RE: [Asterisk-Users] Digium TheVoice recordings' sound terrible
Not that it matters, but I met Allison at Astricon and talked with her about her recording setup. All she does is voice over work. That is it. That is her job. All day long. Yep I was there too while we talked about her setup... She said her house was paid for by voiceover work she did for SafeWay Canada for a few years. www.theivrvoice.com bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with AstTapi
Don't be lazy, check the bug reports for this application - wander over to https://sourceforge.net/tracker/index.php?func=detailaid=1049761group_id=106482atid=644546 It is a known issue with build 0.04 Craig - Original Message - From: Rana Dutt [EMAIL PROTECTED] To: Asterisk Users List [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 12:51 PM Subject: [Asterisk-Users] Problem with AstTapi I wanted to use Outlook 2000 to dial my Contacts using Asterisk. So I installed AstTapi on my Windows XP machine. When I try to dial a contact, the call originates just fine. My SIP phone rings, and when I pick up, Asterisk makes the call to the dialed number correctly. However, Outlook displays an error message saying Unable to complete an operation requested by the automatic phone dialer. Please make sure your modem, phone and phone line are properly configured. After closing the error message dialog, if I then go to dial the Contact again, I get a different error message saying An internal error occurred in the phone dialer. Close the Dial Phone dialog box and then open it again. Well, closing the dialog box and opening it again doesn't work: the same internal error message keeps popping up when trying to make a call. The only way to get rid of it is to exit Outlook and restart it. Has anyone who has used AstTapi seen this problem? I am using Outlook 2000 SP3. My Asterisk TAPI driver is configured as follows: Host: 192.168.2.11 (IP of Asterisk server) Port: 5038 Dial out by using the Dial application - Outgoing chan: Zap/1/ User: john Password: my_secret User channel: SIP/200 My manager.conf is as follows: [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [john] secret = mysecret deny=0.0.0.0/0.0.0.0 permit=192.168.2.17/255.255.255.0 read = system,call,log.verbose,command,agent,user write = system,call,log.verbose,command,agent,user As I said, the first time I place the call from Outlook, it works fine. The trace on Asterisk shows: == Manager 'john' logged on from 192.168.2.17 -- Launching Dial(Zap/1/18005551212) on SIP/200-da5d -- Called 1/18005551212 == Manager 'john' logged off from 192.168.2.17 -- Zap/1-1 answered SIP/200-da5d -- Hungup 'Zap/1-1' Any help would be much appreciated. Rana Dutt Softel, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Type of T1 for T100P card
On Thu, 28 Oct 2004, Steve Underwood wrote: The original poster is asking about 2-way telephony. All the normal forms of telephony on T1 can support 2-way operation, and Asterisk supports them. However, ISDN and SS7 are more robust than the robbed bit signalled forms, like wink start. 2-way just means the same T1 can handle a mixture of incoming and outgoing calls on the same T1. With high call volumes on robbed bit signalled T1s the likelyhood of incoming and outgoing calls clashing (glare) can sometimes be unacceptable. ISDN should be rock solid under these conditions. Isdn can be totally glare free, but not in the Asterisk implementation. Asterisk treats the isdn B channels as normal channels and the D channel as a signalling channel. It allocates a B channel from what it beleives to be a free channel and sends a SETUP message indicating that channel to the network end. There is a risk that the same channel was siezed by the network which will den disconnect the outgoing call and let the incoming call through. So, there is no glare problem as long as the net and the cpe end hunt in opposite order. The collision will only happen when the last channel is contended. However, if the network and the cpe end are not set to opposite hunting order this sort of clash can occur even when not all B channels are used. Since asterisk does neither retry when the channel selection is rejected by the net nor allow the net end to dictate the channel (an option allowed by isdn to prevent glare-like rejection of outgoing calls) it is important to have the opposite ends hunt in opposite order. Just a point in case anyone experiences something resembling glare on an isdn link. Your post is also refering to telephony modes for T1s. RBS gives you all 24 channels, but it doesn't give you 24 *clear* channels. Some bits have been robbed. Most commonly ISDN gives only 23 voice channels. However, ISDN with NFAS and SS7 can give you 24 clear voice channels with Asterisk. The controlling D channel for isdn NFAS has to be delivered somewhere. With the current nfas implementation in asterisk this has to be over an e1/t1 (can it even be delivered in any other way?). This gives you 24*n-1-b B channels where n is the number of T1s and b is the number of backup D channels. I think of nfas more in terms of resiliency then more efficient use of the channels. Peter ___ 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] where do i find openssl-devel to mandrake 10.1
- Original Message - From: Brian West [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 2:17 AM Subject: RE: [Asterisk-Users] where do i find openssl-devel to mandrake 10.1 Go to www.openssl.org download the tarball and compile it. bkw I allready did that, but it does not solve the problem when i compile asterisk.. I still get this message halfway in the compiling, and af far as i can remember from when i compiled asterisk last time, and i got this message, it dissapeared when i installed openssl-devel. /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Fejl 1 Is'nt there a openssl-devel to mandrake 10.1 ? Regards Thomas H. ___ 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] Transfer caller
Give us your extensions.conf and we may be able to help you
___
Not sure if you wanted all of it but here it is with my ID's and domains
changed of course.
*
[general]
static=yes
writeprotect=no
[globals]
[incoming]
exten = s,1,Answer
exten = s,2,Background(ext-or-zero)
exten = s,3,DigitTimeout,3
exten = s,4,ResponseTimeout,30
;Operator Going to Dale for now
exten = 0,1,playback(pls-wait-connect-call)
exten = 0,2,Dial(SIP/102,25,mTt)
exten = 0,3,VoiceMail([EMAIL PROTECTED])
exten = 0,4,Goto,t|1
; 8000 - Get to Vmail
exten = 8000,1,playback(pls-wait-connect-call)
exten = 8000,2,VoiceMailMain(@mydomain.com)
exten = 8000,3,Goto,t|1
; 100 - Todd Office
exten = 100,1,playback(pls-wait-connect-call)
exten = 100,2,Dial(SIP/100,25,mTt)
exten = 100,3,VoiceMail([EMAIL PROTECTED])
exten = 100,4,Goto,t|1
; 1100 - Todd Home
exten = 1100,1,playback(pls-wait-connect-call)
exten = 1100,2,Dial(SIP/1100,25,mTt)
exten = 1100,3,VoiceMail([EMAIL PROTECTED])
exten = 1100,4,Goto,t|1
; 101 - Lewis
exten = 101,1,playback(pls-wait-connect-call)
exten = 101,2,Dial(SIP/101,25,mTt)
exten = 101,3,VoiceMail([EMAIL PROTECTED])
exten = 101,4,Goto,t|1
; 102 - Dale
exten = 102,1,playback(pls-wait-connect-call)
exten = 102,2,Dial(SIP/102,25,mTt)
exten = 102,3,VoiceMail([EMAIL PROTECTED])
exten = 102,4,Goto,t|1
; 103 - Maria
exten = 103,1,playback(pls-wait-connect-call)
exten = 103,2,Dial(SIP/103,25,mTt)
exten = 103,3,VoiceMail([EMAIL PROTECTED])
exten = 103,4,Goto,t|1
; 104 - Jim
exten = 104,1,playback(pls-wait-connect-call)
exten = 104,2,Dial(SIP/104,25,mTt)
exten = 104,3,VoiceMail([EMAIL PROTECTED])
exten = 104,4,Goto,t|1
exten = t,1,Hangup
[outgoing]
; 8000 - Get to Vmail
exten = 8000,1,playback(pls-wait-connect-call)
exten = 8000,2,VoiceMailMain(@mydomain.com)
exten = 8000,3,Goto,t|1
; 100 - Todd
exten = 100,1,playback(pls-wait-connect-call)
exten = 100,2,Dial(SIP/100,25,mTt)
exten = 100,3,VoiceMail([EMAIL PROTECTED])
exten = 100,4,Goto,t|1
; 1100 - Todd Home
exten = 1100,1,playback(pls-wait-connect-call)
exten = 1100,2,Dial(SIP/1100,25,mTt)
exten = 1100,3,VoiceMail([EMAIL PROTECTED])
exten = 1100,4,Goto,t|1
; 101 - Lewis
exten = 101,1,playback(pls-wait-connect-call)
exten = 101,2,Dial(SIP/101,25,mTt)
exten = 101,3,VoiceMail([EMAIL PROTECTED])
exten = 101,4,Goto,t|1
; 102 - Dale
exten = 102,1,playback(pls-wait-connect-call)
exten = 102,2,Dial(SIP/102,25,mTt)
exten = 102,3,VoiceMail([EMAIL PROTECTED])
exten = 102,4,Goto,t|1
; 103 - Maria
exten = 103,1,playback(pls-wait-connect-call)
exten = 103,2,Dial(SIP/103,25,mTt)
exten = 103,3,VoiceMail([EMAIL PROTECTED])
exten = 103,4,Goto,t|1
; 104 - Jim
exten = 104,1,playback(pls-wait-connect-call)
exten = 104,2,Dial(SIP/104,25,mTt)
exten = 104,3,VoiceMail([EMAIL PROTECTED])
exten = 104,4,Goto,t|1
;VoicePulse1
exten =
_1NXXNXX,1,Dial(IAX2/MyUID:[EMAIL PROTECTED]/${EXTEN})
;VoicePulse2
exten =
_1NXXNXX,1,Dial(IAX2/MyUID:[EMAIL PROTECTED]/${EXTEN})
;Local on copper line when not dialing a 1
exten = _NXXNXX,2,Dial(Zap/1/${EXTEN})
;Long distance on copper line
exten = _1NXXNXX,2,Dial(Zap/1/${EXTEN})
exten = t,1,Hangup
*
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Re: [Asterisk-Users] where do i find openssl-devel to mandrake 10.1
It is called libopenssl0.9.7-devel-0.9.7d-1mdk.i586.rpm and may be found in your actual distri or on every mirror server which hosts Mandrake 10.1 Regards Thomas Hupfeldt schrieb: - Original Message - From: "Brian West" [EMAIL PROTECTED] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 2:17 AM Subject: RE: [Asterisk-Users] where do i find openssl-devel to mandrake 10.1 Go to www.openssl.org download the tarball and compile it. bkw I allready did that, but it does not solve the problem when i compile asterisk.. I still get this message halfway in the compiling, and af far as i can remember from when i compiled asterisk last time, and i got this message, it dissapeared when i installed openssl-devel. /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Fejl 1 Is'nt there a openssl-devel to mandrake 10.1 ? Regards Thomas H. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call progress - what are the sticking points?
I have the same problem. callprogress is very experimental and buggy now. and i've lost the .call files feature of asterisk. what do you think about submitting a bug on bugs.digium.com? regards, shabanip Hello, I've been experimenting with the call progress analysis features of *, with mixed success on Zap as well as IAX channels. I've read all the posts about it, including (but not limited to) http://www.voip-info.org/wiki-Asterisk+auto-dial+out and the pages it references. My question is, what's the current state -- is there any work in progress right now to improve the reliability of * call progress detection? last I saw it was still listed as 'experimental'. What are the problems that are preventing a more robust implementation of call progress detection? Would this work better with different hardware (ie. I've had success in the past using Dialogic telephony boards)? Or is this primarily a software issue with *? Thanks much! Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] ISDN-Problem with Quadbri behind Tenovis
Hello everyone, We try to establish a * voicemail system behind a Tenovis (soon to be avaya) Integral 55 with Junghanns quadbri card in the * server. The Tenovis has 4 bri ports configured in nt ptp (edsi 61) which we connected to the quadbri (te, ptp) card. Signaling in one direction seems to work as the asterisk receives a call and seems to answer, but the Tenovis pbx never understands this and switches to 'unreachable' after a short while of ringing. Also, dialing out from an iax-phone via the zap channel results in a ringing signalled in the iax phone but no traffic to the Tenovis (level 2 indicator is alight in tenovis, but d-channel indicator stays dark). We use the bri-stuff-0.1.0-rc4a package from junghanns.net which means asterisk CVS-HEAD-08/13/04. The error asterisk shows when Tenovis dials in: -- Executing Answer(Zap/2-1, ) in new stack -- Accepting call from '7219206012' to '6951' on channel 0/2, span 1 -- Executing MP3Player(Zap/2-1, /usr/share/asterisk/sounds/pioneer.mp3) in new stack Oct 27 19:45:34 WARNING[1088519088]: chan_zap.c:6902 zt_pri_error: PRI: XXX Missing handling for mandatory IE 8 (cs0, Cause) XXX Oct 27 19:45:34 WARNING[1088519088]: chan_zap.c:8128 pri_dchannel: Hangup REQ requested on unconfigured channel 255/255 span 1 Oct 27 19:45:34 WARNING[1088519088]: chan_zap.c:8061 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1 Oct 27 19:45:39 WARNING[1088519088]: chan_zap.c:8061 pri_dchannel: Hangup requested on unconfigured channel 255/255 span 1 Any clues to what happens here? Seems the communication asterisk=Tenovis does not work. And why is the cause not handled in chan_zap? Stefan -- Stefan Märkle Netpioneer GmbH Leiter Knowledge Center Beiertheimer Allee 18 [EMAIL PROTECTED] 76137 Karlsruhe ___ 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] test telephone numbers
On Thursday 28 October 2004 04:15, Steve Totaro wrote: i think he meant numbers that would not be billed for completing a call. No, any number is just fine. Preferably a mix of mobile and fixed numbers for as many countries/regions as possible. So often a customer will say something like I've been trying to get a call through to Uzbekistan all day and nothing works, so i have to try to route Uzbekistan through a carrier who will be able to terminate it properly. Being able to test with a number that won't wake someone up at 3am would be much easier... Finding hotels or companies using an IVR system on the internet will help for landlines, but if anyone has any out of use mobile numbers that will still play a message, this would help a lot to... Thanks for the numbers and suggestions so far, Richard. Andrew Thompson wrote: Richard Bennett wrote: Hi, I was wondering if there already exists a list of worldwide test telephone numbers for us to use to test if we can terminate that destination? Hotels, restaurants and any other public place similar, searched for in google will provide you with almost unlimited opportunities. Ta SJ ___ 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] WRT54GS zaptel timing device
Hi, I'm desperately looking for more info about running Asterisk on WRT54GS. Can you please give some more info how to do this (any pointer to site, more info ...) ? How much room is there on router for software ? Thanks in advance, regards, Robert. - Original Message - From: Kristian Kielhofner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 5:38 AM Subject: [Asterisk-Users] WRT54GS zaptel timing device Hello, I know that I can run Asterisk on the Linksys WRT54GS, but can I do Zaptel as well? I would really like a timing device so I can do IAX2 trunking - but I don't know how to go about it. Has anyone done this? -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SRV lookup fails on dyndns wildcard domains
Let me add that it is not really a SRV problem but a DNS problem caused bei SRV lookup. Of course usually there are no SRVs on dyndns domains. jo jo wrote: I know that SRVs have been discussed here in different flavours but I couldn't find anything about this: When calling SIP URIs like [EMAIL PROTECTED] * fails if wildcards are enabled on that domain. Error message from * is: Oct 27 22:54:12 WARNING[1753105]: No such host: mydomain.dyndns.org If wildcards or srv lookup is disabled it works as expected. No problems at all when calling with other clients. Anyone else observed this behaviour? Any solutions? jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-cvs does not compile on Red Hat 9
Brian West wrote: Yes it does update your zaptel and your other stuff too and you'll be fine. bkw Thanks. That worked. BTW, If anyone has this problem in the future, do what the docs say and: cd zaptel; make install just running make will NOT work (as in asterisk will not compile). :\ cheers, glenn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] asterisk-oh323-0.6.3b
Hi,
First of all thanks for your fast response , but I'm getting errors again .
