display packet type, etc. - Better categorization for colour filtering, etc. - Fixed the timestamps. Still to-do: - Prevent nastiness if someone sends malformed packets down the wire (better bounds error checking). - Understand TRANSFER stuff. - Understand DIALPLAN status updates. Regards, -- Alastair Maw Systems Analyst http://www.mxtelecom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Modem cards??
'These' people don't bother reading, cause when they search for list messages about modem cards, there's one real post, and a dozen smart remarks about the fact they we've been there and done that already. If all you people put your efforts together to create an easy to use web front for searching the lists, and no, Google is not enough for some of us, then we'll all get a better experience. You (not the flamers) who really want to look for information - try looking at the wiki at http://www.voip-info.org/tiki-index.php?page=Asterisk as it is easy to use, and most of the important posts from this list end up there. Flamers, please don't bother to answer. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Thursday, November 27, 2003 11:57 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Modem cards?? For anyone who doesn't know the facts, look at the past emails on this subject on google. I think that the problem is that these people don't bother reading in the first place... Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Modem cards??
Hi! You (not the flamers) who really want to look for information - try looking at the wiki at http://www.voip-info.org/tiki-index.php?page=Asterisk as it is easy to use, and most of the important posts from this list end up there. Question: Why does the Wiki search return nothing when I search for agi or php? Search terms too short? Can that be changed to allow 3 character words? A search for eagi works. Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIMPLE support in Asterisk?
On Thu, 2003-11-27 at 12:03, Mark Spencer wrote: Yea, cause I used both Kphone and Windows messenger, and they successfully registered (and subscribed i think) towards asterisk. Using Kphone I even get a online status on all other users on the asterisk but no interaction with status or IM. So maybe there is some quasi presence avaible? I think it would be a great tool to support IM/Presence. There is so much that can be done with such implementations. SIMPLE could be added within chan_sip, but there is no mechanism within Asterisk to move text from one channel to another *without* the context of a call. *With* the context of a call, we definitely have such a thing (TEXT frames) I haven't read the rest of this thread as I've been away, but I would also love to see Asterisk able to support some sort of IM'ing / Presence. Sorry for the non-informative post :) Leif Madsen. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Grandstream BT-100 and latest CVS
Hello, I was successfully using the BT-100 phone with CVS 11/10. Now that I've upgraded to 11/27, I can't place an outbound call. However the phone is registered and works well with inbound calls. Any suggestions will be appreciated. Thank you. Regards, Christopher ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] door phone
So, anyone got a solution for under AUD$100 ? Surely this is really just a bunch of cheap/commodity electronic components? [EMAIL PROTECTED] wrote: I posted my solution yesterday and it is $50 so not sure why people are still asking for a cheap solution. I have a cheap disposable walmart phone on the channel bank, in immediate mode so when it is picked up it jumps immediately to the default context. I can also dial it like any other extension to talk to people at the door. OK, this part I understood/understand. It's easy and achieves a simple intercom (but depending on where you live, might cost you a LOT if you need to keep replacing the handset because people keep damaging it... Perhaps a cheap (as cheap as possible) speaker phone behind some sort of metal box with a grille where the speaker part is. Modify the 'hook' into a button and you are all done. I have a $20 electric strike on the door which is coupled to my burglar alarm (that I can use a keypad outside to open right now) to interface to asterisk all I need is the following about $5 worth of stuff. Yes, either someone sent it to the list, or I found a AUD$50 door strike for a 'standard' door. You say yours is interfaced to your alarm, what happens if you don't have an alarm? This seems to be the point where I keep getting confused. 