Re: [Asterisk-Users] spandsp rxfax under Linux 2.6 w/TDM400?
On Mon, 2005-03-28 at 18:18 -0800, Derrick Knight wrote:
I have got my Asterisk server running with TDM400 card (2xFXO 2xFXS).
I originally had the system configured with a Panasonic fax machine on
one of the extensions. Due to the high volume of fax spam, I figured it
would be a much better idea to capture the faxes as TIF or PDF files to
minimize wasted paper, etc. I have downloaded, compiled and installed
spandsp and can see the rxfax and txfax applications from within
Asterisk. When a fax comes in, it calls the rxfax application and then
does nothing. Here is my log:
Connected to Asterisk CVS-HEAD-03/28/05-16:44:11 currently running on
video (pid = 22837)
video*CLI
-- Starting simple switch on 'Zap/3-1'
-- Executing SetMusicOnHold(Zap/3-1, default) in new stack
-- Executing Answer(Zap/3-1, ) in new stack
-- Executing Dial(Zap/3-1, Zap/1|15) in new stack
-- Called 1
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Redirecting Zap/3-1 to fax extension
-- Hungup 'Zap/1-1'
-- Executing Macro(Zap/3-1, faxreceive) in new stack
-- Executing SetVar(Zap/3-1, FAXFILE=/data/fax/1112059186.0.tif)
in new stack
-- Executing RxFAX(Zap/3-1, /data/fax/1112059186.0.tif) in new stack
-- Hungup 'Zap/3-1'
I am running:
Fedora Core 3 (2.6.10-1.770_14)
Asterisk (CVS-HEAD-03/28/05-16:44:11)
SpanDSP 0.0.2pre10
libtiff 3.5.7 (including dev source)
Any guidance would be appreciated.
This is what I'm using albeit on an X100, that does work, I seem to
remember having trouble with the macro so used this instead.
[analog-in]
;
exten = s,1,System(/bin/echo -n -e '@CALL${CALLERIDNAME} ~
${CALLERIDNUM}' | nc -q0 -w1 192.168.1.161 10629)
exten = s,2,Answer
exten = s,3,GotoIf($[${CALLERIDNUM} = ${DAVE_MOBILE}]?disa,s,1)
exten = s,4,SetCallerID(9${CALLERIDNUM})
exten = s,5,Dial(${ALLPHONES},20,tr)
exten = s,6,Voicemail(u${INSTITUTE_VM})
exten = s,7,Hangup
exten = s,106,Voicemail(b${INSTITUTE_VM})
exten = s,107,Hangup
;
exten = fax,1,SetCallerID(${CALLERIDNUM})
exten = fax,2,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = fax,3,rxfax(${FAXFILE})
exten = fax,4,System(/usr/bin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERID})
exten = fax,5,Hangup
;
Dave Cotton
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[Asterisk-Users] Voicetronix OpenSwitch12 chan_vpb problem
Hello all, I hope this is not off-topic, if it is please let me know. I'm currently playing with an Asterisk at home, in order to get to know it's ins and outs. Very very impressive indeed. I've got it hooked up to my home phone line via a Wildcard clone board (Intel modem with Ambient chipset), and it works like a charm. Zaptel picks up the card as a generic clone, and works with it without any problems. The versions of Asterisk I'm playing with ranges from stable 1.0.5 to the latest CVS versions of asterisk, libpri and zaptel. I've got a Voicetronix OpenSwitch12 card on loan, and I'm trying to get it to work with Asterisk. No matter how I try, it seems that I've got a bit of a lemon. Here's the problem so far, maybe somebody has stumbled across something similar and knows of a workaround or fix: The Voicetronix driver (version 2.4.0) compiles perfectly, and Asterisk links against it nicely too, resulting in the chan_vpb.o being built. All the test software that comes with the Voicetronix driver sees the card properly, and I can even use the little test PBX program that comes with it's troubleshooting toolbox (I forget the name, it's a tiny little C++ program) to get the card working as a very primitive PBX. In other words, the card works nicely. Btw, this works perfectly with my wildcard in place and the zaptel/wcfxo modules loaded, only Asterisk has not been started. Over to Asterisk, when I try to initialise this card with chan_vpb, my computer locks up solidly (with me ending up having to do a hardware reset). Asterisk starts up beautifully until it reaches chan_vpb, and then it simply hangs. Here is a short snip of what it looks like when I start up Asterisk: ---start--- [chan_mgcp.so] = (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Not found (No such file or directory) == MGCP Listening on 0.0.0.0:2727 == Using TOS bits 0 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) [chan_iax2.so] = (Inter Asterisk eXchange (Ver 2)) == Registered application 'IAX2Provision' == Manager registered action IAXpeers == Parsing '/etc/asterisk/iax.conf': Found == Parsing '/etc/asterisk/iax_additional.conf': Not found (No such file or directory) == Using TOS bits 0 == Binding IAX2 to '0.0.0.0:4569' == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == IAX Ready and Listening on 0.0.0.0 port 4569 == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' [chan_local.so] = (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_skinny.so] = (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Not found (No such file or directory) == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [chan_features.so] = (Feature Proxy Channel) == Registered channel type 'Feature' (Feature Proxy Channel Driver) [skipping chan_oss.so] [chan_modem_i4l.so] = (ISDN4Linux Emulated Modem Driver) [chan_phone.so] = (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [chan_vpb.so] = (VoiceTronix V6PCI/V12PCI/V4PCI API Support) == Parsing '/etc/asterisk/vpb.conf': Found Starting host DSP![1] ---end--- Weird... The last line is also present with you use vpbconf to look for Voicetronix card configuration, the only difference being that the computer doesn't lock up: ---start--- #/usr/sbin/vpbconf Starting host DSP![1] Cards detected:1 BOARD 1 vpb_pconf[0][0] = 1 vpb_pconf[0][1] = 1 vpb_pconf[0][2] = 1 vpb_pconf[0][3] = 1 vpb_pconf[0][4] = 1 vpb_pconf[0][5] = 1 vpb_pconf[0][6] = 1 vpb_pconf[0][7] = 1 vpb_pconf[0][8] = 0 vpb_pconf[0][9] = 0 vpb_pconf[0][10] = 0 vpb_pconf[0][11] = 0 MODEL : V12PCI DATE : 27/11/2003 REVISION : 02.11 SERIAL NUMBER : 34800061 STATIONS[1]: 0 1 2 3 4 5 6 7 TRUNKS[1]: 8 9 10 11 ---end--- Looking in /proc/interrupts shows up both my wildcard clone and the V12PCI card on different IRQ's, so it seems that there are not any clashing between the two. The only thing I can think of is that it's quite an old card (see the date and revision above), which might be an issue. Has anybody stumbled across something like this? If not, do you know who I can contact for help? This particular Asterisk installation is not a production machine at all, merely a getting-to-know-you exercise. However, it would be nice to know wether Voicetronix products are on the trusted list or not. Thanks in advance! -- Regards, Jan Henkins ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spandsp compilation error
Title: Spandsp compilation error Hello everybody, I'm trying to receive and sending faxes with asterisk using spandsp. But while compiling the spandsp0.0.2pre11 (tried also spandsp0.0.1), I get following errormessage: gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c t4.c -Wp,-MD,.deps/t4.TPlo -fPIC -DPIC -o .libs/t4.o In file included from spandsp.h:42, from t4.c:71: spandsp/arctan2.h: In function `arctan2': spandsp/arctan2.h:51: warning: type mismatch in implicit declaration for built-in function `fabs' t4.c: In function `t4_rx_end_page': t4.c:566: `COMPRESSION_CCITT_T4' undeclared (first use in this function) t4.c:566: (Each undeclared identifier is reported only once t4.c:566: for each function it appears in.) t4.c: In function `t4_rx_init': t4.c:915: `COMPRESSION_CCITT_T4' undeclared (first use in this function) t4.c:923: `COMPRESSION_CCITT_T6' undeclared (first use in this function) t4.c: In function `t4_rx_start_page': t4.c:972: `COMPRESSION_CCITT_T4' undeclared (first use in this function) t4.c:974: `TIFFTAG_T4OPTIONS' undeclared (first use in this function) t4.c:983: `COMPRESSION_CCITT_T6' undeclared (first use in this function) make[2]: ** [t4.lo] Erro 1 make[2]: Leaving directory `/usr/src/spandsp-0.0.2/src' make[1]: ** [all] Erro 2 make[1]: Leaving directory `/usr/src/spandsp-0.0.2/src' make: ** [all-recursive] Erro 1 Can anyone tell me what I'm doing wrong? I'm using Debian 3.0r3 with kernel 2.6.6 Thanks in advance, Dennie __This mail has been scanned for all known viruses by AXSWeb powered by SecuTeam NV. _ This mail has been scanned for all known viruses by AXS Mail powered by SecuTeam NV. Register for AXS Mail at http://www.secuteam.com! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaptel based timing for VoIP-only Asterisk
Hi, In a VoIP only environment, Asterisk has to use ztdummy to have any chance of playing back understandable audio files (without drops, hickups etc). I have been using ztdummy to some degree of success, but I also have a Wildcard TDM400P REV E/F Board 1 in the Asterisk machine I'm using. I'm not using this card for anything at all, but I'm wondering how to set it up for timing only. What do I have to do (I have no experience at all with zap channels and the zaptel.conf file)? -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Zaptel based timing for VoIP-only Asterisk
In article [EMAIL PROTECTED], Andreas Sikkema [EMAIL PROTECTED] wrote: Hi, In a VoIP only environment, Asterisk has to use ztdummy to have any chance of playing back understandable audio files (without drops, hickups etc). I have been using ztdummy to some degree of success, but I also have a Wildcard TDM400P REV E/F Board 1 in the Asterisk machine I'm using. I'm not using this card for anything at all, but I'm wondering how to set it up for timing only. What do I have to do (I have no experience at all with zap channels and the zaptel.conf file)? Just make sure the zaptel.conf file is set up correctly for the combination of FXS/FXO modules you have, and that the driver is successfully loaded. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sending many faxes simultaneously with spandsp
I have a potential client that wants to send many faxes simultaneously, over E1 trunks. How CPU intensive is spandsp's txfax? How many concurrent faxes could be sent by a decent CPU (e.g. Xeon 3GHz) before timing starts to get disrupted? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel based timing for VoIP-only Asterisk
Andreas, -Original Message- I have been using ztdummy to some degree of success, but I also have a Wildcard TDM400P REV E/F Board 1 in the Asterisk machine I'm using. I'm not using this card for anything at all, but I'm wondering how to set it up for timing only. What do I have to do (I have no experience at all with zap channels and the zaptel.conf file)? I'm not sure what would be enough to provide timing, but you can easily try by accessing a meetme without a timing source :) - Load the module, test if that is sufficient - Edit /etc/zaptel.conf to provide info about the card, execute ztcfg and test again - Edit zapata.conf to provide info about the card, test again. I am assuming chan_zap was already built for asterisk, it should do that automatically when it detects the libraries. You will need to restart asterisk every step (reloading does not cover Zaptel) Best regards, Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as gateway with oh323 channel to VOIP provider that can provide gateway or gatekeeper feature ?
