[Asterisk-Users] Future WinCE IP Phone
Hi, Found a nice little video about a prototype phone from broadcom currently sitting in Microsoft WinCE lab. The video is at: http://channel9.msdn.com The video in question is an interview with Mike Hall titled "Windows CE and Windows Embedded Lab Tour". The clip dealing with the VOIP phone is right at the start so you don't need to watch the whole thing (although there is some more interesting stuff such as a programmable sewing machine...). Couldn't find any info about the phone on the broadcom site. It will be nice when the phones are this smart (as well as an order of magnitude cheaper) and VOIP starts selling itself. Skype also might have an even tougher time when MSN messenger intergrates voice again; glad I didn't contribute to the 11 sterling million funding round. Aaron __ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users] Asterisk answering only one (dialed-) Number on a PTMP (German "Mehrgeräteanschluss")?
Hi Marco Asterisk will do exactly what you want, as long as your line is point to multipoint. I did my initial experimentation with Asterisk exactly as you describe. You just need to specifiy in your extensions.conf which MSN numbers you want asterisk to respond to. It will ignore all others. Best and cheapest option would be to try a zaphfc card for 20 euros or so.(Billion, Asustek, etc) Make sure it is on its own IRQ or you will have real problems. Use bri-stuff-0.0.2 from www.junghanns.net. When you have played a bit, you can use a second card in NT mode to connect to your other ISDN equipment, and you can use * as the main switch. Magic! And by the way, your English is much better than my German! Good luck! Rgds Tim wendys wrote: Hi, please excuse my poor englisch. Is it possible to connect a (privat Test-Asterisk) to my privat ISDN and allow him to only answer one dialed number? We have 3 up to 10 Numbers on each (Euro-)ISDN (2 b-chanels), it cant't be done by the last Digits cause the numbers are completely different. For Example: I have 3 Numbers (641717, 928752) Is it possible to tell Asterisk (in Extensions.conf?) to Answer 641717 an ignore incomming calls on 928752? I need this solution to work with Asterisk without disconnecting my Girlfriend from the rest of the world. ;-) I realy tried to find an answer in your archive, Asterisk-Manual, "hitchhiker guide to Asterisk" and in Google but I'm affraid that I didn't know the right "searchstring". Marco W. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Future WinCE IP Phone
Hi Aaron, I've been told that messenger (or LCS) wont be incorporating voice inskin for the next 3 years (could be wrong but it's come from more than one person who should be in the know - btw for what it's worth video multiplexing will also be done offboard until this round of architecture changes are implemented as well). Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Clauson Sent: Wednesday, 23 June 2004 5:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Future WinCE IP Phone Hi, Found a nice little video about a prototype phone from broadcom currently sitting in Microsoft WinCE lab. The video is at: http://channel9.msdn.com The video in question is an interview with Mike Hall titled "Windows CE and Windows Embedded Lab Tour". The clip dealing with the VOIP phone is right at the start so you don't need to watch the whole thing (although there is some more interesting stuff such as a programmable sewing machine...). Couldn't find any info about the phone on the broadcom site. It will be nice when the phones are this smart (as well as an order of magnitude cheaper) and VOIP starts selling itself. Skype also might have an even tougher time when MSN messenger intergrates voice again; glad I didn't contribute to the 11 sterling million funding round. Aaron __ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] CISCO 7960 Goes missing
Cheers all. I've upgraded the firmware on my switch and that seems to have fixed the problem. Very odd. Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: 23 June 2004 05:07 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] CISCO 7960 Goes missing You could also install ngrep and watch the traffic go by on port 5060. Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Matt > Sent: Tuesday, June 22, 2004 6:45 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users] CISCO 7960 Goes missing > > I've got a number (10) Cisco 7960's connected to my network. > All the phones work perfectly except one. > > The asterisk console keeps throwing up: > Jun 22 15:39:15 NOTICE[-1147470928]: chan_sip.c:5887 > sip_poke_noanswer: Peer '4001' is now UNREACHABLE! > Jun 22 15:39:27 NOTICE[-1147470928]: chan_sip.c:4925 > handle_response: Peer '4001' is now REACHABLE! > Jun 22 15:42:08 NOTICE[-1147470928]: chan_sip.c:4925 > handle_response: Peer '4001' is now REACHABLE! > Jun 22 15:43:12 NOTICE[-1147470928]: chan_sip.c:5887 > sip_poke_noanswer: Peer '4001' is now UNREACHABLE! > Jun 22 15:43:36 NOTICE[-1147470928]: chan_sip.c:4925 > handle_response: Peer '4001' is now REACHABLE! > > I've checked the cable and even swapped out the phone but > 4001 is always disappearing off of the network. > > Anyone got any hints? > > Obviously I've added qualify=yes to my sip.conf in an attempt to > troubleshoot this but I'm now getting nowhere fast! > > Matt > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] AstriCon Registration Opens Next Monday, June 28th
Hi Steve Olle mailed me a couple of week ago about doing a presentation at this conference, on "dialplan tips and tricks." I'm working on one right now. Just wanted to let you know. Thanks. Really looking forward to the conference. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] No such extension ...
... I've replaced extension name "s" by "_." and now everything is fine. Nevertheless I'm confused, since I think I have already used "s" as a synonym for "_." (asterisk 0.5 or 0.7) and it also worked fine. Maybe I did not remember right. It works now fine anyway. Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Busy message
> There are other users running the latest CVS-HEAD reporting that problem
> (asterisk segfaults when unable to create channel). Maybe you have to
> revert to a previous version till the bug is fixed. ( cvs -D )
OK, thanks, will try that (btw, cvs -D is an invalid command)
Have you any idea why it couldnt create the channel in the first place?
I configured:
exten => _[123456789],1,NoOp(.call for .${EXTEN})
exten => _[123456789],2,Dial(SIP/${EXTEN},60,tr)
exten => _[123456789],3,Voicemail(u${EXTEN})
exten => _[123456789],103,Hangup
I got...
-- Executing NoOp("SIP/54321-b373", ".call for .12345") in new stack
-- Executing Dial("SIP/54321-b373", "SIP/12345|60|tr") in new stack
Jun 22 13:37:58 NOTICE[13326]: app_dial.c:681 dial_exec: Unable to create
channel of type 'SIP'
regards,
Keith
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Re: [Asterisk-Users] Asterisk -- PBX Do Not Disturb
Yes this is where I wanted to use it the most, I use Asterisk as a software-only based PBX using SIP and IAX, I have an ATA device on my network connected to the PBX but it nor the actual analog phone has a DND function. I am hoping to implement an Asterisk-side based DND somehow but was wondering where to go from there. Thanks Steve Dean Collins wrote: That could explain why it wouldn't work on any of my sip extensions I tried it on this morning when I first read about it and thought cool the things you learn. Is there anyway to make it work on Sip extensions? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron J. Angel Sent: Wednesday, 23 June 2004 10:33 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb john lawler [EMAIL PROTECTED] wrote: You don't have to put this in the dialplan. It's one of those low-level functions in Asterisk (possibly controlled at the driver level-- I'm not sure about that). If you have an extension defined, pick up the handset and dial '*78', you should see on the Asterisk CLI: Enabled DND on channel That assumes your using zaptel, no? This doesn't exist for other channels as it's not built in to the channel driver for anything but zaptel. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Stephen Rosebush, [EMAIL PROTECTED] http://www.desynched.org/ // PSTN USA:1-248-724-4452 x201 Netherlands:+31-(0)20-6598858 x63420 x201 // IP Phone FWD:63420 x201 IAXTEL: 1-700-356-6191 x201 SIP:sip:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users] Asterisk answering only one (dialed-) Number on a PTMP (German "Mehrgeräteanschluss")?
Hi Marco, wendys wrote: Hi, please excuse my poor englisch. Is it possible to connect a (privat Test-Asterisk) to my privat ISDN and allow him to only answer one dialed number? We have 3 up to 10 Numbers on each (Euro-)ISDN (2 b-chanels), it cant't be done by the last Digits cause the numbers are completely different. For Example: I have 3 Numbers (641717, 928752) Is it possible to tell Asterisk (in Extensions.conf?) to Answer 641717 an ignore incomming calls on 928752? I need this solution to work with Asterisk without disconnecting my Girlfriend from the rest of the world. ;-) I did this with an AVM Fritz Card and capi_chan from http://www.junghanns.net/asterisk/ You can define incoming and outgoing MSNs in capi.conf so you won't get in conflict with you other MSNs There is some documentation on it in at voip-info.org (seems to be currently down) jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] cdr_mysql compilation error
I am trying to compile current cvs asterisk-addons for mysql cdr but I get the following error. Iam running mysql 4.0.20 and cvs v1 stable version of asterisk. cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:108: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:108: (Each undeclared identifier is reported only once cdr_addon_mysql.c:108: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 Any idea? -- Manuel Marin Garcia TRANSTELCO S.A. DE C.V. Campos Eliseos 9050 B4 – Cd. Juárez, Chih. 32452 - México Oficina: +52 656 692 11 09 – Fax: +52 656 692 1112 - Celular: 915 727 6141 http://www.transtelco.com.mx ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Cisco ata-186 port died
I've seen this when the port on a switch negotiated at 100mb/s not 10mb/s Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jacob Hunter Sent: 23 June 2004 05:29 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ata-186 port died I use both ports on my cisco ata-186. I run them using ulaw. Today I made numerous calls using my analog phone on port 2. I picked it up about an hour after the last call I made and the line was dead. There is no power at all over the phoneline to the phone, and the red light doesnt light up. The configuration is verified as unchanged. Has anyone seen this problem before. I was unsucessful in finding anything on google and wiki about it. jacob ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] pwlib compile error
Thank you. But, I am fairly certain that I am using the proper versions. They certainly match what is in the README!!! Perhaps I have some contagion from a previous install, but finding it seems to be defeating me. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy McNamara Sent: Tuesday, June 22, 2004 9:18 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] pwlib compile error [EMAIL PROTECTED] wrote: > I am trying to compile pwlib in order to get h323 support working. Most > of the compile is fine except it falls over at the point below. Does > anyone have a solution? You are not using the proper version of PWLib (and i'll guess Open H.323 also) Read the README Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Core Dump on app_dial.c
CVS update. Michael. John Baker wrote: Wondering if anybody else is experiencing this: Using June 21st CVS Call made internally from one Polycom IP600 to another. Core dump with the last message in log as: NOTICE[17426]: app_dial.c:681 dial_exec: Unable to create channel of type 'SIP' Happens a couple of times a day. No, I haven't done any backtracing, verbose logging, etc., (first thing in the morning, I promise) I just wanted to see if anybody might have a quick fix. Thanks John Baker ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call generator
Hello, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Thanks by advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[1-9] as E? (was: [Asterisk-Users] Busy message)
Keith Waters schrieb:
...