10k resistor from one of the data lines on the parallel port to base of a 2n transistor, emitter of to ground/earth, collector to a coil of a 12vdc pcb mount relay, 1n400x diode in inverse parallel with the coil to kill spikes, other end of the diode/coil to 12vdc from a disk drive connector in the pc. normally open contacts of the coil in parallel with whatever else drives the door strike (normally a 12-24AC/DC supply - AC if you like that Buzz, DC if a click is more your liking.) I mostly understand that up to the inverse parallel but... I did some basic electronics stuff in school and in project kits, but never quite got up to IC's. Could you please consider making the above into some sort of circuit diagram/drawing? Make a simple agi script either using system command or in C - output a 1 to the bit of the parport, wait 5seconds, output a 0. Done. This bit should be simple :) Much less than $50 - parts avail - Radio Shack or Dick Smith (are they still around in AU ?) Digikey is another fine source. Yes, Dick Smith are still around... So, I understand that there are a few components needed for this to work: Asterisk running on a pc with a parallel port (or maybe serial port?) Small electronic box described above which interfaces from the parallel port to ??? Door strike So, the bit I am missing is does the ??? mean your alarm panel, or does the electronic box connect directly to the door strike? What about some of these door strikes which can 'monitor' the door, can that information be passed back to asterisk? (I'm thinking turn on the lights etc when the door is opened with a key or forced open (can the door strike tell the difference?)) Thanks for your help. I think this should be added to the wiki perhaps? Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT: CISCO-ATA-186
Hi All, It's a long shot but may be someone has CISCO-ATA-186 for sale in Australia? Pls contact me off list if you do. Thanks Alex. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Grandstream BT-100 and
I was successfully using the BT-100 phone with CVS 11/10. Now that I've upgraded to 11/27, I can't place an outbound call. However the phone is registered and works well with inbound calls. Any suggestions will be appreciated. Thank you. Hi! I encounter similar problems. But in my case also incomming calls are not possible. But this might be because of my upgrade to the latest bt-100 beta-firmware (b13p4.22)! Additionally I can't call internal extensions (for example the echo test). Currently i'm using the Asterisk-CVS from 27.11.2003. If i use a softphone (x-lite from x-ten) to connect to * it works perfectly in the same constallation (all with static ip-adresses). Daniel ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTERISK WITHOUT ANY CARD
netstat -nap and see if any ports are open by asterisk, run asterisk by doing asterisk -vc and see if there's any error messages. (don't quit asterisk to do netstat -nap) Should work fine on RH9 - Original Message - From: Hcqm [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 28, 2003 10:41 AM Subject: Re: [Asterisk-Users] ASTERISK WITHOUT ANY CARD Thanks Olle, I asked because I followup all docs on installation and compile it without problems and edited sip.conf inorder to listen on my ip address on 5060 port. But when I run * the port is not open, no firewalling enabled... My system is a RH9. Any help will be appretiated. Regards, Hector. - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Jueves, 27 de Noviembre de 2003 06:54 p.m. Subject: Re: [Asterisk-Users] ASTERISK WITHOUT ANY CARD Hcqm wrote: CAN I USE/COMPILE ASTERISK without any telephone/sound card? I only want to use it as a IP PBX. Yes, please go ahead. I'm running Asterisk on both LInux and FreeBSD servers without any PSTN or ISDN hardware. Have fun! /Olle ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mailing list archives searchable ?
On Thu, 2003-11-27 at 18:49, Arnold Ligtvoet wrote: Hi, I have started development to import the mailinglist archives into a MySQL database and creating a full text search possibility on this. My questions; 1) Is this already done somewhere else ? 2) Is this of interest ? I actually just read someone complaining about this today. I'm sure it'd be a very welcome contribution. The only place that does this is google, and isn't always the best way to search for something specific to the mailing list. -- Leif Madsen leif at hacklocalhost dot com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan
exten = _0119X,1,Congestion
exten = _011[0-8]X,1,Dial(Somechannel,${EXTEN})
See page 27 of the Asterisk Handbook, version 2 for further details.
Steve.
On Friday 28 November 2003 01:53, Isamar Maia wrote:
Hi Folks,
I already know how to make a simple dialplan to specific number pattern.
Now, I need the following:
Calls to 0119XXX - Blocked the calls
Calls to 011 - Route the calls
How can the first exception be done?
Isamar
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Re: [Asterisk-Users] Mailing list archives searchable ?
