[Asterisk-Users] re: where can i find all .C files
hi all, i m using debian to run my asterisk gateway.I want to make some customization in voicemail application.For that i need to modify voicmail's .C(source file) file. can any body tell me where exactly all .C files resides in the system.. thanks tejas __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM Gateway / Terminal for sale
I have used both Telular analog units and Voiceblue SIP units in Australia. PaulH - Original Message - From: Adrian Carter [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, January 07, 2006 1:40 AM Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale Is anyone aware of the details of this in Australia? I'd love to be able to let tech's have calls route straight to their mobiles when 'in-house' Steve Kennedy wrote: On Fri, Jan 06, 2006 at 01:23:26PM -, Chris Bagnall wrote: I don't get it. What is the advantage of using a GSM gateway? VOIP calls are pretty inexpensive as they are now. It largely depends on the country you're calling. Here in the UK, calls to mobiles are maintained at an artificially high rate because the terminating network (the mobile networks) get a cut of call revenue for calls *to* your mobile. By contrast, in the US, the mobile customer often pays a small charge per minute on incoming calls (as I understand the market over there). You'll also find in the UK the mobile phone market is heavily subsidized by the networks such that you can get phones for free if you sign up to 12 month contracts. I often find that it's cost-effective to get a new contract every 12 months (with a free phone), even if I don't want the phone. Flog the phone on ebay and you've got a spare SIM with lots of inclusive minutes for almost nothing. In the UK the wholesale rates are set by Ofcom (like the FCC), which works out about 7p'ish per minute. However the operators can offer retail bundles (including phones) and for a monthly contract they throw in various ammounts of cross network minutes (or free to their own network or whatever). With clever dial-plans and multiple terminals connected to multiple networks you can generally get free calls to mobile users (basically clever least cost routing, time of day sometimes needs to be taken into account as well). However there are some disadvantages, the main being you cant set CLI of the outgoing call as it will always be tied to the SIM of the mobile terminal. Another is that you can NOT run a GSM gateway (as they're known) for 3rd parties. So if you want to connect your office PBX to a gateway to make use of cheap mobile termination for your own company that's fine, but as an ITSP (or traditional telco) you can not allow 3rd party traffic to utilise a gateway. If networks find you are using a gateway (as a telco) they can cut it off, no questions asked. Gateways have been determined to be fixed infrastructure, therefore NOT mobile. There is (or maybe was by now) an Ofcom consultation asking whether this should be changed, the mobile operators will fight it, telcos and other users will be asking for it to be changed. Of course this is UK specific, other countries have more lenient policies (I think Belgium allow gateways, France doesn't allow any kind, and some allow them with the co-operation of the operators). Steve -- Adrian Carter Technical Manager Leading Edge Internet Web http://www.lei.net.au http://support.lei.net.au Direct+61 2 6163 6162 Support 1 300 662 415 E-mail[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: where can i find all .C files
On Sat, 2006-01-07 at 00:26 -0800, Tejas Shah wrote: hi all, i m using debian to run my asterisk gateway.I want to make some customization in voicemail application.For that i need to modify voicmail's .C(source file) file. can any body tell me where exactly all .C files resides in the system.. thanks tejas if you have the deb-src in your sources.lkist then you can install asterisk-source which will be in /usr/src somewhere.. If you want to build from source you can install subversion and get it straight from digium, for 1.2 here are some directions.. svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 svn checkout http://svn.digium.com/svn/zaptel/branches/1.2 zaptel-1.2 svn checkout http://svn.digium.com/svn/libpri/branches/1.2 libpri-1.2 svn checkout http://svn.digium.com/svn/asterisk-addons/branches/1.2 asterisk-addons-1.2 cd zaptel-1.2 make clean install \ cd ../libpri-1.2 make clean install \ cd ../asterisk-1.2 make clean install obviously make any changes between the svn checkout and the make :) -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk initialization
There is a sample php script in the contribs folder that shows who is logged in - one of my clients uses it. PaulH - Original Message - From: Dov Bigio To: asterisk-users@lists.digium.com Sent: Saturday, January 07, 2006 8:24 AM Subject: [Asterisk-Users] Asterisk initialization Hi, I am doing an AGI that logs to a database every Agent login/logoff. My idea is to be able to go to this database and check which agents where logged so that I can force their login in case Asterisk goes down for some reason. The problem is that I would need to reload their status from this AGI when Asterisk initializes. Is there a way to do this? One idea I had was to make safe_asterisk to generate a .call file that calls and extension that would call the AGI to log all the agents back on. Is there another way of running an AGI on initialization? Thank you Dov ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Annoying Notice Message: Don't know what to do with control frame 15
On 6 Jan 2006, at 16:28, Joan Bautista wrote:Hi, I haven't found anything about the message below on the mailing list, Does anyones knows why this notice is being appearing? -- Executing Dial("Local/[EMAIL PROTECTED],2", "IAX2/CallOut/12365533643|30|otT") in new stack -- Called CallOut/12365533643 -- Call accepted by 12.11.11.11 (format ulaw) -- Format for call is ulaw -- IAX2/10.11.240.110:4569-3 is proceeding passing it to Local/[EMAIL PROTECTED],2Jan 6 13:20:41 NOTICE[26911]: channel.c:2416 __ast_request_and_dial: Don't know what to do with control frame 15 -- IAX2/10.11.240.110:4569-3 is circuit-busy -- Hungup 'IAX2/12.11.11.11:4569-3' == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto("Local/[EMAIL PROTECTED],2", "s-CONGESTION|1") in new stack -- Goto (default,s-CONGESTION,1) -- Executing NoOp("Local/[EMAIL PROTECTED],2", ""CONG"") in new stack -- Executing Congestion(" Local/[EMAIL PROTECTED],2", "") in new stack Channel Local/[EMAIL PROTECTED],1 was never answered. == Spawn extension (default, s-CONGESTION, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' My calling scenario is like this: server01 server02 pstn server --IAX trunking-- agents/sip server server01: Asterisk 1.2.1server02: Asterisk 1.2.1Well, according to the iax RFC, control frame 15 (0xf) means 'call is Proceeding', I guess that the NOTICE is just telling you that asterisk can't do anything usefulwith that info, it doesn't sound like a problem to me.Tim. http://www.westhawk.co.uk/ ___
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[Asterisk-Users] wich IAX soft client allow to specify a different server port?
I still having problem with remote SIP client, trying to use IAX client instead but i've to specify TCP port 8080 (because of firewall). I did this on server in bindport=8080 in iax.conf but i cannot find a soft client that allow to set wich server port to use. Any idea? Thanks, Antonio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 is translating vocal sounds into DTMF
I am having the same problem with a male voice at the other end. It is making the spa3k problem for me. Has this been reported to SIPURA ? Is this a common problem ? has anyone done been able to make this happen less often ? I would hope perhaps there's some kind of setting that has to do with the way it detects inband DTMF? I'm pretty sure it's an artifact of this particular ATA; my other SIP devices are just fine. FWIW, I have the same prob on my spa3k. It's peircing and pisses my wife off greatly. Its a fairly common problem with the spa3k, but is somewhat dependent on how it is configured and the distance to your CO. If the spa3k is configured for ring thru (from the fxo to fxs without passing through asterisk), the call remains within the spa3k. Its echo cancellation and tone detection functions are not as good as they could be, and it seems to have more problems with longer length pstn lines then it does with shorter lines (eg, distance between the Central Office and the spa3k). Reducing the audio levels on the pstn/fxo port seems to help somewhat, but you'll reach a point where the audio is rather low when the issues have been minimized. (In other words, the spa3k does not seem to have a very good dynamic range.) Seems the v3.1.7g firmware is worse then some of their earlier releases. Linksys/Sipura support typically suggests very elementary things like check to ensure the impedence is set correctly for your telco line (etc), and then something like try v3.1.5 if you still have problems. I'd love to escalate the issue to level 2 support (assuming that even exists). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problens to link 2 * servers
Hello, I'm traying to link 2 * servers using SIP and the following errors was show: "SIP/AsteriskA:[EMAIL PROTECTED]/100") in new stackDec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such host: 10.0.0.121/100Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this timeDec 13 22:47:07 NOTICE[8767]: rtp.c:435 ast_rtp_read: RTP: Received packet with bad UDP checksumDec 13 22:47:07 WARNING[8767]: pbx.c:1949 ast_pbx_run: Timeout, but no rule 't' in context 'internal'linux*CLI Someone could help me to fix the following problens?? fraternatly, Cleyverson Pereira CostaPhone #: 27+9922-0111Skype: cleyversonMSN: [EMAIL PROTECTED]"Escolhe, pois, a vida, para que vivas, tu e a tua descendência."Deuteronômio 30:19 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Draytek Vigor 2900 Asterisk
I'm in conversation with Draytek's pre-sales dept.. Here's the most recent reply: Hello, We really don't know of anyone who has run an Asterisk server on a Vigor2900. There are doubtless people around, but it's relatively rare. Most people don't run SIP servers. Regards, All I want to know is, if I buy one of these routers, will it break my setup or not - ie. assuming I set up the relevant port-forwarding, can I expect any one-way audio issues. Can't get a definitive answer from suppliers or the manufacturer, so I hope someone here uses this model with Asterisk.? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Draytek Vigor 2900 Asterisk
Jonathan Attwood wrote: I'm in conversation with Draytek's pre-sales dept.. I bought a 2600 2 years ago and I had alot of NAT problem, because the SPI was changing the externhost (sip.conf) ip address with the local private address forwarding the packets, so the audio stream was failing. I sent all the debug logs to the draytek dev team, but they were slow on updates to I bought a new and different brand router. Hope they fixed that issue in the new firmwares Good luck Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Draytek Vigor 2900 Asterisk
Hi, Draytek 2900 is a great router. Easy to setup stable. I want known more detail of your network configuration. I can setup it and make some test. Regards, -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan Attwood Sent: Saturday, January 07, 2006 9:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Draytek Vigor 2900 Asterisk I'm in conversation with Draytek's pre-sales dept.. Here's the most recent reply: Hello, We really don't know of anyone who has run an Asterisk server on a Vigor2900. There are doubtless people around, but it's relatively rare. Most people don't run SIP servers. Regards, All I want to know is, if I buy one of these routers, will it break my setup or not - ie. assuming I set up the relevant port-forwarding, can I expect any one-way audio issues. Can't get a definitive answer from suppliers or the manufacturer, so I hope someone here uses this model with Asterisk.? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Draytek Vigor 2900 Asterisk
Now draytek have some SIP embeded router (e.g., 2100VG, 2900VG...). Maybe you can try these new router. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sergio Chersovani Sent: Saturday, January 07, 2006 9:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Draytek Vigor 2900 Asterisk Jonathan Attwood wrote: I'm in conversation with Draytek's pre-sales dept.. I bought a 2600 2 years ago and I had alot of NAT problem, because the SPI was changing the externhost (sip.conf) ip address with the local private address forwarding the packets, so the audio stream was failing. I sent all the debug logs to the draytek dev team, but they were slow on updates to I bought a new and different brand router. Hope they fixed that issue in the new firmwares Good luck Sergio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] wich IAX soft client allow to specify a differentserver port?
