[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Fri, Feb 24, 2006 at 10:43:31AM +0100, Armin Schindler wrote:
On Fri, 24 Feb 2006, Ralf Schlatterbeck wrote:
On Thu, Feb 23, 2006 at 02:45:25PM +0100, Armin Schindler wrote:
Interesting is, that I receive an INFO_IND *before* the CONNECT_IND.
This looks like an interesting variation of Austrian ISDN to me.
Maybe it is a variation of the ISDN line, but the driver should fix that.
Sending INFO_IND with a call-reference (PLCI) which is assigned by
CONNECT_IND later, is just an error of the isdn driver.
You mean, the capi part of misdn? Should I report a bug against mISDN?
Yes. Maybe it is already fixed in mISDN and you have an older version?
OK, I've reported a bug to mISDN. With the patch from the Karsten Keil
in the mantis tracker:
issue: https://www.isdn4linux.de/mantis/view.php?id=40
I'm now gettig connect_ind and info_ind in the correct order (asterisk
capi debug + verbose 15 log attached). The call still does not proceed
in asterisk (the disconnect comes whe I unplug the ISDN-Line after
several minutes). I've also attached my capi.conf and relevant portions
of the dialplan. The version of mISDN is mqueue from yesterday with the
mantis-tracker patch applied.
Any ideas what I should try next?
Thanks, Ralf
--
Ralf Schlatterbeck
email: [EMAIL PROTECTED] FAX: +43/2243/26465/23
Verbosity is at least 15
CONNECT_IND ID=001 #0x0007 LEN=0046
Controller/PLCI/NCCI= 0x101
CIPValue= 0x10
CalledPartyNumber = 8111
CallingPartyNumber = 21 83650621
CalledPartySubaddress = default
CallingPartySubaddress = default
BC = 80 90 a3
LLC = default
HLC = 91 81
AdditionalInfo
BChannelinformation= default
Keypadfacility = default
Useruserdata = default
Facilitydataarray = default
-- CONNECT_IND (PLCI=0x101,DID=11,CID=650621,CIP=0x10,CONTROLLER=0x1)
ISDN1: msn='*' DNID='11' DID
== ISDN1: Incoming call '0650621' - '11'
INFO_IND ID=001 #0x0008 LEN=0016
Controller/PLCI/NCCI= 0x101
InfoNumber = 0x18
InfoElement = 89
INFO_RESP ID=001 #0x0008 LEN=0012
Controller/PLCI/NCCI= 0x101
-- ISDN1: info element CHANNEL IDENTIFICATION 89
DISCONNECT_IND ID=001 #0x0009 LEN=0014
Controller/PLCI/NCCI= 0x101
Reason = 0x3301
DISCONNECT_RESP ID=001 #0x0009 LEN=0012
Controller/PLCI/NCCI= 0x101
CAPI INFO 0x3301: Protocol error layer 1 (broken line or B-channel
removed by signalling protocol)
-- ISDN1: DISCONNECT_IND on incoming without pbx, doing hangup.
== ISDN1: CAPI Hangingup
== ISDN1: Interface cleanup PLCI=0x101
-- Starting simple switch on 'Zap/2-1'
-- Hungup 'Zap/2-1'
fox*CLI
;
; The General category is for certain variables.
;
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=Capi/contr1
UNTEN=Zap/4
OBEN=Zap/3
BUERO=Zap/2
OFAX=Zap/1
;OFAX=Zap/5
;SERVERRAUM=Zap/6
SERVERRAUM=Zap/2 ; Alias for BUERO
SIPPHONE=SIP/ralf
[extern]
exten = 0,1,Noop()
exten = 0,2,Noop(0)
exten = 0,3,Dial(${UNTEN})
exten = 0,4,Busy()
exten = 0,104,Busy()
exten = 11,1,Noop()
exten = 11,2,Noop(11)
exten = 11,3,Dial(${UNTEN})
exten = 11,4,Busy()
exten = 11,104,Busy()
exten = 12,1,Noop()
exten = 12,2,Noop(12)
exten = 12,3,Dial(${OBEN})
exten = 12,4,Busy()
exten = 12,104,Busy()
exten = 13,1,Noop()
exten = 13,2,Noop(13)
exten = 13,3,Playtones(busy)
exten = 13,4,Busy()
exten = 13,104,Busy()
exten = 14,1,Noop()
exten = 14,2,Noop(16)
exten = 14,3,Dial(${SIPPHONE})
exten = 14,4,Busy()
exten = 14,104,Busy()
exten = 15,1,Noop()
exten = 15,2,Noop(16)
exten = 15,3,Dial(${SERVERRAUM})
exten = 15,4,Busy()
exten = 15,104,Busy()
exten = 16,1,Noop()
exten = 16,2,Noop(16)
; War: BUERO
exten = 16,3,Dial(${BUERO})
exten = 16,4,Busy()
exten = 16,104,Busy()
exten = 17,1,Noop()
exten = 17,2,Noop(17)
exten = 17,3,Dial(${OFAX})
exten = 17,4,Busy()
exten = 17,104,Busy()
exten = 23,1,Noop()
exten = 23,2,Noop(23)
exten = 23,3,Dial(${OFAX})
exten = 23,4,Busy()
exten = 23,104,Busy()
exten = s,1,Noop()
exten = s,2,Noop(s)
exten = s,3,Dial(${UNTEN})
exten = s,4,Busy()
exten = s,104,Busy()
exten = _X.,1,Noop()
exten = _X.,2,Noop(_X.)
exten = _X.,3,Dial(${UNTEN})
exten = _X.,4,Busy()
exten = _X.,104,Busy()
[default]
include = extern
;
; CAPI config
;
;
; general section
[general]
nationalprefix=0
internationalprefix=00
rxgain=0.8
txgain=0.8
language=de ;set default language
;ulaw=yes;set this, if you live in u-law world instead of a-law
; interface sections ...
[ISDN1] ;this example interface gets name 'ISDN1' and may be any
;name not starting with 'g' or 'contr'.
;ntmode=yes ;if isdn
[Asterisk-Users] transferring 3000 SIP calls
Hi, all, we are building a forwarding station in Japan where we would be receiving and forwarding over 3000 SIP calls at the same time. The calls will be offered to us via a carrier as SIP and we will forward the call via the same carrier as SIP. The callflow would look like this: 1. SIP call come in 2. System will authenticate the call based on the number 3. Check the billing information and if it is ok, forward the call to another number (as SIP) 4. If call is not ok, system will connect the call to IVR for an announcement and touch-tone input We are thinking about using Asterisk for this. How big of a system should it be? Can we use one linux box for this (and another for backup) or will it be something humangously huge? Thanks, Vic ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote: On Fri, Feb 24, 2006 at 10:43:31AM +0100, Armin Schindler wrote: On Fri, 24 Feb 2006, Ralf Schlatterbeck wrote: On Thu, Feb 23, 2006 at 02:45:25PM +0100, Armin Schindler wrote: Interesting is, that I receive an INFO_IND *before* the CONNECT_IND. This looks like an interesting variation of Austrian ISDN to me. Maybe it is a variation of the ISDN line, but the driver should fix that. Sending INFO_IND with a call-reference (PLCI) which is assigned by CONNECT_IND later, is just an error of the isdn driver. You mean, the capi part of misdn? Should I report a bug against mISDN? Yes. Maybe it is already fixed in mISDN and you have an older version? OK, I've reported a bug to mISDN. With the patch from the Karsten Keil in the mantis tracker: issue: https://www.isdn4linux.de/mantis/view.php?id=40 I'm now gettig connect_ind and info_ind in the correct order (asterisk capi debug + verbose 15 log attached). The call still does not proceed in asterisk (the disconnect comes whe I unplug the ISDN-Line after several minutes). I've also attached my capi.conf and relevant portions of the dialplan. The version of mISDN is mqueue from yesterday with the mantis-tracker patch applied. Any ideas what I should try next? Well, the error message from mISDN: CAPI INFO 0x3301: Protocol error layer 1 (broken line or B-channel removed by signalling protocol) seems to be very clear. The ISDN line is not working or the used protocol is wrong. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax receive using TDM400P
Ok 1 for Debian, any Fedoras Core 3 out there? fc3, and it doesn't work. If you check the archives, this has all been discussed before. The issue seems to be more oriented to the specific pci bus implementation on the motherboard. You might also want to run /usr/src/zaptel/zttest and read the archives on that as well. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi and Eicon Diva
On Tue, 28 Feb 2006, Paolo Prandini wrote: I am trying to use chan_capi with an Eicon Diva Server BRI. I installed the Eicon drivers from source including CAPU and I can use the board correcly using tty_test and minicom over /dev/ttyds01 or /dev/ttyds01. I need to insmod capi ( why? it is not written anywhere) and then If not already loaded, of course, to use CAPI to need to insmod the modules for that feature. The main module is kernelcapi, which is needed by the divacapi module. the module 'capi' provides the user-space access via /dev/capi20 capiinfo shows me all board parameters, but I have to break it otherwise capiinfo doesn't exit at all, but I don't know if this is an expected behaviour. No, capiinfo may not wait, it should exit immediatly. When I try to use chan_capi I get the message in the asterisk log that CAPI is not installed and in fact the capi20_isinstalled function in chan_capi.c returns 4109, the error code expected when capi is not installed. Why? Has anyone some experience on the matter that is willing to share? I have looked everywhere on google and the usual forums but there are no useful informations. Does /dev/capi20 has the correct permissions set for asterisk? Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] transferring 3000 SIP calls
A thread on running 5000 simultaneous cllas ran on this list recently and it did generate a lot of heat. You might want to look it up the archives - but make sure you read as many posts on it as possible because lots of different opinions formulated over time. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Vic Sent: Tuesday, February 28, 2006 10:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] transferring 3000 SIP calls Hi, all, we are building a forwarding station in Japan where we would be receiving and forwarding over 3000 SIP calls at the same time. The calls will be offered to us via a carrier as SIP and we will forward the call via the same carrier as SIP. The callflow would look like this: 1. SIP call come in 2. System will authenticate the call based on the number 3. Check the billing information and if it is ok, forward the call to another number (as SIP) 4. If call is not ok, system will connect the call to IVR for an announcement and touch-tone input We are thinking about using Asterisk for this. How big of a system should it be? Can we use one linux box for this (and another for backup) or will it be something humangously huge? Thanks, Vic ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_capi and Eicon Diva
Hello Paolo, I put together this page which has instructions on getting Asterisk working with a Diva Server card. Follow the steps for Option 0... http://www.voip-info.org/wiki-Asterisk+Eicon+Diva+CAPI+ISDN -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paolo Prandini Sent: 28 February 2006 07:28 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] chan_capi and Eicon Diva I am trying to use chan_capi with an Eicon Diva Server BRI. I installed the Eicon drivers from source including CAPU and I can use the board correcly using tty_test and minicom over /dev/ttyds01 or /dev/ttyds01. I need to insmod capi ( why? it is not written anywhere) and then capiinfo shows me all board parameters, but I have to break it otherwise capiinfo doesn't exit at all, but I don't know if this is an expected behaviour. When I try to use chan_capi I get the message in the asterisk log that CAPI is not installed and in fact the capi20_isinstalled function in chan_capi.c returns 4109, the error code expected when capi is not installed. Why? Has anyone some experience on the matter that is willing to share? I have looked everywhere on google and the usual forums but there are no useful informations. Thanks. Paolo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Tue, Feb 28, 2006 at 09:20:05AM +0100, Armin Schindler wrote: OK, I've reported a bug to mISDN. With the patch from the Karsten Keil in the mantis tracker: issue: https://www.isdn4linux.de/mantis/view.php?id=40 I'm now gettig connect_ind and info_ind in the correct order (asterisk capi debug + verbose 15 log attached). The call still does not proceed in asterisk (the disconnect comes whe I unplug the ISDN-Line after several minutes). I've also attached my capi.conf and relevant portions of the dialplan. The version of mISDN is mqueue from yesterday with the mantis-tracker patch applied. Any ideas what I should try next? Well, the error message from mISDN: CAPI INFO 0x3301: Protocol error layer 1 (broken line or B-channel removed by signalling protocol) seems to be very clear. The ISDN line is not working or the used protocol is wrong. This error message occurs when I unplug the ISDN-Line from my testing machine LNG after the call already timed out on the remote end (I'm getting No answer). So the call is seen by asterisk, the info_ind is seen by asterisk, but then nothing else happens. What does chan_capi wait for after the connect? Ralf -- Ralf Schlatterbeck email: [EMAIL PROTECTED] FAX: +43/2243/26465/23 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
R: [Asterisk-Users] Re: courtesy message calling mobile phones
Calling any unreachable mobile from Fastweb or Telecom PSTN I don't get any courtesy message. The same happens when calling an inexistent number. I'm configuring two PBX's, connected to two different phone lines, both behave this way. Perhaps there's some missing zapata parameter? Regards, _fangi_ Well, it's funny because here, now (Italy; Telecom Italia PSTN calling Wind mobile), I do get the courtesy message saying that they're moving me to voicemail, if I call myself from the office PBX to my mobile Wind number, and the cellphone is switched off. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zoom 5801 problems with *
I am having problems with a Zoom 5801 and *. It does not appear possible to route voip calls out the FXO, all voip calls get routed to the FXS no matter what. There are tons of menus in the webconfig but about 1/3 of them have no help page, and there is no documentation from Zoom on this device beyond two very brief quickstart cards. There is zero documentation on how to configure the dialplans, or even what half of the menu options mean in this device. Zoom is promoting this device as being compatible with asterisk[1]. Hopefully someone from zoom is on this list who is familiar with this product... -Dan [1] http://www.zoomtel.com/products/voip_products.html ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pickup problem on Asterisk 1.2.4
I was wrong. The problem was with chan_sccp library and was solved downgrading from version 20060207 to 20060204. _fangi_ -Messaggio originale- Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di Francesco Angi Inviato: venerdì 24 febbraio 2006 10.00 A: Asterisk Users Mailing List - Non-Commercial Discussion Oggetto: [Asterisk-Users] pickup problem on Asterisk 1.2.4 Solved the problem downgrading zaptel 1.2.4 to 1.2.3. Mantaining the same configurations now everything works fine. Regards, _fangi_ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: res_features pickupexten
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... the callgroup/pickupgroup settings are correct... dialing *8 or *8# on any client (zap/sip/sccp) results in unknown extension... To pick-up with SIP phone, it has to be defined in sip.conf. Same goes for zap and iax2. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] run with incorrect E1/T1 jumper settings
Hi, I have installed an TE110P but forgot to change the jumper settings to E1. I don't have easy physical access to ther server at the moment so I wonder if it will be possible to run it without changing the jumper settings with a configuration like below or will it be impossible to use the card at all before I fix the jumper? I can't try it myself yet since the operator isn't ready yet, but I would like to know in advance if it is impossible. bchan=1-15,17-24 dchan=16 instead of bchan=1-15,17-31 dchan=16 best regards Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem calling out
Hi All, I installed Asterisk recently and it was working from 2 weeks without a problem until today. Today it started showing strange error Feb 28 03:14:08 WARNING[31430]: chan_sip.c:4826 check_auth: Stale nonce received from 'sip:[EMAIL PROTECTED]' Whatever number I call it displays this, please tell how can I fix this? I have no idea what is happening and the cause of this error? Thanks, Manoj. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] My or provider error?