I configured:
exten => _[123456789],1,NoOp(.call for .${EXTEN})
I consider a short key for [1-9] at least
as useful as N for [2-9], maybe even more useful.
For my internal purposes I'm using E for [1-9].
Am I the only one, who is missing something short
for [1-9]?
Roger.
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RE: [Asterisk-Users] pwlib compile error {Scanned}
If you have not installed PWLib in your home directory (~/pwlib) then you
will have to define the environment variable PWLIBDIR to point to the
correct directory.
Also make sure you have added the $PWLIBDIR/lib directory to your
LD_LIBRARY_PATH environment variable if you intend to use shared libraries
(the default under Linux).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of jc
Sent: Wednesday, June 23, 2004 4:44 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] pwlib compile error {Scanned}
Thank you. But, I am fairly certain that I am using the proper
versions. They certainly match what is in the README!!!
Perhaps I have some contagion from a previous install, but finding it
seems to be defeating me.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeremy
McNamara
Sent: Tuesday, June 22, 2004 9:18 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] pwlib compile error
[EMAIL PROTECTED] wrote:
> I am trying to compile pwlib in order to get h323 support working.
Most
> of the compile is fine except it falls over at the point below. Does
> anyone have a solution?
You are not using the proper version of PWLib (and i'll guess Open H.323
also)
Read the README
Jeremy McNamara
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[Asterisk-Users] Asterisk Database
Anyone know is there a limition on size of asterisk database... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Réf.: [Asterisk-Users] Call generator
Hi, sipp (http://sipp.sourceforge.net/) seems to be a good app. Take a look at http://www.voip-info.org/wiki-SIPP on the wiki to have more info about it... Basically, there is scenario which are describe there and I personnally generated about 3,000,000 calls before having to restart asterisk and i placed about 90 concurrent calls. Good luck! [EMAIL PROTECTED] a écrit : - Pour: <[EMAIL PROTECTED]> De: "GIBERT Frédéric" <[EMAIL PROTECTED]> Envoyé par: [EMAIL PROTECTED] Date: 23-06-2004 10:46 Objet: [Asterisk-Users] Call generator Hello, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Thanks by advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Call generator
GIBERT Frédéric wrote: Hello, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. How about this? http://sipsak.berlios.de/ jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ireland PSTN Number
Hi, Does anyone know of a provider/terminator of Belfast, Ireland telephone numbers? Thanks in advance Regards Dave ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CSV log stopping
If I have ODBC logging enabled (with cdr_odbc), Asterisk logs everything to ODBC *and* to the CSV file (Master.csv). If I issue a "reload", it stops logging to the CSV file, but continues logging to ODBC. To have it log to the CSV file again, I have to issue "unload cdr_csv.so" then "load cdr_csv.so". Is that normal behaviour? Is it supposed to log to the CSV file with ODBC enabled or not? I'm using CVS-06/21/04-22:36:21 Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Accountcode missing in log
I have defined a SIP friend without username and secret, only IP-based. I have also defined an accountcode for that "friend", as follows: [mypeer] type=friend host=192.168.0.100 port=5060 context=mycontext canreinvite=no accountcode=mypeer Unfortunately the accountcode for the calls originating from "mypeer" doesn't show up in the log (either CSV or ODBC). All the other "friend"s I have (which authenticate with username and secret) also have an accountcode, and it shows up in the logs correctly. Is this normal behaviour? Does a "friend" without username/secret *not* log the accountcode when it places calls? Thanks -Manuel ___ Ticinocom SA - Via Stazione 5 - 6600 Muralto Tel 0844 007070 - Fax 0844 007071 http://www.ticinocom.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Call generator
http://lists.digium.com/pipermail/asterisk-users/2004-May/048245.html -- Comgate Telco>Internetmailto:[EMAIL PROTECTED] Behalf Of GIBERT Frédéric Sent: 23 June 2004 09:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call generator Hello, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Thanks by advance. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iax unable to transfer
Dear List
I have notice this kind of problem between my two * box.
My scenario is :
Iax GSM
IaxClient->PBX1>PBX2-->TDM
today CVS Stable V1
I use as Client FireFly with IAX2/GSM and try to call my PBX1 this server call
PBX2 to terminate the call trought a TDM line (TE410P) but after PBX2 join
the two call i can see the log below from my PBX1, i can speak for few second
and after the FireFly hangup.
I have try to change * version from Stable to today CVS but no success same
problem.
I have enabled the IAX Debug and seems the RX side (PBX1) dont accept
something from PBX2 and show the "unable to transfer" (im not expert) :-)
The strange thing is if i call from Sip Phone/client i dont have a problem the
Call is bridged!
The events from the CLI:
-
Executing Dial("[EMAIL PROTECTED]/5",
"IAX2/out:[EMAIL PROTECTED]/[EMAIL PROTECTED]|60|g") in new stack
-- Called out:[EMAIL PROTECTED]/[EMAIL PROTECTED]
-- Call accepted by 213.215.xx.xx (format GSM)
-- Format for call is GSM
-- IAX2[out]/6 stopped sounds
-- IAX2[out]/6 is ringing
-- IAX2[out]/6 stopped sounds
-- IAX2[out]/6 answered [EMAIL PROTECTED]/5
-- Attempting native bridge of [EMAIL PROTECTED]/5 and IAX2[out]/6
-- Channel 'IAX2[out]/6' unable to transfer
-- Hungup 'IAX2[out]/6'
-- Executing Hangup("[EMAIL PROTECTED]/5", "") in new stack
== Spawn extension (incoming,001223445, 4) exited non-zero on 'IAX2
[EMAIL PROTECTED]/5'
-- Executing Hangup("[EMAIL PROTECTED]/5", "") in new stack
---
Thanks in advance for possible help!
Dimitri
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[Asterisk-Users] connecting to Iconnect here using asterisk
Hi,
I wish to connect several ATA186 Phones to each other, to iconnecthere
and to the PSTN using asterisk.
Please tell the appropriate settings for firewall (ports to
open etc…) sip.conf and extensions.conf(part relevant to iconnect).
Also I would be glad to get a working example of your ATA186
configuration.
I tried searching the mailing lists and several sites but
did not find an answer.
My current configuration:
* Asterisk installed on the Gateway
(Bound to internal network and to Internet) (Not behind NAT).
* Several Cisco ATA186 adapters with
SIP firmware (Behind NAT with asterisk set as their “GkOrProxy”).
Current State:
·
I manage to place calls to my other internal phones.
·
Asterisk does not register at Iconnecthere.
·
I am not able to place calls through Iconnecthere.
Thanks,
Configuration files: (commented lines represent options I tried).
n= my Iconnect phone number
p=Iconnect password
u=Iconnect User id
sip.conf
[general]
port = 5060
;bindaddr = 0.0.0.0
;bindaddr = 212.199.7.106
context = from-sip
;callerid=No CallID
;register=nnn:[EMAIL PROTECTED]/ nnn
;register= : @natrelay.deltathree.com/phone
;register=:[EMAIL PROTECTED]/phone
[iconnect]
type=friend
secret=
username=
host=sipauth.deltathree.com
;host=natrelay.deltathree.com
;fromdomain=xxx.xxx.xxx.xxx
;dtmfmode=inband
nat=no
;nat=yes
canreinvite=no
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=G726
;ATA 186 adapter
[dave]
type=friend
username=dave
secret=
nat=yes
host=dynamic
canreinvite=no
;qualify=200
defaultip=192.168.50.2
;ATA 186 adapter
[french1]
type=friend
username=french1
secret=
nat=no
host=dynamic
canreinvite=no
qualify=200
defaultip=192.168.50.3
extensions.conf
[general]
static=yes
writeprotect=no
[globals]
ICONNECT1=16463752819
CONSOLE=Console/SIP
;[macro-dialiconnect]
;exten => s,1,SetCallerID(${ICONNECT1})
;exten => s,2,SetCIDName(${MYNAME})
;exten => s,3,Dial(SIP/[EMAIL PROTECTED],${ARG2})
;exten => s,4,Playback(new/acnt-or-cir-busy-now)
;exten => s,5,Hangup
;exten => s,104,Playback(new/acnt-or-cir-busy-now)
;exten => s,105,Wait,3
;exten => s,106,Playtones(congestion)
;exten => s,107,Wait,30
;exten => s,108,Playback(new/are-you-still-here)
;exten => s,108,Hangup
[from-sip]
exten => _.,1,NoOp
;exten => _.,1,Macro(record-on,${EXTEN},${CALLERIDNUM})
exten => _.,2,Goto(from-sip-post,${EXTEN},1)
exten => i,1,Hangup
exten => h,1,Hangup
[from-sip-post]
exten => 16463752819,1,Dial(${PHONE1},20,Ttm)
exten => 16463752819,2,Playback(transfer)
exten => 16463752819,3,Macro(dialiconnect,${MYCELLPHONE},20)
exten => 16463752819,4,Voicemail(u${PHONE1VM})
exten => 16463752819,5,Hangup
exten => 16463752819,102,Voicemail(b${PHONE1VM})
exten => 16463752819,103,Hangup
exten => 222,1,Dial(SIP/dave,30,t)
exten => 333,1,Dial(SIP/french1,30,t)
;exten => 444,1,Dial(SIP/[EMAIL PROTECTED]:[EMAIL PROTECTED],30,t)
exten => 555,1,Dial(SIP/[EMAIL PROTECTED],30,t)
exten => t,1,Goto(#,1)
; If they take too long, give up
exten => i,1,Playback(invalid)
; "That's not valid, try again"
exten => h,1,Hangup
[intern]
exten => _.,1,NoOp
exten => _.,2,Goto(intern-post,${EXTEN},1)
exten => i,1,Hangup
exten => h,1,Hangup
[iconnect-forced]
; Experimental "forced" dialing through iconnect
to make calls
; prefixed with "6" go out the iconnect
channel. This is to
; test some functionality for inbound connections; feel
free
; to comment it out.
