[Asterisk-Users] A-law distortion
Hi, I set alaw = 1-7 in /etc/zaptel.conf hoping to make my zap channels the same as the PSTN. This caused the levels to be about 20dB too high, as well as being distorted. I adjusted the txgain and rxgain settings in /etc/asterisk/zapata.conf to sane levels, but now I still have rather distorted zap extension - zap extension sound quality. What am I missing here ? zap 1-4 are phone extensions, and zap 5-7 are telco lines. At the moment I'm not even looking at the telco lines. Many thanks, Thomas ___ 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] Handling -1 in dialplans
Hi,
How do you handle the case where a module returns -1 ?
eg consider this:
exten = 123,1,Answer
exten = 123,n,Playback(some-message)
exten = 123,n,etc ...
exten = 123,n,etc ...
exten = 123,n,etc ...
exten = 123,n,Command(${SOME_PARAMETER})
Now what if command returns -1 here ? I would like to branch
accordingly. Also how do you handle jumping to n+101 here - you don't
know what n is ?
Thanks,
Thomas
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[Asterisk-Users] How do you prevent a 3-way conference if an extension is busy ?
If I put an incoming PSTN call on hold, and then dial another extension, which busy, I want to be able to use the hook-switch to cancel the enquiry call, so that I can go back to the original incoming call. Currently if I do this the incoming call gets connected to busy tone and I lose control of the call. How can I prevent this behaviour ? ___ 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] How do you transfer a call to a busy extension ?
Hi, How do you transfer (using say blind transfer) a call to an extension that is currently busy on another call? You don't want the call to be transferred to voicemail, it must stay in 'hold' until the extension becomes available, and then immediately ring that phone. Thanks, Thomas ___ 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] DNID empty on incoming calls
Hi, I see others have had this problem. Is there a solution ? I have a BRI, using zaphfc. If I enable debugging so: bri debug span 1 and then make an incoming call I can see that the DNID info is definitely provided by the PSTN - Here's proof: Called Number (len= 7) [ Ext: 1 TON: Unknown Number Type (0) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '1842' (The '1842' is the part of my telephone number that I'm looking for) But how can I get hold of this info - DNID is empty ? -- Executing NoOp(Zap/1-1, DIALEDPEERNAME = ) in new stack -- Executing NoOp(Zap/1-1, DIALEDPEERNUMBER = ) in new stack -- Executing NoOp(Zap/1-1, DIALEDTIME = ) in new stack -- Executing NoOp(Zap/1-1, DIALSTATUS = ) in new stack -- Executing NoOp(Zap/1-1, DNID = ) in new stack -- Executing NoOp(Zap/1-1, EXTEN= s) in new stack -- Executing NoOp(Zap/1-1, RDNIS= ) in new stack Many thanks Thomas ___ 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] b0rked hfc config
I have 2 Billion cards and I can't get the hfc driver to work. I get this error: ZT_CHANCONFIG failed on channel 9: No such device or address (6) What am I doing wrong ? ztcfg -vvv gives me this: 88--8- Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) SPAN 2: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 09: Individual Clear channel (Default) (Slaves: 09) Channel 10: Individual Clear channel (Default) (Slaves: 10) Channel 11: D-channel (Default) (Slaves: 11) Channel 12: Individual Clear channel (Default) (Slaves: 12) Channel 13: Individual Clear channel (Default) (Slaves: 13) Channel 14: D-channel (Default) (Slaves: 14) 6 channels configured. ZT_CHANCONFIG failed on channel 9: No such device or address (6) 88--8- This is my /etc/zaptel.conf: span=1,1,3,ccs,ami bchan=9-10 dchan=11 span=2,1,3,ccs,ami bchan=12-13 dchan=14 loadzone = us defaultzone=us (I'm using 1.07 of the zaptel driver with bristuff-0.2.0-RC8 and the patches from that package) ___ 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] b0rked hfc config
On Wed, Apr 27, 2005 at 11:08:46AM +0200, Thomas Andrews wrote: I have 2 Billion cards and I can't get the hfc driver to work. I get this error: ZT_CHANCONFIG failed on channel 9: No such device or address (6) What am I doing wrong ? This is my /etc/asterisk/zaptel.conf: [channels] switchtype = euroisdn signalling = bri_cpe_ptmp pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context = incoming channel = 9 channel = 10 channel = 12 channel = 13 ___ 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] semantics terminology