Try this. http://www.virbiage.com/firefly/download/firefly-thirdparty.exe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Antonio Gallo Sent: Saturday, January 07, 2006 8:20 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] wich IAX soft client allow to specify a differentserver port? I still having problem with remote SIP client, trying to use IAX client instead but i've to specify TCP port 8080 (because of firewall). I did this on server in bindport=8080 in iax.conf but i cannot find a soft client that allow to set wich server port to use. Any idea? Thanks, Antonio ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Cell phone dock/switch as Asterisk FXO source
No Bluetooth in the Samsung T309. I couldn't think of why I'd want BT... then of course I started looking at cell sockets, etc. after I got it and found several do not have a cable for the T309 yet. In hindsight, bluetooth would have made this easier. Live and learn! On 1/6/06, Jonathan Attwood [EMAIL PROTECTED] wrote: On 1/5/06, Brian McEntire [EMAIL PROTECTED] wrote: Wow! Thanks for all the responses! Very informative. Erik: I'm just looking for simple dial-out and pass-along incoming cell calls to *. Looks like the doc-n-talk should do it, except I checked with them and, silly me, the new Samsung t309 phone I just got is not supported yet. Hopefully it will be in a few months. Is it not supported, even with the Bluetooth module? (That's assumingthe phone's BTth)___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialer
Yes, I would be very interested in this as well. Thanks, Steve -Original Message- From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, January 06, 2006 4:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dialer Very cool! Is this something you can share the code? Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of trixter aka Bret McDanel Sent: Friday, January 06, 2006 2:17 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dialer On Fri, 2006-01-06 at 11:45 -0700, Wiley Siler wrote: Just to make it easy, I will be reading the caller list from a another server via a web page, parsing it and dialing. After each pass, I just post back to the server web page and it updates the other system. Our tech just needs to review the log once daily. That is basically what I did for a customer. I have a DB that is filtered pursuant to 47 CFR 64.1200 and 16 CFR 310 (US federal laws concerning these types of systems -- not calling to the US, dont worry about it). I wrote some tools to make that a snap. I then have 1-N clients pull from the DB servier via HTTP to get the next number to dial and context to goto. The dialplan updates the DB via HTTP so the status of a given number is known and prevents duplicate calls. I added answering machine detection to my asterisk server and a few other things to make the dialing slightly better. The way it works they can have many many calling systems if they need, nothing has to be local to each other. Reports can be generated off any data that is available (timestamps of events, status of calls, etc). This is perfect for dr appt reminders, batch calls saying 'your product has been shipped' etc. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialer
Darren, I am interested in your project. Let me know if I can help you test. Thanks, Steve -Original Message- From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, January 06, 2006 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dialer If this or any other example is available, I would be most thankful to have it. I got the go ahead on this project to day so now I have to start seeing how to do this. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Tuesday, January 03, 2006 5:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dialer I'm supposed to have a mostly canned script that will do this done already. It will pull the list of people to call out of a db and play them the file specified in the db table. Contact me offlist if you're interested. It will be done real soon but I'm not done testing yet. Darren Wiebe [EMAIL PROTECTED] Kerry Garrison wrote: You actually aren't far from it. If the system only needs to play the same file to each person, a simple script can be used to pull from a database and create call files. Asterisk will use the call files to place the calls and play a sound. A few minutes of searching on that should get you started. I haven't seen anyone else have a canned script ready to go, but would like to know if anyone does. -Kerry *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Wiley Siler *Sent:* Tuesday, January 03, 2006 3:32 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Dialer Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with no, this is absolutely not for doing call marketing. I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students are not logging on. We do this manually now but it would be easier and cheaper to just play them a message. What is the term I should be looking for? I keep thinking auto dialer or something like that but I am not quite getting there. Any help would be appreciated. I have been learning a bit of Perl so I was thinking I could auto generate and AGI file and then just do a Play() of the mp3 when they pick up at the other end? Seems a little kludge though. Thanks, Wiley --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wich IAX soft client allow to specify a different server port?
On Sat, Jan 07, 2006 at 01:19:34PM +0100, Antonio Gallo wrote: I still having problem with remote SIP client, trying to use IAX client instead but i've to specify TCP port 8080 (because of firewall). The IAX protocol is based on UDP, not TCP. I did this on server in bindport=8080 in iax.conf but i cannot find a soft client that allow to set wich server port to use. Any idea? Iaxcomm should work. You can use a complete dial string with username, secret, peer, port number, extension and context if you like, in the following format. [user[:secret[EMAIL PROTECTED]peer[:portno][/exten[@context]] /Mikael ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 3RD REQUEST - Any Help Is Appreciated
-- Executing Goto(Zap/2-1, ext-local|*101|1) in new stack I think the problem here is that you have the timeout set to one second and I am not sure what the * is before 101. My interpretation is ext-local specifies local context. *101 means dial extension 101 but I am unsure of what the * is for. And the final 1 means a one second timeout. -or- I don't know. Thanks, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Fax, txfax -bizarre thing
Hi, I having similar problem. Unfortunately each thread is archive leads to nowhere. I read a post in which similar problem was solved by changing rxgain and txgain to 15. Maybe this would help. Does anyone have common problems? I was wondering why asterisk - great telecommunication program - has such a weak fax support. I am talking about mail to fax and fax to mail. Or maybe I am the only one who has the problems with it. If someone has some experience please help. Best wishes Andrutto -- Kliknij po wiecej! http://link.interia.pl/f18ed ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pattern matching in dialplan problems matching _NNN
Hi,
I have a problem with pattern matching N what should digit 2 to 9
in Asterisk 1.2.1.
If I dial 220 I did not get an PlayBack of invalid. Asterisk jumps into the
context dialout and find there an matching _2. and is using this.
If I change _NNN to _XXX everything works fine. If I dial 220 I hear playtones
invalid. It seemed to be that pattern matching with N is not working as
designed.