Hi, sorry for my h323 dumbness. VOIP provider terminates H323 calls - it can be used as gatekeeper or gateway (they claim so). What option and what setup is best to connect Asterisk to this provider ? Any working examples ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outgoing call immediately disconnected
I have created a call file as shown in the files below. The number is dialled and connected (i.e. the call is placed to the PSTN) but it is immediately disconnected and I get the following message on the console: Starting Zap/3-1 at from-internal-custom,s,1 failed so falling back to exten 's' Extensions.conf [from-internal-custom] exten = s,1,Wait(20) exten = s,2,SendDTMF(1) etc 1.call Channel: Zap/3/1234567 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: from-internal-custom Extension: s Priority: 1 Does anyone have any suggestions as to why Asterisk is failing when it tries to access this extension in this context? Regards Cameron P.S. I tried sacrificing chickens but it didn't work ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Spandsp compilation error
Did you install libtiff libraries prerequisites before compiling It may be an issue with your LD_LIBRARY_PATH as files do not seem to be found. Selon Dennie Verstrepen [EMAIL PROTECTED]: Hello everybody, I'm trying to receive and sending faxes with asterisk using spandsp. But while compiling the spandsp0.0.2pre11 (tried also spandsp0.0.1), I get following errormessage: gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c t4.c -Wp,-MD,.deps/t4.TPlo -fPIC -DPIC -o .libs/t4.o In file included from spandsp.h:42, from t4.c:71: spandsp/arctan2.h: In function `arctan2': spandsp/arctan2.h:51: warning: type mismatch in implicit declaration for built-in function `fabs' t4.c: In function `t4_rx_end_page': t4.c:566: `COMPRESSION_CCITT_T4' undeclared (first use in this function) t4.c:566: (Each undeclared identifier is reported only once t4.c:566: for each function it appears in.) t4.c: In function `t4_rx_init': t4.c:915: `COMPRESSION_CCITT_T4' undeclared (first use in this function) t4.c:923: `COMPRESSION_CCITT_T6' undeclared (first use in this function) t4.c: In function `t4_rx_start_page': t4.c:972: `COMPRESSION_CCITT_T4' undeclared (first use in this function) t4.c:974: `TIFFTAG_T4OPTIONS' undeclared (first use in this function) t4.c:983: `COMPRESSION_CCITT_T6' undeclared (first use in this function) make[2]: ** [t4.lo] Erro 1 make[2]: Leaving directory `/usr/src/spandsp-0.0.2/src' make[1]: ** [all] Erro 2 make[1]: Leaving directory `/usr/src/spandsp-0.0.2/src' make: ** [all-recursive] Erro 1 Can anyone tell me what I'm doing wrong? I'm using Debian 3.0r3 with kernel 2.6.6 Thanks in advance, Dennie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Erratic CPU load
Hi, During tests with a IAX2/PSTN gateway I've been getting strange results for processor idle time and load. I (re)search(ed) this issue for a while, but I didn't get any good explainations. Can somebody help me? I have several sites that rely on a central server for connection to the PSTN. Calls to the PSTN are routed over the Internet to this PSTN gateway using IAX2 in trunk mode. To minimize bandwith usage, the Speex codec is used. The central PSTN gateway is a P4 3.0GHz, 1GByte mem, has a TE110P card supporting ISDN30 and runs Asterisk version 1.0.3 on Debian Sarge. While sustaining 5 connections dialed in through the TE110P (terminated at remote sites through IAX) and running top on the PSTN gateway, I see 98% CPU idle time most of the time. I also see short (around 10sec) bursts of high CPU usage (40-50%) by one of the asterisk processes supporting the connection. The bursts happen in irregular intervals, ranging from 30 to 60 sec. Meanwhile, the reported average load jumps up and down between 0.1 to 0.7. What's happening here? Is the processor load really this erratic, or am I looking at an artefact in cpu usage measurement? Maybe there is an aliasing effect caused by the periodic cpu load (20ms, default trunk frequency) and the cpu usage measurement (also periodic?), but I don't know how to check this. If this top reading is an artefact, is there a way to check the actual (realtime) load? Regarding the actual processor usage for speex encoding: this report suggests my processor is indeed quite busy encoding a few speex channels: http://astertest.com/astricon_performance.ppt. Given the results in this report, I doubt the PSTN gateway will support more than 10 speex encodings. At the same time, the same processor encodes 756x756 PAL television to mpeg-4 on my mythtv box at home. Twice, leaving room for scheduled jobs. Has anyone some references to documentation to put these figures into perspective? Thanks in advance, Eric. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] VoIP Provider problems
Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line about 2 Mb. We feel our quality decrease when in US are about 9:00 or 10:00 in the morning. We do not know if this is it correct or all the people using VoIp provider feel the same quality? Anyone knows any provider without this kind of problems? Witch provider do you use to get the best sounds quality? Any clue will be welcomed. Thanks for your time Obihuan. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Spandsp compilation error
I found my problem, I had installed out of date libraries of libtiff. Now it's running. But thanks anyway. -Oorspronkelijk bericht- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] Verzonden: dinsdag 29 maart 2005 11:55 Aan: Asterisk Users Mailing List - Non-Commercial Discussion Onderwerp: Re: [Asterisk-Users] Spandsp compilation error Did you install libtiff libraries prerequisites before compiling It may be an issue with your LD_LIBRARY_PATH as files do not seem to be found. Selon Dennie Verstrepen [EMAIL PROTECTED]: Hello everybody, I'm trying to receive and sending faxes with asterisk using spandsp. But while compiling the spandsp0.0.2pre11 (tried also spandsp0.0.1), I get following errormessage: gcc -DHAVE_CONFIG_H -I. -I. -I. -I -g -O2 -c t4.c -Wp,-MD,.deps/t4.TPlo -fPIC -DPIC -o .libs/t4.o In file included from spandsp.h:42, from t4.c:71: spandsp/arctan2.h: In function `arctan2': spandsp/arctan2.h:51: warning: type mismatch in implicit declaration for built-in function `fabs' t4.c: In function `t4_rx_end_page': t4.c:566: `COMPRESSION_CCITT_T4' undeclared (first use in this function) t4.c:566: (Each undeclared identifier is reported only once t4.c:566: for each function it appears in.) t4.c: In function `t4_rx_init': t4.c:915: `COMPRESSION_CCITT_T4' undeclared (first use in this function) t4.c:923: `COMPRESSION_CCITT_T6' undeclared (first use in this function) t4.c: In function `t4_rx_start_page': t4.c:972: `COMPRESSION_CCITT_T4' undeclared (first use in this function) t4.c:974: `TIFFTAG_T4OPTIONS' undeclared (first use in this function) t4.c:983: `COMPRESSION_CCITT_T6' undeclared (first use in this function) make[2]: ** [t4.lo] Erro 1 make[2]: Leaving directory `/usr/src/spandsp-0.0.2/src' make[1]: ** [all] Erro 2 make[1]: Leaving directory `/usr/src/spandsp-0.0.2/src' make: ** [all-recursive] Erro 1 Can anyone tell me what I'm doing wrong? I'm using Debian 3.0r3 with kernel 2.6.6 Thanks in advance, Dennie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ This mail has been scanned for all known viruses by AXS Mail powered by SecuTeam NV ... Register for AXS Mail at http://www.secuteam.com! __ This mail has been scanned for all known viruses by AXSWeb powered by SecuTeam NV. __ This mail has been scanned for all known viruses by AXSWeb powered by SecuTeam NV. _ This mail has been scanned for all known viruses by AXS Mail powered by SecuTeam NV ... Register for AXS Mail at http://www.secuteam.com! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP Provider problems
On Tue, 2005-03-29 at 12:36 +0200, Ismael Gil wrote: Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond) If you find the same problem with multiple ITSP's, then it may not be them that is at fault. We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line about 2 Mb. Sounds like it should be quite adequate... how many simultaneous calls are you doing? We feel our quality decrease when in US are about 9:00 or 10:00 in the morning. What time is that for your local time? Is there something that might be happening at/around that time for you? eg, here, around 3 - 6pm is quite busy as school kids get home and go on the internet, same for people getting home from work. In fact, my vague recollection is that things just get busier until around 11pm, before they really slow down. While this doesn't have any relation to *your* adsl connection, think about what this might be doing to your ISP's internet connection We do not know if this is it correct or all the people using VoIp provider feel the same quality? Not that I would know, but I get the feeling that most people get extremely good quality calls over a decent internet connection. Anyone knows any provider without this kind of problems? Witch provider do you use to get the best sounds quality? I've not used any, so can't comment on this. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ADTRAN TA 750 + TE405P + PRI with problem to receive or send fax.
I have a channel bank (TA750) and a PRI with 30 channels connected to a TE405P, in the channel bank I have a extension to a fax machine, but it doesn't work to send or receive fax. There are any advice ? Kind regards, Miguel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] changes to nat =yes?
Hi ne1 know if there has been recent (over easter) cvs changes to what happens when nat =yes especially in relation to sip some things seem to work for me that didn't before ;) thanks walt. _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk-oh323 pre-releases
I have prepared two new, not-final yet, releases of asterisk-oh323: - 0.6.6-pre1 for Asterisk stable - 0.7.2-pre1 for Asterisk CVS HEAD They can be found at: http://www.inaccessnetworks.com/projects/asterisk-oh323/download Please try them and report problems at the bugtracker of the channel driver at: https://skylab.inaccessnetworks.com/mantis Regards, Michael. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Fw: [Asterisk-Users] sip provider
Thore a écrit : Hi ! This work well for incoming calls, but not for outgoing call. Those i call get the wrong number in the display. Thanks to reply to list, not private Thore - Original Message - From: administrator tootai [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sunday, March 27, 2005 6:19 PM Subject: Re: [Asterisk-Users] sip provider Thore a écrit : Hi I have a voip provider use sip. To telephones with exten. 201 and 202. My voip provider give me this numbers 33297540 and 33297545. Is it possible to get exten 201 to ring out on 33297540 and 202 - 33297545 ? [...] ; Voip [33297540] type=friend host=voip.dk dtmfmode=rfc2833 canreinvite=no username=33297540 secret=nisse context=voip_incoming context=voip_incoming-phone1 nat=yes fromuser=33297540 fromdomain=voip.dk insecure=very [33297545] type=friend host=voip.dk dtmfmode=rfc2833 canreinvite=no username=33297545 secret=nisse context=voip_incoming context=voip_incoming-phone2 nat=yes fromuser=33297545 fromdomain=voip.dk insecure=very [...] [voip_incoming] exten = 33297540,1,Dial(Sip/201,120) exten = 33297540,2,Congestion exten = 33297545,1,Dial(Sip/202,120) exten = 33297545,2,Congestion [voip_incoming-phone1] exten = 33297540,1,Dial(Sip/201,120) exten = 33297540,2,Congestion [voip_incoming-phone2] exten = 33297545,1,Dial(Sip/202,120) exten = 33297545,2,Congestion [callfrom201] exten = s,1,SetCallerID(201) exten = s,2,Dial(SIP/[EMAIL PROTECTED],,) [callfrom202] exten = s,1,SetCallerID(202) exten = s,2,Dial(SIP/[EMAIL PROTECTED],,) -- Daniel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_darthvader.c?
Hi, on Alias the badguy(tm) on the phone usually sound like Darth Vader thanks to some cool device from marshall :-) Is something like this possible with asterisk, or, asked a little more generic, can i somehow pipe an rtp-stream to an application via STDOUT and read it back via STDIN? Greetings, aa _ Read the latest Hollywood gossip @ http://xtramsn.co.nz/entertainment ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problems with Suse Linux Enterprise !
