RE: [Asterisk-Users] TE411P Really Bad Echo
I'm using the Varionboards with no problem. Now, about echo... Sagnoma says if YOU have echo, it is THEIR problem and they will fix it. James TaylorMetroTel3505 Summerhill RoadSuite 11Texarkana, Tx 75503903-793-1956 -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Stagg SheltonSent: Sunday, February 12, 2006 5:30 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] TE411P Really Bad EchoI am using asterisk 1.2.4 and zaptel 1.2.3. Also, I tried the latest zaptel out of subversion.Stagg Sheltonwww.oneringnetworks.comIsaac Xiao (KVB Kunlun Pty Limited) wrote: What version of Asterisk and Zaptel you were using? Did you try latest Asterisk 1.2.4 and Zaptel 1.2.3? Anyone has good feedback for TE411P? Isaac Xiao Stagg Shelton wrote: It was Digium's opinion that perhaps the card had a VPM. We got a replacement TE411P, I implemented it tonight and still the exact same echo problem. At this point I feel like I can rule out failed hardware. I contacted Digium support and now they are telling me it's something with my carrier, and I should call them. I called Bellsouth, and they ran a full stress test on the circuit taking me offline for about 30 minutes. The end result is that the circuit test passed with no errors. Bellsouth says it's not in their network, Digium says its not their card, and I have a te411p with VPM disabled in the wct4xx kernel module because something doesn't work the way it should. My customer is wanting to know about sangoma cards with the echo cancellation, and at this point I'm nervous to recommend any hardware. I'm going to look into the sangoma that you suggested. Are there any other kinds of products that I could look into either Passive or Active. Thanks Stagg Shelton www.oneringnetworks.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDD
I've read the WIKI. Any additional information on the TDD mode? Anyone done a TDD to email app? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 ___ --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] kill a .call file
Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Differ between private and out of area?
out-of-area is displayed for calls that originate from LECs that have not implemented caller id. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Monday, September 19, 2005 12:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Differ between private and out of area? I don't believe you can trust the keywords that may or maynot be in the calleridname. The telco folks will frequently honor anything that a company wants inserted as a name (assuming a reasonable request). So, even if you get the correct logic in place for asterisk code, the end result is most likely not going to give you what you want. I know a telco tech that will change the libd database to say God Calling, place a call to a buddy, then change it back to the original string after the call. Also, some itsp's allow you to change that string to anything reasonable. Yes, I know that, but, how to distinguish between them at incoming call? - Original Message - A private call is a call that someone has specifically blocked. An out of area or unknown call is simply a call that the caller-id did not come through on correctly, for some reason. On 9/18/05, Goran Dj. [EMAIL PROTECTED] wrote: Is there any method to make difference between Hidden (Private) and unknown (Out of area) incoming calls on ZAP/x101p? I want to block any hidden call, and to allow unknow calls, but ZAP channel (X101P) always delivering empty CALLERID= in both cases. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] kill a .call file
From my CLI: Attempting call on SIP/gw/19857749166 for [EMAIL PROTECTED]:1 (Retry 114) Attempting call on SIP/gw/19037747603 for [EMAIL PROTECTED]:1 (Retry 83) Attempting call on SIP/gw/19857747603 for [EMAIL PROTECTED]:1 (Retry 80) I want to stop it from any future attempts. Any idea about a command to kill or where the data is stored? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of trixter http://www.0xdecafbad.com Sent: Monday, September 19, 2005 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] kill a .call file On Mon, 2005-09-19 at 13:43 -0500, jltaylor wrote: Any means of killing a .call file that is in progress? You mean once the call has begun? You prolly want to hangup the call ... asterisk -rx soft hangup callid Or is there something else that you wanted? -- 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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hard deskphone via wifi?
I'v used the CB3'S, they are 200mw and work great. At the remote I have Asterisk, 20 extensions, vm, queuing, and always 10 to 20 calls up. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Graves Sent: Wednesday, August 10, 2005 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Hard deskphone via wifi? Has anyone here ever tried using a wifi bridge to place a deskset in someplace where there was no LAN drop? If so what hardware did you use and was it succesful? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 fwd 54245 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: DS3 - VoIP Hardware Recommendations
TNT's have DS3 cards and the DS3 config is cheaper than multiple T1 config. The Lucent MAX TNT is a true carrier class machine. If you need help with TNT's let me know. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andrew Kohlsmith Sent: Wednesday, July 13, 2005 2:20 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] OT: DS3 - VoIP Hardware Recommendations On Wednesday 13 July 2005 13:31, Brian C. Fertig wrote: Trust me dude.. You don't want a lucent TNT. If your going all out for an DS3 and you don't want to multiplex it then you will need something that will take a DS3 which I don't believe TNT's do. Purchase an AS5400HPX they will and work very well. Set yourself up with some dialpeers etc and your good to go. Trust me. I have done it. Speaking as someone who's run a fairly large ISP off of the MaxTNTs... why not? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] mini itx
I've seen the embedded posts. Is anyone running Asterisk on the MINI ITX? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] mini itx
It may not be enough horsepower... I'm looking for a black box, with a PCI slot to put in a telco closet. Needs to be able to take the 4 port T1 card (pci slot) and do g729 for 50-60 calls. Any suggestions? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Iain Young Sent: Thursday, June 23, 2005 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] mini itx On Thu, Jun 23, 2005 at 11:39:21AM -0500, jltaylor wrote: I've seen the embedded posts. Is anyone running Asterisk on the MINI ITX? Yes, no problems, I have an X100P in the PCI slot, but its only a single POTS line. I used the MII board, but only because thats what I had avaliable. Iain ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ASTCC not making calls
Doesn't the ASTCC require 12 digit pins?
James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx 75503
903-793-1956
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Juan Luis
Moyano
Sent: Thursday, June 23, 2005 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] ASTCC not making calls
Hi, im trying to setup ASTCC but I'm getting it difficult. I've
correctly set up the mysql database astcc and added a brand, trunk,
route and a card as follows:
brands
+--+--+--+--+--++--+--+
| name | language | inc | publishednum | did | markup | days | fee |
+--+--+--+--+--++--+--+
| FWD | es | 6| 4| 4| 0 | 30 |0 |
+--+--+--+--+--++--+--+
trunks
+--+--+-+
| name | tech | path|
+--+--+-+
| FWD | IAX2 | 657XXX:[EMAIL PROTECTED] |
+--+--+-+
routes
+-+---++-+-+--+
| pattern | comment | trunks | connectcost | includedseconds | cost |
+-+---++-+-+--+
| ^4. | FWD | FWD| 0 | 0 | 150 |
+-+---++-+-+--+
-Added a card with $25 credit, using 'FWD' brand.
extensions.conf
---
[outbound-fwd]
;
exten = _4.,1,DeadAGI(astcc.agi,${CALLERIDNUM},${EXTEN:1})
exten = _4.,2,Hangup()
iax.conf
register = 657XXX:[EMAIL PROTECTED]
The problem is that when, for example, I dial '4612' i get:
-- Executing DeadAGI(IAX2/[EMAIL PROTECTED]/3, astcc.agi|21|612) in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi
-- Playing 'digits/1' (language 'en')
-- AGI Script astcc.agi completed, returning 0
-- Executing Hangup(IAX2/[EMAIL PROTECTED]/3, ) in new stack
and i hear allison saying I'm sorry that is not a recognized phone
number, goodbye.