Look how I try to install asterisk-oh323-0.6.3b
First I emerged asterisk-1.0.0 (I'm using gentoo 2.4.25), also I saved in a
folder asterisk-1.0.0 src files (/files/asterisk-1.0.0), than I emerged
pwlib-1.6.6 and also saved the src files (extracting pwlib-1.6.6.tar.gz
/files/pwlib) , than I emerged openh323-1.13.5 (/files/openh323). In the
direcory /files I saved also the asterisk-oh323-0.6.3b. Now I edited
Makefile in asterisk-oh323-0.6.3b directory , like this:
DESTDIR=/usr/sbin/
PWLIBDIR=/files/pwlib
OPENH323DIR=/files/openh323
ASTERISKINCDIR=/usr/include/asterisk
ASTERISKMODDIR=/usr/lib/asterisk/modules
ASTERISKETCDIR=/etc/asterisk
OH323WRAPLIBDIR=/usr/local/lib
And I typed make , but I got the following error messages:
gentoo asterisk-oh323-0.6.3b # make
for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done
make: *** No rule to make target `ccflags'. Stop.
make: *** No rule to make target `ccflags'. Stop.
make[1]: Entering directory `/files/asterisk-oh323-0.6.3b/wrapper'
./check_ver /files/pwlib pwlib
./check_ver /files/openh323 openh323
g++ -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5
-DPWLIBVERSION=\1.6.6\ -DOPENH323VERSION=\1.13.5\
-I/files/pwlib/include/ptlib/unix -I/files/pwlib/include
-I/files/openh323/include -I/files/openh323/include/openh323
-I../asterisk-driver -c wrapendpoint.cxx -o wrapendpoint.o
In file included from /files/pwlib/include/ptlib.h:169,
from wrapendpoint.cxx:32:
/files/pwlib/include/ptlib/unix/ptlib/pdirect.h:78: error: parse error
before `
protected'
/files/pwlib/include/ptlib/unix/ptlib/pdirect.h:80: error: syntax error
before
`*' token
In file included from /files/pwlib/include/ptlib.h:181,
from wrapendpoint.cxx:32:
/files/pwlib/include/ptlib/unix/ptlib/config.h:53: error: parse error before
`
public'
/files/pwlib/include/ptlib/unix/ptlib/config.h:55: error: destructors must
be
member functions
/files/pwlib/include/ptlib/unix/ptlib/config.h:57: error: parse error before
`
protected'
In file included from /files/pwlib/include/ptlib.h:187,
from wrapendpoint.cxx:32:
/files/pwlib/include/ptlib/args.h:121: error: parse error before `{' token
/files/pwlib/include/ptlib/args.h:147: error: parse error before `const'
/files/pwlib/include/ptlib/args.h:156: error: parse error before `const'
/files/pwlib/include/ptlib/args.h:165: error: parse error before `int'
/files/pwlib/include/ptlib/args.h:175: error: parse error before `int'
(text omitted)
/files/openh323/include/h323caps.h:401: error: virtual BOOL
H323_RealTimeChannel::H323FramedAudioCodec::DecodeSilenceFrame(void*,
unsigned int)::H323_ALawCodec::H323_muLawCodec::OnReceivedPDU(const
H323_RealTimeChannel::H323FramedAudioCodec::DecodeSilenceFrame(void*,
unsigned int)::H323_ALawCodec::H323_muLawCodec::H245_DataType, int)
/files/openh323/include/h323caps.h:453: confused by earlier errors, bailing
out
make[1]: *** [wrapendpoint.o] Error 1
make[1]: Leaving directory `/files/asterisk-oh323-0.6.3b/wrapper'
make: *** [subdirs_all] Error 1
:(
Thank you
Astrit Morina
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Miroslav
Nachev
Sent: Tuesday, October 26, 2004 2:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] asterisk-oh323-0.6.3b
Dear Morina,
If you use Asterisk 1.0 stable and asterisk-oh323-0.6.3b and the last
OH323 from the CVS you must compile everything without errors.
We had some problems with Asterisk 1.0.1 and asterisk-oh323-0.6.3b
because in the new Asterisk version the callerid variable is struct
comparing with the 1.0 version where is string. When we replace callerid
variable with cid.cid_num the problem was solved.
Best Regards,
Miroslav Nachev
m Hi all,
m I'm trying to compile asterisk-oh323-0.6.3b but I got some comiling
m errors just on start. Can someone tell me the steps and the packages
m required to compile asterisk-oh323-0.6.3b.
m I'm usig asterisk-1.0.0 on Linux gentoo 2.4.25-gentoo-r3 .
m Thank you,
m Astrit Morina
m System Operator
m Tel: 038 20304050
m Fax: 038 20304020
m E-mail: [EMAIL PROTECTED]
m www.ipko.net
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[Asterisk-Users] Nightmare on disconnecting Zap and SIP channel
Hi all,
I have a nightmare working on disconnecting Zap and SIP channels. I've
been battling for almost 3 days with no avail.
My setup is something like this:
[GSM PHONE] -- [GSM Mobile Trunk Gateway] -- [TDM04B] -- [Asterisk]
-- [International VOIP provider]
I called from a GSM mobile phone to GSM trunk gateway then connected
to FXO and Asterisk for outbound calls. When I end up the call, the
Zap and SIP channel does not disconnect and the channels are still
active. So soft hangup is the solution to destroy the active channels
but sometimes the Zap channel is unusable and need to reload the
zaptel and wcfxs driver and restart asterisk to make the zap channel
work. I tried to check my GSM trunk gateway, using a voltage meter
just to know if it is sending a disconnect tone or changing the
voltage and it does. So it seems that Zaptel does not know how to deal
with it and the channel are still active. I would like to ask what are
the description of the Zaptel card, like loop current detection,
polarity reversal, disconnect tones etc etc and how does it deal with
that.
I also tried to use different release of zaptel drivers, from old cvs
to fresh cvs and stable release of zaptel driver from Digium but no
luck. also played some settings on zapata.conf.
my zapata.conf
[channels]
context=gsmmenu
signalling=fxs_ks
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
;echocancel=256
;echotraining=800
callwaiting=no
busydetect=1
busycount=7
relaxdtmf=yes
rxgain=-1.0
txgain=1.0
immediate=no
callprogress=yes
musiconhold=default
usecallerid=no
channel = 1-8
my zaptel.conf
fxsks=1-4
loadzone = us
defaultzone=us
my extensions.conf
[gsmmenu]
exten = s,1,Answer
exten = s,2,Wait,2
exten = s,3,Background(agent-pass)
exten = s,4,Authenticate(/etc/asterisk/pincode,a)
exten = s,5,Wait,2
exten = s,6,DigitTimeout,5
exten = s,7,ResponseTimeout,10
exten = s,8,Background(gsmmenu)
exten = _1NXXNXX,1,Dial(SIP/vpprovider/${EXTEN})
exten = _1NXXNXX,2,Hangup
exten = i,1,Playback,invalid
exten = i,2,Goto(s,6)
exten = t,1,Goto(s,6)
Hope anyone can help me.
Best regards,
Asterisk Mania
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RE: [Asterisk-Users] SIP vs MGCP
Hi Matthew, -Original Message- Does anyone have good pro/con on MGCP vs SIP? We are currently using all SIP, however, I went to a presentation today by Covad/Cisco on a new product they are unveiling and Covad is using all MGCP for Phone-PBX connectivity. This got me thinking: If this huge established company is selling VoIP turn-key solutions and they are using MGCP, why aren't I? So, if anyone has some good reasons for using 1 or the other. Please pass on. I'm sure others on the list would benefit from this as well. Pro MGCP: MGCP signals the PBX for every keypress, allowing the PBX to do quick matching and change dialtones etc. This is more like the classical telephone people expect. In some SIP phones, this is nicely covered, but not in all. Con MGCP: MGCP does not appear to have a widely supported method of authentication other than MAC-address or IP-address based. Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Why I can't hear anything from my sjphone as anh323 endpoint?
