[Asterisk-Users] TE210P Linux SMP
Hi, Does anyone have this card(specifically the wct4xxp driver) working under linux and running a SMP kernel? I'm running it in a dual p4 xeon box and when I compile the kernel for SMP and then recompile libpri/zaptel the module doesn't behave correctly(doesn't pick up the pri's). In addition the lights on the back do the following (when no cable is plugged in):- No module - alternate red really fast Module under UP - alternate slow red Module under SMP - Blank I have tried the following kernels:- 2.4.29, 2.4.32, 2.6.14. I would really like to see it working correctly under 2.6 in SMP (with pre-empt etc). Otherwise half of this machine is kinda useless. Kind Regards, Kris Amy Network Engineer Instant Communications Australia's Favourite ISP Tel: 07 3018 8402 Fax: 07 3278 5666 Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blind transfer question
I did a quick check on the blindxfer config parameter and i cant find any referense to that in the sourcecode for 1.2! In previous version all the call transfer things where handled but the flash button '#' but could also be done by a short hangup (200ms) on the line so im not shure what you should be able to change this to. --On Tuesday, November 29, 2005 13:31:37 -0800 Sean Kennedy [EMAIL PROTECTED] wrote: Hi all, I'm trying to change the keys associated with the blind transfer function. I've been mucking around in features.conf, but nothing I do seems to make any difference ( and I've tried to intentionally break it ). I have restarted the * server between each modification. Is this a known thing? Can anybody give me an idea of how to change the Blind Transfer key sequence to something else? Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +--- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B pgp9CrkfGyC4D.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voice Mail
A SIP phone with the possibility of showing message waiting can get that information from Asterisk. My EyeBeam is showing a small image of a letter in the display to show that there are messages waiting. SO you can use this without mail being sent-out. Best regards jan --On Wednesday, November 30, 2005 15:23:33 +0800 Hiu Yen Onn [EMAIL PROTECTED] wrote: How normally SIP user is informed by having a new incoming voicemail and then, how are they read their mails then i have known that, asterisk will send a mail for the users. then, how to configure the mail smtp and pop3 for asterisk to send mail then. thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- +--- ! Irial / YASK AB ! Att: Jan Saell ! Box 59, S-692 21 KUMLA, SWEDEN ! Tel: 019-58 25 15 Int +46-19 58 25 15 Fax +46-19 58 38 05 ! E-mail: [EMAIL PROTECTED] ! PGP Fingerprint: E957 23C8 9F51 0958 B9AD 7F18 404A 5DA1 F944 A08B pgpskT281ywEx.pgp Description: PGP signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Blind transfer question
Sean Kennedy wrote: Is this a known thing? Can anybody give me an idea of how to change the Blind Transfer key sequence to something else? I assume you're using v1.2. If you change anything in features.conf and then restart asterisk, you can connect to the CLI and do show features to see your current feature list. (that way you are sure * has used your config) cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cdr_manager.conf
Would anybody please tell me, If I keep enabled=yes, cdr_manager would be enable, I know but an 'enabled' cdr_manager would help me? How I can be benifited from this in terms of cdr management? What exactly it does if I keep enabled=yes? As I said: If you set enabled to yes you receive CDR Events via the Manager API (http://www.voip-info.org/wiki-Asterisk+manager+API). If you do not enable it, you dont receive them via the Manager API but only written to the CDR file, database or whatever you configured. or, what are the next step(s)? Next steps? Have a look at the wiki and read about the Manager API, if you come to the conclusion that you don't need it forget about cdr_manager. =Stefan signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE210P Linux SMP
Hi Kris, I have TE406P (same as your but quad span) working on 2.6.13 with pre-empt. I had it working fine with 2.6.14 but I could not switch card's IRQ from CPU0 to CPU1 on the 2.6.14 On 2.6.13 CPU1 is handling IRQ's only for TE406P (with occasional timer IRQ's sneaking in). I suggest that you get the latest source for zaptel from SVN repository. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kris Amy Sent: Wednesday, 30 November 2005 19:31 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TE210P Linux SMP Hi, Does anyone have this card(specifically the wct4xxp driver) working under linux and running a SMP kernel? I'm running it in a dual p4 xeon box and when I compile the kernel for SMP and then recompile libpri/zaptel the module doesn't behave correctly(doesn't pick up the pri's). In addition the lights on the back do the following (when no cable is plugged in):- No module - alternate red really fast Module under UP - alternate slow red Module under SMP - Blank I have tried the following kernels:- 2.4.29, 2.4.32, 2.6.14. I would really like to see it working correctly under 2.6 in SMP (with pre-empt etc). Otherwise half of this machine is kinda useless. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE210P Linux SMP
Hi Boris, I think it might have something to do with the HT(hyperthreading) support. Since I have one working fine under a dual-amd setup. Kind Regards, Kris Amy Network Engineer Instant Communications Australia's Favourite ISP Tel: 07 3018 8402 Fax: 07 3278 5666 Email: [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev Sent: Wednesday, 30 November 2005 6:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] TE210P Linux SMP Hi Kris, I have TE406P (same as your but quad span) working on 2.6.13 with pre-empt. I had it working fine with 2.6.14 but I could not switch card's IRQ from CPU0 to CPU1 on the 2.6.14 On 2.6.13 CPU1 is handling IRQ's only for TE406P (with occasional timer IRQ's sneaking in). I suggest that you get the latest source for zaptel from SVN repository. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kris Amy Sent: Wednesday, 30 November 2005 19:31 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] TE210P Linux SMP Hi, Does anyone have this card(specifically the wct4xxp driver) working under linux and running a SMP kernel? I'm running it in a dual p4 xeon box and when I compile the kernel for SMP and then recompile libpri/zaptel the module doesn't behave correctly(doesn't pick up the pri's). In addition the lights on the back do the following (when no cable is plugged in):- No module - alternate red really fast Module under UP - alternate slow red Module under SMP - Blank I have tried the following kernels:- 2.4.29, 2.4.32, 2.6.14. I would really like to see it working correctly under 2.6 in SMP (with pre-empt etc). Otherwise half of this machine is kinda useless. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IP GSM Gateway is giving uncomplete SIP signalization to PRI interface - can I somehow avoid that in Asterisk ?
Hi, I have following setup : PBX - Voxip from Parlay -PRI- Asterisk -SIP- SIP IP GSM Gateway (2n) on outgoing call from pbx through Voxip and to IP GSM gateway : latter only responds with SIP session progress but no SIP Ringing message when connection starts to ring, so Voxip is hanging up line on approx 13sec timeout I know we could try simulate ringing with r in dial, but that would be quite wrong, cause GSM gateways sometime take more time to establish connection, so user gets false ringing signal... Can we somehow interfere with Asterisk and generate SIP messages to fool Voxip from hanging up the line ? Thanks in advance, regards, Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] BRIStuff and PRI
Hello, on http://www.voip-info.org/wiki-Asterisk+zaphfc it is mentioned, that using BRIStuff breaks PRI support. We are using Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a with a Digium PRI card an a beroNet quadBRI in one server and it's running perfecty for months. It depends only on the order the modules are loaded (/sbin/modprobe zaptel, insmod qozap.ko, insmod wcte11xp.ko - in this order). Has this behaviour changed somehow? Would be a pity and would prevent us from upgrading our server. Regards, Henry ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help transfer call
I want to transfer a telephone call in a house number in determinate established hours, this syntax is correct or it must use a command different from DIAL? exten = _x.,1,GotoIfTime(08:30-12:30|mon-fri|*|*?4) exten = _x.,2,GotoIfTime(14:30-18:30|mon-fri|*|*?4) exten = _x.,3,Goto(cellulare,s,1) exten = _x.,4,Dial(${TELEIN},60) exten = _x.,5,Hangup [cellulare] exten = s,1,Dial(ZAP/g2/${TELETRASFERIMENTO},60) exten = s,2,Hangup Thanks Ciao, Fabio
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Re: [Asterisk-Users] Zaptel errors on Debian
On Wed, Nov 30, 2005 at 05:53:54AM -0800, Geo wrote:
OK, fine, thanks. Now I have 1.0.9 still BRIstuffed-0.2.0-RC8h pre-built
don't know how with hisax harmless (eventhough blacklisted), but all runs.
Halas, I have my initial problem. When I dial zap, I just get a continous
dial tone
example: exten = _9.,1,Dial(ZAP/1,${EXTEN:1})
and
CLI Executing Dial(SIP/xxx-f70a, ZAP/1|01) in new stack
what do you have on your dialplan?
You should have something like:
Dial(Zap/1/NUMBER)
Or:
Dial(Zap/1/NUMBER,options,timeout)
But you seem to have:
Dial(Zap/1,NUMBER)
...
If I debug I see that the interface is just Setting hook state to 2, to 1,
to 0 but does not dial.
If I compose a number during this continous dial tone is ringing. So, I have
to recompose the number.
A.e: you gave it an empty number, right?
Remark: in RH with similar zaptel, zapata config was OK, just dialing.
Is there anything new like setting the hook state first, setting a timer, and
than dialing, ... or is it a problem ?
If this is not the problem, please provide the relevant parts of the
dislplan.
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[Asterisk-Users] Compiling Asterisk 1.2 from Source on Debian Sarge - Problems
Hi I am trying to compile Asterisk 1.2 from source on Debian Sarge but am getting errors. I have looked at the errors, Googled extensively and now at a last resort am posting on this list. Believe me I have tried, but have come up with nothing. I've also installed the following packages from Debian Sarge UNSTABLE: gcc kernel-headers-2.4.27 bison openssl libssl0.9.7: libssl-dev libeditline0 libeditline-dev libedit-dev libedit2 libncurses5 libncurses5-dev zlib1g-dev (Note: needed for cvs head) as well as numerous other packages that I have now lost track of. The error remains the same. It would be great if someone could help me out. I'm aware that I can apt-get Asterisk, but I want to do some tweaking in the code before installing. Here is the first bit of the install message: build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp if cmp -s .cleancount .lastclean ; then echo ; else \ make clean; cp -f .cleancount .lastclean;\ fi build_tools/make_defaults_h defaults.h.tmp if cmp -s defaults.h.tmp defaults.h ; then echo ; else \ mv defaults.h.tmp defaults.h ; \ fi rm -f defaults.h.tmp for x in res channels pbx apps codecs formats agi cdr funcs utils stdtime; do make -C $x depend || exit 1 ; done make[1]: Entering directory `/opt/asterisk-1.2.0/res' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/res' make[1]: Entering directory `/opt/asterisk-1.2.0/channels' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/channels' make[1]: Entering directory `/opt/asterisk-1.2.0/pbx' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/pbx' make[1]: Entering directory `/opt/asterisk-1.2.0/apps' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/apps' make[1]: Entering directory `/opt/asterisk-1.2.0/codecs' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/codecs' make[1]: Entering directory `/opt/asterisk-1.2.0/formats' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/formats' make[1]: Entering directory `/opt/asterisk-1.2.0/agi' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/agi' make[1]: Entering directory `/opt/asterisk-1.2.0/cdr' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/cdr' make[1]: Entering directory `/opt/asterisk-1.2.0/funcs' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/funcs' make[1]: Entering directory `/opt/asterisk-1.2.0/utils' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -DNO_AST_MM `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/utils' make[1]: Entering directory `/opt/asterisk-1.2.0/stdtime' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer *.c make[1]: Leaving directory `/opt/asterisk-1.2.0/stdtime' cd editline unset CFLAGS LIBS test -f config.h || ./configure creating cache ./config.cache checking for gcc... gcc checking whether the C compiler (gcc ) works... yes checking whether the C compiler (gcc ) is a cross-compiler... no checking whether we are using GNU C... yes checking whether gcc accepts -g... yes checking how to run the C
Re: [Asterisk-Users] Help transfer call
_x.
Se una chiamate è in incoming esegue s nel context
quindi sarebbe s,1,GotoIfTime.
poi perchè usi un goto ?
intoltre puoi evitare di fare Goto(cellulare,s,1) puoi fare semplicamente
Dial(ZAP/g2/${TELETRASFERIMENTO},60)
oppure se proprio vuoi il goto puoi ottimizzare goto(cellulare)
Ciao
2005/11/30, asterisk183 [EMAIL PROTECTED]:
I want to transfer a telephone call in a house number in determinate
established hours, this syntax is correct or it must use a command different
from DIAL?
exten = _x.,1,GotoIfTime(08:30-12:30|mon-fri|*|*?4)
exten = _x.,2,GotoIfTime(14:30-18:30|mon-fri|*|*?4)
exten = _x.,3,Goto(cellulare,s,1)
exten = _x.,4,Dial(${TELEIN},60)
exten = _x.,5,Hangup
[cellulare]
exten = s,1,Dial(ZAP/g2/${TELETRASFERIMENTO},60)
exten = s,2,Hangup
Thanks
Ciao, Fabio
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Re: [Asterisk-Users] Newbie question
I dont need to configure zaptel device, you dont use it :) 2005/11/30, [EMAIL PROTECTED] [EMAIL PROTECTED]: Hello friends, I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My question is I am using a Welltech FXO box and ip phones by Welltech. Do I still need to configure zapata.conf and zaptel.conf which I read in the documentation from asterisk pdf file downoladed from asterisk.org ? I think I dont because I dont use a digium card but do I have to still confugure for FXO and FXS ports? Kindly help me solving my doubt. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Truth springs from argument amongst friends. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zapata directory not found in svn .