Anyone knows what could be happening right here?
Many thanks in advance.
--
Juan Luis Moyano
[EMAIL PROTECTED]
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RE: [Asterisk-Users] Using 2 x DSL
You can't really do true bonding unless you control both ends of the link. I had a customer who tried this. It's easy to do with ATM and IMA interfaces on T1/T3 type stuff. The $300-$1000 dual wan routers will not work off the shelf. Policy based routing helped but it's tough to make it work. Now, what you can do is put the Asterisk on ONE network and use policy based routing to share other stuff like surfing, smtp, telnet, etc. You can prioritize the traffic so that the packets to and from the Asterisk are mangled to have the higher priority. If both DSL's are for Asterisk ONLY then you might try round-robin DNS or manually setup traffic. Asteris will work on multiple LAN's - I have both a PUBLIC and PRIVATE ip in the same box on different NIC's. Just set your routing. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of VoIP-PBX Sent: Thursday, June 23, 2005 1:46 PM To: Jorge Carrasquillo; Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Using 2 x DSL Hi all, my client wants to double his bandwidth by using 2 x DSL lines into one Asterisk network How can I do this ? Thanks Henry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] INBAND DTMF G729 ASTERISK
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of [EMAIL PROTECTED] Sent: Thursday, June 23, 2005 2:19 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] INBAND DTMF G729 ASTERISK Hi all. Why don't Asterisk support inband DTMF with G729? Is there a way to do that!? Are you using RFC2833? Doesn't it a security hole? Thanks. Denis. = It's a long story.. It's all about bandwidth, Maximum accepted frequency offset, Minimum rejected frequency offset, Timing, Twists, and Signal to Noise Ratio. It is difficult to meet the ITU standards once it gets mangled through the g.729 codec. James Taylor ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DNIS and DID seeking confirmation
DID number is the number commonly assigned to a PSTN trunk. DNIS and DID may be the same. DNIS refers to the Dialed Number that is passed as signaling with the call (or on ss7). Most calls have ANI and DNIS. Your extensions look ok, assuming that the carrier sends the digits that match. What Asterisk looks for is determined by how you have signaling setup in your config for the card(s) that you have installed. So, this must match the signaling on the carrier side. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Millican Sent: Monday, June 13, 2005 11:02 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] DNIS and DID seeking confirmation Hello all, After much googling I have come to the conclusion that in asterisk land DID(Direct Inward Dial) and DNIS(Dialed Number Identification Service) are used rather interchangeably. If this is an incorrect assumption Please correct me. Based on this assumption if I have everthing set up to land in the [incoming] context and an 800# such as 1-800-123-4567 with 4 digit DNIS I can have an entry in my incoming context exten = _4567, 1, do something this is where the call to my 800 number will land regardless of which trunk the call comes in on. Like wise if I have a DID number 456-7891 with an exten= _7891,1,do something else this will also work. Is this correct or am I way off base? Also what is Asterisk looking for as far as a delimiter or is that in a config file? Something like Seize (Wink) DNIS (Wink) ANI (Wink) Answer or Seize (*) DNIS (*) ANI (*) Answer John M ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Phantom (ghost) Calls with Wildcard TDM400P
http://tzone.the-croc.com/sounds/twiltzon.mid James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Lange Sent: Friday, June 10, 2005 1:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Phantom (ghost) Calls with Wildcard TDM400P No, there is nothing else connected to the lines. Also got a single report of a call coming in, but when answered hearing a ringing sound on the line (as if you were placing an outbound call). Incoming caller doesn't hear the ringing but hears the person say hello. This is very strange. -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location On Thu, 2005-06-09 at 19:10 -0700, Steve Totaro wrote: - Original Message - From: John Lange [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 09, 2005 1:26 PM Subject: [Asterisk-Users] Phantom (ghost) Calls with Wildcard TDM400P We have a client that has a single Wildcard TDM400P with 3 FXO ports on Asterisk 1.0.7. Occasionally the system seems to loose its mind and starts originating calls from that Zap channels that don't exist. The receptionist picks up the phone and nobody is there. This can happen repeatedly over and over again within a few minutes. As far as we can tell these are definitely not real calls as nobody has ever called back and said they couldn't get through. Does anyone have a suggestion for why this might be happening? -- John Lange President OpenIT ltd. www.Open-IT.ca (204) 885 0872 VoIP, Web services, Linux Consulting, Server Co-Location Is there an alarm system, fax or any other sharing the line? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] monitoring
Has anyone done any scripts (or something else) to notify if something goes down? Example: Asterisk_1 is peered to Asterisk_2 Asterisk_1 has qualify=yes Asterisk_1 notices that Asterisk_2 is not responding Asterisk_1 sends email to cell phone Asterisk_2 peer down James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream GSX-2000 - dead :-(
I've got three GS 100 Phones with same problem. Some lights. Some no lights. Some garbled display. I would welcome suggestions for a resurrection. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mark Elkins Sent: Friday, May 27, 2005 10:01 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Grandstream GSX-2000 - dead :-( I have a Grandstream GSX-2000 with .. Software Version:Program-- 1.0.0.3Bootloader-- 1.0.0.3 I tried to do an HTTP update from the Grand Stream web site... After half an hour, I recycled power and now its dead... LED's come on and stay on, screen and buttons are dead. Connectivity to Grandstream.com was always good - whenever I checked (I downloaded the User Manual in a couple of minutes), the site states five minutes to load, so waiting more than 30 mins should have been OK, and they do have this Please Powercycle in red print too... Is there a magic re-incarnation routine ? (Power on whilst holding down some buttons?, Sprinkling chickens blood?) I chose an HTTP upgrade over TFTP - as I read that there were potential issues with TFTP at this firmware level. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] multiples broadvoice lines
[EMAIL PROTECTED]:passwd:[EMAIL PROTECTED]/ X1 [EMAIL PROTECTED]:passwd:[EMAIL PROTECTED]/ X2 This is what I did. I used the BV number as an extension and handled it in a context. There may be better way. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Shaw Sent: Thursday, May 26, 2005 10:26 AM To: Asterisk Users Mailing List Subject: [Asterisk-Users] multiples broadvoice lines Hello All, I have 4 Broadvoice lines. If I call any of the lines it shows that is coming from the first line. exaple [EMAIL PROTECTED]:passwd:[EMAIL PROTECTED] [EMAIL PROTECTED]:passwd:[EMAIL PROTECTED] [EMAIL PROTECTED]:passwd:[EMAIL PROTECTED] [EMAIL PROTECTED]:passwd:[EMAIL PROTECTED] If I call X3 it shows that someone called X1. ANY HELP Please. I'm using [EMAIL PROTECTED] Ver 1 Thanks, David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Astcc
Look at the source code. It provides for an option to be passed from the dial plan to make it silent or give less information. James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Shidan Sent: Friday, May 20, 2005 2:56 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Astcc Is there an easy way to make Astcc silent, so that it does not tell the user how much money he has and the cost to a location, but rather does call control silently. Whats the general consensus of astcc vs areski. Thanks for your advice in advance. Shidan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDMoE emulates a T-1= Is there a product tosimulate a PRI trunk? (Robert Goodyear)