That's your problem, u need mpg123 and not 321. There are instructions on the wiki. Donny -Original Message- From: Willis Wang [mailto:[EMAIL PROTECTED] Sent: October 27, 2004 10:41 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Why I can't hear anything from my sjphone as anh323 endpoint? When I call asterisk(registered as an endpoint on gnugk) from sjphone(also registered on gnugk), I can see following on the asterisk console: *CLI == Starting H323/ip$192.168.1.125:3260/4096 at default,20030060,1 failed so falling back to exten 's' -- Executing Wait(H323/ip$192.168.1.125:3260/4096, 1) in new stack -- Executing Answer(H323/ip$192.168.1.125:3260/4096, ) in new stack -- Executing DigitTimeout(H323/ip$192.168.1.125:3260/4096, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(H323/ip$192.168.1.125:3260/4096, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround(H323/ip$192.168.1.125:3260/4096, demo-congrats) in new stack Is there anything wrong with my mpg123? I can't hear anything from sjphone, and after I dropped the call, I can't use 'stop now' to quit asterisk, and there will always be a process called mpg123 running: mpg123 -q -s --mono -r 8000 -b 2048 -f 4096 fpm-calm-river.mp3 fpm-sunshine.mp3 My linux version is debian woody 3.0 2.4.27-1-686, and the mpg321 package version is 0.2.10.3 thanks a lot! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hardware
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marcelo Pacheco Sent: 26 October 2004 23:23 To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hardware I have a system with VIA chipsets with one T400P (3FXS,1FXO) and 2 E100P (for testing with a cross over cable between them) I have an AMD system with Via chipsets, and have problems with an X100P as I've seen discussed elsewhere on this list. What I'm trying to establish, is whether the problem lies in the hardware (ie: the physical card does not get on with the motherboard), or whether its driver related (ie: the wxfxo driver does not get on with the system drivers for the Via chipset). The reason I'm asking, is that I'd like to upgrade to a TDM400 card with 1 x FXO and 1 x FXS modules. Obviously if the problem is hardware related, then the TDM should be fine, but if software related, I really don't want (and can't afford) to buy a TDM400 + modules only to find out it has the same problem! Does anyone have a TDM400 in an AMD system with a Via chipset, preferably the one as listed below, that could offer any guidance. The motherboard is an MSI KM2M Combo-L SKT A KM266 with a Duron 1800. Any and all help gratefully received, Cheers, Karl bash $ /sbin/lspci :00:00.0 Host bridge: VIA Technologies, Inc. VT8375 [KM266/KL266] Host Bridge :00:01.0 PCI bridge: VIA Technologies, Inc. VT8633 [Apollo Pro266 AGP] :00:07.0 Communication controller: Individual Computers - Jens Schoenfeld Intel 537 :00:10.0 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (rev 80) :00:10.1 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (rev 80) :00:10.2 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (rev 80) :00:10.3 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 82) :00:11.0 ISA bridge: VIA Technologies, Inc. VT8235 ISA Bridge :00:11.1 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06) :00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev 50) :00:12.0 Ethernet controller: VIA Technologies, Inc. VT6102 [Rhine-II] (rev 74) :01:00.0 VGA compatible controller: S3 Inc. VT8375 [ProSavage8 KM266/KL266] This e-mail has been scanned for all viruses by Star. The service is powered by MessageLabs. For more information on a proactive anti-virus service working around the clock, around the globe, visit: http://www.star.net.uk ___ 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] integrating Asterisk to existing TDM-based PBX
Hello, i'm looking for informations in integrating Asterisk to existing TDM-based PBX (particularly Siemens HiPath4000/Hicom300E) similar to the document you can find on www.pham.org/asterisk/asterisk-meridian-a1.pdf for Nortel. Unfortunately the page http://www.voip-info.org/wiki-Siemens+Hicom is not up to date. would be grateful for any pointers. thx. __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HiPath Wild Card T110P interface
Hi, I need to interface the wildcard t100p with the Simens HiPath 3000 PBX's T1 interface. I tried all the possible options for framing and signalling, but could get the card to interface correctly. The LED on the card always shows error. I tried connecting the PBX with a cross as well as straight T1 cable. I would be really grateful if someone can help me in this regard. Regards, - Ashish ___ 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] HiPath Wild Card T110P interface
On Thu, 28 Oct 2004, Ashish Shinde wrote: I need to interface the wildcard t100p with the Simens HiPath 3000 PBX's T1 interface. I tried all the possible options for framing and signalling, but could get the card to interface correctly. The LED on the card always shows error. I tried connecting the PBX with a cross as well as straight T1 cable. I would be really grateful if someone can help me in this regard. Who supplies the clocking? If neither end supplies it there will be problems. Did you use a T1 cross over and not an ethernet cross over? Do you know / have you set the framing and coding correct for the zaptel line in /etc/zaptel.conf? Have you loaded the zaptel module? Sucessfully? Have you run ztcfg? Peter ___ 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] Getting result codes of SIP-dials
Hello list! If I make a SIP call to another host like Dial(SIP/[EMAIL PROTECTED],120,r) I would like to know the error code the other hosts returns if the call failed, i.e. an 404 for Not Found, to play messages for the different possibilities. The problem is, that there seems to be no way of getting this results. HANGUPCAUSE gives me a 16 if everything went OK and a 1 for all other cases. DIALSTATUS gives me a CONGESTION if the call could not be completed. Any ideas how I can get the SIP error codes into a variable? Regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lütticher Straße 10 Tel 0241/701333-11 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ 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] HiPath Wild Card T110P interface
I really don't know who supplies the clocking. How to find that out? I did use a T1 cross - over cable and I tried all possible options for framing and coding in zaptel.conf. Tried ztcfg too. It doesn't complain. Is there any way to find out the framing and coding On Thu, 28 Oct 2004 11:07:01 +0200 (CEST), Peter Svensson [EMAIL PROTECTED] wrote: On Thu, 28 Oct 2004, Ashish Shinde wrote: I need to interface the wildcard t100p with the Simens HiPath 3000 PBX's T1 interface. I tried all the possible options for framing and signalling, but could get the card to interface correctly. The LED on the card always shows error. I tried connecting the PBX with a cross as well as straight T1 cable. I would be really grateful if someone can help me in this regard. Who supplies the clocking? If neither end supplies it there will be problems. Did you use a T1 cross over and not an ethernet cross over? Do you know / have you set the framing and coding correct for the zaptel line in /etc/zaptel.conf? Have you loaded the zaptel module? Sucessfully? Have you run ztcfg? Peter ___ 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
AW: AW: [Asterisk-Users] Firefly 1.9.6 released
Hello Tim I've found a bug in the new code that could have caused this problem. Try downloading and installing from http://www.virbiage.com/firefly/download/firefly-thirdparty.ex e again; this time you should get build 3941, which should solve the problem you ran into. Please let me know if it doesn't, or if you have any other trouble. This seemed to have fixed it, thank you very much! (4 hrs of operation so far) I'm really happy with Firefly, works like a charm ;) Thanks for the great work and for fixing it that fast! Best regards, Pascal. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE: Why I can't hear anything from my sjphone asanh323 endpoint?
Donny, Thanks a lot, I find it in the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20mpg123%20redhat I'll try it soon. Willis ___ 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] HiPath Wild Card T110P interface
I really don't know who supplies the clocking. How to find that out? I did use a T1 cross - over cable and I tried all possible options for framing and coding in zaptel.conf. Tried ztcfg too. It doesn't complain. Is there any way to find out the framing and coding Who is the network end of the link? Asterisk or the HiPath? If it is Asterisk then it should probably supply clocking. Set the clocking source in the span line in zaptel.conf to 0 to use the internal clock as a source. Framing and coding depends on what you have set up the HiPath for. You need to run ztcfg to set up the card according to the zaptel.conf file. Peter ___ 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] mcedit syntax for asterisk conf files
Does anyone cool mcedit syntax for the configuration files to share? :) ___ 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] HiPath Wild Card T110P interface
Hi Peter, Thanks for the help. Will try this and let you know. Thanks and regards, - Ashish On Thu, 28 Oct 2004 11:33:31 +0200 (CEST), Peter Svensson [EMAIL PROTECTED] wrote: On Thu, 28 Oct 2004, Ashish Shinde wrote: I really don't know who supplies the clocking. How to find that out? I did use a T1 cross - over cable and I tried all possible options for framing and coding in zaptel.conf. Tried ztcfg too. It doesn't complain. Is there any way to find out the framing and coding Who is the network end of the link? Asterisk or the HiPath? If it is Asterisk then it should probably supply clocking. Set the clocking source in the span line in zaptel.conf to 0 to use the internal clock as a source. Framing and coding depends on what you have set up the HiPath for. You need to run ztcfg to set up the card according to the zaptel.conf file. Peter ___ 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 AVM C4 with fewer than four lines?
Dear list, We have an * setup with an AVM C4 that works very well. There is one annoying problem though. Since we only have two ISDN lines at the moment, only two are connection to the C4. But the C4 reports 8 working B-channels to *. This means that on dialing out, you will be randomly assigned either a connected or unconnected port (luckily it doesn't affect incoming calls). Does anyone know of a way to disable the ports on the C4 that are not connected? regards Louis ___ 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] Polycom IP 500 and DTMF
Hi all ! I played around for a few hours with a polycom 500 phone and it seems me that the dtmf mode is not configurable, looks like it only has inband mode. While this is ok with G711 I assume that will result in some troubles using G729, altought I cant test it because I havent got any g729 licence yet. Anyone has tried and is willing to share his impressions ? TNX ! -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] disable second call / call waiting via SIP
HI! I have a problem with Sjphone on ipaq. It freeze when I receive a call on second line (seems like CPU is not enough). It there a way to restrict call accepting when I'm already on the phone via SIP in *? because: http://www.voip-info.org/wiki-PBX+Call+Waiting For most POTS providers in the United States, Call Waiting may be turned off by dialing *70 before dialing the telephone number. Is there the same in * ? Many thanks in advance. -- Best regards Vlad ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323-0.6.3b problems
I installed asterisk-oh323-0.6.3b and It had no errors, but now in the asterisk console there are no oh323 commands available... how do I know if Asterisk is is accepting the asterisk-oh323-0.6.3b extension? In oh323.conf I putted this: [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout gatekeeper=(gatekeeper ip) gatekeeperPassword=password gatekeeperTTL=600 userInputMode=TONE amaFlags=default accountCode=fccnasterisk context=voip-h323 [register] alias=asterisk context=all-aliases alias=ASTERISK context=more-aliases context=all-prefixes gwprefix=09 gwprefix=08 context=more-stuff alias=664 gwprefix=02 [codecs] codec=G711A frames=20 Thanks João Pereira ___ 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] IAXy Call Waiting Disable
Leonardo Gomes Figueira wrote: Hi, anyone knows how to disable call waiting on IAXy for every call ? I know that *70 disable for the current call but for each call I have todial it again. On dialplan I can use CheckGroup to limit the number of calls but on Queue with strategy RINGALL new calls keep ringing on the IAXy and the call waiting beep it's pretty noisy. Thanks, Leonardo I tried putting callwaiting=no in iax.conf but no help there. Any other suggestions folks? ___ 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] Xorcom Rapid Asterisk distro beta 0.5.2
I used the Xorcom cd to set up yesterday as well, haven't finished configs yet but I assume it's all ok. One point I will make is that it wouldn't discover my dhcp unless I installed using the expert mode then dhcp auto-discover worked fine. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of steve szmidt Sent: Thursday, October 28, 2004 12:49 AM To: Asterisk Users List Subject: Re: [Asterisk-Users] Xorcom Rapid Asterisk distro beta 0.5.2 On Sunday 10 October 2004 06:41 am, Tzafrir Cohen wrote: Hi folks Hello to all, We have created a simple Debian-based distribution of Asterisk. A CD image of an installer(150MB, requires no extra packages from the 'net) that installs Debian and Asterisk simple and easy. You are invited to take a look at: http://www.xorcom.com/rapid/ The image is free as in GPL. Sources included on the image. Any comments will be appreciated, either via the website or directly to me. I'd like to thank all the users and developers who helped me on #asterisk , #debian-boot and other places. Being posed as an Asterisk distro I decided to reply to the list. This is a nice and fast install ending up using the whole of 334M on a single partition. I used an old 600 MHz machine with 256M RAM and it went pretty fast and smooth. Though I can't for the life of me understand why it defaults to having these ports open by default: porttcp udp service 9 x x discard 13 x daytime 37 x time 2000x callbook I know I don't want to offer any of these services to the Internet. 9/13/37 are never used these days as those services were found too easy to hack through. That was a number of years ago and of course they could be improved. But still does not explain why they are open. My SIP devices uses 123. Port 2000 has been reported as recently as the 25th Oct to be an increasing new IIS PCT exploit. One usually prefer to keep a low profile with servers. This one is asking for attention. To their defense, if you read the release notes, they do recommend against using this in a production environment. I'd like to see a more prominent warning. And during the ever simple install it does not verify the root password. You better know what you type. It does not have ssh installed. Not being a debian user I'm not sure if there's a good reason to not include ssh in the default install. Except to keep things to bare bones. Though I would be hard to not have space for ssh. The game Banner could be skipped if space is the target. All in all it has lots of tools linked through a menu system that works pretty decently. Plenty enough for a server. I guess having an ability to edit asterisk from there could be added. Otherwise it's quite complete. I managed to install ssh, and mc, easily enough (from the CD I think, it seems too fast to have come down over the net). Somehow I've managed to make this my first direct contact with building a Debian system. It would be VERY hard to make it any easier. The one thing I'd like to see is a menu option that opens the services I need After the install. Not open by default. Asterisk from 05/31/04 is running on kernel 2.4.27. There's a minor point of having a broken vm link in /var/spool/asterisk. Having said all that, I think they have done a great job of creating a single Asterisk CD. Some honest work went into getting this done. As a contribution to Asterisk I think it's a very good thing! If the next release continous this well, it should be a very popular distro for our community! -- Steve Szmidt They that would give up essential liberty for 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] Re: Using AVM C4 with fewer than four lines?
Hi, On Thu, 28 Oct 2004 at 12:16, Louis van Dompselaar wrote: Does anyone know of a way to disable the ports on the C4 that are not connected? I think changing devices=4 to devices=2 in capi.conf should do the trick. Of course you have to make sure that your ISDN lines are connected to the two ports that haven't been disabled. cu Reinhard ___ 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] IAXy Call Waiting Disable
see here http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20IAX%20incoming-outgoing%20limit and here http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup - Original Message - From: Todd Lieberman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 7:22 AM Subject: Re: [Asterisk-Users] IAXy Call Waiting Disable Leonardo Gomes Figueira wrote: Hi, anyone knows how to disable call waiting on IAXy for every call ? I know that *70 disable for the current call but for each call I have to dial it again. On dialplan I can use CheckGroup to limit the number of calls but on Queue with strategy RINGALL new calls keep ringing on the IAXy and the call waiting beep it's pretty noisy. Thanks, Leonardo I tried putting callwaiting=no in iax.conf but no help there. Any other suggestions folks? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] disable second call / call waiting via SIP
Sorry, have already found that on wiki. http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup On Thu, 2004-10-28 at 13:31, Vladyslav wrote: HI! I have a problem with Sjphone on ipaq. It freeze when I receive a call on second line (seems like CPU is not enough). It there a way to restrict call accepting when I'm already on the phone via SIP in *? because: http://www.voip-info.org/wiki-PBX+Call+Waiting For most POTS providers in the United States, Call Waiting may be turned off by dialing *70 before dialing the telephone number. Is there the same in * ? Many thanks in advance. -- Best regards Vlad ___ 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] carrier deployment of SIP
I thought this event may interest some people. Cheers, Dean Date: 11/4/2004 2:00 p.m. New York/ 7:00 p.m. London Event: Troubleshooting SIP: Lessons Learned from Deploying SIP Services Sponsors: Empirix Speakers: Ray Le Maistre, International News Editor, Light Reading Click Here to Register SIP is emerging as the most important protocol for VOIP as carriers and enterprises adopt next generation systems and services. We'll look at how service providers can ensure their SIP services are satisfying voice quality demands, and examine the particular testing issues raised by the deployment of SIP-based systems. ___ 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] Polycom IP 500 and DTMF
I played around for a few hours with a polycom 500 phone and it seems me that the dtmf mode is not configurable, looks like it only has inband mode. While this is ok with G711 I assume that will result in some troubles using G729, altought I cant test it because I havent got any g729 licence yet. Anyone has tried and is willing to share his impressions ? I don't (yet) have any Polycom phones, but I have been stuck in situations needing inband DTMF over G.729. If you press the keys for a relatively long time (100 ms or more) then perhaps 1% of the digits may get lost. IIRC, the '0' key is the worst. If your application is for sales, press 1, then it's probably ok. If you need to key in 4-digit extension numbers, it will be annoying. If you need to input account numbers, passwords, etc., with complex menus, forget it. --Stewart ___ 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] Re: Using AVM C4 with fewer than four lines?