If you go to google and add site:lists.digium.com then your keywords.. you can search the list. bwk On Fri, 28 Nov 2003, Arnold Ligtvoet wrote: Hi, I've been on the list for slightly under a month now and noticed; a) a fairly high amount of traffic, b) a lot of questions which come up more than once, c) the archives at lists.digium.com are not searchable. I have started development to import the mailinglist archives into a MySQL database and creating a full text search possibility on this. My questions; 1) Is this already done somewhere else ? 2) Is this of interest ? I'm developing myself now, mainly because I couldn't find a component in PHP or Perl that performs this trick. Does anybody know one, if not I'll continue on my own components. Thanks, Arnold Ligtvoet. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan
exten = _0119.,1,blah exten = _011.,1,blah would that work? On Fri, 28 Nov 2003, Isamar Maia wrote: Hi Folks, I already know how to make a simple dialplan to specific number pattern. Now, I need the following: Calls to 0119XXX - Blocked the calls Calls to 011 - Route the calls How can the first exception be done? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mailing list archives searchable ?
I believe there's a cool site that indexes many mailing lists, including asterisk. google for mailing list archives or similar. Sorry i can't remember the name atm. - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, November 28, 2003 12:13 PM Subject: Re: [Asterisk-Users] Mailing list archives searchable ? On Thu, 2003-11-27 at 18:49, Arnold Ligtvoet wrote: Hi, I have started development to import the mailinglist archives into a MySQL database and creating a full text search possibility on this. My questions; 1) Is this already done somewhere else ? 2) Is this of interest ? I actually just read someone complaining about this today. I'm sure it'd be a very welcome contribution. The only place that does this is google, and isn't always the best way to search for something specific to the mailing list. -- Leif Madsen leif at hacklocalhost dot com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN, BRI, PRI, voice over IP, and more...
On Thu, 2003-11-27 at 16:19, Harry McGregor wrote: Hi, I am thinking of starting a project at Work to pilot to use of Astrisk and VOIP, and would like to tie in several projects together. Currently we are looking at purchasing an ISDN and H.323 based video conferencing system from Polycom. It suggests the use of 3-4 BRI lines (ie 6-8 B channels, but in the form of individual BRIs). We need ISDN for the video conferencing to communicate with the Department of Interior conferencing system. Also we are looking at setting up a US Robotics Total Control V.90 modem rack for field users to be able to dial into the network. We can feed it with a few analog lines and run it at V.34, but would prefer to use the PRI card and feed with a few channels. We are only looking at using about 4 modems at once, at the absolute most. I can bring in ISDN BRI lines from Qwest through University Telcom, or I can bring in a single PRI via any number of CLECs, and have the University transport it to us. The PRI would be far cheaper in the long run. The problem is finding a way to split the PRI, and Astrisk looks like it might be it. I also want to start deploying VOIP phones in a pilot project, and need circuits for it. From reading the archives I have learned that a 4 port T-1 card can accept an incoming PRI and give an outgoing fractional PRI for a modem rack, but I have not been able to find any information about serving ISDN BRI circuits. If the Polycom unit would accept a fractional PRI I would be set, but it does not appear to have that option. Does anyone have any information they can share on this? I noticed that some of those Polycom units supported V.35 connections. You could possibly get a V.35 card for your asterisk computer and split the channels off to the V.35 similar to how the ZapRas works. I don't know if that will work though. Another route would be to help hack up the libpri code to support the 3DSO or Brite signaling to support ISDN lines of T1. CAC has ISDN cards for the ADIT600, and I'm sure there is an equivalent for Adtran. As far as the breaking of a few channels to the Modem rack, that is easy. Just pick a DID number for asterisk to route the call via and send it down one of the second channels. You could ideally set it up to have all channels active instead of a fractional PRI up. This would let you surge up to whatever your demand needed. It is also probable that when your mode needs where high that your office needs are low and it would let you use the channels more effectively. Remember you only route on the channel, it isn't like each channel is a phone number. Any of the channels could be any of the numbers. The VoIP would easily fit into that setup also. Not to mention that if anyone wanted to loan you other equipment to test out, asterisk could provide a PRI connection to it also. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] door phone
