Re: [Asterisk-Users] Can not build zaptel with kernel-2.6.12
harry gaillac wrote: Hello, I try to compile zaptel . I installed kernel-sources but when i run : make linux26 / serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL _CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL _CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm ir You do not appear to have the sources for the 2.6.12-1-386 kernel installed. make: *** [linux26] Error 1 // I have to ask the obvious question. Do you have the same source as you have kernel running? Remember if you have run an upgrade it could have updated the kernel but may not have doen the sources and if you have the sources from the installion media then you would have different versions that will cause this exact problem. David Something don't match in makefile with debian sarge 3.1 here linux26: prereq $(BINS) @echo $(KSRC) @if [ -z $(KSRC) -o ! -d $(KSRC) ]; then echo You do not appear to have the sources for the $(KVERS) kernel installed.; exit 1 ; fi $(MAKE) -C $(KSRC) SUBDIRS=$(PWD) modules Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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
Re: [Asterisk-Users] Asterisk @ Home password recovery
kchase wrote: Can anyone tell me how to recover an Asterisk password if it is forgotten or do I have to do a complete re-install. Which password? Do you have the SSH password for the [EMAIL PROTECTED] server? David Kris ___ --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
Re: [Asterisk-Users] Asterisk @ Home password recovery
kchase wrote: the password to login into asterisk If you still have the SSH password you can log into the box and change the maint password. [EMAIL PROTECTED] ~]# help-aah [EMAIL PROTECTED] - HELP CommandsDescriptions --- config set the local time zone and keyboard type netconfig configure ethernet interface genzaptelconf autoconfig Zaptel cards bundle-crm packup CRM to run on another box restore-aah restore from a backup install-AVMB1ISDN install support for AVB B1 ISDN card install-EiconDiva install support for Eicon Diva ISDN card install-pdf installs support for emailing PDFs of faxes passwd-maintset master password for web GUI passwd-amp set password for amp only passwd-meetme set password for Web MeetMe only passwd set root password for console login passwd adminset admin password for checking system mail setup-cisco create a SIPDefault.cnf in /tftpboot setup-dhcp set up a dhcp server rebuild_zaptel rebuild zaptel driver after kernel update asterisk -r Asterisk CLI yum -y update Get latest patches for CentOS You would run passwd-maint Hope that helps. David - Original Message - From: David Uzzell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, November 16, 2005 3:11 AM Subject: Re: [Asterisk-Users] Asterisk @ Home password recovery kchase wrote: Can anyone tell me how to recover an Asterisk password if it is forgotten or do I have to do a complete re-install. Which password? Do you have the SSH password for the [EMAIL PROTECTED] server? David Kris ___ --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] g729 status in New Zealand
trixter aka Bret McDanel wrote: On Wed, 2005-11-16 at 13:22 +1300, Richard Malcolm-Smith wrote: As far as I was aware a license was only required in contries that had software patents, I know that there arnt here so I am just seeking clarification if thats all there is to it. I say just all get over it. The lic cost is Cheap. I did not want to have to buy lic for the codec but I did when I had my server in a Data Centre. Now I have it at home on my DSL I am not even using the g729 codec anymore cause lan bandwidth is not a issue for me. I don't even have it loaded cause I don't want to waste one of the MAC changes cause I am in the process of getting a new server. It is not as if the Lic Fee is $100 per lic or anything that nasty, it is $10 per channel. If that is used for your personal use then you would likely own need a couple so it is not expensive. And if it is for business and you need many and you can't afford to cover the cost's of the codec then you shouldn't be in business. I mean just all get over it. Been and DONE to death. Enjoy the cheap codec lic for what it is. David This issue was beaten to death before, software patents are not the only issue, while it is true that this software is covered under a patent the application is written broad enough to say that its an algorithm or device not just software. Or so the people who misunderstood me kept yelling towards me. There are some countries where you can only patent a physical device that exists and then only that device. As the codec doesnt qualify in those countries specifically, the patent is not valid there. Those countries specifically forbid the patenting of mathematical algorithms, etc. This was the issue that many people misunderstood or refused to accept and tried to turn the discussion into whether or not one was deserved rather than is it enforcable in a specific location. Be warned you opened a can of worms by asking this question. In short I dont know if patents in new zealand cover nonexistant devices, theories, algorithms, etc. But you may want to broaden your search from just software patents