Does the TDMoE only allow one T1 per segment? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 Free VOIP Telecom ads: http://ads.metrotel.net www.metrotel.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of M O Sent: Friday, May 13, 2005 2:15 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TDMoE emulates a T-1= Is there a product tosimulate a PRI trunk? (Robert Goodyear) Robert, Is there a product to simulate a PRI trunk? (Robert Goodyear) TDMoE emulates a T1. ;) Once the TDMoE link is up, Asterisk just sees 24-lines that appear to be a T1 instead of having to deal with all of the complexities of VoIP. This is useful, since probably 75% of the utility of VoIP is really just the fact that it can run over a network. It's also handy because it unifies the flexibility and cost-savings of a Ethernet with the telephony-friendly aspects of a T1 (alarm codes, bundling trunks, channelization) TDMoE Mini-HOWTO http://voip-info.org/tiki-index.php?page=Asterisk%20TDMoE Sincerely, SoftwareRadioGuy __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calling card
Yes, This is the solution that I am using and it works every time. You can dial a number, put in a pin and it makes calls. I've never received a bill, the minutes are free. I can't understand how these people make any money. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of C F Sent: Monday, May 09, 2005 11:27 AM To: Alberto; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Calling card really?? On 5/9/05, Alberto [EMAIL PROTECTED] wrote: ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Collect calls
Since you are referring to R2 signaling, it works like this: The E1 R2 Call Blocking feature provides two ways to block incoming collect calls-category-based and double answer. With category-based call blocking, collect calls will be blocked based on a specific category. For example, in Brazil, collect calls arrive with a category II-8, for which the gateway should send B-7 as a response instead of an answer signal. This approach is only applicable when switches in the central office support category-based blocking. For legacy switches that do not support category-based blocking, the double answer method is implemented to support the collect-call blocking. For an incoming collect call, the gateway will answer the call with a clearback after one second and re-answer the call after two seconds, causing the collect call to be dropped and normal calls to stay connected. This is what the referenced patches are attempting to do. This does not work in the U.S. or if you have SS7, you don't need it. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael D Schelin Sent: Tuesday, May 03, 2005 6:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Collect calls You Bring up a great point. I understand these codes and my system brings them in via ss7 but as youself I don't know how to protect my network from these charges. I will follow this post to see if anybody has a fix. Rodrigo P. Telles wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Folks, Does someone knows how to identify and block collect calls on Asterisk using PRI channels? I googled it and found this: http://lists.digium.com/pipermail/asterisk-dev/2004-November/007500.html I don't know what does it mean!!! Can someone help me to understand this? I tried to apply that way too, using Flash() but Flash() complains and looks like just work with FXO channels: http://lists.digium.com/pipermail/asterisk-users/2004-October/066360.html Thanks in advance. - -- Rodrigo P. Telles [EMAIL PROTECTED] IVOZ # 1009 TI Manager Devel-IT - http://www.devel.it Bestcom Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCd+zUiLK8unYgEMQRAkChAJ4xDYOvl8yZY+Uqn6v5VFZ4tMzicQCfT8+T 5foewh0m/o3ABMqcNHhtQs4= =rsu2 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Collect calls
In the U.S., its called: Inbound Call Operator Screening (ICOS) automatically screens and blocks incoming third-number-billed or collect calls, or both, so that callers cannot charge these calls to your line. It's a databse thing. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rodrigo P. Telles Sent: Tuesday, May 03, 2005 4:28 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Collect calls -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi Folks, Does someone knows how to identify and block collect calls on Asterisk using PRI channels? I googled it and found this: http://lists.digium.com/pipermail/asterisk-dev/2004-November/007500.html I don't know what does it mean!!! Can someone help me to understand this? I tried to apply that way too, using Flash() but Flash() complains and looks like just work with FXO channels: http://lists.digium.com/pipermail/asterisk-users/2004-October/066360.html Thanks in advance. - -- Rodrigo P. Telles [EMAIL PROTECTED] IVOZ # 1009 TI Manager Devel-IT - http://www.devel.it Bestcom Group -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFCd+zUiLK8unYgEMQRAkChAJ4xDYOvl8yZY+Uqn6v5VFZ4tMzicQCfT8+T 5foewh0m/o3ABMqcNHhtQs4= =rsu2 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR Billing Question.
Are you talking about tracking a single call through three servers, or are your wanting to track all calls made to (through) individual servers and bill a single customer based on ANI? James -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Paul DracevichSent: Wednesday, April 27, 2005 4:41 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] CDR Billing Question. I have three servers and I want to be able to bill a call going from one through all of the serves. The problem is that I am unable to link or pull the data from each server cdr record and have a common bill. I have been looking on google, but anyhelp would be great Regards Paul Dracevich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CDR Billing Question.
Really looks like having a central SQL server is the best way. Run it on a separate, dedicated machine with an IP address all of the others can see. James -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Paul DracevichSent: Wednesday, April 27, 2005 5:11 PMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] CDR Billing Question. Wanting to track all calls made to (through) individual servers and bill a single customer based on ANI -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul DracevichSent: Thursday, April 28, 2005 10:08 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [Asterisk-Users] CDR Billing Question. Yes thats it. -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jltaylorSent: Thursday, April 28, 2005 10:28 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: RE: [Asterisk-Users] CDR Billing Question. Are you talking about tracking a single call through three servers, or are your wanting to track all calls made to (through) individual servers and bill a single customer based on ANI? James -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Paul DracevichSent: Wednesday, April 27, 2005 4:41 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] CDR Billing Question. I have three servers and I want to be able to bill a call going from one through all of the serves. The problem is that I am unable to link or pull the data from each server cdr record and have a common bill. I have been looking on google, but anyhelp would be great Regards Paul Dracevich ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g729 passthrough?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Capouch Sent: Sunday, April 24, 2005 3:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] g729 passthrough? I'm sitting here with my dunce cap on. My weak excuse is that I haven't ever played with g729 before. I have a Sipura 841. I have the phone config set to use g729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) -- Format for call is g729 Apr 24 15:38:38 NOTICE[5586]: channel.c:1833 set_format: Unable to find a path from g729 to slin . . . . I get ringback from Nufone, but as soon as the call answers I get an error: Apr 24 15:43:42 NOTICE[5596]: channel.c:1833 set_format: Unable to find a path from g729 to slin . . . What am I doing wrong to cause it to want to transcode? I assume that's where the complaint is coming from. I thought Asterisk could pass through without transcoding as long as the endpoints are all g729. Thanks. B. ;;; Brian, Add to the [general] section in sip.conf the following: disallow=all allow=g729 allow=ulaw allow=alaw For some reason Asterisk will not pass audio through itself without trying to transcode unless you have this in your config. Don't ask me why it will not work with allow=g729 under the individual peer. This has to go in the [general] section. James Taylor MetroTel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] T1 EM false busy after dial
If Feature Group B signaling is working properly (and you have Feature Group
B trunks), then
to reach your Asterisk box you would dial from the PSTN (1)+950+WXXX {W is 1
or 0 based on the number assigned to you}.