I think changing devices=4 to devices=2 in capi.conf should do the trick. Of course you have to make sure that your ISDN lines are connected to the two ports that haven't been disabled. It's already at devices=2 but that doesn't make a difference. I think chan_capi sees the C4 as one single device with four controllers. Anyway, the current capi.conf has devices=2 (tried 4 and 1; doesn't matter) and controller=1 (tried 1, 2 and 1,2,3,4, doesn't matter). I always get: Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. Contr3: 2 B channels total, 2 B channels free. Contr4: 2 B channels total, 2 B channels free. ___ 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] - ACAN - the Asterisk Comprehensive Archive Network (was RE: GPL thoughts)
On Thu, 2004-10-28 at 04:23, Steve Kann wrote: Adam Goryachev wrote: On Wed, 2004-10-27 at 13:37, Jim Van Meggelen wrote: I think that it's hard to reach a critical mass on a project like this. The way I see it, there's 4 places people will probably look for asterisk add-ons right now: [in no particular order]: 1) asterisk.org 2) The wiki 3) bugs.digium.com 4) google. The part that is important in your ACAN idea is comprehensive, and right now, the most comprehensive place is probably the union of the wiki and bugs.digium.com. True, personally though, I would use them in this order: 1) The wiki 2) bugs.digium.com 3) google I have never seen anything on asterisk.org about any add-on etc... In the land of Big Brother, we also have a wiki (may/may not be working right now) at www.blubrick.com/bb, and things work well. Lots of useful info on the wiki, and lots of files on deadcat. Sometimes a very active mailing list as well (has quietened down in the last couple of years...) So, I certainly wouldn't want to try and duplicate the wiki, but I see the wiki as being for documentation, and ACAN as a file archive. The structure of your site is nice -- just like freshmeat. Is there a real advantage in using this site, as opposed to freshmeat, with appropriate trove categories? Generally it was based on a combination of freshmeat and download.com, plus any other random ideas I happened to have along the way. As far as advantages, AFAIK, freshmeat doesn't allow you to upload a file, it simply points to wherever you decide to publish the file. A lot of people aren't going to setup a web server, and a couple of web pages just so they can share their neat asterisk extensions macro I see that as being one of the main advantages. The next one being that you can simply browse any file on the website, and sooner or later you will come across something that looks very interesting. Just a couple of thoughts on the matter. Oh, and the website again (since I remove my .sig for this mailing list) is www.websitemanagers.com.au/asterisk Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] - ACAN - the Asterisk Comprehensive Archive Network (was RE: GPL thoughts)
On Thu, 2004-10-28 at 04:15, Michael Bielicki wrote: BBversion: ? You are right, I'll have it changed to Asterisk Version tomorrow... Regards, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Motorola Vt1000
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Milk Sent: Wednesday, October 27, 2004 6:56 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Motorola Vt1000 First off -- congratulations to canceling your Vonage account. Eventually, they'll learn. Secondly, when I cancelled one of my Vonage numbers, they offered to perform a reset on my ATA186 for $15, and after that, it was fully usable as a dual FXS on my * box. I opted for that, thinking a $55 ATA isn't all that bad. I had no clue how poorly the Cisco units perform in comparison to my Sipuras, so I sold the ATA186 on ebay for $100 and some change and bought another SPA-2000. Return it, get your fee back (unless you can find a buyer on ebay), and get yourself an SPA-2000 or a PAP2-NA if you can find someone who delivers. I'll second this suggestion--return it. I cancelled my Vonage account, and they happily agreed to unlock my Motorola for $15. Said it was done even before I got off the phone. They didn't unlock it. Googling a little, I found another post from someone who had the same problem. Apparently the Motorolas can't be unlocked. Can anyone confirm this? ___ 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] where do i find openssl-devel to mandrake 10.1
Mandrake specific: urpmi openssl-devel Thomas Hupfeldt wrote: - Original Message - From: Brian West [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 2:17 AM Subject: RE: [Asterisk-Users] where do i find openssl-devel to mandrake 10.1 Go to www.openssl.org download the tarball and compile it. bkw I allready did that, but it does not solve the problem when i compile asterisk.. I still get this message halfway in the compiling, and af far as i can remember from when i compiled asterisk last time, and i got this message, it dissapeared when i installed openssl-devel. /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Fejl 1 Is'nt there a openssl-devel to mandrake 10.1 ? Regards Thomas H. ___ 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 begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ 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] test telephone numbers
Richard Bennett wrote: On Thursday 28 October 2004 04:15, Steve Totaro wrote: i think he meant numbers that would not be billed for completing a call. No, any number is just fine. Preferably a mix of mobile and fixed numbers for as many countries/regions as possible. So often a customer will say something like I've been trying to get a call through to Uzbekistan all day and nothing works, so i have to try to route Uzbekistan through a carrier who will be able to terminate it properly. Being able to test with a number that won't wake someone up at 3am would be much easier... Finding hotels or companies using an IVR system on the internet will help for landlines, but if anyone has any out of use mobile numbers that will still play a message, this would help a lot to... Thanks for the numbers and suggestions so far, Richard. When you get a decent size list, you will post it on the wiki(if you're not doing it now), or at least mail it to any other interested parties, right? -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] test telephone numbers
Yeah, I'd definitely be interested in that list too. I thought I was one of the only find the right carrier game. Sometimes customers can't place calls or have quality issues to certain countries, sometimes even certain Area codes within the U.S, so I then route those calls through NuFone/LookieLoo/VoicePulse/1 of our 6 Voice T1/PRI's, whichever works best. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Thompson Sent: Thursday, October 28, 2004 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] test telephone numbers Richard Bennett wrote: On Thursday 28 October 2004 04:15, Steve Totaro wrote: i think he meant numbers that would not be billed for completing a call. No, any number is just fine. Preferably a mix of mobile and fixed numbers for as many countries/regions as possible. So often a customer will say something like I've been trying to get a call through to Uzbekistan all day and nothing works, so i have to try to route Uzbekistan through a carrier who will be able to terminate it properly. Being able to test with a number that won't wake someone up at 3am would be much easier... Finding hotels or companies using an IVR system on the internet will help for landlines, but if anyone has any out of use mobile numbers that will still play a message, this would help a lot to... Thanks for the numbers and suggestions so far, Richard. When you get a decent size list, you will post it on the wiki(if you're not doing it now), or at least mail it to any other interested parties, right? -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users 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] Ex-girlfriend-logic
Here is the logic that I wanted b/c my bro-n-law is in AU and uses my phone number in the US as his now for his family to contact him as an extension off my * server.
The need was defined to use something like the ex-g/f logic to route calls to his extension instead of me getting all the calls then forwarding them to him. Since his address book is much larger than I had expected or wanted to write up in extensions.conf here is how I achieved it:
I created a *db named 'route'. Basically in there I would input a CID# and an extension to route to.
database put route phone_num extension
In the extensions.conf I put in the following:
[incoming]exten = s,1,NoOp("Incoming:" ${CALLERID})exten = s,2,LookupCIDNameexten = s,3,LookupBlackListexten = s,4,DBget(exten=route/${CALLERIDNUM})exten = s,5,NoOp("Transfering to extension: " ${exten})exten = s,6,Goto(default,${exten},1)exten = s,104,Goto(2200)exten = s,105,GotoIfTime(06:00-22:30|*|*|*?default,2200,1)exten = s,106,Goto(2200)exten = s,2200,Background(press1tospeaktome)exten = s,2201,Wait(3)exten = s,2202,Voicemail(u2200)exten = s,2203,Hangup
Then the next problem was how to deal with my family and what to do with them. Since my parents are in the US and my sister is in AU, I created a menu context and send them to an extension that sends them to a menu context so they can decide to press 1 for me and 2 for my sister bro-n-law.
We have been very happy with this solution and the only draw back is that if there is no caller id number presented, I get those calls but have handled them various ways and plan to re-impliment shortly.
Also, I used the WIKI setup for the PHP and CID name lookup (LookupCIDName)to allow my bro-n-law to input all his phone numbers and point them to his extension so there wasn't any resources of me to input his address book into the *database of route and he could update on demand or his leisure.
Enjoy.
-Jeff
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[Asterisk-Users] Re: call progress - what are the sticking points?
i don't have a specific bug in mind, i was just wondering WHY call progress doesn't work so well -- in particular, on analog lines. ie. is it a hardware or software problem (or both). with more info, i'd like to help to work out the kinks, for myself and everyone. :) I have the same problem. callprogress is very experimental and buggy now. and i've lost the .call files feature of asterisk. what do you think about submitting a bug on bugs.digium.com? not sure what you mean by 'lost the .call files feature', but if you have a specific bug to post, i think it would be great if you posted it. regards, shabanip Hello, I've been experimenting with the call progress analysis features of *, with mixed success on Zap as well as IAX channels. I've read all the posts about it, including (but not limited to) http://www.voip-info.org/wiki-Asterisk+auto-dial+out and the pages it references. My question is, what's the current state -- is there any work in progress right now to improve the reliability of * call progress detection? last I saw it was still listed as 'experimental'. What are the problems that are preventing a more robust implementation of call progress detection? Would this work better with different hardware (ie. I've had success in the past using Dialogic telephony boards)? Or is this primarily a software issue with *? Thanks much! Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WRT54GS zaptel timing device
Kristian Kielhofner wrote: Hello, I know that I can run Asterisk on the Linksys WRT54GS, but can I do Zaptel as well? I would really like a timing device so I can do IAX2 trunking - but I don't know how to go about it. Has anyone done this? You *know* it can? Does that mean you've actually seen it work? A number of people have said it can run, but after questioning it turns out to be hearsay. You can replace the Linksys image with one of serveral. You can build Asterisk with the tools. You can load that image into the Linksys box. You can run it. Then you seem to get stuck, because something fouls up in the threading and nothing works. Anyone who can post a real way to get Asterisk running on these Linksys boxes will be considered a hero :-) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 500 and DTMF
Alessio Focardi wrote: Hi all ! I played around for a few hours with a polycom 500 phone and it seems me that the dtmf mode is not configurable, looks like it only has inband mode. While this is ok with G711 I assume that will result in some troubles using G729, altought I cant test it because I havent got any g729 licence yet. Anyone has tried and is willing to share his impressions ? Polycom IP phones support RFC2833. I don't recall where in the config interface it's set. begin:vcard fn:Eric Wileing n:Wileing;Eric email;internet:[EMAIL PROTECTED] tel;work:504-899-1387 x2120 x-mozilla-html:FALSE version:2.1 end:vcard ___ 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] Re: call progress - what are the sticking points?
We use Asterisk 1.0 Stable CVS, as of 2 days ago. Our paging system relies on the '.call' files. We've paged several times since the upgrade, so the /var/spool/asterisk/outgoing/ system still works. What problems are you having? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen David Sent: Thursday, October 28, 2004 9:46 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Re: call progress - what are the sticking points? i don't have a specific bug in mind, i was just wondering WHY call progress doesn't work so well -- in particular, on analog lines. ie. is it a hardware or software problem (or both). with more info, i'd like to help to work out the kinks, for myself and everyone. :) I have the same problem. callprogress is very experimental and buggy now. and i've lost the .call files feature of asterisk. what do you think about submitting a bug on bugs.digium.com? not sure what you mean by 'lost the .call files feature', but if you have a specific bug to post, i think it would be great if you posted it. regards, shabanip Hello, I've been experimenting with the call progress analysis features of *, with mixed success on Zap as well as IAX channels. I've read all the posts about it, including (but not limited to) http://www.voip-info.org/wiki-Asterisk+auto-dial+out and the pages it references. My question is, what's the current state -- is there any work in progress right now to improve the reliability of * call progress detection? last I saw it was still listed as 'experimental'. What are the problems that are preventing a more robust implementation of call progress detection? Would this work better with different hardware (ie. I've had success in the past using Dialogic telephony boards)? Or is this primarily a software issue with *? Thanks much! Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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] Multiple SIP gateway accounts
On Wed, 2004-10-27 at 15:58, Adam Greenbaum wrote: If you have multiple accounts on the same SIP-PSTN gateway, how do you dial out of a particular one? I think the answer will also involve me setting my domain and username on the outgoing invite, but I have a feeling this might not work because of the authentication. Ok, to answer my own question, it looks as though the correct way of doing this is to use [EMAIL PROTECTED] (from the sip.conf) in address you are dialing. Then fromuser and fromdomain in the SIP entity. Would anyone comment whether this is correct? This now brings me onto another question: How do I associate a SIP entity with a registered account on a PSTN gateway? I have 2 register lines and 2 entities. When I dial into asterisk from the PSTN gateway it always associates with the second entity. (CVS) I've looked through the source and it _seems_ as though you can only match against the from: username [find_user()] and from source address [find_peer()]. Surely you would need to match against the destination sip: username, not the From: username. Am I missing something? I must be, otherwise you would never be able to use multiple accounts on a SIP gateway. Thanks for your help, Adam. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: call progress - what are the sticking points?