to what types of patents are allowed and whether algorithms and other types are patentable in wherever anyone who reads this happens to reside. Regardless digium has no way of giving out the g.729 codec without a license so you will have to look elsewhere to get one for asterisk. There is a patch file to use the intel code. You have to go thruogh a long drawn out process to get the code from intel, registering, agreeing to the license terms blah blah blah. Once you get it its fairly straight forward to apply the patch. The hardest part is getting the code. The patch is released under the GPL but by doing that the patch authors violate the GPL so in effect there is no license for that software. You cant tie a gpl product to something that is licensed contrary to the gpl, that specifically is in their faq. If you know where to look there is another option out there that doesnt use either method, but I have doubts about how legal that one is, so I will not comment on that. ___ --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
Re: [Asterisk-Users] g729 status in New Zealand
trixter aka Bret McDanel wrote: On Wed, 2005-11-16 at 13:21 +1100, David Uzzell wrote: trixter aka Bret McDanel wrote: On Wed, 2005-11-16 at 13:22 +1300, Richard Malcolm-Smith wrote: As far as I was aware a license was only required in contries that had software patents, I know that there arnt here so I am just seeking clarification if thats all there is to it. I say just all get over it. The lic cost is Cheap. I did not want to have to buy lic for the codec but I did when I had my server in a Data Centre. Now I have it at home on my DSL I am not even using the g729 codec anymore cause lan bandwidth is not a issue for me. I don't even have it loaded cause I don't want to waste one of the MAC changes cause I am in the process of getting a new server. well mac based auth is trivial to bypass, many drivers on many systems support mac address changing. But that isnt the point, it may be cheap for small users, but look at vonage who has passed 1 million customers. At $10/license lets say that 7% of their users are on the phone at any given time using g.729 (ok that is prolly high but ...) that is $700k. Lets say that vonage uses a VoIP provider (like global crossing provides) for most of their routes (I know they have some pris in NJ but its unclear what percentage they use those). That doubles the cost to $1.4M. I understand that is a high estimate and would probably be less than half that, so lets go ahead and fudge the math a bit and say they need $500k worth of licenses (25k calls at a time using G.729 for a 1 million customer user base). In the grand scheme of things odds are they wouldnt do this because of the cpu overhead required, but that isnt what most people would consider 'cheap'. Looking at a hobby system or a small VoIP provider it may be cheap but when you think about some of the larger providers it can be a daunting cost. That however isnt the point. the question asked was whether or not its legally enforcable in a given jurisdiction. Ok I will give you that. If it is not enforcable in NZ thats great. Get a copy of the code and build your own codec. If you want to use digium's codec then you have to pay the lic fee even if the patent is not enforcable for you, thats the way the world goes. They have a lic'ed product and only offer that product for sale not as a free product. The question is really un-important for this list, it is ONLY important between the person who thinks that they can use the g729 codec ignoring the patent or considering that it is not legally enforcable for them and their lawyer who will give them concise information about the legal situation in their jurisdiction. It does however not make any difference that for instance vonage would likely use a huge number of lic's other than if they were in a jurisdiction were the patent was legally unenforcable as far as they see it. As I also understand it that people can negotiate with the patent owners directly if they wish to get a cheaper bulk price than what digium is offering to the general public. They can always speak to http://www.voiceage.com/ and work through a large valume pricing structure if it is unavaliable with digium. If you think the codec patent is legally unenforcable for you then go for it get a copy of the code and build your own codec but when it doesn't work or not work the way you see it working else were sorry your on your own. If you buy a lic then you can get support to make sure it works as it should. Round and round we go back to the same place. If a PERSON considers the codec patent legally unenforcable in their jurisdiction then they must make that choice for themselves and going round and round in circle's on this mailling list you will always end up back at the same place over and over again. GO SEE A LAWYER IN YOUR OWN JURISDICTION THAT KNOWS ABOUT THE LAWS THAT EFFECT YOU DIRECTLY AND PAY THEM TO PROVIDE YOU SOUND ADVICE IN THE SITUATION FOR YOU. Trying to change it from that to something else is a questionable tactic, especially when you try to do so by confusing a small hobby system or a small time provider with a larger one. It is not as if the Lic Fee is $100 per lic or anything that nasty, it is $10 per channel. If that is used for your personal use then you would likely own need a couple so it is not expensive. And if it is for business and you need many and you can't afford to cover the cost's of the codec then you shouldn't be in business. See above $500k isnt that common in spare cash. Hell even 10% of that isnt that common in a startup in spare cash. Remember all the other costs would have been accounted for for them to be in business, so that would be extra on top of all those other costs. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk
Re: [Asterisk-Users] IAX only speech one way
Mir wrote: Thanks for your suggestion. Unfortunately, it didnt change anything, A can still not hear B, but B can hear A, strange.. I had the same problem with one of my IAX providers in AUS. Both ends turned of trunking and all was fine with the world again. Not sure what was the cause but that was my solution for EXACTLY the same problem that you explain. David Michael 2005/10/19, Rich Adamson [EMAIL PROTECTED]: I have two Asterisk's connected via IAX, they are sitting on the same network, via a VPN, so there should be no problems with firewalls. My problem is that when a person calls from A to B, A will not hear B speak. B hears A fine. I doesn't matter who initiates the call. One of the Asterisk'ses is a new installation, just installed, but with the Conf-files from an earlier setup, that worked fine. Asterisk version on computer A is Asterisk CVS-v1-0-12/09/04-08:58:31 Asterisk version on computer B is Asterisk CVS-D2005.05.28.22.00.00-10/17/05 Two different versions, but I dont think it should matter? Not sure this applies, but I was having the same problem with teliax.com and turning off the jitterbuffer in iax.conf fixed the problem. Kind of looks like we are running two different versions of asterisk as well, but I'd suspect that teliax has modified their system for other business purposes. Try jitterbuffer=no and see if it helps. Rich ___ --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 ___ --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] Upgrade cause's no Audio on IAX
Ok I upgraded tonight a server from CVS in Late NOV to one just downloaded tonight. It all runs up OK and I can contact it from my ATA 186 using g729a codec and that all works fine. What I am having trouble with is connecting through IAX ATP.org.au in AUS to my server. The connection comes through OK I can see all the tracking info in the console OK but I get 0 audio in either direction. Does anyone know what would have changed to cause this or what I would need to do to look at solving the issue ? I am now offline :( and for some reason rolling back to the older version now does not want to run :( My IAX conf [general] tos=lowdelay jitterbuffer=no disallow=all allow=speex allow=ilbc allow=gsm register = user:[EMAIL PROTECTED] [guest] type=user context=default auth=none [2347] type=friend username=user secret=password auth=md5 host=gw1.austechpartnerships.com context=default trunk=yes qualify=3000 disallow=all allow=ilbc Thanks David ___ 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] Upgrade cause's no Audio on IAX
Further to this I have done a full reinstall of everything including ztdummy and asterisk to the CVS version downloaded yesterday. I get a Loud Buzzing when the line answers now and leaving a voicemail mesg just leaves blank :( So I am thinking as it appears that the ATA works correctly with g729a that it would be either IAX2 problems or iblc codec problems :( Has anyone got any advice? Thanks David David Uzzell wrote: Ok I upgraded tonight a server from CVS in Late NOV to one just downloaded tonight. It all runs up OK and I can contact it from my ATA 186 using g729a codec and that all works fine. What I am having trouble with is connecting through IAX ATP.org.au in AUS to my server. The connection comes through OK I can see all the tracking info in the console OK but I get 0 audio in either direction. Does anyone know what would have changed to cause this or what I would need to do to look at solving the issue ? I am now offline :( and for some reason rolling back to the older version now does not want to run :( My IAX conf [general] tos=lowdelay jitterbuffer=no disallow=all allow=speex allow=ilbc allow=gsm register = user:[EMAIL PROTECTED] [guest] type=user context=default auth=none [2347] type=friend username=user secret=password auth=md5 host=gw1.austechpartnerships.com context=default trunk=yes qualify=3000 disallow=all allow=ilbc Thanks David ___ 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] extension.conf dialplan
Kamran Ahmad wrote:
hi
any one tell me how to make a dialplan
my extensions.conf
exten = _40,1,Dial(OH323/${EXTEN})
i want to dial to 40 number.
could be any number like 923335224005 or
92512213248
at the moment when i am trying to dial 40923335224005
asterisk is dialing
Executing Dial(OH323/R11429, OH323/40923335224005)
but i want him to dial
Executing Dial(OH323/R11429, OH323/923335224005)
exten = _1,1,Dial(OH323/${EXTEN:1})
exten = _10,1,Dial(OH323/${EXTEN:2})
The first one removes the first number and the second one removes the
first 2 chars. and so on, and so on.
