[asterisk-users] seems like call is picked and returned to me
Hi all I hope that someone of you can solve this. Right now I'm stuck! I'm using asterisk with some SIP extensions. Basically I want to establish a call between desktop voip phone (ext 181) and embedded sip system (ext 182) All I can see in CLI is: == Using SIP RTP CoS mark 5 -- Executing [182@default:1] Dial(SIP/181-000a, SIP/182) in new stack == Using SIP RTP CoS mark 5 -- Called SIP/182 -- SIP/182-000b is ringing -- SIP/182-000b is making progress passing it to SIP/181-000a -- SIP/182-000b answered SIP/181-000a -- Remotely bridging SIP/181-000a and SIP/182-000b == Spawn extension (default, 182, 1) exited non-zero on 'SIP/181-000a' Seems like extension 182 (called ext) is getting call and passing them another time to me 181 (origin call) I've try it with siemens pbx and works as expected cheers! Sergio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8, busylevel and CCBS
My question is so complex and I try to explain well.
We have a customer that he wants limits incoming calls to his extensions
to only one. That's not complicated with GROUPCOUNT, DEVICE_STATE or
SIPPEER with curcalls option.But the problem is when you want implement
CCBS service.
If we have next context:
exten=_XXX,1,NOOP()
same=n,GotoIF($[${DEVICE_STATE(${ARG2})}=BUSY]?occupied)
same=n,Dial(SIP/${EXTEN})
same=n,GotoIf($[${DIALSTATUS}=BUSY]?ocupado)
same=n,Hangup()
same=n(occupied),Busy()
same=n,Hangup()
If we call to 100 extensions and that extensions reject call or no
answer call, we can use CallCompletionRequets to request CCNR service
and all work fine.
But when a call is on 100 extension, and you call to 100 extension and
go to occupied label, if you reques a CCBS with
CallCompletionRequest() this application fails with NO_CORE_INSTANCE
error.
It's appear like CCSS only work with DIALSTATUS variable and with Dial
application I don't know how to limit to only one incoming call.
Are there any way to solve this?
Any help would be appreciated.
regards,
Sergio
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RE: [Asterisk-Users] Re: Asterisk Redundency
A good solution is use a program that use sipsack for SIP, something like sipsack for IAX and Linux-HA for asterisk. In this way you check if SIP or IAX is OK, and if these technologies are bad, you can kill asterisk and linux-HA will do the rest. In PSTN Field, you can check rxhooksig in struct zt_params for analog FXO lines, but you must use ISDN Guard for PRI lines. I'm working in this way that failover and I will announce when I have something to test. Regards, srsergio -Mensaje original- De: Matt [mailto:[EMAIL PROTECTED] Enviado el: miércoles, 26 de octubre de 2005 17:21 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Re: Asterisk Redundency Right got it.. sorry I thought I stopped that e-mail from going out. Very cool! Can you give me an idea of what you do for DNS SRV to get the sip devices to flip? Or do you just have the other asterisk server take over the IP of the old one (seems like a good solution). On 10/26/05, Patrick [EMAIL PROTECTED] wrote: On Wed, 2005-10-26 at 11:02 -0400, Matt wrote: On the PRI side you can use the failover equipment from e.g. junghanns.net. Sorry I'm not seeing failover equipment? I'm seeing PRI cards and an ISDN guard? PRI | | ISDNguard / \ / \ / PRI \ / \ Server1 Server2 Afaik if the PRI card on Server1 goes down than the ISDNguard will automatically reroute all calls over the 2nd PRI link to Server2. See the picture in the doc at http://www.junghanns.net/downloads/ISDNguard_en.pdf Regards, Patrick ___ --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 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.12.4/146 - Release Date: 21/10/2005 ___ --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
RE: [Asterisk-Users] E1/T1 failover hardware
http://www.junghanns.net/en/ISDNguard_produkt.html srsergio -Mensaje original- De: John Daragon [mailto:[EMAIL PROTECTED] Enviado el: jueves, 20 de octubre de 2005 17:24 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] E1/T1 failover hardware Warning ! I know zip about electronics. I've been looking for a device to handle the switching of an E1 connection from one Asterisk box to another in the event of a catastrophic server failure. All of the solutions I've seen so far have been designed to handle the situation where the telco line faults so that the local PBX can switch to a secondary E1. I've come across this application note : http://www.maxim-ic.com/appnotes.cfm/appnote_number/2857 which describes T1/E1/J1, N+1 Redundancy With Analog Switches These parts are obviously designed to be built into E1 boards - hence, I think, the protection circuitry. Here's the question, then : what (apart from jumping through regulatory hoops) is to stop a simple array of MOSFETS (and a bit of control circuitry) implementing a failover switch controlled (say) by a pin on a serial or parallel port ? jd -- John Daragon [EMAIL PROTECTED] argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.12.4/143 - Release Date: 19/10/2005 ___ --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
RE: [Asterisk-Users] TDM400P off-hook detection problem
Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog line, and FXS port is for connect analog phone. Are you sure that in 3rd and 4th ports you have immediate=no? regards, srsergio -Mensaje original- De: Alex Ongena [mailto:[EMAIL PROTECTED] Enviado el: jueves, 13 de octubre de 2005 11:17 Para: Asterisk Asunto: [Asterisk-Users] TDM400P off-hook detection problem Hi list, I have a Wildcard TDM400P REV I Board 1 with 4 FXO modules and * 1.0.9 up-and-running. Only 2 FXO ports are used for 2 analog phones and are doing fine. I now wanted to use the 3rd and 4th port, but when I insert an analog phone, take it off hook, I do not get a dial tone. With my 1st and 2nd port, I get messages like: -- Starting simple switch on 'Zap/13-1' -- Hungup 'Zap/13-1' on my CLI, but with port 3 and 4, I don't see anything. I have tried with the same phone that works well in port 1 and 2, so it's not related to the phone. The configuration for port 3 and 4 is idential to 1 and 2. zap show channel xx does not show anything special and what it show is identical between port 1,2 and 3,4. It's a production system, so it's not easy to stop and start troubleshooting it, certainly not easy to open and swap modules Anybody seen something similar ? Thank Alex -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.14/131 - Release Date: 12/10/2005 ___ --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
RE: [Asterisk-Users] TDM400P off-hook detection problem
Check your Revision card, if it is Rev H in zaptel sources you have a zconfig.h with a Flag to Revision H. Try it. regards, -Mensaje original- De: Alex Ongena [mailto:[EMAIL PROTECTED] Enviado el: jueves, 13 de octubre de 2005 12:56 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] TDM400P off-hook detection problem On Thu, 2005-10-13 at 12:44 +0200, Sergio Serrano wrote: Your card must be a TDM with 4 FXS ports. FXO port is to connect and analog line, and FXS port is for connect analog phone. Are you sure that in 3rd and 4th ports you have immediate=no? if it may help, I could just stop *, # rmmod wcfxs # modprobe wcfxs # asterisk and now all ports are working fine ??? I Googled around and found someone with a similar problem 5 okt 2004. It happened after 2 weeks of operation ? I think it's still an issue in the driver... Alex -- NEW: aXs GUARD hands-on Trainings v.7.0 more info at http://www.axsguard.com/indextraining.htm aXs GUARD has completed security and anti-virus checks on this e-mail (http://www.axsguard.com) --- Able NV: ond.nr 0457.938.087 ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.14/131 - Release Date: 12/10/2005 ___ --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
RE: [Asterisk-Users] Quad PRI Problems
You can't put four span in timing, because only one must be like nmaster sincronization. If one of your telco provide time for your card. Put second value in all span to 0. regards, srsergio -Mensaje original- De: Ronald Hartmann [mailto:[EMAIL PROTECTED] Enviado el: martes, 04 de octubre de 2005 14:33 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Quad PRI Problems I have been getting quite a bit of PRI Resets using my Quad PRI Digium card. Prior to the resets I am getting similar notices to the following chan_zap.c:7482 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary D-channel of span 3 Telco claims the PRI's are fine on their end and that it is my unit. Is this timing? (google somewhat leads to this) I am running 1.08 asterisk zaptel libpri. Any help would be greatly appreciated. ~ron Zaptel.conf span=1,1,0,esf,b8zs # connects to an Adtran FXS TA624 em=1-24 span=2,1,0,esf,b8zs # Connects to Bell Company 1 bchan=25-47 dchan= 48 span=3,1,0,esf,b8zs # Connects to Bell Company #2 bchan=49-71 dchan= 72 span=4,1,0,esf,b8zs # Connects to Brook Trout CArd em=1-4 defaultzone=us loadzone=us [channels] context=from-internal-receiver ; Points to the default context of your extensions.conf language=en faxdetect=none usecallerid=yes callerid=asreceived threewaycalling=yes transfer=yes signalling=featd ; FXS for ringing phones group=0 flash=350 rxwink=300 prewink=20~~ echocancel=no ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=no immediate=no channel = 1-24 signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national pridialplan=unknown echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=no echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds group=1 context=from-pstn channel = 25-47 ; Set this to 1-15,17-31 for E1 group=2 signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national channel = 49-71 ; Set this to 1-15,17-31 for E1 group=3 signaling=em_w channel = 73-76 ___ --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
[Asterisk-Users] VideoConference with UMTS
Hi Srs., Do you know if it's possible make a videocall from asterisk to UMTS mobile phone?. Both technologies use H.263 like videocodec. Any idea? ___ --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
RE: [Asterisk-Users] call center software and asterisk
www.inconcertCC.com has a solution based on Asterisk. regards, srsergio -Mensaje original- De: Bartosz Jozwiak [mailto:[EMAIL PROTECTED] Enviado el: jueves, 29 de septiembre de 2005 17:17 Para: Asterisk Users Mailing List - Non-Commercial Discussion CC: Commercial and Business-Oriented Asterisk Discussion Asunto: [Asterisk-Users] call center software and asterisk Hi guys, Need some advise. Is there some kind of call center software which can interconnect with asterisk? So, for example, agents can see on their pc's all info about calling client (based on clid) before they pick up the phone. And that outbound calls are also automated. Commercial solutions more then welcome. Thx, Bartosz Jozwiak ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.8/114 - Release Date: 28/09/2005 ___ --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
RE: [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1)
You must install libncurses5-dev regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Enviado el: martes, 27 de septiembre de 2005 9:20 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: [Asterisk-Users] Termcap missing (compile error[editline/libedit.a] Error 1) Hello Gentlemen :-) I am a little disapointed by an error occured during an update from 1.0.7 to Head in a Debian testing distro. The first error message happens by using the famous script from http://www.szmidt.org/asterisk/asterisk-update.sh : configure: error: termcap support not found make: *** [editline/libedit.a] Erreur 1 ERROR! Compile exited with error. Aborting script! And, if I tempt to compile manualy with make clean; make; make install, I can see that at the end : cd editline unset CFLAGS LIBS test -f config.h || ./configure loading 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 preprocessor... gcc -E checking host system type... i686-pc-linux-gnu cygwin detected checking for a BSD compatible install... install checking for ranlib... ranlib checking for ar... /usr/bin/ar checking for tgetent in -ltermcap... no checking for tgetent in -ltinfo... no checking for tgetent in -lcurses... no checking for tgetent in -lncurses... no configure: error: termcap support not found make: *** [editline/libedit.a] Erreur 1 sarge:/usr/src/asterisk# What occurs ? What I have missed ? Any idea to help me ? What can I describe or search more for a best analyze ? Many thanks in advance, guys ! Best Regards, Francois BERGERET, France. ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 23/09/2005 ___ --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
RE: [Asterisk-Users] IBM x306
Hi all, we have same problem with a x346. Mainly, TE410P shares IRQ with network card and if you change IRQ for this slot, automatically change IRQ in network card. Any idea? srsergio -Mensaje original- De: George Pajari [mailto:[EMAIL PROTECTED] Enviado el: lunes, 26 de septiembre de 2005 10:09 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] IBM x306 I am building an asterisk pbx (1.0.9) on an IBM x306 SCSI machine,my problem is that the BIOS assigns the same IRQ to the SCSI controller, and the TDM400P, i have tried several options of making the bios change the IRQ, but it will always move them together, anyone with some info about my options ? Check the BIOS options -- many others in the x3nn Series as well as the Netfinity before them allow you to specify the IRQ per slot through a deeply buried BIOS config option. I'm not near my rack of IBM servers to boot one to get the exact path but email me offline if you can't find it. g. -- George Pajari, netVOICE communications604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca www.sipura.ca www.snom.ca ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.6/111 - Release Date: 23/09/2005 ___ --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
RE: [Asterisk-Users] Queues
show application Queue is your friend. De: Sander [mailto:[EMAIL PROTECTED] Enviado el: viernes, 23 de septiembre de 2005 13:11Para: 'Asterisk Users Mailing List - Non-Commercial Discussion'Asunto: [Asterisk-Users] Queues Hi there i need to know if there is a wayto play a ringing sound to acallerthe enters a queue so i don't want to have music onhold and i need it to bebehind the answer option like this exten =1,1,Dial(sip/10,10) exten =1,2,Answer exten =1,3,Queue(test) thanks ___ --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
RE: [Asterisk-Users] Asterisk in Spanish
Try in www.asterisk-es.org -Mensaje original- De: Sebastian Milioto [mailto:[EMAIL PROTECTED] Enviado el: lunes, 19 de septiembre de 2005 15:08 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Asterisk in Spanish Hi all, I've been installing [EMAIL PROTECTED] and (of course) all the answering machine (I don't sure that's the right word in english, preatendedora in spanish) speech is in enlgish languaje. Is there anyway to download all those .gsm files speaked in spanish? Or may be another site which contain this kind of stuff (.wav, .gsm files for answering machines in spanish)? Thank you very much, Regards, Sebastian Milioto Telecommunications Engineer IM: [EMAIL PROTECTED] e-mail: [EMAIL PROTECTED] Mobile: 549 3571 543658 ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.11.1/104 - Release Date: 16/09/2005 ___ --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
RE: [Asterisk-Users] E1 configuration problem
Please, send us zaptel.conf and zapata.conf and say us what card you have(TE110P, TE410P...). And what is your country. Regards, srsergio -Mensaje original- De: manish kumar [mailto:[EMAIL PROTECTED] Enviado el: lunes, 19 de septiembre de 2005 6:32 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] E1 configuration problem I am trying to configure E1 card (Digium) but not able to do that. The green light doesn't come up when it starts. What can be the problem. I have also changed the jumper settings of the card from T1 to E1 but still no relief. Thanks in advance Manish ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.25/102 - Release Date: 14/09/2005 ___ --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
RE: [Asterisk-Users] Manipulate CALLERIDNUM
Hi, Try SetCIDNum application before VoiceMail application regards, srsergio -Mensaje original- De: Chad Brown [mailto:[EMAIL PROTECTED] Enviado el: miércoles, 31 de agosto de 2005 8:48 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] Manipulate CALLERIDNUM Can someone tell me how to do this...Given the following line: exten = *97,3,VoicemailMain([EMAIL PROTECTED]) Is it possible to add some logic to manipulate the CALLERIDNUM to send back 801 even if the extension is 601 and 901 even if the extension is 701? I have 2 branch offices where users have both Office and Home SIP phones. I want them to share a VM box. Branch1 = 8XX , Home = 7XX Branch2 = 9XX, Home = 6XX Therefore I would like to manipulate the home CALLERIDNUM in both examples. Make sense? Thanks in advance. ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.17/85 - Release Date: 30/08/2005 ___ --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
RE: [Asterisk-Users] unresolved symbol when loading ztdummy