If you are dialing out {terminating where you look like the carrier} on
FGB then it depends on if you are connected to an Equal Access End Office or
a Access Tandem.
Are you sure about the Feature Group B thing or do you have trunks that just
require MF signaling?
If you want MF, you might try the featdmf setting, however, the telco
needs to know that you want FGD.
AND...
If you are connecting to an Access Tandem instead of and End Office, then
the featdmf in Asterisk will not work.
I have submitted a request for a quote to Digium to modify the code to make
this work properly.
Likewise, true FGB terminating (where it looks like you are the carrier)
works through an Access Tandem and the additional code is missing for that
also.
Take out the featb and add:
em_w
This will let you see if just plain old DTMF works.
James Taylor
903-793-1956
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Ackley
Sent: Sunday, April 24, 2005 4:44 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] T1 EM false busy after dial
TE101P card T1 EM trunk to telco
on a SIP-PSTN call, after dial
SIP phone hears two seconds busy tone (1) then ring tone
how do we get rid of busy tone?
(1) two second busy
(480+620/500 0/500 480+620/500 0/500)
---
extensions.conf:
;
; dial-out to the PSTN with 7 digits
;
exten = _NXX,1,Dial(Zap/g1/${EXTEN})
exten = _NXX,n,Hangup()
zaptel.conf:
span=1,1,0,esf,b8zs
em=1-24
loadzone = us
defaultzone=us
zapata.conf:
[trunkgroups]
[channels]
language=en
context=default
signalling=featb
usecallerid=no
callwaiting=no
threewaycalling=no
transfer=no
cancallforward=no
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
busydetect=yes
busycount=8
channel = 1-24
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.9.18 - Release Date: 4/19/2005
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RE: [Asterisk-Users] T1 EM false busy after dial
Normally, plain old PBX DID trunks are em_w (dtmf).
Strange, the only other problem might be the timing of the wink.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Ackley
Sent: Sunday, April 24, 2005 8:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] T1 EM false busy after dial
thanks info and suggestion
we have a plain old PBX DID trunk from our telco
will try to get more information about the trunk
meanwhile I tried as documented in my zapata.conf:
; JNA tried all below - and even NO signaling same resuts
;Apr 24 21:11:15 WARNING[4430]: chan_zap.c:10198 setup_zap: Ignoring
:signalling
;-- Reconfigured channel 1, Feature Group B (MF) signalling
; etc.
;
;signalling=featb
:signalling=em_w
;signalling=sf_featb
;signalling=sf_featdmf
;signalling=sf
jltaylor wrote:
If Feature Group B signaling is working properly (and you have Feature
Group
B trunks), then
to reach your Asterisk box you would dial from the PSTN (1)+950+WXXX {W is
1
or 0 based on the number assigned to you}.
If you are dialing out {terminating where you look like the carrier} on
FGB then it depends on if you are connected to an Equal Access End Office
or
a Access Tandem.
Are you sure about the Feature Group B thing or do you have trunks that
just
require MF signaling?
If you want MF, you might try the featdmf setting, however, the telco
needs to know that you want FGD.
AND...
If you are connecting to an Access Tandem instead of and End Office, then
the featdmf in Asterisk will not work.
I have submitted a request for a quote to Digium to modify the code to make
this work properly.
Likewise, true FGB terminating (where it looks like you are the carrier)
works through an Access Tandem and the additional code is missing for that
also.
Take out the featb and add:
em_w
This will let you see if just plain old DTMF works.
James Taylor
903-793-1956
--
No virus found in this outgoing message.
Checked by AVG Anti-Virus.
Version: 7.0.308 / Virus Database: 266.9.18 - Release Date: 4/19/2005
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RE: [Asterisk-Users] using * for Internet call waiting
I'll take the scaled down version, just a client that plays voice mail and shows caller id. Any ideas? james -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Nicolás Gudiño Sent: Friday, April 22, 2005 11:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] using * for Internet call waiting On 4/21/05, Gary Carr [EMAIL PROTECTED] wrote: Wondering if it is possible or if something already exist to setup * to offer Internet Call Waiting. For those that do not know what it is, it's a small application that runs on a users computer that will pop up a window letting them know they have a incoming call and who it is from then they can choose to take the call which will disconnect their dialup modem and ring their phone or send the call to voice mail. You need a V92 capable modem for your client and a V92 capable access server for you. The feature is called modem on hold, it lets you pick up a call without loosing your internet connection, and resume the dialup session after hangup. The only feature you need for your telco is call waiting. It does not need forward on busy. Regards, -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Starting with Asterisk-SIP
Don't get many hugs around here...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ruben
cuevas rumin
Sent: Thursday, April 21, 2005 2:28 PM
To: Moises Silva
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Starting with Asterisk-SIP
Hi Mosies,
Thanks for your help, now I have a SIP server using asterisk and I can
communicate my two SIP clients with asterisk in the middle :).
This is the first step, but I have to work a lot of yet, so I think I
will disturbe you and the other people in the list (I'm sorry).
Thank you very much for your help.
Un saludo y un abrazo ;). (It's an spanish expresion)
Rubén.
On 4/20/05, Moises Silva [EMAIL PROTECTED] wrote:
Hi again Ruben. Well, it would be good idea to put here what do you
have in your extensions.conf. Actually i have only includes in this
file, several statements like this:
#include /var/lib/pavoz/extengeneral.iss
So, its easier its administration, but for a simple test you can do
this in extensions.conf:
[testdialplan]
exten = _.,1,Dial(SIP/${EXTEN},40,r)
exten = _.,2,Hangup();
then, in sip.conf:
[general]
port=5060
bindaddr=0.0.0.0
localnet=192.168.1.0/24 ; here you need your net config net_addr/mask
tos=lowdelay
tos=184
defaultexpirey=120
disallow=all
allow=ilbc
allow=alaw
allow=ulaw
defaultcontext=incoming_iss
[15]
type=friend
secret=adminpass
host=dynamic
nat=no
dtmfmode=info
canreinvite=yes
qualify=yes
context=testdialplan
[12]
type=friend
secret=adminpass
host=dynamic
nat=no
dtmfmode=info
canreinvite=yes
qualify=yes
context=testdialplan
So, you need 2 sip phones (can use kphone) with SIP username 12 and
15, using password 'adminpass'.
This is what will happend:
- When you start kphone's, or any other SIP phone, the phones will,
they will try to make a SIP register with the server that you specify,
so you have to configure the phones yo try a register in the Asterisk
Box IP. Asterisk will receive its request for registry and will check
that the username and secret exists in the file sip.conf, if exists,
will save the registry and then Asterisk and the phones will be
connected. Now, when you dial from any sip user, the number will be
sent to Asterisk, and asterisk will try to find a match in the dialed
pattern in the context that the SIP entry specifies (in this case the
parameter context=testdialplan), so , for example, if you dial 12
from sip user 15, the 12 will match in the pattern _., because the
dot match anything, you can be more specific an put in extensions.conf
_XX, instead of _., and Asterisk will only match when you dial a
number of 2 digits length, and that digits are 0-9 (the X means 0-9).