Situation. I call out from SIP phone over h323 trunk and called person decides
not to pick up (on mobile phone they press red button - NO - hang-up). Until
the called person press the NO button, I can hear ringing. When called person
press the button, I don't hear anything. Asterisk waits until timeout and than
ends the call.
How can I get busy or some other appropriate signal on SIP phone headset?
This is what I have in extensions.conf. I use Asterisk 1.2.1 (soon I'll use
1.2.4)
exten = _0.,1,Dial,OOH323/${EXTEN:[EMAIL PROTECTED]
exten = _0.,n,Hangup
--
Tomislav Parcina
tparcina#lama.hr
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote: On Tue, Feb 28, 2006 at 09:20:05AM +0100, Armin Schindler wrote: OK, I've reported a bug to mISDN. With the patch from the Karsten Keil in the mantis tracker: issue: https://www.isdn4linux.de/mantis/view.php?id=40 I'm now gettig connect_ind and info_ind in the correct order (asterisk capi debug + verbose 15 log attached). The call still does not proceed in asterisk (the disconnect comes whe I unplug the ISDN-Line after several minutes). I've also attached my capi.conf and relevant portions of the dialplan. The version of mISDN is mqueue from yesterday with the mantis-tracker patch applied. Any ideas what I should try next? Well, the error message from mISDN: CAPI INFO 0x3301: Protocol error layer 1 (broken line or B-channel removed by signalling protocol) seems to be very clear. The ISDN line is not working or the used protocol is wrong. This error message occurs when I unplug the ISDN-Line from my testing machine LNG after the call already timed out on the remote end (I'm getting No answer). So the call is seen by asterisk, the info_ind is seen by asterisk, but then nothing else happens. What does chan_capi wait for after the connect? Ah, I see. Not very nice to send such a confusing log ;-) Anyway, your config is set to DID mode. So chan_capi will wait for more digits (an INFO_IND with called-party-number) and if the already given destination number does not match the extensions.conf yet. Is your line DID? If not, you should set isdnmode=msn. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] run with incorrect E1/T1 jumper settings
a better way is to to load the driver with all spans set to E1 by running modprobe wcte11xp t1e1override=15 or edit wcte11xp.c and change 'static int t1e1override = -1;' to 'static int t1e1override = 15;' Diyanat From: Robert Andersson [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] run with incorrect E1/T1 jumper settings Date: Tue, 28 Feb 2006 10:36:57 +0100 Precedence: list Hi, I have installed an TE110P but forgot to change the jumper settings to E1. I don't have easy physical access to ther server at the moment so I wonder if it will be possible to run it without changing the jumper settings with a configuration like below or will it be impossible to use the card at all before I fix the jumper? I can't try it myself yet since the operator isn't ready yet, but I would like to know in advance if it is impossible. bchan=1-15,17-24 dchan=16 instead of bchan=1-15,17-31 dchan=16 best regards Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ISDN30E + T1 crossover cable woes
Hi, I'm having problems getting our server to work with our BT ISDN30 box. We are using a Digium TE110P card to connect to the ISDN box on the wall. The card is configured as an E1 (strap on). I've made the T1 crossover cable ( well, made two variations ) and neither work. The light on the Digium card flashes red and the red LED's on the ISDN box stay lit. I've tested the configuration and all modules load ok. Done a ztcfg -v and things look ok. Done zttest and get around 99.7% - 99.9%. After loading asterisk I did a zap show status and alert is red. After googling I found this statement: RED: Loss of signal (LOS): The equipment shall assume loss of signal when the incoming signal amplitude is, for a time duration of at least 1 ms, more than 20 dB below the nominal amplitude. The equipment shall react within 12 ms by issuing AIS. The question is, is this a configuration issue, cable issue or BT issue. Digium's site is down so couldn't look there :-( One of the diagrams I used is here: http://www.gcom.com/home/documents/faqs/cables__t1.htm The other diagram ( can't seem to find that now ) used 1-5, 2-4 etc. Has anyone seen this problem? Thank you in advance. Phil.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] run with incorrect E1/T1 jumper settings
Thanks. Might have saved me a lot of trouble... best regards Robert Diyanat Ali wrote: a better way is to to load the driver with all spans set to E1 by running modprobe wcte11xp t1e1override=15 or edit wcte11xp.c and change 'static int t1e1override = -1;' to 'static int t1e1override = 15;' Diyanat From: Robert Andersson [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] run with incorrect E1/T1 jumper settings Date: Tue, 28 Feb 2006 10:36:57 +0100 Precedence: list Hi, I have installed an TE110P but forgot to change the jumper settings to E1. I don't have easy physical access to ther server at the moment so I wonder if it will be possible to run it without changing the jumper settings with a configuration like below or will it be impossible to use the card at all before I fix the jumper? I can't try it myself yet since the operator isn't ready yet, but I would like to know in advance if it is impossible. bchan=1-15,17-24 dchan=16 instead of bchan=1-15,17-31 dchan=16 best regards Robert ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: How can I debug spandsp?
Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... And how do you turn on Asterisk's debug facilities? Edit logger.conf and uncomment full. Start Asterisk with the the -d option. View debugging information in the /var/log/asterisk/full Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Q: Status of feature Call Deflection / Partial Rerouing in chan-capi and zaphfc
Hello, AFAIK the feature CD (call deflection) is only possible on point-to-multipoint links, is this correct? I've heard about the feature partial rerouting which should do the same on point-to-point-links. Is this implemented in either bristuff or chan-capi(-cm)? Thanks in advance, Karsten ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: res_features pickupexten
the callgroup/pickupgroup settings are correct... dialing *8 or *8# on any client (zap/sip/sccp) results in unknown extension... To pick-up with SIP phone, it has to be defined in sip.conf. Same goes for zap and iax2. callgroup and pickupgoup is configured in the config-files (zap/sip/sccp) - is anything else needed ? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VAD, CNG, for Zap
Hi, I have a few questions that I have been researching for a while. Sorry if it is a bit long winded? I have a huge need for VAD and CNG support in Asterisk, as my bandwith is *very * limited and expensive, and VAD, CNG and DTX will save me alot, at least 30 - 50%. I have installed and am using the SVN-bweschke-bug_5374-r9733, which adds VAD and silence suppression support to Asterisk. However, my understanding(and i am hoping people will correct me) is that VAD and CNG is based on the codec being used, so if g723 is being used, it will detect silence, and the codec will set a silence tag, which will tell asterisk this is silence, so generate comfort noise. So all of this will work if both endpoints of a particular call is using SIP with g723(for example). However,if one or both of the voice endpoints are ZAP, will CNG and VAD still be possible.? My topology is an Asterisk with an E1, calls come into the E1 from a PBX, and then get trunked to another Asterisk box, then go out on that E1. And since ZAP is analogue, is it possible for it to support silence suppresion and VAD? So, my question again, is silence suppresion possible when ZAP is involved I would greatly appreciate any pointers/suggestions. thanks, yusuf ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN30E + T1 crossover cable woes
Hi Phil, we have a very similar setup... ISDN30 plus TE110P... I used a standard cat5 patch lead... Worked a treat... [EMAIL PROTECTED] wrote: Hi, I'm having problems getting our server to work with our BT ISDN30 box. We are using a Digium TE110P card to connect to the ISDN box on the wall. The card is configured as an E1 (strap on). I've made the T1 crossover cable ( well, made two variations ) and neither work. The light on the Digium card flashes red and the red LED's on the ISDN box stay lit. I've tested the configuration and all modules load ok. Done a ztcfg -v and things look ok. Done zttest and get around 99.7% - 99.9%. After loading asterisk I did a zap show status and alert is red. After googling I found this statement: RED: Loss of signal (LOS): The equipment shall assume loss of signal when the incoming signal amplitude is, for a time duration of at least 1 ms, more than 20 dB below the nominal amplitude. The equipment shall react within 12 ms by issuing AIS. The question is, is this a configuration issue, cable issue or BT issue. Digium's site is down so couldn't look there :-( One of the diagrams I used is here: http://www.gcom.com/home/documents/faqs/cables__t1.htm The other diagram ( can't seem to find that now ) used 1-5, 2-4 etc. Has anyone seen this problem? Thank you in advance. Phil. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This e-mail is confidential and is intended solely for the use of the individual to whom it is addressed. Any views or opinions presented are solely those of the author and do not necessarily represent those of Bespoke Textile Computer Systems Ltd. If you are not the intended recipient, be advised that you have received this e-mail in error and that any use, dissemination, forwarding, printing, or copying of this e-mail is strictly prohibited. If you have received this email in error please email [EMAIL PROTECTED] -- Checked by alt-n MDaemon Antivirus Definition count: 168254 Definition date: 2006/02/28 MDAV version: 2.2.8 -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN30E + T1 crossover cable woes
Have you crc check enabled in zaptel.conf? Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk + WiFi Phones
What is the outcome of this finding on f3000. goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Thursday, November 24, 2005 10:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk + WiFi Phones The F3000 is also a clamshell, flip type phone. I should be receiving an eval unit shortly and will post my findings after we work it over in the lab. Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Luki wrote: UTStarCom has the F3000 coming in December, which will have according to their spec * WEP (64 and 128 bit )/WPA/MD5 Auth * Handover/Roaming between different AP and SSID So what else is different compared to the F1000? The 1000 also does WEP 64/128 and WPA with the newest firmware. Not sure about MD5 auth, but SIP nonce/MD5 response certainly is implemented. Roaming kind of works, but could be improved. In one place I made it from 4th floor - elevator - lobby while on the phone and without any noticeable dropouts (ulaw codec). But the building was covered with access points, on average NetStumbler saw 6 at the same time. So it works, but not always. Don't get me wrong, the phone does have issues and in my opinion is not production quality, meaning it will freak out unexpectedly and only a reboot helps, which hardly ever happens to any Sipura adapters or phones. Hopefully the new 3.6 firmware performs better. Luki ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: Re: Delay on Phone ringing
Hi, I have just joined this mail list yesterday and have been searching the Asterisk wiki prior to posting this question. Unfortunately I am not sure if I am searching at the correct places, so I do apologise if this has been posted before. I have currently been tasked to roll out VoIP phones through out our office as the current proprietary Panasonic PBX has no more channels. Thus I have installed [EMAIL PROTECTED] on VIA SP13000,512Mb Ram and using 2 x Digum TDM400P cards with both having 4x TDM40B FXO modules. I have rolled out 12 x Snom320 phones 1 x Snom360 in the office. For the test phase, I wanted to use the current PBX, Therefore Port 1 of the TDM is currently connected to one of the POTS extensions which is spare on the current PBX. Current problems I am facing in the test phase: Whenever I call from outside e.g. from the fax line (separate line) or my mobile, to the main number setup on the Trunk, I get a delay of around 12sec before the VoIP phone actually rings, although the phones connected to the current PBX, ring immediately. I have attached the output file and noticed that the DBget is trying to find something in the AstDB, would that be causing the delay? Or am I looking at the wrong place altogether. Please Help Regards Ash Thakrar asterisk1*CLI soft hangup Zap/1-1 Requested Hangup on channel 'Zap/1-1' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' -- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/1-1, ) in new stack -- Executing Wait(Zap/1-1, 1) in new stack -- Executing SetVar(Zap/1-1, intype=EXT-220) in new stack -- Executing Cut(Zap/1-1, intype=intype|-|1) in new stack -- Executing GotoIf(Zap/1-1, 1?7:9) in new stack -- Goto (from-pstn-reghours,s,7) -- Executing Wait(Zap/1-1, 3) in new stack -- Executing Goto(Zap/1-1, ext-local|220|1) in new stack -- Goto (ext-local,220,1) -- Executing Macro(Zap/1-1, exten-vm|novm|220) in new stack -- Executing Macro(Zap/1-1, user-callerid) in new stack -- Executing DBget(Zap/1-1, AMPUSER=DEVICE//user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=/user -- DBget: Value not found in database. -- Executing DBget(Zap/1-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname -- DBget: Value not found in database. -- Executing GotoIf(Zap/1-1, 1?5) in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp(Zap/1-1, Using CallerID ) in new stack -- Executing SetVar(Zap/1-1, FROMCONTEXT=exten-vm) in new stack -- Executing Macro(Zap/1-1, record-enable|220|IN) in new stack -- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Zap/1-1, recordingcheck|20060227-131600|1141046151.2) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060227-131600|1141046151.2: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(Zap/1-1, No recording needed) in new stack -- Executing Macro(Zap/1-1, dial|15|tr|220) in new stack -- Executing GotoIf(Zap/1-1, 0?4:2) in new stack -- Goto (macro-dial,s,2) -- Executing GotoIf(Zap/1-1, 0?5:4) in new stack -- Goto (macro-dial,s,4) -- Executing AGI(Zap/1-1, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- dialparties.agi: priority = 4 -- dialparties.agi: callingani2 = 0 -- dialparties.agi: accountcode = -- dialparties.agi: channel = Zap/1-1 -- dialparties.agi: callerid = unknown -- dialparties.agi: context = macro-dial -- dialparties.agi: callington = 0 -- dialparties.agi: dnid = unknown -- dialparties.agi: request = dialparties.agi -- dialparties.agi: calleridname = unknown -- dialparties.agi: extension = s -- dialparties.agi: language = en -- dialparties.agi: uniqueid = 1141046151.2 -- dialparties.agi: callingpres = 0 -- dialparties.agi: type = Zap -- dialparties.agi: rdnis = unknown -- dialparties.agi: callingtns = 0 -- dialparties.agi: enhanced = 0.0 dialparties.agi: Caller ID is not set dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 220 to extension map -- dialparties.agi: Extension 220 cf is disabled -- dialparties.agi: Extension 220 do not disturb is disabled -- dialparties.agi: Checking CW and CFB status for extension 220 == Parsing