On Tue, Nov 29, 2005 at 09:37:24PM -0600, Kevin P. Fleming wrote: Mr. James W. Laferriere wrote: Hello All , no zapata diredtory , tho zaptel README says many of the testing programs require its libraries . Please enlighten me . Tia , JimL The zapata directory was not imported into SVN. If anything actually does need it, you can get it from CVS. Is it obsoleted? It looks like a nice toy. See e.g. the recent http://linuxgazette.net/120/smith.html For Debian users: http://packages.debian.org/libzap1 http://packages.debian.org/libzap-dev -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] route call based on codec? (g723 gets message, g729 goes to conf connection)
If u are using 1.2 there is global var SIP_CODEC or IAX_CODE
exten = 88,1,NoOP(${SIP_CODEC})
exten = 88,2,NoOP(${IAX_CODEC})
Try
29 Nov 2005 15:41:38 -0500, jonc [EMAIL PROTECTED]:
I have a rather curious integration problem. I need to direct a call
connection based on the codec used for the connection.
If my softswitch attaches to the Asterisk server using G729 I toss the
connection into a requested conference - that works fine.
On occasion my softswitch will attach to the Asterisk server using G723
(and request joining a conference that is using G729). When that happens
I need to feed the connection a stock announcement (recorded in G723)
and then hang up.
Is there a way to direct a call based on the codec used to attach to the
Asterisk server?
More detail for those scratching their heads...
I'm using Asterisk servers to augment my Vocal Data softswitch. One of
the many things that Asterisk does for me is act as a conference bridge.
This works just dandy except that my softswitch uses the conference
bridge to transcode Voicemail announcements.
My Softswitch automagically transcodes all announcements into G711,
G723, and G729. Whenever someone records a voicemail announcement the VM
server opens a conference using each of the codecs - plays the
announcement in G729 (our default) and then records on the other
connections.
Obviously the G723 connection does not work since Asterisk won't
transcode G723. That's cool. We don't *ever* use G723 - it's just built
into the softswitch.
The problem comes with the fact that the softswitch won't give up on
doing the transcoding to G723. It continues to try and try and try and
try... There is nothing dumber than a machine doing a task it can never
finish. Unless its a machine opening hundereds of connections to my
conferencing bridge trying to do a task it can never complete.
I need to feed it something - anything - in a G723 format. I've got
plenty of G723 audio files. If I can simply play one to the g723
connection then it will be happy and go away. ;-)
Any help is appreciated. Thanks - Jon Carnes
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Re: [Asterisk-Users] Newbie question on 1.2 extension configs
try [EMAIL PROTECTED] http://asteriskathome.sourceforge.net/ 2005/11/29, bram kortleven [EMAIL PROTECTED]: Are there any example configs? Or does anybody have a default config for this setup: 1 analog digium clone card for an analogue line (my home line) Several sip phones (a few of them on the outside of my lan (NAT fw between) and 2 insde my lan) Or a simple way of configging through a frontend/script/management utility... I installed astlinux But it does not allow to install and use AMP... Anyone having another script? Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/29, Francesco Peeters [EMAIL PROTECTED]: try ztcfg -vvv sleep 3 ztcfg -vvv Also helpful is cat /proc/zaptel/* This is what I see: [EMAIL PROTECTED] ~]# lsmod |grep zaptel zaptel206724 7 ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 crc_ccitt 2113 1 zaptel [EMAIL PROTECTED] etc]# rmmod ztdummy [EMAIL PROTECTED] etc]# modprobe zaphfc modes=1 [EMAIL PROTECTED] etc]# ztconfig -vvv [EMAIL PROTECTED] etc]# ztcfg -vvv Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. Changing signalling on channel 1 from Unused to Clear channel ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? [EMAIL PROTECTED] ~]# cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A ISDN card 1 [NT] AMI/CCS 1 2 3 My zaptel.conf has this: span=1,1,3,ccs,ami bchan=1-2 dchan=3 Then, I try to start asterisk: [EMAIL PROTECTED] ~]# asterisk -c Asterisk 1.2.0, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = [ [EMAIL PROTECTED] ~]# Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe [EMAIL PROTECTED] ~]# tail /var/log/asterisk/ [...] 22] chan_zap.c: Unable to specify channel 1: No such device or address Nov 30 12:00:44 ERROR[3422] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Nov 30 12:00:44 ERROR[3422] chan_zap.c: Unable to register channel '1-2' Nov 30 12:00:44 WARNING[3422] loader.c: chan_zap.so: load_module failed, returning -1 Nov 30 12:00:44 WARNING[3422] loader.c: Loading module chan_zap.so failed! -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling Asterisk 1.2 from Source on Debian Sarge - Problems
On Wed, Nov 30, 2005 at 12:38:28PM +0200, Hagen Rode wrote: Hi I am trying to compile Asterisk 1.2 from source on Debian Sarge but am getting errors. I have looked at the errors, Googled extensively and now at a last resort am posting on this list. Believe me I have tried, but have come up with nothing. I've also installed the following packages from Debian Sarge UNSTABLE: Debian Sarge is Stable. gcc kernel-headers-2.4.27 bison openssl libssl0.9.7: libssl-dev libeditline0 libeditline-dev libedit-dev libedit2 libncurses5 libncurses5-dev zlib1g-dev (Note: needed for cvs head) Which version of gcc do you use? Testing and Unstable currently use gcc 4. Mixing gcc 3.3 and gcc 4 could lead to some breakages. Specifically I would assume that your kernel headers are from Sarge (Stable). BTW: If you want debs of 1.2 for Sarge: deb http://rapid.dotsrc.org/ experimental/ deb http://rapid.dotsrc.org/ unstable/ Note that they are bristuffed. Don't like that? the souces are there (s/deb/deb-src/). edit debian/patches/00list to remove the bristuff patch and rebuild. I believe no other patch depends on it. as well as numerous other packages that I have now lost track of. The error remains the same. It would be great if someone could help me out. I'm aware that I can apt-get Asterisk, but I want to do some tweaking in the code before installing. apt-get source asterisk cd asterisk-version number and then either: dpatch-edit-patch a_new_patch # edit the change exit # or brute-force manually edit files # don't forget to log your changes: # the following two are from the package devscripts: dch -n # -n: increment the version number a bit. debuild -uc -us now you'll have fresh new packages in the directory above. I figure you'll only need to upgrade the binary package asterisk itself if the change is a simple tweak. Here is the first bit of the install message: build_tools/make_version_h include/asterisk/version.h.tmp if cmp -s include/asterisk/version.h.tmp include/asterisk/version.h ; then echo; else \ mv include/asterisk/version.h.tmp include/asterisk/version.h ; \ fi rm -f include/asterisk/version.h.tmp if cmp -s .cleancount .lastclean ; then echo ; else \ make clean; cp -f .cleancount .lastclean;\ fi build_tools/make_defaults_h defaults.h.tmp if cmp -s defaults.h.tmp defaults.h ; then echo ; else \ mv defaults.h.tmp defaults.h ; \ fi rm -f defaults.h.tmp for x in res channels pbx apps codecs formats agi cdr funcs utils stdtime; do make -C $x depend || exit 1 ; done make[1]: Entering directory `/opt/asterisk-1.2.0/res' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/res' make[1]: Entering directory `/opt/asterisk-1.2.0/channels' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/channels' make[1]: Entering directory `/opt/asterisk-1.2.0/pbx' make[1]: Nothing to be done for `depend'. make[1]: Leaving directory `/opt/asterisk-1.2.0/pbx' make[1]: Entering directory `/opt/asterisk-1.2.0/apps' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/apps' make[1]: Entering directory `/opt/asterisk-1.2.0/codecs' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/codecs' make[1]: Entering directory `/opt/asterisk-1.2.0/formats' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/formats' make[1]: Entering directory `/opt/asterisk-1.2.0/agi' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/agi' make[1]: Entering directory `/opt/asterisk-1.2.0/cdr' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC `ls *.c` make[1]: Leaving directory `/opt/asterisk-1.2.0/cdr' make[1]: Entering directory `/opt/asterisk-1.2.0/funcs' ../build_tools/mkdep -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Wed, November 30, 2005 12:03, Alejandro Vargas said: 2005/11/29, Francesco Peeters [EMAIL PROTECTED]: try ztcfg -vvv sleep 3 ztcfg -vvv Also helpful is cat /proc/zaptel/* This is what I see: [EMAIL PROTECTED] ~]# lsmod |grep zaptel zaptel206724 7 ztdummy,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2 crc_ccitt 2113 1 zaptel [EMAIL PROTECTED] etc]# rmmod ztdummy [EMAIL PROTECTED] etc]# modprobe zaphfc modes=1 [EMAIL PROTECTED] etc]# ztconfig -vvv [EMAIL PROTECTED] etc]# ztcfg -vvv You are running the HFC-PCI in NT mode. This means you have an ISDN telephone connected to it, rather than using it to connect to the PSTN? Zaptel Configuration SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. Changing signalling on channel 1 from Unused to Clear channel ZT_CHANCONFIG failed on channel 1: Invalid argument (22) Did you forget that FXS interfaces are configured with FXO signalling and that FXO interfaces use FXS signalling? What is in your /etc/asterisk/zapata.conf? I do not recall seeing that info before in this thread... [EMAIL PROTECTED] ~]# cat /proc/zaptel/* Span 1: ZTHFC1 HFC-S PCI A ISDN card 1 [NT] AMI/CCS 1 2 3 My zaptel.conf has this: span=1,1,3,ccs,ami bchan=1-2 dchan=3 That is fine... However zaptel clearly doesn't have any active channels, so asterisk will break on that. What brand HFC-PCI card do you have, and did you apply the Florz patch? Then, I try to start asterisk: [EMAIL PROTECTED] ~]# asterisk -c Asterisk 1.2.0, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] [ [EMAIL PROTECTED] ~]# Warning, flexibel rate not heavily tested! Ouch ... error while writing audio data: : Broken pipe [EMAIL PROTECTED] ~]# tail /var/log/asterisk/ [...] 22] chan_zap.c: Unable to specify channel 1: No such device or address Nov 30 12:00:44 ERROR[3422] chan_zap.c: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Nov 30 12:00:44 ERROR[3422] chan_zap.c: Unable to register channel '1-2' Nov 30 12:00:44 WARNING[3422] loader.c: chan_zap.so: load_module failed, returning -1 Nov 30 12:00:44 WARNING[3422] loader.c: Loading module chan_zap.so failed! That is to be expected as no zap channels are available... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Tone busy in zaptel
I use the Zaptel card and when I call a client busy, Asterisk don't play standard tone of busy, but Asterisk play forbidden tone. What can I doing for play busy tone to Asterisk? Thanks Fabio Yahoo! Messenger: chiamate gratuite in tutto il mondo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/29, Francesco Peeters [EMAIL PROTECTED]: Then add to a startup file like rc.local: modprobe zaptel modprobe zaphfc ztcfg -vv I just made exactly as you sed: removed all bristuff, uncompressed it again, execuded download.sh, downloaded florz patch (zaphfc_0.3.0-PRE-1_florz-10.diff) and applied it, compiled zaphfc, cpied zaptel.conf to /etc and zapata.conf to /etc/asterisk, then: [EMAIL PROTECTED] zaphfc]# modprobe zaphfc [EMAIL PROTECTED] zaphfc]# modprobe zaptel [EMAIL PROTECTED] zaphfc]# ztcfg -vv Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) (I still obtaining this error), then cat /proc/zaptel/* and the system hanged again. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Tone busy in zaptel
U dont manage Dial status after dial command
E' perchè non gestisci il risultato del dial
Ex.
tipo
exten = _X.,1,Dial(ZAP/g0/${EXTEN})
exten = _X.,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Playback(il-numero-chiamato-non-risponde)
exten = s-NOANSWER,2,Hangup
exten = s-CHANUNAVAIL,1,Playback(nessuna-linee-disponibile)
exten = s-CHANUNAVAIL,2,Hangup
exten = s-BUSY,1,Playback(il-numero-chiamato-e-occupato)
exten = s-BUSY,2,Hangup
Anyway u can use playtone to play corret telco tones
ovviamente puoi utilizzare tipo playtone(busy) al posto di playback ecc...
Cheers
Buon lavoro
2005/11/30, asterisk183 [EMAIL PROTECTED]:
I use the Zaptel card and when I call a client busy, Asterisk don't play
standard tone of busy, but Asterisk play forbidden tone.
What can I doing for play busy tone to Asterisk?
Thanks
Fabio
Yahoo! Messenger: chiamate gratuite in tutto il mondo
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--
Giovanni Miano
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Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
Probabily zaphfc not loaded retype ztcfg -vvv 2005/11/30, Alejandro Vargas [EMAIL PROTECTED]: 2005/11/29, Francesco Peeters [EMAIL PROTECTED]: Then add to a startup file like rc.local: modprobe zaptel modprobe zaphfc ztcfg -vv I just made exactly as you sed: removed all bristuff, uncompressed it again, execuded download.sh, downloaded florz patch (zaphfc_0.3.0-PRE-1_florz-10.diff) and applied it, compiled zaphfc, cpied zaptel.conf to /etc and zapata.conf to /etc/asterisk, then: [EMAIL PROTECTED] zaphfc]# modprobe zaphfc [EMAIL PROTECTED] zaphfc]# modprobe zaptel [EMAIL PROTECTED] zaphfc]# ztcfg -vv Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. ZT_SPANCONFIG failed on span 1: Invalid argument (22) (I still obtaining this error), then cat /proc/zaptel/* and the system hanged again. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP GSM Gateway is giving uncomplete SIP signalization to PRI interface - can I somehow avoid that in Asterisk ?