Hi, Where can I get a good reference for terminology terms. For instance I'd like to know if there's an accepted difference between the terms divert and forward. Thanks Thomas ___ 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] ignorepat changing the sound of dialtone
On Sun, Apr 10, 2005 at 09:47:36AM -0500, Andy Hamilton wrote: On Apr 10, 2005 7:30 AM, Thomas Andrews [EMAIL PROTECTED] wrote: Is it possible to play a different dialtone as soon as a user dials say '0' for an outside line ? Ignorepat is an inadequate solution because local users are accustomed to getting a specific PSTN dialtone. I need an audible change in the frequency/modulation of the tone. This depends on what kind of phone you are using. Sorry - With standard POTS phones on a Digium TDM FXS interface. Thanks, Thomas ___ 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] ignorepat changing the sound of dialtone
Howdie folks, Is it possible to play a different dialtone as soon as a user dials say '0' for an outside line ? Ignorepat is an inadequate solution because local users are accustomed to getting a specific PSTN dialtone. I need an audible change in the frequency/modulation of the tone. Thanks, Thomas ___ 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] capi segfault when incoming call is answered
I have a Fritz! card set up to use capi, however when incoming calls to
the card are answered, asterisk segfaults. Here is the output of gdb:
#0 0x4014f7af in memcpy () from /lib/tls/libc.so.6
#1 0x081316b0 in ?? ()
#2 0x08130680 in ?? ()
#3 0x40432da6 in pipe_msg (PLCI=257, CMSG=0x405052c0) at chan_capi.c:1560
#4 0x40436f90 in capi_handle_msg (CMSG=0x101) at chan_capi.c:2379
#5 0x404362f7 in do_monitor (data=0x0) at chan_capi.c:2404
#6 0x400229b4 in start_thread () from /lib/tls/libpthread.so.0
#7 0x in ?? ()
The problem is at line 1560 in chan_capi.c:
memcpy(b3buf[AST_FRIENDLY_OFFSET],(char
*)DATA_B3_IND_DATA(CMSG),DATA_B3_IND_DATALENGTH(CMSG));
I'm using chan_capi-0.3.5 with the patch from
http://www.levigo.de/VoIP/chan_capi-0.3.5-cvs-HEAD-patch.tar.bz2
(A similar thing happens when I make outgoing calls via the Fritz! card
- it segfaults as soon as the phone on the other end starts ringing)
Here's a bit more info from gdb:
8-8---8--
#0 0x4014f7af in memcpy () from /lib/tls/libc.so.6
No symbol table info available.
#1 0x081316b0 in ?? ()
No symbol table info available.
#2 0x08130680 in ?? ()
No symbol table info available.
#3 0x40432da6 in pipe_msg (PLCI=257, CMSG=0x405052c0) at
chan_capi.c:1560
p = (struct capi_pipe *) 0x1e
CMSG2 = {ApplId = 1, Command = 131 '\203', Subcommand = 131
'\203', Messagenumber = 202, adr = {adrController = 131585,
adrPLCI = 131585, adrNCCI = 131585}, AdditionalInfo = CAPI_COMPOSE,
B1configuration = 0x0, B1protocol = 0, B2configuration = 0x0,
B2protocol = 0, B3configuration = 0x0, B3protocol = 0, BC = 0x0,
BChannelinformation = 0x0, BProtocol = CAPI_COMPOSE,
CalledPartyNumber = 0x0, CalledPartySubaddress = 0x0,
CallingPartyNumber = 0x0, CallingPartySubaddress = 0x0, CIPmask = 0,
CIPmask2 = 0, CIPValue = 0, Class = 0, ConnectedNumber = 0x0,
ConnectedSubaddress = 0x0, Data32 = 0, Data64 = 0, DataHandle = 0,
DataLength = 0, FacilityConfirmationParameter = 0x0, Facilitydataarray
= 0x0, FacilityIndicationParameter = 0x0,
FacilityRequestParameter = 0x0, FacilityResponseParameters = 0x0,
FacilitySelector = 0, Flags = 0, Function = 0, HLC = 0x0, Info = 0,
InfoElement = 0x0, InfoMask = 0, InfoNumber = 0, Keypadfacility = 0x0,
LLC = 0x0, ManuData = 0x0, ManuID = 0, NCPI = 0x0, Reason = 0,
Reason_B3 = 0, Reject = 0, Useruserdata = 0x0, SendingComplete = 0x0,
Data = 0xc Address 0xc out of bounds, l = 1, p = 1078220081,
par = 0x40449700 \f, m = 0x0, buf = '\0' repeats 179 times}
error = 160
fr = {frametype = 4, subclass = 4, datalen = 0, samples = 128,
mallocd = 192, offset = 0, src = 0x4020aebc Ä}\023, data = 0x41,
delivery = {tv_sec = 135278508, tv_usec = 135457433}, prev =
0x818b820, next = 0x1}
b3buf = [EMAIL PROTECTED], '\0' repeats 36 times,
[EMAIL PROTECTED]
@[EMAIL PROTECTED]@[EMAIL PROTECTED]@, '\0' repeats 16 times,
N¢C@, '\0' repeats 60 times, \n¼\037@, '\0' repeats 16 times,
¼® @[EMAIL PROTECTED]@[EMAIL PROTECTED] @`UP@/[EMAIL PROTECTED]@[EMAIL
PROTECTED]
@[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL PROTECTED]@[EMAIL
PROTECTED]@`UP@...