Any ideas?
best regards
Thomas
extensions.conf
-
[internal]
#include /etc/asterisk/x_internal.conf
x_internal.conf
-
exten = 210,1,Macro(internalsqldial_stand,${EXTEN})
exten = _NNN,1,NoOp(wrong number dialed)
exten = _NNN,n,PlayBack(invalid)
exten = _NNN,n,Hangup()
include = dialout
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Re: [Asterisk-Users] Voice mail messages aren't sent to e-mail
I added: mailcmd=/usr/bin/sendmail -f hostname -t to the voicemail.conf file under [general] - James On Jan 6, 2006, at 10:31 PM, Pisac wrote: Yes, I found that this is problem with my server. Second server is connected through second provider, and first server and my domain is hosted at fist provider. My (first) provider has some stupid logic that reject e-mails from mailservers which don't have public hostname but private (my second server has server.local), but accepting all e- mails from it's IP address space (first server). So, my temporary solution was that I set up fake (but existing) hostname for second server (gmail.com), and now my (first) provider accepting e-mails. Very stupid. How you changed mailcmd to add a -f? Did you used nail/mail instead of sendmail, in voicemail.conf? Or maybe some .c source changing? Thanks Pisac I had a similar problem, but I was able to see the message getting rejected to rr.com because they were looking up the hostname pbx and rejecting it. I changed the mailcmd to add a -f realhostname.com and it started working. - James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problens to link 2 * servers
network problems. Asterisk wan unable to connect or bind to 10.0.0.121/100 Regards On 1/7/06, Cleyverson P. Costa [EMAIL PROTECTED] wrote: Hello, I'm traying to link 2 * servers using SIP and the following errors was show: SIP/AsteriskA:[EMAIL PROTECTED]/100) in new stack Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such host: 10.0.0.121/100 Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time Dec 13 22:47:07 NOTICE[8767]: rtp.c:435 ast_rtp_read: RTP: Received packet with bad UDP checksum Dec 13 22:47:07 WARNING[8767]: pbx.c:1949 ast_pbx_run: Timeout, but no rule 't' in context 'internal' linux*CLI Someone could help me to fix the following problens?? fraternatly, Cleyverson Pereira Costa Phone #: 27+9922-0111 Skype: cleyverson MSN: [EMAIL PROTECTED] Escolhe, pois, a vida, para que vivas, tu e a tua descendência. Deuteronômio 30:19 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Fax, txfax -bizarre thing
andrutto wrote: I was wondering why asterisk - great telecommunication program - has such a weak fax support. Because it's a PBX and not a fax server. Use IAXmodem and HylaFAX, and then you have a fax server. http://sourceforge.net/projects/iaxmodem http://hylafax.sourceforge.net/ Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pattern matching in dialplan problems matching _NNN
On 07/01/06, Thomas [EMAIL PROTECTED] wrote: Hi, I have a problem with pattern matching N what should digit 2 to 9 in Asterisk 1.2.1. If I dial 220 I did not get an PlayBack of invalid. Asterisk jumps into the context dialout and find there an matching _2. and is using this. If I change _NNN to _XXX everything works fine. If I dial 220 I hear playtones invalid. It seemed to be that pattern matching with N is not working as designed. 220 isn't supposed to match _NNN - N is digits 2-9, not 0. http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Fax, txfax -bizarre thing
andrutto wrote: Hi, I having similar problem. Unfortunately each thread is archive leads to nowhere. I read a post in which similar problem was solved by changing rxgain and txgain to 15. Maybe this would help. Does anyone have common problems? I was wondering why asterisk - great telecommunication program - has such a weak fax support. I am talking about mail to fax and fax to mail. Or maybe I am the only one who has the problems with it. If someone has some experience please help. There are patents related to fax to e-mail and e-mail to fax. These are probably US only patents, but the picture isn't very clear. Some large users of Asterisk and spandsp have been chased by the patent holders. In light of this, I certainly don't want to integrate fax-e-mail support into spandsp. Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Draytek Vigor 2900 Asterisk
Jonathan Attwood wrote: I'm in conversation with Draytek's pre-sales dept.. Here's the most recent reply: Hello, We really don't know of anyone who has run an Asterisk server on a Vigor2900. There are doubtless people around, but it's relatively rare. Most people don't run SIP servers. Regards, All I want to know is, if I buy one of these routers, will it break my setup or not - ie. assuming I set up the relevant port-forwarding, can I expect any one-way audio issues. Can't get a definitive answer from suppliers or the manufacturer, so I hope someone here uses this model with Asterisk.? ___ I have a 2900G behind a Cisco 1720 with dual ADSL WICs in one office, and a 2600VGi standalone in another (I don't use the 2600's built-in FXS ports -- they aren't very good - seem noisy). I have Asterisk servers in both offices, linked via IAX. I have incoming voip services going independently to both Asterisk servers. I've had no problems whatsoever -- everything has worked perfectly. The QoS facility in both routers allows you to reserve a certain amount of bandwidth (in or out) for IAX and SIP and this seems to work fine though it isn't necessary on our networks. I'm using port forwarding on both routers to route IAX and SIP to the private IPs of the Asterisk boxes. But you will need to open the appropriate ports on the firewall in the router, or firewall the Asterisk boxes and DMZ the Asterisk boxes. However, the new Dreytek 3300 series of routers is even more interesting. Multiple WAN ports for backup/load balancing, and optional hardware FXO/FXS ports. I hope this helps. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pattern matching in dialplan problems matching _NNN
thanks... _NXX works for me best regards Thomas On Saturday 07 January 2006 16:37, Peter Bowyer wrote: On 07/01/06, Thomas [EMAIL PROTECTED] wrote: Hi, I have a problem with pattern matching N what should digit 2 to 9 in Asterisk 1.2.1. If I dial 220 I did not get an PlayBack of invalid. Asterisk jumps into the context dialout and find there an matching _2. and is using this. If I change _NNN to _XXX everything works fine. If I dial 220 I hear playtones invalid. It seemed to be that pattern matching with N is not working as designed. 220 isn't supposed to match _NNN - N is digits 2-9, not 0. http://www.voip-info.org/wiki/index.php?page=Asterisk+Dialplan+Patterns Peter -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Fax, txfax -bizarre thing
Lee Howard [EMAIL PROTECTED] wrote: Use IAXmodem and HylaFAX, and then you have a fax server. http://sourceforge.net/projects/iaxmodem http://hylafax.sourceforge.net/ If you're looking for more general information on HylaFAX, see www.hylafax.org. -Darren ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Draytek Vigor 2900 Asterisk
It certainly does. How many rules can you create in the port forwarding section of the V2900? I was told that the V2900 has SIP_ALG. Is this something you've activated? On 1/7/06, Faris Raouf [EMAIL PROTECTED] wrote: Jonathan Attwood wrote: I'm in conversation with Draytek's pre-sales dept.. Here's the most recent reply: Hello, We really don't know of anyone who has run an Asterisk server on a Vigor2900. There are doubtless people around, but it's relatively rare. Most people don't run SIP servers. Regards, All I want to know is, if I buy one of these routers, will it break my setup or not - ie. assuming I set up the relevant port-forwarding, can I expect any one-way audio issues. Can't get a definitive answer from suppliers or the manufacturer, so I hope someone here uses this model with Asterisk.? ___ I have a 2900G behind a Cisco 1720 with dual ADSL WICs in one office, and a 2600VGi standalone in another (I don't use the 2600's built-in FXS ports -- they aren't very good - seem noisy). I have Asterisk servers in both offices, linked via IAX. I have incoming voip services going independently to both Asterisk servers. I've had no problems whatsoever -- everything has worked perfectly. The QoS facility in both routers allows you to reserve a certain amount of bandwidth (in or out) for IAX and SIP and this seems to work fine though it isn't necessary on our networks. I'm using port forwarding on both routers to route IAX and SIP to the private IPs of the Asterisk boxes. But you will need to open the appropriate ports on the firewall in the router, or firewall the Asterisk boxes and DMZ the Asterisk boxes. However, the new Dreytek 3300 series of routers is even more interesting. Multiple WAN ports for backup/load balancing, and optional hardware FXO/FXS ports. I hope this helps. Faris. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help Connecting server districts
I am working on a project to unite several local school districts. We will have 14 different districts, every district would have their own asterisk box on location. We all have fiber lines running to a central location at our isd. This provides connectivity to all the districts. 1. would it be wiser to install a asterisk box at the central location of the ISD and have all asterisk boxes register with that? One asterisk box at each district/school would be wiser, keeping local sip to sip calls on the LAN, and local community calls on local pstn facilities. There is no real need to register one asterisk to another. Simply define a single entry in iax.conf using a type=user, and an assoicated entry in the receiving * box with type=peer. That will minimize the register keep- a-live packets, etc. (That assumes your fiber network is very stable in terms of uptime.) or would it be better to create a mesh-network of registrations. Create a mesh network without the registrations (as noted above). The amount of traffic between districts is likely to be rather low given that a large percentage of communications is likely centered are each districts community of interest. (eg, local calls, room to office calls, etc.) If so can this be acomplished easily using iax or dundi? Yes, with iax very simple. No need for dundi in this case. What would be the best way to link all of these boxes together so that from any location they can call each other using simple extenion headers such as _10xxx for district 1, and _11xxx for distrct two and make that conssistant? Personally, I'd look at setting up your dialplan in such a way as to use real 7-digit or 10-digit telephone numbers in all cases. If school district #8 has local telephone number like 312-456-1000 (assuming no DID numbers), then use something like that in each remote school district's asterisk dialplan. So, if someone in district #12 dials 312-456-1000, the dialplan determines whether to send that call via iax to district #8 or overflow onto the pstn network. If you already have an extension number scheme in each school, the above can be augmented with something like 312-456-1xxx (even though they aren't DID's right now). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Fax, txfax -bizarre thing
I having similar problem. Unfortunately each thread is archive leads to nowhere. I read a post in which similar problem was solved by changing rxgain and txgain to 15. Maybe this would help. Does anyone have common problems? I was wondering why asterisk - great telecommunication program - has such a weak fax support. I am talking about mail to fax and fax to mail. Or maybe I am the only one who has the problems with it. If someone has some experience please help. In addition to what others have already mentioned, fax support requires excellent * timing bus characteristics, and the digium tdm card does not provide that (nor does the older x100p). Those that have spandsp running are primarily T1/E1 users. Running fax over sip channels (and networks) also requires rather tight requirements in terms of latency, jitter, etc, that has been discussed many times on this list. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transfer application
Bill Michaelson wrote:
I am having trouble understanding how to use this. I want to transfer
certain incoming calls from an IAX ITSP based on caller ID. From what I
can make of the docs, I thought I need to do something like this...