Robert Rozman wrote: Hi, I have following setup : PBX - Voxip from Parlay -PRI- Asterisk -SIP- SIP IP GSM Gateway (2n) on outgoing call from pbx through Voxip and to IP GSM gateway : latter only responds with SIP session progress but no SIP Ringing message when connection starts to ring, so Voxip is hanging up line on approx 13sec timeout I know we could try simulate ringing with r in dial, but that would be quite wrong, cause GSM gateways sometime take more time to establish connection, so user gets false ringing signal... Can we somehow interfere with Asterisk and generate SIP messages to fool Voxip from hanging up the line ? You could Answer() the call in Asterisk before passing it off to the gateway. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/30, Francesco Peeters [EMAIL PROTECTED]: You are running the HFC-PCI in NT mode. This means you have an ISDN telephone connected to it, rather than using it to connect to the PSTN? Thanks, now I changed this to mode=0 What is in your /etc/asterisk/zapata.conf? I do not recall seeing that info before in this thread... I've tried modifying the original zapata.conf fron [EMAIL PROTECTED] and also tried copying the one in bristuff zaphfc. ; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode signalling = bri_cpe_ptmp ; p2p TE mode ;signalling = bri_cpe ; p2mp NT mode ;signalling = bri_net_ptmp ; p2p NT mode ;signalling = bri_net pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=demo channel = 1-2 That is fine... However zaptel clearly doesn't have any active channels, so asterisk will break on that. What brand HFC-PCI card do you have, and did you apply the Florz patch? Yes, I applied it today and I'm repeating the tests, but with this patch, when I modprobe zaphfc the system hangs. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Recording Calls
Hi, I´ve recently installed my first Asterisk and it´s working. I can only make outbound calls trough internet. I was willing to record the phone calls in files maybe with wav or gsm extension. Can someboy help me a little with this? Thanks Felix ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Wed, November 30, 2005 12:28, Alejandro Vargas said: 2005/11/29, Francesco Peeters [EMAIL PROTECTED]: Then add to a startup file like rc.local: modprobe zaptel modprobe zaphfc ztcfg -vv I just made exactly as you sed: removed all bristuff, uncompressed it again, execuded download.sh, downloaded florz patch (zaphfc_0.3.0-PRE-1_florz-10.diff) and applied it, compiled zaphfc, cpied zaptel.conf to /etc and zapata.conf to /etc/asterisk, then: [EMAIL PROTECTED] zaphfc]# modprobe zaphfc [EMAIL PROTECTED] zaphfc]# modprobe zaptel [EMAIL PROTECTED] zaphfc]# ztcfg -vv Wrong order: First zaptel, then zaphfc. Do an lsmod to verify both are loaded before ztcfg Zaptel Configuration SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Clear channel (Default) (Slaves: 01) Channel 02: Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. Strange enough, this seems correct... ZT_SPANCONFIG failed on span 1: Invalid argument (22) But this doesn't... (I still obtaining this error), then cat /proc/zaptel/* and the system hanged again. That *might* be caused by the zaphfc not being loaded... (PS: insmod may be better for zaphfc) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Problem with ADIT 600 and FXO configuration
Waiting a bit for 1.2, not yet ready to rewrite the dial-plan.
There were enough fixes, etc, in v1.2 that I'd consider it a priority
to get there fairly soon.
What do you mean Yes the calls out are/were to Zap/g1/xxx?
Your outbound extensions.conf entry should look something like:
exten = _9XXX,1,Dial(Zap/g1/${EXTEN:1})
What is xx in your example? Copy/paste the exact entry that
you are trying to use.
[globals]
TRUNK=Zap/g0; Trunk interface
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
[localexchange]
exten =
_9NXX,1,DBput(RepeatDial/${CALLERIDNUM}=${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,3,Congestion
The above is working just fine with the Digium X101P FXO cards and
has for the last year and 1/2.
And, the above looks fine.
Looking at the X101P when it dials out, it starts at 46.6 volts,
then drops to 8.7 when it dials the DTMF digits, repeated for 10
observations. No reversal or dropping to 0 noted. Switching
to the ADIT 600 FXO card there is 50 volts while not connected
dropping to 10 when it dials.
That's very normal.
I have found two things. The first was one of the POTS lines
was wired with tip/ring reversed by the telco. The second
was that only one of the three lines wouldn't work for
outbound calling (only one DTMF digit is sent). Noticed that
one line is wired next to the ADSL line for Internet service.
Temporary disconnect of the DSL the line, it works, my test
set reported finding data signals on the line also. So it
would appear that the ADIT 600 doesn't tolerate the
interference.
If your test set has a noise level measurement, test each of your
incoming pots lines and use 20 dbrnc of noise (or less) for your
objective. Anything greater then that, dispatch the telco folks
and/or reroute/rewire your inside cable.
So for now I've moved two POTS lines to the ADIT 600 and left
the one connected to the X101P.
I'm sure it's going to cause confusion that one line is
directly on the PBX however the others go though the ADIT.
I'd get rid of the x101p as soon as practical. The functionality and
irregularities of it compared to the channel bank is sure to cause ongoing
support issues that will end up driving up the operational costs.
The two are not in the same league.
I'm not entirely certain that everything is OK however
I think we can move forward. I still have to figure out
why Span 1 stopped functioning (shows the Nop state),
btw. the F1 detail in zttool states Not Open instead
of Not Operational, perhaps a typo?
I would not worry too much about the verbage. Keep in mind that a lot
of terms used in asterisk are those of programmers, not telephony
oriented people. From a programmers perspective, not open and not
operational probably have the same meaning.
Thanks very much for your assistance, it was a very frustrating
problem for me.
Glad I could help a little. I guess a couple of things that have been
learned is always test the external lines early in the process to
eliminate issues, and, don't trust zap/g0 dialing to detect (and
bypass) pots lines that have problems.
The pots line that you mention is next to the dsl circuit (or whatever),
you might try installing another dsl filter in front of the asterisk/
channel bank connection. It should reduce some of the noise and will
likely allow you to use the line.
Rich
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[Asterisk-Users] Re: Would DECT cordless phones work with Asterisk and VOIP?
There is a Kirk distributor for NZ. http://www.wavelink.com.au/ Good luck, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Wed, November 30, 2005 12:55, Alejandro Vargas said: 2005/11/30, Francesco Peeters [EMAIL PROTECTED]: You are running the HFC-PCI in NT mode. This means you have an ISDN telephone connected to it, rather than using it to connect to the PSTN? Thanks, now I changed this to mode=0 What is in your /etc/asterisk/zapata.conf? I do not recall seeing that info before in this thread... I've tried modifying the original zapata.conf fron [EMAIL PROTECTED] and also tried copying the one in bristuff zaphfc. ; ; Zapata telephony interface ; ; Configuration file [channels] ; ; Default language ; ;language=en ; ; Default context ; ; switchtype = euroisdn ; p2mp TE mode signalling = bri_cpe_ptmp ; p2p TE mode ;signalling = bri_cpe ; p2mp NT mode ;signalling = bri_net_ptmp ; p2p NT mode ;signalling = bri_net pridialplan = dynamic prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 echocancel=yes echotraining = 100 echocancelwhenbridged=yes immediate=yes group = 1 context=demo channel = 1-2 That is fine... However zaptel clearly doesn't have any active channels, so asterisk will break on that. What brand HFC-PCI card do you have, and did you apply the Florz patch? Yes, I applied it today and I'm repeating the tests, but with this patch, when I modprobe zaphfc the system hangs. That might be caused by the incorrect loading order (as mentioned in my previous reply) If it remains after correct order (zaptel then zaphfc), please try insmod zaphfc debug=3. In that case also show us the complete output from lspci, and check dmesg and /var/log/messages for any zaptel and zaphfc messages. Good luck! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording Calls
On Wed, November 30, 2005 13:10, Felix Amaral said: Hi, I´ve recently installed my first Asterisk and it´s working. I can only make outbound calls trough internet. I was willing to record the phone calls in files maybe with wav or gsm extension. Can someboy help me a little with this? Thanks Felix If you run Asterisk 1.2, use the automon feature... See http://www.voip-info.org/wiki-Asterisk+config+features.conf -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Recording Calls
Felix Amaral schrieb: Hi, I´ve recently installed my first Asterisk and it´s working. I can only make outbound calls trough internet. I was willing to record the phone calls in files maybe with wav or gsm extension. Can someboy help me a little with this? http://www.voip-info.org/wiki-Asterisk+cmd+monitor signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/30, Francesco Peeters [EMAIL PROTECTED]: Wrong order: First zaptel, then zaphfc. Do an lsmod to verify both are loaded before ztcfg Ok, now I will remove all and try. But when I applied Florz patch every time I load zaphfc the system hangs. First, when compiling zaphfc (after applying Florz patch) i see some warnings: *** Warning: zt_register [/usr/src/bristuff-0.3.0-PRE-1/zaphfc/zaphfc.ko] undefined! *** Warning: zt_receive [/usr/src/bristuff-0.3.0-PRE-1/zaphfc/zaphfc.ko] undefined! *** Warning: zt_transmit [/usr/src/bristuff-0.3.0-PRE-1/zaphfc/zaphfc.ko] undefined! *** Warning: zt_ec_chunk [/usr/src/bristuff-0.3.0-PRE-1/zaphfc/zaphfc.ko] undefined! *** Warning: zt_unregister [/usr/src/bristuff-0.3.0-PRE-1/zaphfc/zaphfc.ko] undefined! Ok, then... modprobe zaptel and... [EMAIL PROTECTED] zaphfc]# insmod ./zaphfc.ko debug=3 (and the system hangs) Kernel panic, fatal exception. Is it necesary the Florz patch? -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Wed, November 30, 2005 13:54, Alejandro Vargas said: 2005/11/30, Francesco Peeters [EMAIL PROTECTED]: Wrong order: First zaptel, then zaphfc. Do an lsmod to verify both are loaded before ztcfg Ok, now I will remove all and try. But when I applied Florz patch every time I load zaphfc the system hangs. First, when compiling zaphfc (after applying Florz patch) i see some warnings: *** Warning: zt_register [/usr/src/bristuff-0.3.0-PRE-1/zaphfc/zaphfc.ko] undefined! *** Warning: zt_receive [/usr/src/bristuff-0.3.0-PRE-1/zaphfc/zaphfc.ko] undefined! *** Warning: zt_transmit [/usr/src/bristuff-0.3.0-PRE-1/zaphfc/zaphfc.ko] undefined! *** Warning: zt_ec_chunk [/usr/src/bristuff-0.3.0-PRE-1/zaphfc/zaphfc.ko] undefined! *** Warning: zt_unregister [/usr/src/bristuff-0.3.0-PRE-1/zaphfc/zaphfc.ko] undefined! Ok, then... modprobe zaptel and... [EMAIL PROTECTED] zaphfc]# insmod ./zaphfc.ko debug=3 (and the system hangs) Kernel panic, fatal exception. Is it necesary the Florz patch? Unless you are using a genuine Junghanns card, yes... It removes the hardware check they put in to make it only work with Junghanns hardware... (A somewhat futile and time-wasting exercise when dealing with Open Source IMHO) These are weird warnings... Have you done make clean before make? Have you first compiled the patched zaptel? -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/30, Francesco Peeters [EMAIL PROTECTED]: If it remains after correct order (zaptel then zaphfc), please try insmod zaphfc debug=3. In that case also show us the complete output from lspci, and check dmesg and /var/log/messages for any zaptel and zaphfc messages. [EMAIL PROTECTED] zaphfc]# lspci 00:00.0 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge 00:00.1 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge 00:00.2 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge 00:00.3 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge 00:00.4 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge 00:00.7 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge 00:01.0 PCI bridge: VIA Technologies, Inc. VT8237 PCI bridge [K8T800/K8T890 South] 00:09.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) 00:0a.0 Network controller: Cologne Chip Designs GmbH ISDN network controller [HFC-PCI] (rev 02) 00:0b.0 Communication controller: Conexant HSF 56k Data/Fax Modem (rev 01) 00:0c.0 Communication controller: Conexant: Unknown device 2f30 (rev 01) 00:0d.0 Communication controller: Conexant: Unknown device 2f30 (rev 01) 00:0f.0 IDE interface: VIA Technologies, Inc. VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06) 00:10.0 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (rev 81) 00:10.1 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (rev 81) 00:10.2 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (rev 81) 00:10.3 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1 Controller (rev 81) 00:10.4 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 86) 00:11.0 ISA bridge: VIA Technologies, Inc. VT8237 ISA bridge [KT600/K8T800/K8T890 South] 00:11.5 Multimedia audio controller: VIA Technologies, Inc. VT8233/A/8235/8237 AC97 Audio Controller (rev60) 00:13.