j = 30
b3len = 0
dtmf = 30 '\036'
dtmflen = 1079005888
rxavg = 0
txavg = 0
8-8---8--
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Re: [Asterisk-Users] capi segfault when incoming call is answered
On Thu, Apr 07, 2005 at 10:15:09AM +0200, Thomas Andrews wrote: I have a Fritz! card set up to use capi, however when incoming calls to the card are answered, asterisk segfaults. Just for the record, my capi.conf looks like this: [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=1842 incomingmsn=* controller=1 softdtmf=1 accountcode= context=isdn-test devices=2 And the relevant bit in extensions.conf looks like this: [isdn-test] exten = s,1,Dial(Zap/7) Many thanks, Thomas ___ 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] :: What does it take to upgrade? :: Newbie Q ::
On Sat, Mar 19, 2005 at 10:40:10AM +0100, Reuben Grech wrote: How can I upgrade Asterisk to the latest version ?? You will have to use cvs: http://www.automated.it/guidetoasterisk.htm#_Toc49248761 Will I need to re-compile?? Yes. -Thomas ___ 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] AGI-like calls in the [globals] section
I'd like to set up some global parameters once at startup using an external program. (eg like one would with AGI) How can I do that ? Thanks, Thomas ___ 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] Channel name (and substring)
How do I get the bit like IAX2/white_phone in extensions.conf eg from
pre-defined variables or variants thereof ?
What I *do* get is strings like this IAX2/[EMAIL PROTECTED]
from ${CHANNEL}, but that's the full channel name.
What am I missing here ?
Thanks,
Thomas
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Re: [Asterisk-Users] Channel name (and substring)
On Thu, Mar 17, 2005 at 02:06:24PM -0600, B. J. Bomar wrote:
Try using the Cut application. For your example channel you can use the
following.
exten = whatever,n,Cut(my_variable=CHANNEL,@,1)
Thanks, I thought of that, but it doesn't account for cases like Zap/1
that becomes Zap/1-1 in the ${CHANNEL} variable because the convention
seems to change from '@' to '-'. It means I can't write a generic
translation.
Thanks,
Thomas
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Re: [Asterisk-Users] Channel name (and substring)
On Thu, Mar 17, 2005 at 12:24:00PM -0800, Sean Kennedy wrote:
Thomas Andrews wrote:
How do I get the bit like IAX2/white_phone in extensions.conf eg from
pre-defined variables or variants thereof ?
What I *do* get is strings like this IAX2/[EMAIL PROTECTED]
from ${CHANNEL}, but that's the full channel name.
This should help:http://www.voip-info.org/wiki-Asterisk+variables
Thanks, that's more or less what I thought I would have to do. It just
seems such a long way around to get something that I would have expected
to be 'ready-made' in some pre-defined $variable.