;
; Dial out on iconnect and wait for 70 seconds for a connect
;
; If no connection in 70 seconds, jump to fastbusy macro
;
exten => _7XXX,1,Macro(dialiconnect,${EXTEN:1},70)
exten => _7XX,2,Macro(fastbusy)
[intern-post]
; if someone dials a "9" in front of their number,
send out via iconnect (commercial PSTN gateway)
include => iconnect-forced
David Leon
Lowes
System Administrator
[EMAIL PROTECTED]
The
Nation Traffic
RE: [Asterisk-Users] Future WinCE IP Phone
There is a very good and working WinCE IP phone available from SJPhone. Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Wednesday, 23 June 2004 18:03 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Future WinCE IP Phone Hi Aaron, I've been told that messenger (or LCS) wont be incorporating voice inskin for the next 3 years (could be wrong but it's come from more than one person who should be in the know - btw for what it's worth video multiplexing will also be done offboard until this round of architecture changes are implemented as well). Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Clauson Sent: Wednesday, 23 June 2004 5:42 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Future WinCE IP Phone Hi, Found a nice little video about a prototype phone from broadcom currently sitting in Microsoft WinCE lab. The video is at: http://channel9.msdn.com The video in question is an interview with Mike Hall titled "Windows CE and Windows Embedded Lab Tour". The clip dealing with the VOIP phone is right at the start so you don't need to watch the whole thing (although there is some more interesting stuff such as a programmable sewing machine...). Couldn't find any info about the phone on the broadcom site. It will be nice when the phones are this smart (as well as an order of magnitude cheaper) and VOIP starts selling itself. Skype also might have an even tougher time when MSN messenger intergrates voice again; glad I didn't contribute to the 11 sterling million funding round. Aaron __ Do you Yahoo!? Yahoo! Mail is new and improved - Check it out! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Skype 4 Linux
Hi All, Since 21 june skype is available to be used on Linux, with a static binary, which includes QT, of 8 meg its big. http://www.skype.com/help_linux_faq.html I presume, with some hacking, there could be a possibility to use the Skype program as a Channel. (Eq. Skype is started, and with a visual scripting thing a connection is made and Asterisk connects via OSS (or the alsa emulation layer)). It is a bit of work, but reverse enginering is too :) Stefan ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Two SIP servers communicating without IAX
Hi! > If I call from an MX250 phone to an Asterisk phone, the conversation is > ok, but there is a noticeable delay in the voice stream. You will probably want to start by analysing your ethernet network (using ping, traceroute, ethereal etc). You might also want to eliminate a switch in between and directly link the two systems to see if things improve. > If I call from an Asterisk phone to an MX250 phone, I can talk FROM the > MX250 phone TO the Asterisk phone, but not the other way around. In > other words, the Asterisk phone will hear everything that is said from > the MX250 phone, but if I say anything on the Asterisk phone the MX250 > phone never gets it. Look at your codec configuration (disallow=all followed byallow= statements). Do a "SIP DEBUG" on the Asterisk CLI to learn more about the codec negotiation before/during a call. Finally check your phones VAD/ silence suppression settings, make sure those are turned off. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Failover of IAX or Spillover as the case may be
Hi! > outbound long distance calls made simultaneously. I just contacted > voicepulse regarding their Connect service. They support 4 outbound calls > at a time. > > My question is - How does one configure Asterix to use: > > 1. Another voicepulse account? (spillover) > 2. If first account is not functional route to another IAX (failover) First of all: Use SetGroup(), CheckGroup() and GetGroupCount() to solve your problem. Secondly you could also use the exit codes of Dial() for failover action, but better prevent the necessity for that with the help of SetGroup(). Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] general install??
(B (B (BNeed to work out if I can use Asterisk as a (Bsoftware PBX or not? (B (BI reside in Japan and my office has a ISDN line (Bwith 2 numbers.I am currently using analogue phones which are connected to (Bmy router(yamaha RT 57i) which has router/Voip/VPN functions.So my ST line (Bis connected to my ST port of the router and tel lines are connected to the (Banalogue ports(2) (B (BNow I am moving to a bigger office and require a (Bpbx.What I want to know can is can I configure a pbx using Asterisk as all (Bthe hardware I saw didn't seem to support ISDN? (B (BAlso I see I may have two options (B (B1. Leave the tel lines connected to the router and (Bjust run asterisk as a software solution that communicates over the lan with my (Brouter for incoming and outgoing calls.Although not sure if this can (Bwork (B (B2. Connect my tel lines to the computer running (Basterisk and do standard config. Can this work when my line is (BISDN??? (B (BPlease give me an idea how and if I can (Bproceed. (B (BThanksMatt
[Asterisk-Users] USB handset for IAX softphone ?
Hi, I am not sure if it's a suitable place to ask this question. Anyone here familiar with USB phone please? I want to buy a cheap one to make it work with IAX softphone. Regards. Steven louse _ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Ireland PSTN Number
On Wednesday 23 June 2004 10:30, David J Carter wrote: > Hi, > > Does anyone know of a provider/terminator of Belfast, Ireland telephone > numbers? Hi, Belfast isn't in Ireland, it's in Northern Ireland, which is part of the UK. Hence any UK provider can give you 0845's, 0870's etc. I know that www.voiptalk.org offer the facility to 'rent' geographic Belfast '028' numbers. Cheers, Gavin. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] capi.so problem on startup
Hi, I'm new to asterisk and try to get it work with capi.so. When I try to start asterisk with "asterisk -c" I get the following errors. I couldn't find any hint on the net what may be wrong in my configs. Has anybody got a hint? Here is the error output: [capi.so] => (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found Jun 23 12:04:33 NOTICE[1075622560]: chan_capi.c:2654 load_module: this box has 1 capi controller(s) -- CAPI[contr1] supports DTMF -- CAPI[contr1] supports supplementary services > HOLD/RETRIEVE > TERMINAL PORTABILITY > ECT > 3PTY > CF > CD > MCID > CCBS > MWI > CCNR == ast_capi_pvt(1272580,1272580,isdndefault,0,2) (0,0,64) == ast_capi_pvt(1272580,1272580,isdndefault,0,2) (0,0,64) Jun 23 12:04:33 WARNING[1075622560]: chan_capi.c:2784 load_module: Unused contr1 Jun 23 12:04:33 WARNING[1075622560]: channel.c:174 ast_channel_register_ex: Already have a handler for type 'CAPI' Jun 23 12:04:33 ERROR[1075622560]: chan_capi.c:2792 load_module: Unable to register channel class CAPI == Unregistered channel type 'CAPI' Jun 23 12:04:33 WARNING[1075622560]: loader.c:328 ast_load_resource: capi.so: load_module failed, returning -1 Jun 23 12:04:33 WARNING[1075622560]: chan_capi.c:2810 unload_module: Unable to unregister from CAPI! == Unregistered channel type 'CAPI' Jun 23 12:04:33 WARNING[1075622560]: loader.c:423 load_modules: Loading module capi.so failed! and here my modules.conf: ; Module Loader configuration file ; ; Asterisk configuration file ; ; Module Loader configuration file ; [modules] autoload=yes ; ; If you want, load the GTK console right away. ; Don't load the KDE console since ; it's not as sophisticated right now. ; noload => pbx_gtkconsole.so ;load => pbx_gtkconsole.so noload => pbx_kdeconsole.so ; ; Intercom application is obsoleted by ; chan_oss. Don't load it. ; noload => app_intercom.so ; ; Explicitly load the chan_modem.so early on to be sure ; it loads before any of the chan_modem_* 's afte rit ; load => chan_modem.so load => res_musiconhold.so ; ; Load either OSS or ALSA, not both ; By default, load OSS only (automatically) and do not load ALSA ; noload => chan_alsa.so noload => chan_oss.so load => res_parking.so load => chan_capi.so ; ; Module names listed in "global" section will have symbols globally ; exported to modules loaded after them. ; [global] chan_modem.so=yes chan_capi.so=yes "modules.conf" 45L, 921C Thanks in advance Tobi ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users] Asterisk answering only one (dialed-) Number on a PT MP (German "Mehrgeräteanschluss")?
Hi! > I have 3 Numbers (641717, 928752) > Is it possible to tellAsterisk (in Extensions.conf?) toAnswer 641717 an > ignore incomming calls on 928752? Sure. However you might want to tell us what ISDN hardware and, more importantly, which Asterisk channel you are using: isdn4linux or chan_capi? Anyway, if you look into the respective configuration files for i4l and chan_capi you will find a setting "incomingmsn=" that will fit your needs. Apart from that you *could* also instruct the ISDN card to listen on all MSNs (incomingmsn=*), but in extensions.conf you then only do an Answer() for the extension that you want to answer. Of course that needs a carefully drafted dialplan where you don't have e.g. an 's' extension in your incoming ISDN context that would still answer the call... Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem when dialing in manager terminal
Hi,
I have configured chan_oss and chan_capi (asterisk from CSV on Monday),
and I can dial from the console:
*CLI> dial 830774
-- Executing Dial("OSS/dsp", "CAPI/872058:b830774|60") in new stack
-- Called 872058:b830774
-- CAPI[contr1/872058]/3 is making progress passing it to OSS/dsp
...
Now I want to dial via telnet for later usage by a soft phone
application, which should be able to remote control asterisk.
I telneted to port 5038, logged in and typed:
> Action: Originate
> Context: local
> Exten: 4080915
> Priority: 1
> Callerid: testtest
> Channel: OSS/dsp
>
Then I get:
> Response: Error
> Message: Originate failed
and in the console:
> channel.c:1828 ast_request: No channel type registered for 'OSS'
> channel.c:1744 __ast_request_and_dial: Unable to request channel
>OSS/dsp
What went wrong?
Thanks for any hints!
Roger.
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Re: [Asterisk-Users] USB handset for IAX softphone ?