This option is under Library routines in your kernel configuration. Regards, srsergio -Mensaje original- De: Christoph Eicke [mailto:[EMAIL PROTECTED] Enviado el: miércoles, 31 de agosto de 2005 10:59 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] unresolved symbol when loading ztdummy On Tuesday 30 August 2005 17:01, Braz wrote: Your kernel has to be compile with CONFIG_CRC_CCITT=y or m. I couldn't find that option in the kernel, but inserting the zaptel module before ztdummy works of course. ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.17/85 - Release Date: 30/08/2005 ___ --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
RE: [Asterisk-Users] queue - ringing members in order
Hi Sr. rrmemory i same like roundrobin, but this policy store which is the next when a call get into your system. For example with next queue: SIP/1 SIP/2 SIP/3 and roundrobin, all calls stars with SIP/1 and with rrmemory first call starts with SIP/1, second call with SIP/2 and so on. Regards, srsergio -Mensaje original- De: Christian Gansberger [mailto:[EMAIL PROTECTED] Enviado el: martes, 30 de agosto de 2005 13:22 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] queue - ringing members in order hi all, i m new to this list, I have a big problem, how to configure a Queue to follow the behaivor of: every incoming call should first ring the member listed first (in queues.conf) - then the second and so on. Is there a way to always start ringing with the first member of the queue? Here is a the queue definition: [res] musiconhold = moh-res strategy = roundrobin timeout = 20 retry = 0 maxlen = 0 member = SIP/1824 member = SIP/1816 member = SIP/1831 member = SIP/1832 Can anyone tell me difference between strategy roundrobin and rrmemory? i m really stuck with that problem so please help Thanks crs ___ --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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.344 / Virus Database: 267.10.17/84 - Release Date: 29/08/2005 ___ --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
RE: [Asterisk-Users] looking for failover ideas
How do you do monitoritng? How Server B knows that Servar A is down? I just do a rsync and MySQL Replication, but I try to do a C program that monitor Server. If you know how can I do this monitoring I will be pleasant with you. regards, srsergio -Mensaje original- De: Senad J [mailto:[EMAIL PROTECTED] Enviado el: martes, 23 de agosto de 2005 20:15 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] looking for failover ideas [EMAIL PROTECTED] wrote: I have had an idea of using two identical servers: Server A with IP x.x.x.a and server B with IP x.x.x.b. Server A is live while server B sits in the background monitoring server A. Server B rsync's asterisk config files daily with server A. In the event of server A going down, server B changes it's IP to x.x.x.a. The calls will obviously drop, but should register with server B. Comments??? We use it... works just fine as you describe it. Senad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] looking for failover ideas
If I use hearbeat I need a failover system for ISDN Lines, not? I waould like that if Server A crashes, Server B Control SIP Registration and ISDN Lines. Do you know about this? regards, srsergio -Mensaje original- De: Senad J [mailto:[EMAIL PROTECTED] Enviado el: martes, 23 de agosto de 2005 22:10 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] looking for failover ideas [EMAIL PROTECTED] wrote: How do you do monitoritng? How Server B knows that Servar A is down? I just do a rsync and MySQL Replication, but I try to do a C program that monitor Server. If you know how can I do this monitoring I will be pleasant with you. 1. use heartbeat for failover between A and B. Setup correctly failover is fully automatic. 2. u can use www.nagios.org or similar installed on C to monitor A and B Senad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Please, excuse me
Title: Mensaje I'm sorry for my holidays message, but I think it's too hard span me from list, don't you think? Could admin return to list, please? Regards, srsergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asunto_mensaje_entrante
Hasta el día 31 de Julio permaneceré de vacaciones, por lo que cualquier tipo de consulta, técnica o comercial debe redirigirla a [EMAIL PROTECTED] o a [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asunto_mensaje_entrante
Hasta el día 31 de Julio permaneceré de vacaciones, por lo que cualquier tipo de consulta, técnica o comercial debe redirigirla a [EMAIL PROTECTED] o a [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: asunto_mensaje_entrante
Hasta el día 31 de Julio permaneceré de vacaciones, por lo que cualquier tipo de consulta, técnica o comercial debe redirigirla a [EMAIL PROTECTED] o a [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sorry
Title: Mensaje I'm sorry for the several messages with holidays message. Regards, srsergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] outgoing call routing
Please, send us zapata.conf. It's possible that you don't have well configure zapata.conf, because in your trace you try to dial through g0 group and your Zap/4(I understand is your Zap connected to PSTN) must be into the 0 group. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jose Vicente Ortega Enviado el: domingo, 19 de junio de 2005 19:26 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] outgoing call routing I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip extensions and a regular phone connected to the box. All routing works fine from the regular phone connected to the box, whether its going to FWD, broadvoice or the PSTN. The problem I am experiencing comes from making calls from the sip phones. They get routed correctly to the sip and iax trunks but when making calls that are routed to the zap channel they ring the regular phone and do not get routed to the PSTN. Below are examples of the verbose from asterisk for calls from internal zap and internal sip channels to the PSTN. -- Starting simple switch on 'Zap/1-1' -- Executing Macro(Zap/1-1, dialout-trunk|1|817XX) in new stack -- Executing Macro(Zap/1-1, record-on|200) in new stack -- Executing AGI(Zap/1-1, set-timestamp.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/set-timestamp.agi -- AGI Script set-timestamp.agi completed, returning 0 -- Executing SetVar(Zap/1-1, CALLFILENAME=20050619-101044-200-817XX) in new stack -- Executing Monitor(Zap/1-1, wav|20050619-101044-200-817XX|mb) in new stack -- Executing GotoIf(Zap/1-1, 0?4) in new stack -- Executing SetCallerID(Zap/1-1, 817XX) in new stack -- Executing Goto(Zap/1-1, 6) in new stack -- Goto (macro-dialout-trunk,s,6) -- Executing SetCallerID(Zap/1-1, ) in new stack -- Executing SetGroup(Zap/1-1, OUT_1) in new stack -- Executing CheckGroup(Zap/1-1, ) in new stack -- Executing SetVar(Zap/1-1, DIAL_NUMBER=817XX) in new stack -- Executing SetVar(Zap/1-1, DIAL_TRUNK=1) in new stack -- Executing AGI(Zap/1-1, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Dial(Zap/1-1, ZAP/g0/817XX) in new stack -- Called g0/817XX -- Hungup 'Zap/4-1' -- Executing Macro(SIP/302-ffef, dialout-trunk|1|817XX) in new stack -- Executing Macro(SIP/302-ffef, record-on|302) in new stack -- Executing AGI(SIP/302-ffef, set-timestamp.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/set-timestamp.agi -- AGI Script set-timestamp.agi completed, returning 0 -- Executing SetVar(SIP/302-ffef, CALLFILENAME=20050619-101314-302-817XX) in new stack -- Executing Monitor(SIP/302-ffef, wav|20050619-101314-302-817XX|mb) in new stack -- Executing GotoIf(SIP/302-ffef, 1?4) in new stack -- Goto (macro-dialout-trunk,s,4) -- Executing Goto(SIP/302-ffef, 6) in new stack -- Goto (macro-dialout-trunk,s,6) -- Executing SetCallerID(SIP/302-ffef, ) in new stack -- Executing SetGroup(SIP/302-ffef, OUT_1) in new stack -- Executing CheckGroup(SIP/302-ffef, ) in new stack -- Executing SetVar(SIP/302-ffef, DIAL_NUMBER=817XX) in new stack -- Executing SetVar(SIP/302-ffef, DIAL_TRUNK=1) in new stack -- Executing AGI(SIP/302-ffef, fixlocalprefix) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf -- AGI Script fixlocalprefix completed, returning 0 -- Executing Dial(SIP/302-ffef, ZAP/g0/817XX) in new stack -- Called g0/817XX -- Zap/1-1 is ringing -- Hungup 'Zap/1-1' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] outgoing call routing
Try with this zapata.conf
[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
axdetect=both
group=1
immediate=no
signalling=fxo_ks
Context=outgoing
channel = 1
group=2
signalling=fxs_ks
immediate=yes
context=from-pstn
channel = 4
And make outgoing call in the way
exten=_.,1,Dial(Zap/g2/${EXTEN})
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jose
Vicente Ortega
Enviado el: lunes, 20 de junio de 2005 20:58
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: RE: [Asterisk-Users] outgoing call routing
Here is is.
; Zapata telephony interface
;
; Configuration file
[trunkgroups]
[channels]
language=en
context=from-pstn
signalling=fxs_ks
rxwink=300 ; Atlas seems to use long (250ms) winks
;
; Whether or not to do distinctive ring detection on FXO lines ;
;usedistinctiveringdetection=yes
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=800
rxgain=0.0
txgain=0.0
group=0
callgroup=1
pickupgroup=1
immediate=no
;faxdetect=both
faxdetect=incoming
;faxdetect=outgoing
;faxdetect=no
;Include AMP configs
#include zapata_additional.conf
;Include genzaptelconf configs
#include zapata-auto.conf
; Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit ;
Zaptel Channels Configurations (zapata.conf) ; ; This is not intended to
be a complete zapata.conf. Rather, it is intended ; to be #include-d by
/etc/zapata.conf that will include the global settings ;
; Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 signalling=fxo_ks ;
Note: this is an extension. Create a ZAP extension in AMP for Channel 1
channel = 1
; channel 2, WCTDM, inactive.
; channel 3, WCTDM, inactive.
signalling=fxs_ks
; Note: this is a trunk. Create a ZAP trunk in AMP for Channel 4
context=from-pstn channel = 4
At 10:10 AM 6/20/2005, Sergio Serrano wrote:
Please,
send us zapata.conf. It's possible that you don't have well
configure zapata.conf, because in your trace you try to dial through g0
group and your Zap/4(I understand is your Zap connected to PSTN) must
be into the 0 group.
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Jose
Vicente Ortega Enviado el: domingo, 19 de junio de 2005 19:26
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] outgoing call routing
I have a Asterisk @home ver 1.0 running with a TDMB11 card. Several sip
extensions and a regular phone connected to the box. All routing works
fine from the regular phone connected to the box, whether its going to
FWD, broadvoice or the PSTN. The problem I am experiencing comes from
making calls from the sip phones. They get routed correctly to the sip
and iax trunks but when making calls that are routed to the zap channel
they ring
the regular phone and do not get routed to the PSTN.
Below are examples of the verbose from asterisk for calls from internal
zap and internal sip channels to the PSTN.
-- Starting simple switch on 'Zap/1-1'
-- Executing Macro(Zap/1-1, dialout-trunk|1|817XX) in new
stack
-- Executing Macro(Zap/1-1, record-on|200) in new stack
-- Executing AGI(Zap/1-1, set-timestamp.agi) in new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/set-timestamp.agi
-- AGI Script set-timestamp.agi completed, returning 0
-- Executing SetVar(Zap/1-1,
CALLFILENAME=20050619-101044-200-817XX) in new stack
-- Executing Monitor(Zap/1-1,
wav|20050619-101044-200-817XX|mb) in new stack
-- Executing GotoIf(Zap/1-1, 0?4) in new stack
-- Executing SetCallerID(Zap/1-1, 817XX) in new stack
-- Executing Goto(Zap/1-1, 6) in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing SetCallerID(Zap/1-1, ) in new stack
-- Executing SetGroup(Zap/1-1, OUT_1) in new stack
-- Executing CheckGroup(Zap/1-1, ) in new stack
-- Executing SetVar(Zap/1-1, DIAL_NUMBER=817XX) in new
stack
-- Executing SetVar(Zap/1-1, DIAL_TRUNK=1) in new stack
-- Executing AGI(Zap/1-1, fixlocalprefix) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/fixlocalprefix
fixlocalprefix: Could not parse /etc/asterisk/localprefixes.conf
-- AGI Script fixlocalprefix completed, returning 0
-- Executing Dial(Zap/1-1, ZAP/g0/817XX) in new stack
-- Called g0/817XX
-- Hungup 'Zap/4-1'
-- Executing Macro(SIP/302-ffef, dialout-trunk|1|817XX) in
new stack
-- Executing Macro(SIP/302-ffef, record-on|302) in new stack
-- Executing AGI(SIP/302-ffef, set-timestamp.agi) in new
stack
RE: [Asterisk-Users] ztcfg server crash
Before change OS try to do next steps: first, stop asterisk. Second, you must do ztcfg -s to shutdown span. Unload modules, load modules if you need and do ztcfg -vv again. Start asterisk Regards Srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jason Walker Enviado el: martes, 14 de junio de 2005 6:07 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] ztcfg server crash I tried to get * stable on a 2.6xxx kernel for about 2 weeks. Then tried it out on a FC1 2.4.xxx kernel and found none of the issues. I am sure others have had success with 2.4.xxx, but I gave up;) BTW - I was using a TE110P and then a TE405P card for the zaptel install. Both were setup as T1s not E1s. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ztcfg server crash I am running Debian Sarge with a custom 2.6.11 kernel. I'll try building another kernel and recompiling the zaptel stuff. Jason Walker wrote: What OS/distro are you running? I experienced the same on Gentoo with the 2.6.xxx kernel. Switched to FC1 (2.4.xxx kernel) with the 1.0.7 CVS and have not had any issues. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rod Bacon Sent: Monday, June 13, 2005 7:31 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ztcfg server crash I was wondering if anyone had experienced the following with asterisk stable. After a period of time (can vary), If I stop asterisk and try to run ztcfg -v to reinitialise my quad e1 card, the server will lock up. Sometimes it's a complete lockup, where it won't even return pings, other times it seems to be partially screwed. ___ 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 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.9 - Release Date: 6/11/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE410P and Siemens HIPATH 3750
Title: Mensaje Hi all, I have to interconnect Asterisk with a Siemens HIPATH 3750. In siemens we can configure ECMA-QSIG Master, ISO-QSIG Master,Point to Point link withCRC4 and Point to Point link withouthCRC4): Siemens has BNC connector. I use a balun with BNC and RH45 connectro. I try with basic RJ45 cable and with crossover RJ45(1-4, 2-5) but I can only see yellow led in TE410P. I have configured siemens like Point to Point with and withouth CRC4 and Asterisk with ccs,hdb3 ( with CRC4 and withouth CRC4), with pri_net and pri_cpe and signalling=euroisdn Anyone has experience with this scenario? Regards, srsergio ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk
Fantastic!! Thanks to your works regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Areski Enviado el: miércoles, 26 de enero de 2005 18:05 Para: Asterisk-Users Mailing-list Asunto: [Asterisk-Users] ANNOUNCEMENT : NEW CallingCard Application forAsterisk Hello everyone, If you want to know why I am so tired today :D Check this CallingCard Solution : http://areski.net/areskicc-doc/ Just finish it yesterday night! Briefly, AreskiCC is an AGI script and PHP-Web application which greatly handle the complete CallingCard System. FEATURES - AGI : * Authenticate with the use of a Cardnumber the Cardnumber can also be defined as accountcode into sip.conf, iax.conf, etc.. * take care of multiple calls using the same Cardnumber * Caller gets informed about his credit Announce the remaining credit * Caller is requested to enter a destination number * Announce the maximal call time for the given destination number It calculates the remaining duration of the actual call (based on tariffrate tables), informs the caller about this and sets a timeout * Interupt the call if the card balance gets zero Warn the caller about the call interupt 60 30 seconds before the call gets interupted * It connects the Caller to the destination through the configured trunk note : different trunks can be configured and associated by prefix * After disconnecting the call AGI updates the credit and stores the concerning Call-Detail-Records with CallingPartyNumber, CalledPartyNumber, CallSetupTime, Duration, Charge and the remaining credit FEATURES - WEB INTERFACE: * CARD/CUSTOMERS * List customers * Refill customer * CARD/CUSTOMERS * List customers/cards * Refill customer/card * Create customer/card * Generate customers/cards * BILLING * View money situation * View Payment * Add new Payment * RATECARD * List Tariffplan * Create new Tariffplan * Define Tariffplan * TRUNK * List Trunk * Add Trunk * CALL REPORT - BALANCE Last note : It's distributed under GNU GPL Licence. I hope there will have a big interest for the soft, I am waiting your feedbacks... Regards, /Areski -_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_-_ Belaïd Arezqui www.areski.net E-mail : areski [EMAIL PROTECTED] gmail (.dot.) 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.4 - Release Date: 25/01/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.300 / Virus Database: 265.7.4 - Release Date: 25/01/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Different EXT lines for different users?