Ok I guess I need to stop being lazy, and actually finish this project. If you don't hear from me on the list about it in the next couple weeks, bug me. I'll complete the hardware, and take photos of it. (I have several loose electric strikes besides the one on the front door so I can photograph them and the electronics I build, and I will throw together some sort of agi or system command that can be executed to activate the strike.) By the way, not to keep you in suspense, but all inverse parallel means is in a DC circuit the diode is installed in the direction that would normally block current, but when the coil in the relay is de-energized, and the magnetic field collapses this is the proper direction for the diode to be pointed so it shunts the voltage spike across the coil and keeps it out of the more sensitive solid state parts of the circuit. So, anyone got a solution for under AUD$100 ? Surely this is really just a bunch of cheap/commodity electronic components? [EMAIL PROTECTED] wrote: I posted my solution yesterday and it is $50 so not sure why people are still asking for a cheap solution. I have a cheap disposable walmart phone on the channel bank, in immediate mode so when it is picked up it jumps immediately to the default context. I can also dial it like any other extension to talk to people at the door. OK, this part I understood/understand. It's easy and achieves a simple intercom (but depending on where you live, might cost you a LOT if you need to keep replacing the handset because people keep damaging it... Perhaps a cheap (as cheap as possible) speaker phone behind some sort of metal box with a grille where the speaker part is. Modify the 'hook' into a button and you are all done. I have a $20 electric strike on the door which is coupled to my burglar alarm (that I can use a keypad outside to open right now) to interface to asterisk all I need is the following about $5 worth of stuff. Yes, either someone sent it to the list, or I found a AUD$50 door strike for a 'standard' door. You say yours is interfaced to your alarm, what happens if you don't have an alarm? This seems to be the point where I keep getting confused. 10k resistor from one of the data lines on the parallel port to base of a 2n transistor, emitter of to ground/earth, collector to a coil of a 12vdc pcb mount relay, 1n400x diode in inverse parallel with the coil to kill spikes, other end of the diode/coil to 12vdc from a disk drive connector in the pc. normally open contacts of the coil in parallel with whatever else drives the door strike (normally a 12-24AC/DC supply - AC if you like that Buzz, DC if a click is more your liking.) I mostly understand that up to the inverse parallel but... I did some basic electronics stuff in school and in project kits, but never quite got up to IC's. Could you please consider making the above into some sort of circuit diagram/drawing? Make a simple agi script either using system command or in C - output a 1 to the bit of the parport, wait 5seconds, output a 0. Done. This bit should be simple :) Much less than $50 - parts avail - Radio Shack or Dick Smith (are they still around in AU ?) Digikey is another fine source. Yes, Dick Smith are still around... So, I understand that there are a few components needed for this to work: Asterisk running on a pc with a parallel port (or maybe serial port?) Small electronic box described above which interfaces from the parallel port to ??? Door strike So, the bit I am missing is does the ??? mean your alarm panel, or does the electronic box connect directly to the door strike? What about some of these door strikes which can 'monitor' the door, can that information be passed back to asterisk? (I'm thinking turn on the lights etc when the door is opened with a key or forced open (can the door strike tell the difference?)) Thanks for your help. I think this should be added to the wiki perhaps? Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Modem cards??
On Thu, 2003-11-27 at 13:39, Andrew Nelson wrote: --On Wednesday, November 26, 2003 9:03 PM +0500 [EMAIL PROTECTED] wrote: On Wed, 26 Nov 2003, Angel Gabriel wrote: I've just been informed, in an IRC room, that it is possible to use a modem card with *. Can someon please confirm this for me? Thanks in advance. why don't you also mention that after being informed of this, about 10 different people also told you it wouldn't work and to just go ahead and by an X100P ? Ok if standard voice modmes do not work with Asterisk as an FXO then why does the X100P look an awful lot like the AMI-IA-96 modem? In fact why do they even have the same FCC ID numbers? I have no problems with supporting Digium by purchasing their hardware, but what makes this modem more equal than others? Because modem drivers do not allow full duplex sound. This is a limitation of the interface chosen by the implementers of the specification. I'd about lay money they did this to not cannibalize their higher profit telephony card market. In the case of Digium's card, they wrote a driver that no longer contains any protected(patented) code for modems, and just exposed the audio in both direction and the telephony sensing. This allows them to bypass large chunks of code that would sit dormant and just be in the way of the asterisk system. So the extra you pay is in part helping Digium repay their development costs up to this point, both in the driver and in the application. The other part you are paying for is 1 hour of support from Digium, and a bit of a free pass from being flamed on this list for scabbing parts. So please support Digium and their fine network of resellers by buying parts from the authorized channels. The alternative may be cheaper on the wallet, but not on your soul. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line TTS Outbound Dialer