Stephen David wrote: i don't have a specific bug in mind, i was just wondering WHY call progress doesn't work so well -- in particular, on analog lines. ie. is it a hardware or software problem (or both). with more info, i'd like to help to work out the kinks, for myself and everyone. :) Back in the days of Stowger exchanges you knew when the called party answered, by a reversal of the DC voltage on your analogue line. With digital exchanges that stopped, and no solid feedback is given to the caller on ordinary analogue lines. You have to infer that someone has answered, and the reliability of that can be poor. Digital lines, like ISDNand SS7, and protocols like MFC/R2 tell you positively that someone has answered. I have the same problem. callprogress is very experimental and buggy now. and i've lost the .call files feature of asterisk. what do you think about submitting a bug on bugs.digium.com? not sure what you mean by 'lost the .call files feature', but if you have a specific bug to post, i think it would be great if you posted it. regards, shabanip Hello, I've been experimenting with the call progress analysis features of *, with mixed success on Zap as well as IAX channels. I've read all the posts about it, including (but not limited to) http://www.voip-info.org/wiki-Asterisk+auto-dial+out and the pages it references. My question is, what's the current state -- is there any work in progress right now to improve the reliability of * call progress detection? last I saw it was still listed as 'experimental'. What are the problems that are preventing a more robust implementation of call progress detection? Would this work better with different hardware (ie. I've had success in the past using Dialogic telephony boards)? Or is this primarily a software issue with *? If you had good results with Dialogic it was merely luck. Because they have to infer the phone has been answered, their detection only works if the calls follow their model of how someone answers the phone. Depending on your circumstances, and the nature of the calls you make, it can be hopelessly unreliable. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 500 and DTMF
On Thu, 2004-10-28 at 23:34, Eric Wieling wrote: Alessio Focardi wrote: Hi all ! I played around for a few hours with a polycom 500 phone and it seems me that the dtmf mode is not configurable, looks like it only has inband mode. Polycom IP phones support RFC2833. I don't recall where in the config interface it's set. __ I have this in ipmid.cfg: DTMF tone.dtmf.level=-15 tone.dtmf.onTime=50 tone.dtmf.offTime=50 tone.dtmf.chassis.masking=1 tone.dtmf.stim.pac.offHookOnly=0 tone.dtmf.viaRtp=0 tone.dtmf.rfc2833Control=1 tone.dtmf.rfc2833Payload=101/ All one line. However, regardless of my sip.conf, I can't use DTMF during the call. Re-reading this (without looking at the manual) perhaps tone.dtmf.viaRtp=0 should be 1 to enable rfc2833 anybody got a suggestion... Thanks, Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 3000 tone table settings for Australia
Hey All, I've got a Sipura 3000 here which I'm currently testing. I'm after either a description of the the Sipura tone format (on the Regional Tab), or a copy of what the settings need to be for Australia, if anyone has then... I've had a look at the indications.conf in Asterisk, but I can't seem to translate it into the format for the Sipura. Thanks Darryl ___ 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] test telephone numbers
Paul Rodan wrote: Yeah, I'd definitely be interested in that list too. I thought I was one of the only find the right carrier game. Sometimes customers can't place calls or have quality issues to certain countries, sometimes even certain Area codes within the U.S, so I then route those calls through NuFone/LookieLoo/VoicePulse/1 of our 6 Voice T1/PRI's, whichever works best. A place I used to work was an inbound call center. The owner would get up every morning at 5 or so and dial every one of the toll free numbers to make sure they were all still working. If any of them failed, the lady that maintained the phone system would be the next person called. That was never a happy phone call... -- Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WRT54GS zaptel timing device
http://www.pbs.org/cringely/pulpit/pulpit20040527.html - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 9:27 AM Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device Kristian Kielhofner wrote: Hello, I know that I can run Asterisk on the Linksys WRT54GS, but can I do Zaptel as well? I would really like a timing device so I can do IAX2 trunking - but I don't know how to go about it. Has anyone done this? You *know* it can? Does that mean you've actually seen it work? A number of people have said it can run, but after questioning it turns out to be hearsay. You can replace the Linksys image with one of serveral. You can build Asterisk with the tools. You can load that image into the Linksys box. You can run it. Then you seem to get stuck, because something fouls up in the threading and nothing works. Anyone who can post a real way to get Asterisk running on these Linksys boxes will be considered a hero :-) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Re: Using AVM C4 with fewer than four lines?
On Thu, 28 Oct 2004 at 14:08, Louis van Dompselaar wrote: It's already at devices=2 but that doesn't make a difference. Sorry, I was looking at the wrong place and drawing the wrong conclusions ... I think chan_capi sees the C4 as one single device with four controllers. A device corresponds to a B-chanel in capi.conf. See the capi.conf form chan_capi source tarball - it uses devices=2 for BRI lines, and devices=30 for PRI lines. So the C4 appears to chan_capi as 4 controllers with 2 devices each. You need to set up a separate interface section for each controller that is actually in use, and just skip the rest: [interfaces] ; The first controller msn=... ... controller=1 devices=2 ; The second controller msn=... ... controller=2 devices=2 Contr1: 2 B channels total, 2 B channels free. Contr2: 2 B channels total, 2 B channels free. Contr3: 2 B channels total, 2 B channels free. Contr4: 2 B channels total, 2 B channels free. capi info reports all the CAPI devices that exist in the system, but that doesn't mean it is really using them all. If you start Asterisk in verbose mode asterisk -vc , you should see a warning like this for every unused CAPI controller: Oct 28 15:41:04 WARNING[1076791936]: chan_capi.c:2786 load_module: Unused contr2 This is what I get on a system with a C2 card where only one of the two controllers is configured in capi.conf. cu Reinhard ___ 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] Zaptel channels
On Wed, 2004-10-27 at 12:10, Paulo Adriano wrote: What is the command to see the zap channels registered. Im getting an error when trying to access my outgoing line. No channel type registered for Zap Drivers are loaded but where do I register this so called zap channels ? Regards The command you are looking for is zap show channels. If * complains that no such command exists then you installed zaptel after you installed Asterisk. Recompile Asterisk and you should be fine. -Seth __ -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ 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] WRT54GS zaptel timing device
Hi Steve, I know about that page, but it does not mention Asterisk. I have built and run several things on a wrt54g. I just haven't had the time to figure out why Asterisk doesn't work. It seems to be a threading problem, from the little time I spent playing with it. I tried some other test code that does threading, and it ran without problems. Steve Steve Totaro wrote: http://www.pbs.org/cringely/pulpit/pulpit20040527.html - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 9:27 AM Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device Kristian Kielhofner wrote: Hello, I know that I can run Asterisk on the Linksys WRT54GS, but can I do Zaptel as well? I would really like a timing device so I can do IAX2 trunking - but I don't know how to go about it. Has anyone done this? You *know* it can? Does that mean you've actually seen it work? A number of people have said it can run, but after questioning it turns out to be hearsay. You can replace the Linksys image with one of serveral. You can build Asterisk with the tools. You can load that image into the Linksys box. You can run it. Then you seem to get stuck, because something fouls up in the threading and nothing works. Anyone who can post a real way to get Asterisk running on these Linksys boxes will be considered a hero :-) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WRT54GS zaptel timing device
Now if one could only find a way to adapt an FXS module! :-) -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Thursday, October 28, 2004 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device http://www.pbs.org/cringely/pulpit/pulpit20040527.html - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 9:27 AM Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device Kristian Kielhofner wrote: Hello, I know that I can run Asterisk on the Linksys WRT54GS, but can I do Zaptel as well? I would really like a timing device so I can do IAX2 trunking - but I don't know how to go about it. Has anyone done this? You *know* it can? Does that mean you've actually seen it work? A number of people have said it can run, but after questioning it turns out to be hearsay. You can replace the Linksys image with one of serveral. You can build Asterisk with the tools. You can load that image into the Linksys box. You can run it. Then you seem to get stuck, because something fouls up in the threading and nothing works. Anyone who can post a real way to get Asterisk running on these Linksys boxes will be considered a hero :-) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP-DTMF
Hi, Can anyone please comment on this? --- Asterisk . [EMAIL PROTECTED] wrote: Hi Alex, --- Alex Barnes [EMAIL PROTECTED] wrote: You should set the type of DTMF on a per SIP PEER basis (sip.conf). Then simply set the SJPhone peer to use dtmfmode=inband. I have used SJPhone without problems along side Snoms that use dtmfmode=rfc2833. Thanks for the response. I know this will work, if the UACs are registered with Asterisk. But none of the UACs that dial this number are registered with Asterisk. They just use the sip uri to dial to that number. ie, like this: sip:[EMAIL PROTECTED]:port. I was trying to make any sip client to reach this number and to the desired extension just by dialing using the sip uri. Hope that explains the problem. Any help appreciated. Thanks again, Girish HTH Alex -Original Message- I have mapped a number in the default context of my dialplan. When someone dials that number, it plays an IVR message and allows the caller to enter 4 digit extensions. If the extension is a valid one, the call wll be routed to that particular extension. 'INFO' is set as the dtmf mode. This works fine if i call from a SIP UAC which sends dtmf as INFO. But When i dial using SJPhone, call doesn't get routed, because SJPhone uses inband dtmf. So, my problem is only people who use UACs that send dtmf using the INFO method can reach the desired extension, where as people who use SJPhone cannot do this. Can i make Asterisk to receive both info and inband dtmf for the same number? Is this possible? If so, can anyone tell me how to do that? Thanks, Girish ___ Do you Yahoo!? Express yourself with Y! Messenger! Free. Download now. http://messenger.yahoo.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WRT54GS zaptel timing device
Using zaptel, zaprtc and rtcsetup you can get simple Zaptel timing, no usb ports or zaptel hardware required. Zaprtc is a simple hack to the rtc driver. Compile your kernel without rtc support, compile this module, and load it along with zaptel. Then run rtcsetup to put it in the background and you've got Zaptel timing. However, if Asterisk has problems running on it, then it's useless. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Schulte Sent: Thursday, October 28, 2004 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] WRT54GS zaptel timing device Now if one could only find a way to adapt an FXS module! :-) -Original Message- From: Steve Totaro [mailto:[EMAIL PROTECTED] Sent: Thursday, October 28, 2004 8:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device http://www.pbs.org/cringely/pulpit/pulpit20040527.html - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 9:27 AM Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device Kristian Kielhofner wrote: Hello, I know that I can run Asterisk on the Linksys WRT54GS, but can I do Zaptel as well? I would really like a timing device so I can do IAX2 trunking - but I don't know how to go about it. Has anyone done this? You *know* it can? Does that mean you've actually seen it work? A number of people have said it can run, but after questioning it turns out to be hearsay. You can replace the Linksys image with one of serveral. You can build Asterisk with the tools. You can load that image into the Linksys box. You can run it. Then you seem to get stuck, because something fouls up in the threading and nothing works. Anyone who can post a real way to get Asterisk running on these Linksys boxes will be considered a hero :-) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WRT54GS zaptel timing device
Curious, can a WRT54G with this firmware, using Wonder Shaper, act as a cheap QOS device? One of our biggest problems is customers with Cable/DSL (256k upload) trying to upload files or browse several webpages at once, it affects the quality of the phone calls, naturally. We were looking into cheap managed switches for their QOS ability, but it seems the LinkSys would be a cheaper/easier solution. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, October 28, 2004 9:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device http://www.pbs.org/cringely/pulpit/pulpit20040527.html - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 9:27 AM Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device Kristian Kielhofner wrote: Hello, I know that I can run Asterisk on the Linksys WRT54GS, but can I do Zaptel as well? I would really like a timing device so I can do IAX2 trunking - but I don't know how to go about it. Has anyone done this? You *know* it can? Does that mean you've actually seen it work? A number of people have said it can run, but after questioning it turns out to be hearsay. You can replace the Linksys image with one of serveral. You can build Asterisk with the tools. You can load that image into the Linksys box. You can run it. Then you seem to get stuck, because something fouls up in the threading and nothing works. Anyone who can post a real way to get Asterisk running on these Linksys boxes will be considered a hero :-) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: call progress - what are the sticking points?
