[Asterisk-Users] UK Patches for Asterisk 1.2
As I am trying to compile a fresh copy of the current svn release of Asterisk 1.2 for a UK system with a combination of X100 and TDM cards, can a kind soul email me the CLID patches for 1.2? Many thanks Vassilis ___ --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] UK Patches for Asterisk 1.2
Thanks Tzafrir, But those patches do not work with the current 1.2. The asterisk_uk patch fails on 3 accounts with chan_zap.c and callerid.h The zaptel patch seems to be ok. Vassilis At 09:00 04/12/2005, you wrote: On Sun, Dec 04, 2005 at 08:14:07AM +, Vassilis Konstantinou wrote: As I am trying to compile a fresh copy of the current svn release of Asterisk 1.2 for a UK system with a combination of X100 and TDM cards, can a kind soul email me the CLID patches for 1.2? As announced before in this list: try http://www.lusyn.com/asterisk/patches.html However it is currently not included in my packages as it seems to conflict with current version of bristuff. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] BT / X100P impedance matching
Steve, The X100P card works ok in UK (I have 3 at the moment). The only problem I encountered with it was when I had my SKY box connected to the same line. This caused random hangups. Apart from that the card works ok and the UK callerid patch is fine for detecting the BT ids. I hope this helps Vassilis At 13:11 16/07/2005, you wrote: I understand that the X100P card is matched to a 600 ohm impedance but the UK BT phone system is not (I haven't been able to find much information on the impedance of the UK system). Has anyone come up with an easy way to match the impedance between the two so the X100P can work in the UK? Presumably a simple transformer won't do the job since it won't pass the DC components? -- - SteveXMPP/Jabber: [EMAIL PROTECTED]Web: http://www.nexusuk.org/ Servatis a periculum, servatis a maleficum - Whisper, Evanescence ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CallerID for UK
Hmmmyes but last time I played with my FXO module on the TDM400 could not detect hangup properly (that is on a London BT line). Has this been fixed? I keep an eye on the CVS but I have not seen any fixes for that. Maybe I missed it. Vassilis Well, the official line is as Mr. Spencer has made in that bugtracker entry... Digium sell the TDM400P which supports polarity detection. CVS supports UK CallerID on that card. Digium no longer sell the X100P so it's not supported any more. The X100P is a fairly crappy choice for the UK since it has a hardcoded 600ohm impedance, suitable really only for the USA... But yes, 'it was only £10 on eBay' been there done that, wasted hours playing with txgain/rxgain/echo cancellation... :) Cheers, Gavin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Detecting shorter hangup tone (UK)
Hello everybody, I recently got a new phone line from Bulldog-CW (UK). Needless to say that I have connected phone line to my Asterisk system. All seems ok but it does not detect hangups. When the caller hangs-up, the Bulldog line gives a continues tone for a few seconds and then it goes silent. I think that the problem is that the tone does not stay on for long enough (compared to say the BT tone) Does anybody now which settings I need to use/change for a X100P to detect it correctly? For info my system uses 3 X100Ps and the one connected to a BT(UK) line correctly detects CLI and Hangups (using the usual UK patches) and the one connected to the Bulldog line detects CLI correctly but it gets confused with the Hangupi.e. does not detect it! :-) Best regards Vassilis ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P random hangups - Please help with suggestions
Just to answer my own question so that this loop is closed and hopefully others who suffer from the same problem may find this useful. The problem in the end had nothing to do with Asterisk of the X100P or at least not directly. After pulling my hair for months and checking various bits of code, drivers and hardware, I realised that the problem was cause by a stupid desktop digital satellite box (UK SKY)connected in parallel (well ...same line other socket) with the X100Ps. The box has a modem and has to phone 'the mother ship' every so often for billing purposes mainly. God knows what it does to the line in between but my suspicion is that the combination of impedance of the two was the main cause of the random hangups. As soon as I removed it from the socket, the problem disappeared! So if you get random hangups with X100Ps and you have other devices connected directly on the main line, check those as well. Many thanks to all the people who contributed ideas. Vassilis At 10:18 09/01/2005, you wrote: Thanks for the reply Bill. I am aware of the interrupts