On Wed, Jun 23, 2004 at 06:13:42PM +0800, steven louse wrote: > Hi, > I am not sure if it's a suitable place to ask this question. Anyone here > familiar with USB phone please? I want to buy a cheap one to make it work > with IAX softphone. > Regards. > Steven louse Look at www.eutecticsinc.com Those are for windows but at least the IPP200T is also working with Linux using latest alsa. (Dunno if they're the cheapest ones) HTH, Rainer -- http://graphics.cs.uni-sb.de/VoIP/ pgpBj3Q25PKgh.pgp Description: PGP signature
Re: [Asterisk-Users] Skype 4 Linux
On Wed, 2004-06-23 at 10:56, Stefan de Konink wrote: > Hi All, > > Since 21 june skype is available to be used on Linux, with a static > binary, which includes QT, of 8 meg its big. > > http://www.skype.com/help_linux_faq.html > > I presume, with some hacking, there could be a possibility to use the > Skype program as a Channel. (Eq. Skype is started, and with a visual > scripting thing a connection is made and Asterisk connects via OSS (or the > alsa emulation layer)). > > It is a bit of work, but reverse enginering is too :) Probably not worth the work because you certainly would conflict with some software patents. Besides that it's going to be very limited, what you can use something like that for. It's after all only their softphone, that works with it. Nothing else. Kind regards, Martin List-Petersen ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: cdr_mysql compilation error
In article <[EMAIL PROTECTED]>, Manuel Marin Garcia <[EMAIL PROTECTED]> wrote: > I am trying to compile current cvs asterisk-addons for mysql cdr but I > get the following error. Iam running mysql 4.0.20 and cvs v1 stable > version of asterisk. That is your problem. asterisk-addons is not in sync with the "v1 stable" branch of asterisk. 1. Update your asterisk from the CVS HEAD 2. "make" AND "make install" in the asterisk tree 3. You can then successfully "make" in asterisk-addons. This has been answered on this list at least twice in the last week! If you are new to the list, you should always look through the last few weeks' messages to see if your question has already been answered recently. See http://lists.digium.com/pipermail/asterisk-users/ Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codecs and pauses
Hi all My * implementation is working brilliantly with only one small fault left to kill. I'm using IAXTalk from Telappliant for my incoming/outgoing calls to the pstn network; if I set my codec to GSM everything works great - no pauses but quality is a bit poor. If it set the codec to alaw (I think I'm using the correct one - I'm in the UK) I get intermittent pauses on the call. Initially I thought it was just a connectivity thing but I get a latency of less than 10ms to iax.voiptalk.org and I'm using a 2mb leased line. To further ensure it wasn't something on the line I've disconnected everything except the * box, a 7960 phone. My phones are all 7960's using SIP. There is a X100P card in the server moh timing etc but it isn't connected to the pstn. The * box itself is a PIII 833 with 256MB. Not at all stressed as this is all it does. Any hints or tips would be really useful as I'm stumped now. Thanks Matt ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Busy message
Hi!
> > I think the whole idea of "busy" or "unavailable" is flawed.
> > Asterisk sets ${CAUSECODE} with the cause of the call being
> > cleared. You can use this to determine what you want to do.
> > For exmaple if the cause code indicates "unallocated" then
> > you should give the caller some indication that they number
> > they dialed is disconnected or no longer in service.
> >
> > I think I posed an example of how I handle this. Check the archives.
>
> What would the contents of CAUSECODE be when set? I can't find
> documentation of this anywhere.
Hm... not working for me:
I dial a non-existing PSTN number via Nikotel:
-- Got SIP response 404 "Not Found" back from 63.214.186.6
-- SIP/nikotel-out-phil-b8e1 is circuit-busy
However CAUSECODE remains empty (checked thru NoOp({CAUSECODE}).
If instead I use HANGUPCAUSE I get "5" as return; that looks better as
this - correctly - translates 404 into "unallocated".
Cheers, Philipp
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Re: [1-9] as E? (was: [Asterisk-Users] Busy message)
Hi! > I consider a short key for [1-9] at least > as useful as N for [2-9], maybe even more useful. > > For my internal purposes I'm using E for [1-9]. > > Am I the only one, who is missing something short > for [1-9]? Certainly not! :-) Create a [request] entry at bugs.digium.com Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Codecs and pauses
Hi, I've been having similar problems to you. I found after reading an unrelated post, about the Jitterbuffer option in iax.conf, setting this to yes has made things much better. Now the problem I have is that telappliant have lost one of there external routers so I can't connect at the moment :-) What is it with networks breaking today, first Dilbert and now Telappliant??? :-) HTH Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 On Wed, 23 Jun 2004, Matt wrote: > Hi all > > My * implementation is working brilliantly with only one small fault left to > kill. > I'm using IAXTalk from Telappliant for my incoming/outgoing calls to the > pstn network; if I set my codec to GSM everything works great - no pauses > but quality is a bit poor. If it set the codec to alaw (I think I'm using > the correct one - I'm in the UK) I get intermittent pauses on the call. > > Initially I thought it was just a connectivity thing but I get a latency of > less than 10ms to iax.voiptalk.org and I'm using a 2mb leased line. To > further ensure it wasn't something on the line I've disconnected everything > except the * box, a 7960 phone. > > My phones are all 7960's using SIP. There is a X100P card in the server moh > timing etc but it isn't connected to the pstn. > > The * box itself is a PIII 833 with 256MB. Not at all stressed as this is > all it does. > > Any hints or tips would be really useful as I'm stumped now. > > Thanks > > Matt > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] 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] Codecs and pauses
>>I've been having similar problems to you. I found after reading an unrelated post, about the Jitterbuffer option in >>iax.conf, setting this to yes has made things much better. Out of interest what have you set for your dropcount maxjitterbuffer Maxexcessbuffer >>Now the problem I have is that telappliant have lost one of there external routers so I can't connect at the moment :-) DOH! >>What is it with networks breaking today, first Dilbert and now Telappliant??? :-) Probably got something to do with the summer solstice, laylines etc :-> -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Glover Sent: 23 June 2004 12:17 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Codecs and pauses Hi, I've been having similar problems to you. I found after reading an unrelated post, about the Jitterbuffer option in iax.conf, setting this to yes has made things much better. Now the problem I have is that telappliant have lost one of there external routers so I can't connect at the moment :-) What is it with networks breaking today, first Dilbert and now Telappliant??? :-) HTH Chris -- Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 On Wed, 23 Jun 2004, Matt wrote: > Hi all > > My * implementation is working brilliantly with only one small fault > left to kill. > I'm using IAXTalk from Telappliant for my incoming/outgoing calls to > the pstn network; if I set my codec to GSM everything works great - no > pauses but quality is a bit poor. If it set the codec to alaw (I > think I'm using the correct one - I'm in the UK) I get intermittent pauses on the call. > > Initially I thought it was just a connectivity thing but I get a > latency of less than 10ms to iax.voiptalk.org and I'm using a 2mb > leased line. To further ensure it wasn't something on the line I've > disconnected everything except the * box, a 7960 phone. > > My phones are all 7960's using SIP. There is a X100P card in the > server moh timing etc but it isn't connected to the pstn. > > The * box itself is a PIII 833 with 256MB. Not at all stressed as > this is all it does. > > Any hints or tips would be really useful as I'm stumped now. > > Thanks > > Matt > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > 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 [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi error
Hi all, with cvs update -D "6/21/04 21:00:00 CET" i'm getting the following error. Jun 23 13:14:03 ERROR[-1284076624]: chan_capi.c:881 capi_write: dont know how to write subclass 64 If i revert back to cvs update -D "6/21/04 18:00:00 CET" the problem is gone. Any ideas? Thanks, Andreas _ Check out news, entertainment and more @ http://xtra.co.nz/broadband ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk user/host registration
Hi Folks,
I am newbie to asterisk. Recentely I have installed asterisk on Linux Fedora 2 box. After reading some document, I tried to configure the server.
When I connect to our server, SIP user-agent shows that I am logged in. But it doesn't show my system(client) IP when I issue command at astrisk CLI. The O/P is as below.
*CLI> sip show peersName/username Host Mask Port Status2001/2001 (Unspecified) (D) 255.255.255.255 0 UNKNOWN2000/2000 (Unspecified) (D) 255.255.255.255 0 UNKNOWN
I am pasting sip.conf & extension.conf
sip.conf
[general]port = 5060bindaddr = 0.0.0.0context = INVALID;Autocreatepeer= yes
[2000]
type=friendusername=2000secret=2000host=dynamiccontext=from-sipmailbox=100canreinvite=noqualify=300nat=no
[2001]
type=friendusername=2001secret=2001host=dynamiccontext=from-sipmailbox=100canreinvite=noqualify=300nat=no
extensions.conf
[globals]CONSOLE=Console/dsp;CONSOLE=Zap/1;CONSOLE=Phone/phone0IAXINFO=guest;IAXINFO=myuser:mypassTRUNK=Zap/g2TRUNKMSD=1;TRUNK=IAX2/user:[EMAIL PROTECTED]
[bogon-calls]
exten => _.,1,Congestion
[from-sip]
exten => 2000,1,Dial(SIP/2000,20)exten => 2000,2,Voicemail(u${EXTEN})exten => 2000,3,Hangupexten => 2000,102,Voicemail(b${EXTEN})exten => 2000,103,Hangup
exten => 2001,1,Dial(SIP/2001,20)exten => 2001,2,Voicemail(u${EXTEN})exten => 2001,3,Hangupexten => 2001,102,Voicemail(b${EXTEN})exten => 2001,103,Hangup
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
Immediate help is highly appreciated.
Thank you,
Mandar
Yahoo! India Matrimony: Find your partner
online.
RE: [Asterisk-Users] Codecs and pauses
On Wed, 23 Jun 2004, Matt wrote: > >>I've been having similar problems to you. I found after reading an > unrelated post, about the Jitterbuffer option in > >>iax.conf, setting this to yes has made things much better. > Out of interest what have you set for your > dropcount > maxjitterbuffer > Maxexcessbuffer > I haven't! Just tried turning on the jitterbuffer, but I will have a play, as I do get occassional dropout when my machine checks for mail, but as I'm only on ADSL I shouldn't be surprised really. Probably the longest sentance in the world! > > >>What is it with networks breaking today, first Dilbert and now > Telappliant??? :-) > Probably got something to do with the summer solstice, laylines etc :-> > If it's anything to do with that, it's two days late! At least Dilbert has come back now :-) Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [1-9] as E? (was: [Asterisk-Users] Busy message)
... Certainly not! :-) Create a [request] entry at bugs.digium.com Done. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Codecs and pauses
Chris, >>Just tried turning on the jitterbuffer, but I will have a play, as I do get occassional dropout when my machine checks >>for mail I'll also have a play once Telappliant is back up and post the results. Cheers Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Glover Sent: 23 June 2004 12:34 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Codecs and pauses On Wed, 23 Jun 2004, Matt wrote: > >>I've been having similar problems to you. I found after reading an > unrelated post, about the Jitterbuffer option in > >>iax.conf, setting this to yes has made things much better. > Out of interest what have you set for your dropcount maxjitterbuffer > Maxexcessbuffer > I haven't! Just tried turning on the jitterbuffer, but I will have a play, as I do get occassional dropout when my machine checks for mail, but as I'm only on ADSL I shouldn't be surprised really. Probably the longest sentance in the world! > > >>What is it with networks breaking today, first Dilbert and now > Telappliant??? :-) > Probably got something to do with the summer solstice, laylines etc > :-> > If it's anything to do with that, it's two days late! At least Dilbert has come back now :-) Chris -- E Mail: [EMAIL PROTECTED] SIP: [EMAIL PROTECTED] IAXTEL: 17003366726 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] sidetone noticeably loud on analog handsets on T100P
Has anyone experienced this? I did some googling on it through the archives of course, but don't see much discussion of sidetone issues with analog handsets. I'm wondering if there's some way I could be adjusting the sidetone in Asterisk or should I be looking at my FXS channel bank? I have the same problem with an Adit 600 channel bank that I've been trying to resolve without any success. The sidetone is loud and very annoying. The only way to avoid it is to keep the phone mic further away from your mouth. Unfortunately, I think it is a channel bank issue. I've disconnected my channel bank from the Asterisk server and the sidetone is still very loud. I've changed the gain on the channel bank but that hasn't made any difference. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [1-9] as E? (was: [Asterisk-Users] Busy message)
Philipp von Klitzing [EMAIL PROTECTED] wrote: >> I consider a short key for [1-9] at least as useful as N for [2-9], >> maybe even more useful. >> >> For my internal purposes I'm using E for [1-9]. >> >> Am I the only one, who is missing something short for [1-9]? > > Certainly not! :-) Create a [request] entry at bugs.digium.com Does Z not work? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing CLI
Hello I have contacted my line provider who is saying that in order to get my 0845 or 0870 number to id as the incoming number on a landline that i may call i need the following. User must provide - NPI set to E.163/E.164 User must provide - TON = "national or international I have had a good search around and can't seem to find a good answer to this. Does anyone have any idea where i would set this and how. Best Regards Simon ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Eicon Diva 2.0 PCI ISDN Card
Shaun Ewing wrote: On Tue, 22 Jun 2004 14:00:40 -0600, Michael Welter <[EMAIL PROTECTED]> wrote: Does anyone have this card working with Asterisk? Thanks Eicon Diva 2.02 PCI for the most part works perfectly here using ISDN4Linux and the HiSax driver. Outbound DTMF doesn't work though (inbound works fine). -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Are you using it in North America (NI1 protocol)? Thanks, Mike -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [1-9] as E? (was: [Asterisk-Users] Busy message)
Hi, > -Original Message- > > I consider a short key for [1-9] at least as useful as N for [2-9], > > maybe even more useful. > > > > For my internal purposes I'm using E for [1-9]. > > > > Am I the only one, who is missing something short for [1-9]? > > Certainly not! :-) Create a [request] entry at bugs.digium.com http://www.voip-info.org/wiki-Asterisk+Dialplan+Patterns X matches any digit from 0-9 Z matches any digit form 1-9 N matches any digit from 2-9 [1237-9] matches any digit or letter in the brackets (in this example, 1,2,3,7,8,9) . wildcard, matches one or more characters Does this work ?? Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy message and extensions are hanging.