[Asterisk-Users] Re: Re: How can I debug spandsp?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Edit logger.conf and uncomment full. Start Asterisk with the the -d option. View debugging information in the /var/log/asterisk/full Is -d option necessary? Anyway, done that. Just thought that you think about something else. Thank you! -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: res_features pickupexten
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... callgroup and pickupgoup is configured in the config-files (zap/sip/sccp) - is anything else needed ? Sorry, I'm not up to this. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Q: Status of feature Call Deflection / Partial Rerouing in chan-capi and zaphfc
On Tue, Feb 28, 2006 at 11:43:55AM +0100, Karsten Wemheuer wrote: Hello, AFAIK the feature CD (call deflection) is only possible on point-to-multipoint links, is this correct? At least on my ptp link capiinfo reports: [...] Supplementary services support: 0x0033 Hold / Retrieve Terminal Portability Call Forwarding Call Deflection which suggests that CD is available for ptp too. I have not checked if CD works with chan_capi though. Ralf -- Ralf Schlatterbeck email: [EMAIL PROTECTED] FAX: +43/2243/26465/23 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] monitor outgoing calls in queue / campaings
hi i'm migrating a callcenter to asterisk, inbound calls, queue monitorig is ok, but how can i monitot outgoing calls? for example my agent can be associated with more than one campaigns, so if i monitor his calls in a day, how can i learn about how many calls has he made for campaings A or campaings B? i'm thinking to add some extensions, for example: exten = 99XX,1;Register in db the call how campaings 99; exten = 99XX,2;Dial exten = 98XX,1;Register in db the call how campaings 98; exten = 98XX,2;Dial but, maybe there is something that do it automaticly...is it possible? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Tue, Feb 28, 2006 at 10:46:40AM +0100, Armin Schindler wrote: Ah, I see. Not very nice to send such a confusing log ;-) I'm sorry. Anyway, your config is set to DID mode. So chan_capi will wait for more digits (an INFO_IND with called-party-number) and if the already given destination number does not match the extensions.conf yet. Maybe that is the problem. The full DID information already is contained in the connect_ind according to the log I sent in the last Mail: CalledPartyNumber = 8111 [...] -- CONNECT_IND (PLCI=0x101,DID=11,CID=650621,CIP=0x10,CONTROLLER=0x1) ISDN1: msn='*' DNID='11' DID In Austria the PTP will send only the DID information, not the whole number, so 11 is all that is sent for DID. And according to my dialplan 11 should call one of my ZAP channels. Maybe chan_capi should have a timeout waiting for more info_ind and if the timeout is reached pass the call to asterisk anyway? How about checking if Sending Complete is set (don't know where this would be transmitted, in capi). What should I do to make chan_capi not wait for more info_ind (that apparently never come)? Is your line DID? If not, you should set isdnmode=msn. The line is DID (PTP or Anlagenanschluss). Ralf -- Ralf Schlatterbeck email: [EMAIL PROTECTED] FAX: +43/2243/26465/23 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Polycom Default Ring Volume [OT]
Hi Anton (et al) - Well.. I already sent my email to them :) Kind of OT here, but just out of curiosity, how do you email them? Do you have an actual address, or do you just use the form on their web site? I've sent a bunch of requests via that form, and even though it says I should receive a response, I never have. I've tried going through a couple of resellers, too, and haven't gotten anything. I'm hoping you have some more direct (and effective!) means. Thanks, Noah ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Re: Delay on Phone ringing
The only time I see recorded in your log is that of the recording check -- Executing AGI(Zap/1-1, recordingcheck|20060227-131600|1141046151.2) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060227-131600|1141046151.2: Inbound recording not enabled which doesn't seem to take any time. Only you would know at what phase the dialplan was in at each point of the 12 seconds. How long did it take before this took place: -- Starting simple switch on 'Zap/1-1' How long did this phase take: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- AGI Script dialparties.agi completed, returning 0 MARK. Ash Thakrar wrote: Hi, I have just joined this mail list yesterday and have been searching the Asterisk wiki prior to posting this question. Unfortunately I am not sure if I am searching at the correct places, so I do apologise if this has been posted before. I have currently been tasked to roll out VoIP phones through out our office as the current proprietary Panasonic PBX has no more channels. Thus I have installed [EMAIL PROTECTED] on VIA SP13000,512Mb Ram and using 2 x Digum TDM400P cards with both having 4x TDM40B FXO modules. I have rolled out 12 x Snom320 phones 1 x Snom360 in the office. For the test phase, I wanted to use the current PBX, Therefore Port 1 of the TDM is currently connected to one of the POTS extensions which is spare on the current PBX. Current problems I am facing in the test phase: Whenever I call from outside e.g. from the fax line (separate line) or my mobile, to the main number setup on the Trunk, I get a delay of around 12sec before the VoIP phone actually rings, although the phones connected to the current PBX, ring immediately. I have attached the output file and noticed that the DBget is trying to find ‘something’ in the AstDB, would that be causing the delay? Or am I looking at the wrong place altogether. Please Help Regards Ash Thakrar asterisk1*CLI soft hangup Zap/1-1 Requested Hangup on channel 'Zap/1-1' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' -- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/1-1, ) in new stack -- Executing Wait(Zap/1-1, 1) in new stack -- Executing SetVar(Zap/1-1, intype=EXT-220) in new stack -- Executing Cut(Zap/1-1, intype=intype|-|1) in new stack -- Executing GotoIf(Zap/1-1, 1?7:9) in new stack -- Goto (from-pstn-reghours,s,7) -- Executing Wait(Zap/1-1, 3) in new stack -- Executing Goto(Zap/1-1, ext-local|220|1) in new stack -- Goto (ext-local,220,1) -- Executing Macro(Zap/1-1, exten-vm|novm|220) in new stack -- Executing Macro(Zap/1-1, user-callerid) in new stack -- Executing DBget(Zap/1-1, AMPUSER=DEVICE//user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=/user -- DBget: Value not found in database. -- Executing DBget(Zap/1-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname -- DBget: Value not found in database. -- Executing GotoIf(Zap/1-1, 1?5) in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp(Zap/1-1, Using CallerID ) in new stack -- Executing SetVar(Zap/1-1, FROMCONTEXT=exten-vm) in new stack -- Executing Macro(Zap/1-1, record-enable|220|IN) in new stack -- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Zap/1-1, recordingcheck|20060227-131600|1141046151.2) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060227-131600|1141046151.2: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(Zap/1-1, No recording needed) in new stack -- Executing Macro(Zap/1-1, dial|15|tr|220) in new stack -- Executing GotoIf(Zap/1-1, 0?4:2) in new stack -- Goto (macro-dial,s,2) -- Executing GotoIf(Zap/1-1, 0?5:4) in new stack -- Goto (macro-dial,s,4) -- Executing AGI(Zap/1-1, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- dialparties.agi: priority = 4 -- dialparties.agi: callingani2 = 0 -- dialparties.agi: accountcode = -- dialparties.agi: channel = Zap/1-1 -- dialparties.agi: callerid = unknown -- dialparties.agi: context = macro-dial -- dialparties.agi: callington = 0 -- dialparties.agi: dnid = unknown -- dialparties.agi:
Re: [Asterisk-Users] ISDN30E + T1 crossover cable woes
Phil To connect a BT ISDN30e to the TE110P card you do NOT require a T1 crossover. You need a straight through cable, and any cat5 cable will be just fine. Rgds Tim [EMAIL PROTECTED] wrote: Hi, I'm having problems getting our server to work with our BT ISDN30 box. We are using a Digium TE110P card to connect to the ISDN box on the wall. The card is configured as an E1 (strap on). I've made the T1 crossover cable ( well, made two variations ) and neither work. The light on the Digium card flashes red and the red LED's on the ISDN box stay lit. I've tested the configuration and all modules load ok. Done a ztcfg -v and things look ok. Done zttest and get around 99.7% - 99.9%. After loading asterisk I did a zap show status and alert is red. After googling I found this statement: RED: Loss of signal (LOS): The equipment shall assume loss of signal when the incoming signal amplitude is, for a time duration of at least 1 ms, more than 20 dB below the nominal amplitude. The equipment shall react within 12 ms by issuing AIS. The question is, is this a configuration issue, cable issue or BT issue. Digium's site is down so couldn't look there :-( One of the diagrams I used is here: http://www.gcom.com/home/documents/faqs/cables__t1.htm The other diagram ( can't seem to find that now ) used 1-5, 2-4 etc. Has anyone seen this problem? Thank you in advance. Phil. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for Q.Sig success story
Hi Mimmus, I have just ordered a Sangoma A101, that I should receive pretty soon. As you may remember, I will try to connect it to an Alcatel 4400, using either EuroISDN or Q.Sig. In order to save me some time and effort, would you mind sending me some sample configuration, like the wanpipe and zaptel configs ? Again, my first goal is to have basic call setup running in both directions. My final goal is to interconnect heterogeneous legacy systems using a VoIP cloud (based on Asterisk and IAX2). Thanks in advance for any help you may provide. BR, - Patrick - -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mimmus Sent: jeudi, 26. janvier 2006 11:36 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Looking for Q.Sig success story Hi Mimmus, and thanks for the quick reply. You are welcome. It is actually very good to hear that most of it works. The difference in my project is that we'll keep the PSTN link on the Alcatel, and use the asterisk only as a inter-site trunking solution. The reason is that I have no Alcatel knowledge (will rely on other people), and I want to be as un-intrusive as possible. If you don't mind, I would have some additional questions: I have no knowledge of Alcatel too! I think that putting Asterisk in front of Alcatel is the best way to offer * advanced features (voicemail, audioconference, fax, ...) to all users. 1) Can you confirm that Q.Sig is the only option for me ? No idea! 2) What hardware are you using on the Asterisk (Digium ?) Tried both Digium TE410P and Sangoma A102. Better results with latest one. Once my pilot starts I may come back to you for some examples and advice, but this probably won't happen before a month or so. No problem. Bye Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk + WiFi Phones
Goksie - I have found the F3000 works fine with Asterisk, however, the general release of this phone has been pushed back several times by UTStarcom. At present, we have none of these available. I might suggest the Linksys WIP300 as an alternative. Cory J Andrews VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 ++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] AIM - B2CORY - Original Message - From: ADEGOKE ARUNA [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, February 28, 2006 6:49 AM Subject: RE: [Asterisk-Users] Asterisk + WiFi Phones What is the outcome of this finding on f3000. goksie -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Thursday, November 24, 2005 10:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk + WiFi Phones The F3000 is also a clamshell, flip type phone. I should be receiving an eval unit shortly and will post my findings after we work it over in the lab. Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Luki wrote: UTStarCom has the F3000 coming in December, which will have according to their spec * WEP (64 and 128 bit )/WPA/MD5 Auth * Handover/Roaming between different AP and SSID So what else is different compared to the F1000? The 1000 also does WEP 64/128 and WPA with the newest firmware. Not sure about MD5 auth, but SIP nonce/MD5 response certainly is implemented. Roaming kind of works, but could be improved. In one place I made it from 4th floor - elevator - lobby while on the phone and without any noticeable dropouts (ulaw codec). But the building was covered with access points, on average NetStumbler saw 6 at the same time. So it works, but not always. Don't get me wrong, the phone does have issues and in my opinion is not production quality, meaning it will freak out unexpectedly and only a reboot helps, which hardly ever happens to any Sipura adapters or phones. Hopefully the new 3.6 firmware performs better. Luki ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asttapi - what's wrong?