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10) 00:18.0 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] HyperTransport Technology Configuration 00:18.1 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Address Map 00:18.2 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] DRAM Controller 00:18.3 Host bridge: Advanced Micro Devices [AMD] K8 [Athlon64/Opteron] Miscellaneous Control 01:00.0 VGA compatible controller: nVidia Corporation NV34 [GeForce FX 5200] (rev a1) When loading the module with de Florz patches, the kernel panic generates this messges: Nov 30 13:51:37 asterisk1 kernel: zaphfc: no version for zt_receive found: kernel tainted. Nov 30 13:51:37 asterisk1 kernel: zaphfc: jitterbuffer size: 1 Nov 30 13:51:37 asterisk1 kernel: ACPI: PCI interrupt :00:0a.0[A] - GSI 11 (level, low) - IRQ 11 Nov 30 13:51:37 asterisk1 kernel: zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xf899a000 fifo 0xf6b 68000(0x36b68000) IRQ 11 HZ 1000 Nov 30 13:51:37 asterisk1 kernel: zaphfc: Card 0 configured for TE mode Nov 30 13:51:37 asterisk1 kernel: zaphfc: Card 0 configured for master mode Nov 30 13:51:37 asterisk1 kernel: zaphfc: no version for zt_receive found: kernel tainted. Nov 30 13:51:37 asterisk1 kernel: zaphfc: jitterbuffer size: 1 Nov 30 13:51:37 asterisk1 kernel: ACPI: PCI interrupt :00:0a.0[A] - GSI 11 (level, low) - IRQ 11 Nov 30 13:51:37 asterisk1 kernel: zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xf899a000 fifo 0xf6b 68000(0x36b68000) IRQ 11 HZ 1000 Nov 30 13:51:37 asterisk1 kernel: zaphfc: Card 0 configured for TE mode Nov 30 13:51:37 asterisk1 kernel: zaphfc: Card 0 configured for master mode Nov 30 13:51:37 asterisk1 kernel: Unable to handle kernel paging request at virtual address 00100100 Nov 30 13:51:37 asterisk1 kernel: printing eip: Nov 30 13:51:37 asterisk1 kernel: c0150557 Nov 30 13:51:37 asterisk1 kernel: *pde = Nov 30 13:51:37 asterisk1 kernel: Oops: [#1] Nov 30 13:51:37 asterisk1 kernel: Modules linked in: zaphfc(U) zaptel(U) crc_ccitt md5 ipv6 autofs4 i2c_de v i2c_core sunrpc dm_mirror dm_mod button battery ac uhci_hcd ehci_hcd snd_via82xx snd_ac97_codec snd_pcm_ oss snd_mixer_oss snd_pcm snd_timer snd_page_alloc snd_mpu401_uart snd_rawmidi snd_seq_device snd soundcor e 8139too mii floppy ext3 jbd Nov 30 13:51:37 asterisk1 kernel: CPU:0 Nov 30 13:51:37 asterisk1 kernel: EIP:0060:[c0150557] Tainted: GF VLI Nov 30 13:51:37 asterisk1 kernel: EFLAGS: 00010016 (2.6.9-22.EL) Nov 30 13:51:37 asterisk1 kernel: EIP is at kfree+0x23/0x49 Nov 30 13:51:37 asterisk1 kernel: eax: 00100100 ebx: f8a1d000 ecx: edx: c100 Nov 30 13:51:37 asterisk1 kernel: esi: f8a1d000 edi: 0002 ebp: 0086 esp: f6b14dac Nov 30 13:51:37 asterisk1 kernel: ds: 007b es: 007b ss: 0068 Nov 30 13:51:37 asterisk1 kernel: Process insmod (pid: 3527, threadinfo=f6b14000 task=f6908780) Nov 30 13:51:37 asterisk1 kernel: Stack: f8a1d000 0002
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/30, Francesco Peeters [EMAIL PROTECTED]: These are weird warnings... Have you done make clean before make? Have you first compiled the patched zaptel? AHH!! I must compile and install the zaptel module included with bristuff replacing the one included whith asteriskathome, is it?? When it worked I will go to voip-info.org and add some detailed instructions there... -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail and sendmail
On Tue, Nov 29, 2005 at 03:25:19PM -0500, Michaël Gaudette wrote: Hi, I`m a beginning Asterisk and Sendmail user. Note that the sendmail need not be sendmail. It can be basically ant mail transfer agent (MTA). Postfix, exim and maybe qmail will do s well. I am trying to setup my voicemail to send emails to a certain email address. It doesn't work, and I think I've figured out what it is. There is probably a spam-feature at my provider (that I am using as smart host in sendmail) to not accept emails coming from [EMAIL PROTECTED] You can configure your MTA to add a domain name if the sender name contains a username alone. It shoukd then help you with other mail messages sent from this system. E.g: the outputs of cron jobs. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Wed, Nov 30, 2005 at 01:14:35PM +0100, Francesco Peeters wrote: On Wed, November 30, 2005 12:28, Alejandro Vargas said: 2005/11/29, Francesco Peeters [EMAIL PROTECTED]: Then add to a startup file like rc.local: modprobe zaptel modprobe zaphfc ztcfg -vv I just made exactly as you sed: removed all bristuff, uncompressed it again, execuded download.sh, downloaded florz patch (zaphfc_0.3.0-PRE-1_florz-10.diff) and applied it, compiled zaphfc, cpied zaptel.conf to /etc and zapata.conf to /etc/asterisk, then: [EMAIL PROTECTED] zaphfc]# modprobe zaphfc [EMAIL PROTECTED] zaphfc]# modprobe zaptel [EMAIL PROTECTED] zaphfc]# ztcfg -vv Wrong order: First zaptel, then zaphfc. Do an lsmod to verify both are loaded before ztcfg Why should it matter? (if you remove the automatic runs of ztcfg from modprobe.conf/modules.conf) ? And does it matter even if you do run ztcfg multiple times? I've heard all sorts of conflicting reports about this. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Wed, November 30, 2005 14:19, Alejandro Vargas said:
2005/11/30, Francesco Peeters [EMAIL PROTECTED]:
If it remains after correct order (zaptel then zaphfc), please try
insmod
zaphfc debug=3.
In that case also show us the complete output from lspci, and check
dmesg
and /var/log/messages for any zaptel and zaphfc messages.
[EMAIL PROTECTED] zaphfc]# lspci
00:00.0 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge
00:00.1 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge
00:00.2 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge
00:00.3 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge
00:00.4 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge
00:00.7 Host bridge: VIA Technologies, Inc. K8T800Pro Host Bridge
00:01.0 PCI bridge: VIA Technologies, Inc. VT8237 PCI bridge
[K8T800/K8T890 South]
00:09.0 Ethernet controller: Realtek Semiconductor Co., Ltd.
RTL-8139/8139C/8139C+ (rev 10)
00:0a.0 Network controller: Cologne Chip Designs GmbH ISDN network
controller [HFC-PCI] (rev 02)
00:0b.0 Communication controller: Conexant HSF 56k Data/Fax Modem (rev 01)
00:0c.0 Communication controller: Conexant: Unknown device 2f30 (rev 01)
00:0d.0 Communication controller: Conexant: Unknown device 2f30 (rev 01)
00:0f.0 IDE interface: VIA Technologies, Inc.
VT82C586A/B/VT82C686/A/B/VT823x/A/C PIPC Bus Master IDE (rev 06)
00:10.0 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1
Controller (rev 81)
00:10.1 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1
Controller (rev 81)
00:10.2 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1
Controller (rev 81)
00:10.3 USB Controller: VIA Technologies, Inc. VT82x UHCI USB 1.1
Controller (rev 81)
00:10.4 USB Controller: VIA Technologies, Inc. USB 2.0 (rev 86)
00:11.0 ISA bridge: VIA Technologies, Inc. VT8237 ISA bridge
[KT600/K8T800/K8T890 South]
00:11.5 Multimedia audio controller: VIA Technologies, Inc.
VT8233/A/8235/8237 AC97 Audio Controller (rev60)
00:13.0 Ethernet controller: Realtek Semiconductor Co., Ltd.
RTL-8139/8139C/8139C+ (rev 10)
00:18.0 Host bridge: Advanced Micro Devices [AMD] K8
[Athlon64/Opteron] HyperTransport Technology Configuration
00:18.1 Host bridge: Advanced Micro Devices [AMD] K8
[Athlon64/Opteron] Address Map
00:18.2 Host bridge: Advanced Micro Devices [AMD] K8
[Athlon64/Opteron] DRAM Controller
00:18.3 Host bridge: Advanced Micro Devices [AMD] K8
[Athlon64/Opteron] Miscellaneous Control
01:00.0 VGA compatible controller: nVidia Corporation NV34 [GeForce FX
5200] (rev a1)
When loading the module with de Florz patches, the kernel panic
generates this messges:
Nov 30 13:51:37 asterisk1 kernel: zaphfc: no version for zt_receive
found: kernel tainted.
Nov 30 13:51:37 asterisk1 kernel: zaphfc: jitterbuffer size: 1
Nov 30 13:51:37 asterisk1 kernel: ACPI: PCI interrupt :00:0a.0[A]
- GSI 11 (level, low) - IRQ 11
Nov 30 13:51:37 asterisk1 kernel: zaphfc: CCD/Billion/Asuscom 2BD0
configured at mem 0xf899a000 fifo 0xf6b
68000(0x36b68000) IRQ 11 HZ 1000
Nov 30 13:51:37 asterisk1 kernel: zaphfc: Card 0 configured for TE mode
Nov 30 13:51:37 asterisk1 kernel: zaphfc: Card 0 configured for master
mode
Nov 30 13:51:37 asterisk1 kernel: zaphfc: no version for zt_receive
found: kernel tainted.
Nov 30 13:51:37 asterisk1 kernel: zaphfc: jitterbuffer size: 1
Nov 30 13:51:37 asterisk1 kernel: ACPI: PCI interrupt :00:0a.0[A]
- GSI 11 (level, low) - IRQ 11
Nov 30 13:51:37 asterisk1 kernel: zaphfc: CCD/Billion/Asuscom 2BD0
configured at mem 0xf899a000 fifo 0xf6b
68000(0x36b68000) IRQ 11 HZ 1000
Nov 30 13:51:37 asterisk1 kernel: zaphfc: Card 0 configured for TE mode
Nov 30 13:51:37 asterisk1 kernel: zaphfc: Card 0 configured for master
mode
Nov 30 13:51:37 asterisk1 kernel: Unable to handle kernel paging
request at virtual address 00100100
Nov 30 13:51:37 asterisk1 kernel: printing eip:
Nov 30 13:51:37 asterisk1 kernel: c0150557
Nov 30 13:51:37 asterisk1 kernel: *pde =
Nov 30 13:51:37 asterisk1 kernel: Oops: [#1]
Nov 30 13:51:37 asterisk1 kernel: Modules linked in: zaphfc(U)
zaptel(U) crc_ccitt md5 ipv6 autofs4 i2c_de
v i2c_core sunrpc dm_mirror dm_mod button battery ac uhci_hcd ehci_hcd
snd_via82xx snd_ac97_codec snd_pcm_
oss snd_mixer_oss snd_pcm snd_timer snd_page_alloc snd_mpu401_uart
snd_rawmidi snd_seq_device snd soundcor
e 8139too mii floppy ext3 jbd
Nov 30 13:51:37 asterisk1 kernel: CPU:0
Nov 30 13:51:37 asterisk1 kernel: EIP:0060:[c0150557]
Tainted: GF VLI
Nov 30 13:51:37 asterisk1 kernel: EFLAGS: 00010016 (2.6.9-22.EL)
Nov 30 13:51:37 asterisk1 kernel: EIP is at kfree+0x23/0x49
Nov 30 13:51:37 asterisk1 kernel: eax: 00100100 ebx: f8a1d000 ecx:
edx: c100
Nov 30 13:51:37 asterisk1 kernel: esi: f8a1d000 edi: 0002 ebp:
0086 esp: f6b14dac
Nov 30 13:51:37 asterisk1 kernel: ds: 007b es: 007b ss: 0068
Nov 30 13:51:37 asterisk1
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Wed, November 30, 2005 14:31, Alejandro Vargas said: 2005/11/30, Francesco Peeters [EMAIL PROTECTED]: These are weird warnings... Have you done make clean before make? Have you first compiled the patched zaptel? AHH!! I must compile and install the zaptel module included with bristuff replacing the one included whith asteriskathome, is it?? Yep. Even worse: you must replace ALL of Asterisk... (except config files) It can be most easily done with compile.sh in the BRIstuff folder, which should - normally - compile and install everything in the correct order... When it worked I will go to voip-info.org and add some detailed instructions there... I am planning of writing up what I had to do to make it work on my machine with 2 HFC-PCI cards, once I have everything running as it should... (One of my problems right now is that features like automon and atxfer/blindxfer only work for the caller, not the callee when connecting calls between the two ISDN cards... That is fine for outgoing calls, but a PITA for incoming calls...) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Wed, November 30, 2005 14:44, Tzafrir Cohen said: On Wed, Nov 30, 2005 at 01:14:35PM +0100, Francesco Peeters wrote: On Wed, November 30, 2005 12:28, Alejandro Vargas said: 2005/11/29, Francesco Peeters [EMAIL PROTECTED]: Then add to a startup file like rc.local: modprobe zaptel modprobe zaphfc ztcfg -vv I just made exactly as you sed: removed all bristuff, uncompressed it again, execuded download.sh, downloaded florz patch (zaphfc_0.3.0-PRE-1_florz-10.diff) and applied it, compiled zaphfc, cpied zaptel.conf to /etc and zapata.conf to /etc/asterisk, then: [EMAIL PROTECTED] zaphfc]# modprobe zaphfc [EMAIL PROTECTED] zaphfc]# modprobe zaptel [EMAIL PROTECTED] zaphfc]# ztcfg -vv Wrong order: First zaptel, then zaphfc. Do an lsmod to verify both are loaded before ztcfg Why should it matter? (if you remove the automatic runs of ztcfg from modprobe.conf/modules.conf) ? zaphfc connects to zaptel, which it cannot if started before zaptel, which'll make it fail... (And thus unloads it!) And does it matter even if you do run ztcfg multiple times? I've heard all sorts of conflicting reports about this. Me too, but I haven't seen any difference whether I run it once, twice or a gazillion times... I don't see a difference between using the -s switch or not either! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/30, Francesco Peeters [EMAIL PROTECTED]: AHH!! I must compile and install the zaptel module included with bristuff replacing the one included whith asteriskathome, is it?? Yep. Even worse: you must replace ALL of Asterisk... (except config files) It can be most easily done with compile.sh in the BRIstuff folder, which should - normally - compile and install everything in the correct order... But... I already tried it at first, and asterisk stopped working... Well... I'll do it and check why it is not working. -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe with the V (video) option
I am trying to allow my
conference participants to see who they are talking to.