Hello, i am running suse linux enterprise edition of kernel version 2.6.5-7.97-smp, i have latest stable asterisk zaptel asterisk stuff compile fines i have TDM400P card with 1FXS and 3FXO modules, every time i probe with modprobe and issue ztcfg -vv commandit shows the following errors: also issue modprobe wcfxs but no luck asterisk2:/lib # modprobe zaptel asterisk2:/lib # modprobe wct4xxp asterisk2:/lib # ztcfg -vv Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) Channel 03: FXS Kewlstart (Default) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 4 channels configured. ZT_CHANCONFIG failed on channel 1: No such device or address (6) i am also sets udev configuration files udev.rules and permissions.udev as describe on wiki am i doing something wrong. please i have want some quick tips suggestions guidelines. zaptel.conf: fxoks=1 fxsks=2-4 loadzone = us defaultzone=us Thanks in advance please helping me out. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk and XLite on same machine (OSX)?
why not using a IAX phone, is running great on OS X http://iaxclient.sourceforge.net/iaxcomm/ ·· Adrià Vidal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Erratic CPU load
Hi, Im the astertest guy. If you are referring to the graphs on page 41 - 42, please note that those are done on a embedded via 800mhz cpu and not on a system similar to yours. So i'm pretty sure you can do more than 10 speex encodings at the same time. (also some things changes since we did those tests, some optimizations and configuration options were done to the speex codec implementation in asterisk. Now you can choose your own quality vs cpu usage balance on asterisk) I suggest you go to cvs-head and check out the changes to the speex codec. Without changes to your asterisk machine, i estimate you will be able to do around 30 to 40 channels on your machine if it was without zaptel. Zaptel does not take a lot cpu especially compared to speex encodings. (i know from experience) About the periodic load, please see if any calls are being setup or tear down, or specific applications are used in those cases. Saddly enough, i still didnt find the time to do any load measurements on pri cards. Although i have a test setup ready to go. Zoa. Eric Giesselbach wrote: Hi, During tests with a IAX2/PSTN gateway I've been getting strange results for processor idle time and load. I (re)search(ed) this issue for a while, but I didn't get any good explainations. Can somebody help me? I have several sites that rely on a central server for connection to the PSTN. Calls to the PSTN are routed over the Internet to this PSTN gateway using IAX2 in trunk mode. To minimize bandwith usage, the Speex codec is used. The central PSTN gateway is a P4 3.0GHz, 1GByte mem, has a TE110P card supporting ISDN30 and runs Asterisk version 1.0.3 on Debian Sarge. While sustaining 5 connections dialed in through the TE110P (terminated at remote sites through IAX) and running top on the PSTN gateway, I see 98% CPU idle time most of the time. I also see short (around 10sec) bursts of high CPU usage (40-50%) by one of the asterisk processes supporting the connection. The bursts happen in irregular intervals, ranging from 30 to 60 sec. Meanwhile, the reported average load jumps up and down between 0.1 to 0.7. What's happening here? Is the processor load really this erratic, or am I looking at an artefact in cpu usage measurement? Maybe there is an aliasing effect caused by the periodic cpu load (20ms, default trunk frequency) and the cpu usage measurement (also periodic?), but I don't know how to check this. If this top reading is an artefact, is there a way to check the actual (realtime) load? Regarding the actual processor usage for speex encoding: this report suggests my processor is indeed quite busy encoding a few speex channels: http://astertest.com/astricon_performance.ppt. Given the results in this report, I doubt the PSTN gateway will support more than 10 speex encodings. At the same time, the same processor encodes 756x756 PAL television to mpeg-4 on my mythtv box at home. Twice, leaving room for scheduled jobs. Has anyone some references to documentation to put these figures into perspective? Thanks in advance, Eric. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users signature.asc Description: OpenPGP digital signature ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC-S
Hi! I have just installed Redhat 9 and Asterisk to my computer, and now i have problems with my non-zaptel Card, I don't know how to set it up since all instructions are for digium's hardware. I have searched from the Internet for hours now, can you help me to understand all this HFC-s thing and how it is related to CAPI, ISDN4Linux, bristuff and so on. I have to say that I am not so familiar with Linux. Thank you in advance This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call files run at certain times
I like your idea, Ill play with it for a while and see what comes out.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Martes, 29 de Marzo de 2005 12:36 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call files run at certain times
Ok,
so you have a context:
[mycallouts]
exten = _X.,1,Answer
exten = _X.,2,Background(mycall${EXTEN}) exten =
_X.,3,Dial(ZAP/g1/${EXTEN})
and when you do the record you can do
[myrecording]
exten = 98,1,Answer
exten = 98,2,Background(please_enter_99_followed_by_number)
exten = 99.,1,Record(mycall${EXTEN:2})
Or something similar. Hope that makes sense. So the record will create a
file called mycall5551234 or whatever the number is, and then from the call
file you'd send it to the context mycallouts with the extension set to
whatever the number was and then it would play the correct file, and then
call the number.
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] CIC Code
I guess I should have included more information to get assistance after
reading what I posted,
I am running MF not SS7, the local Telco does see me sending the appropriate
info but not in the correct protocol according to them. They say it looks
more like I am dialing the dial code and CIC code manually instead of
Asterisk sending it before the number dialed. So in essence asterisk needs
to send two packets, one with the 017CICCode and the second packet with the
phone number. Also I need to be sending ANI information as well. I am
stumped to what to do at this point and have tried everything I have found
on google and what Digium tech support suggested, if anyone could assist me
in this it would be greatly appreciated and if need be I would be willing to
pay for outsourced technical support if you have experience setting this up
since I need to get it up and going quickly instead of learning how to do
it. I can learn later... HEHE
Thanks again,
Jason
Red Hat Linux release 9 (Shrike)
Kernel \r on an \m
2 Each Digium T100X
(Snippets of my config files, some phone/server specific info changed for
post IE sip username/secret, CICCode and default IP)
-Extensions.conf-
[general]
static=yes
writeprotect=no
[globals]
CONSOLE = Console/dsp
IAXINFO = guest
TRUNK = Zap/g1
TRUNKMSD = 1
TRUNK2 = Zap/g2
TRUNKMSD = 2
TRUNK3 = Zap/g3
TRUNKMSD = 3
exten = _1NXXNXX,1,Dial(Zap/g2/017CICCode${EXTEN:1})
exten = _1NXXNXX,2,Hangup()
-zapata.conf-
context=default
usecallerid=yes
callwaiting=yes
immediate=no
group=1
echocancel=yes
signalling=em
channel = 1-24
context=twoway
usecallerid=yes
callwaiting=yes
immediate=no
group=2
echocancel=yes
signalling=featdmf
channel = 25-36
context=incoming
usecallerid=yes
immediate=no
group=3
echocancel=yes
signalling=featb
channel = 37-48
-Zaptel.conf-
# Zaptel Configuration File
span=1,1,0,esf,b8zs
em=1-24
defaultzone=us
span=2,1,0,esf,b8zs
em=25-48
loadzone=us
defaultzone=us
- Sip.conf -
[User]
username=User
secret=nothing
type=friend
host=dynamic
defaultip=1.1.1.1
dtmfmode=info
context=incoming ;twoway ;default
canreinvite=no
disallow=all
nat=yes
allow=ulaw
allow=alaw
mailbox=107
- Lsmod -
Module Size Used byNot tainted
soundcore 6404 0 (autoclean)
wct1xxp13024 48
zaptel179712 98 [wct1xxp]
autofs 13268 0 (autoclean) (unused)
natsemi19552 1
keybdev 2944 0 (unused)
mousedev5492 0 (unused)
hid22148 0 (unused)
input 5856 0 [keybdev mousedev hid]
usb-uhci 26348 0 (unused)
usbcore78784 1 [hid usb-uhci]
ext3 70784 2
jbd51892 2 [ext3]
From: Tom Chandler [EMAIL PROTECTED]
Date: Mon, 28 Mar 2005 19:46:01 -0600
To: [EMAIL PROTECTED]
Subject: Fw: [Asterisk-Users] CIC Code
Jason,
If you get any answers, I too would be interested.
I believe on terminating, the CIC is not sent, AMA recording uses the
CIC assigned to the trunk group. If in SS7, then the CIC is passed
in the IAM message.
I have not worked on the originating side, so I can not help.
Thank You
Tom Chandler
- Original Message -
From: Jason Miller [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, March 28, 2005 7:22 PM
Subject: [Asterisk-Users] CIC Code
Has anyone ever setup Asterisk to pass Feature Group D access while using
a
CIC code for outbound calls? If so can you please email the configuration
you have done? I have tried to get this up and running but with no luck. I
have also contacted support and I cant seem to get this going.
Thanks in Advance,
Jason Miller
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Re: [Asterisk-Users] app_darthvader.c?
On Tue, 29 Mar 2005 11:33:16 +, Andreas Anderson [EMAIL PROTECTED] wrote: Hi, on Alias the badguy(tm) on the phone usually sound like Darth Vader thanks to some cool device from marshall :-) Is something like this possible with asterisk, or, asked a little more generic, can i somehow pipe an rtp-stream to an application via STDOUT and read it back via STDIN? Have you ever tried LPC10 codec? Sounds like Darth Vader to me. -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CIC Code
On Mon, 28 Mar 2005, Jason Miller wrote: Has anyone ever setup Asterisk to pass Feature Group D access while using a CIC code for outbound calls? If so can you please email the configuration you have done? I have tried to get this up and running but with no luck. I have also contacted support and I cant seem to get this going. I got it working briefly. I had to talk with the switch techs at the other end for a couple hours and modify the source code to reformat what was being sent down the line. Their definition of FGD and asterisk's definition were not the same. It was a nortel DMS-100 or 250 set up for CLASS 4. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
Thank you for your story Paul, nice work with the dialplans! I have one question, so you say that for server 2, asterisk is behind nat and you have sip clients inside and outside the nat. Which ports are you forwarding to asterisk from your firewall and in the case of sip clients outside nat, did you have to open certain ports for each client or all clients use the same? For inside clients it should be a charm! Very nice job Paul, intercity dialing and everything well connected... That was a good story.. Thx for sharing. Anton -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Martes, 29 de Marzo de 2005 12:52 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate - Original Message - From: Anton Krall [EMAIL PROTECTED] would like to hear some actual setups and how people are solving the nat issue within scenarios like: Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones I've been playing with this with my friends for awhile now. We've got four different Asterisk servers set up in four different cities: 1. 2 nics - one on internal network, other on external network. TDM400 card with 2 FXO and 1 FXS, 2 different analog lines, and a LiveVoip IAX2 dialout. Various SIP phones connected, both from within the internal network and out on the internet from behind other NATs. 2. 1 nic - behind NAT (ports forwarded). X100p with 1 analog line. Various SIP phones, internal network and from behind other NATs. 3 4. Like #2 but no X100p. All four servers are connected via IAX2 - in all cases we can dial extensions for each other's systems and the call gets dumped to the correct server. Also between server 1 2 we have local inter-city dialing working (if you dial an outside number that is local to the other city and don't put a 1 in front of the number it dumps to the other server and dials out). NAT hasn't proven to be a problem for us - the only thing we can't do as a result of all the SIP clients being natted is Reinvites - this just means that all conversation *must* go through the server as opposed to direct client-client transfer. Servers that are behind nats have the correct IP settings set in SIP.CONF. As long as I set the STUN server on the sip clients to a good working STUN server everything works like a hot damn. Nothing special regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on a dialup connection?