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... When I try to call from asttapi one number, I get message No one is available to answer at this time (1:0/0/0). Immediately after that I try to call the same number from SIP phone (the same one that is used with asttapi) and call goes true. What have I done wrong? Solved! Problem vas that manager adds default caller ID (not the one that was defined in sip.conf for the phone from which I'll will speak). And I need to sent to provider specific caller ID. Now, I have question. In agents conf, can I define Caller ID for every user (manager)? If not, that is something that defiantly should be implemented. -- Tomislav Parcina tparcina#lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] newbie debugger needs a little guidance
Hi guys, I am trying to step our asterisk server. All the internal phones / extensions work and I had the outgoing / incoming calls working before. But for some reason, unknown to me, it has stopped working. I have switched on sip debug and the main thing I notice is the recurring appearance of Noisy feedback tells: pid=2359 req_src_ip=217.155.69.86 req_src_port=5060 in_uri=sip:sip.jnctn.net out_uri=sip:sip.jnctn.net via_cnt==1 Can anyone help me with this? Thanks, James p.s. Here is a bit of the console debug output. Sip read: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.3.70:5060;branch=z9hG4bK0438ec30 From: sip:[EMAIL PROTECTED];tag=as58d6dd22 To: sip:[EMAIL PROTECTED];tag=1835cbfecbeb5b3c6b80319fb44e3d9b.f68f Call-ID: [EMAIL PROTECTED] CSeq: 120 REGISTER Contact: sip:[EMAIL PROTECTED]:5060;expires=120 Server: OpenSer (1.0.0-pre0 (i386/linux)) Content-Length: 0 Warning: 392 66.227.100.20:5060 Noisy feedback tells: pid=2359 req_src_ip=217.155.69.86 req_src_port=5060 in_uri=sip:sip.jnctn.net out_uri=sip:sip.jnctn.net via_cnt==1 10 headers, 0 lines Feb 28 07:20:09 NOTICE[8591]: chan_sip.c:6831 handle_response: Outbound Registration: Expiry for sip.jnctn.net is 120 sec (Scheduling reregistration in 105000 ms) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote: On Tue, Feb 28, 2006 at 10:46:40AM +0100, Armin Schindler wrote: Ah, I see. Not very nice to send such a confusing log ;-) I'm sorry. Anyway, your config is set to DID mode. So chan_capi will wait for more digits (an INFO_IND with called-party-number) and if the already given destination number does not match the extensions.conf yet. Maybe that is the problem. The full DID information already is contained in the connect_ind according to the log I sent in the last Mail: CalledPartyNumber = 8111 [...] -- CONNECT_IND (PLCI=0x101,DID=11,CID=650621,CIP=0x10,CONTROLLER=0x1) ISDN1: msn='*' DNID='11' DID In Austria the PTP will send only the DID information, not the whole number, so 11 is all that is sent for DID. That is okay, the dialplan just need to have a match. And according to my dialplan 11 should call one of my ZAP channels. Okay, but in DID mode chan_capi waits for additional INFO_INDs until a) a dialplan match is found or b) the line signals SENDING-COMPLETE or SETUP message (both are not sent by mISDN, but they are necessary). Maybe chan_capi should have a timeout waiting for more info_ind and if the timeout is reached pass the call to asterisk anyway? What for? The dialplan rules should give you enough to make this possible. How about checking if Sending Complete is set (don't know where this would be transmitted, in capi). Checking this as part of other messages is not implemented yet, but the CAPI driver must send this as INFO_IND as well, because it was requested by chan_capi. What should I do to make chan_capi not wait for more info_ind (that apparently never come)? As a workaround, you can check if isdnmode=msn with immediate=yes will work. But as far as I can see, mISDN must be fixed to provide correct and full INFO_IND. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with incoming call, Please help
Hi All, I was able to install Asterisk and make outgoing calls. Recently I purchased two DID's and I am facing a problem configuring them to my Asterisk, I hope with the help I get from this list I will be able to configure successfully. Mu errors are Feb 28 08:31:58 NOTICE[19133]: pbx.c:1331 pbx_extension_helper: Cannot find extension context 'context_mantra2' Feb 28 08:31:58 NOTICE[19135]: chan_iax2.c:5761 socket_read: Rejected connect attempt from 208.139.204.245, request '[EMAIL PROTECTED] (or a valid context in your extensions.conf of your choosing.)' does not exist I have incoming and default contexts like this [incoming] exten = 18883003532,1,Answer() exten = 18883003532,2,DIAL(SIP/manoj,20) exten = 8883003532,1,Answer() exten = 8883003532,2,DIAL(SIP/manoj,20) Please help me configure this. Thanks, Manoj ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Question
On 28/02/06, Alexander Lopez [EMAIL PROTECTED] wrote: read STDIN while [ x$STDIN != x ] do export VARNAME=`echo $STDIN | cut -f1 -d :` export VARVALUE=`echo $STDIN |cut -f2 -d : | cut -c2-255` case $VARNAME in (agi_request) export AGIREQUEST=$VARVALUE;; (agi_channel) export AGICHANNEL=$VARVALUE;; (agi_language) export AGILANGUAGE=$VARVALUE;; (agi_type) export AGITYPE=$VARVALUE;; (agi_uniqueid) export AGIUNIQUEID=$VARVALUE;; (agi_callerid) export AGICALLERID=$VARVALUE;; (agi_calleridname) export AGICALLERIDNAME=$VARVALUE;; (agi_dnid) export AGIDNID=$VARVALUE;; (agi_rdnis) export AGIRDNIS=$VARVALUE;; (agi_context) export AGICONTEXT=$VARVALUE;; (agi_extension) export AGIEXTENSION=$VARVALUE;; (agi_priority) export AGIPRIORITY=$VARVALUE;; (agi_enhanced) export AGIENHANCED=$VARVALUE;; (agi_accountcode) export AGIACCOUNTCODE=$VARVALUE;; esac read STDIN done } # # You now have all the stuff Asterisk gives you stored in Variables # # # Do what ever you want here: # # -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, February 27, 2006 10:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Asterisk Question I was going to see if I can execute a bash script as an AGI - just looking around the internet for examples at the moment. Anybody got an example spare? I'm just a bit stuck on how to start this, but I am quite comfortable writing asterisk dialplan stuff and bash scripts later, PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FW: 7960-tones.xml (Schochet, Wes)
As the thread from the other mailing list he sent this to states, it is illegal to share the file(s) he is asking for. Below is the thread from the sccp users mailing list that he sent this to. sccp mailing list 2006/2/28, picciuX In fact: the one you mention is not a config file; it is part of the Locale-Installer for Cisco Call Manager. You need a valid service contract to download it. Sorry... picciuX 2006/2/28, Schochet, Wes [EMAIL PROTECTED]: I know that's true of firmware, there seems to be a lot of XML config file examples out there on just about every web site you find. of course, not these particular one that I am looking for From: Kaleb L. Kunzler [ ] Sent: Monday, February 27, 2006 7:13 PM To: [EMAIL PROTECTED] Subject: RE: [Chan-sccp-users] 7960-tones.xml Wes, it would be illegal for anyone on the list to send you this file (or any Cisco file). To get the necessary files you need to contact Cisco, who will most likely tell you that you need a service contract with them. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Schochet, Wes Sent: Monday, February 27, 2006 6:05 PM To: '[EMAIL PROTECTED]' Subject: [Chan-sccp-users] 7960-tones.xml I am afraid I have the wrong version of this file that somehow got loaded. Does anyone have a US version? How about 7960-fonts.xml? Thanks, Wes ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] variables internas
I've seen in the asterisk configuration the way to call some internal variables like caller-id-number, caller-id-name, language, etc. but.. What is the variable for changing the DID? Is there a manual with this details? -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Set CallerIDNum on a PRI
Hi, I have a PRI line with DID (from 100 to 499) in Italy. Now I'm seeing all calls from same DID 'main' number. Can I set outgoing CallerIdNum to the right extension? Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] AGI Scripts Terminate too Soon
In the 1.0.x branch asterisk does not always send SIGHUP to agi scripts on call hangup. In 1.2.x a SIGHUP is always sent, even using DEADAGI - From the UPGRADE.txt in the source: AGI: * AGI scripts did not always get SIGHUP at the end, previously. That behavior has been fixed. If you do not want your script to terminate at the end of AGI being called (e.g. on a hangup) then set SIGHUP to be ignored within your application. Craig - Original Message - From: Darren Wiebe [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 28, 2006 10:09 AM Subject: Re: [Asterisk-Users] AGI Scripts Terminate too Soon In that case, asterisk sends -HUP to the agi script (I believe). Darren Michael Collins wrote: If that's true, why does dial() return control to the script when the callee hangs up? Doug, if I understand the AGI limitation correctly, the 'dead' in DeadAGI() refers to the other end of a dial() connection. I *think*, but I'm not positive on that. Does anyone know the answer to this one? Thanks, MC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Darren Wiebe [EMAIL PROTECTED] Aleph Communications ASTPP - Open Source Voip Billing Calling Cards www.aleph-com.net/astpp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] playing hold time announcement without queue position announcement
Greetings fellow list members, I have what I think is a relatively simple question, but it did not appear to be addressed on the wiki. I am trying to setup a queue so that it plays an estimated holdtime announcement, but not a queue position announcement. Currently my dialplan does both, and while I know how to take out the estimated holdtime without affecting the queue position announcement, I do not see how to do the oppositte. Does anyone know how to do this? Here is a sample of one of my queues from queues.conf: ;; ;;* Development Test Queue * ;; [10001] announce=beep2 ;* a beep to alert the agent of the call servicelevel=30 ;* target service level (maximum time in queue in seconds) musiconhold=default ;* sets music for this queue strategy=rrmemory ;* sets method of allocating calls to reps timeout=20 ;* how long do we let phone ring before it is a timeout retry=5 ;* how long to wait before trying all members again weight=1 ;* weight against other queues sharing agents wrapuptime=4 ;* how long to wait before freeing up for another call maxlen=0 ;* maximum people in queue (0 is no limit) announce-frequency=60 ;* time between position/estimated hold time announcements announce-holdtime=yes ;* announce estimated hold time (yes|no|once) monitor-format=pcm;* record calls in pcm format monitor-join=yes ;* join recordings joinempty=no ;* callers can join an empty queue leavewhenempty=yes;* remove callers from queue if no agents on I know I can set "announce-holdtime" to "no" and remove the hold time, but I'm unsure how to keep the hold time but remove the queue position. in this section of the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+config+queues.conf you can see it says: ;Howoftentoannouncequeuepositionand/orestimatedholdtimetocaller(0=off) ; ;announce-frequency=90 ; ;Shouldweincludeestimatedholdtimeinpositionannouncements? ;Eitheryes,no,oronlyonce;holdtimewillnotbeannouncedif1minute ;;announce-holdtime=yes|no|once the fact that is says "and/or" leads me to believe there is a way to only play the hold time without the queue position, but I do not see any suggestions on how to do this. Thanks in advance for any advice, Franklin Webb Inter Media Marketing Solutions ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Set CallerIDNum on a PRI
Hi, Mimmus wrote: I have a PRI line with DID (from 100 to 499) in Italy. Now I'm seeing all calls from same DID 'main' number. Can I set outgoing CallerIdNum to the right extension? Yes, assuming your telco allows you to. Be sure to figure out what number format is required in your case. Your telco can tell you. (Often this is the full DID without a leading 0) Best regards, Florian ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Tue, Feb 28, 2006 at 02:32:27PM +0100, Armin Schindler wrote:
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
On Tue, Feb 28, 2006 at 10:46:40AM +0100, Armin Schindler wrote:
Anyway, your config is set to DID mode. So chan_capi will wait for more
digits (an INFO_IND with called-party-number) and if the already given
destination number does not match the extensions.conf yet.
Maybe that is the problem. The full DID information already is contained
in the connect_ind according to the log I sent in the last Mail:
CalledPartyNumber = 8111
[...]
-- CONNECT_IND
(PLCI=0x101,DID=11,CID=650621,CIP=0x10,CONTROLLER=0x1)
ISDN1: msn='*' DNID='11' DID
In Austria the PTP will send only the DID information, not the whole
number, so 11 is all that is sent for DID.
That is okay, the dialplan just need to have a match.
And according to my
dialplan 11 should call one of my ZAP channels.
Okay, but in DID mode chan_capi waits for additional INFO_INDs until
a) a dialplan match is found or
That condition should be fulfilled, I have
[...]
exten = 11,1,Noop()
exten = 11,2,Noop(11)
exten = 11,3,Dial(${UNTEN})
exten = 11,4,Busy()
exten = 11,104,Busy()
in my dialplan. But the call isn't seen by my dialplan.
b) the line signals SENDING-COMPLETE or SETUP message
(both are not sent by mISDN, but they are necessary).
OK, I'll append that to the already-open bug-report in the misdn
mantis-tracker.
Maybe chan_capi should have a timeout waiting for more info_ind and if
the timeout is reached pass the call to asterisk anyway?
What for? The dialplan rules should give you enough to make this possible.
See above: Seems the dialplan match isn't detected??
[...]
As a workaround, you can check if isdnmode=msn with immediate=yes will work.
But as far as I can see, mISDN must be fixed to provide correct and full
INFO_IND.
OK, I'll try.