Folks!
1) I have modified the original sip.conf and extension.conf file instead of writing
mine. This looks like a mistake.
2)I have fired off Asterisk Extensions conf with 2 extensions i.e 2000 and 2001 and
made one test call.
I forgot to set a time out. The calls between these two extensions were partially
successful.
After writing my own files, it started working. I went through following steps...
1. Dialed with 2000 as user and password same as user.
2. It logged into the server.
3. When I dialed for 2001, it went to voicemail box and asked me to drop message as
the 2001 was offline.
4. But when I changed username and password from the same system... its not allowing
me to test the system .
5. It has cached all the entries which I am unable to remove.
But now the problem is that these two extensions are still waiting and I do not know
how to kill them.
Please guide me to remove these entries as I have tried by restarting SER and ASTERISK
servers.
Can somebody help?
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Aaron J.
Angel
Sent: Tuesday, June 22, 2004 8:32 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Busy message
Eric Wieling [EMAIL PROTECTED] wrote:
> On Tue, 2004-06-22 at 18:43, Simon Brown wrote:
>> This should be listed as a bug - it is not logical to go to busy,
>> when in fact the extension is unavailable.
>
> I think the whole idea of "busy" or "unavailable" is flawed.
> Asterisk sets ${CAUSECODE} with the cause of the call being
> cleared. You can use this to determine what you want to do.
> For exmaple if the cause code indicates "unallocated" then
> you should give the caller some indication that they number
> they dialed is disconnected or no longer in service.
>
> I think I posed an example of how I handle this. Check the archives.
What would the contents of CAUSECODE be when set? I can't find
documentation of this anywhere.
___
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Re: [1-9] as E? (was: [Asterisk-Users] Busy message)
Aaron J. Angel schrieb: ... Does Z not work? ... Yes, it does. I had to look in the most recent docs first. Sorry! Roger. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Busy message
Hi Keith Keith Waters wrote: There are other users running the latest CVS-HEAD reporting that problem (asterisk segfaults when unable to create channel). Maybe you have to revert to a previous version till the bug is fixed. ( cvs -D ) OK, thanks, will try that (btw, cvs -D is an invalid command) 'cvs -D' is incomplete, you have to specify the date of the version you are requesting after the 'D'. Anyways, it seems that the problem is fixed on CVS. Do a 'cvs update' -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Call generator
On Wednesday 23 June 2004 04:46, GIBERT Frédéric wrote: > Has someone know a good call generator for asterisk including SIP protocol > (freeware if possible)? > I need to stress a plateform and I don't find any. Are there any IAX2 call generators? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] sidetone noticeably loud on analog handsets on T100P
On Wednesday 23 June 2004 07:59, Paul Zimm wrote: > I have the same problem with an Adit 600 channel bank that I've been > trying to resolve without > any success. The sidetone is loud and very annoying. The only way to > avoid it is to keep the > phone mic further away from your mouth. Interesting; I'm using a T100P connected to an Adit600 and I have no sidetone issues. I've used scenarios involving FXS ports connecting to trunk lines on a Norstart MICS KSU as well as regular old phones connected to the FXS ports. The Adit600 does have its own tx/rxgain settings; have you played with those? Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Failover of IAX or Spillover as the case may be
On Wednesday 23 June 2004 06:01, Philipp von Klitzing wrote: > Secondly you could also use the exit codes of Dial() for failover action, > but better prevent the necessity for that with the help of SetGroup(). IAX2 does not return the disconnect cause from the far end; this is necessary in cases where the provider accepts connections but is having PSTN connectivity issues of their own; Dial() jumps to n+101 but you don't know if the number you were calling was actually busy or if there were problems, so your only choice is to try dialing them again with another provider. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Call Generator for ISDN (PRI/BRI)
Hi, I am looking for Call Generator for PRI ISDN and BRI ISDN signals. From where I can found some cheap or 2nd hand call generator (tester/analyzer)? Maybe PCI based will be cheaper than standalone solution. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Ireland PSTN Number
On 23 Jun 2004, at 12:15, Gavin Hamill wrote: Belfast isn't in Ireland, it's in Northern Ireland, which is part of the UK. Hence any UK provider can give you 0845's, 0870's etc. Belfast is on the island of Ireland. This is an important distinction since national calls in the Republic of Ireland are to all 32 counties (that includes Northern Ireland). This means that a Dublin - Belfast call is a national rate call whereas a Dublin - London call is an international rate call. This could be useful information to anyone wanting to terminate calls for the Republic of Ireland. It's what we're doing :-) Stephan Managing Director ANU Internet Services Dangan Business Park Galway Ireland http://www.anu.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Call Generator for ISDN (PRI/BRI)
why not use asterisk with QaudBRI and/or E100P ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Miroslav Nachev Sent: woensdag 23 juni 2004 16:43 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Generator for ISDN (PRI/BRI) Hi, I am looking for Call Generator for PRI ISDN and BRI ISDN signals. From where I can found some cheap or 2nd hand call generator (tester/analyzer)? Maybe PCI based will be cheaper than standalone solution. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: The content of this e-mail message does not constitute a commitment of DKMA bvba This e-mail and any attachments thereto may contain information which is confidential and/or protected by intellectual property rights and are intended for the intended recipient only. Any use of the information contained herein ( including, but not limited to, total or partial reproduction, communication or distribution in any form ) by persons other than the designated recipient(s) is prohibited.If an addressing or transmission error has misdirected this e-mail, please notify the author, either by telephone or by e-mail and delete the material from any computer. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Outgoing CLI
> User must provide - TON = "national or international Add pridialplan=national before just above your "channel =>..." line in zapata.conf to set TON to national for outgoing calls. NPI will be always be set to E.164 afaik. Now set callerid to 845 or 870 without the leading 0. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with incominglimit and outgoinglimit
Hi, I seem to have a problem with chanisavail and the call limits on sip phones(incoming and outgoing) The problem seems to be that chanisavail when trying create to create channels and hanging them up afterwards screw up the current usage limit on the phones. Example with chanisavail: Phone A calls voicemail (usage now 1) Phone B tries to call Phone A and uses ChanIsAvail in the dialplan. ChanIsAvail creates a channel thru ast_request. But when hanging up the created channel the limit is decremented. The call is established Example without chanisavail: Phone A calls voicemail (usage now 1) Phone B tries to call Phone A and just use dial in the dialplan. The call is not established Has any one else had this error?? Kind Regards Claus --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.708 / Virus Database: 464 - Release Date: 18-06-2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Outgoing CLI
Simon wrote: Hello I have contacted my line provider who is saying that in order to get my 0845 or 0870 number to id as the incoming number on a landline that i may call i need the following. User must provide - NPI set to E.163/E.164 User must provide - TON = "national or international I have had a good search around and can't seem to find a good answer to this. Does anyone have any idea where i would set this and how. Best Regards Simon * makes bundles out of NPI and TON, and calls them the pridialplan. I find this annoying, but then some people find me annoying, Look in zapata.conf Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Eicon Diva 2.0 PCI ISDN Card
On Wed, 23 Jun 2004 06:30:21 -0600, Michael Welter <[EMAIL PROTECTED]> wrote: > Are you using it in North America (NI1 protocol)? I'm using it in Australia with the Euro ISDN (ETSI ISDN) protocol. -Shaun > Thanks, > Mike > > -- > Michael Welter > Introspect Telephony Corp. > Denver, Colorado > +1 303 674 2575 > [EMAIL PROTECTED] > www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] AstriCon Registration Opens Next Monday, June 28th
Hello Brian, Wednesday, June 23, 2004, 4:04:30 AM, you wrote: BC> Hi Steve BC> Olle mailed me a couple of week ago about doing a presentation at this BC> conference, on "dialplan tips and tricks." BC> I'm working on one right now. Just wanted to let you know. BC> Thanks. Really looking forward to the conference. BC> B. BC> ___ BC> Asterisk-Users mailing list BC> [EMAIL PROTECTED] BC> http://lists.digium.com/mailman/listinfo/asterisk-users BC> To UNSUBSCRIBE or update options visit: BC>http://lists.digium.com/mailman/listinfo/asterisk-users I know this is kind of off topic on the list.. but who's attending this.. I sure as hell will be at this.. Hope to meet others from the list and the IRC channel. -- Best regards, Frankie ([EMAIL PROTECTED]) mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [1-9] as E? (was: [Asterisk-Users] Busy message)
I don't think it's documented, but Z specifies 1-9
exten => _Z,1,NoOp(.call for .${EXTEN})
On Wed, 2004-06-23 at 03:52, Roger Schreiter wrote:
> Keith Waters schrieb:
> ...