Hi! firefly(iLBC) works over dial-up connection. also u can use x-Lite (GSM/iLBC). C ya, Madhawa On Tue, 29 Mar 2005 16:26:13 +1200, Matt Riddell [EMAIL PROTECTED] wrote: Kerry Garrison wrote: This is what I get: speex - - - - - - - - - - - In other words, it's not installed. First install speex, then reinstall asterisk. Details on speex are available here: http://www.speex.org and Asterisk info is (obviously) here: http://www.asterisk.org -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This e-mail and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended addressee, or the person responsible for delivering it to them, you may not copy, forward disclose or otherwise use it or any part of it in any way. To do so may be unlawful. If you receive this e-mail by mistake, please advise the sender immediately. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CIC Code
Do you remember what you actually changed to make it work cause that is the same switch that I am dealing with myself if I am not mistaken. Thank you, Jason Miller From: Dave Weis [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 29 Mar 2005 06:23:35 -0600 (CST) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] CIC Code On Mon, 28 Mar 2005, Jason Miller wrote: Has anyone ever setup Asterisk to pass Feature Group D access while using a CIC code for outbound calls? If so can you please email the configuration you have done? I have tried to get this up and running but with no luck. I have also contacted support and I cant seem to get this going. I got it working briefly. I had to talk with the switch techs at the other end for a couple hours and modify the source code to reformat what was being sent down the line. Their definition of FGD and asterisk's definition were not the same. It was a nortel DMS-100 or 250 set up for CLASS 4. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SMS configuration
Hi, I've been trying to setup SMS on asterisk - would be useful to have for things like server outages, email from important customers, etc. I can send SMS with no issues, although I have to send it over the Zap line.. none of the VOIP providers will route the call. It arrives on my mobile phone a couple of minutes later (usually.. had to wait half an hour for one). Incoming however just isn't working. I've got a nice list of numbers from which SMS messages come: 080057875290 and 0845602111 - BT message centre I think 01612745990 - Mentioned in BT docs but never had one from there yet 07953966066 - T-Mobile phones SMS from here - I presume there are other ones for Orange, etc. but I don't have one of those phones to check. My incoming sms block is just: [sms_in] exten = s,1,SMS(sms,a) exten = s,2,NoOp exten = s,3,Hangup Outgoing is fine.. eg. smsq reset: -- Zap/1-1 answered Local/[EMAIL PROTECTED],2 Channel Local/[EMAIL PROTECTED],1 was answered. Lauching SMS(0) on Local/[EMAIL PROTECTED],1 -- SMS RX 93 00 6D -- SMS TX 91 0F 01 02 05 81 00 00 F0 00 F1 05 F2 F2 BC 4C 07 FE -- SMS TX 92 01 FF 6E -- SMS RX 95 09 01 00 50 30 62 10 40 42 00 ED -- SMS TX 94 00 6C -- SMS TX 92 01 FF 6E -- SMS RX 95 09 01 00 50 30 62 10 40 42 00 ED -- SMS TX 94 00 6C -- Hungup 'Zap/1-1' Mar 26 01:04:26 NOTICE[15806]: pbx_spool.c:244 attempt_thread: Call completed to Local/17094009 Then the 0 service tries to text me back with an OK message: -- Starting simple switch on 'Zap/1-1' -- Executing Goto(Zap/1-1, sms_in|s|1) in new stack -- Goto (sms_in,s,1) -- Executing SMS(Zap/1-1, sms|a) in new stack -- SMS TX 93 00 6D -- Hungup 'Zap/1-1' It continues doing this every few minutes until it eventually gives up about half an hour later. What seems to be happening is the SMS application is bailing out after the first line of the output. I've enabled debugging and verbose logging and there's nothing printed anywhere from the SMS app. Has anyone seen anything like this? Tony ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk@Home H323
I am looking for a step-by-step on adding H323 to [EMAIL PROTECTED] So far I have installed [EMAIL PROTECTED], upgraded to the CVS-HEAD and followed instructions according to voip-info and this list's archives. I keep getting critical errors on compilation of H323, both Open 323 and OH323. Has anyone managed to install H323 with [EMAIL PROTECTED] If so, what steps did you perform. With Thanks Mike - Original Message - From: Matt [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 11:41 AM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] 0.7 released Web Meetme is now installed by default and the meetme2 application is no longer needed. What does this mean exactly? Does this use the regular meetme as opposed to the meetme2 we had to setup before? On Mon, 28 Mar 2005 17:35:37 -0800 (PST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2 application is no longer needed. - we now have ZAP extension thanks to AMP 007 - weather.agi reads the current weather report using text to speech __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HFC PCI
Greetings to everyone! I am new to Asterisk and ISDN modules so i tried to follow many of the articles about capi, bristuff, mISDN and so on. Now I am working in mISDN but every way I try I have compiling errors! The HFC PCI card is a Digi Datafire Micro V. The bristuff give me the error invalid module format zaphfc.ko and I don't know how to compile and load it. The capi solution give me many many compiling errors. Some one can give me a good step by step how to or suggest witch is the best solution for an hfc pci card? Thanks in Advance Denis ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zultys 4x5 phone
Does anyone on-list have any advice about the version 2.2 firmware for the Zultys 4x5 phone? It has a new gateway mode that is supposed to direct calls on the analogue line to the PBX for VM, etc. Zultys has not yet presented any docs, and the keep refering me to the reseller who doesn't know anything about t at all. Thanks, Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fail over
For all my PBX installations I want to have Fail Over on the main incoming PSTN line so that a power outage does not leave the offices stranded. Is there any commercial solution to this? I would rather a finished product than a home soldering project. Chris Mason [EMAIL PROTECTED] Box 340, The Valley, Anguilla, British West Indies Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483 Fax: (264) 497-8463 - US Fax (815)301-9759 Yahoo IM: [EMAIL PROTECTED] Skype ID: netconcepts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicetronix OpenSwitch12 chan_vpb problem
On Tue, March 29, 2005 3:48 am, Jan Henkins said: it would be nice to know wether Voicetronix products are on the trusted list or not. I've got an OpenSwitch6 working in a development (soon to be production, fingers crossed, box); never had such a lockup. Have you sent any queries to [EMAIL PROTECTED] Ben's been very helpful when I've had troubles. He's pointed out my self-inflicted troubles very gently and was quick to address those in his court. Paul -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No D-channels available!
Title: No D-channels available! Hi all, I´ve ran into a problem regarding D-channels. I have setup Asterisk (CVS-HEAD-03/21/05-16:41:57) on RH Fedora Core 3(2.6.10-1.770_FC3) I also have a digium wcte11xp card for connetivity to the PSTN(E1). When I start zttool i see that Current Alarms changes between Recovering and Blue Alarm/Recovering. I started to see these problems after i moved the digiumcard from one PCI slot to another, this was to solve a IRQ missmatch. Please let me know if you need anymore info Best regards Rikard Westlund My zaptel.conf looks like this: span=1,1,0,ccs,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 loadzone = se defaultzone=se This is the output from lsmod: Module Size Used by wcte11xp 40096 0 zaptel 204676 1 wcte11xp md5 4033 1 ipv6 231681 12 autofs4 23493 0 sunrpc 156325 1 dm_mod 55637 0 i2c_i801 8141 0 i2c_core 20801 1 i2c_i801 hisax 517149 0 crc_ccitt 2113 2 zaptel,hisax isdn 131905 1 hisax slhc 6849 1 isdn tg3 84933 0 floppy 57841 0 ext3 116297 3 jbd 69977 1 ext3 This is from /var/log/messages Mar 29 14:19:18 sepbx kernel: PCI: Assigned IRQ 9 for device :03:01.0 Mar 29 14:19:18 sepbx kernel: Controller version: 24 Mar 29 14:19:18 sepbx kernel: FALC version: Mar 29 14:19:18 sepbx kernel: TE110P: Setting up global serial parameters for E1 FALC V1.2 Mar 29 14:19:18 sepbx kernel: TE110P: Successfully initialized serial bus for card Mar 29 14:19:18 sepbx kernel: Found a Wildcard: Digium Wildcard TE110P T1/E1 Mar 29 14:19:18 sepbx kernel: Registered tone zone 16 (Sweden) Mar 29 14:19:18 sepbx kernel: TE110P: Span configured for CCS/HDB3/CRC4 Mar 29 14:19:18 sepbx kernel: Calling startup (flags is 4099) Mar 29 14:19:18 sepbx kernel: TE110P: Span configured for CCS/HDB3/CRC4 Mar 29 14:19:18 sepbx kernel: Calling startup (flags is 4099) Mar 29 14:19:18 sepbx kernel: Registered tone zone 16 (Sweden) Mar 29 14:19:18 sepbx zaptel: Running ztcfg: succeeded Mar 29 14:19:19 sepbx kernel: NMF workaround on! Mar 29 14:19:19 sepbx kernel: wcte1xxp: Setting yellow alarm This is from more /proc/zaptel/1 Span 1: WCT1/0 Digium Wildcard TE110P T1/E1 Card 0 HDB3/CCS/CRC4 RECOVERING 1 WCT1/0/1 Clear 2 WCT1/0/2 Clear 3 WCT1/0/3 Clear 4 WCT1/0/4 Clear 5 WCT1/0/5 Clear 6 WCT1/0/6 Clear 7 WCT1/0/7 Clear 8 WCT1/0/8 Clear 9 WCT1/0/9 Clear 10 WCT1/0/10 Clear 11 WCT1/0/11 Clear 12 WCT1/0/12 Clear 13 WCT1/0/13 Clear 14 WCT1/0/14 Clear 15 WCT1/0/15 Clear 16 WCT1/0/16 HDLCFCS 17 WCT1/0/17 Clear 18 WCT1/0/18 Clear 19 WCT1/0/19 Clear 20 WCT1/0/20 Clear 21 WCT1/0/21 Clear 22 WCT1/0/22 Clear 23 WCT1/0/23 Clear 24 WCT1/0/24 Clear 25 WCT1/0/25 Clear 26 WCT1/0/26 Clear 27 WCT1/0/27 Clear 28 WCT1/0/28 Clear 29 WCT1/0/29 Clear 30 WCT1/0/30 Clear 31 WCT1/0/31 Clear When I start Asterisk(asterisk -vc) I get this: Mar 29 15:02:15 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:17 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:18 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:19 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:21 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:22 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:23 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:25 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:26 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:27 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:29 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:31 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down Mar 29 15:02:32 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel
[Asterisk-Users] rfc2833 cisco 7960 DTMF issue
I'm having an issue sending DTMF to cisco dialing this extension I should hear the dtmf tone RTP playload 101 has been sent to the cisco phone, but no audio. in the dialplan exten = 8603,1,Answer(1) exten = 8603,n,sipdtmfmode(rfc2833) exten = 8603,n,SendDTMF(1|100) exten = 8603,n,hangup() sip.conf dtmfmode=rfc2833 SIPDefault.conf I did play with all possible settings for dtmf_outofband: avt, avt_always, none and 0,1 for dtmf_inband nothing happens cisco 7905g is working OK with this example cisco 7960 firmware issue? Any help? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HFC-S
Hi, Look at this page page http://www.junghanns.net/asterisk/downloads/ and get the latest version of bristuff which should be : http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC7k.tar.gz This package is specific for BRI adapter using the cologne chipset. It works great ! Best regards David Masure -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Envoyé : mardi 29 mars 2005 14:04 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] HFC-S Hi! I have just installed Redhat 9 and Asterisk to my computer, and now i have problems with my non-zaptel Card, I don't know how to set it up since all instructions are for digium's hardware. I have searched from the Internet for hours now, can you help me to understand all this HFC-s thing and how it is related to CAPI, ISDN4Linux, bristuff and so on. I have to say that I am not so familiar with Linux. Thank you in advance This mail sent through L-secure: http://www.l-secure.net/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicetronix OpenSwitch12 chan_vpb problem
Hello Paul, On Tue, 2005-03-29 at 08:04 -0500, Paul Dugas wrote: I've got an OpenSwitch6 working in a development (soon to be production, fingers crossed, box); never had such a lockup. Have you sent any queries to [EMAIL PROTECTED] Ben's been very helpful when I've had troubles. He's pointed out my self-inflicted troubles very gently and was quick to address those in his court. Thanks a mil, I've just fired off a mail to them. I'll report back what happened. -- Regards, Jan Henkins ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over
There's many solutions.. One being www.voiceguard.com I think might be what you want. - Original Message - From: Chris Mason [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 8:01 AM Subject: [Asterisk-Users] Fail over For all my PBX installations I want to have Fail Over on the main incoming PSTN line so that a power outage does not leave the offices stranded. Is there any commercial solution to this? I would rather a finished product than a home soldering project. Chris Mason [EMAIL PROTECTED] Box 340, The Valley, Anguilla, British West Indies Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483 Fax: (264) 497-8463 - US Fax (815)301-9759 Yahoo IM: [EMAIL PROTECTED] Skype ID: netconcepts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom SP300 questions
Just Look at the missed calls screen and exit, the counter will clear. Not sure on an IP300, but on an IP600 fastest is to press down arrow then left arrow. On Mar 28, 2005, at 6:32 PM, Paul Hales wrote: 1. You can set up items in the Digitmap (under SIP conf) to know when a number is complete. 2. The up and down arrows let you look through the missed calls. Later, PaulH -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, 8 March 2005 10:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom SP300 questions Hi, all I have two questions regarding usage of Polycom SP300 with Asterisk. No sure if it is Astersisk or phone related, though. 1. When dialing an extension, one has to perss Dial or Send on the phone after number is entered. Is it possible to avoid this and only enter the number? 2. This is probably phone only related, but hopefully someone know the answer. If there wwas a missed call, phone shows 1 call missed. I am trying to figure out how to clear this message from the phone. There are no buttons as far as I can see to get rid of this message on the phone. Thanks, Rudolf ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. CAUTION: This email message and accompanying data may contain information that is confidential. If you are not the intended recipient, you are notified that any use, dissemination, distribution or copying of this message or data is prohibited. If you have received this email message in error, please notify us immediately and erase all copies of this message and attachments. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Debugging Asterisk in GDB (DDD)
On Mon, 2005-03-28 at 15:04 -0800, Jay Ray wrote: Hi, I am running Asterisk on Fedora Core 3. I am trying to use DDD to debug Asterisk. However, when I attach the debugger to the Asterisk Process, the Asterisk CLI promt hangs. Does not give any output, and Asterisk stops processing calls... What could be wrong and what is the best way to debug Asterisk...? You might want to ask this question on the asterisk-dev list since those there are probably more familiar with a debugger + Asterisk. I have personally run Asterisk through gdb once or twice and never had a problem, but I started the process from within gdb. I've never tried to attach to an already running Asterisk process. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No D-channels available!