Stephen David wrote: i don't have a specific bug in mind, i was just wondering WHY call progress doesn't work so well -- in particular, on analog lines. ie. is it a hardware or software problem (or both). with more info, i'd like to help to work out the kinks, for myself and everyone. :) Back in the days of Stowger exchanges you knew when the called party answered, by a reversal of the DC voltage on your analogue line. With digital exchanges that stopped, and no solid feedback is given to the caller on ordinary analogue lines. You have to infer that someone has answered, and the reliability of that can be poor. Digital lines, like ISDNand SS7, and protocols like MFC/R2 tell you positively that someone has answered. That's a good explanation. I'll expand upon it a bit by saying that even with reversal, there's a limited amount of information you can represent with that. POTS was always intended to be cheap basic phone service, and keeping it simple was not considered a downside by the phone company. As it is, you run into an information representation issue with the existing technology: the entire traditionally used bandwidth of the channel during a call is used for audio data (that is, to say, that they send an analog signal). As a call originator, you really can not tell the difference between a ringing signal generated by the phone company and a ringing signal caused by the called party picking up the phone and playing an identical sound. Reversal fixed that, but was largely made obsolete by out of band supervision - since the real purpose of reversal was for the telephone company to be able to bill correctly for completed calls (IIRC, ICBW). More difficult is the problem of knowing when the remote end has gone away. Reversal, loop break, dial tone, and just plain silence are not all that unusual as methods of detection. In some cases, you do actually need to infer that the remote has gone away. There's no real excuse for us to be using this technology anymore, with the availability of things like ISDN BRI, which allows for digital signalling of call progress. However, we continue to use it because the ILEC's have done such a fab job of making ISDN a dead technology. Funny thing is, it'll end up biting them where it hurts, as customers drift to VoIP to gain the features that ISDN promised, at a fraction of the cost. (I say that as someone who currently brings in dialtone on BRI, btw) ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] Re: call progress - what are the sticking po ints?
It looks for tones (currently hardwired as US). I have updated to include UK tones but is hard to get it to reliably recognise. For example the tones in the switch here at work are 5-10% off frequency. Correcting for this, and doing a lot of fiddling it did recognise the tones but was unreliable. I have a problem in that our office switch clears to dialtone rather than busy if the other end hangs up. I would like a way of recognising unexpected dialtone and hanging-up. So far, this has not been easy. I have changed the busydetect to clear if it gets continupus tone for 8 seconds but this does false hangups and would be useless for a fax machine. Peter -Original Message- From: Steve Underwood [mailto:[EMAIL PROTECTED] Sent: 28 October 2004 14:37 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: call progress - what are the sticking points? Stephen David wrote: i don't have a specific bug in mind, i was just wondering WHY call progress doesn't work so well -- in particular, on analog lines. ie. is it a hardware or software problem (or both). with more info, i'd like to help to work out the kinks, for myself and everyone. :) Back in the days of Stowger exchanges you knew when the called party answered, by a reversal of the DC voltage on your analogue line. With digital exchanges that stopped, and no solid feedback is given to the caller on ordinary analogue lines. You have to infer that someone has answered, and the reliability of that can be poor. Digital lines, like ISDNand SS7, and protocols like MFC/R2 tell you positively that someone has answered. I have the same problem. callprogress is very experimental and buggy now. and i've lost the .call files feature of asterisk. what do you think about submitting a bug on bugs.digium.com? not sure what you mean by 'lost the .call files feature', but if you have a specific bug to post, i think it would be great if you posted it. regards, shabanip Hello, I've been experimenting with the call progress analysis features of *, with mixed success on Zap as well as IAX channels. I've read all the posts about it, including (but not limited to) http://www.voip-info.org/wiki-Asterisk+auto-dial+out and the pages it references. My question is, what's the current state -- is there any work in progress right now to improve the reliability of * call progress detection? last I saw it was still listed as 'experimental'. What are the problems that are preventing a more robust implementation of call progress detection? Would this work better with different hardware (ie. I've had success in the past using Dialogic telephony boards)? Or is this primarily a software issue with *? If you had good results with Dialogic it was merely luck. Because they have to infer the phone has been answered, their detection only works if the calls follow their model of how someone answers the phone. Depending on your circumstances, and the nature of the calls you make, it can be hopelessly unreliable. Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This e-mail and any attachment is for authorised use by the intended recipient(s) only. It may contain proprietary material, confidential information and/or be subject to legal privilege. It should not be copied, disclosed to, retained or used by, any other party. If you are not an intended recipient then please promptly delete this e-mail and any attachment and all copies and inform the sender. Thank you. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail.conf
I have set it on my box but it didn't work either. I have it set up like this: [mailbox number] = [password],[name],[email],delete=yes -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Wednesday, October 27, 2004 4:38 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] voicemail.conf Delete=yes is a per box option. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Duraid Abbas Sent: Wednesday, October 27, 2004 3:14 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voicemail.conf I have delete=yes and attach=yes. But my messages are not getting deleted after they're sent. I'm running asterisk as root so it can't be a permission issue. Any ideas? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WRT54GS zaptel timing device
yes. - Original Message - From: Paul Rodan [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 10:18 AM Subject: RE: [Asterisk-Users] WRT54GS zaptel timing device Curious, can a WRT54G with this firmware, using Wonder Shaper, act as a cheap QOS device? One of our biggest problems is customers with Cable/DSL (256k upload) trying to upload files or browse several webpages at once, it affects the quality of the phone calls, naturally. We were looking into cheap managed switches for their QOS ability, but it seems the LinkSys would be a cheaper/easier solution. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, October 28, 2004 9:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device http://www.pbs.org/cringely/pulpit/pulpit20040527.html - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 9:27 AM Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device Kristian Kielhofner wrote: Hello, I know that I can run Asterisk on the Linksys WRT54GS, but can I do Zaptel as well? I would really like a timing device so I can do IAX2 trunking - but I don't know how to go about it. Has anyone done this? You *know* it can? Does that mean you've actually seen it work? A number of people have said it can run, but after questioning it turns out to be hearsay. You can replace the Linksys image with one of serveral. You can build Asterisk with the tools. You can load that image into the Linksys box. You can run it. Then you seem to get stuck, because something fouls up in the threading and nothing works. Anyone who can post a real way to get Asterisk running on these Linksys boxes will be considered a hero :-) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: call progress - what are the sticking points?
Joe Greco wrote: Stephen David wrote: i don't have a specific bug in mind, i was just wondering WHY call progress doesn't work so well -- in particular, on analog lines. ie. is it a hardware or software problem (or both). with more info, i'd like to help to work out the kinks, for myself and everyone. :) Back in the days of Stowger exchanges you knew when the called party answered, by a reversal of the DC voltage on your analogue line. With digital exchanges that stopped, and no solid feedback is given to the caller on ordinary analogue lines. You have to infer that someone has answered, and the reliability of that can be poor. Digital lines, like ISDNand SS7, and protocols like MFC/R2 tell you positively that someone has answered. That's a good explanation. I'll expand upon it a bit by saying that even with reversal, there's a limited amount of information you can represent with that. POTS was always intended to be cheap basic phone service, and keeping it simple was not considered a downside by the phone company. As it is, you run into an information representation issue with the existing technology: the entire traditionally used bandwidth of the channel during a call is used for audio data (that is, to say, that they send an analog signal). As a call originator, you really can not tell the difference between a ringing signal generated by the phone company and a ringing signal caused by the called party picking up the phone and playing an identical sound. Reversal fixed that, but was largely made obsolete by out of band supervision - since the real purpose of reversal was for the telephone company to be able to bill correctly for completed calls (IIRC, ICBW). Actually it was not really intentional. The reversal back to the calling party was just a byproduct of the way a Strowger exchange worked. Within the network it was used for billing purposes. More difficult is the problem of knowing when the remote end has gone away. Reversal, loop break, dial tone, and just plain silence are not all that unusual as methods of detection. In some cases, you do actually need to infer that the remote has gone away. Hangup is relatively easy. Most lines now give a strong distinct beeping either the moment the phone is dropped, or a short time after. The problem in * is its detector is not very good, or very voice immune. I have a much better one in my spandsp library, but it isn't integrated with * right now. Detecting answer is the tough one. There is nothing unambiguous about it. There's no real excuse for us to be using this technology anymore, with the availability of things like ISDN BRI, which allows for digital signalling of call progress. However, we continue to use it because the ILEC's have done such a fab job of making ISDN a dead technology. Funny thing is, it'll end up biting them where it hurts, as customers drift to VoIP to gain the features that ISDN promised, at a fraction of the cost. As someone whose colleagues built one of the first ISDN muxes in the 80's, I can tell you attitudes made it dead from day one. (I say that as someone who currently brings in dialtone on BRI, btw) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: call progress - what are the sticking points?
On Thu, 28 Oct 2004, Steve Underwood wrote: Stephen David wrote: i don't have a specific bug in mind, i was just wondering WHY call progress doesn't work so well -- in particular, on analog lines. ie. is it a hardware or software problem (or both). with more info, i'd like to help to work out the kinks, for myself and everyone. :) Back in the days of Stowger exchanges you knew when the called party answered, by a reversal of the DC voltage on your analogue line. With digital exchanges that stopped, and no solid feedback is given to the caller on ordinary analogue lines. You have to infer that someone has answered, and the reliability of that can be poor. Digital lines, like ISDNand SS7, and protocols like MFC/R2 tell you positively that someone has answered. At least in Sweden pots interfaces in the pstn all have answer and disconnect supervision through polarity reversals. If you had good results with Dialogic it was merely luck. Because they have to infer the phone has been answered, their detection only works if the calls follow their model of how someone answers the phone. Depending on your circumstances, and the nature of the calls you make, it can be hopelessly unreliable. It could be that it handles answer supervision and that his pots line has tbat option as well. Peter ___ 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] Re: call progress - what are the sticking points?
Joe Greco wrote: More difficult is the problem of knowing when the remote end has gone away. Reversal, loop break, dial tone, and just plain silence are not all that unusual as methods of detection. In some cases, you do actually need to infer that the remote has gone away. I understand that the phone company (sometime) doesn't provide information about remote hangup on POTS lines. What bugs me is the simple question - how does your average 10$ answering machine detects the hang up? I'm guessing the obvious - DSP and some heuristics as to what a hangup sounds like and it sounds to me that it isn't all that hard to do in Asterisk (since it's done in those cheap machines) but I would be very glad to hear some tips from someone that knows a little better then me. Thanks, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voicemail.conf
Duraid Abbas wrote: I have set it on my box but it didn't work either. I have it set up like this: [mailbox number] = [password],[name],[email],delete=yes [mailbox number] = [password],[name],[email],[pageremail],delete=yes So, you probably want something like: 100 = 1234,Test User,[EMAIL PROTECTED],,delete=yes Regards Darryl ___ 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] Re: call progress - what are the sticking points?
Joe Greco wrote: More difficult is the problem of knowing when the remote end has gone away. Reversal, loop break, dial tone, and just plain silence are not all that unusual as methods of detection. In some cases, you do actually need to infer that the remote has gone away. I understand that the phone company (sometime) doesn't provide information about remote hangup on POTS lines. What bugs me is the simple question - how does your average 10$ answering machine detects the hang up? They don't, necessarily. I'm guessing the obvious - DSP and some heuristics as to what a hangup sounds like and it sounds to me that it isn't all that hard to do in Asterisk (since it's done in those cheap machines) but I would be very glad to hear some tips from someone that knows a little better then me. Answering machines get by on several mechanisms. The ones that come to mind are: 1) Silence detection. 2) Session time limit. Both of these are effective at doing something vaguely right within the requirements of an answering machine. If you've never heard an answering machine that's recorded a minute's worth of dialtone followed by the loud the phone is off the hook tone, then I'm shocked. :-) Just because you can engineer around a problem doesn't make the solution right. ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] Re: call progress - what are the sticking points?
Hi Joe, On Thu, 28 Oct 2004 16:41:06 +0200, Gilad Ben-Yossef [EMAIL PROTECTED] wrote: Joe Greco wrote: More difficult is the problem of knowing when the remote end has gone away. Reversal, loop break, dial tone, and just plain silence are not all that unusual as methods of detection. In some cases, you do actually need to infer that the remote has gone away. I understand that the phone company (sometime) doesn't provide information about remote hangup on POTS lines. What bugs me is the simple question - how does your average 10$ answering machine detects the hang up? Asterisk detects hangups with busydetect and busycount just fine. At least for me. The problem is ANSWER detection for billing purposes. -- Nicolás Gudiño Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P hardware problems
Dear All, I am using TDM400P card for about 4 months right now with 4 FXO modules. During this 4 months of use I needed to shutdown server (power off) becuase the card just stopped working. The card did not picked up calls, on console there is nothing. Shutting down asterisk does not help or loding drivers again. The server has to be powerd off and then turned on again. Could it be hardware (TDM400P) problem ? Did somebody notice the same problems ? Bartosz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No dial tone from fxs port
HengWee Chin wrote: Hi, I am having problem with the fxs port. I have compiled the zaptel, zapata and asterisk version 1.0.1. But after I start asterisk, I do not hear any dial tone coming on the fxs port. I am not able to dial out too. Perhaps you could post your zaptel.conf and zapata.conf settings. Most likely there's something mixed up somewhere. Are the modules loaded correctly? Which card are you using? Flynn Thanks Flynn, I have narrow it down to be the PCI slot. Because if I was to move the TDM400P card to another slot, I will not have any problem. The system that I am currently working on is a 5 5-volt PCI slot motherboard. For some reasons, the PCI slot 1 and 2 gives me problem. I have checked the interrupts, for slot 1 and slot 2 they have dedicated interrupts that is not shared. I even tried to change the interrupts via cmos. Nothings seems to work. Perhaps it is something wrong with the motherboard or the slot. The strange thing is that, zaptel is able to detect the port on slot 1 and 2. Just that I cannot heard any dial tone on the FXS port. I would like to know is there anyone have any success stories getting asterisk up and running on a 5 5-volt PCI slot using system. If possible, can share the brand and model of the motherboard. Regards, Chin ___ 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] Re: call progress - what are the sticking points?