My dialplan
calls: Meetme(${ARG1} | vMd)
I get audio and no video.
I thought the v option might
do the trick? Am I way off? Any tips?
Thanks..
trond
___
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Re: [Asterisk-Users] MeetMe with the V (video) option
Trond G. Andersen wrote:
I am trying to allow my conference participants to see who they are
talking to.
My dialplan calls: Meetme(${ARG1} | vMd)
I get audio and no video.
I thought the v option might do the trick? Am I way off? Any tips?
Doesn't work.
Some people have developed patches, but need money before they can share them.
So we're kinda stuck till app_conference does it I guess.
--
Cheers,
Matt Riddell
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Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
OK. Thank you everybody It is working now. The short solution is this: download bristuff, execute download apply patch, execute compile and check configs of asterisk in order to run it. To add the module to the start, it is easy to add this to /etc/modprobe.conf options zaphfc modes=0 install zaphfc /sbin/modprobe --ignore-install zaphfc /sbin/ztcfg And this to /etc/sysconfig/zaptel: MODULES=$MODULES zaphfc Thank you again. The next step is to try 2 cards... -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zapata directory not found in svn .
Tzafrir Cohen wrote: Is it obsoleted? It looks like a nice toy. See e.g. the recent http://linuxgazette.net/120/smith.html No, it's still on our CVS servers and will be there indefinitely. If there is demand (I assumed there wouldn't be) I can easily import it into SVN as well... ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Astfax problem
ok got the patchfile to work but now i have compiling errors: gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g3 -Iinclude -I../include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DZAPTEL_OPTIMIZATIONS -fomit-frame-pointer -fPIC -c -o app_rxfax.o app_rxfax.cIn file included from app_rxfax.c:15:../include/asterisk/file.h:55: Fehler: syntax error before »*« token../include/asterisk/file.h:55: Warnung: Funktionsdeklaration ist kein Prototyp../include/asterisk/file.h:56: Fehler: syntax error before »*« token../include/asterisk/file.h:56: Warnung: Funktionsdeklaration ist kein Prototypapp_rxfax.c: In Funktion »phase_e_handler«:app_rxfax.c:77: Warnung: implizite Deklaration der Funktion »fax_get_transfer_statistics«app_rxfax.c:78: Warnung: implizite Deklaration der Funktion »fax_get_far_ident«app_rxfax.c:79: Warnung: implizite Deklaration der Funktion »fax_get_local_ident«app_rxfax.c: In Funktion »rxfax_exec«:app_rxfax.c:189: Warnung: Zeigerziele bei Übergabe des Arguments 1 von »__builtin_strncpy« unterscheiden sich im Vorzeichenbesitzapp_rxfax.c:259: Warnung: Übergabe des Arguments 1 von »fax_init« von inkompatiblem Zeigertypapp_rxfax.c:260: Fehler: »t30_state_t« hat kein Element namens »verbose«app_rxfax.c:263: Warnung: implizite Deklaration der Funktion »fax_set_local_ident«app_rxfax.c:266: Warnung: implizite Deklaration der Funktion »fax_set_header_info«app_rxfax.c:267: Warnung: implizite Deklaration der Funktion »fax_set_rx_file«app_rxfax.c:269: Warnung: implizite Deklaration der Funktion »fax_set_phase_d_handler«app_rxfax.c:270: Warnung: implizite Deklaration der Funktion »fax_set_phase_e_handler«app_rxfax.c:281: Warnung: implizite Deklaration der Funktion »fax_rx_process«app_rxfax.c:284: Warnung: implizite Deklaration der Funktion »fax_tx_process«app_rxfax.c:321: Warnung: Übergabe des Arguments 1 von »fax_release« von inkompatiblem Zeigertypmake[1]: *** [app_rxfax.o] Fehler 1make[1]: Leaving directory `/usr/src/asterisk/apps'make: *** [subdirs] Fehler 1 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Wed, November 30, 2005 15:15, Alejandro Vargas said: OK. Thank you everybody It is working now. The short solution is this: download bristuff, execute download apply patch, execute compile and check configs of asterisk in order to run it. To add the module to the start, it is easy to add this to /etc/modprobe.conf options zaphfc modes=0 install zaphfc /sbin/modprobe --ignore-install zaphfc /sbin/ztcfg And this to /etc/sysconfig/zaptel: MODULES=$MODULES zaphfc Thank you again. The next step is to try 2 cards... When you do, make VERY sure the PCI slots are NOT sharing an IRQ! That'll break it every time! cat /proc/pci | less and then check the IRQs for both cards... -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 360, Hold Button Asterisk v1.2
We're running Asterisk at Home but upgraded to version 1.2 of Asterisk. After the upgrade the 'Hold' button on our Snom 360 phones now immediately hangs up a call instead of putting the call on hold. Has anyone else had this problem and figured out how to fix it? I ran 'sip debug' in the Asterisk CLI and it looks like when I hit the hold button it's sending a Cancel message if I understand correctly. Here is the output: -- SIP read from 192.168.1.220:2057: CANCEL sip:[EMAIL PROTECTED];user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.220:2057;branch=z9hG4bK-n8r9ndx0xkfb;rport From: Sascha sip:[EMAIL PROTECTED];tag=vny1xnywlu To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 2 CANCEL Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED]:2057;line=vtp1p31i Content-Length: 0 --- (9 headers 0 lines)--- Sending to 192.168.1.220 : 2057 (NAT) Reliably Transmitting (NAT) to 192.168.1.220:2057: SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 192.168.1.220:2057;branch=z9hG4bK-n8r9ndx0xkfb;received=192.168.1.220;rport=2057 From: Sascha sip:[EMAIL PROTECTED];tag=vny1xnywlu To: sip:[EMAIL PROTECTED];user=phone;tag=as7ad32eb1 Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 192.168.1.220:2057: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.220:2057;branch=z9hG4bK-n8r9ndx0xkfb;received=192.168.1.220;rport=2057 From: Sascha sip:[EMAIL PROTECTED];tag=vny1xnywlu To: sip:[EMAIL PROTECTED];user=phone;tag=as7ad32eb1 Call-ID: [EMAIL PROTECTED] CSeq: 2 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Content-Length: 0 Thanks in advance for any advice! Sascha ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_misdn crashes : init_stack: success but entitylist not empty
Hi, I have the same problem, same error but loading modules changes nothing. I'm using debian sarge and Asterisk 1.2: after compiling asterisk I launched install-misdn from beronet site. When I started Asterisk the same error arose: Nov 30 15:43:06 ERROR[4914]: chan_misdn.c:3455 load_module: Unable to initialize mISDN Maybe I'm missing something? How can I know if I have mISDN module working? TIA Giorgio Incantalupo Yoann Le Bihan wrote: 2005/11/25, Jose Limeres [EMAIL PROTECTED]: Yoann, I am going through a similar problem you reported in a past posting: Nov 24 17:49:31 ERROR[9326] chan_misdn.c: Unable to initialize mISDN Nov 24 17:49:31 WARNING[9326] loader.c: chan_misdn.so: load_module failed, returning -1 Nov 24 17:49:31 WARNING[9326] chan_misdn.c: cb_log called with out-of-range port number! (0) Nov 24 17:49:31 WARNING[9326] loader.c: Loading module chan_misdn.so failed! How did you solve it? I looked back to this error. In fact, it happens when you forget to initialize driver, so do it : /etc/init.d/misdn-init scan If everything goes well you can do : /etc/init.d/misdn-init config /etc/init.d/misdn-init start Then, you can start asterisk :-) Cheers, YLB. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer call error
When I call a internal Sip telephone, the calling transfer to Teleohone external, but Asterisk show this error: Executing Dial("SIP/201-1e2a", "ZAP/g1/3472543320|60") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/3472543320 Nov 30 15:52:09 WARNING[1866]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 2: Red Alarm Nov 30 15:52:09 WARNING[1866]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 2 Nov 30 15:52:09 NOTICE[1862]: chan_zap.c:8451 pri_dchannel: PRI got event: Alarm (4) on Primary D-channel of span 1 Nov 30 15:52:09 NOTICE[1862]: chan_zap.c:8458 pri_dchannel: pri_shutdown Nov 30 15:52:09 NOTICE[1866]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 2 Nov 30 15:52:09 NOTICE[1862]: chan_zap.c:8451 pri_dchannel: PRI got event: No more alarm (5) on Primary D-channel of span 1 Nov 30
15:52:09
WARNING[1866]: chan_zap.c:6511 handle_init_event: Detected alarm on channel 1: No Alarm Nov 30 15:52:09 WARNING[1866]: chan_zap.c:1586 zt_disable_ec: Unable to disable echo cancellation on channel 1 Nov 30 15:52:09 NOTICE[1866]: chan_zap.c:6506 handle_init_event: Alarm cleared on channel 1 -- Hungup 'Zap/1-1' == No one is available to answer at this time (1:0/0/0) -- Executing Hangup("SIP/201-1e2a", "") in new stack == Spawn extension (local, 203, 2) exited non-zero on 'SIP/201-1e2a' My extension.conf: exten = 203,1,Dial(ZAP/g1/3472543320,60) exten = 203,2,Hangup Why? Thanks
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Re: [Asterisk-Users] zapata directory not found in svn .
Hello Kevin , On Tue, 29 Nov 2005, Kevin P. Fleming wrote: Mr. James W. Laferriere wrote: Hello All , no zapata diredtory , tho zaptel README says many of the testing programs require its libraries . Please enlighten me . Tia , JimL The zapata directory was not imported into SVN. If anything actually does need it, you can get it from CVS. Any reason why ? Tia , JimL -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zapata directory not found in svn .