You can read more about this in:
http://voip-info.org/wiki-Asterisk+config+extensions.conf
Once the pattern is matched, Asterisk will attempt to execute the
commands that are there, in this case a Dial() command, that say Open
a Channel type SIP, and try to dial to the ${EXTENSION}, ${EXTENSION}
is a special var, you can read more about asterisk vars in:
http://voip-info.org/wiki-Asterisk+variables
So it will try to dial to a SIP user with the dialed extensión.
So that all, it should work for a small test.
I have studied in Universidad de Guadalajara, in Guadalajara, México.
Any other people from México here :-)
Good Look!
On 4/19/05, ruben cuevas rumin [EMAIL PROTECTED] wrote:
Hi Moises,
Thanks for the reply, and thanks Dana too.
I know that I can to communicate two SIPs phones without Asterisk in
the middle. But this isn't my final objective, This is the first step
in my project, it mean, I firstly want make works a simple testbed
(the one I described in the previous mail), and then step by step
configure more difficult testbed.
So if you, please, could help me to configure this simple test, I'm
will be happy :).
I think my problem is the dial plan in the extensions.conf.
Ah, I'm studing electronics and comunnication eng, in the University
Carlos III of Madrid. Congratulations for your graduation, I hope end
in September of this year.
Which University do you have study?
Best Regards and thank you for your help.
On 4/19/05, Moises Silva [EMAIL PROTECTED] wrote:
Hi Ruben. You can make a direct IP call. If the 2 sip phones can ping
each other (that is, both are reachable in the network), then in
kphone select the option File New Call, then type
sip:[EMAIL PROTECTED] , the 'number' is the number wich is configured
in kphone, sipdeviceip will be the IP of the machine that is running
the kphone application. Note that this kind of call does not have
nothing to do with Asterisk, the phones are using sip protocol without
asterisk in the middle. When kphone makes a register to asterisk, then
you dont need to specify sip:[EMAIL PROTECTED] you only dial a
number and the number is
RE: [Asterisk-Users] using * for Internet call waiting
Would like to see a small client for this. It could be SIP or IAX without all of the phone features. It would need to provide a URL to the .wav file so it could be played. Any ideas? James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Gary Carr Sent: Thursday, April 21, 2005 3:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] using * for Internet call waiting Wondering if it is possible or if something already exist to setup * to offer Internet Call Waiting. For those that do not know what it is, it's a small application that runs on a users computer that will pop up a window letting them know they have a incoming call and who it is from then they can choose to take the call which will disconnect their dialup modem and ring their phone or send the call to voice mail. Thanks, Gary ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] using * for Internet call waiting
I'm an ISP, what I would like is a client for the dialup customer to run.
They would use call fwd busy to my did on an asterisk box.
I'd signal and they could click on button (URL) to download .wav file in
asterisk voice mail.
James
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Mojo with
Horan Company, LLC
Sent: Thursday, April 21, 2005 5:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] using * for Internet call waiting
I once tried the pagoo service. Seems I had to ask the telco for Call
Forward Busy, and provide them with the toll free number pagoo gave me
for their service. When the forwarded call is received by their
systems, they would see _my_ callerid information, and thus know to
contact my computer for the notification purpose.
Also, not sure if this is on track with what you want, but I've used
jabber_client.pl tied into my dialplan to popup the callerid info of an
incoming call on my screen.. I could then choose to answer the call or
let it ring to voicemail. Seems the jabber client Neos has
well-designed popups.
links:
http://jabberd.jabberstudio.org/2/
for the jabber_alert.pl script, allows sending jabber msgs from cmd line.
http://www.neosmt.com/
for a jabber client that pops up incoming messages. Note, this is also
an H.323 client. Haven't tried it with * yet, but I have been meaning to.
Here's the specific Dialplan line I use:
[inpstn]
exten = s,2,TrySystem(echo Incoming call from :${CALLERID} |
jabber_alert.pl -e [EMAIL PROTECTED] -n [EMAIL PROTECTED] -w
senders_password)
Because it can sometimes take 2 or 3 seconds to send the jabber message
on my network, I use TrySystem instead of System, which blocks, waiting
for the return code from the command I passed. Because the return code
is prolly irrelevant, you'd most likely want to use TrySystem too...
hope this helps :)
Moj
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary Carr
Sent: Thursday, April 21, 2005 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] using * for Internet call waiting
Wondering if it is possible or if something already exist to setup * to
offer Internet Call Waiting. For those that do not know what it is, it's
a small application that runs on a users computer that will pop up a
window letting them know they have a incoming call and who it is from
then they can choose to take the call which will disconnect their dialup
modem and ring their phone or send the call to voice mail.
Thanks,
Gary
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RE: [Asterisk-Users] MF instead of DTMF
MF works with FGD FGC signaling. Are you taking FGD with a tandem connection? James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael B. Murdock Sent: Tuesday, April 19, 2005 8:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] MF instead of DTMF I am looking into using Asterisk for an application where the upsteam switch will provide MF digits instead of DTMF after establishing a call. This is not during the call set up but after the call is established additional MF digits will be passed to indicated features to provide to the caller. Trunking will be EM T1. Does asterisk support MF detection in addition to DTMF? Has anyone done anything like this before? -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for ATAs
The SIPURA 3000 allows some dialplan programming. You might check them out. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anton Krall Sent: Monday, April 18, 2005 5:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Looking for ATAs Im looking for ATAs that have 1 FXO and 2 FXS ports, they will connect to a central Asterisk server and they idea is to share the FXO between the ATAs for people in location 1 can call the persons extension in location 2 or use locations 2's POTS lines to dial as a local call. Any recommendations? Also, how do you go about a dialplan for this? Does asterisk have to manage routing for this? Configure the ATAs so that if call not local or between the 2 local FXS, then route thru asterisk for termination on some other ATA (to their FXS or FXO)? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VPN/Asterisk combo
James Taylor I use MikroTik for a multi-LAN-multi-WAN router. It has a GUI interface and is easy to setup routes, rules, and queues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Mason (Lists) Sent: Tuesday, April 19, 2005 9:33 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] VPN/Asterisk combo Wow, that's quite a setup. What do you use for routing and firewalling? Chris Mason www.anguillaguide.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI - T1 feasibility
Works, looks simple. James Taylor -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ronald Hartmann Sent: Tuesday, April 19, 2005 12:14 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] PRI - T1 feasibility Looking for advice on the following feasibility --PRI (Goes to span 1)--Asterisk (4 span PRI Card)-- Sip Phones (Receive ANI and DNIS) | | | T1 Robbed Bit Span 2 | | | IVR Application I need to be able to take a call and look at the number dialed (DID) and if it matches a list, then Send the call directly to the T1 Span 2 to be handled by the IVR App. The system must pass the ANI AND DNIS information from Span-1 to Span-2 as the IVR Application requires this information to perform its services. Finally, the IVR System after performing its function will transfer the call to an extension (one of the Sip Phones) The IVR System will perform a Flash Hook followed by the extension number of the SIP PHONE. If the incoming call does not match a list of DID numbers, then the call is sent straight to an auto attendant on the asterisk box, bypassing the span-2 T1 stuff. Any thoughts on asterisks ability to do this would be appreciated. I feel confident in all the sections with exception to Will asterisk be able to handle the flash hook from IVR and thus pass the call to the extension. Thanks Very Much for you time in helping me with this. ~ron ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] MF instead of DTMF
Are you talking about SIT and all of the announcements that are for: Work stoppage no dial tone switch blockage emergency announcments misdialing vacant disconnects etc? James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael B. Murdock Sent: Tuesday, April 19, 2005 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] MF instead of DTMF Thank for the reply Jim, I realize FGD uses MF and obviously there is a MF decoder in asterisk. What I am trying to determine is if asterisk can detect MF digits after the call has been presented using FGD call setup. What I am trying to determine is if Asterisk can be used as a replacement for the Class Announcement periphial for a Class-5 (DMS-10) switch. I am trying to get the specific Nortel or Telcordia spec on this feature but have been told by one switch tech that the specific announcement (or string of announcements) to play is indicated by a variable number of outpulsed MF digits after the trunk is seized. -- Mike - Original Message - From: jltaylor [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, April 19, 2005 9:56 AM Subject: RE: [Asterisk-Users] MF instead of DTMF MF works with FGD FGC signaling. Are you taking FGD with a tandem connection? James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael B. Murdock Sent: Tuesday, April 19, 2005 8:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] MF instead of DTMF I am looking into using Asterisk for an application where the upsteam switch will provide MF digits instead of DTMF after establishing a call. This is not during the call set up but after the call is established additional MF digits will be passed to indicated features to provide to the caller. Trunking will be EM T1. Does asterisk support MF detection in addition to DTMF? Has anyone done anything like this before? -- Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for ATAs