exten = _NXXNXX,n(nocid),transfer(1000)
exten = _NXXNXX,n,noop(boo,${TRANSFERSTATUS})
exten = _NXXNXX,n,hangup
exten = 1000,1,Dial(IAX2/jnctn_out/16665551234,45,t)
exten = 1000,n,hangup
Why don't you just do:
exten = _NXXNXX,n(nocid),Dial(IAX2/jnctn_out/16665551234,45,t)
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] RE: Fax, txfax -bizarre thing
Hi,I certainly don't want to integrate fax-e-mail support into spandsp.I think our problem is not connected with spandsp and fax - email integration. All the applications I mean spandsp txfax and rxfax are enough to have emial - fax functionality in Asterisk. I wrote a program which allows me to convert emial to tiff, it is also not a problem to turn Asterisk to be a small email server (I use qmail). Only problem is with the txfax (or I don't with what, maybe with synchronization to my Telco). When Asterisk launch the txfax application sometimes (quite often) something cause it to hangup the Zap channel and fax transmission goes to the space!. On http://soft-switch.org/spandsp_faq/ar01s08.html I found that this can be cause by the synchronization problem, correct me if I'm wrong. But there is a little, that I can do about it. What should I change to be synchronized with Telco (linux version, hardware or.) Of course that's not a problem to use hylafax, but I just want to have it on one machine (I'm afraid that Asterisk and hylafax won't run on the same machine :( )And one more thing. I use fax machine connected to TDM400P I can receive and send faxes without problems. I get crazy things Best wishesAndrew Nowrot ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Non-PRI T1
Any type of circuit available as an analog line is also available over a T1. It just minimizes the amount of copper required to deliver service. You must look at you original order from your telephone company to determine the type of circuits they are delivering. They may be POTS 1FB in which case there are no digitis. They may be EM with none. Or they may be DID which would be sending digits. Although only the order would identify the exact number of digits. If they are DID then you were probably assignen at least one block of 20 numbers to be delivered on the circuit. Of course then you need to know how many trunks/circuits you have active on the T1. For outgoing calls you control which ones to use for any given call. for incoming they may be divided into multiple trunk groups with different numbers coming in on different channels. If they are non DID circuits then they should hit the s exten in whatever context you have defined for them in zapata.conf on an incoming call. Good luck On Jan 6, 2006, at 6:02 PM, O'Connor, Jonathan wrote: You can use a normal T1. I have one between an Asterisk box and a Vodavi switch. I use 10 channels between them with EM signalling. Zaptel.conf: loadzone= us defaultzone = us span=1,1,0,esf,b8zs #bchan=1-10 # set this to 1-15,17-31 for E1 #dchan=24 # set this to 16 for E1 em=1-10 #fxsgs=1-10 Then Zapata.conf has: immediate=no overlapdial=yes switchtype=national signalling=em_w emdigitwait=1000 channel=1-10 You have to be careful of timing and such, because basically its really no fancier then a set of lines that happen to be on a T1. It opens and line and literally sends 4 DTMF codes to the server to tell it where its calling. For all intents and purposes it could be a bundle of 10 normal phone lines. It is our only site using this method because they wanted a fortune for a PRI card. Our Avaya and Nortel PBXs all talk to their Asterisk PBXs using PRI, MUCH easier. The basic T1 signalling and EM Wink is truly horrid to work with when the PBX is as dumb as a box of rocks like our Vodavi system. -Jonathan Jonathan O'Connor Senior System Administrator Inoveris LLC Direct Line (614) 791-3742 Fax (614) 791-3748 Helpdesk 866-456-1566 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rusty Shackleford Sent: Friday, January 06, 2006 6:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Non-PRI T1 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Phillips Sent: Friday, January 06, 2006 3:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Non-PRI T1 Are they configured for inbound calls? If so how? Usually the telco sends the last 4 digits of the called phone number down the line. Uhm, don't you need PRI signalling for that? -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.371 / Virus Database: 267.14.14/222 - Release Date: 01/05/2006 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk initialization
Do not know what version you are running, But there are a few ways to do this. There is a persistant setting: from agents.conf ;; Define whether callbacklogins should be stored in astdb for; persistence. Persistent logins will be reloaded after; Asterisk restarts.;persistentagents=yes If you want to handle it outside of Asterisk via an AGI you can have your AGI execute: AgentCallbackLogin([AgentNo][|[options][|[EMAIL PROTECTED]): this is providing that you have the information saved in your DB. Personal Opinion: Use the builtin features with the persistentagents options and use the php script in the contribs directory to see who is on. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dov BigioSent: Friday, January 06, 2006 4:24 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Asterisk initialization Hi, I am doing an AGI that logs to a database every Agent login/logoff. My idea is to be able to go to this database and check which agents where logged so that I can force their login in case Asterisk goes down for some reason. The problem is that I would need to reload their status from this AGI when Asterisk initializes. Is there a way to do this? One idea I had was to make safe_asterisk to generate a .call file that calls and extension that would call the AGI to log all the agents back on. Is there another way of running an AGI on initialization? Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help Connecting server districts
I would agree with all but a few issue: I would incoparate dundi, After using it I have fallen in love with it for distributed applications such as this. It makes configuration at first a bit steeper but it picks up monentum as your deploy. With Dundi you basicaly broadcast what extensions or numbers are served by your machine and using a set of keys (which negats having to configure a perr for every machine to create a mesh netowrk) There is no need to connect to the Public Dundi Peering fabric unless you want to. You can run your own 'private' Dundi peers. Let me know if you need any help.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Saturday, January 07, 2006 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Help Connecting server districts I am working on a project to unite several local school districts. We will have 14 different districts, every district would have their own asterisk box on location. We all have fiber lines running to a central location at our isd. This provides connectivity to all the districts. 1. would it be wiser to install a asterisk box at the central location of the ISD and have all asterisk boxes register with that? One asterisk box at each district/school would be wiser, keeping local sip to sip calls on the LAN, and local community calls on local pstn facilities. There is no real need to register one asterisk to another. Simply define a single entry in iax.conf using a type=user, and an assoicated entry in the receiving * box with type=peer. That will minimize the register keep- a-live packets, etc. (That assumes your fiber network is very stable in terms of uptime.) or would it be better to create a mesh-network of registrations. Create a mesh network without the registrations (as noted above). The amount of traffic between districts is likely to be rather low given that a large percentage of communications is likely centered are each districts community of interest. (eg, local calls, room to office calls, etc.) If so can this be acomplished easily using iax or dundi? Yes, with iax very simple. No need for dundi in this case. What would be the best way to link all of these boxes together so that from any location they can call each other using simple extenion headers such as _10xxx for district 1, and _11xxx for distrct two and make that conssistant? Personally, I'd look at setting up your dialplan in such a way as to use real 7-digit or 10-digit telephone numbers in all cases. If school district #8 has local telephone number like 312-456-1000 (assuming no DID numbers), then use something like that in each remote school district's asterisk dialplan. So, if someone in district #12 dials 312-456-1000, the dialplan determines whether to send that call via iax to district #8 or overflow onto the pstn network. If you already have an extension number scheme in each school, the above can be augmented with something like 312-456-1xxx (even though they aren't DID's right now). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Latency
That would depend heavaly on your netowrk. Would your Swtiches (not routers as TMDoE is layer 2) I pulled up an old posting from Mark on TDMoE. http://www.marko.net/asterisk/archives/0301/0566.html -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Friday, January 06, 2006 5:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Latency Does anyone know if using TDMoE instead of straight SIP between servers would increase or decrease latency? Aaron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording Calls at the phone