You can try to set one context for each extension.
regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Alen Salamun
Enviado el: martes, 25 de enero de 2005 16:04
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] Different EXT lines for different users?
Hello!
I would like to make asterisk to use different ISDN external lines
dependant on which internal user makes the call. Right now I have
(12345678 represents my MSN):
[pstn] ; ISDN to PSTN
exten = _0.,1,Dial(CAPI/12345678:b${EXTEN:1})
exten = _0.,2,Hangup
This ofcourse means that whenever someone call's out to number 0this
call goes to outside line 12345678.
Now i would like asterisk to behave like that:
When user 100 calls go outside on line 12345670
When user 101 calls go outside on line 12345671
...
How can I do that?
Thank you,
Alen
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RE:[Asterisk-Users] Problems with loading TE110 module
Title: Mensaje Have you solve your Problem?, I have same problem after with recompile kernel. Regards, srsergio Monday, December 20, 2004, 12:44:36 PM, Matt wrote:MR Have you tried doing a modprobe -r first?Before reboot I did rmmod wcte11xp. If you mean that.now modprobe -r wcte11xp doesn't do anything, still can't load themodule. :(Tamas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE110P doesn't appear in /proc/zaptel
Title: Mensaje Hi all, I have installed a TE110P in a BOX but when I load zaptel module I can't see any device in /proc/zaptel. And led of the card is green. My zaptel.conf is the next: span=1,1,0,ccs,hdb3,crc4bchan=1-15,17-31dchan=16loadzone=esdefaultzone=es and cat /proc/pci throguh next: PCI devices found: Bus 0, device 0, function 0: Host bridge: Intel Corp. 82845 845 (Brookdale) Chipset Host Bridge (rev 4). Prefetchable 32 bit memory at 0xd000 [0xd7ff]. Bus 0, device 1, function 0: PCI bridge: Intel Corp. 82845 845 (Brookdale) Chipset AGP Bridge (rev 4). Master Capable. Latency=64. Min Gnt=14. Bus 0, device 30, function 0: PCI bridge: Intel Corp. 82801BA/CA/DB PCI Bridge (rev 5). Master Capable. No bursts. Min Gnt=6. Bus 0, device 31, function 0: ISA bridge: Intel Corp. 82801BA ISA Bridge (LPC) (rev 5). Bus 0, device 31, function 1: IDE interface: Intel Corp. 82801BA IDE U100 (rev 5). I/O at 0xf000 [0xf00f]. Bus 0, device 31, function 2: USB Controller: Intel Corp. 82801BA/BAM USB (Hub #1) (rev 5). IRQ 10. I/O at 0xd000 [0xd01f]. Bus 0, device 31, function 3: SMBus: Intel Corp. 82801BA/BAM SMBus (rev 5). IRQ 9. I/O at 0x500 [0x50f]. Bus 0, device 31, function 4: USB Controller: Intel Corp. 82801BA/BAM USB (Hub #2) (rev 5). IRQ 12. I/O at 0xd800 [0xd81f]. Bus 0, device 31, function 5: Multimedia audio controller: Intel Corp. 82801BA/BAM AC'97 Audio (rev 5). IRQ 9. I/O at 0xdc00 [0xdcff]. I/O at 0xe000 [0xe03f]. Bus 1, device 0, function 0: VGA compatible controller: ATI Technologies Inc Radeon VE QY (rev 0). IRQ 5. Master Capable. Latency=32. Min Gnt=8. Prefetchable 32 bit memory at 0xd800 [0xdfff]. I/O at 0xc000 [0xc0ff]. Non-prefetchable 32 bit memory at 0xe100 [0xe100]. Bus 2, device 1, function 0: Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8029(AS) (rev 0). IRQ 10. I/O at 0xa000 [0xa01f]. Bus 2, device 5, function 0: Network controller: Tiger Jet Network Inc. Model 300 128k (rev 0). IRQ 12. Master Capable. Latency=32. Min Gnt=1.Max Lat=128. I/O at 0xa400 [0xa4ff]. Non-prefetchable 32 bit memory at 0xe300 [0xe3000fff]. Any idea? regards, srsergio -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel
Yes I'm sure that I load wcte11xp. When I do wcte11xp I obtain next error: /lib/modules/2.4.20/misc/wcte11xp.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20/misc/wcte11xp.o: insmod /lib/modules/2.4.20/misc/wcte11xp.o failed /lib/modules/2.4.20/misc/wcte11xp.o: insmod wcte11xp failed Any idea? Srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: miércoles, 29 de diciembre de 2004 18:45 Para: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel On Wed, Dec 29, 2004 at 11:59:58AM +0100, Sergio Serrano wrote: Hi all, I have installed a TE110P in a BOX but when I load zaptel module I can't see any device in /proc/zaptel. And led of the card is green. From /proc/pci, it looks like you pci bus saw the card. Are you sure that you loaded the wcte11xp module? Matthew Fredrickson ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel
This card is the only card in the system, and other thing, led of the card is fixed green. In dmesg I obtain nothing. Regards, -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de [EMAIL PROTECTED] Enviado el: miércoles, 29 de diciembre de 2004 22:31 Para: Asterisk Users Mailing List - Non-Commercial Discussion; [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] TE110P doesn't appear in /proc/zaptel On Wed, Dec 29, 2004 at 09:17:45PM +0100, Sergio Serrano wrote: Yes I'm sure that I load wcte11xp. When I do wcte11xp I obtain next error: /lib/modules/2.4.20/misc/wcte11xp.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg /lib/modules/2.4.20/misc/wcte11xp.o: insmod /lib/modules/2.4.20/misc/wcte11xp.o failed /lib/modules/2.4.20/misc/wcte11xp.o: insmod wcte11xp failed Is that the only digium card you have in that machine? If not, that device I saw on the PCI bus could be another card. What does it say in dmesg about it when you try to load it (as per instructions received upon failure to load module)? Matthew Fredrickson ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.6 - Release Date: 28/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Link an Asterisk Box with a PBX (E1 connection)
Try exten= _X.,1, Dial(Zap/g2) If your 2nd TE110 has defined a group. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jeremy SALMON Enviado el: miércoles, 22 de diciembre de 2004 15:01 Para: asterisk-users@lists.digium.com Asunto: [Asterisk-Users] Link an Asterisk Box with a PBX (E1 connection) Hi, I have to link an Asterisk Box with a PBX Matra 6501. System look like this : E1--Te110P Asterisk Te110P-E1Matra 6501-Phones | | Ip Phones Incoming call from E1 will enter on asterisk, if incoming number is _800n then go to IP phones. In this case no problem. But if it's an another call i want to return call to my old MATRA 6501. I don't known in my dial plan how to send call to an outgoing E1 channel... Thanks ! Jeremy SALMON ___ 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 -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.3 - Release Date: 21/12/2004 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.6.3 - Release Date: 21/12/2004 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and HylaFax
Title: Mensaje Hi all, again I try configure Hylafax with asterisk. I would like configure Asterisk in the next way: 1)An incoming fax go into through X100P 2)Asterisk detects Fax and redirect fax to Hylafax Is it possible? Any idea woluld be great idea? regards, srsergio -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.296 / Virus Database: 265.5.4 - Release Date: 15/12/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Error when install E100P
Please, could you send us cat /proc/pci?. Could you compile libpri?
Regards,
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Ning Zhou
Enviado el: martes, 23 de noviembre de 2004 16:10
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Error when install E100P
Hi, all
I am trying to install E100P card, the 'modprobe zaptel' is ok, but when I
did 'modprobe wct1xxp', I got such error, so can not load the driver for the
card.
/lib/modules/2.4.20-8/misc/wct1xxp.o: init_module: No such device
Hint: insmod errors can be caused by incorrect module parameters, including
invalid IO or IRQ parameters.
You may find more information in syslog or the output from dmesg
/lib/modules/2.4.20-8/misc/wct1xxp.o: insmod
/lib/modules/2.4.20-8/misc/wct1xxp.o failed
/lib/modules/2.4.20-8/misc/wct1xxp.o: insmod wct1xxp failed
Could anyone please give me some hints?
Thank you so mmuch!!
my configuration file is like this:
Zapata.conf:
[channels]
context=default
group=1
callgroup=1
pickupgroup=1
switchtype=euroisdn
signalling=pri_cpe
context=default
channel = 1-15,17-31
zaptel.conf:
span=1,1,0,ccs,hdb3
bchan=1-15,17-31
dchan=16
loadzone = no
defaultzone = no
extensions.conf
[default]
exten = _XX,1,Dial,Zap/g1/${EXTEN}
exten = s,1,Wait(1)
exten = s,2,Answer
exten = s,3,Hangup ___
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RE: [Asterisk-Users] hangup()???
Hi, this call is from? Zap channel, Capi channel or other channel? It is possible that you don't detect well hangup from incoming channel. Regards. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Altus Snyman Enviado el: lunes, 22 de noviembre de 2004 9:48 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: [Asterisk-Users] hangup()??? Good day all I want to tell asterisk that it should hangup a channel in a certain step For example: exten = s,5,Dial(SIP/302,25) exten = s,6,Hangup exten = s,7,Hangup(SIP/302) What happens is that if someone calls into the pbx and hangs up before it gets answered it still keeps on ringing on the internal side and if you pick up there is nothing Please Help ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.797 / Virus Database: 541 - Release Date: 15/11/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.797 / Virus Database: 541 - Release Date: 15/11/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI 0x3301 Problem
Hi all, I have a PBX working for a year with an Eicon Diva Server 4BRI. One day it was a storm and nothing occurs, but after a a few days I can't send and receive any calls. I have connected TEIs to Asterisk and other PBX and when I try to dial, I hear correct tone two times, but then line hangup, with the next trace: -- Executing ChanIsAvail(SIP/824-b0cd,CAPI/971844367CAPI/971846015CAPI/971846034CAP I/971846036CAPI/971846094CAPI/971846141CAPI/971846142CAPI/971846143CAPI /971846146CAPI/971846147CAPI/971846148) in new stack -- data = 971844367 -- capi request omsn = 971844367 == found capi with omsn = 971844367 -- CAPI Hangingup -- Executing Dial(SIP/824-b0cd, CAPI/@971844367:687754642|17) in new stack -- data = @971844367:687754642 -- capi request omsn = @971844367 == found capi with omsn = 971844367 == CAPI Call CAPI[contr1/971844367]/1 == CAPI Call CAPI[contr1/971844367]/1 -- creating pipe for PLCI=-1 sent CONNECT_REQ MN =0x7 -- Called @971844367:687754642 -- CONNECT_CONF ID=001 #0x0007 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 == received CONNECT_CONF PLCI = 0x301 INFO = 0 == DISCONNECT_IND PLCI=0x301 REASON=0x3301 sent DISCONNECT_RESP PLCI=0x301 -- CAPI Hangingup -- removed pipe for PLCI = 0x301 I obtain 0x3301 error message(Protocol error layer 1 (broken line or B-channel removed by signalling protocol)). I'm in Spain. I think that problem could be something with: 1) Eicon Diva Server is bad: I have tested this card in other machine and other lines and it works well. 2) MotherBoard is bad: this is a probable error but I'm not sure because this motherboard has worked during a year. 3) Wire from Card to TEI is broken, But I don't think this because I obtain a valida MSN to put this call. 4) Telecom Operator has modified something in these lines. I start Eicon diva card with divactrl load -c 1 -f ETSI -u 2. And my capi.conf is the next: [general] nationalprefix=0 internationalprefix=00 rxgain=0.5 txgain=0.5 [interfaces] msn=971844367,971846015,971846034,971846036,971846094,971846141,971846142,97 1846143,971846146,971846147,971846148 isdnmode=multipoint incomingmsn=* mode=immediate controller=1,2,3,4 ;softdtmf= ;accountcode= context=default callgroup=1 ;echosquelch= echocancel=1 echotail=64 devices=8 Any idea? Regards, srsergio --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.796 / Virus Database: 540 - Release Date: 13/11/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_icd compile problem
Hi all, I try to compile app_icd to test it but I can't compile it. I have installed asterisk 1.0.2 and I download ICD and put files into /usr/src/asterisk/apps/icd directory. I think that make.conf in icd directory is ok but when I try to compile icd I obtain next error: === Compile: /usr/src/asterisk/apps/icd/app_icd.c (app_icd.o) app_icd.c:66: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) make: *** [app_icd.o] Error 1 If I change line 66 from static ast_mutex_t icdlock = AST_MUTEX_INITIALIZER; To static ast_mutex_t icdlock = AST_MUTEX_DEFINE_STATIC; I obtain next error: === Compile: /usr/src/asterisk/apps/icd/app_icd.c (app_icd.o) app_icd.c:67: `AST_MUTEX_DEFINE_STATIC' undeclared here (not in a function) make: *** [app_icd.o] Error 1 My make.conf is the next: # what compiler CC=gcc # uncomment this if your asterisk is version 1.0 CFLAGS += -DAST_POST_10 # where is the asterisk source tree ASTSRC = /usr/src/asterisk # # copy these from the asterisk top level Makefile INSTALL_PREFIX= ASTLIBDIR=$(INSTALL_PREFIX)/usr/lib/asterisk ASTVARLIBDIR=$(INSTALL_PREFIX)/var/lib/asterisk ASTETCDIR=$(INSTALL_PREFIX)/etc/asterisk ASTSPOOLDIR=$(INSTALL_PREFIX)/var/spool/asterisk ASTLOGDIR=$(INSTALL_PREFIX)/var/log/asterisk ASTHEADERDIR=$(INSTALL_PREFIX)/usr/include/asterisk ASTCONFPATH=$(ASTETCDIR)/asterisk.conf ASTBINDIR=$(INSTALL_PREFIX)/usr/bin ASTSBINDIR=$(INSTALL_PREFIX)/usr/sbin ASTVARRUNDIR=$(INSTALL_PREFIX)/var/run MODULES_DIR=$(ASTLIBDIR)/modules AGI_DIR=$(ASTVARLIBDIR)/agi-bin INCLUDES = -I$(ASTSRC) -I$(ASTSRC)/include -I. -I/usr/src/asterisk/include CFLAGS += $(INCLUDES) CFLAGS += -DINSTALL_PREFIX=\$(INSTALL_PREFIX)\ -DASTETCDIR=\$(ASTETCDIR)\ -DASTLIBDIR=\$(ASTLIBDIR)\ CFLAGS += -DASTVARLIBDIR=\$(ASTVARLIBDIR)\ -DASTVARRUNDIR=\$(ASTVARRUNDIR)\ -DASTSPOOLDIR=\$(ASTSPOOLDIR)\ -DASTLOGDIR=\$(ASTLOGDIR)\ CFLAGS += -DASTCONFPATH=\$(ASTCONFPATH)\ -DASTMODDIR=\$(MODULES_DIR)\ -DASTAGIDIR=\$(AGI_DIR)\ -D_GNU_SOURCE CFLAGS += -O0 -g CFLAGS += -Wall #CFLAGS += -DNDEBUG LDFLAGS = Anyone could help me? Best reagards, srsergio --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.797 / Virus Database: 541 - Release Date: 15/11/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Software SIP Phones
Hi, Voicemoil capabilities are in Asterisk. You can use Asterisk voicemail from any SIP Software. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Ashling O'Driscoll Enviado el: miércoles, 17 de noviembre de 2004 18:28 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Software SIP Phones Hi, I am also interested in what softphones other asterisk people are using. I am using xlite but that doesnt seem to have any voicemail capabilities (correct me if im wrong). You have to purchase xpro for that. Does anyone have any suggestions?. Apologies to the person who first sent this mail-I don't mean to be rude and butt in on your thread. Aisling. Original Message From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Software SIP Phones Date: Wed, 17 Nov 2004 12:15:55 -0500 On Wednesday 17 November 2004 12:15, Diego Aguirre wrote: Hi, I am using X-Lite with Wine! wow! I triied to get it working under wine but it was a no go. I'm very familiar with Wine, we run a few apps under wine here. Coudl you share your config or somet tips to help me get it running? Are you using stock wine, or a flavour from Codeweavers or Transgaming? Thanks, Pete ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.796 / Virus Database: 540 - Release Date: 13/11/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.796 / Virus Database: 540 - Release Date: 13/11/2004 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Chan zap not loaded(ast_pickup_call)