I believe the routing is only to either VOIP or the local pots line. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anton Krall Sent: Tuesday, April 19, 2005 10:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Looking for ATAs James. With sipuras 3000 would I be able to deploy multiple atas on diff. locations and be able to use the PSTN (FXO) between them? Also, for example, I was thiking about this scenario: Location 1: 1 FXO line only and 2 FXS, can I use a sipura 3000 (1 FXO and 1 FXS) and a sipura 2000 (1 FXS no router) and allow both FXS to call each other thru the routing on the sipura 3000 and share the FXO between both FXS and also incoming sip calls from other locations to use the 3000 FXO? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jltaylor Sent: Martes, 19 de Abril de 2005 09:59 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Looking for ATAs The SIPURA 3000 allows some dialplan programming. You might check them out. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Anton Krall Sent: Monday, April 18, 2005 5:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Looking for ATAs Im looking for ATAs that have 1 FXO and 2 FXS ports, they will connect to a central Asterisk server and they idea is to share the FXO between the ATAs for people in location 1 can call the persons extension in location 2 or use locations 2's POTS lines to dial as a local call. Any recommendations? Also, how do you go about a dialplan for this? Does asterisk have to manage routing for this? Configure the ATAs so that if call not local or between the 2 local FXS, then route thru asterisk for termination on some other ATA (to their FXS or FXO)? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Calling Card
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Huddleston, Robert Sent: Monday, April 18, 2005 10:45 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Calling Card Anyone experimented with Calling Card support in * Am I wrong in presuming that if I have one calling card caller call in and want to complete a call I will use 2 lines (1 for the customers inbound and another to complete the remote call)?? Thanks You are not wrong ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Motherboard failure with 2 Digium TE405P car ds
I'll have to agree. Check power supplies under load and see what kind of voltage you are getting (12/5v legs). James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of mattf Sent: Monday, April 18, 2005 2:00 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Motherboard failure with 2 Digium TE405P car ds I have 2 temperature probes in the server, they record peak temperature and neither have gotten within 5 degrees of our peak usage Asterisk servers' average temperature. Also the current machine has all new components and no dust buildup or fan blockage. Our server room is monitored by two independant room temperature sensors that log temperature every 15 minutes and if it gets over 85F the system will phone 3 of us every 15 minutes until the temperature goes down or the system is turned off. We have not had any AC problems since we put the new AC system in 6 months ago. We went to this length because we have had several of the things you mentioned happen to us as well, The server room has a dedicated AC unit, you need a key to change the thermostat temperature, and our machines have very good air flow front to back with either very few or no significant heat traps. This seems to be a power or motherboard issue that I cannot figure out. Does anyone have the actual power usage rating of the Digium TE405P card? Thanks, MATT--- -Original Message- From: Race Vanderdecken [mailto:[EMAIL PROTECTED] Sent: Monday, April 18, 2005 2:27 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Motherboard failure with 2 Digium TE405P cards Just from long term experience it might be a heat problem. Check the really basic stuff first. The air flow might not be adequate for the box. Make sure your ribbon cables and such are not blocking flow. Two cards might draw too much power, causing the power supply to overheat causing everything to overheat. Don't add more fans, put in better/more efficient fans or a better power supply. After six months are you getting dust build up on the fans or vents? More dust traps more heat which cause more power to be needed to run fans and convert AC/DC which causes more heat, and so on. But you are reporting a five week breakdown. Put a recording thermometer in your boxes. It could be the cooling is not running as expected in the room. Do you own the room? I once had a room where the janitor would shut the air-conditioning off at night because he knew nobody was in there. Then he would turn it back on in the morning before I got there. The machine was dead, but the room was ice cold. That took three weeks and a lot of IBM repair guys later to discover. I only found it because I checked the room on a weekend and it was 90+ in there. Don't over tax the air-conditioners. I once had a room where the company insisted on keeping it at 60 because things kept over heating, every time there was an over heating they pushed the thermostat lower. Turns out the air-conditioner was turning itself off because it was overheating from the demand. Then after a few hours off it would comeback on, cool and overheat itself because it was unable to keep the room as cold as a meat locker. Even better, another time someone brought in a portable cooler to keep a room/closet with a switch in it like an icebox. They vented the heat from the portable cooler out of the room into the dropped ceiling via a 20 foot exhaust hose. By stuffing the exhaust hose into the plenum the hose was accidentally pointed at the thermostat of the HVAC thermostat for the entire office. So with 90+ degree air pointed at the office HVAC thermostat the office HVAC thought it was 90 inside and kept the place so cold we could barely work there during the winter. Moral of the story, sometimes it ain't anything you are doing. Race the Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mattf Sent: Monday, April 18, 2005 10:35 AM To: 'asterisk-users@lists.digium.com' Subject: [Asterisk-Users] Motherboard failure with 2 Digium TE405P cards Hello, I have spend a long time trying to figure out exactly what is the problem with one of my Asterisk servers, it is the only one at any of our locations that has two Digium quad T1 cards in it with 7 T1s connected to it. Most of the rest of our Asterisk servers run identical hardware except that they only have a single TE405P board in them. Here's what seems to happen to this system starting 6 months ago: Take brand new Asus motherboard with P4 processor, 2GB RAM, SATA or SCSI drives and two TE405P Digium quad T1 boards. Hook up one local and one long distance T1, hook up 4 crossover PRIs to other telco equipment, hook up one channelbank. The system will run perfectly for about 5 weeks, then randomly the channel bank users will notice a weird audio cracking sound and the system will crash. Upon
RE: [Asterisk-Users] problem connecting multiple boxes via IAX2
Send me a copy of your iax.conf and your extensions.conf. I'll look at it. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of MobilPete Sent: Saturday, April 16, 2005 6:18 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] problem connecting multiple boxes via IAX2 Senerio multiple * boxes connecting to a central * box with T1 card via IAX2. 1box 1 abd 2 work fine all the time box 3 - after approx 10-15 minutes with no calls - central box with T1 card fails to deliver incoming calls to box 3. Connectivity is good, * exten-2-exten good in order to allow incoming calls again, we only need to make 1 outbound call from box 3. Then everything works well again. Can anyone shine some light on this problem? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT: google groups Asterisk-test and nowAsterisk-Users marked as spam on Gmail
My spam filter started showing that the list IP was blacklisted this morning. It seems to be cleared up now. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andy Hamilton Sent: Friday, April 15, 2005 2:44 PM To: Sig Lange; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: google groups Asterisk-test and nowAsterisk-Users marked as spam on Gmail Right here. I find this quite upsetting. You might think that after telling Gmail that a hundred messages are not spam, let alone having a filter to apply a label to all messages with [Asterisk-Users], would be enough. Apparently not. I'm also not aware (correct me if I'm wrong) of any method to tame the spam filter for my particular mailbox, but although Gmail is grand, the spam filter sure doesn't seem to be a learning one. It would also appear that Asterisk-Bis, Asterisk-BSD, and Asterisk-Security are unaffected. -Andy On 4/15/05, Sig Lange [EMAIL PROTECTED] wrote: Starting around Apr 14th Gmail has started marking all messages for Asterisk-Users as spam. Prior to that on google groups someone created a asterisk-test group (seperate from this group). Is this perhaps related? I believe it all has happened within a week time frame. Gmail is a great service but if this is what's going to happen I will quit using gmail. I'm giving a shot out to see any other gmail users out there having this problem. My Asterisk-Dev seems to be unaffected. Who's having similiar issues? TIA, Sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone already pionered outing calling with userselcted background noise?