On Tuesday 29 March 2005 14:08, Rikard Westlund wrote: [...] When I start Asterisk(asterisk -vc) I get this: Mar 29 15:02:15 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down [...] I'll hazzard a guess and say you have the card jumpered for T1 instead of E1. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call files run at certain times
Matt.
I gave your ideas a try and made it work with a twist. Use a macro but...
Here is the good part, call the macro from a call file using application,
passed parameters like name of the sound file, telephone to call, etc.
Voila! Works great!
Thx for the hints Matt.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell
Sent: Martes, 29 de Marzo de 2005 12:36 a.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call files run at certain times
Ok,
so you have a context:
[mycallouts]
exten = _X.,1,Answer
exten = _X.,2,Background(mycall${EXTEN}) exten =
_X.,3,Dial(ZAP/g1/${EXTEN})
and when you do the record you can do
[myrecording]
exten = 98,1,Answer
exten = 98,2,Background(please_enter_99_followed_by_number)
exten = 99.,1,Record(mycall${EXTEN:2})
Or something similar. Hope that makes sense. So the record will create a
file called mycall5551234 or whatever the number is, and then from the call
file you'd send it to the context mycallouts with the extension set to
whatever the number was and then it would play the correct file, and then
call the number.
--
Cheers,
Matt Riddell
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RE: [Asterisk-Users] No D-channels available!
Nope! that I have checked. Rikard -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bob Goddard Sent: den 29 mars 2005 15:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No D-channels available! On Tuesday 29 March 2005 14:08, Rikard Westlund wrote: [...] When I start Asterisk(asterisk -vc) I get this: Mar 29 15:02:15 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down [...] I'll hazzard a guess and say you have the card jumpered for T1 instead of E1. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fail over
No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. Chris Mason [EMAIL PROTECTED] Box 340, The Valley, Anguilla, British West Indies Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483 Fax: (264) 497-8463 - US Fax (815)301-9759 Yahoo IM: [EMAIL PROTECTED] Skype ID: netconcepts -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Marlowe Sent: Tuesday, March 29, 2005 9:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Fail over There's many solutions.. One being www.voiceguard.com I think might be what you want. - Original Message - From: Chris Mason [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 8:01 AM Subject: [Asterisk-Users] Fail over For all my PBX installations I want to have Fail Over on the main incoming PSTN line so that a power outage does not leave the offices stranded. Is there any commercial solution to this? I would rather a finished product than a home soldering project. Chris Mason [EMAIL PROTECTED] Box 340, The Valley, Anguilla, British West Indies Tel. (264) 497-5670 - Cell: (264) 235-5670 - Also (305)-735-3483 Fax: (264) 497-8463 - US Fax (815)301-9759 Yahoo IM: [EMAIL PROTECTED] Skype ID: netconcepts ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] constant ringing on Zap channels
As soon I do a reload I see contant ringing like this on the CLI: -- Zap/14-1 is ringing -- Zap/23-1 is ringing -- Zap/22-1 is ringing -- Zap/20-1 is ringing -- Zap/19-1 is ringing -- Zap/14-1 is ringing -- Zap/23-1 is ringing -- Zap/22-1 is ringing -- Zap/20-1 is ringing -- Zap/19-1 is ringing This goes on continuously and no phones are ringing. I am using a digium T1 card and ADIT 600. Does anyone know what this means and if I should be concerned about it? Thanks, Richard __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Erratic CPU load
-Original Message- From: Zoa [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 29, 2005 1:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Erratic CPU load Hi, Im the astertest guy. If you are referring to the graphs on page 41 - 42, please note that those are done on a embedded via 800mhz cpu and not on a system similar to yours. So i'm pretty sure you can do more than 10 speex encodings at the same time. (also some things changes since we did those tests, some optimizations and configuration options were done to the speex codec implementation in asterisk. Now you can choose your own quality vs cpu usage balance on asterisk) I suggest you go to cvs-head and check out the changes to the speex codec. I did test a IAX/Speex connection between two test machines with cvs head installed. They also showed bursty cpu usage, but max cpu usage during the bursts might be lower, I'll check again. And I didn't change any speex config options, so there's hope :) Without changes to your asterisk machine, i estimate you will be able to do around 30 to 40 channels on your machine if it was without zaptel. Zaptel does not take a lot cpu especially compared to speex encodings. (i know from experience) About the periodic load, please see if any calls are being setup or tear down, or specific applications are used in those cases. 30 channels would be great. Enough to service the 30 ISDN (E1) channels. The periodic load was not due to call setup or tear down, and as far as I could see (ps, top) there were no processes getting in Asterisk's way. Zaptel on it's own gave low (and steady) load during a test with calls from PSTN via TE110P to MOH on the gateway itself. But I guess only a test with iax, with codec translation and *without* zap can rule out zap as part of the cause. Saddly enough, i still didnt find the time to do any load measurements on pri cards. Although i have a test setup ready to go. I can continue testing using your hints, thanks. Eric. Zoa. Eric Giesselbach wrote: Hi, During tests with a IAX2/PSTN gateway I've been getting strange results for processor idle time and load. I (re)search(ed) this issue for a while, but I didn't get any good explainations. Can somebody help me? I have several sites that rely on a central server for connection to the PSTN. Calls to the PSTN are routed over the Internet to this PSTN gateway using IAX2 in trunk mode. To minimize bandwith usage, the Speex codec is used. The central PSTN gateway is a P4 3.0GHz, 1GByte mem, has a TE110P card supporting ISDN30 and runs Asterisk version 1.0.3 on Debian Sarge. While sustaining 5 connections dialed in through the TE110P (terminated at remote sites through IAX) and running top on the PSTN gateway, I see 98% CPU idle time most of the time. I also see short (around 10sec) bursts of high CPU usage (40-50%) by one of the asterisk processes supporting the connection. The bursts happen in irregular intervals, ranging from 30 to 60 sec. Meanwhile, the reported average load jumps up and down between 0.1 to 0.7. What's happening here? Is the processor load really this erratic, or am I looking at an artefact in cpu usage measurement? Maybe there is an aliasing effect caused by the periodic cpu load (20ms, default trunk frequency) and the cpu usage measurement (also periodic?), but I don't know how to check this. If this top reading is an artefact, is there a way to check the actual (realtime) load? Regarding the actual processor usage for speex encoding: this report suggests my processor is indeed quite busy encoding a few speex channels: http://astertest.com/astricon_performance.ppt. Given the results in this report, I doubt the PSTN gateway will support more than 10 speex encodings. At the same time, the same processor encodes 756x756 PAL television to mpeg-4 on my mythtv box at home. Twice, leaving room for scheduled jobs. Has anyone some references to documentation to put these figures into perspective? Thanks in advance, Eric. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CIC Code
On Tue, 29 Mar 2005, Jason Miller wrote:
Do you remember what you actually changed to make it work cause that is the
same switch that I am dealing with myself if I am not mistaken.
Approximately line 1673 in chan_zap.c, it looks like this:
if (p-sig == SIG_FEATD) {
l = ast-cid.cid_num;
if (l)
snprintf(p-dop.dialstr,
sizeof(p-dop.dialstr), T*%s*%s*, l, c + p-stripmsd);
else
snprintf(p-dop.dialstr,
sizeof(p-dop.dialstr), T**%s*, c + p-stripmsd);
} else
if (p-sig == SIG_FEATDMF) {
l = ast-cid.cid_num;
if (l)
snprintf(p-dop.dialstr,
sizeof(p-dop.dialstr), M*00%s#*%s#, l, c + p-stripmsd);
else
snprintf(p-dop.dialstr,
sizeof(p-dop.dialstr), M*02#*%s#, c + p-stripmsd);
} else
The switch tech told me that the 00 indicates operator assistance on their
switch, he thought 01 or the 02 in the next line was more correct. Also,
the * and # aren't called * and #, I found out that the switch tech laughs
at you. :-) They are KP and ST.
One strange thing is that asterisk seemed to be putting pauses in the dial
string. In zaptel.c on line 2502 I inserted a printk to look at the final
dial string. There were pauses inserted in the dial string. It would be
nice if monitor could get audio during the dialing instead of just after
answer.
dave
From: Dave Weis [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Date: Tue, 29 Mar 2005 06:23:35 -0600 (CST)
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] CIC Code
On Mon, 28 Mar 2005, Jason Miller wrote:
Has anyone ever setup Asterisk to pass Feature Group D access while using a
CIC code for outbound calls? If so can you please email the configuration
you have done? I have tried to get this up and running but with no luck. I
have also contacted support and I cant seem to get this going.
I got it working briefly. I had to talk with the switch techs at the other
end for a couple hours and modify the source code to reformat what was
being sent down the line. Their definition of FGD and asterisk's
definition were not the same. It was a nortel DMS-100 or 250 set up for
CLASS 4.