Mr. James W. Laferriere wrote: Any reason why ? Tia , JimL There were many 'stale' projects that I didn't bother to import. Given that nothing has been changed in that project for over a year, and that nothing in Zaptel (in normal use) relies on it. it seemed a good candidate to be 'pruned'. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
2005/11/30, Francesco Peeters [EMAIL PROTECTED]: When you do, make VERY sure the PCI slots are NOT sharing an IRQ! That'll break it every time! Did you try to use APIC? This is suposed to solve the problem of IRQs -- Alejandro Vargas ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX Service providers in Australia for unlimited inbound
Can anyone on the list recommend any IAX Service providers in Australia for unlimited inbound in the 02 area code? Ive been using Faktortel for A$9.50 per month and although the outbound is fantastic (I mean the quality is fantastic the fixed price 10c per call Australia wide is pretty good as well) the inbound has been getting worse and worse. I keep getting calls with un-usable echo etc which means I need to hang-up and call them back etc. Is there anyone who can recommend an alternative, they must be able to offer multiple inbound calls (faktortel allows me 4 simultaneous inbound calls at the moment). Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Got SIP response 400 Invalid Subscription-State
I keep getting this error message from one of my Avaya 4620SW hard phone. Got SIP response 400 Invalid Subscription-State back from 192.168.xx.xx which is the IP address assigned to that hard phone. Also the phone will still have dial tone but cannot make or recieve any calls. Thanks ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: IAX Service providers in Australia for unlimited inbound
Dean Collins [EMAIL PROTECTED] uttered the following thing: Can anyone on the list recommend any IAX Service providers in Australia for unlimited inbound in the 02 area code? You can try www.austechpartnerships.com.au though their outbound is a bit more expensive. BB ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with IAX2 jitterbuffer and DTMF reception with 1.2.0
Rich Adamson wrote: I've noticed that if I enable the jitterbuffer in iax.conf with Asterisk 1.2.0 that Asterisk stops responding to incoming DTMF frames for calls between Teliax and my server. I've used iax2 debug and Ethereal to confirm that Teliax is, in fact, sending the frames. I only have two IAX2 connections (one a softphone on my local network and the Teliax registration). The softphone does not experience the problem (DTMF frames sent from the softphone to * are recognized with the jitter buffer enabled). Everything else about the connection with Teliax is fine when the jitter buffer is enabled. iax2 show channels reports a 40 ms delay with the Teliax connection. Any ideas? My iax.conf is: [general] bandwidth=low jitterbuffer=no ; setting this to yes causes DTMF frames from Teliax to be ignored (for my server anyway), apparently forcejitterbuffer=no register = XX:[EMAIL PROTECTED] tos=lowdelay autokill=yes [bboatrig] type=friend host=dynamic context=default auth=md5,rsa,plaintext secret=XX callerid=Laptop 7020 accountcode=bboatrig0 [teliax] context=incoming-voip-trusted type=friend host=voip-co3.teliax.com auth=md5 secret=X disallow=all allow=ulaw allow=alaw allow=gsm Had the same issue with them for some time using cvs-head. I simply left the jitterbuffer turned off and really haven't had an issue. If you guys ran ethereal on the packets coming from teliax, did you find that the DTMF frames had timestamps that were around the same as the surrounding voice frames? If asterisk isn't seeing the DTMF frames at all, it could just be that teliax is sending them with timestamps far into the future or something, so the jitterbuffer is just putting them back in order. For example, if teliax sends frames that look like this Voice: 0ms Voice: 20ms Voice: 40ms [...] Voice: 1000ms Voice: 1020ms DTMF: 22333444ms Voice: 1040ms Then, the jitterbuffer will end up holding onto the DTMF frame, for.. a while.. -SteveK I opened a trouble ticket with them and they quickly blamed asterisk for not fixing a problem. Best guess is they are running an older version of iax2 (probably modified) and haven't taken the time to upgrade. There is also a known issue with using trunk=yes and ilbc. Bug has been opened on that one. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXmodem fax polling
In checking this out, how does one implement it.. the readme is very vague. I really like the IAXmodem with hylafax for incoming, and has been working great. I would like to explore the outbound faxing capabilities, but havent had a chance to go down that road. Right now I can fax out using the Brother MFC software printer driver, because I have a connection to the MFC with a USB cable ... I have downloaded a few of the Hylafax clients to no avail in making them work. anyways ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Debian Sarge + Asterisk 1.2 + chan_mISDN not starting
Hi, I'm setting up an Asterisk 1.2 PBX based on a Debian Sarge distro with a quadBRI beroNet card. I've followed beroNet instructions so I compiled Zaptel, Libpri and Asterisk and then launched install-mISDN script downloaded from beronet site (install-mISDN.tar.gz). I try to start Asterisk but it gives me the following error: *[chan_misdn.so] = (Channel driver for mISDN Support (Bri/Pri)) Nov 30 17:02:17 ERROR[4246]: chan_misdn.c:3455 load_module: Unable to initialize mISDN Nov 30 17:02:17 WARNING[4246]: loader.c:414 __load_resource: chan_misdn.so: load_module failed, returning -1 Nov 30 17:02:17 WARNING[4246]: chan_misdn.c:3623 chan_misdn_log: cb_log called with out-of-range port number! (0) Nov 30 17:02:17 WARNING[4246]: loader.c:554 load_modules: Loading module chan_misdn.so failed!* Supposing mISDN is already present in Asterisk 1.2, why isn't Asterisk starting?? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_sip.c error
Why Asterisk show this message: Nov 30 17:05:17 WARNING[1351]: chan_sip.c:9600 handle_response_register: Got 200 OK on REGISTER that isn't a register Thanks Yahoo! Messenger: chiamate gratuite in tutto il mondo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disable IAX2 native bridging / Monitor() app
I am working with a 3rd party provider who is providing me with IAX2 dialtone. I am using GSM codec end-to-end and my provider insists on ULAW only. When my remote IAX clients attempt to use the provider for PSTN calls by calling my primary * box, and my primary's dialplan is set to dial the provider, native bridging is attempted. Because the codecs are different, the call fails and the caller hears dead audio. What I want is to disable native bridging altogether and to force my primary to stay in the media path and transcode. My primary staying in the media path is also essential for call recording. I have found: If you don't want native bridging, you need to disable it in chan_iax2.c * by undefining BRIDGE_OPTIMIZATION. Can anyone confirm if this is actually the case? Or is there a simpler, undocumented Dial() switch I can use? Running 1.0.9 on remote IAX boxes, 1.2 Beta 1 on primary. Second question: Can someone summarize for me the circumstances under which only a single part of the channel / one side of the conversation is recorded using Monitor() under 1.2? For example, I have heard that natively bridged Zap channels only record one side of the conversation. This may not be the case in 1.2. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] pbx or asterisk?
hi all i have a pbx siemens connect via E1 to my asterisk box. the asterisk box can call without problems to pbx extensions. but when y press the numbers form example 402 in the pbx phones asterisk give me this -- Saved useragent X-Lite release 1103m for peer 402 -- Going to extension s|1 because of Complete received -- Executing Playback(Zap/31-1, vm-goodbye) in new stack -- Accepting call from '' to 's' on channel 0/31, span 1 == Spawn extension (default, s, 1) exited non-zero on 'Zap/31-1' -- Hungup 'Zap/31-1' -- Accepting call from '' to 's' on channel 0/31, span 1did not receive any number or i have miss configure somenthing in asterisk box? -- .- Pablo Allietti LACNIC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: route call based on codec? (g723 gets message, g729 goes to conf connection)
You may have already done this, but my first approach would be to look hard
at the Vocal Data switch and see if you can disable G723 support on the
switch.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- - -- -- - --
Giovanni Miano [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
If u are using 1.2 there is global var SIP_CODEC or IAX_CODE
exten = 88,1,NoOP(${SIP_CODEC})
exten = 88,2,NoOP(${IAX_CODEC})
Try
29 Nov 2005 15:41:38 -0500, jonc [EMAIL PROTECTED]:
I have a rather curious integration problem. I need to direct a call
connection based on the codec used for the connection.
If my softswitch attaches to the Asterisk server using G729 I toss the
connection into a requested conference - that works fine.
On occasion my softswitch will attach to the Asterisk server using G723
(and request joining a conference that is using G729). When that happens
I need to feed the connection a stock announcement (recorded in G723)
and then hang up.
Is there a way to direct a call based on the codec used to attach to the
Asterisk server?
More detail for those scratching their heads...
I'm using Asterisk servers to augment my Vocal Data softswitch. One of
the many things that Asterisk does for me is act as a conference bridge.
This works just dandy except that my softswitch uses the conference
bridge to transcode Voicemail announcements.
My Softswitch automagically transcodes all announcements into G711,
G723, and G729. Whenever someone records a voicemail announcement the VM
server opens a conference using each of the codecs - plays the
announcement in G729 (our default) and then records on the other
connections.
Obviously the G723 connection does not work since Asterisk won't
transcode G723. That's cool. We don't *ever* use G723 - it's just built
into the softswitch.
The problem comes with the fact that the softswitch won't give up on
doing the transcoding to G723. It continues to try and try and try and
try... There is nothing dumber than a machine doing a task it can never
finish. Unless its a machine opening hundereds of connections to my
conferencing bridge trying to do a task it can never complete.
I need to feed it something - anything - in a G723 format. I've got
plenty of G723 audio files. If I can simply play one to the g723
connection then it will be happy and go away. ;-)
Any help is appreciated. Thanks - Jon Carnes
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--
Giovanni Miano
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RE: [Asterisk-Users] Disable IAX2 native bridging / Monitor() app
Answered my own question, partially from the route call based on codec
thread:
If u are using 1.2 there is global var SIP_CODEC or IAX_CODE
exten = 88,1,NoOP(${SIP_CODEC})
exten = 88,2,NoOP(${IAX_CODEC})
So I can modify my dialplan to check the codec. If it's anything but GSM,
route to the IAX provider. If it's GSM, dial out via the PSTN.
-Original Message-
From: Colin Anderson [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 30, 2005 9:25 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Disable IAX2 native bridging / Monitor() app
I am working with a 3rd party provider who is providing me with IAX2
dialtone. I am using GSM codec end-to-end and my provider insists on ULAW
only. When my remote IAX clients attempt to use the provider for PSTN calls
by calling my primary * box, and my primary's dialplan is set to dial the
provider, native bridging is attempted. Because the codecs are different,
the call fails and the caller hears dead audio.
What I want is to disable native bridging altogether and to force my primary
to stay in the media path and transcode. My primary staying in the media
path is also essential for call recording. I have found:
If you don't want native bridging, you need to disable it in chan_iax2.c
* by undefining BRIDGE_OPTIMIZATION.
Can anyone confirm if this is actually the case? Or is there a simpler,
undocumented Dial() switch I can use?
Running 1.0.9 on remote IAX boxes, 1.2 Beta 1 on primary.
Second question: Can someone summarize for me the circumstances under which
only a single part of the channel / one side of the conversation is recorded
using Monitor() under 1.2? For example, I have heard that natively bridged
Zap channels only record one side of the conversation. This may not be the
case in 1.2.
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Re: [Asterisk-Users] Disable IAX2 native bridging / Monitor() app
You could try using one of the dial functions that listen to DTMF i.e. t or T -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] asterisk starting problem. Warning 2224 (app_capiCD.so)
Hi all, I'm new in Asterisk so I'll thank a lot any help. When I start Asterisk: asterisk -cvv, the output is as follows: == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk , Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer [EMAIL PROTECTED] = == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Parsing '/etc/asterisk/dnsmgr.conf': Found Asterisk Dynamic Loader loading preload modules: == Parsing '/etc/asterisk/modules.conf': Found == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/cdr.conf': Found Nov 30 18:20:29 NOTICE[2625]: cdr.c:1185 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 1 - 2 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [ExecIfTime] == Registered application 'ExecIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [Set] == Registered application 'Set' [SetVar] == Registered application 'SetVar' [ImportVar] == Registered application 'ImportVar' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [res_musiconhold.so] = (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' == Parsing '/etc/asterisk/musiconhold.conf': Found Nov 30 18:20:29 WARNING[2625]: res_musiconhold.c:833 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. [res_features.so] = (Call Features Resource) == Parsing '/etc/asterisk/features.conf': Found Nov 30 18:20:29 WARNING[2630]: res_musiconhold.c:421 spawn_mp3: Found no files in '/usr/share/asterisk/mohmp3' Nov 30 18:20:29 WARNING[2630]: res_musiconhold.c:488 monmp3thread: Unable to spawn mp3player == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [chan_capi.so] = (Common ISDN API for Asterisk) == Parsing '/etc/asterisk/capi.conf': Found == Registered channel type 'CAPI' (Common ISDN API Driver ($Revision: 1.115 $) ) == Registered application 'capiCommand' == Registered custom function VANITYNUMBER [res_indications.so] = (Indications Configuration) == Parsing '/etc/asterisk/indications.conf': Found == Registered application 'PlayTones' == Registered application 'StopPlayTones' [res_monitor.so] = (Call Monitoring Resource) == Registered application 'Monitor' == Registered application 'StopMonitor' == Registered application 'ChangeMonitor' == Manager registered action Monitor == Manager registered action StopMonitor == Manager registered action ChangeMonitor [res_adsi.so] = (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Found [res_agi.so] = (Asterisk Gateway Interface (AGI)) == Registered application 'DeadAGI' == Registered
[Asterisk-Users] Sipura SPA-3000 SPA-2002 - Unable to dial *99
Hi folks, I am having a small problem with a few Sipura units. The settings are pretty much factory stock: the unit is set up to not register and the IP address for the unit is static and defined in the SIP setup for that unit. All other calls are sent and received properly, this is the only problem I have. When I dial *99 from the phone connected to line 1, I cannot complete a call. Instead of completing the call, I still have a dialtone. The only thing I can think of is that the units are somehow setup to ignore 9 and still play the dialtone, but I haven't seen anything in the interface to specify that. I have tried specifying various dialplans to make sure that *xx is sent to the Asterisk server, but no joy. Another oddity is that dialing *98 works just fine, only *99 fails. If anyone else has run into this and found a solution, I'd love a pointer in the right direction. Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] iaxmodem
Here is the example The output of asterisk: -- Registered IAX2 '4000' (AUTHENTICATED) at 127.0.0.1:33384 The output of iaxmodem: [EMAIL PROTECTED] iaxmodem-0.0.5]# ./iaxmodem ttyIAX Setting device = '/dev/ttyIAX' Setting port = 4569 Setting refresh = 300 Setting server = '127.0.0.1' Setting peername = '4000' Setting secret = 'password' Setting cidname = 'faxnameconfig' Setting cidnumber = '4000' Setting codec = slinear Opened pty, slave device: /dev/pts/7 Created /dev/ttyIAX symbolic link Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 28154 DCall: 0 [127.0.0.1:4569] USERNAME: 4000 REFRESH : 300 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 2ms SCall: 3 DCall: 28154 [127.0.0.1:4569] AUTHMETHODS : 3 CHALLENGE : 137050741 USERNAME: 4000 Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 9ms SCall: 28154 DCall: 3 [127.0.0.1:4569] USERNAME: 4000 MD5 RESULT : 7d5bc6839f3e8f7d931493ed3a029214 REFRESH : 300 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00019ms SCall: 6 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00019ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00019ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00021ms SCall: 3 DCall: 28154 [127.0.0.1:4569] USERNAME: 4000 DATE TIME : 192826213 REFRESH : 300 APPARENT ADDRES : IPV4 127.0.0.1:33384 MESSAGE COUNT : 0 CALLING NUMBER : 4000 CALLING NAME: modem4000 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00021ms SCall: 28154 DCall: 3 [127.0.0.1:4569] Registration completed successfully. Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00019ms SCall: 6 DCall: 28155 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02000ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 6 DCall: 28155 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 2ms SCall: 7 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 2ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 2ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 2ms SCall: 7 DCall: 28156 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02001ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 7 DCall: 28156 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00014ms SCall: 4 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00014ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00014ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00014ms SCall: 4 DCall: 28157 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02001ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 28157 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 3 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 7ms SCall: 28158 DCall: 3 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 7ms SCall: 28158 DCall: 3 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 7ms SCall: 3
Re: [Asterisk-Users] Comedian Voicemail? PROBLEMS?