Hi all, I'm sorry, but I'm stupid because I haven't load res_parking.so. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Enviado el: viernes, 10 de septiembre de 2004 9:35 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: [Asterisk-Users] Chan zap not loaded(ast_pickup_call) Hi all I have installed an E100P. I have loaded zaptel and wct1xxp. My zaptel.conf is the next: span=1,1,0,ccs,hdb3,crc4,yellow bchan=1-15,17-31 dchan=16 loadzone=es defaultzone=es My zapata.conf is the next: [channels] switchtype = euroisdn language=es signalling = pri_cpe pridialplan = local prilocaldialplan = local echocancel = yes context = default group=1 channel = 1-15,17-31 When I start asterisk it says: [chan_zap.so]Sep 10 09:22:09 WARNING[1076253312]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_zap.so: undefined symbol: ast_pickup_call Sep 10 09:22:09 WARNING[1076253312]: loader.c:374 load_modules: Loading module chan_zap.so failed! Any idea? Regards, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Swissvoice de-register
Hi all, I'm trying to configure a swissvoice IP10S but after a minutes this phones appears like UKNOWN in sip show peers and it is unaccesible. This phone can make call but it can't receive calls. Any idea? Regards, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP Swissvoice de-register
SIP version IP10 SP v0.0.1 (Build 5) Regards, -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Florian Overkamp Enviado el: lunes, 06 de septiembre de 2004 13:42 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] SIP Swissvoice de-register Hi, -Original Message- I'm trying to configure a swissvoice IP10S but after a minutes this phones appears like UKNOWN in sip show peers and it is unaccesible. This phone can make call but it can't receive calls. What firmware are you running with ? Bog-standard IP10's come with H323 or MGCP. SIP is still in early stages (but coming soon). Florian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk SIP between two networks
Hi all, I have any information more. I have configured sip.conf with bindaddr=0.0.0.0. I have observed traffic with tethereal and I have seen the next. First REGISTER goes out from my asterisk to my SIP Provider. My SIP Provider respond to my with a 401 Unauthorized meesage, but Asterisk doesn't read this message and try to resend first REGISTER. In the second localnet(in previous message) there is a Hicom Siemens with a HG1500 interface with Intel propietary protocol, but without SIP protocol. I have noticed that this interface goes down when I start asterisk. Has anyone had same problem? Could anyone help me with this problem? Best regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Enviado el: miércoles, 01 de septiembre de 2004 0:46 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: [Asterisk-Users] Asterisk SIP between two networks Hi all, I have next configuration: SIP Provider---ADSL router---localnet 192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones first localnet 192.168.20.0 second localnet 172.28.240.0 in second localnet we have softphone and the first localnet is connected to ADSL router to connect to our SIP provider. The problem is the next: if I put in [general] section in sip.conf, bindaddr=0.0.0.0, I can't register in my SIP provider. If I put 192.168.20.10 in bindaddr I can register in my SIP provider but softphones can't register into asterisk. I 'm using asterisk RC1. Any idea? srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk SIP between two networks
Hi, I have any information more, I have noticed that asterisk receives 401 Unauthorized message but If I do a sip denbug I can read next: Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK6a231e4d From: sip:[EMAIL PROTECTED];tag=as0f12fef4 To: sip:[EMAIL PROTECTED];tag=84448f3c7053227cca70775302748de3.a036 Call-ID: [EMAIL PROTECTED]: 122 REGISTER WWW-Authenticate: Digest realm=voztele.com, nonce=4135f73170422e2f6d1bd01c77eca25260de8f4b Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 If you can see next line Call-ID: [EMAIL PROTECTED]: 122 REGISTER, Cseq field is incompleted. How it is possible?. If I put asterisk in only one localnet with bindaddr=192.168.20.10 I haven't the problem, but If I put asterisk in two localnet with bindaddr=0.0.0.0 I obtain this command. Any idea? Please, I need help Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Enviado el: miércoles, 01 de septiembre de 2004 12:51 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks Hi all, I have any information more. I have configured sip.conf with bindaddr=0.0.0.0. I have observed traffic with tethereal and I have seen the next. First REGISTER goes out from my asterisk to my SIP Provider. My SIP Provider respond to my with a 401 Unauthorized meesage, but Asterisk doesn't read this message and try to resend first REGISTER. In the second localnet(in previous message) there is a Hicom Siemens with a HG1500 interface with Intel propietary protocol, but without SIP protocol. I have noticed that this interface goes down when I start asterisk. Has anyone had same problem? Could anyone help me with this problem? Best regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Enviado el: miércoles, 01 de septiembre de 2004 0:46 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: [Asterisk-Users] Asterisk SIP between two networks Hi all, I have next configuration: SIP Provider---ADSL router---localnet 192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones first localnet 192.168.20.0 second localnet 172.28.240.0 in second localnet we have softphone and the first localnet is connected to ADSL router to connect to our SIP provider. The problem is the next: if I put in [general] section in sip.conf, bindaddr=0.0.0.0, I can't register in my SIP provider. If I put 192.168.20.10 in bindaddr I can register in my SIP provider but softphones can't register into asterisk. I 'm using asterisk RC1. Any idea? srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk SIP between two networks
I just use localnet parameter in next way: localnet=192.168.20.0/255.255.255.0 localnet=172.28.240.0/255.255.240.0 Any idea more? Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Kevin Walsh Enviado el: miércoles, 01 de septiembre de 2004 19:16 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks Sergio Serrano [EMAIL PROTECTED] wrote: SIP Provider---ADSL router---localnet 192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones first localnet 192.168.20.0 second localnet 172.28.240.0 in second localnet we have softphone and the first localnet is connected to ADSL router to connect to our SIP provider. if I put in [general] section in sip.conf, bindaddr=0.0.0.0, I can't register in my SIP provider. If I put 192.168.20.10 in bindaddr I can register in my SIP provider but softphones can't register into asterisk. I 'm using asterisk RC1. You probably need to use the localnet setting in sip.conf. See here for more sip.conf-related information: http://www.voip-info.org/wiki-Asterisk+config+sip.conf On the other hand, you could use an IAX2 provider and side-step the issue altogether. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk SIP between two networks
Hi all, I'm desperate, if I put bindaddr=192.168.20.10, I obtain the next: Sip read: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK2174f136 From: sip:[EMAIL PROTECTED];tag=as05db6abc To: sip:[EMAIL PROTECTED];tag=84448f3c7053227cca70775302748de3.a866 Call-ID: [EMAIL PROTECTED] CSeq: 102 REGISTER WWW-Authenticate: Digest realm=voztele.com, nonce=41365c4cf9c69cc73a429f27813652ded65fc483 Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 But If i put bindaddr=0.0.0.0, I obtain yhe next: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.20.10:5060;branch=z9hG4bK54679e05 From: sip:[EMAIL PROTECTED];tag=as294baf04 To: sip:[EMAIL PROTECTED];tag=84448f3c7053227cca70775302748de3.e5c8 Call-ID: [EMAIL PROTECTED]: 103 REGISTER WWW-Authenticate: Digest realm=voztele.com, nonce=41365cfc1947f24b5cd03bb5bca062540243dc39 Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux)) Content-Length: 0 It is a bug? Why if I put bindaddr=0.0.0.0 packet received by asterisk is broken? Could anyone help me? Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Enviado el: jueves, 02 de septiembre de 2004 0:28 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks I just use localnet parameter in next way: localnet=192.168.20.0/255.255.255.0 localnet=172.28.240.0/255.255.240.0 Any idea more? Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Kevin Walsh Enviado el: miércoles, 01 de septiembre de 2004 19:16 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: RE: [Asterisk-Users] Asterisk SIP between two networks Sergio Serrano [EMAIL PROTECTED] wrote: SIP Provider---ADSL router---localnet 192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones first localnet 192.168.20.0 second localnet 172.28.240.0 in second localnet we have softphone and the first localnet is connected to ADSL router to connect to our SIP provider. if I put in [general] section in sip.conf, bindaddr=0.0.0.0, I can't register in my SIP provider. If I put 192.168.20.10 in bindaddr I can register in my SIP provider but softphones can't register into asterisk. I 'm using asterisk RC1. You probably need to use the localnet setting in sip.conf. See here for more sip.conf-related information: http://www.voip-info.org/wiki-Asterisk+config+sip.conf On the other hand, you could use an IAX2 provider and side-step the issue altogether. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP between two networks
Hi all, I have next configuration: SIP Provider---ADSL router---localnet 192.168.20.0---ASTERISK---localnet 172.24.240.0---softphones first localnet 192.168.20.0 second localnet 172.28.240.0 in second localnet we have softphone and the first localnet is connected to ADSL router to connect to our SIP provider. The problem is the next: if I put in [general] section in sip.conf, bindaddr=0.0.0.0, I can't register in my SIP provider. If I put 192.168.20.10 in bindaddr I can register in my SIP provider but softphones can't register into asterisk. I 'm using asterisk RC1. Any idea? srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone
Title: Mensaje Push send after you number, srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de James DuttonEnviado el: jueves, 05 de agosto de 2004 12:28Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Call Transfer Problems with Grandstream Budgetone 100 Phone I have two Grandstream Budgetone 100 phones connected to my local asterisk server. I am able to receive incoming calls, and place outgoing calls, but have two problems... 1) I cannot transfer calls between the two phones. Pressing transfer takes me to a dial tone, I key in the internal number then press # or transfer, and the original call is cut off and the other internal phone does not ring. 2) I cannot hear an outgoing ringing tone when placing the call. I would be extremely grateful to anyone out who has experience of these phones and can help. Regards James Dutton
RE: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn
Hi, in Spain that process is correct. If you setup a communication between a caller and a called, if called phone hangs, in caller side hear a silence, but is a correct process. It's is due to in the called side you can hangup a phone and pickup other phone without lost communication. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Walter Klomp Enviado el: jueves, 29 de julio de 2004 16:44 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Zaptel doesn't see remote hangup ? euro-isdn Hi Just received my spanky new TE405P today to replace my Cisco gateway... After much fiddling (I forgot to switch it to E1) I got it to work and everything seems to work perfectly on our ISDN PRI. If I dial-in from the PSTN to a SIP phone, the call goes through and if I hangup either the SIP phone or the remote end, the call gets disconnected and destroyed However, if I dial-in from the SIP phone to my PSTN and then hang up my PSTN phone, the call does not get disconnected. My SIP phone goes quiet but doesn't disconnect. If I a few seconds later pick up the PSTN phone again, the connection is still there. Only if I hangup the SIP phone, the call gets destroyed. It seems that Zap doesn't see the remote hangup... Here is my Zaptel config and my Zapata config. I presume the extensions config etc are OK as my call-flow never changed and things were working fine with my AS5300. Am I missing something ? How do I debug the Zap channels ? Cheers, Walter Klomp /etc/zaptel.conf span=1,1,0,ccs,hdb3,crc4 # This is the line in question... span=2,1,0,ccs,hdb3,crc4 # not used yet span=3,0,0,ccs,hdb3,crc4 # not used yet span=4,0,0,ccs,hdb3,crc4 # not used yet # Span 1 bchan=1-15 dchan=16 bchan=17-31 # Span 2 bchan=32-46 dchan=47 bchan=48-62 # Span 3 bchan=63-77 dchan=78 bchan=79-93 # Span 4 bchan=94-108 dchan=109 bchan=110-124 alaw=1-124 loadzone=uk defaultzone=uk /etc/asterisk/zapata.conf [channels] context=default switchtype=euroisdn signalling=pri_cpe usecallerid=yes hidecallerid=no callwaiting=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes echocancel=yes rxgain=0.0 txgain=0.0 immediate=no ; Channels inherit configuration above them ; Span 1 group=1 signalling=pri_cpe channel = 1-15 channel = 17-31 ; Span 2 group=2 signalling=pri_cpe channel = 32-46 channel = 48-62 ; Span 3 group=3 signalling=pri_cpe channel = 63-77 channel = 79-93 ; Span 4 group=4 signalling=pri_cpe channel = 94-108 channel = 110-124 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] debian install zaptel
Title: Mensaje It's more easy download tarball and compile it. srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de YanEnviado el: jueves, 22 de julio de 2004 13:31Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] debian install zaptel Hi: Did anyone use apt-get install zaptel successfully? After apt-get instal zaptel, use "modprobe zaptel", get a "FATAL modul zaptel notfound". Thanks. Yan
RE: [Asterisk-Users] Problems with festival
Title: Mensaje I have the same problem.I'm usinr asterisk-1.0-RC1. Anyone could help us? regards, srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Dan FernandezEnviado el: viernes, 16 de julio de 2004 20:42Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Problems with festival I cannot get Festival to work with asterisk. I have the following: exten = 555,1,Answerexten = 555,2,Festival(mary has a little lamb)exten = 555,3,Hangup I get the following from asterisk: "Festival returned ER" and the festival logs shows the following: client(1) Fri Jul 16 15:35:54 2004 : disconnectedclient(2) Fri Jul 16 15:40:26 2004 : accepted from localhost Festival seems to be running fine. For example if I do: echo this is a test | --tts --language english it works just fine I'm starting festival from the script festival_server and the logs shows no errors. I had to rename the festival directory to festival-1.4.3 to apply the patch Any ideas what can the problem be?