Just a warning that you will not know when the line is picked up with a X100P. Later when you upgrade to T1/E1 service you will know when it is picked up. So I assume that means that I should just wait until I hear some level of noise and then I know that the line has been picked up? You may want to rethink this just a bit, wouldn't want to restart the message because someone coughed on the line. Maybe a limit of how far in to the message before you ignore noise. Or maybe noise would have to last for more than a certain period of time before it triggered another waiting sequence. Like, say, if noise lasts for longer than 2 full seconds or something. Shouldn't be a problem. If you write your own DSP stuff, you could write this using EAGI, otherwise you will need to write it into an asterisk app to get at the already written silence detection. Either way, you could use festival for the TTS, or for certain messages, you could go ahead a record prompts to be played. This way you are sure your message will be understood. What is EAGI? I will probably use festival for the time being, but I thing that I would eventually like to use ScanSoft's RealSpeak SDK because it is so life-like. Unfortunately our text alerts are fully customizeable, so we can't pre-record them. Thanks for the help! Carl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan
On Thu, 2003-11-27 at 19:17, Brian West wrote: exten = _0119.,1,blah exten = _011.,1,blah would that work? Unless you are in a location that needs to support varying length outbound phone numbers, you should really fully define the pattern. This will let asterisk know ahead of time how many digits it is expected to get from the user before commencing with the dialing. On Fri, 28 Nov 2003, Isamar Maia wrote: Hi Folks, I already know how to make a simple dialplan to specific number pattern. Now, I need the following: Calls to 0119XXX - Blocked the calls Calls to 011 - Route the calls How can the first exception be done? Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN, BRI, PRI, voice over IP, and more...
Thank you for the reply, please see questions in-line. On Thu, 2003-11-27 at 18:36, Steven Critchfield wrote: snip I noticed that some of those Polycom units supported V.35 connections. You could possibly get a V.35 card for your asterisk computer and split the channels off to the V.35 similar to how the ZapRas works. I don't know if that will work though. Ok, I will look at the V.35 options. Another route would be to help hack up the libpri code to support the 3DSO or Brite signaling to support ISDN lines of T1. CAC has ISDN cards for the ADIT600, and I'm sure there is an equivalent for Adtran. So you don't know of any support to use a ISDN4Linux or similar card as an ISDN FXS port? I have found a device that will take a single PRI and break it into multiple PRI and BRI groups, but it is quite expensive. http://www.patapsco.co.uk I was looking at the Liberator 1PxB and Liberator 2PxB As far as the breaking of a few channels to the Modem rack, that is easy. Just pick a DID number for asterisk to route the call via and send it down one of the second channels. You could ideally set it up to have all channels active instead of a fractional PRI up. This would let you surge up to whatever your demand needed. It is also probable that when your mode needs where high that your office needs are low and it would let you use the channels more effectively. Remember you only route on the channel, it isn't like each channel is a phone number. Any of the channels could be any of the numbers. Sounds great. The VoIP would easily fit into that setup also. Perfect. We are looking at providing some remote telephone connections for our telecommuting users. Not to mention that if anyone wanted to loan you other equipment to test out, asterisk could provide a PRI connection to it also. Harry -- Harry McGregor, Computing Manager Tucson Support Group - U.S. Geological Survey University of Arizona - Environment and Natural Resource Building 520-670-5574 (office) - [EMAIL PROTECTED] 520-661-7875 (Cell) - [EMAIL PROTECTED] The opinions/statements expressed herein are my own and should not be taken as a position, opinion, or endorsement of the University of Arizona or the U.S. Geological Survey. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial Plan
Yes I recall simlar from the handbook.
bkw
exten = _0119X,1,Congestion
exten = _011[0-8]X,1,Dial(Somechannel,${EXTEN})
On Thu, 27 Nov 2003, Steven Critchfield wrote:
On Thu, 2003-11-27 at 19:17, Brian West wrote:
exten = _0119.,1,blah
exten = _011.,1,blah
would that work?