Martin Joseph wrote: It's format=wav49|gsm|wav Try swapping the wav49 and the wav; my voicemail messages were garbled until I did this: format=wav|gsm|wav49 You should try not to just tack one line on top of a long message to list... ;~) ok, sorry :] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-3000 SPA-2002 - Unable to dial *99
When I dial *99 from the phone connected to line 1, I cannot complete a call. Go to the Regional tab in the advanced admin menu, find the Vertical Service Activation Codes section. Remove which ones you don't want the Sipura to handle (i.e. *99). Luki ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-3000 SPA-2002 - Unable to dial *99
I am having a small problem with a few Sipura units. The settings are pretty much factory stock: the unit is set up to not register and the IP address for the unit is static and defined in the SIP setup for that unit. All other calls are sent and received properly, this is the only problem I have. When I dial *99 from the phone connected to line 1, I cannot complete a call. Instead of completing the call, I still have a dialtone. The only thing I can think of is that the units are somehow setup to ignore 9 and still play the dialtone, but I haven't seen anything in the interface to specify that. I have tried specifying various dialplans to make sure that *xx is sent to the Asterisk server, but no joy. Another oddity is that dialing *98 works just fine, only *99 fails. If anyone else has run into this and found a solution, I'd love a pointer in the right direction. Since those codes are also listed as Vertical Service Activation Codes on the Regional tab, did you try to disable them on that page? Maybe the sipura is still interpreting those as opposed to following your dialplan. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] hierarchical VoIP system
Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao Pereira www.fccn.pt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Static on inside end of conversation
I have a very similar server, pstn setup, phones, and user base, and I switched over to G729 codec 'cause the polycoms support it. While the call quality has dropped ever so slightly (I have received no complaints from my users however), snaps, crackles, clicks and pops are gone. I did not have as extreme a case of static it seems, though. This will take more processor power, and will probably make any lost interrupts more evident. Moj Jeff Busch wrote: Hello, I am running the following configuration: 2.8ghz P4 with 1GB of RAM Audiocodes MP-108 connected to 5 POTS lines Polycom IP-500 phones [EMAIL PROTECTED] 1.3 (this is Asterisk 1.0.9) End users are complaining of an echo and static on the inside end (the internal side), but the outside end of the conversation doeesn't notice anything. Does anyone have any suggestions on troubleshooting / fixing this problem? Thanks! Jeff ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] [EMAIL PROTECTED] isdn
On Wed, November 30, 2005 16:29, Alejandro Vargas said: 2005/11/30, Francesco Peeters [EMAIL PROTECTED]: When you do, make VERY sure the PCI slots are NOT sharing an IRQ! That'll break it every time! Did you try to use APIC? This is suposed to solve the problem of IRQs Yep, tried APIC, NOAPIC, ACPI=OFF, etc. (capitals only for clarity!) but to no avail! As soon as both share the same IRQ, the zaphfc driver stops passing data to asterisk... The easiest fix was to swap the cards with other cards in the system to spread out the IRQ... Problem solved! -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura SPA-3000 SPA-2002 - Unable to dial *99
On Nov 30, 2005, at 12:39 PM, Luki wrote: When I dial *99 from the phone connected to line 1, I cannot complete a call. Go to the Regional tab in the advanced admin menu, find the Vertical Service Activation Codes section. Remove which ones you don't want the Sipura to handle (i.e. *99). DOH! Don't know why I didn't try those, because I did see them there. What's weird is that if I remove the entry for *99, I can now dial *99 and it works. However, there is still a definition for *98 in there as well, and that has worked all along. Weird. Tom Tom Rymes Cascade Link Systems www.cascadelinksystems.com (603) 375-1414 Intelligent technology solutions for small businesses. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] US e911 reminder
Just a reminder tonight at midnight is the deadline for pstn connected VoIP providers operating in the US to provide E911 or face fines upto $11,000 per day. There is also a filing requirement with the FCC which is due tonight as well. Enforcement Bureau Outlines Requirements of November 28, 2005 Interconnected Voice Over Internet Protocol 911 Compliance Letters http://www.fcc.gov/eb/Public_Notices/DA-05-2945A1.html Consumer page but has some basic info http://ftp.fcc.gov/cgb/consumerfacts/voip911.html -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] hierarchical VoIP system
On Wed, 2005-11-30 at 17:45 +, Joao Pereira wrote: Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed Have you considered enum for the voip enabled phones and failing through to either realtime or extensions.conf if enum fails? tip I found enum is easier to manage with powerdns and the mysql backend (although it can do postgress, isc bind, and other stuff for its backend, it seems faster for me and many have reported a much lower memory footprint when doing thousands of zones). That would seem to accomplish what you want and make it easier to port people over to voip as needed. Infact depending on how you configure everything, everyone could be in enum even the old legacy routes, then its a simple matter of editing what is already there. At least that has been my experience. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? SER tends to deal with large numbers of sip registrations better than asterisk on the same hardware. Mostly because it is specifically written for just that task. realtime may change that (I havent seen any specific studies done on load issues post realtime so I cant comment as I havent done any personally). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] hierarchical VoIP system
You should take a look to ENUM protocol: http://www.voip-info.org/wiki/view/ENUM. It could provide a decentralized and simple solution for your requirements. Regards -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Joao Pereira Enviado el: miércoles, 30 de noviembre de 2005 18:45 Para: [EMAIL PROTECTED]; asterisk-users@lists.digium.com Asunto: [Asterisk-Users] hierarchical VoIP system Hello Im managing a WAN with a lot of Universities. Some of them already installed a VoIP solution based on SER (to manage SIP clients) and Asterisk (for services and PSTN GW). The DNS routing provided by SER is working perfectly, but we want to start routing all calls thru IP transparently. We want our legacy PBXs (that are connected to Asterisk) to forward all calls to IP. The idea is to forward all calls to a central VoIP server, that has all the numbers that already are VoIP enabled, and then: - if the called number is VoIP enabled, he routes the call to that Univ. VoIP server - if the called number isnt in the list, the call goes back to the PBX and a PSTN call is dialed This way, ppl starts using the VoIP infrastructure, without even knowing what VoIP means, and the telecom bill starts decreasing. I know thats a statical and hierarchical structure and we dont want that, but is a good solution for this migration phase, where a lot of places are still using TDM systems. Now, the top of the hierarchy should be an Asterisk or SER? I dont know which of the systems is the best choice for the job. Does someone has an idea of what should we use? Thanks Joao Pereira www.fccn.pt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] US e911 reminder
On Mon, 2005-11-28 at 15:01 -0800, trixter aka Bret McDanel wrote: Just a reminder tonight at midnight is the deadline for pstn connected VoIP providers operating in the US to provide E911 or face fines upto $11,000 per day. There is also a filing requirement with the FCC which is due tonight as well. sorry for this being lagged, moved servers and mail was just being queued. To clarify the deadline was midnight last monday. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 http://www.sacaug.org/ Sacramento Asterisk Users Group signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] pbx or asterisk?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Looks like your zap channels are droping into the default context... better to set up a from-pstn context and start there. Pablo Allietti wrote: hi all i have a pbx siemens connect via E1 to my asterisk box. the asterisk box can call without problems to pbx extensions. but when y press the numbers form example 402 in the pbx phones asterisk give me this -- Saved useragent X-Lite release 1103m for peer 402 -- Going to extension s|1 because of Complete received -- Executing Playback(Zap/31-1, vm-goodbye) in new stack -- Accepting call from '' to 's' on channel 0/31, span 1 == Spawn extension (default, s, 1) exited non-zero on 'Zap/31-1' -- Hungup 'Zap/31-1' -- Accepting call from '' to 's' on channel 0/31, span 1did not receive any number or i have miss configure somenthing in asterisk box? -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFDjey8y9wPyZpnL2URAiVCAJ4hQCz+eb1/MaABy2gxUMOcMw1AMwCfYEJI VTt9lDiRDMLZhJ2aOL4Qpnw= =KqmL -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Snom 320s and the hint priority
Hi everyone, Does anyone have this working? I'm looking at these phones for my receptionist phone, with the requirement that the two bars of buttons and lights on the side show line presence for programmable extensions ( ie: line 1 show the presense of my 101 user, line 2 = 102 user, ect.. ). I don't want to buy them only to find they can't do this, so I was hoping someone on the list had these suckers up and running. Thanks Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX Service providers in Australia for unlimitedinbound
Try www.oztell.com they have a somewhat complicated website interface but once you figure it out its ok and I found them to be by far the cheapest provider in Australia. They offer DIDs at $1.95 per month. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Thursday, 1 December 2005 2:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAX Service providers in Australia for unlimitedinbound Can anyone on the list recommend any IAX Service providers in Australia for unlimited inbound in the 02 area code? Ive been using Faktortel for A$9.50 per month and although the outbound is fantastic (I mean the quality is fantastic the fixed price 10c per call Australia wide is pretty good as well) the inbound has been getting worse and worse. I keep getting calls with un-usable echo etc which means I need to hang-up and call them back etc. Is there anyone who can recommend an alternative, they must be able to offer multiple inbound calls (faktortel allows me 4 simultaneous inbound calls at the moment). Cheers, Dean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iaxmodem
Are you using the libiax2 that came with iaxmodem? If you are, then I'm not sure what to say... the client-server behavior looks bizarre. Lee. Miguel Soto wrote: Here is the example The output of asterisk: -- Registered IAX2 '4000' (AUTHENTICATED) at 127.0.0.1:33384 The output of iaxmodem: [EMAIL PROTECTED] iaxmodem-0.0.5]# ./iaxmodem ttyIAX Setting device = '/dev/ttyIAX' Setting port = 4569 Setting refresh = 300 Setting server = '127.0.0.1' Setting peername = '4000' Setting secret = 'password' Setting cidname = 'faxnameconfig' Setting cidnumber = '4000' Setting codec = slinear Opened pty, slave device: /dev/pts/7 Created /dev/ttyIAX symbolic link Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 28154 DCall: 0 [127.0.0.1:4569] USERNAME: 4000 REFRESH : 300 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 2ms SCall: 3 DCall: 28154 [127.0.0.1:4569] AUTHMETHODS : 3 CHALLENGE : 137050741 USERNAME: 4000 Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 9ms SCall: 28154 DCall: 3 [127.0.0.1:4569] USERNAME: 4000 MD5 RESULT : 7d5bc6839f3e8f7d931493ed3a029214 REFRESH : 300 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00019ms SCall: 6 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00019ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00019ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00021ms SCall: 3 DCall: 28154 [127.0.0.1:4569] USERNAME: 4000 DATE TIME : 192826213 REFRESH : 300 APPARENT ADDRES : IPV4 127.0.0.1:33384 MESSAGE COUNT : 0 CALLING NUMBER : 4000 CALLING NAME: modem4000 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00021ms SCall: 28154 DCall: 3 [127.0.0.1:4569] Registration completed successfully. Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00019ms SCall: 6 DCall: 28155 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02000ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 6 DCall: 28155 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 2ms SCall: 7 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 2ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 2ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 2ms SCall: 7 DCall: 28156 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02001ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 7 DCall: 28156 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00014ms SCall: 4 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00014ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00014ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00014ms SCall: 4 DCall: 28157 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02001ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 28157 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 3 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 7ms SCall: 28158 DCall: 3 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 7ms SCall: 28158 DCall: 3 [127.0.0.1:4569]
RE: [Asterisk-Users] iaxmodem
The Remote hangup messages disappear if I set qualify=no in the iax.conf file. But is this correct? Miguel -Original Message- From: Miguel Soto [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 30, 2005 10:31 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] iaxmodem Here is the example The output of asterisk: -- Registered IAX2 '4000' (AUTHENTICATED) at 127.0.0.1:33384 The output of iaxmodem: [EMAIL PROTECTED] iaxmodem-0.0.5]# ./iaxmodem ttyIAX Setting device = '/dev/ttyIAX' Setting port = 4569 Setting refresh = 300 Setting server = '127.0.0.1' Setting peername = '4000' Setting secret = 'password' Setting cidname = 'faxnameconfig' Setting cidnumber = '4000' Setting codec = slinear Opened pty, slave device: /dev/pts/7 Created /dev/ttyIAX symbolic link Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 28154 DCall: 0 [127.0.0.1:4569] USERNAME: 4000 REFRESH : 300 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 2ms SCall: 3 DCall: 28154 [127.0.0.1:4569] AUTHMETHODS : 3 CHALLENGE : 137050741 USERNAME: 4000 Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 9ms SCall: 28154 DCall: 3 [127.0.0.1:4569] USERNAME: 4000 MD5 RESULT : 7d5bc6839f3e8f7d931493ed3a029214 REFRESH : 300 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00019ms SCall: 6 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00019ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00019ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00021ms SCall: 3 DCall: 28154 [127.0.0.1:4569] USERNAME: 4000 DATE TIME : 192826213 REFRESH : 300 APPARENT ADDRES : IPV4 127.0.0.1:33384 MESSAGE COUNT : 0 CALLING NUMBER : 4000 CALLING NAME: modem4000 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00021ms SCall: 28154 DCall: 3 [127.0.0.1:4569] Registration completed successfully. Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00019ms SCall: 6 DCall: 28155 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02000ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 6 DCall: 28155 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 2ms SCall: 7 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 2ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 2ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 2ms SCall: 7 DCall: 28156 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02001ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 7 DCall: 28156 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00014ms SCall: 4 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00014ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00014ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00014ms SCall: 4 DCall: 28157 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02001ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 28157 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 3 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 7ms