> > I configured:
> >
> > exten => _[123456789],1,NoOp(.call for .${EXTEN})
>
> I consider a short key for [1-9] at least
> as useful as N for [2-9], maybe even more useful.
>
> For my internal purposes I'm using E for [1-9].
>
> Am I the only one, who is missing something short
> for [1-9]?
>
>
> Roger.
>
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--
Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."
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Re[2]: [Asterisk-Users] Call Generator for ISDN (PRI/BRI)
Dear Michael, MD> why not use asterisk with QaudBRI and/or E100P ? Because I have to be sure that I am Euro ISDN compliant. My target is Bulgaria which is in Europe. Best Regards, Miroslav Nachev -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Miroslav Nachev Sent: woensdag 23 juni 2004 16:43 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Call Generator for ISDN (PRI/BRI) Hi, I am looking for Call Generator for PRI ISDN and BRI ISDN signals. From where I can found some cheap or 2nd hand call generator (tester/analyzer)? Maybe PCI based will be cheaper than standalone solution. Best Regards, Miroslav Nachev COSMOS Software Enterprises, Ltd. Tel:(+359-2) 983-32-62 Mobile: (+359-88) 897-31-95 E-Mail: [EMAIL PROTECTED] [EMAIL PROTECTED] http://www.space-comm.com Post address: P. O. Box 941, 1000 Sofia, Bulgaria Office address: ap. 9, fl. 4, "11 August" str., No. 43, 1202 Sofia, Bulgaria ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Two SIP servers communicating without IAX
On Wed, 2004-06-23 at 05:01, Philipp von Klitzing wrote: --snip-- > > If I call from an Asterisk phone to an MX250 phone, I can talk FROM the > > MX250 phone TO the Asterisk phone, but not the other way around. In > > other words, the Asterisk phone will hear everything that is said from > > the MX250 phone, but if I say anything on the Asterisk phone the MX250 > > phone never gets it. > > Look at your codec configuration (disallow=all followed byallow= > statements). Do a "SIP DEBUG" on the Asterisk CLI to learn more about the > codec negotiation before/during a call. Finally check your phones VAD/ > silence suppression settings, make sure those are turned off. That took care of it. I added a disallow=all followed by an allow=ulaw and it worked fine. What's strange is that I had manually set the codec on the phone earlier to use ulaw, but it didn't appear to want to listen. Forcing it via sip.conf took care of the problem. Thanks a lot for the help. -- Alex Malinovich Golden Technologies, Inc. (219) 462-7200 x 216 http://www.golden-tech.com signature.asc Description: This is a digitally signed message part
Re: [Asterisk-Users] Call generator
Andrew Kohlsmith wrote: On Wednesday 23 June 2004 04:46, GIBERT Frédéric wrote: Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Are there any IAX2 call generators? Regards, You can use asterisk to generate the calls, just put a few hundred files in asterisk's spool directory. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk user/host registration
It seems your sip.conf is correct. Maybe the problem is on your client
side. You can use SJphone to do a test. Look at this websit
http://www.voip-info.org/wiki-Asterisk+phone+sjphone
It tell you how to set the configuration file.
Mandar Pise wrote:
Hi Folks,
I am newbie to asterisk. Recentely I have installed asterisk on Linux
Fedora 2 box. After reading some document, I tried to configure the server.
When I connect to our server, SIP user-agent shows that I am logged in.
But it doesn't show my system(client) IP when I issue command at astrisk
CLI. The O/P is as below.
*CLI> sip show peers
Name/usernameHost Mask Port Status
2001/2001(Unspecified) (D) 255.255.255.255 0UNKNOWN
2000/2000(Unspecified) (D) 255.255.255.255 0UNKNOWN
I am pasting *sip.conf* & *extension.conf*
**
*/sip.conf/*
[general]
port = 5060
bindaddr = 0.0.0.0
context = INVALID
;Autocreatepeer= yes
[2000]
type=friend
username=2000
secret=2000
host=dynamic
context=from-sip
mailbox=100
canreinvite=no
qualify=300
nat=no
[2001]
type=friend
username=2001
secret=2001
host=dynamic
context=from-sip
mailbox=100
canreinvite=no
qualify=300
nat=no
*/extensions.conf/*
[globals]
CONSOLE=Console/dsp
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest
;IAXINFO=myuser:mypass
TRUNK=Zap/g2
TRUNKMSD=1
;TRUNK=IAX2/user:[EMAIL PROTECTED]
[bogon-calls]
exten => _.,1,Congestion
[from-sip]
exten => 2000,1,Dial(SIP/2000,20)
exten => 2000,2,Voicemail(u${EXTEN})
exten => 2000,3,Hangup
exten => 2000,102,Voicemail(b${EXTEN})
exten => 2000,103,Hangup
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(u${EXTEN})
exten => 2001,3,Hangup
exten => 2001,102,Voicemail(b${EXTEN})
exten => 2001,103,Hangup
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
Immediate help is highly appreciated.
Thank you,
Mandar
*Yahoo! India Matrimony*
.
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Re: [Asterisk-Users] Call generator
Andrew Kohlsmith wrote: On Wednesday 23 June 2004 04:46, GIBERT Frιdιric wrote: Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. Are there any IAX2 call generators? You can use Asterisk as a generic (H.323/SIP/IAX/ZAP/ISDN) call generator. I have written a small shell script to make things easier which you can find it below: http://www.inaccessnetworks.com/projects/asterisk-oh323/utils Regards, Andrew Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P Noise
Title: Message Hi I have 2 x X100P on UK BT, both have been working fine for a while, but now I have started to get a beeping sound my end every 8/10 sec, and break-up in the voice call inbound/outbound. Any ideas??? Rgds
Re: [Asterisk-Users] trying to set an internal ivr
Grandsteams can be configured to dial an extension as soon as you pick it up. Great for doorphones and your application. I forget the exact vebiage but I know its in there. Oh here it is, Offhook Auto-Dial: - Original Message - From: "PAZ" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Tuesday, June 22, 2004 5:08 PM Subject: RE: [Asterisk-Users] trying to set an internal ivr That works OK for my Zaptel Hardware, because Asterisk detects when I pick Up the phone. For my Grandstream BudgeTone 100 not so much, because there is no traffic between phone and asterisk until I call some number. Thank you anyway. On Fri, 18 Jun 2004, Jay Milk wrote: You're basically looking for hotline functionality. I'm using Sipuras for my FXS ports, and they can be configured to dial a phone number upon pickup. I played with that before, and the call was established so quickly that I had to add a "Wait" instruction in there so the receiver could make it to the ear :) If you're using zap channels for FXS, you could do something line (in zapata.conf): context => instantpickup immediate => yes channel => 10-60 And then in extensions.conf: [instantpickup] exten => s,1,Answer exten => s,1,Wait(1) exten => VoiceMailMain() (or whereever you want to go from here) > -Original Message- > From: Greg Hill [mailto:[EMAIL PROTECTED] > Sent: Thursday, June 17, 2004 6:53 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] trying to set an internal ivr > > > On Thu, 17 Jun 2004, PAZ wrote: > > I'm trying to implement an IVR for internal use for the > enterprise I > > work for, but the goal I'm trying to reach is that the main menu of > > this IVR present itself to the user after 5 seconds he picks up his > > extension (and only if the user doesn't press any key, off > course). I > > imagine the solution (if exists) maybe relies in timeout > properties, > > but I can't see it. Any suggestions for my extension.conf file ?. > > once a connection is established to the server, you could > have exten => t,1,Goto(yourIVR) or similar. But that depends > on the phone making a connection to * as soon as the handset > is lifted. The xten softphone (currently my only SIP device > :( ) doesn't actually connect to the SIP server until you > push the call button. I guess that if a hardware phone > actually connects immediately, then you could probably make > the timeout extension work. Maybe you can adjust the timeout > length with exten => s,1,DigitTimeout(5) or something > similar. ResponseTimeout might work for that too.. I'm just > guessing, though.. I had an idea but no hardware to test on. > > Greg > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roberto Paz "The box said, 'Requires Windows 95 or better', so i installed Linux" TKK 5 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] Problem with incominglimit and outgoinglimit
Hi Philipp, Thanks for the reply.. I was just wondering how to make this at a per user basis. Could you provide me with an example. The Wiki doesn't provide that much info when it comes to pratical use. Kind Regards cf "The box said 'Requires Windows 95, NT, or better,' so I installed Linux." This message is for the designated recipient only and may contain privileged or confidential information. If you have received it in error, please notify the sender immediately and delete the original. Any other use of the email by you is prohibited. - Original Message - From: "Philipp von Klitzing" <[EMAIL PROTECTED]> To: "Claus Futtrup" <[EMAIL PROTECTED]> Sent: Wednesday, June 23, 2004 4:08 PM Subject: Re: [Asterisk-Users] Problem with incominglimit and outgoinglimit > Use SetGroup() and GetGroupCount() and CheckGroup() insetad of > incominglimit > > Cheers, Philipp > > > --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.708 / Virus Database: 464 - Release Date: 18-06-2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk -- PBX Do Not Disturb