Joe Greco wrote: Stephen David wrote: i don't have a specific bug in mind, i was just wondering WHY call progress doesn't work so well -- in particular, on analog lines. ie. is it a hardware or software problem (or both). with more info, i'd like to help to work out the kinks, for myself and everyone. :) Back in the days of Stowger exchanges you knew when the called party answered, by a reversal of the DC voltage on your analogue line. With digital exchanges that stopped, and no solid feedback is given to the caller on ordinary analogue lines. You have to infer that someone has answered, and the reliability of that can be poor. Digital lines, like ISDNand SS7, and protocols like MFC/R2 tell you positively that someone has answered. That's a good explanation. I'll expand upon it a bit by saying that even with reversal, there's a limited amount of information you can represent with that. POTS was always intended to be cheap basic phone service, and keeping it simple was not considered a downside by the phone company. As it is, you run into an information representation issue with the existing technology: the entire traditionally used bandwidth of the channel during a call is used for audio data (that is, to say, that they send an analog signal). As a call originator, you really can not tell the difference between a ringing signal generated by the phone company and a ringing signal caused by the called party picking up the phone and playing an identical sound. Reversal fixed that, but was largely made obsolete by out of band supervision - since the real purpose of reversal was for the telephone company to be able to bill correctly for completed calls (IIRC, ICBW). Actually it was not really intentional. The reversal back to the calling party was just a byproduct of the way a Strowger exchange worked. Within the network it was used for billing purposes. Okay, I remembered that it had been used for billing, and started to disappear with the advent of digital... sometimes hard to remember which came first, the chicken or the egg. ;-) More difficult is the problem of knowing when the remote end has gone away. Reversal, loop break, dial tone, and just plain silence are not all that unusual as methods of detection. In some cases, you do actually need to infer that the remote has gone away. Hangup is relatively easy. Most lines now give a strong distinct beeping either the moment the phone is dropped, or a short time after. The problem in * is its detector is not very good, or very voice immune. I have a much better one in my spandsp library, but it isn't integrated with * right now. Detecting answer is the tough one. There is nothing unambiguous about it. Yup. In fact, I just picked up one of these Sipura 3000's, and it's really kind of interesting, the phone rings twice (to gather CID) and then the Sipura answers and continues to generate ringing tones while it does SIP stuff. Weird. There's no real excuse for us to be using this technology anymore, with the availability of things like ISDN BRI, which allows for digital signalling of call progress. However, we continue to use it because the ILEC's have done such a fab job of making ISDN a dead technology. Funny thing is, it'll end up biting them where it hurts, as customers drift to VoIP to gain the features that ISDN promised, at a fraction of the cost. As someone whose colleagues built one of the first ISDN muxes in the 80's, I can tell you attitudes made it dead from day one. I know, I know. What a damn shame. They've been trying to kill DSL too, but I think the realization has finally dawned that the cable company (and soon the power company, and maybe next year the water company) are all working to bring high speed access. With that will inevitably come VoIP. It won't be a serious contender in the short term, but at the point where the broadband technology is stable and 98% reliable, and VoIP has mature E911 support, many people will feel fine giving up their land lines in exchange for a cell phone plus VoIP. In the meantime, I'm looking for a cheap but good BRI-to-SIP gateway. Just ran across the Patton SmartNode 1200. Anyone know anything about this? :-) ... JG -- Joe Greco - sol.net Network Services - Milwaukee, WI - http://www.sol.net We call it the 'one bite at the apple' rule. Give me one chance [and] then I won't contact you again. - Direct Marketing Ass'n position on e-mail spam(CNN) With 24 million small businesses in the US alone, that's way too many apples. ___ 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] WRT54GS zaptel timing device
Steve Underwood wrote: Kristian Kielhofner wrote: Hello, I know that I can run Asterisk on the Linksys WRT54GS, but can I do Zaptel as well? I would really like a timing device so I can do IAX2 trunking - but I don't know how to go about it. Has anyone done this? You *know* it can? Does that mean you've actually seen it work? A number of people have said it can run, but after questioning it turns out to be hearsay. You can replace the Linksys image with one of serveral. You can build Asterisk with the tools. You can load that image into the Linksys box. You can run it. Then you seem to get stuck, because something fouls up in the threading and nothing works. Anyone who can post a real way to get Asterisk running on these Linksys boxes will be considered a hero :-) Steve Steve, My application for Asterisk on the WRT is for remote offices where you could have SIP phones register with Asterisk on the WRT, and it would do IAX2 trunking back to the main Asterisk machine. No voicemail, meetme, transcoding, cdr, etc. Just a few modules. But I need Zaptel and I timer even for this limited application (IAX2 trunking). -- Kristian Kielhofner ___ 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] WRT54GS zaptel timing device
Paul Rodan wrote: Curious, can a WRT54G with this firmware, using Wonder Shaper, act as a cheap QOS device? One of our biggest problems is customers with Cable/DSL (256k upload) trying to upload files or browse several webpages at once, it affects the quality of the phone calls, naturally. We were looking into cheap managed switches for their QOS ability, but it seems the LinkSys would be a cheaper/easier solution. You can use OpenWRT and wondershaper (tc), or Sveasoft if you want something simpler. I have been told that even the LinkSys firmware includes QoS support. -- Kristian Kielhofner ___ 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] WRT54GS zaptel timing device
Paul Rodan wrote: Using zaptel, zaprtc and rtcsetup you can get simple Zaptel timing, no usb ports or zaptel hardware required. Zaprtc is a simple hack to the rtc driver. Compile your kernel without rtc support, compile this module, and load it along with zaptel. Then run rtcsetup to put it in the background and you've got Zaptel timing. However, if Asterisk has problems running on it, then it's useless. The problem is, you need to have an RTC for zaprtc to work. The WRT does not :) -- Kristian Kielhofner ___ 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] WRT54GS zaptel timing device
At 09:18 AM 10/28/2004, you wrote: Curious, can a WRT54G with this firmware, using Wonder Shaper, act as a cheap QOS device? Yes. Join the Sveasoft forum and there is good info on this. http://www.sveasoft.com/ I personally have not tried the QOS yet but it is supposed to work. Tom One of our biggest problems is customers with Cable/DSL (256k upload) trying to upload files or browse several webpages at once, it affects the quality of the phone calls, naturally. We were looking into cheap managed switches for their QOS ability, but it seems the LinkSys would be a cheaper/easier solution. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Thursday, October 28, 2004 9:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device http://www.pbs.org/cringely/pulpit/pulpit20040527.html - Original Message - From: Steve Underwood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 9:27 AM Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device Kristian Kielhofner wrote: Hello, I know that I can run Asterisk on the Linksys WRT54GS, but can I do Zaptel as well? I would really like a timing device so I can do IAX2 trunking - but I don't know how to go about it. Has anyone done this? You *know* it can? Does that mean you've actually seen it work? A number of people have said it can run, but after questioning it turns out to be hearsay. You can replace the Linksys image with one of serveral. You can build Asterisk with the tools. You can load that image into the Linksys box. You can run it. Then you seem to get stuck, because something fouls up in the threading and nothing works. Anyone who can post a real way to get Asterisk running on these Linksys boxes will be considered a hero :-) Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] how-to invoke the Busy voice mailbox menu in Asterisk
Hi everyone I have two user agents communicating with each other. In case another user calls to any one of those user agents i want to show asterisk that they are busy and therefore cannot entertain the call. Currently i am sending a 486 Busy Here message, but that makes asterisk invoke the Not Available voice mailbox menu. Where as i want asterisk to invoke the Busy voice mainlbox menu. Which message should be sent by my user agents to do this. thank you __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: call progress - what are the sticking points?
On Thu, 28 Oct 2004, Nicolás Gudiño wrote: Asterisk detects hangups with busydetect and busycount just fine. At least for me. The problem is ANSWER detection for billing purposes. Does asterisk support polarity reversal detection for answer/disconnect supervision? For a quick look at the source it does not appear to do so, only as a CallerId trigger. It does not look that hard to add since a polarity message is already sent from the zaptel driver. However, I have no such cards so I can not try myself. We use isdn. :) Or maybe I am wrong and Asterisk does support polarity reversal supervision. Peter ___ 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] TDM400P hardware problems
- Original Message - From: Bartosz Jozwiak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 10:55 AM Subject: [Asterisk-Users] TDM400P hardware problems | Dear All, | | I am using TDM400P card for about 4 months right now with | 4 FXO modules. | During this 4 months of use I needed to shutdown server (power off) becuase | the card just stopped working. | The card did not picked up calls, on console there is nothing. | Shutting down asterisk does not help or loding drivers again. | The server has to be powerd off and then turned on again. | | Could it be hardware (TDM400P) problem ? | Did somebody notice the same problems ? | | Bartosz | ___ I was just going to ask if this has been fixed yet I have the exact same card, hasn't worked properly yet (with respect to the reboot situation). the same behaviour happens with zaptel channels configured or not what version of the zaptel drivers are you using? Greg ___ 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] Re: call progress - what are the sticking points?
Joe Greco wrote: Answering machines get by on several mechanisms. The ones that come to mind are: 1) Silence detection. 2) Session time limit. Both of these are effective at doing something vaguely right within the requirements of an answering machine. If you've never heard an answering machine that's recorded a minute's worth of dialtone followed by the loud the phone is off the hook tone, then I'm shocked. :-) I never did. I also never owned an answering machine :-) Just because you can engineer around a problem doesn't make the solution right. Agreed and I was not thinking about this as a solution, but rather as a better kludge then my current method of using session time limit which is good enough for voice mail but is not good enough with conference bridge. Cheers, Gilad -- Gilad Ben-Yossef [EMAIL PROTECTED] Codefidence. A name you can trust(tm) Web: http://codefidence.com | SIP: [EMAIL PROTECTED] Tel: +972.9.8650475 ext. 201 | Fax: +972.9.8850643 I am Jack's Overwritten Stack Pointer -- Hackers Club, the movie ___ 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] TDM400P hardware problems
- Original Message - From: Cirelle Enterprises [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 12:15 PM Subject: Re: [Asterisk-Users] TDM400P hardware problems - Original Message - From: Bartosz Jozwiak [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 10:55 AM Subject: [Asterisk-Users] TDM400P hardware problems | Dear All, | | I am using TDM400P card for about 4 months right now with | 4 FXO modules. | During this 4 months of use I needed to shutdown server (power off) becuase | the card just stopped working. | The card did not picked up calls, on console there is nothing. | Shutting down asterisk does not help or loding drivers again. | The server has to be powerd off and then turned on again. | | Could it be hardware (TDM400P) problem ? | Did somebody notice the same problems ? | | Bartosz | ___ I was just going to ask if this has been fixed yet I have the exact same card, hasn't worked properly yet (with respect to the reboot situation). the same behaviour happens with zaptel channels configured or not what version of the zaptel drivers are you using? Greg ___ I have just installed Asterisk and Zaptel 1.0.2 So I will see if this helps. Sometimes I even here clicking noise and I need to power off server then everything goes back to normal. My server is Dell. Bartosz ___ 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] MFC/R2 Argentina variant ANI problems
I'm trying to get ANI info without succes in the Unicall channel. Apparently (as told by the PBX technical support), Argentina R2 implementation of ANI request needs a different answer than the current UniCall support. When Asterisk receives a 0x5 (ANI request) it must answer with an 0x1 and then the individual digits of the actual extension number (601 in my case). Unicall only answers the extension number, generating a protocol error. Also, when the call is established by asterisk, ANI request isn't generated at all. Of course, I'll try to fix the mising 0x1 myself, at least if I find where to add the extra digit :) Any help appreciated. Guillermo _ Charla con tus amigos en línea mediante MSN Messenger: http://messenger.latam.msn.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400P hardware problems
- Original Message - From: Bartosz Jozwiak [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 11:36 AM Subject: Re: [Asterisk-Users] TDM400P hardware problems | | | - Original Message - | From: Cirelle Enterprises [EMAIL PROTECTED] | To: Asterisk Users Mailing List - Non-Commercial Discussion | [EMAIL PROTECTED] | Sent: Thursday, October 28, 2004 12:15 PM | Subject: Re: [Asterisk-Users] TDM400P hardware problems | | | | - Original Message - | From: Bartosz Jozwiak [EMAIL PROTECTED] | To: [EMAIL PROTECTED] | Sent: Thursday, October 28, 2004 10:55 AM | Subject: [Asterisk-Users] TDM400P hardware problems | | | | Dear All, | | | | I am using TDM400P card for about 4 months right now with | | 4 FXO modules. | | During this 4 months of use I needed to shutdown server (power off) | becuase | | the card just stopped working. | | The card did not picked up calls, on console there is nothing. | | Shutting down asterisk does not help or loding drivers again. | | The server has to be powerd off and then turned on again. | | | | Could it be hardware (TDM400P) problem ? | | Did somebody notice the same problems ? | | | | Bartosz | | ___ | | I was just going to ask if this has been fixed yet | | I have the exact same card, hasn't worked properly yet (with | respect to the reboot situation). | | the same behaviour happens with zaptel channels configured or not | | what version of the zaptel drivers are you using? | Greg | | ___ | | I have just installed Asterisk and Zaptel 1.0.2 | So I will see if this helps. | Sometimes I even here clicking noise and I need to power off server | then everything goes back to normal. My server is Dell. | | Bartosz | This is what I get just trying to modprobe wcfxo /lib/modules/2.4.22-1.2199.nptlcustom/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.22-1.2199.nptlcustom/misc/wcfxo.o: insmod /lib/modules/2.4.22-1.2199.nptlcustom/misc/wcfxo.o failed /lib/modules/2.4.22-1.2199.nptlcustom/misc/wcfxo.o: insmod wcfxo failed It appears the card does not release it's IO unless it is powered down, we are using a P4SPA+ board Greg ___ 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] TDM400P hardware problems
Hi, | I am using TDM400P card for about 4 months right now with | 4 FXO modules. | During this 4 months of use I needed to shutdown server (power off) | the card just stopped working. I have the exact same card, hasn't worked properly yet (with respect to the reboot situation). Sometimes I even here clicking noise and I need to power off server then everything goes back to normal. My server is Dell. These cards have always been problematic for me. Make sure the cards aren't sharing interrupts. Always make sure you use filtered power (UPS or a good powerbar). You might get lucky and find that re-seating the card or changing PCI slots helps. Failing that, try using a completely different PC. We've found a system from Shuttle (all intel components) that tends to keep the card very stable - so we standardize on it when building Asterisk boxes. You'll probably find that stopping * and re-loading the card module (wcfxs) will correct the issue (temporarily). Aside from that, make sure your customers know how to hit ctrl-alt-del on the pbx... Ryan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voicemail.conf
Yup, it worked. Thanks. I have another question. My fromstring is not working would you know why? But email body is working. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darryl Ros Sent: Thursday, October 28, 2004 10:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail.conf Duraid Abbas wrote: I have set it on my box but it didn't work either. I have it set up like this: [mailbox number] = [password],[name],[email],delete=yes [mailbox number] = [password],[name],[email],[pageremail],delete=yes So, you probably want something like: 100 = 1234,Test User,[EMAIL PROTECTED],,delete=yes Regards Darryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Using AVM C4 with fewer than four lines?