A (too) simple sollution to your problem is to take the analog audio from your IP phone using a module atached between the curly handset cord and the base unit of the IP phone - like http://www.quasarelectronics.com/tre156.htm So, basically you need to change the old RJ11 - 1/8 inch recording - RJ11 system you have used to a new one with RJ10 - 1/8 inch recording - RJ10. Sure, this solution works only if the handeset it is attached through a RJ10 port to the handset. I do not know exactly how your software will deal with this change as there should be a mechnism to start stop recording based on the audio level injected into PC's audio card (mic port). Hope it helps. Ioan Indreias Modulo Consulting - http://www.modulo.ro I'm not really trying to monitor anything on the asterisk box at all. I guess this is more of an SIP phone question. Really all I need is to get the audio from an SIP phone, both the caller and callie, to a 1/8th inch stereo jack that I can plug into a mic input. Michael Sampson Information Systems Manager Customer Contact Services [EMAIL PROTECTED] 952-936-4000 Douglas Garstang wrote: On Demand-monitoring? If your referring to monitoring specific agents calls, I'm still trying to work out how to do that. You can either monitor all calls for a queue, or all calls for all agents, but not all calls for a specific agent. I tried to use the Monitor() command on it's own to start recording when an agent receives a call, but that does not appear to work. -Original Message- From: Francesco Peeters (Asterisk) [mailto:[EMAIL PROTECTED] Sent: Friday, January 06, 2006 7:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Recording Calls at the phone On Fri, January 6, 2006 15:37, Michael Sampson said: I work for a call center and we are looking at using asterisk to have our operators take calls. Our message taking software records all the calls on the operators computers. Right now we use these recording controls from radio shack that plug in between the wall jack and the phone and plug in via a 1/8 inch stereo connector to the mic input on the computer. If I buy an IP phone I can't do that. I could get an FXO adapter and regular phones, but I'm looking to get as little equipment as possible. Radio shack makes a recording control that plugs in to a 2.5 mm headset jack, but it takes batteries so thats not going to work Does anyone else do something similar? Does anyone have any ideas about what producs/setup would work for this. Asterisk has a built in monitoring system. You can chose to do Always, Never or On Demand monitoring, depending on your setup and dialplan http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor Good luck! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dialer
http://www.astpp.org/index.php?n=Misc.AutoDialOut I put together what I have on that site. Darren wiebe [EMAIL PROTECTED] Steve Totaro wrote: Darren, I am interested in your project. Let me know if I can help you test. Thanks, Steve -Original Message- From: Wiley Siler [mailto:[EMAIL PROTECTED] Sent: Friday, January 06, 2006 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Dialer If this or any other example is available, I would be most thankful to have it. I got the go ahead on this project to day so now I have to start seeing how to do this. Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Tuesday, January 03, 2006 5:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dialer I'm supposed to have a mostly canned script that will do this done already. It will pull the list of people to call out of a db and play them the file specified in the db table. Contact me offlist if you're interested. It will be done real soon but I'm not done testing yet. Darren Wiebe [EMAIL PROTECTED] Kerry Garrison wrote: You actually aren't far from it. If the system only needs to play the same file to each person, a simple script can be used to pull from a database and create call files. Asterisk will use the call files to place the calls and play a sound. A few minutes of searching on that should get you started. I haven't seen anyone else have a canned script ready to go, but would like to know if anyone does. -Kerry *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Wiley Siler *Sent:* Tuesday, January 03, 2006 3:32 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Dialer Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminology clear. First off let me premise this with no, this is absolutely not for doing call marketing. I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if students are not logging on. We do this manually now but it would be easier and cheaper to just play them a message. What is the term I should be looking for? I keep thinking auto dialer or something like that but I am not quite getting there. Any help would be appreciated. I have been learning a bit of Perl so I was thinking I could auto generate and AGI file and then just do a Play() of the mp3 when they pick up at the other end? Seems a little kludge though. Thanks, Wiley --- - ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SPA-3000 is translating vocal sounds into DTMF
Rich Adamson wrote: Its a fairly common problem with the spa3k, but is somewhat dependent on how it is configured and the distance to your CO. Just to add a couple of data points: I don't know why, but for me the problem has been worse lately than it had been during the early time I spent with this unit. I have been using it pretty exclusively for about a month and a half; only in the past few days have those tones really become objectionable. Second, I'm not *any* distance from the CO: my FXO is the port on an ATA186 (original issue, 2002 vintage) from Vonage. Finally, for sure some of the calls on which I'm experiencing the problem aren't even going out the FXO port on the thing; they are ITSP calls made using the FXS port only. FWIW. B. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Fax, txfax -bizarre thing
Of course that's not a problem to use hylafax, but I just want to have it on one machine (I'm afraid that Asterisk and hylafax won't run on the same machine :( ) [Colin Anderson]I am experimenting with IAXmodem to Hylafax running on an Asterisk server. It works. Last Thur, I had 98 virtual modems recognized and running under Hylafax. So far, I can send and recieve faxes reasonably well but there's some configuration issues I have to get out of the way before I would beta it on my users. I expect that Hylafax to a couple of plain old USR modems running on /ttys0 and /ttys1 would work fine enough even if Asterisk was on the box. Even though the postscript conversion is done on the server that can be controlled with -nice.After that all that Hylafax has to do is service the modems and the clients.This seems to be pretty low overhead - I think the client protocol is FTP on a nonstandard port, and servicing a couple of modems at 9600 can't be too taxing. Hell, 25 modems shouldn't be taxing, on a modern machine. BTW I have used, installed, admin'd etcabout a dozen big Windows faxing solutions(basically all of the big players) and I've never used Hylafax before, and I was really impressed. It's better than 80% of the Windows product offerings, IMO. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to Unload app_rxfax.so
Hello All, Dunno what happen but Asterisk is refusing to start... Went over the log and found out that app_rxfax.so is failing to load. Jan 7 11:57:28 VERBOSE[4320] logger.c: [app_rxfax.so]Jan 7 11:57:28 WARNING[4320] loader.c: /usr/lib/asterisk/modules/ app_rxfax.so: undefined symbol: fax_set_phase_d_handler Jan 7 11:57:28 WARNING[4320] loader.c: Loading module app_rxfax.so failed! Is there any way to bypass this module and start Asterisk... I think it was a bad idea to compile Asterisk with fax capability... Thanks, Neal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to Unload app_rxfax.so
Yes, you could do that making some changes on modules.conf noload = app_rxfax.so Regards Alberto Nitesh Divecha wrote: Hello All, Dunno what happen but Asterisk is refusing to start... Went over the log and found out that app_rxfax.so is failing to load. Jan 7 11:57:28 VERBOSE[4320] logger.c: [app_rxfax.so]Jan 7 11:57:28 WARNING[4320] loader.c: /usr/lib/asterisk/modules/ app_rxfax.so: undefined symbol: fax_set_phase_d_handler Jan 7 11:57:28 WARNING[4320] loader.c: Loading module app_rxfax.so failed! Is there any way to bypass this module and start Asterisk... I think it was a bad idea to compile Asterisk with fax capability... Thanks, Neal ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Problens to link 2 * servers
From this server can you ping 10.0.0.121? What is your network mask? 10.0.0.121/100 is not a valid address (mask are in the range of /0 to /32) This is where you should start. What is your network definition? Tudo bem? Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Moises Silva Sent: Saturday, January 07, 2006 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Problens to link 2 * servers network problems. Asterisk wan unable to connect or bind to 10.0.0.121/100 Regards On 1/7/06, Cleyverson P. Costa [EMAIL PROTECTED] wrote: Hello, I'm traying to link 2 * servers using SIP and the following errors was show: SIP/AsteriskA:[EMAIL PROTECTED]/100) in new stack Dec 13 22:46:57 WARNING[8767]: chan_sip.c:1398 create_addr: No such host: 10.0.0.121/100 Dec 13 22:46:57 NOTICE[8767]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time Dec 13 22:47:07 NOTICE[8767]: rtp.c:435 ast_rtp_read: RTP: Received packet with bad UDP checksum Dec 13 22:47:07 WARNING[8767]: pbx.c:1949 ast_pbx_run: Timeout, but no rule 't' in context 'internal' linux*CLI Someone could help me to fix the following problens?? fraternatly, Cleyverson Pereira Costa Phone #: 27+9922-0111 Skype: cleyverson MSN: [EMAIL PROTECTED] Escolhe, pois, a vida, para que vivas, tu e a tua descendência. Deuteronômio 30:19 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] choppy music on hold - only on PRI PSTN
Hello to all I do not know what is causing choppy music on hold when call comes in through E1 card (PRI).. but this channel info is somehow strange.. We use Alaw over PRI (and I think its format number 8), But why is WriteFormat at 2 ? Thanks! show channel Zap/1-1 -- General -- Name: Zap/1-1 Type: Zap UniqueID: 1136667936.0 Caller ID: 04573573 Caller ID Name: (N/A) DNID Digits: 349 State: Up (6) Rings: 1 NativeFormat: 72 WriteFormat: 2 ReadFormat: 8 1st File Descriptor: 14 Frames in: 3516 Frames out: 3352 Time to Hangup: 0 Elapsed Time: 0h1m10s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: OZ0800 Extension: s Priority: 7 Call Group: 0 Pickup Group: 0 Application: Queue Data: OZ0800|Tt|||300 Blocking in: ast_waitfor_nandfds ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Immediate routing on 0 (DNIS)?
I've got a T1 (EM wink). Our four-digit inbound DNIS numbers are in the range of 0600 - 1699. However, the second that the 0 is seen on an in-bound 06xx call, it stops listening for any more digits, and immediately tries to route the call. My 16xx numbers wait for all four digits before trying to route. Is there something, somewhere, that tells it to do an immediate route on seeing 0? I don't have much of anything in my extensions.conf file. I'm seeing what's going on via tail -f /var/log/asterisk/full Any suggestions? Thanks! -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Immediate routing on 0 (DNIS)?