Interestingly enough, I just tried this on a SIP extension and it worked fine...although I did not see a report on the command line as expected. I used a Polycom IP600 to call an extension which is an analog phone plugged into one side of a Spiura SPA-2000. Keying *78 into the phone one the SPA generated a pulsing dialtone for 3 seconds then the dialtone returned to normal. I hungup then dialed that extension again and was routed directly to voicemail. I do have an X100p in my * server and zaptel is loaded. However, I don't use it as my fxo anymore, I just left it there a timing source for the conferences. Michael On Wed, 23 Jun 2004 10:08:46 +0200, Stephen Rosebush wrote: >Yes this is where I wanted to use it the most, I use Asterisk as a >software-only based PBX using SIP and IAX, I have an ATA device on my >network connected to the PBX but it nor the actual analog phone has a >DND function. I am hoping to implement an Asterisk-side based DND >somehow but was wondering where to go from there. > >Thanks > >Steve > >Dean Collins wrote: > >>That could explain why it wouldn't work on any of my sip extensions I >>tried it on this morning when I first read about it and thought cool the >>things you learn. >> >>Is there anyway to make it work on Sip extensions? >> >>Cheers, >>Dean >> >> >> >>-Original Message- >>From: [EMAIL PROTECTED] >>[mailto:[EMAIL PROTECTED] On Behalf Of Aaron J. >>Angel >>Sent: Wednesday, 23 June 2004 10:33 AM >>To: [EMAIL PROTECTED] >>Subject: RE: [Asterisk-Users] Asterisk -- PBX Do Not Disturb >> >>john lawler [EMAIL PROTECTED] wrote: >> >> >>>You don't have to put this in the dialplan. It's one of >>>those low-level functions in Asterisk (possibly controlled at >>>the driver level-- I'm not sure about that). If you have an >>>extension defined, pick up the handset and dial '*78', you >>>should see on the Asterisk CLI: >>> >>> Enabled DND on channel >>> >>> >> >>That assumes your using zaptel, no? This doesn't exist for other >>channels >>as it's not built in to the channel driver for anything but zaptel. >> >>___ >>Asterisk-Users mailing list >>[EMAIL PROTECTED] >>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 >>[EMAIL PROTECTED] >>http://lists.digium.com/mailman/listinfo/asterisk-users >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> >> > > >-- >Stephen Rosebush, >[EMAIL PROTECTED] >http://www.desynched.org/ > >// PSTN >USA: 1-248-724-4452 x201 >Netherlands: +31-(0)20-6598858 x63420 x201 > >// IP Phone >FWD: 63420 x201 >IAXTEL:1-700-356-6191 x201 >SIP: sip:[EMAIL PROTECTED] > >___ >Asterisk-Users mailing list >[EMAIL PROTECTED] >http://lists.digium.com/mailman/listinfo/asterisk-users >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 "One day in your life shouldn't be a problem" - 54-40 from One Day ** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Outgoing CLI
Hi Simon, the bad news is that you cannot change pridialplan on a per call basis (or if you can I don't know how it's done). So even if setting pridialplan=national works for your 0845 presentation number calls it's unlikely to work for ordinary calls that present your geographic PSTN (0207) DDI range. I have just setup a test using a PRI into your carrier (using pridialplan=national instead of local) and here's the result from a normal call to a UK geographic PSTN number with a geographic PSTN DDI range CLI being presented: -- Making new call for cr 32775 > Protocol Discriminator: Q.931 (8) len=44 > Call Ref: len= 2 (reference 7/0x7) (Originator) > Message type: SETUP (5) > Bearer Capability (len= 3) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) > Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) > Ext: 1 User information layer 1: A-Law (35) > Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 >ChanSel: Reserved > Ext: 1 Coding: 0 Number Specified Channel Type: 3 > Ext: 1 Channel: 1 ] > Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) > Called Number (len=14) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0161XXX' ] > Sending Complete (len= 0) < Protocol Discriminator: Q.931 (8) len=9 < Call Ref: len= 2 (reference 32775/0x8007) (Terminator) < Message type: RELEASE COMPLETE (90) < Cause (len= 2) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) < Ext: 1 Cause: Invalid number format (28), class = Normal Event (1) ] The good news is that the NPI is exactly as you need (E.164) but as you can see from the line above, when pridialplan=local is NOT used, the call is rejected with release cause 28 (Invalid number format). Some switches accept a National TON and others don't, it varies from carrier to carrier. The strange part is that the geographic DDI number that I used for the Calling Number does not appear on the trace above; does anyone know if this is a bug with * ? NB. I tried the Calling Number both with and without a leading 0. Maybe your datafill for this PRI is now different (after your request for an 08XX presentation number to the carrier) and it will accept "TON: National Number" instead of "TON: Subscriber Number" for all calls. Please drop a note to the list and let us know what happens when you make the changes. Use 'pri debug span 1' to gather some trace information if you need to post more detail back with your next e-mail. HTH Darren -- Comgate Telco>Internetmailto:[EMAIL PROTECTED] Behalf Of Thilo Salmon Sent: 23 June 2004 14:49 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Outgoing CLI > User must provide - TON = "national or international Add pridialplan=national before just above your "channel =>..." line in zapata.conf to set TON to national for outgoing calls. NPI will be always be set to E.164 afaik. Now set callerid to 845 or 870 without the leading 0. Thilo ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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] FXO impedance matching
Robert Hajime Lanning wrote: Echo echo ech ech ec ec e e . . :) What's the importance of the impedance matching in a FXO interface ? If impedance matching is that important, then how is it accomplished? I'm fairly sure our X101P is not impedance matching properly. I've never not had echo, and I've followed all the procedures in the wiki and from help that I've gotten from IRC. The only way to hear anything is by turning up the gain in /etc/asterisk/zapata.conf, and I've been told this is a big no-no, as it increases echo problems. But without it, the line is so low that it's useless. However, if you plug a handset directly into the line coming from the CO, it's at a decent volume level, etc. No echo, of course. -- Jason A. Pattie [EMAIL PROTECTED] Xperience, Inc. (http://www.xperienceinc.com) -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. MailScanner thanks transtec Computers for their support. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Call generator
Hello Adam, I'm interested by this solution, but can you please give me more info because I don't know how to generate calls with asterisk and the spool directory. How don't know wich files do I need to use. Thanks. Fred Date: Thu, 24 Jun 2004 00:02:13 +1000 From: Adam Hart <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Call generator Reply-To: [EMAIL PROTECTED] Andrew Kohlsmith wrote: > On Wednesday 23 June 2004 04:46, GIBERT Frédéric wrote: > >>Has someone know a good call generator for asterisk including SIP protocol >>(freeware if possible)? >>I need to stress a plateform and I don't find any. > > > Are there any IAX2 call generators? > > Regards, You can use asterisk to generate the calls, just put a few hundred files in asterisk's spool directory. -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-addons compilation error
[EMAIL PROTECTED] wrote: > Hello, > You need to have your versions of asterisk and asterisk-addons > match, if you are getting a lock error. Download both from the cvs > and compile. This should fix your error. > - Original Message - > From: "Hekuran Doli" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Tuesday, June 22, 2004 3:19 PM > Subject: Re: [Asterisk-Users] asterisk-addons compilation error > > >> have you install mysql-devel? >> >> >> I am getting the following error as of today after updating both asterisk and asterisk-addons. These are both under /usr/src. ... >>> cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class >>> >>> cdr_addon_mysql.c:108: error: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:420: error: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 >>> >>> >>> Hi >>> >>> I'm having the same problem. >>> I've tried to solve the problem using the all the tips&tricks from >>> the mailingslists - But with no luck :-( >>> >>> System: RH9, mysql 4.0.20 >>> >>> NRB >>> >>> >>> >>> ___ >>> Asterisk-Users mailing list >>> [EMAIL PROTECTED] >>> 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 >> [EMAIL PROTECTED] >> 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 > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users I am also working on getting this installed. I cvs asterisk & asterisk-addons. Make install asterisk, then make asterisk-addons. I still receive the mysql error: make: *** [cdr_addon_mysql.o] Error 1 I have mysql and devel installed package MySQL-3.23.58-1 is already installed package MySQL-devel-3.23.58-1 is already installed When I did the make install from the asterisk folder, it added the CFLAGS+=-I../asterisk/include into the Makefile. Im running RH Fedora Core 1. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] clarent hardware
Just wondering if anyone has had any luck with the Clarent CPGs ( uses MGCP ). I have a couple CPG 201s laying around that I am trying to get working but am having difficulties. They successfully register with the asterisk box, but when I lift the handset of the phone plugged into any of its ports, there is no dial tone, I hear no DTMF tones when I press keys, etc. But I can make the phone ring by using a soft phone, although when I lift the handset it just keeps ringing. Any help would be great, I did see a post I found with google about someone getting CPG 101s going well, but they never posted any config files or how they did it. I also can't find any good information about using MGCP with asterisk, any links, tips, help would be appreciated. Thanks in advance! Josh Krueger <[EMAIL PROTECTED]> Urban Communications http://www.urbancom.net/
Re: [Asterisk-Users] sidetone noticeably loud on analog handsets on T100P
I have the same problem with an Adit 600 channel bank that I've been trying to resolve without any success. The sidetone is loud and very annoying. The only way to avoid it is to keep the phone mic further away from your mouth. Interesting; I'm using a T100P connected to an Adit600 and I have no sidetone issues. I've used scenarios involving FXS ports connecting to trunk lines on a Norstart MICS KSU as well as regular old phones connected to the FXS ports. The Adit600 does have its own tx/rxgain settings; have you played with those? I have played with the tx/rxgain settings on the Adit600. They don't seem to have any effect on the sidetone issue. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help needed with read()
Hi,
Greatly appreciate
if some one help me with the application read().
asterisk*CLI>
show application readasterisk*CLI> -= Info about application
'Read' =-
[Synopsis]:Read
a variable
[Description]:
Read(variable[|filename]): Reads a '#' terminated string of digits
fromthe user, optionally playing a given filename first. Returns -1 on
hangup orerror and 0 otherwise.
I need to
collect a variable length digit string terminated by # and then pass those
digits to an agi script. I can do this as follows (when I know the
length of the string).
exten =>
s,1,BackGround(please-enter-the-fourdigit-pin)exten =>
s,2,DigitTimeout,5exten => s,3,ResponseTimeout,10exten => _,1,
agi, agiscript.agi
I would like to use Read() here,
like
exten =>
s,1,BackGround(please-enter-the-pin)exten =>
s,2,Background(followed-by-pound)exten => s,3,DigitTimeout,5exten
=> s,4,ResponseTimeout,10exten => s,5,Read(${EXTEN})exten => ???/
This will read
the digits but I do not know how to proceed after the
reading.