Well this is about as cool as having Asterisk do voice changing (can we just make the guy sound like a girl?). I'll work on this. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of magnus Sent: Friday, April 15, 2005 3:21 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Anyone already pionered outing calling with userselcted background noise? Hello, We are at the beginning of an asterisk project to be able to have callers call in on premium rate number, (Which funds outgoing call) be presented with ivr menu choice of background noises, then be presented with external dial tone, outgoing dialled digits collected and then dialled and once connected, background music invoked as well as connecting original caller and dialled person. For example, guy's late home, needs an excuse, wants to call home and pretend his flight is/was delayed, thus he would dial in, select airport background music with canned tannoy recording of flight delays and then enter his home telephone number. Or variations of theme. We think it can be done with Asterisk, simple IVR, E1 Zap channels, but thoughts on how would we mix canned selected audio file and outgoing call? Input from anyone's that's tried this welcome. Thanks Magnus ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bridging 2 Zap channels
I've seen this. I think that I wasn't getting CPC from the phone company. I had an ISDN BRI into a TA with two pots lines coming out. When I converted to just two post lines (no isdn) then it never happened again. James -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Matthew Drobnak Sent: Friday, April 15, 2005 7:23 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bridging 2 Zap channels Here I was thinking I was the only person to see this - somehow, one day, we tied up two lines for 6 hours, from my uncle dialing his cell phone...Hanging up, dialing it again...Both lines stayed up, and yes, I think connected to each other. Mind you, this is with FXO channels, but same idea. Anyone? -Matt Paul Hewlett wrote: I am running * 1.0.6 with 8 analogue phone lines connected to 2 cards - lspci reveals these as : 03:04.0 Communication controller: Tiger Jet Network Inc. Model 300 128k 03:05.0 Communication controller: Tiger Jet Network Inc. Model 300 128k The wcfxs module is loaded successfully and I have the first 3 lines actually connected. /etc/asterisk/zapata.conf is correct (channels = 1-3) The problem is that under certain circumstances (which I am unable to determine) * bridges 2 of the Zap channels together even though I can see no possible way in the dialplan. This then permanently consumes 2 lines leaving only one available. I have been watching the system for 2 days now and have managed to trap it into this condition twice - the system is only under light load. Can anyone suggest a means of tracking this down via debug commands and suchlike ? Has anyone else seen this and what was the fix ? Paul He ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] distribute outbound calls
Any ideas on how to rotate (evenly distribute) outbound calls over a number of 'trunks' or contexts? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: list format vs newsgroup format
IF there was a consideration for a change, I prefer: phpbb it's open source and easy to use. www.phpbb.com you can still get emails from the posts. -- Original Message -- From: Chris Earle (CBL) [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Fri, 18 Jul 2003 10:02:22 -0400 Agh I hate trying to sift through all these messages and keep track of the various threads going on . Who else on here prefers the newsgroup/threaded approach? If you haven't already, check out news.gmane.org for mailing lists turned into newsgroups readable by news readers... only problem being that this list requires list membership before postingShrug C h r i sE a r l e System Solutions Specialist ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Cygwin?
Is the hassle in running it or setting it up? This gets back to my interest in a CD to boot and install a basic system on a hard drive. Something like a 2 line 4 station version and then a single T1, 4 station, 2 line. This is why there is a users list and a developers list. As a user, I just want a CD with typical config ready to go. As a developer, I want to play with and tweak everything, including the OS. Others are asking for a GUI or web interface. There's a place for all of this. Look at what's involved in getting started: You either have to download 600+MB Linux, install, compile, etc. Or run out and buy a Linux version and install, compile, etc. Now for most of us this is not a big problem. But, just look at the time envolved in setting up a couple of 266mz boxes to play and test with. Put an ISO on the site and watch hardware sales fly... And then watch the consultants market grow. There will be posts like: ...well I bought the hardware, installed, it works but I need xxxyyy, can any one log into my system and program this thing?... James Taylor [EMAIL PROTECTED] 903-793-1953 -- Original Message -- From: Chris Earle \(CBL\) [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Wed, 16 Jul 2003 02:23:12 -0400 Hey all, quick question: does asterisk work okay in a Cygwin environment? I want to install it on my cygwin setup for local testing/demoing and save me the hassle of using a pure linux machine As long as it doesn't take a huge huge performance hit from running out of Cygwin, then I'll have a go there for a start confirmation appreciated! thanks -- C h r i sE a r l e System Solutions Specialist ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on Cygwin?