Unless you are in a location that needs to support varying length
outbound phone numbers, you should really fully define the pattern. This
will let asterisk know ahead of time how many digits it is expected to
get from the user before commencing with the dialing.
On Fri, 28 Nov 2003, Isamar Maia wrote:
Hi Folks,
I already know how to make a simple dialplan to specific number pattern.
Now, I need the following:
Calls to 0119XXX - Blocked the calls
Calls to 011 - Route the calls
How can the first exception be done?
Isamar
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--
Steven Critchfield [EMAIL PROTECTED]
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RE: [Asterisk-Users] Mailing list archives searchable ?
Come on guys how hard is it to add site:lists.digium.com into the google search box along with your keywords? Or is that like too hard? On Thu, 27 Nov 2003, Dustin Knuttgen wrote: Would really love to see a searchable archive. I think it would be very helpful. Thanks for taking this project on. Dustin Knuttgen -Original Message- From: Arnold Ligtvoet [mailto:[EMAIL PROTECTED] Sent: Thu 11/27/2003 6:49 PM To: [EMAIL PROTECTED] Cc: Subject: [Asterisk-Users] Mailing list archives searchable ? Hi, I've been on the list for slightly under a month now and noticed; a) a fairly high amount of traffic, b) a lot of questions which come up more than once, c) the archives at lists.digium.com are not searchable. I have started development to import the mailinglist archives into a MySQL database and creating a full text search possibility on this. My questions; 1) Is this already done somewhere else ? 2) Is this of interest ? I'm developing myself now, mainly because I couldn't find a component in PHP or Perl that performs this trick. Does anybody know one, if not I'll continue on my own components. Thanks, Arnold Ligtvoet. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line TTS Outbound Dialer
Carl Youngblood wrote: What is EAGI? I will probably use festival for the time being, but I thing that I would eventually like to use ScanSoft's RealSpeak SDK because it is so life-like. Unfortunately our text alerts are fully customizeable, so we can't pre-record them. Beware the likelike TTS, that sucks up thousands of dollars and gets thrown away. RealSpeak is great for demos :-) Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk behind NAT How to do it.
Thanks to ww and his patch on bug #104, I have successfully implemented Asterisk behind NAT without using STUN or anything crazy. It's quite straight forward. Until this gets tested enough and put into CVS, you will have to patch your chan_sip.c file to do this. I'm sure within the next few days this will get put merged into CVS if no one finds any problems. I tried this on chan_sip.c version 1.249 (the version the patch was written for) and the latest as of today 1.258. Both work great. Open ports 5060 and your RTP range (found in /etc/asterisk/rtp.conf). Default is 1 - 2 Forward ports 5060 and your RTP range to your internal Asterisk box. For your sip.conf, you need to add three lines: ; sip.conf snippet [general] port=5060 ; make sure you have this line :) inside_net=192.168.1.100; this is the internal ip address of the; asterisk server inside_mask=255.255.255.0 ; internal ip mask. /24 as this example outside_addr=216.239.33.100 ; this can also be a FQDN! ie. ; my.domain.com ; ... plus whatever else you have in your sip.conf Download the patch at: http://bugs.digium.com/file_download.php?file_id=430type=bug Either update your Asterisk or verify you have at least version 1.249 of chan_sip.c: cd /usr/src/asterisk/channels/ cvs status chan_sip.c === File: chan_sip.cStatus: Locally Modified Working revision:1.258 Repository revision: 1.258 /usr/cvsroot/asterisk/channels/chan_sip.c,v While in pwd /usr/src/asterisk/channels/ patch -p0 /path/to/patch Nothing should fail. cd /usr/src/asterisk/ make cp /usr/src/asterisk/channels/chan_sip.so /usr/lib/asterisk/modules/ Restart your Asterisk and try it. If you want to call a NAT'd Asterisk box, my Free World Dialup number is 18924. Currently online. -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] AGI (IF/ELSE)
Ok.. I was thinking about this.. It is not a very wise decsion to put
the user input in a loop.. So how could I do some error checking outside
of the loop?