problem. To solve it I have already disabled my serial ports freeing up interrupts 3 and 4 and these are allocated to the two cards. This was done 2 months ago and has not solved the problem. Is there any way that something can wake up every now and then and generate these two interrupts? My current /proc/interrupts is as follows: CPU0 0: 185392655 XT-PIC timer 1: 13 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 1853793865 XT-PIC wcfxo 4: 1853787231 XT-PIC wcfxo 8: 2 XT-PIC rtc 9: 0 XT-PIC acpi 10: 0 XT-PIC Intel 82801BA-ICH2 12: 57674469 XT-PIC eth0, PS/2 Mouse 14:209 XT-PIC ide0 15: 10619 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 The wcfxo's are clearly allocated their own INT but can something else mess-up with these interrupts? Vassilis At 09:54 09/01/2005, you wrote: Both of the X100Ps seem to randomly hang-up both incoming and outgoing calls. I think most people who use X100P cards (clone or originals) have had your experience. So far as I can tell, the cause is always an interrupt problem. Specifically that affected X100P cards share an interrupt with one or more other devices. Have you checked for shared interrupts using the command: cat /proc/interrupts to see if any interrupts are shared? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] X100P random hangups - Please help with suggestions
This one is driving me crazy. So any suggestions will be very welcome. My setup: Suse Linux 9.0 (Pentium 4, 1GB) Asterisk: current stable (1.0.3?) - tried the head CVS before Christmas but did not fix it 2 X100P clones - one for a UK BT line, one connected to an ATA186 configured for a UK BT Broabband-Voice service (MGCP) 1 ATA186 (SIP) connected to two dect internal phones (configured as extensions 5000-5001) The problem: Both of the X100Ps seem to randomly hang-up both incoming and outgoing calls. There is no fixed dureation but it always happens. Sometimes as soon as a call is answered and sometimes at any point up to 10-15 minutes. All calls through my true VOIP lines (I use sipcall in UK and fwd) are fine and never disconnect during the call. The X100Ps seem to detect the real hangup properly (of course). Things I have tried: 1) The latest CVS (up to early December). No change 2) The current stable. No change 3) Playing with the rxgain in the zapata.conf file (no change) 4) Using the Loopstart instead of Kewlstart. No false hangups here BUT as expected lots of line noise. Is this a good clue to what is happening? Are there any parameters I can tweak to make the Kewlstart driver a bit more reliable? Please help. This is driving me (and the people using the system crazy). Vassilis My current zapata.conf is attached below: ;; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; group = 1 language=en ; ; Default context ; context=incoming switchtype=national usedistinctiveringdetection=no useincomingcalleridonzaptransfer=yes rxwink=300 usecallerid=yes hidecallerid=no callwaiting=no usecallingpres=no callwaitingcallerid=no threewaycalling=no transfer=yes cancallforward=no callreturn=no echocancel=yes echocancelwhenbridged=yes echotraining=yes relaxdtmf=yes rxgain=0.0 txgain=1.5 ; ; Specify whether the channel should be answered immediately or ; if the simple switch should provide dialtone, read digits, etc. ; immediate=no ;musiconhold=default callprogress=no progzone=uk ; usecallerid = yes ; we want Caller*ID support cidsignalling = v23 ; UK (BT) Caller*ID uses the V.23 std cidstart = history ; use the Zaptel history (X100P) busydetect=no signalling=fxs_ks channel = 1 ;BT Broadband Voice - Uses US ID and busysignal on Hangup busydetect=yes busycount=6 cidstart = ring ; ring starts Caller*ID cidsignalling = bell ; Cid US signalling=fxs_ks channel = 2 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P random hangups - Please help with suggestions
Thanks for the reply Bill. I am aware of the interrupts problem. To solve it I have already disabled my serial ports freeing up interrupts 3 and 4 and these are allocated to the two cards. This was done 2 months ago and has not solved the problem. Is there any way that something can wake up every now and then and generate these two interrupts? My current /proc/interrupts is as follows: CPU0 0: 185392655 XT-PIC timer 1: 13 XT-PIC keyboard 2: 0 XT-PIC cascade 3: 1853793865 XT-PIC wcfxo 4: 1853787231 XT-PIC wcfxo 8: 2 XT-PIC rtc 9: 0 XT-PIC acpi 10: 0 XT-PIC Intel 82801BA-ICH2 12: 57674469 XT-PIC eth0, PS/2 Mouse 14:209 XT-PIC ide0 15: 10619 XT-PIC ide1 NMI: 0 LOC: 0 ERR: 0 MIS: 0 The wcfxo's are clearly allocated their own INT but can something else mess-up with these interrupts? Vassilis At 09:54 09/01/2005, you wrote: Both of the X100Ps seem to randomly hang-up both incoming and outgoing calls. I think most people who use X100P cards (clone or originals) have had your experience. So far as I can tell, the cause is always an interrupt problem. Specifically that affected X100P cards share an interrupt with one or more other devices. Have you checked for shared interrupts using the command: cat /proc/interrupts to see if any interrupts are shared? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] X100P random hangups - debug info