Hope that helps with examples.
David
Kamran Ahmad
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Re: [Asterisk-Users] Hi there..
Bharat M. Sarvan wrote: Hello Everybody, This is Bharat here. I am on the way of learning Asterisks, and I just wished to know how I go about if got to write dailplans for outbound calls and inbound calls. If you could provide me with a simple example, I could get thru. Waiting for your response If you go to http://www.voip-info.org/tiki-index.php and search for extensions you will find exactly what you are in need off. David Regards Bharat M. Sarvan ___ 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] Do you need to recompile the Linux 2.6 kernel for zaptel modules?
Geoff Nordli wrote: Hi Everyone. On the Linux 2.6 kernel do I need to recompile the kernel in order to compile the zaptel modules? Using Mandrake 10.1 which runs standard a 2.6 kernel I did not have to recompile the kernel to get them working. cheers, David Thanks, Geoff ___ 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] Kernel 2.4 or 2.6 for the latest asterisk ??
Brett, Gary wrote: Hi there Just a quick question. I have been playing around with asterisk CVS-1.0.02 on fedora core 1 (2.4 kernel) and I would like to have a look at asterisk v 1.0.6 but am still a little uncertain which linux kernel is best to run on ?, can I use Fedora Core 3 (is it the preferred kernel) or should I stick with FC1 Ps - the only additional hardware in the box will be a digium single port E1 I would have said that if you were not going to be running any hardware zap devices the 2.6 kernel option would be the best as you don't need any special motherboard to get ztdummy to run. As you have hardware, I would say that the kernel version would be dependant on which Distro you wanted to use and what kernel that comes with that or if you want to compile a kernel. I have several * servers all without zap devices running on Mandrake 10.1 which is a 2.6 kernel and I was happy with it. so thats my opion and personal choice. David Any advice would be greatly appreciated Gary ___ 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] Extentions Variable Dialing QUESTION.
I have this exten code to dial out to a specific group of numbers.
exten = _10X,1,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:1})
I want to be able to have one line to handle generic calling so we could
dial say 10 or 10X or 10XXX or something in between.
Does any one know what the exten line would be to be that generic or
point me to something that would explain it?
Thanks
David
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[Asterisk-Users] g729 Lic ordered from Digium Question.
Does anyone know how long the orders take? I ordered some a couple of days ago and it said normally 24hours, and I am guessing that the weekend cause's some delays but it did not say anything abouy that. Any one got any ideas on how long generally over the weekend it takes? Thanks David ___ 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] ATA 186 Codec Question.
I have seen the list of codecs for the ATA 186's but not sure if it was 100% or not. I want to know really is it possible to run GSM or ilbc on them or is a G729 lic the only way to get a low bandwidth codec? This is the list of codecs that I have seen. RxCodec and TxCodecConfigure the codec ID. * G.723.1Codec ID 0 * G.711aCodec ID 1 * G.711ucodec ID 2 * G.729acodec ID 3 Thanks David ___ 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] Possibility of getting someone to delete a user from the list???