-gcc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of PBX
Posted At: Thursday, November 27, 2003 9:19 PM
Posted To: Asterisk User Group
Conversation: AGI (IF/ELSE)
Subject: [Asterisk-Users] AGI (IF/ELSE)
I need some help with some statements.
#!/usr/bin/perl
use Asterisk::AGI;
$AGI = new Asterisk::AGI;
my %input = $AGI-ReadParse();
my $callerid = $input{'callerid'};
if ($optemp != 1) {
my $empid = $AGI-get_data('employee',-1,5);
$AGI-stream_file(entered);
$AGI-say_digits($empid);
my $optemp = $AGI-get_data('correct',-1,1);
}else{
my $strid = $AGI-get_data('store',-1,5);
$AGI-stream_file(entered);
$AGI-say_digits($strid);
my $optstr = $AGI-get_data('correct',-1,1);
}
exit;
I can't seem to figure out what I am doing wrong. When the script is
run. The user puts in there employee ID and then hears it back to them.
Then they are asked if this is correct press 1 for yes or 9 for no. If
they press 1, it should go onto the next piece of the script
But if I press 1 the script ends Any ideas
Thanks,
-gcc
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Re: [Asterisk-Users] Multi-line TTS Outbound Dialer
On Thu, 2003-11-27 at 19:48, Carl Youngblood wrote: Just a warning that you will not know when the line is picked up with a X100P. Later when you upgrade to T1/E1 service you will know when it is picked up. So I assume that means that I should just wait until I hear some level of noise and then I know that the line has been picked up? For testing, that would do fine. Like I said, when you move up to digital, it all is much easier to deal with. You may want to rethink this just a bit, wouldn't want to restart the message because someone coughed on the line. Maybe a limit of how far in to the message before you ignore noise. Or maybe noise would have to last for more than a certain period of time before it triggered another waiting sequence. Like, say, if noise lasts for longer than 2 full seconds or something. That may be fine. Although you may have trouble with some line that also is feeding back echo. That would cause you a bad loop. Like I said, you may wish to limit the number of restarts just in case you end up misdialing a system that is just repeating it's menu. Shouldn't be a problem. If you write your own DSP stuff, you could write this using EAGI, otherwise you will need to write it into an asterisk app to get at the already written silence detection. Either way, you could use festival for the TTS, or for certain messages, you could go ahead a record prompts to be played. This way you are sure your message will be understood. What is EAGI? I will probably use festival for the time being, but I thing that I would eventually like to use ScanSoft's RealSpeak SDK because it is so life-like. Unfortunately our text alerts are fully customizeable, so we can't pre-record them. EAGI is Extended AGI. Basically it ends the audio data on file descriptor 3. The original use(I think) was to hook sphinx up to asterisk in a way that didn't cause licensing issues. In this case, you can use the audio coming in on file descriptor 3 to run your own DSP. Of course, with your TTS choice, I guess you will also being executing a text to speech command to file then stream the file. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN, BRI, PRI, voice over IP, and more...
On Thu, 2003-11-27 at 20:06, Harry McGregor wrote: On Thu, 2003-11-27 at 18:36, Steven Critchfield wrote: Another route would be to help hack up the libpri code to support the 3DSO or Brite signaling to support ISDN lines of T1. CAC has ISDN cards for the ADIT600, and I'm sure there is an equivalent for Adtran. So you don't know of any support to use a ISDN4Linux or similar card as an ISDN FXS port? One of the guys here is working on a ISDN card. I saw the site recently. I think I saw it mention that it could do network side signaling. But other than that, I think all the cards you would come into contact with are CPE only. I have found a device that will take a single PRI and break it into multiple PRI and BRI groups, but it is quite expensive. http://www.patapsco.co.uk I was looking at the Liberator 1PxB and Liberator 2PxB That looks like an interesting product. What price did you find on it? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RFC3389 support incomplete
Hi When i make a call using IAX2, the log of the remote asterisk say Nov 17 20:20:12 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:22 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Nov 17 20:20:26 NOTICE[28686]: File rtp.c, Line 263 (process_rfc3389): RFC3389 s upport incomplete. Turn off on client if possible Who i turn off and how i fix this thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] door phone