Thanks a
bunch
Sathya
Re: [Asterisk-Users] X100P Noise
On Wednesday 23 June 2004 08:17, Lee Norvall wrote: > I have 2 x X100P on UK BT, both have been working fine for a while, but > now I have started to get a beeping sound my end every 8/10 sec, and > break-up in the voice call inbound/outbound. > Any ideas??? Sounds like your x100p cards are sharing interrupts with another device. Check /proc/interrupts. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] FXO impedance matching
Jason A. Pattie wrote: Robert Hajime Lanning wrote: Echo echo ech ech ec ec e e . . :) What's the importance of the impedance matching in a FXO interface ? If impedance matching is that important, then how is it accomplished? I'm fairly sure our X101P is not impedance matching properly. I've never not had echo, and I've followed all the procedures in the wiki and from help that I've gotten from IRC. The only way to hear anything is by turning up the gain in /etc/asterisk/zapata.conf, and I've been told this is a big no-no, as it increases echo problems. But without it, the line is so low that it's useless. However, if you plug a handset directly into the line coming from the CO, it's at a decent volume level, etc. No echo, of course. My experience is with excessive buzz and hum on the line. When I plug a vintage Western Electric phone into the line, there is no buzz or hum because the phone has its own impedance matching circuitry. When I plug my AT&T 954 set into the line, I hear a lot of hum. I'm told the X100P does not have impedance matching. Rich Adamson is the fellow to talk with about impedance. Apparently the hum on my lines is caused by a partial ground on either the tip or ring (or both) wire. If both leads have the same resistance to ground (matched) then there is no hum. I don't experience echo with the buzz and hum. I've been told that echo is caused when the circuit goes from four wire to two wire. I'm trying to locate a schematic of an impedance matching circuit so I can breadboard a device but haven't found one so far. I anyone has experience with this I invite him to reply. Mike -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference calling
Hello all, I have an ISDN card lspci: 07:06.0 Network controller: Eicon Technology Corporation DIVA Server PRI-30M 2.0 which has 8 channels active. I am wondering if a:, this card is supported/can be made to work with Asterisk, and b:, if it is possible to make Asterisk initiate 2 outgoing voice calls, which it conferences together. Unfortunately I only have binary drivers for the Eicon card ( gah ). I am not a PBX expert, but am pretty handy with Linux, and can get my hands dirty with C if necessary. -- Random russian saying: If you will not hear reason, she will surely rap your knuckles. jabber: [EMAIL PROTECTED] pgp: http://gk.umtstrial.co.uk/~calum/keys.php Linux 2.6.5-gentoo 16:42:55 up 15 days, 5:11, 1 user, load average: 0.24, 0.33, 0.27 ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Failover of IAX or Spillover as the case may be
On Wed, 23 Jun 2004, Andrew Kohlsmith wrote: > On Wednesday 23 June 2004 06:01, Philipp von Klitzing wrote: > > Secondly you could also use the exit codes of Dial() for failover action, > > but better prevent the necessity for that with the help of SetGroup(). > > IAX2 does not return the disconnect cause from the far end; this is necessary > in cases where the provider accepts connections but is having PSTN > connectivity issues of their own; Dial() jumps to n+101 but you don't know if > the number you were calling was actually busy or if there were problems, so > your only choice is to try dialing them again with another provider. Why don't you put that on bugs.digium.com - perhaps someone will implement it. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as a SIP UA and voicemail with SER not working anymore
Hi,
I downloaded the stable branch of
asterisk a couple of month ago, and I'm using it as a SIP UA voicemail
server with SER, and my setup works fine.
I do have a list of phones defined in
voicemail.conf, in the sip.conf file I only have the
setup of asterisk as a peer registering to ser. The extensions.conf
file contain the extensions that link to the voicemail
application. This setup is working as expected so far.
I downloaded the latest cvs yesterday, and with the same config
files nothing work anymore: asterisk denies a caller to leave a voicemail with
Forbidden-403 code as if the caller needed to authenticate with asterisk,
instead of just establishing an rtp session with it
and just act as a SIP UA.
Has anything changed recently in regards
to have asterisk acting not as a sip server but just as a sip ua ?
This
is a snippet of what I have in the extensions.conf
file:
exten
=> _1959XX,1,Answer
exten
=> _1959XX,2,Wait(1)
exten
=> _1959XX,3,Voicemail(u${EXTEN})
exten
=> _1959XX,4,Wait(1)
exten
=> _1959XX,5,Hangup
Thanks.
Samy.
RE: [Asterisk-Users] Call generator
GIBERT Frédéric wrote: > Hello Adam, > > I'm interested by this solution, but can you please give me more info > because I don't know how to generate calls with asterisk and the > spool directory. How don't know wich files do I need to use. > > Thanks. > Fred Look in your ./asterisk directory, you'll have a sample.call file. Read it and it will show you what to do. Nik > > > > You can use asterisk to generate the calls, just put a few hundred > files in asterisk's spool directory. > > -Adam > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users l ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] FXO impedance matching
Michael Welter wrote: > Jason A. Pattie wrote: >> Robert Hajime Lanning wrote: >> >>> Echo echo ech ech ec ec e e . . >>> >>> :) >>> >>> >>> What's the importance of the impedance matching in a FXO interface ? >> >> >> > My experience is with excessive buzz and hum on the line. When I > plug a vintage Western Electric phone into the line, there is no buzz > or hum because the phone has its own impedance matching circuitry. > When I plug my AT&T 954 set into the line, I hear a lot of hum. I'm > told the X100P does not have impedance matching. > > Rich Adamson is the fellow to talk with about impedance. Apparently > the hum on my lines is caused by a partial ground on either the tip > or ring (or both) wire. If both leads have the same resistance to > ground (matched) then there is no hum. > > I don't experience echo with the buzz and hum. I've been told that > echo is caused when the circuit goes from four wire to two wire. > > I'm trying to locate a schematic of an impedance matching circuit so I > can breadboard a device but haven't found one so far. I anyone has > experience with this I invite him to reply. > > Mike If you KNOW the impedances of the two lines, a simple impedance matching transformer available at any electronics distributer (Mouser, Digi-Key, etc) carries many differdnt types, that are just for this purpose. Nik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk + appradius & freeradius
Here is the jist: Freeradius is up running and functional using SIP Express radius how to. My asterisk box has app radius installed. Is there any documents on how-to link asterisk to freeradius? documentation is lacking on app radius, at least not as detailed as I need. Anyone know of a how-to or a link that covers this ? Thanks. __ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] help needed with read()
On Wed, 2004-06-23 at 10:12, Sathya wrote:
> asterisk*CLI>
> -= Info about application 'Read' =-
>
> [Synopsis]:
> Read a variable
>
> [Description]:
> Read(variable[|filename]): Reads a '#' terminated string of digits from
> the user, optionally playing a given filename first. Returns -1 on hangup or
> error and 0 otherwise.
>
> I need to collect a variable length digit string terminated by # and
> then pass those digits to an agi script. I can do this as follows
> (when I know the length of the string).
>
> exten => s,1,BackGround(please-enter-the-fourdigit-pin)
> exten => s,2,DigitTimeout,5
> exten => s,3,ResponseTimeout,10
> exten => _,1, agi, agiscript.agi
> I would like to use Read() here, like
>
> exten => s,1,BackGround(please-enter-the-pin)
> exten => s,2,Background(followed-by-pound)
> exten => s,3,DigitTimeout,5
> exten => s,4,ResponseTimeout,10
> exten => s,5,Read(${EXTEN})
> exten => ???/
First, if you are going to be going to AGI, why not just ask for the
digits inside of AGI? It seems like it is more appropriate that way.
Read is for when you can accomplish what you want without jumping to
AGI.
Also, be wary that Background will allow a user to interupt the prompt
with a digit press. This is the normal expected behavior. You would want
to use playback.
--
Steven Critchfield <[EMAIL PROTECTED]>
___
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[Asterisk-Users] Voicemail Password Changes Lost on Asterisk Restart
Ok I have googled and googled and combed through the wiki for an answer to this and have come up empty. What I'm finding is that when a user changes their VM password, it is saved somewhere like maybe the CSV database or something because when you log in, the new password works fine, but it's not saving to voicemail.conf. So new passwords are lost when asterisk is restarted and people become annoyed... I get the feeling from reading the wikis about VoiceMailMain() that you really need to use cdr_ to keep passwords on restart but it's not said out right. Is this true? Or can asterisk save to the voicemail.conf and I just need to update my CVS? -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] X100P Noise
Hi I remembered that I had disabled USB 2.0 on the motherboard last week. I rebooted the server, enabled USB 2.0 and all seems a lot better. I guess this may have cleared any conflicts.. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ryan Courtnage Sent: 23 June 2004 16:13 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] X100P Noise On Wednesday 23 June 2004 08:17, Lee Norvall wrote: > I have 2 x X100P on UK BT, both have been working fine for a while, > but now I have started to get a beeping sound my end every 8/10 sec, > and break-up in the voice call inbound/outbound. Any ideas??? Sounds like your x100p cards are sharing interrupts with another device. Check /proc/interrupts. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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 [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk + appradius & freeradius
> [EMAIL PROTECTED] wrote: > Here is the jist: Freeradius is up running and > functional using SIP Express radius how to. My > asterisk box has app radius installed. Is there > any documents on how-to link asterisk to freeradius? > documentation is lacking on app radius, at least > not as detailed as I need. Anyone know of a how-to or > a link that covers this ? > > Thanks. > Have you looked at http://appradius.minitelecom.org/ Doesnt look very detailed tho. Harold ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Conference application !
Hi, I´m just compiling the app_conference but I can´t locate the common.h file , those it´s requered. Someone help me to locate Common.h file Thanks
Re: [Asterisk-Users] Voicemail Password Changes Lost on Asterisk Restart
On Wed, 23 Jun 2004 09:22:40 -0700, Chris Shaw <[EMAIL PROTECTED]> wrote: > Is this true? Or can asterisk save to the voicemail.conf and I just need to > update my CVS? > >-Chris I don't know about the rest, but my Asterisk install certainly updates voicemail.conf when changing passwords. I just tested then to make sure and it updated. Using stable CVS-06/07/04-16:18:54 -Shaun ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Voicemail Password Changes Lost on Asterisk Restart
Chris Shaw wrote: > Ok I have googled and googled and combed through the wiki for an > answer to this and have come up empty. What I'm finding is that when > a user changes their VM password, it is saved somewhere like maybe > the CSV database or something because when you log in, the new > password works fine, but it's not saving to voicemail.conf. So new > passwords are lost when asterisk is restarted and people become > annoyed... > > I get the feeling from reading the wikis about VoiceMailMain() that > you really need to use cdr_ to > keep passwords on restart but it's not said out right. > > Is this true? Or can asterisk save to the voicemail.conf and I just > need to update my CVS? > > -Chris > Nah, it's because the file isn't writeable by whover asterisk runs as, I'm pretty sure. Works fine running as root on my machine. ___ Asterisk-Users mailing list [EMAIL PROTECTED] 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] Skype 4 Linux
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wednesday 23 June 2004 03:56 am, Stefan de Konink wrote: > Hi All, > > Since 21 june skype is available to be used on Linux, with a static > binary, which includes QT, of 8 meg its big. > > http://www.skype.com/help_linux_faq.html I downloaded it the other day to play with. On a virtually inactive DSL to Cable connection the sound quality was HORRID! Has anyone had better experience with skype? - -- Mike gpg key: http://diehlnet.com/~mdiehl/mdiehl.asc 83AD D927 758D 4BFC A800 0277 4B26 75A4 F0D1 C7EB -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFA2bU2SyZ1pPDRx+sRAnEpAKC7N02NLx8O1Znq/mww61tF4SB+4QCfWbIz TjFf+cO+wtqkMrH+WNa2TTY= =SW+g -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