Martijn van Oosterhout wrote: On Mon, Feb 28, 2005 at 10:04:27AM +0100, Dave Cotton wrote: On Sun, 2005-02-27 at 17:54 -0500, Robert Webb wrote: This is getting VERY annoying. Is there anyone in here that has access to the list administration to delete the user below??? Pray tell me why. The list isn't being flooded by these messages as far as I see. Their mailserver is broken in that it sends bounces to the From address (ie the person who sent the email) rather than the Sender (the asterisk mail server). So you only get an message from them when you send something. That is, one email for *every* message you send. There's also a server somewhere sending each message back to me with this attached: --- Spam detection software, running on the system zeus.avanzada7.com, has identified this incoming email as possible spam. The original message has been attached to this so you can view it (if it isn't spam) or label similar future email. If you have any questions, see the administrator of that system for details. --- However, the content analysis tells me the score is 0.1 of the necessary 5.0. Unfortunatly it's not helpful enough in determining the email address with the problem. Well I must be lucky cause I don't get any of these types of things from the list in quiet a while. So I must be lucky somehow. David Have a nice day, ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoIP/Asterisk presentation
Duane wrote: For those interested, I'm giving a talk about VoIP/enum.164/asterisk tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS build #2, 4th floor, room 10. Sorry for the late notice, it didn't occur to me that there might be people on this list interested and able to attend etc... I'd have been there like a flash but late notice was the problem :( And to think I was in the city all day today and did not leave till late! I could have stayed in there and been there :( Oh well next time. David ___ 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] multiple sip phones behind firewall
Paul P. Pongco wrote: Hello List, Can you please point me to the right resources on making multiple sip phones behind a firewall w/ private address work with asterisk w/c is on a public network. I have seen STUN on the grandstream and Xtunnels on X-lite. What is most deployed by members here with similar setups? Thanks. My sugestion would be if you have more than a couple of phones behind this one firewall, why wouldn't you run a very basic * server to coonect all those SIP phones to and then IAX to the external * server. This does not have as much issues with the firewall and then you could possibly take advantage of some of the bandwidth saving features in IAX for your connection. Just a thought and I don't know how that will work with your operation but that is what I thought might be an Option. David Uzzell ___ 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] Getting speex to work
Jonathan Lin wrote: Hi All, I am having trouble getting speex to work on asterisk. I downloaded 1.0.4 from speex.org, download libogg from vorbis. ./configure, make and make install for both and then recompile my asterisk 1.0.5, yet I am getting error when trying to load codec_speex.so. Below is the error from the log 18:29:04 WARNING[27687]: libspeex.so.1: cannot open shared object file: No such file or directory I can see libspeex.so.1 in /usr/local/lib and it's symbolic linked to libspeex.so.1.2.0 so the only thing I can think of is the permission. I changed the permission to 777 for libspeex.so.1.2.0 just for testing but it's still crashing. Has anyone encounter this problem or maybe point me in the right direction for debugging this? Jon Lin When you recompile * do you make clean first? I had some issues when I first started out sometimes after compiling up in the first time and then recompiling without a make clean it did not always pick up the changes. David ___ 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] * Mobile Phone Mobile Network
Ok I have a question. Seen it come and go around the mailling list for a while but never really seen an answer that seems to sort it out. What is needed is some interface from * Mobile Phone Mobile Network Service. At this point all the providers in AUS that I have found are charging a Premium Rate for Land Line Mobile Network services. What I would like to do is be able to purchase a low rate Mobile SIM that I can chuck into a Mobile Phone and have it setup so that I route the Mobile calls through it. Rembering that most if not all mobile phones can be accessed via RS232 interface. Anyone done this or seen it done or know how to do it using * and whatever? Cheers David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zaptel Needed
[EMAIL PROTECTED] wrote: Hello All, Can someone please tell me about Zaptel? Is it only needed if you are going to have an interface card like TDM400P installed on the Asterisk server? Do you really need it if you do not have the interface card? You don't need to load Zaptel if your not going to run any cards but if your going to use IAX and/or meetme you will need some form of timming and that comes from zaptel if you have hardware installed or ztdummy if you don't have any hardware installed. http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer%20ztdummy ztdummy is a Linux kernel module that will provide your Asterisk with a Zaptel timer even if you haven't got any Digium hardware installed in your Linux server. cheers, David Thanks, Lonnie ___ 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] ATA's