Ralf
--
Ralf Schlatterbeck
email: [EMAIL PROTECTED] FAX: +43/2243/26465/23
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk hangs up - h323
This is third time today that my Asterisk hangs up. It seams that I have problems with h323. I'm using ooh323 from Asterisk add-ons. I have the following configuration Asterisk 1.2.1 Asterisk-addons 1.2.1 Fedora Core 4 I'm using SIP phones and h323 trunk to my VoIP provider Like I said this is third time today that he hang's up. First time, I came at work and Asterisk was down. Second time I tried to call, and Asterisk was down (not sure at that wary moment or before I tried to call). So, I decide to start logging and this is what I received just before Asterisk died. Anyway, I tried to reload from CLI and that is when he died. What can I do to check why it's happening? I have plenty of disk space, lots of free ram and processor is idle for more than 80%. I think it could be because of alaw codec that I use (my provider requires it) and this is what is in ooh323.conf file (ONLY ulaw, gsm, g729 and g7231 supported as of now). But Like I said, it works for several hours and then it dies... So I don't think that is it. ooh323.conf [general] bindaddr=xxx.xxx.xxx.xxx h323id=ObjSysAsterisk e164=100 callerid=asterisk gatekeeper = DISABLE context=incomingh323 disallow=all allow=alaw allow=ulaw allow=gsm dtmfmode=rfc2833 full.pbx Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/manager.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/manager.conf': Found Feb 28 14:04:15 NOTICE[5018] cdr.c: CDR simple logging enabled. Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/rtp.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/rtp.conf': Found Feb 28 14:04:15 VERBOSE[5018] logger.c: == RTP Allocating from port range 1 - 2 Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_musiconhold.so' (Music On Hold Resource) Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/musiconhold.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/musiconhold.conf': Found Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_indications.so' (Indications Configuration) Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_adsi.so' (ADSI Resource) Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_features.so' (Call Features Resource) Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/features.conf': Feb 28 14:04:15 VERBOSE[5018] logger.c: == Parsing '/etc/asterisk/features.conf': Found Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature Blind Transfer (blindxfer) to sequence '#1' Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature Attended Transfer (atxfer) to sequence '#2' Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature One Touch Monitor (automon) to sequence '#3' Feb 28 14:04:15 VERBOSE[5018] logger.c: == Remapping feature Disconnect Call (disconnect) to sequence '#0' Feb 28 14:04:15 DEBUG[5018] res_features.c: Removed old parking extension [EMAIL PROTECTED] Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Added extension '700' priority 1 to parkedcalls Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_config_mysql.so' (MySQL RealTime Configuration Driver) Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime Host: Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime Port: 0 Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime User: Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime Password: Feb 28 14:04:15 ERROR[5018] res_config_mysql.c: MySQL RealTime: Failed to connect database server on . Check debug for more info. Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime: Cannot Connect: Can't connect to local MySQL server through socket '' (111) Feb 28 14:04:15 WARNING[5018] res_config_mysql.c: MySQL RealTime: Couldn't establish connection. Check debug. Feb 28 14:04:15 DEBUG[5018] res_config_mysql.c: MySQL RealTime: Cannot Connect: Can't connect to local MySQL server through socket '' (111) Feb 28 14:04:15 VERBOSE[5018] logger.c: == MySQL RealTime reloaded. Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'res_crypto.so' (Cryptographic Digital Signatures) Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'chan_iax2.so' (Inter Asterisk eXchange (Ver 2)) Feb 28 14:04:15 ERROR[5018] chan_iax2.c: Unable to load config iax.conf Feb 28 14:04:15 VERBOSE[5018] logger.c: == Loaded firmware 'iaxy.bin' Feb 28 14:04:15 NOTICE[5018] iax2-provision.c: No IAX provisioning configuration found, IAX provisioning disabled. Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'chan_skinny.so' (Skinny Client Control Protocol (Skinny)) Feb 28 14:04:15 NOTICE[5018] chan_skinny.c: Unable to load config skinny.conf, Skinny disabled Feb 28 14:04:15 VERBOSE[5018] logger.c: -- Reloading module 'chan_local.so' (Local Proxy Channel) Feb 28 14:04:15
RE: [Asterisk-Users] fax receive using TDM400P
Using Asterisk 1.2.1, why not 1.2.4? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Ioan Indreias |Sent: Tuesday, February 28, 2006 1:29 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] fax receive using TDM400P | |We have just installed one machine with FC3 (with last |updates) + asterisk |1.2.1 + spandsp-0.0.2pre21. From our tests it shows OK. | |Ioan Indreias |Modulo Consulting |www.tenora.ro | |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Anton Krall |Sent: Tuesday, February 28, 2006 6:35 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] fax receive using TDM400P | |Ok 1 for Debian, any Fedoras Core 3 out there? | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Thomas ||Artner ||Sent: Monday, February 27, 2006 4:57 PM ||To: Asterisk Users Mailing List - Non-Commercial Discussion ||Subject: Re: [Asterisk-Users] fax receive using TDM400P || ||Am Monday 27 February 2006 23:15 schrieb Anton Krall: || Guys.. I just thought of something.. Anybody who is sucessfuly || receviing faxes using spandsp and running Fedora Core 3? || What are you running? || ||Debian stable - and it works perfectly. || || | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax receive using TDM400P
That what worries me, the 2 systems Im testing are completely different. One has x100p cards (2) and the other has 2 TDM400P with 4 FXO and 1 TE110P.. All same results... No go. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Thomas Artner |Sent: Tuesday, February 28, 2006 1:35 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] fax receive using TDM400P | |Anton Krall wrote: | Ok 1 for Debian, any Fedoras Core 3 out there? | |I think it doesn't depend on the linux distribution whether it |works or not. |It's rather an hardware issue. | | | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf |Of Thomas | |Artner | |Sent: Monday, February 27, 2006 4:57 PM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] fax receive using TDM400P | | | |Am Monday 27 February 2006 23:15 schrieb Anton Krall: | | Guys.. I just thought of something.. Anybody who is sucessfuly | | receviing faxes using spandsp and running Fedora Core 3? | | What are you running? | | | |Debian stable - and it works perfectly. | | | | | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Austria isdn p2p empty DID
Hi there! I've set up an [EMAIL PROTECTED] with AVM C2 P2P ... Everything works fine ... BUT ... If someone is dialing the PBX head number without any extension, asterisk can't handle this ... the DID in this case is empty Any ideas how to handle this? Regards, Marcus Hofbauer -- |** realität ist da wo der pizzamann herkommt **| ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax receive using TDM400P
Yep, been there, done that. How about this results: [EMAIL PROTECTED] asterisk]# /usr/src/zaptel-1.2.4/zttest -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% --- Results after 15 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.997559 Anything above 99.98 is good so.. Why isnt faxing working :( |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Tuesday, February 28, 2006 2:18 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] fax receive using TDM400P | | | Ok 1 for Debian, any Fedoras Core 3 out there? | |fc3, and it doesn't work. | |If you check the archives, this has all been discussed before. |The issue seems to be more oriented to the specific pci bus |implementation on the motherboard. You might also want to run |/usr/src/zaptel/zttest and read the archives on that as well. | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Polycom Default Ring Volume [OT]
No, same as you, thru the form on their website... :( |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Noah I. Miller |Sent: Tuesday, February 28, 2006 6:56 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: [Asterisk-Users] Re: Polycom Default Ring Volume [OT] | |Hi Anton (et al) - | | Well.. I already sent my email to them :) | |Kind of OT here, but just out of curiosity, how do you email |them? Do you have an actual address, or do you just use the |form on their web site? I've sent a bunch of requests via |that form, and even though it says I should receive a |response, I never have. I've tried going through a couple of |resellers, too, and haven't gotten anything. I'm hoping you |have some more direct (and effective!) means. | |Thanks, |Noah |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot boot machine up after working on zaptel....
Hi all, hard for me to explain this, but it keeps happening on a number of machines I attempt to upgrade zaptel, or do something to zaptel modules. and then I reboot the machine, and for whatever reason, it hangs on loading the modules Either the install wasn't complete, the zaptel modules settings are wrong, whatever but the problem is now I can't get past the boot up and the machine is basically lost Is there any way to bypass the module load attempt or anything? I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD, but no go I'm on Debian 2.4.18, with Zaptel 1.0.9.2 I understand that there was something wrong in the modules config, but surely I should be able to bypass and get back in to fix it! Any ideas greatly appreciated, as I would rather not have to use an old clone drive and start over -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Re: Delay on Phone ringing
Hi Mark, Thanks for your reply. For the phase you have indicated the time it took was immediate, no delays there. Regards Ash -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Hulber Sent: 28 February 2006 13:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] FW: Re: Delay on Phone ringing The only time I see recorded in your log is that of the recording check -- Executing AGI(Zap/1-1, recordingcheck|20060227-131600|1141046151.2) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060227-131600|1141046151.2: Inbound recording not enabled which doesn't seem to take any time. Only you would know at what phase the dialplan was in at each point of the 12 seconds. How long did it take before this took place: -- Starting simple switch on 'Zap/1-1' How long did this phase take: -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- AGI Script dialparties.agi completed, returning 0 MARK. Ash Thakrar wrote: Hi, I have just joined this mail list yesterday and have been searching the Asterisk wiki prior to posting this question. Unfortunately I am not sure if I am searching at the correct places, so I do apologise if this has been posted before. I have currently been tasked to roll out VoIP phones through out our office as the current proprietary Panasonic PBX has no more channels. Thus I have installed [EMAIL PROTECTED] on VIA SP13000,512Mb Ram and using 2 x Digum TDM400P cards with both having 4x TDM40B FXO modules. I have rolled out 12 x Snom320 phones 1 x Snom360 in the office. For the test phase, I wanted to use the current PBX, Therefore Port 1 of the TDM is currently connected to one of the POTS extensions which is spare on the current PBX. Current problems I am facing in the test phase: Whenever I call from outside e.g. from the fax line (separate line) or my mobile, to the main number setup on the Trunk, I get a delay of around 12sec before the VoIP phone actually rings, although the phones connected to the current PBX, ring immediately. I have attached the output file and noticed that the DBget is trying to find 'something' in the AstDB, would that be causing the delay? Or am I looking at the wrong place altogether. Please Help Regards Ash Thakrar asterisk1*CLI soft hangup Zap/1-1 Requested Hangup on channel 'Zap/1-1' == Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm' == Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' -- Starting simple switch on 'Zap/1-1' -- Executing GotoIf(Zap/1-1, 1?from-pstn-reghours|s|1:) in new stack -- Goto (from-pstn-reghours,s,1) -- Executing GotoIf(Zap/1-1, 0?from-pstn-reghours-nofax|s|1:2) in new stack -- Goto (from-pstn-reghours,s,2) -- Executing Answer(Zap/1-1, ) in new stack -- Executing Wait(Zap/1-1, 1) in new stack -- Executing SetVar(Zap/1-1, intype=EXT-220) in new stack -- Executing Cut(Zap/1-1, intype=intype|-|1) in new stack -- Executing GotoIf(Zap/1-1, 1?7:9) in new stack -- Goto (from-pstn-reghours,s,7) -- Executing Wait(Zap/1-1, 3) in new stack -- Executing Goto(Zap/1-1, ext-local|220|1) in new stack -- Goto (ext-local,220,1) -- Executing Macro(Zap/1-1, exten-vm|novm|220) in new stack -- Executing Macro(Zap/1-1, user-callerid) in new stack -- Executing DBget(Zap/1-1, AMPUSER=DEVICE//user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=/user -- DBget: Value not found in database. -- Executing DBget(Zap/1-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname -- DBget: Value not found in database. -- Executing GotoIf(Zap/1-1, 1?5) in new stack -- Goto (macro-user-callerid,s,5) -- Executing NoOp(Zap/1-1, Using CallerID ) in new stack -- Executing SetVar(Zap/1-1, FROMCONTEXT=exten-vm) in new stack -- Executing Macro(Zap/1-1, record-enable|220|IN) in new stack -- Executing GotoIf(Zap/1-1, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(Zap/1-1, recordingcheck|20060227-131600|1141046151.2) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060227-131600|1141046151.2: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(Zap/1-1, No recording needed) in new stack -- Executing Macro(Zap/1-1, dial|15|tr|220) in new stack -- Executing GotoIf(Zap/1-1, 0?4:2) in new stack -- Goto (macro-dial,s,2) -- Executing GotoIf(Zap/1-1, 0?5:4) in new stack -- Goto (macro-dial,s,4) -- Executing AGI(Zap/1-1,
[Asterisk-Users] Conference bridge dimensioning
We are using an Asterisk box to do conferencing right now. I have had about sixteen active lines in conference and the quality was acceptable. We now have a need for 50 people to conference at one time. Does anyone have enough experience doing this to give me some pointers. Will it even be reasonable to try this? Is the mixing done on the the hardware, I plan on using a quad span t-1 card from Digium? The server is a fedora box with a dual core xeon at 2.0 Ghz and 2 gigs of Ram. Is there a rule of thumb to go by as far as conferencing resources? Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 1-800-666-2833 x299 (608) 783-7560 x299 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax receive using TDM400P
FC2 SpanDSP -pre25, Te110P. Works perfect. -Original Message- From: Anton Krall [mailto:[EMAIL PROTECTED] Sent: Monday, February 27, 2006 9:35 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] fax receive using TDM400P Ok 1 for Debian, any Fedoras Core 3 out there? |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Thomas Artner |Sent: Monday, February 27, 2006 4:57 PM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: Re: [Asterisk-Users] fax receive using TDM400P | |Am Monday 27 February 2006 23:15 schrieb Anton Krall: | Guys.. I just thought of something.. Anybody who is sucessfuly | receviing faxes using spandsp and running Fedora Core 3? | What are you running? | |Debian stable - and it works perfectly. | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Austria isdn p2p empty DID
Marcus Hofbauer [EMAIL PROTECTED] writes: BUT ... If someone is dialing the PBX head number without any extension, asterisk can't handle this ... the DID in this case is empty Any ideas how to handle this? Try the WaitExten application. cu, Wolfgang ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] changing source email address of pager notifications
Anyone have a clue how to get the voicemail pager notification (actually, text message) source email address to change? We use both the email and pager feature, so just using the email feature to send test messages is not an option. We also do not manage the users email, so creating aliases the go to two addresses is not really practical. We want the email to include an attachment, so the pager feature must be used for text messages. Everything is working as planned, but the pager message source email address is the [EMAIL PROTECTED], not what is specified in serveremail= Thx ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cannot boot machine up after working on zapt el....
What happens if you take out the Zaptel I/F's? If it boots, you can correct whatever you did then replace them. hth -Original Message- From: Chris Earle (CBL) [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 28, 2006 7:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cannot boot machine up after working on zaptel Hi all, hard for me to explain this, but it keeps happening on a number of machines I attempt to upgrade zaptel, or do something to zaptel modules. and then I reboot the machine, and for whatever reason, it hangs on loading the modules Either the install wasn't complete, the zaptel modules settings are wrong, whatever but the problem is now I can't get past the boot up and the machine is basically lost Is there any way to bypass the module load attempt or anything? I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD, but no go I'm on Debian 2.4.18, with Zaptel 1.0.9.2 I understand that there was something wrong in the modules config, but surely I should be able to bypass and get back in to fix it! Any ideas greatly appreciated, as I would rather not have to use an old clone drive and start over -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Austria isdn p2p empty DID
On Tue, Feb 28, 2006 at 03:40:28PM +0100, Marcus Hofbauer wrote: I've set up an [EMAIL PROTECTED] with AVM C2 P2P ... Everything works fine ... BUT ... If someone is dialing the PBX head number without any extension, asterisk can't handle this ... the DID in this case is empty Do you have an s extension in your dialplan for the context where incoming isdn calls are handled? -- Ralf Schlatterbeck email: [EMAIL PROTECTED] FAX: +43/2243/26465/23 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
On Tue, Feb 28, 2006 at 02:32:27PM +0100, Armin Schindler wrote:
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote:
On Tue, Feb 28, 2006 at 10:46:40AM +0100, Armin Schindler wrote:
Anyway, your config is set to DID mode. So chan_capi will wait for more
digits (an INFO_IND with called-party-number) and if the already given
destination number does not match the extensions.conf yet.
Maybe that is the problem. The full DID information already is contained
in the connect_ind according to the log I sent in the last Mail:
CalledPartyNumber = 8111
[...]
-- CONNECT_IND
(PLCI=0x101,DID=11,CID=650621,CIP=0x10,CONTROLLER=0x1)
ISDN1: msn='*' DNID='11' DID
In Austria the PTP will send only the DID information, not the whole
number, so 11 is all that is sent for DID.
That is okay, the dialplan just need to have a match.
And according to my
dialplan 11 should call one of my ZAP channels.
Okay, but in DID mode chan_capi waits for additional INFO_INDs until
a) a dialplan match is found or
That condition should be fulfilled, I have
[...]
exten = 11,1,Noop()
exten = 11,2,Noop(11)
exten = 11,3,Dial(${UNTEN})
exten = 11,4,Busy()
exten = 11,104,Busy()
in my dialplan. But the call isn't seen by my dialplan.
That is correct so far.
Maybe chan_capi should have a timeout waiting for more info_ind and if
the timeout is reached pass the call to asterisk anyway?
What for? The dialplan rules should give you enough to make this possible.
See above: Seems the dialplan match isn't detected??
It will be detected when chan_capi gets the signal to check it
(SENDING-COMPLETE/SETUP INFO_IND, or normal CONNECT_IND in MSN mode with
immediate=yes).
Armin
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot boot machine up after working on zaptel....