Ken D'Ambrosio wrote: I've got a T1 (EM wink). Our four-digit inbound DNIS numbers are in the range of 0600 - 1699. However, the second that the 0 is seen on an in-bound 06xx call, it stops listening for any more digits, and immediately tries to route the call. My 16xx numbers wait for all four digits before trying to route. Is there something, somewhere, that tells it to do an immediate route on seeing 0? I don't have much of anything in my extensions.conf file. I'm seeing what's going on via tail -f /var/log/asterisk/full Any suggestions? Thanks! -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Post your extensions.conf and what's on the CLI (asterisk -r) -- Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Possible bug with GotoIfTime
Running a fairly recent subversion release of Asterisk, I'm running into a problem using labels (as opposed to priorities) with this application. Here is the dialplan segment: ; isolate gotoiftime bug with labels ;exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4) exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?light:dark) exten = 806,n(light),noop(light) exten = 806,n,hangup exten = 806,n(dark),noop(dark) exten = 806,n,hangup As coded, this is what happens when it executes: -- Executing GotoIfTime(IAX2/hack-2, 8:00-20:00|*|*|*?light:dark) in new stack Jan 7 18:38:09 NOTICE[28137]: pbx.c:1705 pbx_extension_helper: No such label 'light:dark' in extension '806' in context 'default' Jan 7 18:38:09 WARNING[28137]: pbx.c:6312 ast_parseable_goto: Priority 'light:dark' must be a number 0, or valid label == Spawn extension (default, 806, 1) exited non-zero on 'IAX2/hack-2' -- Hungup 'IAX2/hack-2' But if I disable the second exten line instead of the first, it works properly. Beware. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Jobs
I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? - Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Draytek Vigor 2900 Asterisk
Hiya, I've got a 2900g series, and it works fine (I have the 2200we before I upgraded and that was ok too!). I have used its built in wifi to go to an ipaq iax extension, and also have asterisk doing sip and iax through to fwd and sipgate. There's some port forwarding rules to get the protocols through to my asterisk box and that's about it! From what I may have heard (although I've never tried this) the 'V' or voice version have to terminate SIP calls on itself - i.e. you cant pass the ports through to another SIP box - similarly the analogue ports have to go to the router. Like I say - I've never tried this - its what I found out from the owners forums. HTH. Wayne. Jonathan Attwood wrote: I'm in conversation with Draytek's pre-sales dept.. Here's the most recent reply: Hello, We really don't know of anyone who has run an Asterisk server on a Vigor2900. There are doubtless people around, but it's relatively rare. Most people don't run SIP servers. Regards, All I want to know is, if I buy one of these routers, will it break my setup or not - ie. assuming I set up the relevant port-forwarding, can I expect any one-way audio issues. Can't get a definitive answer from suppliers or the manufacturer, so I hope someone here uses this model with Asterisk.? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Possible bug with GotoIfTime
Bill Michaelson wrote: Running a fairly recent subversion release of Asterisk, I'm running into a problem using labels (as opposed to priorities) with this application. Here is the dialplan segment: ; isolate gotoiftime bug with labels ;exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?2:4) exten = 806,1,GotoIfTime(8:00-20:00,*,*,*?light:dark) exten = 806,n(light),noop(light) exten = 806,n,hangup exten = 806,n(dark),noop(dark) exten = 806,n,hangup As coded, this is what happens when it executes: -- Executing GotoIfTime(IAX2/hack-2, 8:00-20:00|*|*|*?light:dark) in new stack Jan 7 18:38:09 NOTICE[28137]: pbx.c:1705 pbx_extension_helper: No such label 'light:dark' in extension '806' in context 'default' Jan 7 18:38:09 WARNING[28137]: pbx.c:6312 ast_parseable_goto: Priority 'light:dark' must be a number 0, or valid label == Spawn extension (default, 806, 1) exited non-zero on 'IAX2/hack-2' -- Hungup 'IAX2/hack-2' But if I disable the second exten line instead of the first, it works properly. Beware. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users GotoIfTime(8:00-20:00|*|*|*?light:dark|s|1) -- . -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- . signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how to configure iax account for iaxmodem?
Hi, I'm running asterisk 1.2.1 and started to play with iaxmodem 0.0.7 running on the same box. I wonder how to setup the iax account correctly so that I may initiate outbound calls via iaxmodem? registration upon iaxmodem startup is okay and I can direct calls to it. -- Registered IAX2 'iaxmodem' (AUTHENTICATED) at 127.0.0.1:33874 But upon an outbound call setup request from iaxmodem (ATD123) I get the following asterisk error: Jan 8 01:06:29 NOTICE[10273]: chan_iax2.c:6778 socket_read: Rejected connect attempt from 127.0.0.1, who was trying to reach '123@' What is missing in the iax.conf to make asterisk accept the outbound calls? [iaxmodem] type=peer callerid=iaxmodem username=iaxmodem secret=password host=dynamic disallow=all allow=alaw peercontext=from-internal context=from-internal TIA, Bruno ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wich IAX soft client allow to specify a different server port?
[user[:secret[EMAIL PROTECTED]peer[:portno][/exten[@context]] Well but i don't need to dial out, i need to register to asterisk using IAX and 8080 port and all the client i've tested will not allow that into their account config section: they just have the server name/ip not the port. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to configure iax account for iaxmodem?
Bruno Voigt wrote: Hi, I'm running asterisk 1.2.1 and started to play with iaxmodem 0.0.7 running on the same box. I wonder how to setup the iax account correctly so that I may initiate outbound calls via iaxmodem? registration upon iaxmodem startup is okay and I can direct calls to it. -- Registered IAX2 'iaxmodem' (AUTHENTICATED) at 127.0.0.1:33874 But upon an outbound call setup request from iaxmodem (ATD123) I get the following asterisk error: Jan 8 01:06:29 NOTICE[10273]: chan_iax2.c:6778 socket_read: Rejected connect attempt from 127.0.0.1, who was trying to reach '123@' What is missing in the iax.conf to make asterisk accept the outbound calls? [iaxmodem] type=peer callerid=iaxmodem username=iaxmodem secret=password host=dynamic disallow=all allow=alaw peercontext=from-internal context=from-internal Hrmmm... iaxmodem registers properly but gets a rejection when trying to place a call. It's probably a codec problem. Try adding allow=ulaw and allow=slinear to your iaxmodem context and see if that helps. Lee. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Jobs
Most of the Asterisk work I have found out and about is either done by internal staff or by companies wanting work done by external contractors. Like you, I have found very little in the way of full time jobs for 'asterisk people' PaulH - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 08, 2006 10:47 AM Subject: [Asterisk-Users] Asterisk Jobs I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? - Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Market Share
You could probably pay $15-20 for a paul budde report with relatively accurate figures. www.budde.com.au (even if he does believe asterisk is a passing fad - hi Paul :) he's still one of the best telco resources in Australia. Telsyste might be another option. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dakota Sent: Saturday, 7 January 2006 8:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Asterisk Market Share Does anyone know where I can find some stats on the following The % market share for the following types of Ip PBX Mitel Nortel Asterisk etc Thanks! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Kudzu and Zaptel Cards
Redhat has a 'Hardware Discovery Utility' called Kudzu. When I change cards, kudzu pops up and ask to remove/config the card. Most of the time kudzu has trouble recognizing the Digium Zaptel cards and calls them something wrong, like calling the TDM card a network card. I'm having a devil of a time getting 3 TE410Pcards to come up with all green lights. For example one or two cards full green, and the other has one red and yellow. Swap cards give me some other form of workingness. My questionare: 1) How necessary is Kudzu? 2) Should it ran at all? 3) If I choose ignore or disable kudzu, will it stop the zaptel cards from being detected or working? Any hints will be welcome TIA Bart ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Up to 4 seconds delay to play prompt?