Thanks for your enthuastic response. There's this Linux project out there for 802.11 at: www.station-server.com They have figured out how to make this type of distribution package work. Don't get me wrong, Asterisk seems to have just about everything from a feature standpoint. The open source concept is one that I support. We run FreeBSD for routers and I love it. You are absolutely correct about the need to learn about Linux distributions and installers. Most people don't and some find it too difficult (I suppose that they are the ones who should stick to Windows?). Hardware costs? I guess these guys that have a hardware cost problem have never priced a Dialogic 240xx/T1 or the quad card. Used single T1 Dialogic cards are $1100. *** The digium hardware offerings are the best price that I've seen for any solution. *** -- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: 16 Jul 2003 09:14:06 -0500 On Wed, 2003-07-16 at 09:10, jltaylor wrote: Is the hassle in running it or setting it up? This gets back to my interest in a CD to boot and install a basic system on a hard drive. Something like a 2 line 4 station version and then a single T1, 4 station, 2 line. This is why there is a users list and a developers list. As a user, I just want a CD with typical config ready to go. As a developer, I want to play with and tweak everything, including the OS. But there is _NO_ typical config. This is enough of a problem. Plus there is no need to host an ISO of the OS and cost digium money in bandwidth that other people are more than willing to do. The maintaining of a OS ISO is immense and best left to other projects. In fact the only thing really needed is for someone to set up a nightly build and package of asterisk into the couple of different package formats and make it available to the world. Others are asking for a GUI or web interface. There's a place for all of this. Look at what's involved in getting started: You either have to download 600+MB Linux, install, compile, etc. Or run out and buy a Linux version and install, compile, etc. Now for most of us this is not a big problem. But, just look at the time envolved in setting up a couple of 266mz boxes to play and test with. See you need to learn about other distros and installers. I know that Mandrake and RH offer network installs, and debian shouldn't be installed any other way. I'm only commenting on debians network install because I know it, but you only download 28 megs of files and then only what you are going to install after that. Total download for an asterisk machine should be under 150 megs. Put an ISO on the site and watch hardware sales fly... Do you think the ISO will change all these VoIP only users into hardware users? If you listen to the comments from them, it is a cost issue mostly on the hardware, not the software. No amount of software bundling is going to change the budget of a user. And then watch the consultants market grow. There will be posts like: ...well I bought the hardware, installed, it works but I need xxxyyy, can any one log into my system and program this thing?... I doubt this. The consultants market will be more of the kind like VCCH is doing which is going out to a site and saying, We can provide you this, that, and these other things all for a price under that quote you have in your hands now. The difference here is that most users that already found their way here and went ahead with a purchase of hardware will either already know how to do it themselves, or are patient enough to wait till that feature comes forward. Those who need consultants usually will not be the ones we see. -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Line Override Device
This power failure thing does not have to be complicated. A few solutions come to mind: 1) A 3,5,12 (whatever is needed) power supply (wall wart)used with a relay (DPDT). When the wall wart has power, the computer takes the call. When power fails, the POTS line falls in to place. Now, this does not delay while the computer is booting up. 2) A basic stamp computer - about $25-30. It has 8 programmable i/o pins that will drive relays. One pin monitors either a wall wart or 5v from one of the plugs on your computer's power supply. When pin 1 goes low (no power) relay kicks in to bypass computer and connect POTS line direct. When power returns program jumps to a sleep or delay statement for xMINS until computer boots. And then releases relay for normal operation. www.parallaxinc.com and resellers. James Taylor [EMAIL PROTECTED] 903-793-1953 -- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: 13 Jul 2003 17:35:55 -0500 On Sun, 2003-07-13 at 15:55, John Laur wrote: You can build a UPS for that, but the better option here is to attach a phone to the phone side of the X100P that is always connected to the POTS line so that even when the computer goes down you can send and receive calls. If you don't want it to ring *unless* the power is out, you could wire it through a normally-closed relay hooked to something simple like the parallel port (there are schematics everywhere for this). When the computer is off, the relay closes, and the phone rings with the line. Heck, if you have an analog set on FXS you want to ring when power goes, you could get a SPDT relay and wire one line into open and one line into closed and switch between them. If you don't care much about incoming calls during the outage, just plugging a phone into the other end of X100p and turning off the ringer will do the trick. It is easier to wire to a 12 volt(yellow) wire off of the PSU, plus this lets you drive larger relays. The specs are available on the net to show you how to wire POE (Power over ethernet). In fact I did my own so I can use the 7960 before we found a suitable wall wart. Basicaly all I did was punch down a keystone with the ethernet data lines, then punched down the power lines so that one side had power and the other didn't so I didn't chance blowing up my switch that was made before they thought of doing POE. I used the power supply from a CAC AB1 that had the ringer module broke on it. It produces 1amp of 48volts and was more than adequate for the 7960. If I had a lot of phones to power, I have a 6amp 48volt PSU from a Premisys channel bank that I picked up at a hamfest for $10. If you do this and plug anything other than the 7960 into it like a NIC you can easily damage it! (google for 'etherkiller' for more) Real power over ethernet injectors provide power only to devices that 'ask' for it, but for small setups they are very much more expensive than the price of a UPS that could power the 7960 for hours (a $30 ups running only the 7960 should go for at least a couple hours) - Compare this to paying $100+ per port for PoE injectors! Putting 'raw' 48V on the Ethernet in an office environment where someone else might accidentally plug something into the wall jack incorrectly would be a disaster! Of course there are some cost savings associated with not having to maintain and upkeep 48 UPS's for 48 phones that make PoE worth it, but I'd say that for less than 12 users it becomes harder to justify. etherkillers are 110 volts AC to data pins, POE is 48 volts DC on non data pins. This should not blow devices that are not expecting PoE. Think about it, how would a device ask for power if it doesn't have power to make the request? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] EZ-Install
Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a basic Linux/Asterisk system? Just re-boot and config. -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] EZ-Install
Not CD based. Just CD install. When you reboot Linux with asterisk is installed. You could add any other tools you think are necessary. User then just does config. -- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: 14 Jul 2003 10:18:24 -0500 On Mon, 2003-07-14 at 10:34, jltaylor wrote: Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a basic Linux/Asterisk system? Just re-boot and config. Might be interesting to build based off of a knoppix cd, but then what do you store the configs to? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] EZ-Install
That sounds interesting... -- Original Message -- From: Matthew Hardeman [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: Mon, 14 Jul 2003 11:11:35 -0500 Maybe it's just me... But I fail to see the reasoning behind branching to a whole new distribution just to support an easy, out of the box Asterisk install. Perhaps just the creation of an RPM package with a basic configuration would be the ticket? The one potential exception to this would be if you wrote a distribution with advanced hardware detection and preconfiguration such that during the install process, Digium hardware is detected and you can go ahead and configure spans and channels, etc. In that case, the distribution might have some unique value. Short of that, I cannot imagine a new distribution just to package together a pre-configured Asterisk configuration. Even if you wrote an installation process like that, couldn't it be just as well implemented with a clever RPM-based installation and some nice plain old userspace configuration tools? Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of jltaylor Sent: Monday, July 14, 2003 11:00 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] EZ-Install Not CD based. Just CD install. When you reboot Linux with asterisk is installed. You could add any other tools you think are necessary. User then just does config. -- Original Message -- From: Steven Critchfield [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED] Date: 14 Jul 2003 10:18:24 -0500 On Mon, 2003-07-14 at 10:34, jltaylor wrote: Has anyone thought about an ISO file that could be used to make a CD for a bootable install for a basic Linux/Asterisk system? Just re-boot and config. Might be interesting to build based off of a knoppix cd, but then what do you store the configs to? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Member
Greetings, I'm ready to start and setup Asterisk. Any preference on which Linux to use? Windows FreeBSD in use here. I'd like to get up and running as quickly and easily as possible. Thanks -- James Taylor [EMAIL PROTECTED] 903-793-1953 -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