Thanks Yeah, you would think so wouldn't you. Tried that , and still wouldn't boot Really annoying. beacuse I've been doing work with the zaptel drivers and such and this happened once already... Thanks for the suggestion, Chris - Original Message - From: Colin Anderson [EMAIL PROTECTED] To: 'Chris Earle (CBL)' [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, February 28, 2006 10:05 AM Subject: RE: [Asterisk-Users] Cannot boot machine up after working on zaptel What happens if you take out the Zaptel I/F's? If it boots, you can correct whatever you did then replace them. hth -Original Message- From: Chris Earle (CBL) [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 28, 2006 7:45 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Cannot boot machine up after working on zaptel Hi all, hard for me to explain this, but it keeps happening on a number of machines I attempt to upgrade zaptel, or do something to zaptel modules. and then I reboot the machine, and for whatever reason, it hangs on loading the modules Either the install wasn't complete, the zaptel modules settings are wrong, whatever but the problem is now I can't get past the boot up and the machine is basically lost Is there any way to bypass the module load attempt or anything? I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD, but no go I'm on Debian 2.4.18, with Zaptel 1.0.9.2 I understand that there was something wrong in the modules config, but surely I should be able to bypass and get back in to fix it! Any ideas greatly appreciated, as I would rather not have to use an old clone drive and start over -- Chris Earle System Solutions Specialist -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Austria isdn p2p empty DID
On Tue, 28 Feb 2006, Marcus Hofbauer wrote: Hi there! I've set up an [EMAIL PROTECTED] with AVM C2 P2P ... Everything works fine ... BUT ... If someone is dialing the PBX head number without any extension, asterisk can't handle this ... the DID in this case is empty Any ideas how to handle this? When setting immediate=yes in capi.conf, such a call will be sent into extension 's' of your context. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[OFFLIST] Re: [Asterisk-Users] run with incorrect E1/T1 jumper settings
Robert Andersson wrote: Hi, I have installed an TE110P but forgot to change the jumper settings to E1. I don't have easy physical access to ther server at the moment so I wonder if it will be possible to run it without changing the jumper settings with a configuration like below or will it be impossible to use the card at all before I fix the jumper? I can't try it myself yet since the operator isn't ready yet, but I would like to know in advance if it is impossible. bchan=1-15,17-24 dchan=16 instead of bchan=1-15,17-31 dchan=16 best regards Robert Tjena. Fick du det att fungera? Jag är lite osäker om det går att tvinga det. Såg något svar på listan där, men jag har faktiskt aldrig provat det själv. /Mvh Micke - Mikael Andersson dCAp Certified ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot boot machine up after working on zaptel....
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 28.02.2006, at 15:44, Chris Earle ((CBL)) wrote: Hi all, hard for me to explain this, but it keeps happening on a number of machines I attempt to upgrade zaptel, or do something to zaptel modules. and then I reboot the machine, and for whatever reason, it hangs on loading the modules Either the install wasn't complete, the zaptel modules settings are wrong, whatever but the problem is now I can't get past the boot up and the machine is basically lost Is there any way to bypass the module load attempt or anything? I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD, but no go I'm on Debian 2.4.18, with Zaptel 1.0.9.2 I understand that there was something wrong in the modules config, but surely I should be able to bypass and get back in to fix it! Any ideas greatly appreciated, as I would rather not have to use an old clone drive and start over Hi Chris, How about you use a Live CD distribution and disable the loading of the driver in some config? Unfortunately I'm not very familiar with Debian, in Gentoo you would edit /etc/modules.autoload/kernel-2.6 and then uncomment the line that loads the module. You should then be able to boot normally and do what you have to do in order to get it to work. Does this also happen when you load the driver using modprobe? Christoph - -- GPG Key ID: 33D6AA8C AIM: zeitgeist2600 ICQ: 271512600 Jabber: [EMAIL PROTECTED] http://www.geisterstunde.org http://www.ceicke.de -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEBGzw8e/ZGTPWqowRAleSAJ0WIcjiORoRTnd1mTWJNYUj9WuWDACfX7zn 8cadgA0CfHPAgB0Rww5XCHw= =AT6i -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
I have also issues with jitter over wan (cdma), I'm trying to debug how dejitter buffer is working (using iax2 jb debug), but nothing happens/no debug output on asterisk console :-( is any way how to monitor iax jitter buffer? thx PJ I'm really hoping to see some working settings from some people here. The jitterbuffer is one of the main features I've been looking forward to in 1.2. Here are my current settings, if anyone notices a major problem please let me know. I'm using dropcount of 2 hoping that a shrink in the jitterbuffer will happen a little faster as a trade-off. Am I thinking correctly on this? I moved the resyncthreshold way up since people are having issues with it. My thoughts on minexcessbuffer=60 is to immediately get a decent buffer going, as this is much higher than the jitter I usually see (~20ms). jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=60 jittershrinkrate=1 maxjitterinterps=10 resyncthreshold=1500 Thanks, -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN30E + T1 crossover cable woes
I had a similar issue here in Aus where I was chasing crossover cables around. Eventually the cows actually did come home and I called up the telco. They rebuilt (or reinitialized) the ISDN service and everything worked a treat from there on in. Took a couple of days to get to this point. Suggest you will probably be OK with straight-through Cat5e. Phone up BT and give em some stick. Mark. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Tuesday, 28 February 2006 9:34 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ISDN30E + T1 crossover cable woes Hi, I'm having problems getting our server to work with our BT ISDN30 box. We are using a Digium TE110P card to connect to the ISDN box on the wall. The card is configured as an E1 (strap on). I've made the T1 crossover cable ( well, made two variations ) and neither work. The light on the Digium card flashes red and the red LED's on the ISDN box stay lit. I've tested the configuration and all modules load ok. Done a ztcfg -v and things look ok. Done zttest and get around 99.7% - 99.9%. After loading asterisk I did a zap show status and alert is red. After googling I found this statement: RED: Loss of signal (LOS): The equipment shall assume loss of signal when the incoming signal amplitude is, for a time duration of at least 1 ms, more than 20 dB below the nominal amplitude. The equipment shall react within 12 ms by issuing AIS. The question is, is this a configuration issue, cable issue or BT issue. Digium's site is down so couldn't look there :-( One of the diagrams I used is here: http://www.gcom.com/home/documents/faqs/cables__t1.htm The other diagram ( can't seem to find that now ) used 1-5, 2-4 etc. Has anyone seen this problem? Thank you in advance. Phil. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Echo and other reasons to migrate to BRI from POTS? Was (Echo on PRI/BRI?)
Paul, Ah, I see. Our echo is largly under control now. It took me a while to figure out the gains and get them tuned, and now the echo only leaves very small artifacts. Nonetheless, this still provokes the odd complaint here and there. We use VOIP for outgoing calls when our POTS lines are congested, and we find zero echo during those calls. Therefore, I assume that our handsets (Cisco _79[46]0's) handle echo properly, and the source is our local loop. I suppose then I cannot promise that migrating to a pair of BRI circuits (4 channels) will eliminate echo. It would be safe to say that echo would PROBABLY be eliminated? Other reasons to migrate: eliminate static/line noise (from our local loop, can't do anything about the other end), speed up call setup time, eliminate the 1 in 1000 chance that you will accidentally answer an incoming call when trying to place an outgoing call. Reasons not to migrate: more costly (about $15/month/channel), harder to configure (I'm a bit intimidated - can't have downtime), if the * server blows up one cannot simply plug in a $10 handset from Walmart to get some bit of functionality - MUST use ISDN hardware. Any experianced opinions on this? Brent- Echo can occur for all sorts of reasons- analog conversions as someone else already mentioned, 4 wire to 2 wire in particular- but could also occur in the IP path due to network issues, and can occur on any sort of digital or analog circuit due to various electrical or audio components. (one of the more commonly neglected causes is poor handsets, that do a bad job of isolating the speaker and microphone, or attempt to add sidetone incorrectly (sidetone is the slight echo you should hear of your own voice- it's very hard to hear, but without it, you get the feeling that you're talking to a dead wire). Conversion to BRI/PRI is a last step only, in my opinion, unless you have other compelling reasons to do so- there's a lot of other places to look first. If you could describe more of your particular setup, I'd be happy to give more detailed description of where the problems might lie. -Paul Davidson PlanCommunications, LLC Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918x325 Voice 219.836.1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to determine duration call when is used Attended Transfer
Hi,
I am trying to determine the actual call duration (billsec) when is
used Attended Transfer but this is very dificult because there is no
relation between channels. Are there any suggestions how can be solved
this?
I have an idea where in the CDR must be added new column where to
be stored the CDR UniqueID from another channel which is linked.
Or to have another database/table (res_features) where to store all
events like transfer, hold and conferences.
How to add some very useful patch for Attended Transfer? In the
standard source code of function builtin_atxfer(...) is writen:
newchan = ast_feature_request_and_dial(
transferer, Local,
ast_best_codec(transferer-nativeformats),
dialstr,
15000,
outstate, cid_num, cid_name);
We replace this line with this one:
newchan = ast_feature_request_and_dial(
transferer, Local,
ast_best_codec(transferer-nativeformats),
dialstr,
atxfernoanswertimeout,
outstate, cid_num, cid_name);
where atxfernoanswertimeout is static int which can be configured
in features.conf. the default value is 15000.
Follow the main useful patch where when the other party is busy the
channel return busy signal instead return to the caller. The function
is ast_feature_request_and_dial(...):
...
else if ((f-subclass == AST_CONTROL_BUSY) ||
(f-subclass == AST_CONTROL_CONGESTION))
{
state = f-subclass;
- new lines:
if (option_verbose 2)
ast_verbose( VERBOSE_PREFIX_3 %s is busy\n, chan-name);
ast_indicate(caller, AST_CONTROL_BUSY);
- old lines:
/*
ast_frfree(f);
f = NULL;
break;
*/
Best Regards,
Miroslav Nachev
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax receive using TDM400P
Ah! A spandsp pre25... Ok.. The plot thickens :) |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Colin Anderson |Sent: Tuesday, February 28, 2006 8:55 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] fax receive using TDM400P | |FC2 SpanDSP -pre25, Te110P. Works perfect. | |-Original Message- |From: Anton Krall [mailto:[EMAIL PROTECTED] |Sent: Monday, February 27, 2006 9:35 PM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] fax receive using TDM400P | | |Ok 1 for Debian, any Fedoras Core 3 out there? | ||-Original Message- ||From: [EMAIL PROTECTED] ||[mailto:[EMAIL PROTECTED] On Behalf Of Thomas ||Artner ||Sent: Monday, February 27, 2006 4:57 PM ||To: Asterisk Users Mailing List - Non-Commercial Discussion ||Subject: Re: [Asterisk-Users] fax receive using TDM400P || ||Am Monday 27 February 2006 23:15 schrieb Anton Krall: || Guys.. I just thought of something.. Anybody who is sucessfuly || receviing faxes using spandsp and running Fedora Core 3? || What are you running? || ||Debian stable - and it works perfectly. || || | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] why incoming DATA CALLS are answered as VOICE by asterisk IVR?