RE: [Asterisk-Users] Chan_Capi 0.3.4a error
Try to compile with lastest CVS srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Martin List-Petersen Enviado el: jueves, 15 de julio de 2004 1:12 Para: [EMAIL PROTECTED] CC: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Chan_Capi 0.3.4a error Are you compiling against stable ? Check the Makefile .. you need to comment out the line that says CFLAGS= -DUNSTABLE_CVS Kind regards, Martin List-Petersen On Wed, 2004-07-14 at 21:16, [EMAIL PROTECTED] wrote: I just downloaded chan_capi.0.3.4a.tar.gz but it will not compile on my system (Suse 9.1). I compiled and installed Asterisk and it is running, did I miss a configuration or dependency for Chan_Capi somewhere ? cheers, Mike linux:/usr/src/chan_capi-0.3.4a # make gcc -pipe -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -DCAPI_ES -DCAPI_GAIN -DCAPI_SYNC -DUNSTABLE_CVS -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c In file included from /usr/include/linux/kernelcapi.h:13, from /usr/include/linux/capi.h:18, from chan_capi.c:34: /usr/include/linux/list.h:604:2: warning: #warning don't include kernel headers in userspace chan_capi.c:61: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_capi.c:61: warning: parameter names (without types) in function declaration chan_capi.c:61: warning: data definition has no type or storage class chan_capi.c:62: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_capi.c:62: warning: parameter names (without types) in function declaration chan_capi.c:62: warning: data definition has no type or storage class chan_capi.c:63: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_capi.c:63: warning: parameter names (without types) in function declaration chan_capi.c:63: warning: data definition has no type or storage class chan_capi.c:64: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_capi.c:64: warning: parameter names (without types) in function declaration chan_capi.c:64: warning: data definition has no type or storage class chan_capi.c:65: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_capi.c:65: warning: parameter names (without types) in function declaration chan_capi.c:65: warning: data definition has no type or storage class chan_capi.c:66: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_capi.c:66: warning: parameter names (without types) in function declaration chan_capi.c:66: warning: data definition has no type or storage class chan_capi.c:67: warning: type defaults to `int' in declaration of `AST_MUTEX_DEFINE_STATIC' chan_capi.c:67: warning: parameter names (without types) in function declaration chan_capi.c:67: warning: data definition has no type or storage class chan_capi.c: In function `_capi_put_cmsg': chan_capi.c:105: error: `capi_put_lock' undeclared (first use in this function) chan_capi.c:105: error: (Each undeclared identifier is reported only once chan_capi.c:105: error: for each function it appears in.) chan_capi.c: In function `capi_echo_canceller': chan_capi.c:180: error: `contrlock' undeclared (first use in this function) chan_capi.c: In function `capi_detect_dtmf': chan_capi.c:230: error: `contrlock' undeclared (first use in this function) chan_capi.c: In function `capi_send_digit': chan_capi.c:308: error: `contrlock' undeclared (first use in this function) chan_capi.c: In function `remove_pipe': chan_capi.c:480: error: `pipelock' undeclared (first use in this function) chan_capi.c: In function `capi_hangup': chan_capi.c:612: error: `usecnt_lock' undeclared (first use in this function) chan_capi.c: In function `capi_call': chan_capi.c:684: error: `pipelock' undeclared (first use in this function) chan_capi.c: In function `capi_read': chan_capi.c:825: error: structure has no member named `delivery' chan_capi.c:826: error: structure has no member named `delivery' chan_capi.c: In function `capi_write': chan_capi.c:898: error: `capi_send_buffer_lock' undeclared (first use in this function) chan_capi.c: In function `capi_new': chan_capi.c:1021: error: structure has no member named `delivery' chan_capi.c:1022: error: structure has no member named `delivery' chan_capi.c:1077: error: `usecnt_lock' undeclared (first use in this function) chan_capi.c: In function `capi_request': chan_capi.c:1129: error: `iflock' undeclared (first use in this function) chan_capi.c:1145: error: `contrlock' undeclared (first use in this function) chan_capi.c: In function `find_pipe': chan_capi.c:1180: error: `pipelock' undeclared (first use in this function) chan_capi.c: In function `pipe_frame': chan_capi.c:1213: error: too few arguments to function
RE: [Asterisk-Users] Help with chan_capi
Title: Mensaje send us a debug file, but first Have you load CAPI driver for Fritz? Avanzada 7, S.L. Sergio Serrano RevueltoRD Manager Avda. Juan López de Peñalver 17Edificio Centro de Empresas Planta 3ª, Pasillo BParque Tecnológico de Andalucía29590 Campanillas(Málaga) [EMAIL PROTECTED] tel: tel2:fax: mobile: (+0034) 951014947(+0034) 951014943. Ext 705(+0034) 951010922618747717 Signature powered by Plaxo Want a signature like this? Add me to your address book... -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Markus KleinEnviado el: jueves, 24 de junio de 2004 12:37Para: [EMAIL PROTECTED]@[EMAIL PROTECTED]Asunto: [Asterisk-Users] Help with chan_capiHi all,I´m a newbie @ asterisk and i´m getting in trouble while configuring asterisk for ISDN first run. I´m using Debian testing an the standard packages of asterisk and chan_capi. My Fritz Card Capis are working fine. I´m already using i4l and hylafax on this machine. When I include the chan_capi to the modules.conf I´m getting an Error like this:[chan_capi.so] = (Common ISDN API for Asterisk)== Parsing '/etc/asterisk/capi.conf': FoundIllegal instruction (core dumped)Asterisk ist no stopped. My modules.conf looks like:[modules]autoload=nonoload = pbx_gtkconsole.sonoload = pbx_kdeconsole.sonoload = app_intercom.sonoload = chan_modem.sonoload = chan_modem_i4l.sonoload = chan_modem_bestdata.sonoload = chan_modem_aopen.soload = res_musiconhold.soload = res_parking.soload = chan_capi.sonoload = chan_iax2.sonoload = chan_zap.sonoload = chan_alsa.sonoload = chan_oss.so[global]chan_capi.so=yesAnd the capi.conf is:[general]nationalprefix=0internationalprefix=00[interfaces]msn=9142829incomingmsn=9142829controller=1softdtmf=1devices=2Does anyone have an idea whats going wrong here?THX4helpMarkus
[Asterisk-Users] Transfer with Budgetone
Hi all, I try to do next transfer: A person contact with me, I would like transfer to other person in next manner. I call to other person and when I say who wants talk with him I hangup phones an call is redirect automatically to other person: 1. call to me 2. Hold the call and call to other person. 3. I say Anyone want talk to you, OK, thanks, 4. I hangup and first person is directly redirect to second person? It is possible with asterisk and budgetone phones? Regards, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Transfer with Budgetone
I know that way, but some person ask for me for first way to do transfers. srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Stephen R. Besch Enviado el: miércoles, 02 de junio de 2004 15:37 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Re: Transfer with Budgetone Sergio Serrano wrote: Hi all, I try to do next transfer: A person contact with me, I would like transfer to other person in next manner. I call to other person and when I say who wants talk with him I hangup phones an call is redirect automatically to other person: 1. call to me 2. Hold the call and call to other person. 3. I say Anyone want talk to you, OK, thanks, 4. I hangup and first person is directly redirect to second person? It is possible with asterisk and budgetone phones? Sergio, Not as far as I know, at least not exactly the way you have outlined it. Try this: 1. call comes to you 2. You hold the call and call other person. 3. You say Someone wants to talk to you, OK, thanks 3a. Other person then hangs up. 3b. You flash back to the original caller 3c. You tell them that you are transferring the call 3d. You transfer the call using the transfer feature on the phone 4. You hangup and first person is transferred to other person? Stephen R. Besch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Transfer with Budgetone
I have just to talk with Grandstream and they say to me that they ar working in 3-way conferencing for BT-100 series. I hope they have FW soon. One question more? How can I do parking call with Budgetone. Before # works fine, but Now it doesn't work. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de John Fraizer Enviado el: miércoles, 02 de junio de 2004 19:29 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] Re: Transfer with Budgetone Tony Hoyle wrote: Stephen R. Besch wrote: Not as far as I know, at least not exactly the way you have outlined it. Try this: 1. call comes to you 2. You hold the call and call other person. 3. You say Someone wants to talk to you, OK, thanks 3a. Other person then hangs up. 3b. You flash back to the original caller 3c. You tell them that you are transferring the call 3d. You transfer the call using the transfer feature on the phone 4. You hangup and first person is transferred to other person? Ugh. So Asterisk doesn't handle transfer? Every company phone system I've ever used has not required 3a-3d. It looks like a real hack to do so. It anyone working on implementing this? Tony Asterisk handles transfer just fine. It's the P-O-S Grandstreams that don't. John ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Codec G729 uninstall
Hi all, Are there any way to clean codec_g729b license ffrom Asterisk. I would like to clean a license to install other more big, but when I do ../codec_g729b/Registration --XX I obtain a segmentation fault. Any idea? srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Chan_capi and modem-fax
Hi all, I have just put a message from a few days with a problem with CAPI hangup. I have noticed that line with 97% of hangs, is a line connected with a ATA286 with a modem-fax. Could it be the problem? Regards, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] multiplle isdn card
First thing you must is read next url http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO and if you hav done this, please attach your capi.conf. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] nombre de massimo Enviado el: martes, 04 de mayo de 2004 19:31 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] multiplle isdn card Hi to all, I added a second isdn fritz card to my asterisk box to manage a second isdn line. But when I start capi it sees only one controller. How I can enable the second isdn card. Thank you Bye ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI Eicon Diva Server 4BRI
Hi all, I have a PC working with a DIVA Eicon Server 4BRI during a lot of time. Now I can't make call but I can receive calls. I load diva with command: divactrl load -c 1 -f ETSI -u -t 0 Country: Spain Isdnmode: point to point My capi.conf is the next: [global] mode=immediate isdnmode=ptp txgain=0.8 rxgain=0.5 [interfaces] msn=952901652,952901987 incomingmsn=* controller=1,2 softdtmf=0 context=default echocancel=1 echotail=64 callgroup=1 devices=4 I obtnain next trace in console: -- Attempting call on CAPI/@952901987:B951014947||r for [EMAIL PROTECTED]:2 (Retry 1) -- data = @952901987:B951014947||r -- capi request omsn = @952901987 == found capi with omsn = 952901987 == CAPI Call CAPI[contr1/952901987]/7 with B3 == CAPI Call CAPI[contr1/952901987]/7 with B3-- creating pipe for PLCI=-1 -- CONNECT_CONF ID=001 #0x0f52 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 == received CONNECT_CONF PLCI = 0x301 INFO = 0 == DISCONNECT_IND PLCI=0x301 REASON=0x3302 -- CAPI Hangingup -- removed pipe for PLCI = 0x301 Apr 19 12:03:17 NOTICE[25619]: pbx_spool.c:199 attempt_thread: Call failed to go through, reason 1 Any idea? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CAPI Eicon Diva Server 4BRI
Hi, Executing divactrl dchannel -dmonitor -Debug I obtain the next messages: MDL-ERROR(G) SIG-EVENT 0A MDL-ERROR(G) SIG-EVENT 0A MORE SIG-X(045) 08 01 12 05 A1 04 03 80 90 A3 18 01 81 6C 0B 00 A0 39 35 32 39 30 31 39 38 37 70 0D 80 39 35 31 30 31 34 39 34 37 7C 7C 72 7D 02 91 81 Q.931 CR12 SETUP Sending complete Bearer Capability 80 90 a3 Channel Id 81 Calling Party Number 00 a0 '952901987' Called Party Number 80 '951014947||r' HLC 91 81 MDL-ERROR(G) SIG-EVENT 0A SIG-EVENT 0A EVENT: Call failed in State 'Call initiated' Link disconnected, TEI error MDL-ERROR(G) SIG-EVENT 0A MDL-ERROR(G) SIG-EVENT 0A MDL-ERROR(G) SIG-EVENT 0A Any idea? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sergio Serrano Revuelto Enviado el: miércoles, 19 de mayo de 2004 12:00 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] CAPI Eicon Diva Server 4BRI Hi all, I have a PC working with a DIVA Eicon Server 4BRI during a lot of time. Now I can't make call but I can receive calls. I load diva with command: divactrl load -c 1 -f ETSI -u -t 0 Country: Spain Isdnmode: point to point My capi.conf is the next: [global] mode=immediate isdnmode=ptp txgain=0.8 rxgain=0.5 [interfaces] msn=952901652,952901987 incomingmsn=* controller=1,2 softdtmf=0 context=default echocancel=1 echotail=64 callgroup=1 devices=4 I obtnain next trace in console: -- Attempting call on CAPI/@952901987:B951014947||r for [EMAIL PROTECTED]:2 (Retry 1) -- data = @952901987:B951014947||r -- capi request omsn = @952901987 == found capi with omsn = 952901987 == CAPI Call CAPI[contr1/952901987]/7 with B3 == CAPI Call CAPI[contr1/952901987]/7 with B3-- creating pipe for PLCI=-1 -- CONNECT_CONF ID=001 #0x0f52 LEN=0014 Controller/PLCI/NCCI= 0x301 Info= 0x0 == received CONNECT_CONF PLCI = 0x301 INFO = 0 == DISCONNECT_IND PLCI=0x301 REASON=0x3302 -- CAPI Hangingup -- removed pipe for PLCI = 0x301 Apr 19 12:03:17 NOTICE[25619]: pbx_spool.c:199 attempt_thread: Call failed to go through, reason 1 Any idea? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi Fax
Hi all, I would like to know if chan_capi is prepared to receive faxes. I have a eicon deiva server 4bri with chan_capi and Grandstream HandyTone connected to a Fax, but this fax can't receive faxes. Any idea? Thanks, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk + GrandStream SIP phones
Try to add a qualify= to sip.conf, and try to exec a sip show peers. In spite of phones appears like register, if you use NAT, your firewall can cut communication. Try the next: Just after phone register call to it, and then wait for a minutes and try to call again. Could you call first time but not in second one? It is due to your firewall. Try to configure wuth next config: [1004] .. . qualify= ... .. In you grandstream configuration try to put time to expire register 1 minute and then try to do the previous test. I'm sorry for my english, but I hope this let you call. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de pesb Enviado el: lunes, 29 de marzo de 2004 20:26 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Asterisk + GrandStream SIP phones -This is my 'sip.conf' file: ;* ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls tos=184 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing registration disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw [1004] type=friend username=1004 secret= reinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1004 nat=1 disallow=all allow=ulaw allow=alaw [1005] type=friend username=1005 secret= reinvite=no canreinvite=no host=dynamic dtmfmode=inband mailbox=1005 nat=1 disallow=all allow=ulaw allow=alaw ;*** -And this is the basic seting of my two GrandStream SIP phones: ***[1005] IP Address:192.168.0.105 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:empty SIP User ID:1005 Authenticate ID:1005 Authenticate Password:123 Name:1005 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ***[1004] IP Address:192.168.0.104 Subnet Mask:255.255.255.0 SIP Server: 192.168.0.103 Outbound Proxy:empty SIP User ID:1004 Authenticate ID:1004 Authenticate Password:123 Name:1004 Preferred Vocoder: choice 1: PCMU choice 2: PCMA choice 3: G723 choice 4: G729 choice 5: G726-32 choice 6: G728 G723 rate: 6.3kbps Silence Suppression:No Send DTMF:in-audio ** I have 2 SIP GrandStream phones, both phones are correctly registered to the Asterisk server. But, when I try to make a call from registered phone '1005' to registered phone '1004', dialing 1004, Asterisk responds with the 'Status: 404 Not Found' message. How do I have to dial? What else do I need to set? Find attached my traffic captured on ethereal. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 and SCSI
Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] G.729 and SCSI
Yes I have mounted CDROM first with automount(/dev/cdrom) and second manually(/dev/hde) but nothing. Any idea? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Andrew Thompson Enviado el: jueves, 25 de marzo de 2004 17:59 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] G.729 and SCSI Sergio Serrano wrote: Hi all, I try to install a G.729 license in SCSI system with a IDE CDROM but I can't do it. Any one has experience to do this? Regards, srsergio Here is the wiki page for g729: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing It's not specifically listed there, but the licensing process has issues with SCSI only systems. - Andrew Thompson http://aktzero.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CAPI
Hi,
I have a problem with a Eicon Diva Server 4 BRI. I have 4 BRI ISDN and
11 number for these 4 ISDN. At first I have connected one of these 4 ISDN.