OK Some more information. I have changed interupts for the 2 X100Ps and the problem did not go away. Looking at the detailed debug information I always get the following from before the disconnect: Jan 9 12:13:40 DEBUG[1992]: Exception on 17, channel 1 Jan 9 12:13:40 DEBUG[1992]: Got event On hook(1) on channel 1 (index 0) Jan 9 12:13:40 DEBUG[1992]: disabled echo cancellation on channel 1 Jan 9 12:13:40 DEBUG[1992]: Didn't get a frame from channel: Zap/1-1 Jan 9 12:13:40 DEBUG[1992]: Bridge stops bridging channels SIP/5001-2b20 and Zap/1-1 Jan 9 12:13:40 DEBUG[1992]: Hangup: channel: 1 index = 0, normal = 17, callwait = -1, thirdcall = -1 Jan 9 12:13:40 DEBUG[1992]: disabled echo cancellation on channel 1 Jan 9 12:13:40 DEBUG[1992]: Set option TDD MODE, value: OFF(0) on Zap/1-1 === Zap1 is the UK BT line. SIP/5001 is one of the extensions connected to the Cisco ATA186. Can somebody help me with what is happening with the On hook event? Is there something I could try to not drop the line? Vassilis ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK X100P CID patch for the latest CVS
Is it possible for someone to post the UK CID patch for the latest CVS. The one I have fails to patch channels/chan_zap.c? Many thanks Vassilis ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on SuSE 9.1?
I compiled version 1.0.3 over teh weekend on a Suse 9.1 box. It was a clean installation straight out of the SUSE cds. Make sure that the kernel sources are loaded and that you do a full online update before you proceed. Asterisk compiles without any problems. Vassilis At 17:44 13/12/2004, you wrote: I am trying to do my first asterisk install on a SuSE 9.1 box, using the asterisk-update script mentioned a few days ago on this list. I did read the 'quickstart' document on onlamp.com, and made sure the following packages were installed via yast: bison, cvs, gcc, kernel-source, libtermcap-devel, ncurses-devel, newt-devel, openssl096b, and openssl-devel. The SuSE 9.1 DVD contained openssl-0.9.7d-15. I hope that's compatible, since its 'later' than 096b, right? ...there was a termcap package, which contains the termcap libraries, but no libtermcap-devel. If there are header files necessary in addition to the libraries, does someone know where I can obtain them packaged for SuSE 9.1? After pulling down all the source from CVS, the script begins compiling. It finds a 2.6 kernel, and tells me that kernel sources aren't necessary with 2.6 kernels. It then proceeds to compile several modules, but quits when the compile of zttest returns a message telling me that I need kernel sources in order to compile. Can someone point me in the right direction? -- Rick Green They that can give up essential liberty to obtain a little temporary safety, deserve neither liberty nor safety. -Benjamin Franklin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Can a TDM21 and a X100P co-exist
Well.. subject says it all really. I have a TDM with 2 FXS modules and 1 FXO and a X100P. If I load teh zaptel and wctdm drivers. Asterisk sees the TDM ports fine but not the X100P I have tried several combinations of port numbering but can some kind person with a similar setup to send me the correct zaptel.conf, zapata.conf and which drivers to load with modprobe? many thanks Vassilis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK callerid Patch for the latest CVS
Hello! Is it possible for somebody to email me the patch for the UK callerid (for the X100P cards). I know that the TDM100P patches are included but .. I still use the X100Ps Many thanks Vassilis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] False Hangups on Asterisk
I have the same problem. My setup is: Suse 8.0 with 2 single FXO cards with 2 lines coming in and 1 ATA186 providing the connection to two analogue phones. What I get is an almost random hung up during calls (both incoming and dialed). Sometimes tha call can last for 30 minutes without any problems sometimes 1min. In all cases the phone switches to the busy tone and the caller hears nothing, The line does not hang-up though. I tried various tests trying to pin down the problem but I was unsuccessful so far. My latest theory was that this may happen when the ATA re-registers with Asterisk but as you dont have any ATAs it seems that my theory is wrong :-( Does anybody else have similar problems or knows a procedure to