RE: [Asterisk-Users] iaxmodem
Yes, I am using libiax2 that came with iaxmodem :) -Original Message- From: Lee Howard [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 30, 2005 11:43 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] iaxmodem Are you using the libiax2 that came with iaxmodem? If you are, then I'm not sure what to say... the client-server behavior looks bizarre. Lee. Miguel Soto wrote: Here is the example The output of asterisk: -- Registered IAX2 '4000' (AUTHENTICATED) at 127.0.0.1:33384 The output of iaxmodem: [EMAIL PROTECTED] iaxmodem-0.0.5]# ./iaxmodem ttyIAX Setting device = '/dev/ttyIAX' Setting port = 4569 Setting refresh = 300 Setting server = '127.0.0.1' Setting peername = '4000' Setting secret = 'password' Setting cidname = 'faxnameconfig' Setting cidnumber = '4000' Setting codec = slinear Opened pty, slave device: /dev/pts/7 Created /dev/ttyIAX symbolic link Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 28154 DCall: 0 [127.0.0.1:4569] USERNAME: 4000 REFRESH : 300 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 2ms SCall: 3 DCall: 28154 [127.0.0.1:4569] AUTHMETHODS : 3 CHALLENGE : 137050741 USERNAME: 4000 Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: REGREQ Timestamp: 9ms SCall: 28154 DCall: 3 [127.0.0.1:4569] USERNAME: 4000 MD5 RESULT : 7d5bc6839f3e8f7d931493ed3a029214 REFRESH : 300 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00019ms SCall: 6 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00019ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00019ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: REGACK Timestamp: 00021ms SCall: 3 DCall: 28154 [127.0.0.1:4569] USERNAME: 4000 DATE TIME : 192826213 REFRESH : 300 APPARENT ADDRES : IPV4 127.0.0.1:33384 MESSAGE COUNT : 0 CALLING NUMBER : 4000 CALLING NAME: modem4000 Tx-Frame Retry[-01] -- OSeqno: 002 ISeqno: 002 Type: IAX Subclass: ACK Timestamp: 00021ms SCall: 28154 DCall: 3 [127.0.0.1:4569] Registration completed successfully. Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00019ms SCall: 6 DCall: 28155 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02000ms SCall: 28155 DCall: 6 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 6 DCall: 28155 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 2ms SCall: 7 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 2ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 2ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 2ms SCall: 7 DCall: 28156 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02001ms SCall: 28156 DCall: 7 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 7 DCall: 28156 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 00014ms SCall: 4 DCall: 0 [127.0.0.1:4569] Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00014ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG Timestamp: 00014ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Unknown IE 046 : Present Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 00014ms SCall: 4 DCall: 28157 [127.0.0.1:4569] Tx-Frame Retry[010] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: PING Timestamp: 02001ms SCall: 28157 DCall: 4 [127.0.0.1:4569] Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: INVAL Timestamp: 0ms SCall: 4 DCall: 28157 [127.0.0.1:4569] Remote hangup. Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: POKE Timestamp: 7ms SCall: 3 DCall: 0
Re: [Asterisk-Users] Snom 320s and the hint priority
On 10:25, Wed 30 Nov 05, Sean Kennedy wrote: Hi everyone, Does anyone have this working? I'm looking at these phones for my receptionist phone, with the requirement that the two bars of buttons and lights on the side show line presence for programmable extensions ( ie: line 1 show the presense of my 101 user, line 2 = 102 user, ect.. ). I don't want to buy them only to find they can't do this, so I was hoping someone on the list had these suckers up and running. Works great here :) -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? signature.asc Description: Digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 320s and the hint priority
Hey, you are my new best friend. I have never had a phone to use with the hint priority, would you mind giving me a sample of your configuration so I can figure it out? Much apprecaited! Sean Michiel van Baak wrote: On 10:25, Wed 30 Nov 05, Sean Kennedy wrote: Hi everyone, Does anyone have this working? I'm looking at these phones for my receptionist phone, with the requirement that the two bars of buttons and lights on the side show line presence for programmable extensions ( ie: line 1 show the presense of my 101 user, line 2 = 102 user, ect.. ). I don't want to buy them only to find they can't do this, so I was hoping someone on the list had these suckers up and running. Works great here :) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Queue calls...
I want to play a file for an agent that answers a queue call, before the agent is actually connected with the call. I want something along the lines of,Answer as member of team X, or similar, before the agent is connected with the caller. Is this possible? And how would I do it? -- Trey Blancher Systems Administrator, USA Debt Management LLC (251)445-0683 ext 8601 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 320s and the hint priority
Hi Sean, Works fine for me as well. Took some working to get right. There's a very recent thread on this, see: http://lists.digium.com/pipermail/asterisk-users/2005-November/136343.html Also, you'll need to go into the web interface for your Snom phones and configure each button for each line you want to monitor. We have Snom 360's. For us, in the web menu, you go to Function Keys and program as many of the 'P' (e.g. P1, P2, etc) with the extension number and set the drop down menu to 'Destination'. Hope that helps. Sascha Sean Kennedy wrote: Hi everyone, Does anyone have this working? I'm looking at these phones for my receptionist phone, with the requirement that the two bars of buttons and lights on the side show line presence for programmable extensions ( ie: line 1 show the presense of my 101 user, line 2 = 102 user, ect.. ). I don't want to buy them only to find they can't do this, so I was hoping someone on the list had these suckers up and running. Thanks Sean ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] problem with zaptel 1.2.0 and pulse dialing
Hi all, Pulse dialing is not working on my asteriskathome configuration with asterisk 1.2.0 and zaptel 1.2.0 in France. I've pulsedial=yes in zapata.conf. Tone dialing is working 100%. In file zapata.h (zaptel 1.2.0), I've found the following : #define ZT_DEFAULT_PULSEMAKETIME 50 /* 50 ms of line closed when dial pulsing */ #define ZT_DEFAULT_PULSEBREAKTIME 50/* 50 ms of line open when dial pulsing */ #define ZT_DEFAULT_PULSEAFTERTIME 750 /* 750ms between dial pulse digits */ #define ZT_MINPULSETIME (15 * 8)/* 15 ms minimum */ #define ZT_MAXPULSETIME (200 * 8) /* 200 ms maximum */ Is this configuration compatible with French analog PSTN ? Is there any possibilities to change these values in any configuration file without need to compile again ? Is there still anyone using pulse dialing ? What your configuration is ? Thanks. Cyrille DERORY ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] problem with zaptel 1.2.0 and pulse dialing
I use 1.2 Beta 1 in pulse dial mode, as an interface to an EM switch. A bunch of collectors in the US and 2 in the UK have a private network of historic switches interconnected via the Internet and Asterisk. If the make break time is a problem you can change in the source but will have to recompile. Also, we have found that with the TDM400 FXS card, the detection of dial pulses is intolerant of slightly out of spec dials, either speed or make break. You also need to be aware that the w or wait parameter in the dial command only works with DTMF. That coupled with Asterisk not detecting dial tone can result in many mis (pulse) dialed calls to the PSTN So YES, quite a few still use, and will continue to use pulse dial [EMAIL PROTECTED] was not used as the configuration for our more or less special needs proved too cumbersome. Good luck. John Novack Cyrille DERORY wrote: Hi all, Pulse dialing is not working on my asteriskathome configuration with asterisk 1.2.0 and zaptel 1.2.0 in France. I've pulsedial=yes in zapata.conf. Tone dialing is working 100%. In file zapata.h (zaptel 1.2.0), I've found the following : #defineZT_DEFAULT_PULSEMAKETIME 50/* 50 ms of line closed when dial pulsing */ #defineZT_DEFAULT_PULSEBREAKTIME 50/* 50 ms of line open when dial pulsing */ #defineZT_DEFAULT_PULSEAFTERTIME 750/* 750ms between dial pulse digits */ #defineZT_MINPULSETIME (15 * 8)/* 15 ms minimum */ #defineZT_MAXPULSETIME (200 * 8)/* 200 ms maximum */ Is this configuration compatible with French analog PSTN ? Is there any possibilities to change these values in any configuration file without need to compile again ? Is there still anyone using pulse dialing ? What your configuration is ? Thanks. Cyrille DERORY ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disposition failed in Asterisk-1.2.0-stable
We just upgraded our current asterisk cluster to the release version of Asterisk 1.2.0. Strange enough, out of the 11000+ calls, only 720 (and counting) have a disposition of FAILED in the cdr's. These 720+ have only occurred after the upgrade, and I'm rather confused as to why it would show up in the CDR's like this. If anyone has a clue, please let me know, any help would be appreciated. Aaron Daniel [EMAIL PROTECTED] Sam Houston State University 936-496-3000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Snom 320s and the hint priority
On 11:33, Wed 30 Nov 05, Sean Kennedy wrote: Hey, you are my new best friend. I have never had a phone to use with the hint priority, would you mind giving me a sample of your configuration so I can figure it out? Much apprecaited! Hey hey new pal ;) First of all, have a look at this thread: http://lists.digium.com/pipermail/asterisk-users/2005-November/136343.html Second, here some parts of my extensions.conf for the lights on the phone: exten = 101,1,Dial(SIP/101) ;dial the phone exten = 101,2,Hangup() ;hangup channel, this is just for safety exten = 101,hint,SIP/101;notify snoms about status exten = 102,1,Dial(SIP/102) ;dial the phone exten = 102,2,Hangup() ;hangup channel, this is just for safety exten = 102,hint,SIP/102;notify snoms about status On the phones website make sure to configure the led-buttons in this way: Destination - number to monitor/speeddial That's all I had to do to make it work. Good luck -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Why is it drug addicts and computer afficionados are both called users? signature.asc Description: Digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to exit from Asterisk console.
I am new to Asterisk. Asterisk 1.2 I started * like this: asterisk -vgc now I am in CLI mode: *CLI How do I get out this CLI mode to linux shell without kill asterisk process? I tried EXIT, QUIT, exit and quit. None of them work. If I use ^c, this also kill asterisk process. GC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to exit from Asterisk console.
gc wrote: I started * like this: asterisk -vgc now I am in CLI mode: *CLI How do I get out this CLI mode to linux shell without kill asterisk process? if you want to run it like this, first do a screen (more info: man screen) so you can run it in a background shell. But I recommend on running it with the appropriate init script.. cheers ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE : [Asterisk-Users] How to exit from Asterisk console.
Title: Message just press Ctrl-C or type exit You will kill asterisk, of course... Start asterisk by typing asterisk and then go toCLI by typing asterisk -r then, when u will quit, asterisk will not be killed U will be then in CLI mode have fun -Message d'origine-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de gcEnvoyé: mercredi 30 novembre 2005 22:15À: asterisk-users@lists.digium.comObjet: [Asterisk-Users] How to exit from Asterisk console. I am new to Asterisk. Asterisk 1.2 I started * like this: asterisk -vgc now I am in CLI mode: *CLI How do I get out this CLI mode to linux shell without kill asterisk process? I tried EXIT, QUIT, exit and quit. None of them work. If I use ^c, this also kill asterisk process. GC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] How to exit from Asterisk console.
Kill the Asterisk process Launch Asterisk as a background process by typing asterisk or use the safe_asterisk shell script (better) type asterisk r to connect to the console Press Ctrl C to exit the console. Use ps a | grep Asterisk to determine if the Asterisk process is still running (it should be) Modify /etc/rc.d/rc.local to have step 3 in it as the last line. This will launch Asterisk as a background process on boot. hth -Original Message- From: gc [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 30, 2005 2:15 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to exit from Asterisk console. I am new to Asterisk. Asterisk 1.2 I started * like this: asterisk -vgc now I am in CLI mode: *CLI How do I get out this CLI mode to linux shell without kill asterisk process? I tried EXIT, QUIT, exit and quit. None of them work. If I use ^c, this also kill asterisk process. GC ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR issues
I'm having problems setting up the CDR functionality. Namely, it doesn't always wok (but I do have some records). When typing cdr mysql status in the Asterisk console, it does say connected for 3 minutes 22 seconds, with 0 records added since last restart. But I did call a few times into my PBX, so what is the issue? Thing is, somehow (and I didn't change any config) there are three records into the CDR table. And they correspond with real calls. It just stopped taking in more, somehowand those three weren't in sequence, the system missed a few calls. My biggest grip is I don't know where to troubleshoot this. Any log files I can look at? The message log in var/log/asterisk only shows that I am using simple CDR. Next natural question: When I dial into my PBX, and my PBX dials out to make a bridge, can the CDR DB show the two calls (the incoming one and the outgoing one) separately? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