Thanks,
Thomas
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[Asterisk-Users] leaky reload
If I comment out the following line in zapata.conf I would expect asterisk to forget the cli information for that channel when I reload: callerid=Uniden Dead (256) 428-6125 ... but it doesn't; I have to restart asterisk for it to take effect. The funny thing is that the reverse is *not* true - ie if I uncomment the line and reload then it learns about the caller id Uniden Dead. Why is this a one-way process ? Thanks, Thomas ___ 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] skinny error
On Mon, Nov 15, 2004 at 07:32:29AM -0500, Jason p wrote: check to make sure you have a ip address added to teh skinny.conf file.. if your even using skinny. Yup, that's it. Thanks Jason. Regards, Thomas ___ 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] skinny error
Hi Darren, On Mon, Nov 15, 2004 at 02:21:27PM +, Storer, Darren wrote: In older versions of code bind = 0.0.0.0 was sufficient. I now find that you must indicate the actual IP address of the LAN card on the Asterisk server or skinny support will not startup correctly. That's exactly what I found. I just put the IP of the ethernet card in there and the error went away. Regards, Thomas ___ 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] (newbie) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 02:22:18PM +0100, Soren Rathje wrote: /etc/zaptel.conf fxols=1 #S100U fxsls=2 #X100P loadzone = us defaultzone=us Looks fine allthough the comments are wrong :-) Thanks Soren. I made all the changes you suggested, but do I have to change the above to ... fxoks=1 fxsks=2 ... if I changed to kewel-start in zapata.conf ? I assumed so, and went ahead and did so. Still no dial-tone though. Thanks, Thomas ___ 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] (newbie) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 03:01:28PM +0100, Soren Rathje wrote: Hmm.. Does Asterisk load chan_zap ? I believe so: [chan_zap.so] = (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found -- Registered channel 1, FXO Kewlstart signalling -- Registered channel 2, FXS Kewlstart signalling -- Automatically generated pseudo channel == Registered channel type 'Zap' (Zapata Telephony Driver) == Registered application 'CallingPres' == Manager registered action ZapTransfer == Manager registered action ZapHangup == Manager registered action ZapDialOffhook == Manager registered action ZapDNDon == Manager registered action ZapDNDoff == Manager registered action ZapShowChannels Thanks again! Thomas ___ 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] (newbie) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 03:16:13PM +0100, Soren Rathje wrote: Hang on... What line pair do you use on the phone; 1+4 or 2+3 ?? I believe the correct pair to use should be 2+3. It's the middle pair. I assume that's 2+3 on an RJ connector ? ___ 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] (newbie) no dialtone on a TDM400P card
I think I know what the problem is. I think that asterisk cannot generate dialtone because it had a problem with the soundcard. [chan_oss.so] = (OSS Console Channel Driver) Nov 14 16:35:49 WARNING[2078]: chan_oss.c:994 load_module: XXX I don't work right with non-full duplex sound cards XXX == Registered channel type 'Console' (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found Nov 14 16:35:49 WARNING[2091]: chan_oss.c:240 sound_thread: Read error on sound device: Resource temporarily unavailable I had to put this in modules.conf to get rid of the error: noload = chan_oss.so So I assume now that it's not capable of making dialtone ? Regards, Thomas ___ 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] (newbie) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 03:35:06PM +0100, Soren Rathje wrote: Just for verification, do you have any green led's lit on the back of your card ?? Yes, and I have tested with a different telephone and cable that I know works. ___ 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] (newbie) no dialtone on a TDM400P card
Hi Dave, On Sun, Nov 14, 2004 at 03:38:27PM +0100, Dave Cotton wrote: Try cat /proc/interrupts a number of times, do the interrupts on wctdm show an increase? They do! What also bothers me is that the interrupt is shared: 16661397 IO-APIC-level ohci1394, wctdm I have no idea what ohci1394 is. I don't have any infra-red devices connected, but I assume this (Intel) motherboard has support, hence this driver ?? Can you dial the extension from another? Watch out mine rings on the rack but nothing from the phone then if I try again it locks completely and requires a complete power reset. Sorry to be so dumb, but how would I do that ? I only have one FXS module. Or is it possible to simulate a call from the *CLI console ? Thanks, Thomas ___ 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] (newbie) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 04:06:37PM +0100, Soren Rathje wrote:
modprobe zaptel debug=1
kernel: Zapata Telephony Interface Registered on major 196
insmod wctdm debug=1
kernel: Setting FXS hook state to 0 (00)
last message repeated 3 times
kernel: Registered Span 1 ('WCTDM/0') with 4 channels
kernel: Span ('WCTDM/0') is new master
kernel: Freshmaker version: 71
kernel: Freshmaker passed register test
kernel: ProSLIC on module 0, product 0, version 5
kernel: ProSLIC on module 0 seems sane.
kernel: ProSLIC on module 0 powered up to -72 volts (c2) in 20 ms
kernel: Loop current set to 20mA!