When I try to call I receive the next trace:
-- Executing ChanIsAvail(SIP/716-b0cd,
CAPI/971844367CAPI/971846015CAPI/971846034CAPI/971846036CAPI/971846094
CAPI/971846141CAPI/971846142CAPI/971846143CAPI/971846146CAPI/971846147C
API/971846148) in new stack
-- data = 971844367
-- capi request omsn = 971844367
== found capi with omsn = 971844367
-- CAPI Hangingup
-- Executing SubString(SIP/716-b0cd,
CANAL=CAPI[contr1/971844367]/0|12|9) in new stack
Mar 22 17:51:00 WARNING[262161]: app_substring.c:63 substring_exec: The use
of Substring application is deprecated. Please use ${variable:a:b} instead
-- Executing Dial(SIP/716-b0cd, CAPI/@971844367:687754642|17) in new
stack
-- data = @971844367:687754642
-- capi request omsn = @971844367
== found capi with omsn = 971844367
== CAPI Call CAPI[contr1/971844367]/1 == CAPI Call
CAPI[contr1/971844367]/1 -- creating pipe for PLCI=-1
sent CONNECT_REQ MN =0x7
-- Called @971844367:687754642
-- CONNECT_CONF ID=001 #0x0007 LEN=0014
Controller/PLCI/NCCI= 0x301
Info= 0x0
== received CONNECT_CONF PLCI = 0x301 INFO = 0
== DISCONNECT_IND PLCI=0x301 REASON=0x3301
sent DISCONNECT_RESP PLCI=0x301
-- CAPI Hangingup
-- removed pipe for PLCI = 0x301
== No one is available to answer at this time
My extensions.conf is the next:
exten=_X,1,ChanIsAvail(CAPI/971844367CAPI/971846015CAPI/971846034
CAPI/971846036CAPI/971846094CAPI/971846141CAPI/971846142CAPI/
971846143CAPI/971846146CAPI/971846147CAPI/971846148)
;exten=_X,1,ChanIsAvail(CAPI/971844367)
exten=_X,2,SubString,CANAL=${AVAILCHAN}|12|9
exten=_X,3,Dial(CAPI/@${CANAL}:${EXTEN}|17)
exten=_X,104,Playback(invalid)
exten=_X,105,Hangup()
and my capi.conf is the next:
[global]
mode=immediate
isdnmode=multipoint
txgain=0.8
rxgain=0.8
[interfaces]
msn=971844367,971846015,971846034,971846036,971846094,971846141,971846142,97
1846143,971846146,971846147,971846148
;msn=971844367
incomingmsn=*
controller=1
context=default
echocancel=1
echotail=64
devices=8
any idea?
srsergio
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RE: [Asterisk-Users] PCphoneline FXO to FXS box??
We are going to do this test next week. I will say the result Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jim Rosenberg Enviado el: domingo, 29 de febrero de 2004 1:15 Para: Asterisk Asunto: [Asterisk-Users] PCphoneline FXO to FXS box?? pcphoneline.com sells a little box with two RJ-11 jacks that is supposed to convert an FXS port into an FXO port. According to their blurb, when a call comes in it basically conferences the two lines together. Is anyone out there using this box with Asterisk? Any problems? What happens to callerid when you get an incoming call? I'm thinking about using one of these things with the Grandstream ATA-286 for a spot where I may not have a PC available to put a Digium FXO card into. (Don't have Ethernet where the PSTN jack is, so the easiest thing to do is WiFi it. Seems a shame to dedicate a whole PC to just a single FXO port ...) -T.i.A., Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] GS Budgetone 101 canot receive calls
If your BG 101 is in intranet, try to adjust your qualify parameter to 60. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Matthew B Marlowe Enviado el: viernes, 27 de febrero de 2004 2:08 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] GS Budgetone 101 canot receive calls Show us your extensions.conf Sincerely, Matthew Marlowe Gear 3 Technologies, LLC 609.252.1155 x614 www.gear3.com (00) Choose a job you love, and you will /||\ never have to work a day in your life. =/\= -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carlos Chavez Sent: Thursday, February 26, 2004 7:59 PM To: Asterisk Subject: [Asterisk-Users] GS Budgetone 101 canot receive calls I just got a Budgetone 101 phone today and after configuring it I can make calls to any other phone on my * server. The problem is that no matter what I do, when I dial the extension assigned to the phone it will always send me directly to voicemail with the busy message. I tried searching through the mailing list but have not been able to find a solution. Can anybody help? Here is the entry in sip.conf: [4010] username=4010 type=friend secret=(secret) host=dynamic amaflags=default callerid=Roberto IP Phone 4010 mailbox=4010 canreinvite=no ;reinvite=no ;nat=yes qualify=no dtmfmode=info defaultip=192.168.0.102 I can see on the * console that the phone is registering. If I do a sip show peers I ge thw following: Name/usernameHost Mask Port Status 4010/4010192.168.0.102 (D) 255.255.255.255 5060 Unmonitored I tried the phone both on the local network and from another network. -- Carlos Chavez Computer Engineer, CCNA Corporativo Lacer S.A. de C.V. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Extrange Problem
Title: Mensaje Hi all, For a few days we have a veryextrange problem. We have an intranet with Budgetone and others SIP Phones. In the extranet We HaveBudgetone Phones. The whole system was working well between the extranet and the intranet until a few days ago. When we try to speak with a Budgetone of the intranet, we can speak during a few seconds but after a time the audio is cut in the sense of intranet-extranet. The problem is not only it, but if a budgetone of the intranet speaks with another phone of the intranet the same thing happens. After a time of conversation the audio is cut in the sense of the budgetone to another phone. I see the next meesage in debug: Feb 26 10:50:04 DEBUG[50193]: Didn't get a frame from channel: SIP/707-996a I have checked the files of configuration. It does not appear at all any more in the files of logs and I do not know that to do. Can it be a problem of the internal network? of the switches? Is there any bug in the budgetones? Any idea? Thanks, srsergio
[Asterisk-Users] RE: [Asterisk-Users] Spanish indications configurationº
Title: Mensaje Hola, ahi va la sección [es] para el indications.conf [es] description = Spain ringcadence = 1500,3000 dial = 425 busy = 425/200,0/200 ring = 425/1500,0/3000 congestion = 425/200,0/200,425/200,0/200,425/200,0/600 callwaiting = 425/175,0/175,425/175,0/3500 dialrecall = !425/200,!0/200,!425/200,!0/200,!425/200,!0/200,425 record = 1400/500,0/15000 info = 950/330,0/1000 dialout = 500 www.avanzada7.com Sergio Serrano RevueltoRD Manager Avanzada 7 [EMAIL PROTECTED] tel: fax: mobile: (+0034) 951014947(+0034) 951010922618747717 Signature powered by Plaxo Want a signature like this? Add me to your address book... -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de dfmEnviado el: viernes, 13 de febrero de 2004 12:18Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Spanish indications configurationº Hi all We've been using * for a while here in Spain, but some people has told us that they have problems when they type an extension calling to us. I've been trying to find out what's going on, and it's an issue that only happens with some ISDN and analog calls, not from mobile calls as long as i have observe. My concern is about the indications.conf Spanish telco lines configuration, Is in the * list any Spanish user that can share this configuration with me and see if it's ok?? i would really appreciate it. Diego
RE: [Asterisk-Users] Eicon Diva Server
Hi all, I will ptobe your answers tomorrow. I'll say the results. Thanks for all. -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Sascha Knific Enviado el: martes, 10 de febrero de 2004 22:08 Para: [EMAIL PROTECTED] Asunto: AW: [Asterisk-Users] Eicon Diva Server Hi Sergio, I don´t have any setup like you but looking over you config I saw this: My capi.conf is the next: [global] mode=immediate isdnmode=multipoint txgain=0.5 rxgain=0.5 [interfaces] msn=951014943 incomingmsn=951014943 controller=1 context=default echocancel=1 echotail=64 devices=2 msn=951014944 incoming=951014944 controller=1 ^^^ Maybe you should try controller=2 here. context=default echocancel=1 echotail=64 devices=2 Tell me if it helps. Sascha --- Sascha Knific K Systems Design Tel. +49-8151-773260Wittelsbacherstr. 6a Fax. +49-8151-77326282319 Starnberg, Germany Leo +49-8151-773261WGS84: N57°59,875' E011°20,568' [EMAIL PROTECTED] http://www.k-sysdes.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Eicon Diva Server
Hi all,
anyone could help me with capi.conf?. I have installed an Eicon
Diva Server 4BRI. I have 2 EuroISDN BRI lines,
First line number: 951014943
Second line number: 951014944
I try to do 4 calls but, I can't do more than two call.
My capi.conf is the next:
[global]
mode=immediate
isdnmode=multipoint
txgain=0.5
rxgain=0.5
[interfaces]
msn=951014943
incomingmsn=951014943
controller=1
context=default
echocancel=1
echotail=64
devices=2
msn=951014944
incoming=951014944
controller=1
context=default
echocancel=1
echotail=64
devices=2
And mi extensions.conf for Dial CAPI are the next:
exten=_X,1,ChanIsAvail(CAPI/951014943CAPI/951014944)
exten=_X,2,SubString,CANAL=${AVAILCHAN}|12|9
exten=_X,3,Dial(CAPI/@${CANAL}:B${EXTEN}|17)
Any idea?
Thanks,
srsergio
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RE: [Asterisk-Users] Call recording
You must use Monitor Application Happy New Year, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Edoardo Borghesi [fabbricadigitale] Enviado el: viernes, 02 de enero de 2004 12:33 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] Call recording Hello, I need a way to record every call made to asterisk on a file. The app_record application works but it is blocking, so I can't connect a phone-operator and an user while recording. I thought to use the MeetMe application and using a fake user to record the call but in this way I can't know if the phone-operator is ready to answer or is answering another user (i.e., the operator is always in conference and I obviously don't want to have more than one user connected to the conference). Does anyone know a way to achieve this goal? I can also modify some code if this is needed. Thanks Edoardo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RxFAX application
Hi mack_jpn I think problem is CFR 84 sending. In console appears that CFR 84 si sent but te other fax doesn't receive CRF 84, and then RXFAX is waiting for the fax but the other fax doesn't send it ever. I try to see source code. Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Masakazu Nakano Enviado el: domingo, 21 de diciembre de 2003 5:37 Para: [EMAIL PROTECTED] CC: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] RxFAX application Hi sergio On Fri, 19 Dec 2003 14:49:15 +0100 Sergio Serrano Revuelto [EMAIL PROTECTED] wrote: Hi all, I have tested RxFAX application through X100P card. When Fax arrive i obtain the next trace: snip 5 (0.01679,-0.16590) - 0.02781 6 ( -0.04451, 0.75304) - 0.56904 7 ( -0.01415,-0.29305) - 0.08608 Fast carrier down Segmentation fault And i obtain 8 byte tif file. Any Idea? I have installed tiff-3.5.7 and spandsp-20031021. I get same result. but the end part looks like that. Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down -- Hungup 'Zap/1-1' with no segfault I'm tryed with tiff-v3.6.0 ( use with tar balled headers ) and spandsp-20031021 Does anyone have good result? Regards. mack_jpn ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFAX application
Hi all, I have tested RxFAX application through X100P card. When Fax arrive i obtain the next trace: -- Starting simple switch on 'Zap/1-1' -- Executing Answer(Zap/1-1, ) in new stack -- Executing SetMusicOnHold(Zap/1-1, random) in new stack -- Executing WaitMusicOnHold(Zap/1-1, 5) in new stack -- Started music on hold, class 'random', on Zap/1-1 -- Redirecting Zap/1-1 to fax extension -- Stopped music on hold on Zap/1-1 == Spawn extension (default, fax, 0) exited non-zero on 'Zap/1-1' -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/fax/uno.tif) in new stack Changed from phase 0 to 1 Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 38 37 36 35 34 33 32 31 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Store and forward Internet fax: no Real-time Internet fax: no Preferred octets: 256 Can receive fax Data signalling rate: V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Inch-based resolution preferred: no Metric-based resolution preferred: no Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 c6 f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 TSI: 43 32 32 39 30 31 30 31 35 39 20 20 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: 951010922 DCS: 83 00 c6 f0 80 80 00 DCS with final frame tag In state 9 DCS: Store and forward Internet fax: no Real-time Internet fax: no Can receive fax Data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Inch-based resolution preferred: no Metric-based resolution preferred: no Minimum scan line time for higher resolutions: T15.4 = T7.7 Get at V.29 Changed from phase 3 to 5 Fast carrier up Fast carrier down Changed from phase 5 to 4 0 bad bits in trainability test Start rx document - compression 2 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Fast carrier down Fast carrier up Equalizer state: -7 (0.0, 0.0) - 0.0 -6 (0.0, 0.0) - 0.0 -5 (0.0, 0.0) - 0.0 -4 (0.0, 0.0) - 0.0 -3 (0.0, 0.0) - 0.0 -2 (0.25105, 0.74039) - 0.61121 -1 ( -0.87268,-0.36304) - 0.89337 0 ( -1.79414,-2.01854) - 7.29345 1 ( -0.87268,-0.36304) - 0.89337 2 (0.25105, 0.74039) - 0.61121 3 (0.0, 0.0) - 0.0 4 (0.0, 0.0) - 0.0 5 (0.0, 0.0) - 0.0 6 (0.0, 0.0) - 0.0 7 (0.0, 0.0) - 0.0 Equalizer state: -7 (0.00649,-0.03380) - 0.00118 -6 (0.02073,-0.00508) - 0.00046 -5 (0.03125,-0.04397) - 0.00291 -4 (0.02094,-0.05189) - 0.00313 -3 (0.00954,-0.02848) - 0.00090 -2 (0.26182, 0.72429) - 0.59315 -1 ( -0.84074,-0.41198) - 0.87657 0 ( -1.76021,-2.05082) - 7.30420 1 ( -0.85500,-0.38151) - 0.87658 2 (0.22413, 0.68563) - 0.52033 3 ( -0.03512,-0.10808) - 0.01291 4 ( -0.02204,-0.03244) - 0.00154 5 (0.04513, 0.11010) - 0.01416 6 (0.05265, 0.07431) - 0.00829 7 ( -0.01433,-0.11280) - 0.01293 Equalizer state: -7 (0.14017, 0.08799) - 0.02739 -6 ( -0.18079,-0.02633) - 0.03338 -5 (0.00565, 0.03149) - 0.00102 -4 (0.15813,-0.07292) - 0.03032 -3 ( -0.28991,-0.34523) - 0.20324 -2 (0.10369, 0.59990) - 0.37063 -1 (0.03284,-0.02606) - 0.00176 0 ( -0.11463,-0.90551) - 0.83309 1 (0.03712, 0.71173) - 0.50794 2 ( -0.46280, 0.55488) - 0.52208 3 ( -1.38062,-1.69745) - 4.78743 4 ( -0.95295,-1.72961) - 3.89964 5 (0.03678, 0.07704) - 0.00729 6 (0.29737, 0.77668) - 0.69166 7 ( -0.14636,-0.40090) - 0.18214 Fast carrier training failed Equalizer state: -7 (0.10167, 0.06545) - 0.01462 -6 (
[Asterisk-Users] Asterisk as SIP Server
Hey Srs. I have a little problem with the next scenario: Internal Phone(801)--Asterisk(public IP) --INTERNET--ADSL Router--Budgetone(716) |--ADSL Router--Budgetone(717) My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls srvlookup = yes ; Enable SRV lookups on outbound calls tos=lowdelay domain=AVANZADA7 maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=120 ; Default length of incoming/outoing disallow=all; Disallow all codecs allow=g729 allow=alaw [716] type=friend username=716 fromuser=716 fromdomain=AVANZADA7 secret=716 host=dynamic allow=g729 allow=alaw qualify=yes canreinvite=yes context=outgoing dtmfmode=info nat=yes [717] type=friend username=717 fromuser=717 fromdomain=AVANZADA7 secret=717 host=dynamic allow=g729 allow=alaw qualify=yes canreinvite=yes context=outgoing dtmfmode=info nat=yes [801] type=friend username=801 fromuser=801 secret=801 fromdomain=AVANZADA7 host=dynamic defaultip=192.168.0.185 allow=g729 allow=alaw allow=ulaw mailbox=801 context=outgoing canreinvite=no dtmfmode=info nat=no If I make a call from 716 or 717 to 801 have no problem, but If I try make a call from 717 to 716 or 716 to 717, signalling is correct but when a side pickup call, these call hangup. In console appear( warning: detected firewall/NAT type is UDP blocked) Any idea, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Still TDM400P problem
Next configuration must work: zaptel.conf fxoks=1-4 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1-4 Srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de JanM Enviado el: jueves, 20 de noviembre de 2003 11:27 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Still TDM400P problem -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michiel Betel Sent: den 20 november 2003 11:24 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Still TDM400P problem JanM wrote: Hi again all, I have searched the list for help with my problem but I can´t find an answer. I only manage to get one port of my TDM400P card working. When I do dmesg I get following, seems like four discovered ports: --- Zapata Telephony Interface Registered on major 196 PCI: Found IRQ 11 for device 02:00.0 PCI: Sharing IRQ 11 with 02:07.1 PCI: Sharing IRQ 11 with 02:0c.0 Freshmaker version: 63 Freshmaker passed register test Module 0: Installed -- AUTO Module 1: Installed -- AUTO Module 2: Installed -- AUTO Module 3: Installed -- AUTO Found a Wildcard FXS: Wildcard TDM400P REV E/F (4 modules) Registered tone zone 5 (Finland) But when I do ztcfg -vv I only get one port configured: Zaptel Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) 1 channels configured. How do I configure/load the rest of the ports? Add them in /etc/zaptel.conf... ztcfg reads this file and configures zap ports accordingly Michiel When I try that I get an error: Ouch ... error while writing audio data: : Broken pipe This works: zaptel.conf fxoks=1 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1 But when I try to set up more ports/channels on the first card it stops working. The following configuration doesn´t makes the Ouch... error: zaptel.conf fxoks=1-4 loadzone=fi defaultzone=fi Zapata.conf [channels] group=1 context=internt signalling=fxo_ks channel=1-4 What does it mean with audio data in the error message? ---JanM--- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Open Source Linux PBX!