Hi, Some background... I have the following directories: /var/lib/asterisk/sounds/custom/ - Here are french prompts /var/lib/asterisk/sounds/custom/en - Here are the english prompts If I do: SetLanguage(en) Playback(custom/mypromp) The prompt file is played immediatly If I do: SetLanguage(fr) Playback(custom/myprompt) There is a good 4 seconds delay before tha prompt is played... Any hints? Andre Courchesne ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Jobs
Asterisk is still virtually unknown to endusers. The only reason why it's even a blip on the radar of PBX manufacturers is because how quickly the community is growing, and how feature rich the system is already. The biggest threat is that it is free and not proprietary which totally flies in the face of the whole tradition of the greedy industry. They see what happened to the music industry when it moved too slowly and did not anticipate the market paradigm shift. 3com seems to have knowledge and insight into the future, at least that is my take on the V3000. A 1U IP PBX with one FXS port and four FXOs at a slightly higher price to a decent Asterisk box. I could easily sell one of these boxes to an end user for slightly more than a comparable Asterisk rig based on name recognition, the fact that they can call any number of dealers in the area for support, the GUI, and the nice glossy brochures! If I demo the system, it is a no-brainer for the bean counters, the suits, and even the overburdened techs. Of course I would mention that the system is very expandable and all you have to do is plug in a phone and it will be ready to go automagically. What I would leave out is the fact that if, lets say you wanted to upgrade to a T1/E1 you would have to buy a different several U sized chassis and a card that will probably set them back about $4k or so. Also, I would not mention that they were locked into 3com phones and that besides the high price of the phone, they will need to also buy a license for it to work. I think I read that Digium did something like $20 or $80 million in business last year (obviously I am not sure of the figure but it was impressive and more than I would have guessed for them). That is a nice chunk of change, but it is chump change to the overall PBX industry. It is like the penny that someone drops and doesn't even bother to pick up. Looking quickly, I found this reference, The PBX market at $13.2 billion in 2002 is forecast to reach $17.9 billion by 2008 during the forecast period. http://www.researchandmarkets.com/reportinfo.asp?report_id=34161 . It is no wonder why you see very few jobs listed for Asterisk skilled workers compared to the real market share holders. The numbers just aren't there. There are plenty of consultants around the globe that can SSH into a box on the other side of the world and configure it. If I was in the market to hire an Asterisk consultant, I would watch the list and see who knows what the heck they are talking about and has a good attitude or I would look at the wiki. So far I have a few guys I would call on if I needed some work done that I know would be high quality. I would call on Nicolas Gudino (the creator of AMP), Darren Wiebe (who knows all about pre-paid, post paid, and what I am especially looking forward to, the integration of OSCommerce to his platform, or the Coalescent Systems guys. I have some others but I keep them secret so they are available when I need them. I also have a feeling that most Asterisk jobs are self created. Will Asterisk get a chunk of the market share? Probably some years down the road but I will not hold my breath. Not until there is something similar to the marketing, documentation, sales channels, and name recognition. None of this will be easy since anyone can learn Asterisk and start a company. It is not like NEC for instance, where there can only be so many distributors in an area and they must be certified. If they screw up, they lose their distributorship. The Open Source business model makes it almost impossible to emulate the same type of marketing, sales, and business model. Thanks, Steve I am going to cross post this to the biz list, since that is really where it belongs. Most of the Asterisk work I have found out and about is either done by internal staff or by companies wanting work done by external contractors. Like you, I have found very little in the way of full time jobs for 'asterisk people' PaulH - Original Message - From: Douglas Garstang [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, January 08, 2006 10:47 AM Subject: [Asterisk-Users] Asterisk Jobs I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? - Doug. -- -- ___
Re: [Asterisk-Users] Screening incoming calls.
Thanks Sorry, I missed that local/8600 channel. On 1/6/06, Philipp von Klitzing [EMAIL PROTECTED] wrote: Hi! Thanks for that post thats a good one :-) just one thing, what happens if the user doesn't want to connect to the caller? does it get saved as VM? Looking thru the code I couldn't see where that happens. The 1st MeetMe has three participants: caller AGI .call file -- VoiceMail() AGI .call file -- VM box owner (listen-only mode) VoiceMail() simply records the MeetMe conference. Cheers, Philipp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Jobs
Douglas Garstang wrote: I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? I think that the Asterisk customer profile can shed some light on this. If you are a big company, you'll buy into an expensive system because you can afford it and rely on it. If you are a small company, you will look to Asterisk as an inexpensive way to set up your telephone system. You will also likely have staff that is willing to work with it and not enough money or need to hire external consultants exclusively for Asterisk. You may have a telecom or networking consultant that will put together the network and set up the system but Asterisk is a small piece of it. I'd say Asterisk is more of a plus in a job description but not a requirement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Some advice on routing DID's
Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote server. Thanks, -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some advice on routing DID's
I have written an agi script that I use for that. Then I can just have a list of dids and extensions in a db. Tom Vile wrote: Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote server. Thanks, -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some advice on routing DID's
It's funny you mentioned that Darren, I was looking at your scripts today. I will evaluate it some more. On 1/7/06, Darren Wiebe [EMAIL PROTECTED] wrote: I have written an agi script that I use for that. Then I can just have a list of dids and extensions in a db. Tom Vile wrote: Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote server. Thanks, -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Immediate routing on 0 (DNIS)?
Post your extensions.conf and what's on the CLI (asterisk -r)
As requested:
# cat /etc/asterisk/extensions.conf
[incoming]
exten = s,1,Answer()
exten = s,n,NoOp(CallerID is ${CALLERID})
exten = s,n,NoOp(DID is ${DNID})
exten = s,n,Background(enter-ext-of-person)
exten = 1625,1,Playback(digits/1)
exten = 1625,n,Goto(digits/1)
exten = i,1,NoOp(CallerID is ${CALLERID})
exten = i,n,NoOp(DID is ${DNID})
And the console stuff for a 06xx DID:
-- Starting simple switch on 'Zap/1-1'
== Unknown extension '0' in context 'incoming' requested
-- Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/1-1'
For completeness' sake, here's the stuff from /var/log/asterisk/full; note
that my 16xx extensions get all four DTMF digits before continuing:
Jan 7 22:45:30 VERBOSE[5819] logger.c: -- Starting simple switch on
'Zap/1-1'
Jan 7 22:45:30 DEBUG[5819] chan_zap.c: DTMF digit: 0 on Zap/1-1
Jan 7 22:45:30 DEBUG[5819] chan_zap.c: No echo cancellation requested Jan
7 22:45:30 VERBOSE[5819] logger.c: == Unknown extension '0' in context
'incoming' requested
Jan 7 22:45:33 DEBUG[5819] channel.c: Scheduling timer at 160 sample
intervals
Jan 7 22:45:33 VERBOSE[5819] logger.c: -- Playing 'ss-noservice'
(language 'en')
[...]
So, I'm kind of stumped. As far as I can see, 0 is being treated
special, and is trying to route immediate to the operator or something.
[I've deleted the extensions.ael sample file, so that's not playing a
role, either.]
Thanks for any suggestions/ideas/etc...
-Ken
Ken D'Ambrosio wrote:
I've got a T1 (EM wink). Our four-digit inbound DNIS numbers are in the
range of 0600 - 1699. However, the second that the 0 is seen on an
in-bound 06xx call, it stops listening for any more digits, and
immediately tries to route the call. My 16xx numbers wait for all four
digits before trying to route. Is there something, somewhere, that
tells
it to do an immediate route on seeing 0? I don't have much of anything
in my extensions.conf file. I'm seeing what's going on via
tail -f /var/log/asterisk/full
Any suggestions?
Thanks!
-Ken
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Post your extensions.conf and what's on the CLI (asterisk -r)
--
Roman Volf
Keystreams Internet Solutions
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Immediate routing on 0 (DNIS)?
On 1/7/06, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
Post your extensions.conf and what's on the CLI (asterisk -r)
As requested:
# cat /etc/asterisk/extensions.conf
[incoming]
exten = s,1,Answer()
exten = s,n,NoOp(CallerID is ${CALLERID})
exten = s,n,NoOp(DID is ${DNID})
exten = s,n,Background(enter-ext-of-person)
exten = 1625,1,Playback(digits/1)
exten = 1625,n,Goto(digits/1)
exten = i,1,NoOp(CallerID is ${CALLERID})
exten = i,n,NoOp(DID is ${DNID})
And the console stuff for a 06xx DID:
-- Starting simple switch on 'Zap/1-1'
== Unknown extension '0' in context 'incoming' requested
-- Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/1-1'
For completeness' sake, here's the stuff from /var/log/asterisk/full; note
that my 16xx extensions get all four DTMF digits before continuing:
Jan 7 22:45:30 VERBOSE[5819] logger.c: -- Starting simple switch on
'Zap/1-1'
Jan 7 22:45:30 DEBUG[5819] chan_zap.c: DTMF digit: 0 on Zap/1-1
Jan 7 22:45:30 DEBUG[5819] chan_zap.c: No echo cancellation requested Jan
7 22:45:30 VERBOSE[5819] logger.c: == Unknown extension '0' in context
'incoming' requested
Jan 7 22:45:33 DEBUG[5819] channel.c: Scheduling timer at 160 sample
intervals
Jan 7 22:45:33 VERBOSE[5819] logger.c: -- Playing 'ss-noservice'
(language 'en')
[...]
So, I'm kind of stumped. As far as I can see, 0 is being treated
special, and is trying to route immediate to the operator or something.
[I've deleted the extensions.ael sample file, so that's not playing a
role, either.]
I am stumped as well, you don't have any extension defined for either
0, _0X, or _0X.
So I got no clue why *you* are stumped, in fact 1625 is treated
special, because it got an extension.
Thanks for any suggestions/ideas/etc...
-Ken
Ken D'Ambrosio wrote:
I've got a T1 (EM wink). Our four-digit inbound DNIS numbers are in the
range of 0600 - 1699. However, the second that the 0 is seen on an
in-bound 06xx call, it stops listening for any more digits, and
immediately tries to route the call. My 16xx numbers wait for all four
digits before trying to route. Is there something, somewhere, that
tells
it to do an immediate route on seeing 0? I don't have much of anything
in my extensions.conf file. I'm seeing what's going on via
tail -f /var/log/asterisk/full
Any suggestions?
Thanks!