kernel: Post-leakage voltage: 25 volts
kernel: ProSLIC on module 0 powered up to -72 volts (c0) in 10 ms
kernel: Loop current set to 20mA!
kernel: Calibration Vector Regs 98 - 107:
kernel: 98: 10
kernel: 99: 11
kernel: 100: 11
kernel: 101: 0f
kernel: 102: 07
kernel: 103: 64
kernel: 104: 09
kernel: 105: d7
kernel: 106: 07
kernel: 107: 08
kernel: Init Indirect Registers completed successfully.
kernel: Proslic module 0 loop current is 20mA
kernel: Module 0: Installed -- AUTO FXS/DPO
kernel: ProSLIC on module 1, product 0, version 0
kernel: VoiceDAA System: 04
kernel: ISO-Cap is now up, line side: 03 rev 03
kernel: Module 1: Installed -- AUTO FXO (FCC mode)
kernel: ProSLIC on module 2, product 0, version 0
kernel: Module 2: Not installed
kernel: ProSLIC on module 3, product 0, version 0
kernel: Module 3: Not installed
kernel: Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules)
kernel: NO BATTERY on 1/2!
/sbin/ztcfg
kernel: Setting FXS hook state to 0 (00)
kernel: Registered tone zone 0 (United States / North America)
kernel: Power alarm on module 1, resetting!
last message repeated 9 times
asterisk -vvvgc
kernel: Setting FXS hook state to 0 (00)
kernel: Setting FXS hook state to 0 (00)
I don't like the look of that NO BATTERY message. What do you think
Soren ?
-Thomas
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Re: [Asterisk-Users] (newbie) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 04:36:21PM +0100, Soren Rathje wrote: NO BATTERY applies to FXO ports and says that Span 1/Card 2 does not receive power from the line, i.e. it is not plugged into the wall socket. (if I read the source correctly) ok. I connected it to the PABX and I got this so I assume that ports ok Nov 14 17:49:30 zorg kernel: 63761 Polarity reversed (0 - -1) Nov 14 17:49:30 zorg kernel: BATTERY on 1/2 (-)! BTW. Does the hookstate change change is you lift the handset ?? In /var/log/messages ? nothing happens as far as I can see. ___ 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] (newbie) no dialtone on a TDM400P card
Hi Dave, On Sun, Nov 14, 2004 at 04:05:04PM +0100, Dave Cotton wrote: Firewire, either disable it from the BIOS or move your cards around, Digium cards do not like shared interrupts. Yes firewire :) I couldn't disable it in the BIOS, so I took your advice and swapped cards. Now it's not shared: 22:1043486 IO-APIC-level wctdm The interrupt count is still steadily increasing. (and asterisk isn't running at the moment.) Yes, but from your earlier post you cast doubt on the sound card. Can't you use a softphone from another machine? If there is something I can install on either linux or windows I'm happy to try. What would you suggest as the easiest softpone to install ? Thanks, Thomas ___ 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] (newbie) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 05:42:36PM +0100, Soren Rathje wrote: May I suggest you call the nearest medicine man and have him drive out the gremlin... Local tradition dictates that I slaughter the whitest goat :) Or, look for contact problems in the sockets/connectors, you may have a faulty FXS module since the FXO module and the base card seems to function as expected. I agree. I looped the fxo back to the fxs to see if battery voltage was being supplied by the fxs: it looks fine because /var/log/messages shows the change of battery state when I (un)plug it, so at least the connectors are ok. I'm going to get hold of the supplier and see if he can test the module for me. Thanks so much for your help Soren. I have *really* appreciated it! Regards, Thomas ___ 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] skinny error
What does this error mean: Nov 14 10:35:12 WARNING[24733]: Unable to get our IP address, Skinny disabled I looked in channels/chan_skinny.c and it looks like ourhost[] is never initialised ? $ ___ 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] (newbie) no dialtone on a TDM400P card