On Thursday, 28 October, 2004 14:08 : Louis van Dompselaar [EMAIL PROTECTED] wrote: I think changing devices=4 to devices=2 in capi.conf should do the trick. Of course you have to make sure that your ISDN lines are connected to the two ports that haven't been disabled. It's already at devices=2 but that doesn't make a difference. I think chan_capi sees the C4 as one single device with four controllers. Anyway, the current capi.conf has devices=2 (tried 4 and 1; doesn't matter) and controller=1 (tried 1, 2 and 1,2,3,4, doesn't matter). I always get: If your devices= is to low, for example =1, then when you receive a 2nd call, you'll get the following error : ERROR [3075]: chan_capi.c:1696 capi_handle_msg: did not find device for msn = 1234xxx and the caller get a busy signal. -- Carl ___ 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] WRT54GS zaptel timing device
Ahh, well then, that would be a problem. Doesn't it seem minor compared to the problem that nobody has successfully gotten Asterisk to run for an extended time on the WRT54G? Do you already have Asterisk successfully running on one? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kristian Kielhofner Sent: Thursday, October 28, 2004 11:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] WRT54GS zaptel timing device Paul Rodan wrote: Using zaptel, zaprtc and rtcsetup you can get simple Zaptel timing, no usb ports or zaptel hardware required. Zaprtc is a simple hack to the rtc driver. Compile your kernel without rtc support, compile this module, and load it along with zaptel. Then run rtcsetup to put it in the background and you've got Zaptel timing. However, if Asterisk has problems running on it, then it's useless. The problem is, you need to have an RTC for zaprtc to work. The WRT does not :) -- Kristian Kielhofner ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MFC/R2 Argentina variant ANI problems
Hi Guillermo, The reason I don't send ANI when I get a 5 for .ar protocol is the information I have says something different :-) It says that when I get A5 I should respond with the calling party category, which I think I do. The current code seems to work OK against a Dialogic card in .ar mode. R2 documentation is somewhat unreliable in various countries. I will try to sort this out at the weekend. I have an update I am trying to finish off, with various fixes in it. Steve Guillermo Freige wrote: I'm trying to get ANI info without succes in the Unicall channel. Apparently (as told by the PBX technical support), Argentina R2 implementation of ANI request needs a different answer than the current UniCall support. When Asterisk receives a 0x5 (ANI request) it must answer with an 0x1 and then the individual digits of the actual extension number (601 in my case). Unicall only answers the extension number, generating a protocol error. Also, when the call is established by asterisk, ANI request isn't generated at all. Of course, I'll try to fix the mising 0x1 myself, at least if I find where to add the extra digit :) Any help appreciated. Guillermo ___ 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] TDM400P hardware problems
- Original Message - From: Ryan Courtnage [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 28, 2004 11:58 AM Subject: Re: [Asterisk-Users] TDM400P hardware problems | Hi, | | | I am using TDM400P card for about 4 months right now with | | 4 FXO modules. | | During this 4 months of use I needed to shutdown server (power off) | | the card just stopped working. | | I have the exact same card, hasn't worked properly yet (with | respect to the reboot situation). | | Sometimes I even here clicking noise and I need to power off server | then everything goes back to normal. My server is Dell. | | These cards have always been problematic for me. | | Make sure the cards aren't sharing interrupts. The card is in a dedicated slot | Always make sure you use filtered power (UPS or a good powerbar). Always filtered, using APC | You might get lucky and find that re-seating the card or changing PCI doesn't help | slots helps. Failing that, try using a completely different PC. | Don't think so - the board shouldn't be made as a general use component if it doesn't work properly in a universal implementation (PCI). The only requirement is the bios has to be at least level 2.2 and a min of 3.3 V |snip | You'll probably find that stopping * and re-loading the card module | (wcfxs) will correct the issue (temporarily). Asterisk has nothing to do with zaptel Aside from that, make | sure your customers know how to hit ctrl-alt-del on the pbx... ctrl-alt-del, reboot from the command line same thing the card needs power off, even though the board still has power applied Greg ___ 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] WRT54GS zaptel timing device
Paul Rodan wrote: Ahh, well then, that would be a problem. Doesn't it seem minor compared to the problem that nobody has successfully gotten Asterisk to run for an extended time on the WRT54G? Do you already have Asterisk successfully running on one? Paul, I am going to try later this evening, with a very stripped down version of Asterisk - see some of my earlier posts. -- Kristian Kielhofner ___ 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] re: Linksys PAP2-NA
Brian, I would like to know what approach you took? Im working on the software hack approach but that 8k flash chip on the back is starting to look tempting to me just for kicks. Did you literally toast the pap2? Dooz Owings -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian C. Fertig Sent: Friday, October 08, 2004 1:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] re: Linksys PAP2-NA Good luck.. I made a PAP2 and RT31P2 smoke when I was trying to unlock them.. It didn't work. I gave up and bought 2 Sipura's.. .o---o. Brian Fertig Network Engineer Planet Telecom, Inc. Tampa, FL Office 813.901.5182x107 Office 813.864.3164 Direct 813.817.9961 Cellular 813.881.9762 Fax Web: www.planet-telecom.com email: [EMAIL PROTECTED] --IM's--- MSN: [EMAIL PROTECTED] AIM: ptelebrian Yahoo: ptele_brian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Friday, October 08, 2004 3:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] re: Linksys PAP2-NA If it indeed is Sipura, I may be able to give a try to reset the box, if you can put it on a public IP. I know how to unlock VoicePulse,which is SIPURA so this may not be difficult. Seshu Kanuri [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of steve szmidt Sent: Friday, October 08, 2004 3:19 PM To: Brandon Patterson; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] re: Linksys PAP2-NA On Friday 08 October 2004 03:03 pm, Brandon Patterson wrote: MessageThat is correct. There is no way to unlock the box. This is technically incorrect as there IS a way to unlock it. All you need is the password. There might even be other ways to get around it. Google on PAP2 and unlock to see about 100 pages on the subject. -- Steve Szmidt They that would give up essential liberty for 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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ 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 attachment: winmail.dat___ 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] Need Asterisk to generate ringing tone on inbound SIP calls
Hello I have an SIP carrier defined on my Asterisk which delivers DID calls direct to extensions. The extensions are all either SNOM 200's or Cisco 7905's (SIP). The SIP carrier sends the extension number only in the invite, this then rings the phone. Below is an ethereal trace of an inbound call from the SIP carrier to extension 204 on *: SourceDestination Protocol Info xxx.xxx.xxx.xxx xx.xxx.xx.xx SIP/SDP Request: INVITE sip:[EMAIL PROTECTED], with session description xx.xxx.xx.xx xxx.xxx.xxx.xxx SIP Status: 100 Trying xx.xxx.xx.xx xxx.xxx.xxx.xxx SIP Status: 180 Ringing xx.xxx.xx.xx xxx.xxx.xxx.xxx SIP/SDP Status: 200 OK, with session description xxx.xxx.xxx.xxx xx.xxx.xx.xx SIP Request: ACK sip:[EMAIL PROTECTED] The problem I have is that the SIP carrier is playing a non-UK ring tone from a media gateway back in their network somewhere. They are playing ringing because asterisk sends them a Status 180 and expects ringing to be played at source. Since the call originates from the PSTN, source is actually their media gateway where they meet the PSTN. Callers are complaining as they are expecting to hear a UK ring tone since they are calling us in the UK. I want to be able to configure asterisk to play its own UK style ring tone, and to send a Status 183 instead of a Status 180 back to the SIP carrier so that it opens the backward speech path and lets the caller hear my UK ring tone. The SIP carrier has confirmed that if a Status 183 is sent instead of a 180 they will allow the caller to hear whatever progress tone Asterisk plays. I have trawled the WIKI, and drawn a blank. Can anyone please point me in the right direction. Many Thanks Chris Joseph ___ 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] Multiple Bandwidth Providers and Asterisk
This is a possible near future situation of ours: 10 different bandwidth providers, each providing 2 T1s. 15 different call carriers about the US. Each BW provider has its own Cisco 2610 into which the T1s go. Then 1 ethernet cable from each router goes into * box. If caller A needs to go to carrier 1, can * tell how many calls are already on ethernet cable #2? Cause if ether #2 (the default) is full/close to full, * needs to send caller A to carrier 1 via a different ethernet port/provider. Any one in a similar situation with multiple bandwidth providers? How do you handle BW choice? Normally this would be done in a router. But the routers don't handle calls, they handle packets. So if the router sends packet #1 of call #1 out 1 provider and packet #2 out another, you get jitter. Help/Advice appreciated. Thanks, Matthew ___ 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] TDM400P hardware problems
On Thu, 2004-28-10 at 12:08 -0400, Cirelle Enterprises wrote: Don't think so - the board shouldn't be made as a general use component if it doesn't work properly in a universal implementation (PCI). I'm not arguing that it shouldn't ... just sharing my experiences. Could be your mobo, your power-supply, etc. I've had PCs that meet the requirements in which the tdm400p was very unstable (daily). | You'll probably find that stopping * and re-loading the card module | (wcfxs) will correct the issue (temporarily). Asterisk has nothing to do with zaptel You can't unload the wcfxs module if asterisk is running. You'll get Device or resource busy. ___ 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] Multiple Bandwidth Providers and Asterisk
-Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: Thursday, October 28, 2004 12:27 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Multiple Bandwidth Providers and Asterisk Each BW provider has its own Cisco 2610 into which the T1s go. Then 1 ethernet cable from each router goes into * box. If caller A needs to go to carrier 1, can * tell how many calls are already on ethernet cable #2? Cause if ether #2 (the default) is full/close to full, * needs to send caller A to carrier 1 via a different ethernet port/provider. Any one in a similar situation with multiple bandwidth providers? How do you handle BW choice? Just to make sure I understand: You want to know how to load balance your bandwidth among a number of ethernet interfaces with the emphasis on making sure that once a connection is established that it will use the same interface, right? If so here are a couple of links: * http://lartc.org/howto/ - Linux Advanced Routing Traffic Control Howto * http://lartc.org/howto/lartc.rpdb.multiple-links.html - Specific page regarding multiple links If that isn't your goal please disregard. Good luck, Robert Jackson ___ 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] TDM400P hardware problems
Some of the REV H boards have been problematic. If you have one of these and you are having trouble, you should contact Digium. Michael Crown Managing Partner The VoIP Connection http://www.thevoipconnection.com vox: 321.989.6728 ext. 611 fax: 321.989.0284 -Original Message- From: Ryan Courtnage [mailto:[EMAIL PROTECTED] Sent: Thursday, October 28, 2004 11:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM400P hardware problems Hi, | I am using TDM400P card for about 4 months right now with | 4 FXO modules. | During this 4 months of use I needed to shutdown server (power off) | the card just stopped working. I have the exact same card, hasn't worked properly yet (with respect to the reboot situation). Sometimes I even here clicking noise and I need to power off server then everything goes back to normal. My server is Dell. These cards have always been problematic for me. Make sure the cards aren't sharing interrupts. Always make sure you use filtered power (UPS or a good powerbar). You might get lucky and find that re-seating the card or changing PCI slots helps. Failing that, try using a completely different PC. We've found a system from Shuttle (all intel components) that tends to keep the card very stable - so we standardize on it when building Asterisk boxes. You'll probably find that stopping * and re-loading the card module (wcfxs) will correct the issue (temporarily). Aside from that, make sure your customers know how to hit ctrl-alt-del on the pbx... Ryan ___ 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