-Ken
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Post your extensions.conf and what's on the CLI (asterisk -r)
--
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Keystreams Internet Solutions
[EMAIL PROTECTED]
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Re: [Asterisk-Users] Some advice on routing DID's
Do I need to install the complete ASTPP package or just utilize your AGI script with the context for AMP? Thanks On 1/7/06, Tom Vile [EMAIL PROTECTED] wrote: It's funny you mentioned that Darren, I was looking at your scripts today. I will evaluate it some more. On 1/7/06, Darren Wiebe [EMAIL PROTECTED] wrote: I have written an agi script that I use for that. Then I can just have a list of dids and extensions in a db. Tom Vile wrote: Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote server. Thanks, -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Immediate routing on 0 (DNIS)?
I am stumped as well, you don't have any extension defined for either 0, _0X, or _0X. So I got no clue why *you* are stumped, in fact 1625 is treated special, because it got an extension. Okay; thanks! I mis-understood the mechanism. I didn't think extensions.conf actually came into play until it had all the DID stuff. I guess DTMF is DTMF, regardless of the source, huh? In which case, I'm all set -- again, thanks for setting me on the right road! -Ken Thanks for any suggestions/ideas/etc... -Ken Ken D'Ambrosio wrote: I've got a T1 (EM wink). Our four-digit inbound DNIS numbers are in the range of 0600 - 1699. However, the second that the 0 is seen on an in-bound 06xx call, it stops listening for any more digits, and immediately tries to route the call. My 16xx numbers wait for all four digits before trying to route. Is there something, somewhere, that tells it to do an immediate route on seeing 0? I don't have much of anything in my extensions.conf file. I'm seeing what's going on via tail -f /var/log/asterisk/full Any suggestions? Thanks! -Ken ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Post your extensions.conf and what's on the CLI (asterisk -r) -- Roman Volf Keystreams Internet Solutions [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Some advice on routing DID's
Just grab the script. I can help you with it off the mailing list if you like Darren Tom Vile wrote: Do I need to install the complete ASTPP package or just utilize your AGI script with the context for AMP? Thanks On 1/7/06, Tom Vile [EMAIL PROTECTED] wrote: It's funny you mentioned that Darren, I was looking at your scripts today. I will evaluate it some more. On 1/7/06, Darren Wiebe [EMAIL PROTECTED] wrote: I have written an agi script that I use for that. Then I can just have a list of dids and extensions in a db. Tom Vile wrote: Would like some advice on the best way to route DID's to remote asterisk servers. Currently I have multiple DID's on my main Asterisk server in a datacenter and have remote servers that connect via an IAX trunk and when a call comes into my server I pass it to the iax peer. Just wondering what the best way it is to do this without having to have multiple line contexts for each remote server. Thanks, -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Agi Perl Talk Time
Hi all, Can anyone tell me from where i can call my update query when the call is pickuped using AGI Perl. I writen the code to store active calls in MySQL DB. and display the real time call counter. i mean to insert record when the call is pickuped -- Thank You, Code Lover ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Jobs
Thanks all for the replies. I started working for a CLEC a few months ago and we've chosen to implement Asterisk. I'm not sure if the fact that my boss is an open source advocate is a good thing or not... ie yes it's great to work with Asterisk and see all the features coming together (especially with Polycom phones). On the other hand I wonder how useful this experience will really be. I see a lot of VOIP jobs requiring Cisco experience. I worked with VOIP back in 1998, for a global VOIP wholesaler called OzEmail Interline in Australia before there where any standards... before SIP even. Until a few months ago I was working with SAN's and storage. Anywho... Doug. -Original Message- From: Robert La Ferla [mailto:[EMAIL PROTECTED] Sent: Sat 1/7/2006 8:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [Asterisk-Users] Asterisk Jobs Douglas Garstang wrote: I'm curious why the number of jobs out there requiring Asterisk seems to be pretty low. After looking around dice, monster, careerbuilder etc, I was surprised to find no more than 3-4 employment opportunities with Asterisk throughout the US. Is it really that low? There seems to be a job of opportunities for Cisco and other vendors solutions (duh... GUI's are good... duh). I wonder if demand will increase, or am I just looking in the wrong places? I think that the Asterisk customer profile can shed some light on this. If you are a big company, you'll buy into an expensive system because you can afford it and rely on it. If you are a small company, you will look to Asterisk as an inexpensive way to set up your telephone system. You will also likely have staff that is willing to work with it and not enough money or need to hire external consultants exclusively for Asterisk. You may have a telecom or networking consultant that will put together the network and set up the system but Asterisk is a small piece of it. I'd say Asterisk is more of a plus in a job description but not a requirement. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Line Sharing or Better Call Pickup
I have been trying to figure out for quite a while now how to better setup asterisk in a small office environment.. For example, small offices usually want to be able to have shared lines, so one can put a line on hold and another person can pick that call up if its on hold. The astra and snom phones have the ability to use parking spots, but asterisk doesnt seem to support them, as in you cant put a call in a slot and have a button and light for that slot light up on all the phones in a group I found a patch for snoms so that a call could be picked up but its extreme beta and doesnt seem stable enough for production use. People who are use to working in a small 4 line 5 10 phone office dont want to go through trouble of parking calls and having to tell someone else the parking spot number, and transfer is no good cause sometimes they have to put the call to a person but that person is on the phone or busy so they have to wait for a while which makes transfer not the best option either. I am hoping there is a solution for this I am not seeing, maybe someone has some experience with small office setups because this would seem to be major for any small setup. Any suggestions or experience would be greatly appreciated Thanks This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Jobs
On 1/8/06, Douglas Garstang [EMAIL PROTECTED] wrote: I'm not sure if the fact that my boss is an open source advocate is a good thing or not... ie yes it's great to work with Asterisk and see all the features coming together (especially with Polycom phones). On the other hand I wonder how useful this experience will really be. I see a lot of VOIP jobs requiring Cisco experience. The answer to your question probably depends on what type of IT Guy you are (nope, we're not all the same), and the nature of your career goals. If you are the kind of guy who will learn, from working with Asterisk, a good overall view of the technologies behind and issues with semi-traditional and IP telephony, in addition to developing a sense of how things work and how to get things done with telephony systems in general, then I think it will be valuable experience. If you either aren't particularly interested in gaining a high-level overview of telephony systems, and/or if you are just not the kind of guy who will get that kind of a big picture by working with only one specific platform or system, then it may not be so valuable to you. Even if the latter is true, I don't think that would make you any less-qualified or less-skilled as an IT worker; I have simply noticed that people in the field see things and learn about systems in totally different ways. Some learn more by focusing on the specifics of various systems or platforms one by one, and some learn by constructing and updating a conceptual framework that contains within it the specifics of whatever system (Asterisk, in this case, but may include Cisco, etc in your future) is being worked on. If you learn things the former way, by focusing on a specific system at a time, your Asterisk experience will probably not be as valuable to you in terms of future jobs as would equivalent Cisco experience. But, if you tend to learn things in the latter way, your Asterisk experience is probably pretty much just as useful as if it were Cisco experience, or experience with any other vendor. Also, if you want to work for a large, formal company that places a lot of importance on titles and buzzwords, your lack of specific, major-vendor experience may present a problem. On the other hand, if you want to work for an outfit that is perhaps less formal and more unconventional, your Asterisk experience could stand you in very good stead, since Asterisk is such a flexible system, and since experience with it may indicate that you are a flexible programmer/net/sysadmin :). Anyway, those are my two cents on it. -Rusty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Processor Update?
I've been Googling around for some time now (a few hours on dial-up). I find all kinds of questions similar to mine, but either there is no answer or the answer has nothing to do with the question. Hopefully this post isn't another one of those. Does Asterisk favor FPU performance or clock speed? (Meaning AMD or Intel) I see Asterisk can be compiled for x64 systems. Does it run any better\worse on x64 versus i386? Dual-core CPU performance isn't as good as dual CPU's, but is more often than not a better deal as it's close, but a lot cheaper. Dual-core vs. Dual CPU performance depends on the application. How does Asterisk respond? Looking to build out a system capable of 100 concurrent calls (IP pass-through, so in effect 200). Looking at the dimensioning, I can get some ideas, but few of them state the call quality. I'm hoping to be able to use a single CPU that is dual core as the price to go to a dual CPU that is single core puts me at least 60% to a redundant server. More information about the setup -- No cards of any kind (Tyan board with integrated video, NIC, SATA nothing more needed) some IAX2 trunks (ztdummy) mainly SIP clients, but an increasing number of IAX2 mainly ulaw\alaw, if not solely (the transcoding chews CPU cycles needlessly) Mike HammettIntelligent Computing Solutionshttp://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to check Digium TE410P firmware version?
On 12/27/05 22:39 Kevin P. Fleming said the following: Steve 4 wrote: Field-upgradeable? Does that mean that I can do it myself? That would be great since some systems are in production and sending the board to Digium takes time. The 2nd gen firmware has field-upgradeability. The 1st gen firmware does not, unfortunately. There is not currently any 3rd gen firmware, but when there is, you'll be able to do it yourself :-) we've got a number of TE410P 1st gen firmware cards. could we send them in to digium for a firmware upgrade ? the cards were purchased in october 2004 from atp in melbourne. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