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--
Dave Weis I believe there are more instances of the abridgment
[EMAIL PROTECTED] of the freedom of the people by gradual and silent
encroachments of those in power than by violent
and sudden usurpations.- James Madison
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[Asterisk-Users] Call-ID and Unique-ID
Could anyone explain to me what is the difference between Call-ID and UniqueID of SIP calls, please? Which one could be used as an ID to trace, for example, the status of a call with Manager API and PHP? Thanks, Alex ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] constant ringing on Zap channels
On March 29, 2005 08:40 am, Richard Reina wrote: This goes on continuously and no phones are ringing. I am using a digium T1 card and ADIT 600. Do you have the Adit600 configured correctly? It's not stuck in a test mode or anything? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CIC Code
On Tue, 29 Mar 2005 06:31:11 -0600, Jason Miller [EMAIL PROTECTED] wrote: Do you remember what you actually changed to make it work cause that is the same switch that I am dealing with myself if I am not mistaken. Thank you, Jason Miller From: Dave Weis [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Tue, 29 Mar 2005 06:23:35 -0600 (CST) To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] CIC Code On Mon, 28 Mar 2005, Jason Miller wrote: Has anyone ever setup Asterisk to pass Feature Group D access while using a CIC code for outbound calls? If so can you please email the configuration you have done? I have tried to get this up and running but with no luck. I have also contacted support and I cant seem to get this going. I got it working briefly. I had to talk with the switch techs at the other end for a couple hours and modify the source code to reformat what was being sent down the line. Their definition of FGD and asterisk's definition were not the same. It was a nortel DMS-100 or 250 set up for CLASS 4. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users 1. EAEO siezes outgoing trunk 2. Carrier responds with wink-start 3. After wink, delay of 40 to 200 ms before you outpulse If carrier requested ANI, you outpulse KP+II+ANI(10 digits)+ST, otherwise you send KP+ST 4. After customer completes dialing, EAEO outpulses KP+(0)+7/10 digits+ST 5. Carrier responds with wink 6. Carrier returns true called party answer supervision. If you are connecting via an Access Tandem: 1st state dialing: KP+0ZZ+XXX+ST 0ZZ=spare tandem center code XXX=dialed or presubscribed CIC 2nd state as above. I'm not sure how Asterisk handles this but I will tak e a look. You might request Feature Group C signaling, just to test and see if you can make it work. It is: KP+DNIS+ST Then if you can talk go back to working on the FGD. -- James Taylor MetroTel 3505 Summerihll Road Suite 11 Texarkana, Texas 75503 903-793-1956 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kernel panic loading second fritz card
Hello David, I've spent many hours to make my 2 Fritz PCI v2 work with Asterisk :-) I was not able to make them work with the fcpci drivers (even with custom driver modifications). The solution was to use mISDN (with chan_capi) instead of fcpci. You have a guideline at http://rcum.uni-mb.si/~uvp00845b/ However, there is a caveat : when installing the avm driver, use the special following syntax, otherwise the 2nd card will not work : insmod avmfritz.ko protocol=2,2 type=28,28 Hope it helps, Olivier David Phelan wrote: Hi Everyone, Long time reader, first time poster. FINALLY got my First AVM Fritz Card up and running under Centos 3.4 Installed the secondmodified the drivers etc as per the instructions found at the wiki System boots Modprobe capi all good modprobe fcpci all good modprobe f2pci the kernel then goes into Panic If Modprobe f2pci before fcpci the kernel still goes into panic. Config as follows CentOS 3.4 Kernel 2.4.21-27.0.2.EL fcpci - fcpci-suse8.2-03.11.02 chan_capi-0.3.5 _ David Phelan Blue Ridge Systems Ph:+61 7 3624 8777 _ -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.4 - Release Date: 27/03/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] No D-channels available!
On Tuesday 29 March 2005 14:40, Rikard Westlund wrote: Nope! that I have checked. 1. Double check 2. Change the D channel to be 24 and retry 3. Cycle all channels through all possibilities. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Bob Goddard Sent: den 29 mars 2005 15:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] No D-channels available! On Tuesday 29 March 2005 14:08, Rikard Westlund wrote: [...] When I start Asterisk(asterisk -vc) I get this: Mar 29 15:02:15 WARNING[4557]: chan_zap.c:2049 pri_find_dchan: No D-channels available! Using Primary on channel anyway 16! == Primary D-Channel on span 1 down [...] I'll hazzard a guess and say you have the card jumpered for T1 instead of E1. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7970 Color
Hey Everyone, I bought a Cisco 7970 Color IP phone. I wanted to reset it back to factory defaults. I went through the sequence of holding down the pound key when the unit is powereing on and then when the sequence changes to press 123456789*0#. The phone seemed to do something different after that. Now it is stuck in the constant cycle of going down the line buttons in a row of green lights. Can anyone help me with this? Thanks a million Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
Basically, I'm forwarding the standard Asterisk ports: tcp 5060 udp 5060 udp 4569 udp 5036 tcp 5038 udp 5038 udp 1:2 I'm not sure that i needed both tcp and udp on the mgmt port 5038, but what the heck. :) In sip.conf: externip = xx.xx.xx.xx localnet=192.168.1.0 In the sip client contexts they *all* have: nat=yes canreinvite=no This is so that they can be hopped both in and out of NATs without reconfiging. No special ports being forwarded for the clients. They seem to work behind whatever NATs we throw at them without difficulties... later, Paul - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 5:28 AM Subject: RE: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate Thank you for your story Paul, nice work with the dialplans! I have one question, so you say that for server 2, asterisk is behind nat and you have sip clients inside and outside the nat. Which ports are you forwarding to asterisk from your firewall and in the case of sip clients outside nat, did you have to open certain ports for each client or all clients use the same? For inside clients it should be a charm! Very nice job Paul, intercity dialing and everything well connected... That was a good story.. Thx for sharing. Anton -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Martes, 29 de Marzo de 2005 12:52 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate - Original Message - From: Anton Krall [EMAIL PROTECTED] would like to hear some actual setups and how people are solving the nat issue within scenarios like: Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones I've been playing with this with my friends for awhile now. We've got four different Asterisk servers set up in four different cities: 1. 2 nics - one on internal network, other on external network. TDM400 card with 2 FXO and 1 FXS, 2 different analog lines, and a LiveVoip IAX2 dialout. Various SIP phones connected, both from within the internal network and out on the internet from behind other NATs. 2. 1 nic - behind NAT (ports forwarded). X100p with 1 analog line. Various SIP phones, internal network and from behind other NATs. 3 4. Like #2 but no X100p. All four servers are connected via IAX2 - in all cases we can dial extensions for each other's systems and the call gets dumped to the correct server. Also between server 1 2 we have local inter-city dialing working (if you dial an outside number that is local to the other city and don't put a 1 in front of the number it dumps to the other server and dials out). NAT hasn't proven to be a problem for us - the only thing we can't do as a result of all the SIP clients being natted is Reinvites - this just means that all conversation *must* go through the server as opposed to direct client-client transfer. Servers that are behind nats have the correct IP settings set in SIP.CONF. As long as I set the STUN server on the sip clients to a good working STUN server everything works like a hot damn. Nothing special regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp rxfax under Linux 2.6 w/TDM400?
Thanks Dave,
I will strip out the macro and give it a try. It did appear from the
log that the macro was being called and the last command was rxfax.
The sending fax shows that it is getting connected, and then fails to
send the fax with a message that the receiving fax did not respond and
no file was sent.
I'll repost my results after the stripping of the macro.
Thanks!
Derrick
Dave Cotton wrote:
On Mon, 2005-03-28 at 18:18 -0800, Derrick Knight wrote:
I have got my Asterisk server running with TDM400 card (2xFXO 2xFXS).
I originally had the system configured with a Panasonic fax machine on
one of the extensions. Due to the high volume of fax spam, I figured it
would be a much better idea to capture the faxes as TIF or PDF files to
minimize wasted paper, etc. I have downloaded, compiled and installed
spandsp and can see the rxfax and txfax applications from within
Asterisk. When a fax comes in, it calls the rxfax application and then
does nothing. Here is my log:
Connected to Asterisk CVS-HEAD-03/28/05-16:44:11 currently running on
video (pid = 22837)
video*CLI
-- Starting simple switch on 'Zap/3-1'
-- Executing SetMusicOnHold("Zap/3-1", "default") in new stack
-- Executing Answer("Zap/3-1", "") in new stack
-- Executing Dial("Zap/3-1", "Zap/1|15") in new stack
-- Called 1
-- Zap/1-1 is ringing
-- Zap/1-1 is ringing
-- Redirecting Zap/3-1 to fax extension
-- Hungup 'Zap/1-1'
-- Executing Macro("Zap/3-1", "faxreceive") in new stack
-- Executing SetVar("Zap/3-1", "FAXFILE=/data/fax/1112059186.0.tif")
in new stack
-- Executing RxFAX("Zap/3-1", "/data/fax/1112059186.0.tif") in new stack
-- Hungup 'Zap/3-1'
I am running:
Fedora Core 3 (2.6.10-1.770_14)
Asterisk (CVS-HEAD-03/28/05-16:44:11)
SpanDSP 0.0.2pre10
libtiff 3.5.7 (including dev source)
Any guidance would be appreciated.
This is what I'm using albeit on an X100, that does work, I seem to
remember having trouble with the macro so used this instead.