When incoming DATA call arrive on ISDN BRI, asterisk (zaphfc) recognise that this is DATA call, but answering anyway (playing IVR messages, etc...) How to stop that? I want that only VOICE calls are answered, and DATA/FAX to be ignored. (I'm using Asterisk 1.2.1 Brisftuffed 0.3.0-PRE-1f, ZapHFC) Log: -- Accepting data call from '' to '3001' on channel 0/2, span 1 -- Executing Answer(Zap/2-1, ) in new stack -- Executing BackGround(Zap/2-1, ivr_intro) in new stack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax receive using TDM400P
How about this: --- Results after 33 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.988163 Faxing is working just fine. Mabe it's mother board related? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Tuesday, February 28, 2006 4:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] fax receive using TDM400P Yep, been there, done that. How about this results: [EMAIL PROTECTED] asterisk]# /usr/src/zaptel-1.2.4/zttest -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8193 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% 8192 samples in 8192 sample intervals 100.00% --- Results after 15 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.997559 Anything above 99.98 is good so.. Why isnt faxing working :( |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Rich Adamson |Sent: Tuesday, February 28, 2006 2:18 AM |To: Asterisk Users Mailing List - Non-Commercial Discussion |Subject: RE: [Asterisk-Users] fax receive using TDM400P | | | Ok 1 for Debian, any Fedoras Core 3 out there? | |fc3, and it doesn't work. | |If you check the archives, this has all been discussed before. |The issue seems to be more oriented to the specific pci bus |implementation on the motherboard. You might also want to run |/usr/src/zaptel/zttest and read the archives on that as well. | | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Echo and other reasons to migrate to BRI
Brent-There is no good way to say what changing the hardware and PSTN hookup will probably do for the echo problems. I'm not sure if you mentioned (lost in the past history of your post now) what sort of hardware you're using for PSTN connection now- TDMs, X100s, ATA's, etc- but that could also be a potential cause. I've heard tell of aftermarket X100s and certain ATA's being very finicky with echo- and age of hardware can sometimes make a difference. I personally use and recommend the Cisco handsets- it's hard (IMHO) to get echo generated there. Adjusting the gains, as you've already done, is generally step two, and should be done with caution, but it sounds like you've got that part down. I will say that, if you have generally poor audio quality on your PSTN circuits- (and I would measure the difference between 'through the 7960' and 'through an analog handset plugged into the line' by ear to confirm- you may have already done this), it is definitely time to start looking for alternate PSTN termination. Each new method, however, brings with it additional chances for echo problems- however, if you're working with a single Asterisk box, all lines and handsets terminated to it, a digital circuit directly to it (via Digium TDM card) presents the *smallest* possible chance for echo problems- you're left with network issues, or possible server performance issues. Instead of BRI (which has more complicated hardware and channel drivers), you might consider a fractional PRI, or an Integrated Access circuit, where you're bringing in the full T1, but paying only for a few channels. I know that that's available here in Illinois, at very competitive pricing to BRI circuits. You may also want to switch to 100% VoIP provided termination, porting your number to a carrier (I'm recently a fan of NuFone, and they're relatively local to you, with centers in Michigan and Chicago- but YMMV), as you know that's an echo free solution. -Paul DavidsonPlanCommunications, LLC Date: Tue, 28 Feb 2006 09:52:02 -0600From: Brent Torrenga [EMAIL PROTECTED]Subject: [Asterisk-Users] Re: Echo and other reasons to migrate to BRIfrom POTS? Was (Echo on PRI/BRI?)To: asterisk-users@lists.digium.comMessage-ID: 000b01c63c7e$eae0ba60$7200a8c0@oscarContent-Type: text/plain;charset=us-asciiPaul, Ah, I see. Our echo is largly under control now. It took me a while tofigure out the gains and get them tuned, and now the echo only leaves verysmall artifacts. Nonetheless, this still provokes the odd complaint here and there. We use VOIP for outgoing calls when our POTS lines are congested, andwe find zero echo during those calls. Therefore, I assume that our handsets(Cisco _79[46]0's) handle echo properly, and the source is our local loop. I suppose then I cannot promise that migrating to a pair of BRI circuits (4channels) will eliminate echo. It would be safe to say that echo wouldPROBABLY be eliminated?Other reasons to migrate: eliminate static/line noise (from our local loop, can't do anything about the other end), speed up call setup time, eliminatethe 1 in 1000 chance that you will accidentally answer an incoming callwhen trying to place an outgoing call.Reasons not to migrate: more costly (about $15/month/channel), harder to configure (I'm a bit intimidated - can't have downtime), if the * serverblows up one cannot simply plug in a $10 handset from Walmart to get somebit of functionality - MUST use ISDN hardware.Any experianced opinions on this? Brent-Echo can occur for all sorts of reasons- analog conversions as someone elsealready mentioned, 4 wire to 2 wire in particular- but could also occur inthe IP path due to network issues, and can occur on any sort of digital or analog circuit due to various electrical or audio components. (one of themore commonly neglected causes is poor handsets, that do a bad job ofisolating the speaker and microphone, or attempt to add sidetone incorrectly(sidetone is the slight echo you should hear of your own voice- it's veryhard to hear, but without it, you get the feeling that you're talking to adead wire). Conversion to BRI/PRI is a last step only, in my opinion, unless you have other compelling reasons to do so- there's a lot of otherplaces to look first.If you could describe more of your particular setup, I'd be happy to givemore detailed description of where the problems might lie. -Paul Davidson PlanCommunications, LLCSincerely,Brent A. Torrenga[EMAIL PROTECTED] Torrenga Engineering, Inc.907 Ridge RoadMunster, Indiana 46321-1771219.836.8918x325 Voice219.836.1138 Facsimile www.torrenga.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] fax receive using TDM400P
One board is intel and the other is also intel (supermicro). :( |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Cosmin Prund |Sent: Tuesday, February 28, 2006 10:28 AM |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] fax receive using TDM400P | |How about this: | |--- Results after 33 passes --- |Best: 100.00 -- Worst: 99.987793 -- Average: 99.988163 | |Faxing is working just fine. Mabe it's mother board related? | | -Original Message- | From: [EMAIL PROTECTED] |[mailto:asterisk-users- | [EMAIL PROTECTED] On Behalf Of Anton Krall | Sent: Tuesday, February 28, 2006 4:41 PM | To: 'Asterisk Users Mailing List - Non-Commercial Discussion' | Subject: RE: [Asterisk-Users] fax receive using TDM400P | | Yep, been there, done that. | | How about this results: | | [EMAIL PROTECTED] asterisk]# /usr/src/zaptel-1.2.4/zttest -v Opened pseudo | zap interface, measuring accuracy... | | 8192 samples in 8191 sample intervals 99.987793% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8191 sample intervals 99.987793% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8193 sample intervals 99.987793% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | 8192 samples in 8192 sample intervals 100.00% | --- Results after 15 passes --- | Best: 100.00 -- Worst: 99.987793 -- Average: 99.997559 | | Anything above 99.98 is good so.. Why isnt faxing working :( | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf Of Rich | |Adamson | |Sent: Tuesday, February 28, 2006 2:18 AM | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: RE: [Asterisk-Users] fax receive using TDM400P | | | | | | Ok 1 for Debian, any Fedoras Core 3 out there? | | | |fc3, and it doesn't work. | | | |If you check the archives, this has all been discussed before. | |The issue seems to be more oriented to the specific pci bus | |implementation on the motherboard. You might also want to run | |/usr/src/zaptel/zttest and read the archives on that as well. | | | | | |___ | |--Bandwidth and Colocation provided by Easynews.com -- | | | |Asterisk-Users mailing list | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | | | ___ | --Bandwidth and Colocation provided by Easynews.com -- | | Asterisk-Users mailing list | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | |___ |--Bandwidth and Colocation provided by Easynews.com -- | |Asterisk-Users mailing list |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | | ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with HT 488 FXO
Hi Pasqualotto, Actually, I've seen your post on Asterisk-Users list yesterday, but I could not understand back then. Now, I've checked your sip configuration again, I think you make a mistake in type of sip account. I use friend not peer. I am not sure though. Following is what I had in my sip.conf file for the FXO port of HT488: [41] username=41 type=friend secret=put your password here host=dynamic context=put your context here callerid=Outside-line 41 dtmfmode=inband group=1 callgroup=1 pickupgroup=1 Of course, you should configure HT488 FXO sip account accordingly too. You should make sure that HT488 registers with Asterisk. Also read again the following thread: http://lists.digium.com/pipermail/asterisk-users/2005-August/123548.html Now, when you call 41 from another phone, you should be able to hear the dial tone. And if you configured HT488 to answer incomming calls to FXO and where they should be directed to (Forward to VoIP box), then you should be able to call in HT488 FXO and talk to Asterisk after a few rings. (HT488 configuration is also very important, I don't know what settings you have there.) I don't have a HT488 these days, so I cannot test your configurations, sorry. Soner - Original Message - From: Pasqualotto Enrico [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, February 27, 2006 9:54 PM Subject: [Asterisk-Users] Asterisk with HT 488 FXO Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug: -- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport From: Unknown sip:[EMAIL PROTECTED];tag=as073738f8 To: sip:192.168.1.157:5062;tag=ebc4a8e2 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Grandstream HT488 1.0.2.16 Contact: sip:[EMAIL PROTECTED]:5062;user=phone Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 192.168.1.157:5062: SIP/2.0 481 No Such Call Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport From: Unknown sip:[EMAIL PROTECTED];tag=as073738f8 To: sip:192.168.1.157:5062;tag=52242a6b Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Grandstream HT488 1.0.2.16 Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.1.157:5060: REGISTER sip:192.168.1.157 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport From: sip:[EMAIL PROTECTED];tag=as558874a4 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 120 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI -- SIP read from 192.168.1.157:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport From: sip:[EMAIL PROTECTED];tag=as558874a4 To: sip:[EMAIL PROTECTED];tag=3a733fa7 Call-ID: [EMAIL PROTECTED] CSeq: 120 REGISTER User-Agent: Grandstream HT488 1.0.2.16 Content-Length: 0 --- The register string ?? Can anyone help me?? Thanks -- Pasqualotto Enrico email: [EMAIL PROTECTED] web: http://www.pasqualotto.org -BEGIN GEEK CODE BLOCK- Version: 3.12 GIT d? s: a-- C+++ UL P L++ E--- W++ N++ o K- w--- O-- M V-- PS+ PE+ Y PGP- t--- 5 X R tv-- b+ DI- D+ G e h++ r+ y+ --END GEEK CODE BLOCK-- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cannot boot machine up after working on zapt el....
Boot up with this: http://www.sysresccd.org/Main_Page Mount the partition in question and remove the Zaptel module. Reboot, and you should be good (except for Zaptel of course) hth -Original Message- From: Christoph Eicke [mailto:[EMAIL PROTECTED] Sent: Tuesday, February 28, 2006 8:32 AM To: Chris Earle (CBL); Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cannot boot machine up after working on zaptel -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 28.02.2006, at 15:44, Chris Earle ((CBL)) wrote: Hi all, hard for me to explain this, but it keeps happening on a number of machines I attempt to upgrade zaptel, or do something to zaptel modules. and then I reboot the machine, and for whatever reason, it hangs on loading the modules Either the install wasn't complete, the zaptel modules settings are wrong, whatever but the problem is now I can't get past the boot up and the machine is basically lost Is there any way to bypass the module load attempt or anything? I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD, but no go I'm on Debian 2.4.18, with Zaptel 1.0.9.2 I understand that there was something wrong in the modules config, but surely I should be able to bypass and get back in to fix it! Any ideas greatly appreciated, as I would rather not have to use an old clone drive and start over Hi Chris, How about you use a Live CD distribution and disable the loading of the driver in some config? Unfortunately I'm not very familiar with Debian, in Gentoo you would edit /etc/modules.autoload/kernel-2.6 and then uncomment the line that loads the module. You should then be able to boot normally and do what you have to do in order to get it to work. Does this also happen when you load the driver using modprobe? Christoph - -- GPG Key ID: 33D6AA8C AIM: zeitgeist2600 ICQ: 271512600 Jabber: [EMAIL PROTECTED] http://www.geisterstunde.org http://www.ceicke.de -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin) iD8DBQFEBGzw8e/ZGTPWqowRAleSAJ0WIcjiORoRTnd1mTWJNYUj9WuWDACfX7zn 8cadgA0CfHPAgB0Rww5XCHw= =AT6i -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cheapest provider for Philippine route
Do anyone know who can provide some cheap PH routes/. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk 1.2.4 IAX2 New Jitterbuffer Tuning
Ron, keep in mind, that yoy mix parameters for new and old iax jitterbuffer implementation, these: dropcount=2 maxexcessbuffer=300 minexcessbuffer=60 jittershrinkrate=1 maxjitterinterps=10 are ae valid only for _old_ implementation, and I thing, that asterisk 1.2 use new iax buffer by default... so, I'm using only: jiterbuffer=yes forcejitterbuffer=yes maxjitterbuffer=1500 resyncthreshold=-1 but I don't know, how to monitor if jb is even working, because no output from iax2 jb debug :-( can anybody explain? PJ Ron Senykoff wrote: I'm really hoping to see some working settings from some people here. The jitterbuffer is one of the main features I've been looking forward to in 1.2. Here are my current settings, if anyone notices a major problem please let me know. I'm using dropcount of 2 hoping that a shrink in the jitterbuffer will happen a little faster as a trade-off. Am I thinking correctly on this? I moved the resyncthreshold way up since people are having issues with it. My thoughts on minexcessbuffer=60 is to immediately get a decent buffer going, as this is much higher than the jitter I usually see (~20ms). jitterbuffer=yes dropcount=2 maxjitterbuffer=500 maxexcessbuffer=300 minexcessbuffer=60 jittershrinkrate=1 maxjitterinterps=10 resyncthreshold=1500 Thanks, -Ron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zoom 5801 problems with *
On Feb 28, 2006, at 1:22 AM, [EMAIL PROTECTED] wrote: I am having problems with a Zoom 5801 and *. It does not appear possible to route voip calls out the FXO, all voip calls get routed to the FXS no matter what.snip If there is a routing function of some kind on the modem setup, perhaps you can change the default to the FXO? I only suggest this, because this is how I got the wellgate 3701a to dial out the fxo... Good Luck, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] callthru and CDR
is it possible
resetcdr and/or start newcdrAFTER pickup of
dialout?
[dialthru]exten
= s,1,Answer()exten = s,2,DigitTimeout,4exten =
s,3,ResponseTimeout,10 exten = s,4,Playtones(dial);exten =
i,1,Playback(invalid)exten = i,2,Goto(dialthru,s,2);exten =
t,1,Playback(timeout)exten = t,2,Goto(dialthru,s,2)
exten =
_X.,1,Dial(TRUNK/${EXTEN})
...
turby
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheapest provider for Philippine route
Sam Tam wrote: Do anyone know who can provide some cheap PH routes/.’ I've been looking myself. Cheapest DIDs in Metro Manila I've seen are $27.50/month; cheapest termination to same (non-mobile) from US I've seen is $0.23/minute. Expensive chismis :-) -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cannot boot machine up after working on zapt el....