Title: Mensaje try to cvs srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Quan Le TrungEnviado el: jueves, 13 de noviembre de 2003 10:43Para: [EMAIL PROTECTED]CC: [EMAIL PROTECTED]; [EMAIL PROTECTED]Asunto: [Asterisk-Users] Open Source Linux PBX! Hi! I have just bought the Wildcard TDM400P 4-port FXS PCI Card, and Wildcard X100P is a single-port FXO PCI Card to install on my computer to implement the PBX (Private Packet Exchange). However, I cannot download the corresponding softwares (asterisk, libpri and zaptel) at the following address: ftp://ftp.asterisk.org/pub/telephony . If anyone has already downloaded these softwares, please kindly send them to me via the following e-mail: [EMAIL PROTECTED] . Thanks in advance! P.S Please kindly send files in separate e-mails to me because of limited size of received e-mails. Best regards, Quan L. T.
RE: [Asterisk-Users] 2 X100Ps give error
Try to load module manually: modprobe wcfxo; ztcfg - srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Cameron Palmer Enviado el: viernes, 31 de octubre de 2003 6:27 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] 2 X100Ps give error I have two X100Ps in my zaptel.conf and have it configured: fxsks=1-2 When starting rc.zaptel I get the error: Loading zaptel framework... Loading zaptel hardware modules: wcfxo wcusb Running ztcfg... ZT_CHANCONFIG failed on channel 2: No such device or address (6) As far as I can tell from the archived mailing lists I shouldn't have this problem. I do notice a difference in lspci which concerns me: 00:0d.0 Communication controller: Tiger Jet Network Inc. Intel 537 00:0e.0 Serial controller: Tiger Jet Network Inc. Intel 537 Why is one listed as a Communication controller and the other a serial controller? What I'm trying to do... I have activated the second line option and put it into a call hunt with the first line on my Vonage account. At the office I have a Konexx DWI connecting my Nortel phone to my desktop running RedHat and a X100P. My goal is seamless (as far as my boss in Santa Cruz can tell) connectivity between home and office but maintaining a low ongoing cost. Namely about $60 per month. Other details... The home setup - 2 7960G SIP phones 1 ATA-186 Linux (slackware 9.1) 2.4.22 on PIII 650MHz Connected to Vonage over the POTS. 2 X100P FXO Cards The office setup - Dell Precision 650 Redhat 9 1 X100P FXO Card Konexx DWI connected to Nortel phone ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 X100Ps give error
Try to do cat /proc/pci. You must verify that card doesn't share IRQ with USB or other component. srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Cameron Palmer Enviado el: viernes, 31 de octubre de 2003 8:32 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] 2 X100Ps give error RESULTS of your suggestion. I had tried similar variations from the mailing list archives the result was the same. # modprobe wcfxo ZT_CHANCONFIG failed on channel 2: No such device or address (6) /lib/modules/2.4.22/misc/wcfxo.o: post-install wcfxo failed /lib/modules/2.4.22/misc/wcfxo.o: insmod wcfxo failed # ztcfg - Zaptel Configuration == Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) Channel 02: FXS Kewlstart (Default) (Slaves: 02) 2 channels configured. ZT_CHANCONFIG failed on channel 2: No such device or address (6) cameron. On Fri, 31 Oct 2003, Sergio Serrano Revuelto wrote: Try to load module manually: modprobe wcfxo; ztcfg - srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Cameron Palmer Enviado el: viernes, 31 de octubre de 2003 6:27 Para: [EMAIL PROTECTED] Asunto: [Asterisk-Users] 2 X100Ps give error I have two X100Ps in my zaptel.conf and have it configured: fxsks=1-2 When starting rc.zaptel I get the error: Loading zaptel framework... Loading zaptel hardware modules: wcfxo wcusb Running ztcfg... ZT_CHANCONFIG failed on channel 2: No such device or address (6) As far as I can tell from the archived mailing lists I shouldn't have this problem. I do notice a difference in lspci which concerns me: 00:0d.0 Communication controller: Tiger Jet Network Inc. Intel 537 00:0e.0 Serial controller: Tiger Jet Network Inc. Intel 537 Why is one listed as a Communication controller and the other a serial controller? What I'm trying to do... I have activated the second line option and put it into a call hunt with the first line on my Vonage account. At the office I have a Konexx DWI connecting my Nortel phone to my desktop running RedHat and a X100P. My goal is seamless (as far as my boss in Santa Cruz can tell) connectivity between home and office but maintaining a low ongoing cost. Namely about $60 per month. Other details... The home setup - 2 7960G SIP phones 1 ATA-186 Linux (slackware 9.1) 2.4.22 on PIII 650MHz Connected to Vonage over the POTS. 2 X100P FXO Cards The office setup - Dell Precision 650 Redhat 9 1 X100P FXO Card Konexx DWI connected to Nortel phone ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Channel Bank with E1
I need connect up to 100 analog phone to a H.323 network through *. I think use TE410P, But I need to know what channel bank is better. I use E1 lines Any idea? Thanks in advance, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de DUSTIN WILDES Enviado el: miércoles, 29 de octubre de 2003 14:30 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] Answering Machine Detection Thanks for all the info! So I take it I would need to either build an additional APP to asterisk like (voice_detection) or into an AGI and have that application or AGI run after the call is Answered? Fortunately it's not a telemarketing system! :-) It's an appointment reminder system for some of our employees. Calls them up and reminds them of important tasks like meetings and stuff. -Original Message- From: Michiel Betel [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 8:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Answering Machine Detection See http://resource.intel.com/telecom/support/documentation/unix/SR50_linux/ html _files/vox_feat/contents.html#TopOfPage chapter 2 for a basic insight on Dialogic does it... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: woensdag 29 oktober 2003 3:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Answering Machine Detection Humans tend to say Hello? (short burst of audio followed by silence), and answering machines tend to say I'm sorry I'm not here right now, please leave a message after the beep (long burst of audio followed by a beep and silence). So, basically you need to decide 1) what is audio and what is background noise and 2) how long should there be audio followed by silence. On Tue, 2003-10-28 at 19:25, Alastair Maw wrote: On 27/10/03 21:57, DUSTIN WILDES wrote: Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? There's obviously no nice way of doing this. If you're doing telemarketing, and you're playing pre-recorded audio, which of course is a nasty thing to do, the algorithm is something like: 1. Dial out. 2. Wait for answer. 3. Start playing audio. 4. If you hear something that sounds like a beep, either hang up and try again later, or stop the audio, pause for two seconds and start playing it again. 5. Hang up when finished playing audio. Step 4 is accomplished by doing a FFT on the incoming audio into frequency buckets and taking a rolling average of the mean and standard deviation, such that you can detect when a fixed monotone beep occurs at the other end. If you don't want to play audio files and wait for beeps, and want to connect real humans to each other, then there's no decent way to do this, as the only difference between humans and arbitrary answering machines is that the answering machines give you a beep prompt to record your message. Regards, -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN BRI card
Title: Mensaje AVM Fritz it good for Asterisk. A little difficult to configure but not impossible. srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Tomica CrnekEnviado el: jueves, 16 de octubre de 2003 14:36Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] ISDN BRI card Anyone knows of a good ISDN BRI card to use with Asterisk?
[Asterisk-Users] SIP phone hangs after some hours
Hi, I have a problem with sip.conf. After some hours my sip phone(netergy) hangs. In clonse appears the next logs repeatly: 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.155 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK5ffceb80 From: asterisk sip:[EMAIL PROTECTED];tag=as4b104f64 To: sip:192.168.0.155 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.155:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as4b104f64 To: sip:192.168.0.155 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK5ffceb80 Supported: timer,100rel Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS,PRACK Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 13 headers, 0 lines DEBUG[12301]: File chan_sip.c, Line 533 (__sip_ack): Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Found DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' My sip.conf is the next: ; ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to bindaddr = 192.168.0.207; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay maxexpirey=10 ; Max length of incoming registration we allow defaultexpirey=10 ; Default length of incoming/outoing registration [705] type=friend username=705 host=192.168.0.155 dtmfmode=inband mailbox=705 callerid=705 context=outgoing reinvite=yes canreinvite=no qualify=yes nat=-1 My sip phone doesn't register in asterisk due to my decision. I can send and receive call, but if phones is inactive during some hours it hangs. It is due to asterisk or my sip phone? Any idea? Thanks, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using Asterisk in an netted scenario
Title: Mensaje Yes yo can do it. srsergio -Mensaje original-De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de e-smithEnviado el: miércoles, 24 de septiembre de 2003 15:02Para: [EMAIL PROTECTED]Asunto: [Asterisk-Users] Using Asterisk in an netted scenario Hi, Just to get myideeas confirmed: Is it possible to useasterisk in a scenario where : - One Asterisk connects to another asterisk over tcp/ip with qos to another asterisk.- The otherasterisk has an connection to the PSTN whitch users connected to the first asterisk uses to get to the public telephone network. Kind regards Mats Karlsson
RE: [Asterisk-Users] how to dial a h323 destination ?
exten=XXX,1,Dial(h323/3|17|tTm)
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Enviado el: martes, 23 de septiembre de 2003 11:07
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] how to dial a h323 destination ?
Please, can somebody tell me how do a h323 call correctly with the dial
app ?
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Montag, 22. September 2003 18:26
An: Asterisk User
Betreff: [Asterisk-Users] how to dial a h323 destination ?
Hi all,
i have big problems to make a h323 call over the gatekeeper from my
provider. The provider demanded following account data:
H323 ID:XXX-XXX-XX-X
DetinationNumer: XXX
I have configured the oh323.conf following:
gatekeeper=XX.XX.XXX.XXX
alias=XXX-XXX-XX-X
Isx the alias equal to the h323id ?
And how i have to make a call with the dial app ?
I have following config:
exten = _01099X.,1,Dial,OH323/${EXTEN:7}
exten = _01099X.,2,Hangup
I thought it would be enough when i give the destination number if i
registered at the gk, isn't it ? Or is a ip and something like a
userbname necessary ? And if how can i dial so?
Can somebody help please ?
Thanks,
Thomas.
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RE: [Asterisk-Users] how to dial a h323 destination ?
Could you send me your h323.conf and you gnugk.ini?
Sergio Serrano Revuelto
Responsable de Consultoría
Avanzada 7, S.L.
Teléfono / Fax: +34 951 01 49 47 / +34 951 01 09 22
www.avanzada7.com
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger
Enviado el: martes, 23 de septiembre de 2003 11:28
Para: [EMAIL PROTECTED]
Asunto: AW: [Asterisk-Users] how to dial a h323 destination ?
OK. This is what i know too...
But this don't work. The gatekeeper tells me everytime caller not
registered. If i start *, the registration at the gatekeeper is ok. If
i make i call it is not ok. Is there any other info that i have to
send with ?
like : Dial(OH323/[EMAIL PROTECTED]/H323ID or similar like this ?
Thanks for help,
Thomas.
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Sergio
Serrano Revuelto
Gesendet: Dienstag, 23. September 2003 11:15
An: [EMAIL PROTECTED]
Betreff: RE: [Asterisk-Users] how to dial a h323 destination ?
exten=XXX,1,Dial(h323/3|17|tTm)
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Thomas
Haeger Enviado el: martes, 23 de septiembre de 2003 11:07
Para: [EMAIL PROTECTED]
Asunto: RE: [Asterisk-Users] how to dial a h323 destination ?
Please, can somebody tell me how do a h323 call correctly with the dial
app ?
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Auftrag von Thomas
Haeger
Gesendet: Montag, 22. September 2003 18:26
An: Asterisk User
Betreff: [Asterisk-Users] how to dial a h323 destination ?
Hi all,
i have big problems to make a h323 call over the gatekeeper from my
provider. The provider demanded following account data:
H323 ID:XXX-XXX-XX-X
DetinationNumer: XXX
I have configured the oh323.conf following:
gatekeeper=XX.XX.XXX.XXX
alias=XXX-XXX-XX-X
Isx the alias equal to the h323id ?
And how i have to make a call with the dial app ?
I have following config:
exten = _01099X.,1,Dial,OH323/${EXTEN:7}
exten = _01099X.,2,Hangup
I thought it would be enough when i give the destination number if i
registered at the gk, isn't it ? Or is a ip and something like a
userbname necessary ? And if how can i dial so?
Can somebody help please ?
Thanks,
Thomas.
***
beroNet technologies GmbH
Dipl.- Ing. Thomas Häger
Potsdamer Str. 18 A
14513 Teltow
FON:+49 (0) 3328 3077731
FAX:+49 (0) 3328 334779
Email: [EMAIL PROTECTED]
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RE: [Asterisk-Users] how to dial a h323 destination ?