Jon Pounder wrote: I posted my solution yesterday and it is $50 so not sure why people are still asking for a cheap solution. I have a cheap disposable walmart phone on the channel bank, in immediate mode so when it is picked up it jumps immediately to the default context. I can also dial it like any other extension to talk to people at the door. I have a $20 electric strike on the door which is coupled to my burglar alarm (that I can use a keypad outside to open right now) to interface to asterisk all I need is the following about $5 worth of stuff. 10k resistor from one of the data lines on the parallel port to base of a 2n transistor, emitter of to ground/earth, collector to a coil of a 12vdc pcb mount relay, 1n400x diode in inverse parallel with the coil to kill spikes, other end of the diode/coil to 12vdc from a disk drive connector in the pc. normally open contacts of the coil in parallel with whatever else drives the door strike (normally a 12-24AC/DC supply - AC if you like that Buzz, DC if a click is more your liking.) Just as another data point, you could look at something like http://www.x10.com/products/x10_um506.htm It's an X10 module (powerline control system) that provides basically an on/off switch for low-voltage devices (up to [EMAIL PROTECTED]). Pretty much all you need then is the power supply and the strike. The main advantage is that being an X10 device, you don't have to run wires all the way to your computer room. All you need is your X10 computer interface (like the FireCracker PC Interface (CM17A), which every good geek has already, right? :-) -- Matt White [EMAIL PROTECTED] Arts and Science Computer Labs University of Saskatchewan It sure is Monday... Ain't it a sin I've gotta work my way thru the week again. - Mark Chesnutt...Sure Is Monday ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Mailing list archives searchable ?
On Thu, 27 Nov 2003, Brian West wrote: Come on guys how hard is it to add site:lists.digium.com into the google search box along with your keywords? Or is that like too hard? This doesn't always work well. For example, ten days ago a message came through the list with the text voicepulse working? in the subject line. (It's still in my mail folder, and I just happened to pick it at random.) The search string 'site:lists.digium.com voicepulse working' doesn't return any results. Same thing happened when I searched just now for this thread, which has been going for a few days. Anybody know how to tell how frequently Google reindexes a web site? Some things might be found by searching with google, but I feel that a separate, more frequenly updated method of searching the list archive has its place also. Greg Hill ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line TTS Outbound Dialer
What is EAGI? I will probably use festival for the time being, but I thing that I would eventually like to use ScanSoft's RealSpeak SDK because it is so life-like. Unfortunately our text alerts are fully customizeable, so we can't pre-record them. Beware the likelike TTS, that sucks up thousands of dollars and gets thrown away. RealSpeak is great for demos Do you mean that it doesn't work very well in practice, or that it works well but is simply not worth the money? Thanks, Carl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mailing list archives searchable ?
Greg Hill wrote: On Thu, 27 Nov 2003, Brian West wrote: Come on guys how hard is it to add site:lists.digium.com into the google search box along with your keywords? Or is that like too hard? This doesn't always work well. For example, ten days ago a message came through the list with the text voicepulse working? in the subject line. (It's still in my mail folder, and I just happened to pick it at random.) The search string 'site:lists.digium.com voicepulse working' doesn't return any results. Same thing happened when I searched just now for this thread, which has been going for a few days. Anybody know how to tell how frequently Google reindexes a web site? I got embarrassed a few weeks ago during a particularly interesting thread when I more or less ordered some people to use google to read the thread. At this time the original message in the thread had been out there for a week or two. I don't need to finish the story: nothing doing via google. So those of you who roast the newbie in a heartbeat, while performing in the main a valuable function, need to be aware that sometimes search the archives, stupid is not really going to reveal the information that your tart messages imply will be forthcoming. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Multi-line TTS Outbound Dialer
Or maybe noise would have to last for more than a certain period of time before it triggered another waiting sequence. Like, say, if noise lasts for longer than 2 full seconds or something. That may be fine. Although you may have trouble with some line that also is feeding back echo. That would cause you a bad loop. Like I said, you may wish to limit the number of restarts just in case you end up misdialing a system that is just repeating it's menu. Thanks for the help. All of these telephony issues are fairly new to me. So just for me to understand better, echo is basically something that is difficult to control, right? I mean, if a telco's line has echo, asterisk can't do anything about that, right? Thanks, Carl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