try in order to identify the source of the problem? Vassilis At 21:43 19/08/2004, you wrote: I have an asterisk server running on Redhat 8.0 with a Digium TDM400P w/4 FXO modules (TDM04P) There are 2 lines going into the Digium card. One line is a Vonage digital line, and the other line is a Comcast voice line. I have a SIP Grandstream 100 phone connected to the Asterisk server. I also have IAX configured with FWD. The problem is that on occasionally, after talking for about 20 minutes or so, the call gets hung up and I get a fast paced busy signal. The caller gets dead air. So far this has happened on incoming calls to either the Comcast Line, the Vonage Line and it just happened while makine and 800 call on Freeworlddialup. The problem occurs on both lines for incoming calls and it just happened again today on an outgoing call after 15 minutes of talk. I have tried busydetect=no and yes and neither one make a difference. suggestions? /etc/asterisk/zapata.conf busydetect=yes busycount=10 -- Ruben Fagundo NPV Corporation Tel: 617-848-0890 x100 Fax: 617-249-1994 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help! BT UK has striken again ! I lost CLID!
I am a very happy Asterisk user/installer for the last couple of months. I regularly compile teh latest CVS. I got RC1 working and I as using the UK patches for the BT caller ID with no problems ...that is until this morning :-( I have an Asterisk X100 card going through the master socket of an engineer installed ADSL connection. Up to this morning all was trouble free. Then BT upgraded my 512K connection to 1mb. Everything seems to be working ok apart from the caller ID identification. In the past Asteriks used to detect it properly but now I get: NOTICE[294930]: callerid.c:245 callerid_feed: Caller*ID failed checksum I have seen in the past some people reporting some twicks for the rtgain to resolve this but I can't find what I need to change. Can somebody in a similar position help? Vassilis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Help! BT UK has striken again ! I lost CLID!
Thanks for the reply Kevin, The loss of Caller ID happened while i was using a properly patched RC1 without rebooting. All calls before the ADSL upgrade were displayed with their correct Caller ID. Then the ADSL upgrade came... it was very obvious when it happened because I lost INTENET connectivity and then when my router resynced I had 1mb transfers. Since then though all calls give me the Caller ID error. A caller id display unit connected directly to the BT line reports the CLID properly so it is transmitted. I tried today's CVS with the UK patch but the result is the same. Everything else seems to be ok. Vassilis At 10:24 22/07/2004, you wrote: Vassilis Konstantinou [EMAIL PROTECTED] wrote: I have an Asterisk X100 card going through the master socket of an engineer installed ADSL connection. Up to this morning all was trouble free. Then BT upgraded my 512K connection to 1mb. Everything seems to be working ok apart from the caller ID identification. In the past Asteriks used to detect it properly but now I get: NOTICE[294930]: callerid.c:245 callerid_feed: Caller*ID failed checksum I have seen in the past some people reporting some twicks for the rtgain to resolve this but I can't find what I need to change. Can somebody in a similar position help? Apart from the BT changes, I know that you also upgraded Asterisk. Are you sure you applied all of the patches with no patch failures? You would have needed to install both the Zaptel and Asterisk patch sets and would have needed to restart everything. Also, if you switched from an old patch set to a new one then you would need to remove ukcallerid = yes from your zapata.conf file and change usecallerid = yes to usecallerid = uk. That depends upon the age of your previous patches, of course. Double-check everything. Also, plug a phone, with Caller*ID support, into your BT line and verify that you are being sent the ID. Perhaps BT switched it off by mistake, or there's a problem on their part. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Using call redirection numbers
Hello everybody, I am trying to setup asterisk to redirect international calls via a carrier which uses a fixed price tel number. The scenario is Dial 087..something (UK number) Pause for answer at the other end Send required telephone number 003..etc followed by # What is the easiest way of doing this? I have trouble with both the pause and adding the # at the end of the number. Best regards Vassilis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Why? oh why can't I dial out?
I have been struggling with my Asterisk setup for 3 days now and I think I
have done well...apart from the small detail that I cannot dial out on my
phone (PSTN) line.