On 08:05, Tue 28 Feb 06, Colin Anderson wrote: I've tried holding SHIFT down to get the LILO menu, and loading LinuxOLD, but no go Do this, pick the kernel you want to load, and add: single So in my laptops case it sais: Linux single That will boot your pc into singleuser mode and it won't enter the modprobe stage. You can now remove the .ko file and reboot. Good luck -- Michiel van Baak [EMAIL PROTECTED] http://michiel.vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheapest provider for Philippine route
i think we can help. we do have there some contacts and if the total volume would be significant, we can give a nice quotation. let me know off list what you exactly need. BTW, $0.23/minute is much much high compared to our solution. On 2/28/06, Johnathan Corgan [EMAIL PROTECTED] wrote: Sam Tam wrote: Do anyone know who can provide some cheap PH routes/.'I've been looking myself.Cheapest DIDs in Metro Manila I've seen are $27.50/month; cheapest termination to same (non-mobile) from US I'veseen is $0.23/minute.Expensive chismis :-)-Johnathan___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] Cheapest provider for Philippine route
Can be as low as 15€cents from us on fix and 20€cents for mobiles We don't have dids yet for Philipine -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Johnathan Corgan Envoyé : mardi 28 février 2006 18:07 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [Asterisk-Users] Cheapest provider for Philippine route Sam Tam wrote: Do anyone know who can provide some cheap PH routes/.’ I've been looking myself. Cheapest DIDs in Metro Manila I've seen are $27.50/month; cheapest termination to same (non-mobile) from US I've seen is $0.23/minute. Expensive chismis :-) -Johnathan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T38 fax pass thru to Cisco as53xx
Dear all, Did anyone successfully test T38 fax pass thru to Cisco as53xx? Weve tried 1.2.4 with latest patch and latest svn trunk and T38 patch but still not work. Reinvites from Cisco are correctly passed back to the originating gateway, but fax never able to connect. Cisco IOS 12.3.x configuration voice service voip fax protocol t38 ls-redundancy 5 hs-redundancy 2 fallback pass-through g711alaw h323 sip Thanks Ray ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound quality issue in one direction and wctdm problem with APIC enabled kernel
I'm chasing down a pop/click type of disturbance on a PBX system. Strangely, the disturbance is only heard by the outside caller, the internal recipient hears the caller crystal clear. This seems to have crept up when upgrading the zaptel driver to the 1.2 series while running 1.0.10. I went ahead and upgraded the entire system to 1.2.4. The system is a ~2Ghz AMD 32bit system, with 512MB of memory and nothing other than Asterisk running. Phone traffic is minimal, perhaps 3 simultaneous calls max, but the problem occurs with just one call. It's located in a data center with ~20ms pings to the ITSP and ~20ms pings to the remote office IP phones. Up to this point, ztdummy was in use without problems, although the timing (zttest) was a hair under the recommended threshold. I dropped in a TDM400P for testing, and although the timing improved, the symptom remained. The system has an IDE drive, and I verified the hdparm dma/irq settings were enabled. The TDM card was sharing interrupts, so I recompiled the kernel with APIC support. Unfortunately the wctdm module will no longer load after recompile and install into the new kernel directory. I went back to the ztdummy driver with the same problem. Below is the relevant errors and info. Chris # modprobe wctdm FATAL: Error inserting wctdm (/lib/modules/2.6.12-prep/misc/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm # dmesg wctdm: disagrees about version of symbol zt_receive wctdm: Unknown symbol zt_receive wctdm: disagrees about version of symbol zt_qevent_lock wctdm: Unknown symbol zt_qevent_lock wctdm: disagrees about version of symbol zt_ec_chunk wctdm: Unknown symbol zt_ec_chunk wctdm: disagrees about version of symbol zt_transmit wctdm: Unknown symbol zt_transmit wctdm: disagrees about version of symbol zt_unregister wctdm: Unknown symbol zt_unregister wctdm: disagrees about version of symbol zt_hooksig wctdm: Unknown symbol zt_hooksig wctdm: disagrees about version of symbol zt_register wctdm: Unknown symbol zt_register # cat /proc/interrupts CPU0 0: 34991774IO-APIC-edge timer 1: 10IO-APIC-edge i8042 8: 1IO-APIC-edge rtc 9: 1 IO-APIC-level acpi 12:111IO-APIC-edge i8042 14: 170392IO-APIC-edge ide0 15: 383872IO-APIC-edge ide1 18: 0 IO-APIC-level SiS SI7012, SiS SI7013 Modem 19: 164220 IO-APIC-level eth0 20: 0 IO-APIC-level ohci_hcd:usb2 21: 0 IO-APIC-level ohci_hcd:usb3 22: 0 IO-APIC-level ohci_hcd:usb4 23: 0 IO-APIC-level ehci_hcd:usb1 NMI: 0 LOC: 34991738 ERR: 0 MIS: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Replicating functionality from our prior PBX
We have just installed Asterisk in our new office and we have some teething problems, but so far nothing we did not expect/could not handle. However, our CEO was very attached to a function in our old Nortel PBX that I am not sure how to approach. If someone could point me in the right direction, I would be most grateful. The function is this: CEO records message, then specifies a list of extensions for that message to be sent to Any thoughts, questions, comments would be appreciated. Thanks, Patrick Foster ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] GSM phone reception range extendor
I think I have seen a post about that before. But cant find it again Can some people light me up with the detail ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Re: Delay on Phone ringing
On Feb 28, 2006, at 6:50 AM, Ash Thakrar wrote: Hi Mark, Thanks for your reply. For the phase you have indicated the time it took was immediate, no delays there. I have seen on the list several discussions of how additional delay on ringing can be due to Asterisk trying to get caller ID info... Might try searching the list archive for how to turn that off? Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conference bridge dimensioning
Hi Jordan,We are planning on building the same thing. We are still waiting for the hardware. We are using a Dell PE 2850 3GHz with 2G of RAM and a TE210P.I asked Digium support if this can suupport 50 users in one conference and the tech support guy said yes. Here's also a response from this list http://lists.digium.com/pipermail/asterisk-users/2006-February/147956.htmlLet's share our experiences.Goodluck.richardJordan Novak [EMAIL PROTECTED] wrote:We are using an Asterisk box to do conferencing right now. I have had about sixteen active lines in conference and the quality was acceptable. We now have a need for 50 people to conference at one time. Does anyone have enough experience doing this to give me some pointers. Will it even be reasonable to try this? Is the mixing done on the the hardware, I plan on using a quad span t-1 card from Digium? The server is a fedora box with a dual core xeon at 2.0 Ghz and 2 gigs of Ram. Is there a rule of thumb to go by as far as conferencing resources?Jordan Novak Communications Technician Logistics Health Inc. 1319 Saint Andrews Street La Crosse WI 54603 1-800-666-2833 x299 (608) 783-7560 x299 ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Asterisk Question
Paul,
Just curious - what kind of stuff are you reading from the file?
-MC
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Monday, February 27, 2006 7:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Asterisk Question
I was going to see if I can execute a bash script as an AGI - just
looking
around the internet for examples at the moment.
Anybody got an example spare?
I'm just a bit stuck on how to start this, but I am quite comfortable
writing asterisk dialplan stuff and bash scripts
later,
PaulH
steve [EMAIL PROTECTED] wrote:
From: Paul Hales [EMAIL PROTECTED]
Subject: [Asterisk-Users] Asterisk question
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain
Any idea how to read an external file, grab some stuff and push it
back
into an Asterisk variable?
I can do it the other way with:
system(echo ${UNIQUEID} = /home/ast/curr_calls)
but I'm a bit stumped on how to go the other way around
much thanks,
Paul Hales
I'll go out on a limb here and take a guess that it could be done as
an
AGI script that incorporates SED (http://www.gnu.org/software/sed/)
and
AWK (http://www.gnu.org/software/gawk/gawk.html). I've used both
for
some bash scripting in the past. . .
Regards,
Steve Cayona
Super Technologies, Inc.
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Comfort noise support incomplete in Asterisk (RFC 3389)
Hi guys,I'm using Zyxel Prestige 2602R, as router/SIP-ua with my architecture SER+Asterisk.Normally, everything is fine. In these days I'm experiencing some problems: some guests said me that, if he call everything is right, but if is called, he cannot hear the caller. Immediately, I though into an RTP-Proxy problem, but is not.Then I saw that message appear on the Asterisk CLI, during the incoming call:NOTICE[3575]: rtp.c:330 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: XXX.XXX.XXX.XXXNow I've checked into the router, and the VAD was already unset.Using normal IP-telephones, everything is perfect.Does anyone, got an idea or already got problems with that router?Thanks to all -- .:FaberK:. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM phone reception range extendor
On Wed, Mar 01, 2006 at 12:45:45AM +0800, Sam Tam wrote: I think I have seen a post about that before. But can't find it again Can some people light me up with the detail GSM extenders I don't think are legal in the UK, except if installed/operated by a GSM network operator (as they re-transmit and you need a license to operate in the GSM bands). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo stevekennedyuk / MSN [EMAIL PROTECTED] Euro Tech News Blog http://eurotechnews.blogspot.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H.323 ( HW PBX to *)
Hi, Im trying to connect * to Nortel BCM 50, This PBX use H.323 v3 to interface with other PBX. The port use to connect is TCP 1720 but I cant configure this port on my * box. Im using a H.323.conf file sample to activate the port but the * isnt listening there. Somebody have any idea or tip? This is mi H.323.conf [general] port = 1720 bindaddr = 192.168.0.200 ;tos=lowdelay ; ; amaflags = default ; ; ;accountcode=lss0101 ; ; allow=all ; turns on all installed codecs ;disallow=g723.1 ; Hm... Proprietary, don't use it... ;allow=gsm ; Always allow GSM, it's cool :) ;allow=ulaw ; User-Input Mode ( DTMF ) ; ; valid entries are: rfc2833, inband ; default is rfc2833 dtmfmode=rfc2833 ; ; Set the gatekeeper ; DISCOVER - Find the Gk address using multicast ; DISABLE - Disable the use of a GK ; IP address or Host name - The acutal IP address or hostname of your GK ;gatekeeper = DISABLE ; Tell Asterisk whether or not to accept Gatekeeper routed calls or not. Normally this should always be set to yes, unless you want to have finer control over wh ch users are allowed access to Asterisk. Default: YES ; AllowGKRouted = yes ; Default context gets used in siutations where you are using the GK routed model or no type=user was found. This gives you the ability to either play an invalid message or to simply not use user authentication at all. Thanks in advance. Pedro. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zoom 5801 problems with *
Here is a link to some additional resources which may be helpful in configuring the 5801 and other Zoom products http://www.zoom.com/salessupport/index.php?pd=ATA_Documentation/ Cory Andrews Purchasing Manager ++ VOIPSupply.com A Division of b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 direct - 716.250.3402 mobile - 716.907.4054 email - [EMAIL PROTECTED] AIM - b2Cory - Original Message - From: Martin Joseph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, February 28, 2006 11:50 AM Subject: Re: [Asterisk-Users] Zoom 5801 problems with * On Feb 28, 2006, at 1:22 AM, [EMAIL PROTECTED] wrote: I am having problems with a Zoom 5801 and *. It does not appear possible to route voip calls out the FXO, all voip calls get routed to the FXS no matter what.snip If there is a routing function of some kind on the modem setup, perhaps you can change the default to the FXO? I only suggest this, because this is how I got the wellgate 3701a to dial out the fxo... Good Luck, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: chan_capi-cm-0.6.4
On Tue, 28 Feb 2006, Ralf Schlatterbeck wrote: ... - But I guess the workaround would yield to my current situation (I'm running a patched version of 0.35 currently as mentioned at the start of this tread): When a caller uses overlap sending (e.g from a POTS line) instead of block dialling (as from my mobile phone) I'll usually lose the DID information. You are using chan_capi 0.3.5? I didn't remember that. Then you should forget all I wrote, because 0.3.5 is very different to current stable version. Armin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with HT 488 FXO
Soner Tari wrote: Hi Pasqualotto, Actually, I've seen your post on Asterisk-Users list yesterday, but I could not understand back then. Now, I've checked your sip configuration again, I think you make a mistake in type of sip account. I use friend not peer. I am not sure though. Ok, thanks, now with my new type the call from FXO (300) are correctly forwarded to my extension (204) after n second. Now I have another problem: I want that the calls from 300 to 204 are redirected to my ring-group. With [EMAIL PROTECTED] Inbound routing I have add these lines in extension.conf: -- cut --- [ext-did] include = ext-did-custom exten = s/204,1,SetVar(FROM_DID=s/204) exten = s/204,2,Goto(ext-group,1,1) exten = _X./204,1,Goto(s/204) [ext-group] include = ext-group-custom exten = 1,1,Macro(rg-group,ringall,60,,201-202-203-204-205-206) exten = 1,2,Goto(ext-group,1,1); jump -- cut - The calls from context from-pstn (SIP account) is also redirected to ring-group and these work. I found this in Asterisk CLI: -- Executing Macro(SIP/300-3bb9, exten-vm|novm|204) in new stack -- Executing Macro(SIP/300-3bb9, user-callerid) in new stack -- Executing DBget(SIP/300-3bb9, AMPUSER=DEVICE/300/user) in new stack -- DBget: varname=AMPUSER, family=DEVICE, key=300/user -- DBget: set variable AMPUSER to 300 -- Executing DBget(SIP/300-3bb9, AMPUSERCIDNAME=AMPUSER/300/cidname) in new stack -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=300/cidname -- DBget: set variable AMPUSERCIDNAME to ht488 -- Executing GotoIf(SIP/300-3bb9, 0?5) in new stack -- Executing SetCallerID(SIP/300-3bb9, ht488 300) in new stack -- Executing NoOp(SIP/300-3bb9, Using CallerID ht488 300) in new stack -- Executing SetVar(SIP/300-3bb9, FROMCONTEXT=exten-vm) in new stack -- Executing Macro(SIP/300-3bb9, record-enable|204|IN) in new stack -- Executing GotoIf(SIP/300-3bb9, 0 0?2:4) in new stack -- Goto (macro-record-enable,s,4) -- Executing AGI(SIP/300-3bb9, recordingcheck|20060228-133504|1141151704.8) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20060228-133504|1141151704.8: Inbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing NoOp(SIP/300-3bb9, No recording needed) in new stack -- Executing Macro(SIP/300-3bb9, dial|15|tr|204) in new stack -- Executing GotoIf(SIP/300-3bb9, 0?4:2) in new stack -- Goto (macro-dial,s,2) -- Executing GotoIf(SIP/300-3bb9, 0?5:4) in new stack -- Goto (macro-dial,s,4) -- Executing AGI(SIP/300-3bb9, dialparties.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi -- dialparties.agi: priority = 4 -- dialparties.agi: callingani2 = 0 -- dialparties.agi: accountcode = -- dialparties.agi: channel = SIP/300-3bb9 -- dialparties.agi: callerid = 300 -- dialparties.agi: context = macro-dial -- dialparties.agi: callington = 0 -- dialparties.agi: dnid = 204 -- dialparties.agi: request = dialparties.agi -- dialparties.agi: calleridname = ht488 -- dialparties.agi: extension = s -- dialparties.agi: language = en -- dialparties.agi: uniqueid = 1141151704.8 -- dialparties.agi: callingpres = 0 -- dialparties.agi: type = SIP -- dialparties.agi: rdnis = unknown -- dialparties.agi: callingtns = 0 -- dialparties.agi: enhanced = 0.0 dialparties.agi: Caller ID name and number are '300' dialparties.agi: Methodology of ring is 'none' -- dialparties.agi: Added extension 204 to extension map -- dialparties.agi: Extension 204 cf is disabled -- dialparties.agi: Extension 204 do not disturb is disabled -- dialparties.agi: Checking CW and CFB status for extension 204 == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 -- dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS == Manager 'admin' logged off from 127.0.0.1 dialparties.agi: Extension 204 is available...skipping checks -- dialparties.agi: DbSet CALLTRACE/204 to 300 -- AGI Script dialparties.agi completed, returning 0 -- Executing Dial(SIP/300-3bb9, SIP/204|15|tr) in new stack -- Called 204 -- SIP/204-1d0a is ringing -- SIP/204-1d0a answered SIP/300-3bb9 -- Attempting native bridge of SIP/300-3bb9 and SIP/204-1d0a -- Started music on hold, class 'default', on channel 'SIP/300-3bb9' -- Stopped music on hold on SIP/300-3bb9 == Spawn extension (macro-dial, s, 10) exited non-zero on 'SIP/300-3bb9' in macro 'dial' == Spawn extension (macro-exten-vm, s, 4) exited non-zero on 'SIP/300-3bb9' in macro 'exten-vm' == Spawn extension (from-internal, 204, 1) exited non-zero on 'SIP/300-3bb9' -- Executing