[analog-in]
;
exten = s,1,System(/bin/echo -n -e "'@CALL${CALLERIDNAME} ~
${CALLERIDNUM}'" | nc -q0 -w1 192.168.1.161 10629)
exten = s,2,Answer
exten = s,3,GotoIf($[${CALLERIDNUM} = ${DAVE_MOBILE}]?disa,s,1)
exten = s,4,SetCallerID(9${CALLERIDNUM})
exten = s,5,Dial(${ALLPHONES},20,tr)
exten = s,6,Voicemail(u${INSTITUTE_VM})
exten = s,7,Hangup
exten = s,106,Voicemail(b${INSTITUTE_VM})
exten = s,107,Hangup
;
exten = fax,1,SetCallerID(${CALLERIDNUM})
exten = fax,2,SetVar(FAXFILE=/var/spool/asterisk-fax/${UNIQUEID}.tif)
exten = fax,3,rxfax(${FAXFILE})
exten = fax,4,System(/usr/bin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERID})
exten = fax,5,Hangup
;
Dave Cotton
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Re: [Asterisk-Users] constant ringing on Zap channels
Do you have the Adit600 configured correctly? It's not stuck in a test mode or anything? I have no idea if it's configured correctly. We just kind of hooked it up when the install was done a couple months ago. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with 401 Unauthorized
Mike Miller wrote: I'm not sure where to start even -- It seems that the problem is with the response to the digest authentication, but I'm not sure how to fix that. The log below is from linphone, but I see the exact same thing with kphone and xten from a indows box as well. You are right (obviously), Asterisk is rejecting the REGISTER requests. However, without seeing the relevant portions of your sip.conf file, we cannot begin to tell you what is wrong, since we are not telepathic (as much as we may wish to be!). Some things that can cause this: - no type=peer entry that matches - the peer entry is not host=dynamic - the peer entry has permit/deny that disallow this device's IP address ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fail over
On Tue, 29 Mar 2005 09:40:08 -0400, Chris Mason [EMAIL PROTECTED] wrote: No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. The Sipura 3000 does this. That is what I use at home. -Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Problem parsing unusual SIP/SDP
Stewart Nelson wrote: I never get such good support for commercial software, even on high-end packages that charge an arm and a leg for maintenance. Many thanks to Mark, Kevin, and the Asterisk team. Thanks for the kind words, we appreciate it! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
Any problems with RTP or voice just on one side? So as long as you use some STUN server, the RTP packets have the right IP. Did you install your own stund or are you using a public one? You didn't have to use SER at all right? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Martes, 29 de Marzo de 2005 08:27 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate Basically, I'm forwarding the standard Asterisk ports: tcp 5060 udp 5060 udp 4569 udp 5036 tcp 5038 udp 5038 udp 1:2 I'm not sure that i needed both tcp and udp on the mgmt port 5038, but what the heck. :) In sip.conf: externip = xx.xx.xx.xx localnet=192.168.1.0 In the sip client contexts they *all* have: nat=yes canreinvite=no This is so that they can be hopped both in and out of NATs without reconfiging. No special ports being forwarded for the clients. They seem to work behind whatever NATs we throw at them without difficulties... later, Paul - Original Message - From: Anton Krall [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, March 29, 2005 5:28 AM Subject: RE: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate Thank you for your story Paul, nice work with the dialplans! I have one question, so you say that for server 2, asterisk is behind nat and you have sip clients inside and outside the nat. Which ports are you forwarding to asterisk from your firewall and in the case of sip clients outside nat, did you have to open certain ports for each client or all clients use the same? For inside clients it should be a charm! Very nice job Paul, intercity dialing and everything well connected... That was a good story.. Thx for sharing. Anton -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Fielding Sent: Martes, 29 de Marzo de 2005 12:52 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate - Original Message - From: Anton Krall [EMAIL PROTECTED] would like to hear some actual setups and how people are solving the nat issue within scenarios like: Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones I've been playing with this with my friends for awhile now. We've got four different Asterisk servers set up in four different cities: 1. 2 nics - one on internal network, other on external network. TDM400 card with 2 FXO and 1 FXS, 2 different analog lines, and a LiveVoip IAX2 dialout. Various SIP phones connected, both from within the internal network and out on the internet from behind other NATs. 2. 1 nic - behind NAT (ports forwarded). X100p with 1 analog line. Various SIP phones, internal network and from behind other NATs. 3 4. Like #2 but no X100p. All four servers are connected via IAX2 - in all cases we can dial extensions for each other's systems and the call gets dumped to the correct server. Also between server 1 2 we have local inter-city dialing working (if you dial an outside number that is local to the other city and don't put a 1 in front of the number it dumps to the other server and dials out). NAT hasn't proven to be a problem for us - the only thing we can't do as a result of all the SIP clients being natted is Reinvites - this just means that all conversation *must* go through the server as opposed to direct client-client transfer. Servers that are behind nats have the correct IP settings set in SIP.CONF. As long as I set the STUN server on the sip clients to a good working STUN server everything works like a hot damn. Nothing special regards, Paul ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Manager API how see if call is on hold
I am using the manager API for "show channels". If I have a multi line phone extenstions 510 - 515 and 510 has a call on hold and 511 has a call on hold and I am answering 512 the manager API show channels doesnt seem to tell me that 510 and 511 are on hold? They are reported as Up. How do I find this information out? Thanks, Jerry ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call-ID and Unique-ID
Alex wrote: Could anyone explain to me what is the difference between Call-ID and UniqueID of SIP calls, please? Which one could be used as an ID to trace, for example, the status of a call with Manager API and PHP? The Call-ID is internal to the SIP protocol, and not exposed inside Asterisk (or via manager/AGI). The UniqueID is assigned by Asterisk to the call itself and should be used for tracking the call via the Asterisk interfaces. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] constant ringing on Zap channels
Since signalling info is carried in the A/B bits which is how * talks to the Adit regarding the state of each channel, any framing misconfig or timing misconfig will cause this. Perform a print config on the adit and closely compare with zapata.conf and zaptel.conf On Mar 29, 2005, at 8:02 AM, Andrew Kohlsmith wrote: On March 29, 2005 08:40 am, Richard Reina wrote: This goes on continuously and no phones are ringing. I am using a digium T1 card and ADIT 600. Do you have the Adit600 configured correctly? It's not stuck in a test mode or anything? -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Partially receiving a fax
Title: Partially receiving a fax Hello everybody, I've succesfully installed spandsp and libtiff4 on my Debian linux platform. I want to receive faxes on my Asterisk server through a tdm10b PCI card. But when I send a fax to Asterisk I get following output from Asterisk and I only receive a part of the fax: Coarse carrier frequency 1707.59 (42) Coarse carrier frequency 1699.90 (66) Training error 1.005662 Training succeeded (constellation mismatch 1.207576) Start rx document Start rx page - compression 2 Coarse carrier frequency 1699.79 (66) Training error 1.040207 Training succeeded (constellation mismatch 1.196346) EOP with final frame tag In state 5 DCN with final frame tag In state 8 That's it! My zapata.conf file looks like this [channels] switchtype=national signalling=fxo_ks rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=1.5 txgain=1.0 group=1 callgroup=1 pickupgroup=1 immediate=no echotraining=yes callerid=asreceived context=fax faxdetect=both relaxdtmf=yes Any suggestions, Thx in advance Dennie __This mail has been scanned for all known viruses by AXSWeb powered by SecuTeam NV. _ This mail has been scanned for all known viruses by AXS Mail powered by SecuTeam NV. Register for AXS Mail at http://www.secuteam.com! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] -lssl problem on debian
Hello Just installed fresh Debian testing box, checked out Asterisk and others from CVS stable (-r 1.0), and now trying to 'make install' in Asterisk. I get this error: if [ -d CVS ] ! [ -f .version ]; then echo CVS-v1-0-03/29/05-15:19:53 .version; fi gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv -lssl /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Error 1 I have installed openssl. I added /usr/lib/ssl into my /etc/ld.so.conf. Looked for other pkgs such as dev pkgs for openssl but couldn't find any. Can anyone help me to find out the right package? Thank you. Best, fred signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: FW: [Asterisk-Users] polycom 500 help!!
WHen you say cannot communicate you mean it keeps giving you a busy signal when you try and dial? and could you post ur sip.conf along with the messages asterisk prints out. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] -lssl problem on debian
I get this error: if [ -d CVS ] ! [ -f .version ]; then echo CVS-v1-0-03/29/05-15:19:53 .version; fi gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv -lssl /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Error 1 I have installed openssl. I added /usr/lib/ssl into my /etc/ld.so.conf. Looked for other pkgs such as dev pkgs for openssl but couldn't find any. After editing /etc/ld.so.conf did you run ldconfig -v to re-read the file? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk compare with Skype
The fact is, there is not ONE sip or iax softphone that is as easy to use as skype for the average user. The sad thing is it doesn't have to be that way. Spend the $100 and get her a IAXy that's pre configured to your local Asterisk server. Then she can use an analog phone to call you for free. -lee ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime mysql problem?
Ok, that was straight from the wiki. Still does not work, I tried it from the iax.conf, etc files and it works just fine. I even tried terminating/placing calls on the same server with realtime and it works fine. Is realtime broken? Is there anything else I can test with? Thanks, Matt -Original Message- From: Matthew Boehm [mailto:[EMAIL PROTECTED] Sent: Monday, March 28, 2005 8:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Realtime mysql problem? Matt Schulte wrote: ++---+-+--+---+--+ ++---+-+--+---+--+-- ---+---+- Here goes it's going to be messy :-) I followed the directions off the wiki. This *should* work just fine right? I built the table according to the directions, every field is varchar though, could that be a problem? The value of nat should be no or yes, not 0 (zero). Try that and reload everything. -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] -lssl problem on debian
You need to have your openssl development package installed. It's trying to link to librairies that are not availables. Fred Blaise wrote: Hello Just installed fresh Debian testing box, checked out Asterisk and others from CVS stable (-r 1.0), and now trying to 'make install' in Asterisk. I get this error: if [ -d CVS ] ! [ -f .version ]; then echo CVS-v1-0-03/29/05-15:19:53 .version; fi gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o term.o ulaw.o alaw.o callerid.o fskmodem.o image.o app.o cdr.o tdd.o acl.o rtp.o manager.o asterisk.o ast_expr.o dsp.o chanvars.o indications.o autoservice.o db.o privacy.o astmm.o enum.o srv.o dns.o aescrypt.o aestab.o aeskey.o utils.o editline/libedit.a db1-ast/libdb1.a stdtime/libtime.a -ldl -lpthread -lncurses -lm -lresolv -lssl /usr/bin/ld: cannot find -lssl collect2: ld returned 1 exit status make: *** [asterisk] Error 1 I have installed openssl. I added /usr/lib/ssl into my /etc/ld.so.conf. Looked for other pkgs such as dev pkgs for openssl but couldn't find any. Can anyone help me to find out the right package? Thank you. Best, fred ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jean-Francois Theroux Systems administrator PrivalODC 450.761.9973 http://www.privalodc.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] adding extension ChanSpy
Hi ALL, I have downloaded app_chanspy.c and chanspy_sounds.tgz. But I haven't found any instructions on how to compile and where to untar these files... I tried to put the .c file on asterisk-src/apps and remake asterisk, but it seems it was not enough... Thank you!Dov ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fail over
No, that's a service, or at least I think it is, the sales garbage obscures what it really is so who knows. What I need is a little box that diverts calls if the PBX goes down. FYI, the topic has been discussed previously on the list, and the problem that you're trying to address is far more difficult that what you might think. The issue is... how do you know when the pbx is down? - machine is up, asterisk is down - machine is up, asterisk is up but not responding - machine is down hard (somewhat easier to address) Some of the previous postings noted using a relay to transfer t1's, pri's, etc, to a second machine; however, tripping the relay still requires some sort of watchdog timer that would sense inactivity. There is no code in asterisk to trigger that process today. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dell 1750 TDM400P - Power
Has anyone come up with a way to get power to a TDM400P card installed in a Dell PowerEdge 1750? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell 1750 TDM400P - Power
Adam Robins wrote: Has anyone come up with a way to get power to a TDM400P card installed in a Dell PowerEdge 1750? I've not tried, but based on what I see in my 1750s, I would say 'good luck'. There are no drive power connectors anywhere, and you can't steal power from a fan connector because they are all monitored (and probably not enough current anyway). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell 1750 TDM400P - Power
Has anyone come up with a way to get power to a TDM400P card installed in a Dell PowerEdge 1750? The TDM card only needs the external power connector if fxs modules are installed. The fxo modules don't use it that power. If fxs modules are present, only the 12 volt lead is used. Therefore creating your own single-wire jumper from another 12 volt source anywhere in the system should be relatively easy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell 1750 TDM400P - Power
I thought the TDM was broke on 1750's...?? I could never get passed that NMI issue. -Original Message- From: Rich Adamson [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 29, 2005 9:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dell 1750 TDM400P - Power Has anyone come up with a way to get power to a TDM400P card installed in a Dell PowerEdge 1750? The TDM card only needs the external power connector if fxs modules are installed. The fxo modules don't use it that power. If fxs modules are present, only the 12 volt lead is used. Therefore creating your own single-wire jumper from another 12 volt source anywhere in the system should be relatively easy. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Verizon ISDN
My backup plan is to use an Adtran Express 3000 to analog and then a Digium card but I'm not sure I can preserve the signaling for the centrex features. I guess that's a cheap way to try this if I can't find a reasonably prices ISDN card. Brian On Mon, 2005-03-28 at 15:52, Kevin P. Fleming wrote: Steven Critchfield wrote: Does the Adtran way differ significantly enough to make this become easy? Yeah, the Adtran actually does ISDN PRI to ISDN BRI conversion (it's a very simple switch), not just encapsulation. It's not cheap, though, so it's not something you want to use unless PRI is not available to you or is horrendously expensive. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Realtime mysql problem?