Try to add gwprefix in oh323.conf after your alias. You must know that you can configure * gw in gnugk.ini or in oh323.conf. I recommend you put in your oh323.conf. srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Thomas Haeger Enviado el: martes, 23 de septiembre de 2003 12:07 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] how to dial a h323 destination ? Here is my oh323.conf ... ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; ; Port to listen to. ; Default value is 1720. ; listenPort=1720 ; ; Port to connect to. ; (Used only when we don't have a gatekeeper) ; Default value is 1720. ; connectPort=1720 ; ; Enable fast start (yes,no). ; fastStart=yes ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=yes ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=none ; ; Set the maximum number of inbound/outbound H.323 connections. ; outboundMax=50 inboundMax=10 ; ; Set tracing options for the H.323 wrapper library. ; libTraceFile can be 'stdout' or a full path name to a logfile ; libTraceLevel=3 ;libTraceFile=stdout libTraceFile=/var/log/asterisk/oh323.log ; ; Disable gatekeeper or specify a gatekeeper. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; gatekeeper's DNS name, ; gatekeeper's ip, ; GKID:gatekeeper's id ; gatekeeper=80.86.166.196 ;gatekeeper=DISABLE ; ; Set the gatekeeper password ; ;gatekeeperPassword=secret ; ; Set the mode for sending user-input ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; userInputMode=Q931 ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Set the default context of H.323 calls. ; context=voipout ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] alias=BER-BER-GW-1 ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ;alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; ;context=all-aliases ;alias=ASTERISK ;alias=666 ; ; Aliases/prefixes routed in more-aliases context. ; ;context=more-aliases ;alias=665 ; ; Aliases/prefixes routed in all-prefixes context. ; ;context=all-prefixes ;gwprefix=00 ;gwprefix=01 ; ; Aliases/prefixes routed in more-stuff context. ; ;context=more-stuff ;alias=664 ;gwprefix=02 ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every codec option may have a frames option ; associated with it. ; Valid values for the codec option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G728- G.728 ; G729- G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft GSM Audio Capability ; LPC10 - LPC-10 ; Number of frames in RTP packet (if not specified) is 1. ; ;codec=G729A ;frames=20 ;codec=GSM0610 ;frames=4 ;codec=G7231 ;frames=20 ;codec=G72316K3 ;codec=G72315K3 ;codec=G7231A6K3 ;codec=G7231A5K3 codec=G711A ;codec=G711U ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] how to dial a h323 destination ?
When you register your * gw in gatekeeper you must say to gatekeeper which are the number that you must redirect to your * gw. For example, if you dial 555xx, you input in your oh323.conf must be like this: [register] alias=BER-BER-GW-1 Gwprefix=555 Regards, srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Thomas Haeger Enviado el: martes, 23 de septiembre de 2003 12:36 Para: [EMAIL PROTECTED] Asunto: AW: [Asterisk-Users] how to dial a h323 destination ? What is the gwprefix ? I try to connect the gk directly from our * gw -Ursprungliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Auftrag von Sergio Serrano Revuelto Gesendet: Dienstag, 23. September 2003 12:27 An: [EMAIL PROTECTED] Betreff: RE: [Asterisk-Users] how to dial a h323 destination ? Try to add gwprefix in oh323.conf after your alias. You must know that you can configure * gw in gnugk.ini or in oh323.conf. I recommend you put in your oh323.conf. srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Thomas Haeger Enviado el: martes, 23 de septiembre de 2003 12:07 Para: [EMAIL PROTECTED] Asunto: RE: [Asterisk-Users] how to dial a h323 destination ? Here is my oh323.conf ... ; ; Configuration file of OpenH323 channel driver ; ;- ; General configuration options ; (ports, jitter, GK, ...) ;- [general] ; ; Address to bind to for incoming connections. ; Default is ALL. ; listenAddress=0.0.0.0 ; ; Port to listen to. ; Default value is 1720. ; listenPort=1720 ; ; Port to connect to. ; (Used only when we don't have a gatekeeper) ; Default value is 1720. ; connectPort=1720 ; ; Enable fast start (yes,no). ; fastStart=yes ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=yes ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=yes ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...1). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=none ; ; Set the maximum number of inbound/outbound H.323 connections. ; outboundMax=50 inboundMax=10 ; ; Set tracing options for the H.323 wrapper library. ; libTraceFile can be 'stdout' or a full path name to a logfile ; libTraceLevel=3 ;libTraceFile=stdout libTraceFile=/var/log/asterisk/oh323.log ; ; Disable gatekeeper or specify a gatekeeper. ; Valid values for this option are: ; DISABLE, ; DISCOVER, ; gatekeeper's DNS name, ; gatekeeper's ip, ; GKID:gatekeeper's id ; gatekeeper=80.86.166.196 ;gatekeeper=DISABLE ; ; Set the gatekeeper password ; ;gatekeeperPassword=secret ; ; Set the mode for sending user-input ; Valid values for this option are: ; Q931- Q.931 Keypad Information Element ; STRING - H.245 string ; TONE- H.245 tone ; RFC2833 - RFC2833 ; userInputMode=Q931 ; ; AMA flags (default, omit, billing, documentation) ; amaFlags=default ; ; Account code ; accountCode=H323 ; ; Set the default context of H.323 calls. ; context=voipout ;- ; Configure H.323 aliases, prefixes and ; related ASTERISK's contexts ;- [register] alias=BER-BER-GW-1 ; ; Aliases/prefixes associated with the default context ; defined in section [general]. ; ;alias=asterisk ;alias=123 ; ; Aliases/prefixes routed in all-aliases context. ; ;context=all-aliases ;alias=ASTERISK ;alias=666 ; ; Aliases/prefixes routed in more-aliases context. ; ;context=more-aliases ;alias=665 ; ; Aliases/prefixes routed in all-prefixes context. ; ;context=all-prefixes ;gwprefix=00 ;gwprefix=01 ; ; Aliases/prefixes routed in more-stuff context. ; ;context=more-stuff ;alias=664 ;gwprefix=02 ;- ; Specify and configure CODEC related ; options ;- [codecs] ; ; Define the codec list of the channel driver. ; Every codec option may have a frames option ; associated with it. ; Valid values for the codec option are: ; G711U - G.711 u-Law ; G711A - G.711 A-Law ; G7231 - G.723.1(6.3k) ; G72316K3- G.723.1(6.3k) ; G72315K3- G.723.1(5.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G7231A6K3 - G.723.1A(6.3k) ; G728- G.728 ; G729- G.729 ; G729A - G.729A ; G729B - G.729B ; G729AB - G.729AB ; GSM0610 - GSM 0610 ; MSGSM - Microsoft
RE: [Asterisk-Users] ISDN BRI hardware
You can try AVM FRITZ with chan_capi from kapejod. srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de YO Internet Information Enviado el: lunes, 22 de septiembre de 2003 0:03 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] ISDN BRI hardware We sell: AVM B1 for development environment Eicon Diva Server BRI card for live system (on-board echo canceller) Tan www.telappliant.com - Original Message - From: Mark Hagler [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, September 21, 2003 10:43 PM Subject: [Asterisk-Users] ISDN BRI hardware Hi, Anybody have lots of experience with PCI ISDN cards and Asterisk? I'm thinking of getting a BRI in my house to deliver more advanced signaling to my PBX (yes, I'm a geek :-)) but I've never played with isdn4linux. Is there any particular BRI card that works better with Asterisk than any other? Also, can the BRI interface cards participate in conference, etc., since they aren't a Zaptel interface? Thanks, M ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Registration NOTIFY EVENT
Hi all, when I try register my netergy SIP Phone with *, I can't do it due to the next message: 1 headers, 2 lines Reliably Transmitting: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK3438300a From: asterisk sip:[EMAIL PROTECTED];tag=as34fa433f To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 37 Messages-Waiting: yes Voicemail: 1/2 (no NAT) to 192.168.0.155:5060 Sip read: SIP/2.0 405 Method Not Allowed Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as34fa433f To: sip:[EMAIL PROTECTED] CSeq: 102 NOTIFY Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK3438300a Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS,PRACK Content-Length: 0 NOTIFY meesage is nos supported by asterisk? Anyone can help me? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP stage
Title: Mensaje
Hi,
I
would like to configure a stage for SIP phones. This stage would be the
next:
two
netergy SIP phones connected to Asterisk through chan_sip.
one
X100P or AVM FRITZ to outside lines.
I
think that sip.conf would be the next:
;;
SIP Configuration for Asterisk;[general]port =
5060
; Port to bind tobindaddr =
192.168.0.207
; Address to bind tocontext =
outgoing
; Default for incoming
callsdisallow=allallow=alawmaxexpirey=3600
; Max length of incoming registration we
allowdefaultexpirey=120
; Default length of incoming/outoing registration
[704]type=friendusername=704;secret=704host=192.168.0.154dtmfmode=rfc2833mailbox=704callerid=704context=outgoingreinvite=yescanreinvite=yesqualify=yesnat=-1
[705]type=friendusername=705;secret=705host=192.168.0.155;defaultip=192.168.0.5dtmfmode=rfc2833mailbox=705callerid=705context=outgoingreinvite=yescanreinvite=yesqualify=yesnat=-1
And my
extensions.conf would be the next:
[outgoing]
exten=i,1,Playback(invalid)exten=t,1,Hungup()
exten=_7XX,1,Goto(SIP|${EXTEN}|1)exten=_X,1,ChanIsAvail(CAPI/951014943CAPI/951014944)exten=_X,2,SubString,CANAL=${AVAILCHAN}|12|9exten=_X,3,Dial(CAPI/@${CANAL}:B${EXTEN}|17)
[SIP]
exten=704,1,Dial(SIP/704|tTm)exten=705,1,Dial(SIP/705|tTm)
are
these files correct?
Why
hwen I try call from one phone to other only rings once and then
hungup?
Any
idea,
thanks,
srsergio
RE: [Asterisk-Users] SIP registration
I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration
Thanks, my phone has the next sip setting. Can you help me with correct parameters with the below sip.conf? SIP Server Settings * Server Address: (IP or FQDN) * Port: * Domain Name: * Send Registration Request: (true or false) Gateway Settings Dial Plan: Transport: (UDP tor TCP ) Phone Number: CallerID Name: Port: AEC: (On or OFF) User Name: Password: Thanks for all srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jan Janak Enviado el: viernes, 19 de septiembre de 2003 8:59 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SIP registration
Thanks, my phone has the next sip setting. Can you help me with correct parameters with the below sip.conf? SIP Server Settings * Server Address: (IP or FQDN) * Port: * Domain Name: * Send Registration Request: (true or false) Gateway Settings Dial Plan: Transport: (UDP tor TCP ) Phone Number: CallerID Name: Port: AEC ON: (On or OFF) User Name: Password: Thanks for all srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Jan Janak Enviado el: viernes, 19 de septiembre de 2003 8:59 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, I don't know if it is the problem, but the message below is syntactically invalid, there must be space between the name token in From and To (704) and the URI, i.e. correct From should look like this: From: 704 sip:[EMAIL PROTECTED];tag=230b0-e0 instead of this: From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 Jan. On 19-09 08:38, Sergio Serrano Revuelto wrote: I have the same problem, Asterisk debug is the next: REGISTER sip:AVANZADA7 SIP/2.0 Call-ID: [EMAIL PROTECTED] From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED] CSeq: 101 REGISTER Via: SIP/2.0/UDP 192.168.0.154:5060 Contact: sip:[EMAIL PROTECTED]:5060 Max-Forwards: 70 Expires: 1800 Supported: timer Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 192.168.0.154 : 5060 (non-NAT) Transmitting (no NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.0.154:5060 From: 704sip:[EMAIL PROTECTED];tag=230b0-e0 To: 704sip:[EMAIL PROTECTED];tag=as539680e1 Call-ID: [EMAIL PROTECTED] CSeq: 101 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 192.168.0.154:5060 DEBUG[12301]: File chan_sip.c, Line 835 (__sip_destroy): Destorying call '[EMAIL PROTECTED]' 10 headers, 0 lines Reliably Transmitting: OPTIONS sip:192.168.0.154 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc From: asterisk sip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Length: 0 (no NAT) to 192.168.0.154:5060 Sip read: SIP/2.0 200 OK Call-ID: [EMAIL PROTECTED] From: asterisksip:[EMAIL PROTECTED];tag=as6c232c12 To: sip:192.168.0.154 CSeq: 102 OPTIONS Via: SIP/2.0/UDP 192.168.0.207:5060;branch=z9hG4bK160c3fcc Supported: timer Allow: REFER,UPDATE,INFO,MESSAGE,OPTIONS Accept: application/sdp Accept-Encoding: Accept-Language: en;q=0.8 User-Agent: Netergy MicroElectronics Content-Length: 0 My sip.conf is the next: [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls disallow=all allow=alaw tos=lowdelay [704] type=friend username=704 secret=704 host=192.168.0.154 dtmfmode=inband mailbox=704 callerid=704 context=outgoing reinvite=no canreinvite=no qualify=300 nat=1 ANY IDEA ABOUT THIS? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Hielke Christian Braun Enviado el: jueves, 18 de septiembre de 2003 19:05 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] SIP registration Hello, try to change [siptestphone] to [atrg613test] in sip.conf. Maybe that helps. Regards, Christian. On Thu, Sep 18, 2003 at 05:52:19PM +0200, Tais M. Hansen wrote: Hi, I'm having problems letting a SIP endpoint register at Asterisk. Here's the debug output from Asterisk: ... sip.conf: [general] port=5060 bindaddr=s.s.s.s context=cxnet-in tos=lowdelay [siptestphone] type=friend user=atrg613test host=dynamic defaultip=c.c.c.c ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_capi and poor voice quality
HI, I am probing chan_modem_i4l again with AVM FRITZ but I can hear nothing in phone outside of asterisk, I explain Phone 1-- AVM_FRITZ--Asterisk-- Phone 2 From Phone1 to Phone 2 I can hear, but From phone 2 to phone 1 I can't hera nothing. Any idea? srsergio -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Peer Oliver schmidt Enviado el: miércoles, 23 de julio de 2003 12:19 Para: [EMAIL PROTECTED] Asunto: Re: [Asterisk-Users] chan_capi and poor voice quality Peter Zeltins wrote: Calling * via SIP produces very good sound. Calling * via the chan_capi produces horrible sound. However, if I dial 500 in the demo menu to connect to the IAX at digium the sound is good again. ie: ISDNCall-AVM-B1-Card-Asterisk = All prompts sound horrible SIP-Asterisk = Prompts are good Stupid question... why don't you use I4L instead of chan_capi? I've wanted to use chan_capi myself but due to lack of time haven't been able to get it running yet. However I4L produces good audio quality, although I miss extended ISDN features... I'm using AVM Fritz PCI card My reason for chan_capi are messages in the mailing list suggesting a better quality with the chan_capi driver. :( So, you are saying the voice quality you experience with the Fritz PCI card is satisfactory? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR question
Hi, I would like to know how suppress number for outside dialling in CDR table. For example, if I need press 9 key to make an outside call, I would like that the number in dst field in cdr table was the outside number without 9 key. It's possible? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two Question
Hi, I would like to know how do two things. First, it is possible simulate PBX scenary?, I explain. I would like that when an user press 9(outgoing key) asterisk will generate a new dial tone in H.323 EP and then the user could press number for dial. Second, it's possible modify time interdigit. If I doubt in a number to dial, asterisk has a timeout very short and it starts to dial some early. Any ideas? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Two Question
Hi, I would like to know how do two things. First, it is possible simulate PBX scenary?, I explain. I would like that when an user press 9(outgoing key) asterisk will generate a new dial tone in H.323 EP and then the user could press number for dial. Second, it's possible modify time interdigit. If I doubt in a number to dial, asterisk has a timeout very short and it starts to dial some early. Any ideas? Thanks in advance, srsergio ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