I've never set up asterisk before. I hear no dialtone on the telephone plugged into the TDM400 card. This is what ztcfg -vv gives me: Zaptel Configuration == Channel map: Channel 01: FXO Loopstart (Default) (Slaves: 01) Channel 02: FXS Loopstart (Default) (Slaves: 02) 2 channels configured. I've used this link to set up the ports: http://www.digium.com/index.php?menu=faq#Configuration_0 How do I debug this ? Is there supposed to be a log in /var/log/asterisk/event_log ? Regards, Thomas ___ 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] (newbie) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 11:19:26AM +0200, Thomas Andrews wrote: I hear no dialtone on the telephone plugged into the TDM400 card. Here's the relevant output from dmesg: Zapata Telephony Interface Registered on major 196 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Not installed Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 0 (United States / North America) Power alarm on module 1, resetting! Power alarm on module 1, resetting! Power alarm on module 1, resetting! Power alarm on module 1, resetting! Power alarm on module 1, resetting! Power alarm on module 1, resetting! Power alarm on module 1, resetting! Power alarm on module 1, resetting! Power alarm on module 1, resetting! Power alarm on module 1, resetting! ___ 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] (newbie) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 10:32:40PM +1300, Matt Riddell wrote: Power alarm on module 1, resetting! Have you plugged the power into the TDM400P? I have yes. ___ 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] (newbie) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 10:32:40PM +1300, Matt Riddell wrote: Power alarm on module 1, resetting! Have you plugged the power into the TDM400P? The wierd thing is that asterisk refuses to start *until* I've had those Power alarm error messages. Until then I get these errors: = (Zapata Telephony) Parsing '/etc/asterisk/zapata.conf': Found Nov 14 12:13:56 WARNING[1490]: chan_zap.c:774 zt_open: Unable to specify channel 1: No such device or address Nov 14 12:13:56 ERROR[1490]: chan_zap.c:6247 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Nov 14 12:13:56 ERROR[1490]: chan_zap.c:9202 setup_zap: Unable to register channel '1' Nov 14 12:13:56 WARNING[1490]: loader.c:385 ast_load_resource: chan_zap.so: load_module failed, returning -1 Unregistered channel type 'Zap' Nov 14 12:13:56 WARNING[1490]: loader.c:480 load_modules: Loading module chan_zap.so failed! ___ 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] (newbie) no dialtone on a TDM400P card
Hi Rich, On Sun, Nov 14, 2004 at 04:27:44AM -0600, Rich Adamson wrote: 1. in the /usr/src/zaptel directory, do a 'make config' Sorry, I didn't mention that I'm running debian, so that init.d script will need tweaking. I'll look at it to see if it's doing anything that I'm not. 2. execute a 'modprobe wctdm' I already have the zaptel and wctdm modules loaded. I assume that's enough ? 3. execute a 'ztcfg -vvv' I posted the results to that in my first mail. But I'm assuming that you're thinking that the init.d script may have changed something ? 3. execute 'service zaptel stop' followed by 'service zaptel start' You must be running Red Hat. On debian that's invoke-rc.d zaptel restart ... 4. start asterisk Any difference? The wierd thing is that after an arbitrary amount of time, asterisk *does* start. I have linked this with the Power alarms. It won't start before I get them, and it always does afterwards Thanks for the help! Thomas ___ 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] (newbie) no dialtone on a TDM400P card
On Sun, Nov 14, 2004 at 11:46:15AM +0100, Soren Rathje wrote: Can you post your actual configuration ? /etc/zaptel.conf fxols=1 #S100U fxsls=2 #X100P loadzone = us defaultzone=us /etc/asterisk/zapata.conf [trunkgroups] [channels] context=default switchtype=national signalling=fxo_ls rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no context=internal signalling=fxo_ls channel=1 context=incoming signalling=fxs_ls channel=2 Regards, Thomas ___ 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] (newbie) no dialtone on a TDM400P card
Hi Cirelle, On Sun, Nov 14, 2004 at 07:28:56AM -0500, Cirelle Enterprises wrote: you might have to power the box down - no power for the modules to load (appears to be common for this card) if that is the case, do a search on tdm in the email archive as there is a fix for the reboot problem I'm afraid I don't understand this sentence. I did power the machine down to make sure that the power was indeed plugged in. Are you saying there's a startup problem with these cards. module 1 (closest to the top of the bracket (furthest from the pci connector) is for the phone line module 2 is for the handset I don't agree, but perhaps I'm wrong. The green module (fxs) is number 1 and the red module (fxo) is number 2. As I understand it you plug the handset into the green one (fxs). Not so ? Thanks for the help. Thomas ___ 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] wctdm to replaces wcfxs module ?
Hi, Am I correct in saying that the wcfxs kernel module is something of the past, and is now replaced by wctdm ? Regards, Thomas ___ 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