Matt Schulte wrote: Ok, that was straight from the wiki. Still does not work, I tried it from the iax.conf, etc files and it works just fine. I even tried terminating/placing calls on the same server with realtime and it works fine. Is realtime broken? Is there anything else I can test with? Hrm. Have you tried realtime load iaxpeers 622 ? That command should confirm that the info is indeeded being read via RealTime. Do you have RealTime cache turned on/off? You might try turning it off/on (for the sake of trying something else). -Matthew ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with 401 Unauthorized
On Tue, 29 Mar 2005 07:37:24 -0700, Kevin P. Fleming [EMAIL PROTECTED] wrote: Mike Miller wrote: I'm not sure where to start even -- It seems that the problem is with the response to the digest authentication, but I'm not sure how to fix that. The log below is from linphone, but I see the exact same thing with kphone and xten from a indows box as well. You are right (obviously), Asterisk is rejecting the REGISTER requests. However, without seeing the relevant portions of your sip.conf file, we cannot begin to tell you what is wrong, since we are not telepathic (as much as we may wish to be!). Some things that can cause this: - no type=peer entry that matches - the peer entry is not host=dynamic - the peer entry has permit/deny that disallow this device's IP address Based on what you wrote -- I'm using type=friend, not type=peer. This should be ok, though, correct? (As friend == peer + user, right?) sip.conf: [general] context=default; Default context for incoming calls realm=192.168.1.100; Realm for digest authentication port=5060; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes; Enable DNS SRV lookups on outbound calls [203] type=friend username=203 context=internal secret=203 qualify=no; linphone will become unreachable if qualify=yes host=dynamic nat=no canreinvite=yes disallow=all; only the sensible codecs allow=ulaw allow=alaw allow=gsm ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF detection/generation
I'm hoping Asterisk can help me solve an unusual problem. I need two SIP endpoints (VoiceXML gateways) to transfer DTMF tones to each other. Both of them can detect DTMF according to rfc2833. However, one of them (host2) must generate DTMF inband. Happily, I came up with the following sip.conf to allow host1 to detect DTMF tones generated by host2. [in] type=peer host=host1 dtmfmode=rfc2833 canreinvite=no [out] type=peer host=host2 dtmfmode=inband But this is not enough, because it doesn't allow host2 to detect tones generated by host1. :-( I'm an Asterisk newbie, but thrilled that it got me this far. I'm kinda stuck now, though, and I'm hoping someone on the list can point me in the right direction. Thanks, Jim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with 401 Unauthorized
Mike Miller wrote: Based on what you wrote -- I'm using type=friend, not type=peer. This should be ok, though, correct? (As friend == peer + user, right?) Yes, type=friend is fine. sip.conf: [general] context=default; Default context for incoming calls realm=192.168.1.100; Realm for digest authentication Please remove/comment this line as a test, it should not be necessary. port=5060; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes; Enable DNS SRV lookups on outbound calls [203] type=friend username=203 context=internal secret=203 qualify=no; linphone will become unreachable if qualify=yes host=dynamic nat=no canreinvite=yes disallow=all; only the sensible codecs allow=ulaw allow=alaw allow=gsm This looks fine, although 'username' is not needed. What version of Asterisk are you running? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@Home 0.7 released
Has it been updated for AMP 1-10-007a? Or manual update is required? Thanks Robert Btw: great work!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, March 28, 2005 8:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] [EMAIL PROTECTED] 0.7 released We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2 application is no longer needed. - we now have ZAP extension thanks to AMP 007 - weather.agi reads the current weather report using text to speech __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with 401 Unauthorized
On Tue, 29 Mar 2005 09:01:37 -0700, Kevin P. Fleming [EMAIL PROTECTED] wrote: Mike Miller wrote: Based on what you wrote -- I'm using type=friend, not type=peer. This should be ok, though, correct? (As friend == peer + user, right?) Yes, type=friend is fine. sip.conf: [general] context=default; Default context for incoming calls realm=192.168.1.100; Realm for digest authentication Please remove/comment this line as a test, it should not be necessary. Same results -- I thought that perhaps since my sip id was via ip instead of a domain name, I should try this. port=5060; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes; Enable DNS SRV lookups on outbound calls [203] type=friend username=203 context=internal secret=203 qualify=no; linphone will become unreachable if qualify=yes host=dynamic nat=no canreinvite=yes disallow=all; only the sensible codecs allow=ulaw allow=alaw allow=gsm This looks fine, although 'username' is not needed. What version of Asterisk are you running? 1.0.6 from an ubuntu package. I'd also tried a version compiled from source, but with the same results. I tried taking out username, but it didn't help. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dell 1750 TDM400P - Power
Short of finding somewhere to tap 12v off the board that 1) would'nt make the danged thing beep and 2) voiding the warrantee cdrom??) , I'd just juryrig an external 12v supply along the lines of http://www.soekris.com/PowerAccessories.htm. I'm assumong the tdm400p only taps the 12V for RI and not the 5v ... Adam Robins wrote: Has anyone come up with a way to get power to a TDM400P card installed in a Dell PowerEdge 1750? Thanks, Adam The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Question
Le mardi 29 Mars 2005 18:13, Parker, Blake (MIS) a écrit : What is the command to create a new voicemail box? addmailbox in /asterisk_directory/contrib/scripts Blake ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home 0.7 released
yes --- Matt [EMAIL PROTECTED] wrote: Web Meetme is now installed by default and the meetme2 application is no longer needed. What does this mean exactly? Does this use the regular meetme as opposed to the meetme2 we had to setup before? On Mon, 28 Mar 2005 17:35:37 -0800 (PST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2 application is no longer needed. - we now have ZAP extension thanks to AMP 007 - weather.agi reads the current weather report using text to speech __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with 401 Unauthorized
Mike Miller wrote: 1.0.6 from an ubuntu package. I'd also tried a version compiled from source, but with the same results. I tried taking out username, but it didn't help. OK, then we need a _full_ log, with: - sip debug - set verbose 255 - set debug 255 There should be (at least) a message on the console about why Asterisk is rejecting the REGISTER request. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk, SER, NAT, STUN and the whole debate
Anton Krall wrote: Any problems with RTP or voice just on one side? So as long as you use some STUN server, the RTP packets have the right IP. Did you install your own stund or are you using a public one? You didn't have to use SER at all right? Setting nat=yes does pretty much the same as a STUN server. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@Home 0.7 released
I'm sure some users would use it. Once it's done post it and I'll add it to [EMAIL PROTECTED] Also is it possible to do it without MeetMe2? The new WebMeetMe from Areski uses the normal conferencing app and this is much cleaner and simpler than meetme2. Also is it posible to do it without MeetMe2? --- Dan Austin [EMAIL PROTECTED] wrote: I'm finally back from my trip to Asia and am starting in on the CBMySQL and MeetMe2 apps. I really like where Areski is heading with WebMeetMe and think I will likely merge my scheduling features into WebMeetMe. My 'C' skills are going to prevent any real improvements in MeetMe2, other than compile cleanup, and small bug fixes. The bulk of what we want to accomplish is tied to the web-gui and CBMySQL. So the question I have is would the [EMAIL PROTECTED] user base find any value in a conference scheduler addon? Dan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, March 28, 2005 5:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] [EMAIL PROTECTED] 0.7 released We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2 application is no longer needed. - we now have ZAP extension thanks to AMP 007 - weather.agi reads the current weather report using text to speech __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] small qos switch
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Latham Sent: Sunday, March 27, 2005 12:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] small qos switch I heard a great solution at Linux World Boston. A rather talented young man mentioned using a IPV6 VPN on the IPV4 internet. IPV6 supports QOS by default. Just VPN straight back to the CO and have your POP there so you only need one firewall too. He may have been talented, just not in network engineering. While your IPv6 encapsulated VPN would have QOS, the underlying transport medium (IPv4) still would not (if it didn't have it before). Furthermore, if any Ipv4 hops in between would have prioritized your traffic higher based on its type, they now have no idea what is is, because it's encapsulated. Daryl ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@Home 0.7 released
No. But 0.8 will be out soon with AMP 1-10-007a and some other fixes and features. --- Robert Augustyn [EMAIL PROTECTED] wrote: Has it been updated for AMP 1-10-007a? Or manual update is required? Thanks Robert Btw: great work!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, March 28, 2005 8:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] [EMAIL PROTECTED] 0.7 released We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2 application is no longer needed. - we now have ZAP extension thanks to AMP 007 - weather.agi reads the current weather report using text to speech __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@Home 0.7 released
Does the backup feature preserve enough info so that I won't have to rebuild all of my extensions/etc? JD [EMAIL PROTECTED] wrote: yes --- Matt [EMAIL PROTECTED] wrote: Web Meetme is now installed by default and the meetme2 application is no longer needed. What does this mean exactly? Does this use the regular meetme as opposed to the meetme2 we had to setup before? On Mon, 28 Mar 2005 17:35:37 -0800 (PST), [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2 application is no longer needed. - we now have ZAP extension thanks to AMP 007 - weather.agi reads the current weather report using text to speech __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Turnkey alternatives to fonality or switchvox?
Just a follow-up to my message. I hope I didn't come off as negative about voipconnection. They're a great crew over their, and they defintely know their stuff :) Give them a look because I think you'll be happy Andy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andy Slezak Sent: Monday, March 28, 2005 11:50 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Turnkey alternatives to fonality or switchvox? Hey all, First-time poster and been following the list for some time. Great postings, all... Currently we're implementing the voipconnection vs-1 for a customer. It's a decent device, and the included admin interface leaves a lot to be desired... The idea of the storing settings compact flash is nice, although sometimes they choose not to stick. All in all, we're thinking about returning it in favor of something a bit more tested (and easier on an asterisk newbie). :) Currently, potentials include Fonality's PBXtra Switchvox PBX because they are turnkey in nature AND have ultra-friendly admin interfaces (read: highly customizable * implementation without touching the command-line - nice for a quick powerful deployment). Anyone out there know of alternatives to either of these products? Has anyone had any experiences (good/bad) with either product or company? I really look forward to anyone's feedback. Thanks in advance, Andy Slezak ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe flags in * 1.0.7
While researching Areski's new Web-MeetMe management gui, I found some odd (from what I expected) behaviour). Using the CLI to set un/mute status works but does not update the flags, or so it appears. Starting with a fresh conference (1 user) *CLI meetme list 3456 User #: 1 Channel: OH323/R61 Using the CLI to mute the caller (no change in the user status0 *CLI meetme mute 3456 1 *CLI meetme list 3456 User #: 1 Channel: OH323/R61 Using the *-DTMF menu to mute oneself *CLI meetme list 3456 User #: 1 Channel: OH323/R61 (Listen only) Is this the desired behaviour? Dan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk SMS configuration
Incoming however just isn't working. I've got a nice list of numbers from which SMS messages come: snip You are sending the extra digit to say which mailbox the message is for, right? In this country, if you do not send that digit, it will try to vocalize the message during the calls. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kernel panic loading second fritz card
On Tue, 2005-03-29 at 16:11 +0200, Oga wrote: I've spent many hours to make my 2 Fritz PCI v2 work with Asterisk :-) I was not able to make them work with the fcpci drivers (even with custom driver modifications). The solution was to use mISDN (with chan_capi) instead of fcpci. You have a guideline at http://rcum.uni-mb.si/~uvp00845b/ However, there is a caveat : when installing the avm driver, use the special following syntax, otherwise the 2nd card will not work : insmod avmfritz.ko protocol=2,2 type=28,28 I have systems running 2 fritz cards with no problems at all, which is good because one is a round trip of 1500Kms away. Dave Cotton ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@Home 0.7 released
Any possibility to support a zero extension and operator extension automatically in the Auto-attendant? Thanks, Wiley -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Tuesday, March 29, 2005 9:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] 0.7 released No. But 0.8 will be out soon with AMP 1-10-007a and some other fixes and features. --- Robert Augustyn [EMAIL PROTECTED] wrote: Has it been updated for AMP 1-10-007a? Or manual update is required? Thanks Robert Btw: great work!! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Monday, March 28, 2005 8:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] [EMAIL PROTECTED] 0.7 released We had added a lot to this release to our one button install of Asterisk. Now you can have even more features automatically installed and configured. Asterisk 1.0.7 AMP 1-10-007 Flash Operator Panel 0.20 Redesigned WebMeetme weather agi scripts Midnight Commander We have added some of our most requested features. - Web Meetme is now installed by default and the meetme2 application is no longer needed. - we now have ZAP extension thanks to AMP 007 - weather.agi reads the current weather report using text to speech __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Small Business - Try our new resources site! http://smallbusiness.yahoo.com/resources/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