My setup is:
Suse Linux 9.0
1 fxo card connected to a BT(UK) line
1 Cisco ATA186 sip v3.0 with two analogue phones attached to it
Asterix CVS-HEAD-05/30/04-06:56:31
with the UK Userid patch applied. Asterisk loads without any warnings.
The problem?
I can receive calls with the userid reported correctly. I can forward them
to the two SIP ATA lines. I can dial internally (between the two phones) BUT
I cannot dial out :-(
I have tried everything (and yes I searched the world using google but
nothing seems to apply to my case). So can somebody please direct me to
possible causes.
The scenario is: if I dial 9123 (for the UK clock) then output from the
console is:
-- Executing Dial(SIP/5000-96f1, Zap/1/123) in new stack
-- Called 1/123
-- Zap/1-1 answered SIP/5000-96f1
-- Hungup 'Zap/1-1'
SIP/5000 is one of my ATA phones
ZAP/1 is the fxo card
The call is transferred to Zap/1 as I can hear the dial tone but then
nothing happens (it does not dial 123). It just stays on the tone until it
times out. I also tried pressing buttons on my ATA phone but nothing is
transferred. HELP!
Here is a collection of my conf files:
zaptel.conf
fxsks=1
loadzone=uk
defaultzone=uk
---
zapata.conf
[channels]
;
; Default language
;
language=en
;
; Default context
;
;context=default
context=incoming
switchtype=national
signalling=fxs_ks
usedistinctiveringdetection=no
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=no
transfer=no
ancallforward=no
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
relaxdtmf=yes
rxgain=0.0
txgain=4.0
immediate=no
musiconhold=default
busydetect=no
callprogress=no
usecallerid=uk
channel = 1
part of extensions.conf
[incoming]
exten = s,1,SetCallerId(${CALLERID})
exten = s,2,dial(SIP/5000SIP/5001,10,tr)
exten = s,3,Voicemail,u1000
exten = s,102,Voicemail,b1000
exten = _9.,1,Dial(${CONSOLE}/${EXTEN:1})
exten = _9.,2,Congestion
Vassilis
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Re: [Asterisk-Users] Why? oh why can't I dial out?
Thanks for the reply Greg,
The definition for the console is
[globals]
;CONSOLE=Console/dsp; Console interface for demo
CONSOLE=Zap/1
so if I am mistaken I have commented out the dsp and I am using Zap/1 the
X100P card. Is this ok?
the clock is 123 so dialing 9123 should get me there.
Best regards
Vassilis
At 17:12 27/06/2004, you wrote:
So, assuming that calls from your SIP device are in the same context as
the above extensions, all extensions beginning with a 9 should be dialled
on ${CONSOLE}. On my box, ${CONSOLE}=console/dsp... the sound card. Is
yours set to something similar (or is it really set to dial the zap
interface?)
Not being from the UK myself, I don't know whether the clock's number is
123 or 9123. If it's 9123, then you should be dialing 99123 in order to
get through your dialplan with the 9123 still intact to send to the PSTN.
Greg
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Re: [Asterisk-Users] Compiling under Suse 9.0
Just to answer my own question (for the benefit of future generations :-) My setup needed installation of anything that had to do with postgresql, and webdot (for the docs). Now it has compiled ok and so far looks ok. Vassilis At 00:01 30/05/2004, you wrote: Hello Everybody... probably this is an FAQ item but I can't find it anywhere so here it goes. I am trying to compile the latest CVS release of Asterisk under Suse 9.0 zaptel and libpri compile without any problems but when I do a 'make install' in asterisk, it compiles ok until it reaches the cdr directory. There I get a number of error messages mainly whenever CONNECTION_BAD and CONNECTION_OK appear. Does anybody know what I am doing wrong? Or if there is something else I need to compile/include before trying the asterisk directory? Many thanks Vassilis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Compiling under Suse 9.0
Hello Everybody... probably this is an FAQ item but I can't find it anywhere so here it goes. I am trying to compile the latest CVS release of Asterisk under Suse 9.0 zaptel and libpri compile without any problems but when I do a 'make install' in asterisk, it compiles ok until it reaches the cdr directory. There I get a number of error messages mainly whenever CONNECTION_BAD and CONNECTION_OK appear. Does anybody know what I am doing wrong? Or if there is something else I need to compile/include before trying the asterisk directory? Many thanks Vassilis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