Shaun Ewing wrote: I'm using the ATA 186 and think it's great. The latest firmware also changes the web interface - it's similar to the 7905G/7912G phones now. Did you have much trouble getting the ATA 186 working? I have one running Version: v3.1.0 atasip (Build 040211A) I have it setup and it does poll the * server but does not work to use and errors in sip. Followed the instructions on the wiki page for them and it still wants to be a pain :( Other problem is that it is in Denmark and I am in AUS :) so timming is an issue. Any advice would be appreciated. David Uzzell This is the sip debug from * end. Sip read: REGISTER sip:203.29.98.221 SIP/2.0 Via: SIP/2.0/UDP 62.79.110.156:5060 From: Test901 sip:[EMAIL PROTECTED];user=phone;tag=4212316678 To: Test901 sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER Contact: Test901 sip:[EMAIL PROTECTED]:5060;user=phone;transport=udp;expires=120 User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A) Content-Length: 0 9 headers, 0 lines Using latest request as basis request Sending to 62.79.110.156 : 5060 (NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 62.79.110.156:5060;received=62.79.110.156;rport=5060 From: Test901 sip:[EMAIL PROTECTED];user=phone;tag=4212316678 To: Test901 sip:[EMAIL PROTECTED];user=phone;tag=as560861ed Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 62.79.110.156:5060 Transmitting (NAT): SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 62.79.110.156:5060;received=62.79.110.156;rport=5060 From: Test901 sip:[EMAIL PROTECTED];user=phone;tag=4212316678 To: Test901 sip:[EMAIL PROTECTED];user=phone;tag=as560861ed Call-ID: [EMAIL PROTECTED] CSeq: 1 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=pbx.unifiedau.net, nonce=4a523e7e Content-Length: 0 to 62.79.110.156:5060 Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Destroying call '[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
Re: [Asterisk-Users] Linphone / Kphone / lipz4
Ralph Green, Jr. wrote: On Mon, 2005-02-14 at 13:08 +0100, Jens Kübler wrote: Maybe you wanna check out the softphone zip4x5 made by Zultys. It's the software which is used by the same hardphone. Howdy, Do you use this product and do you have any relationship with Zultys? It looks interesting, but it is documented to support only old RedHat versions and they don't release source to let me recompile. I am not a big RedHat fan, but if I have to use it on the desktop, I would want something newer than RedHat 9. If you can tell me you are using it with a newer distro, that would help. Well I can tell you that it won't work with mandrake 10.1 and it does not work. :( Wish I could get source and compile it up as it looks like it may be a nice softphone. David Have a good day, Ralph ___ 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] Meetme
Nitesh Divecha wrote: Hey All, Just finished installing Asterisk and configured all the necessary parameters to start. I cant seem to find the Meetme application in my asterisk directory. I downloaded asterisk from CVS and installed it and all my Snom phones are working and voicemail too. I am getting error: - Feb 11 17:10:19 WARNING[13042]: pbx.c:1280 pbx_extension_helper: No application 'Meetme' for extension (sip, 5557, 1) == Spawn extension (sip, 5557, 1) exited non-zero on 'SIP/phone1-f88d' Do I need zaptel to be installed? http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe Please observe * The MeetMe application needs a timer to work. There are different ways to get the timer to work, but it won't work by default if you haven't got a Digium Zaptel hardware interface card installed. At this time only zaptel devices may be used. If you do not have a Zaptel device see the ztdummy instructions for timing. Your problem could be different but that answers you Question Exactly. Hope that helps. David Any help will be appreciated. Nitesh ___ 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] RE:mandrake linux install of zaptel
Jens Vagelpohl wrote: On Feb 11, 2005, at 16:28, [EMAIL PROTECTED] wrote: Extreme N00b, I am getting the error message a target does not exist when running the make install inside the zap directory, probably pretty common, possibly a package I didn't install, just need some insight on it. The same occurs with the libpri and asterisk. Would be good for some more info. But just of the top of my head you might want to check out the www.voip.info.org wiki and look under mandrake cause it has some instructions specific to the 2.6 kernel. And there are also some very complete instructions on there for complete installs from nothing through all the additional software required and *. That might be a good place to start. David I think everyone would appreciate if... - you wrote a new mail instead of highjacking an existing thread by answering it and replacing the subject line - you would not keep 5 miles of completely unrelated stuff in your email message - you could provide a better problem description that includes specific error messages and message stacks. Thanks! jens ___ 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] Voicemail timeouts after 30sec's everytime.
Ok I have a challange that I can't seem to find a way to fix it. My Voicemail in * timesout after 30secs without fail everytime no matter what I do. I have incomming calls comming in through Freshtel IAX2, if it goes to SIP extension when it is online it can hang on for what ever time the call goes for. If however it goes to the Voicemail it will timeout at 30sec and I can't seem to figure out why it just hangs up the call. From below is my voicemail.conf were I have tried everything I can think of from making the maxmessage large and turning off silencesuppresion but nothing changines the fact that it will time out at 30sec's. Does any have any ideas as to why this would do this and how I could go about correcting the issue? Thanks. David [general] format=wav49|gsm|wav [EMAIL PROTECTED] attach=yes maxmessage=360 minmessage=1 maxgreet=60 maxsilence=0 maxlogins=3 sendvoicemail=yes review=yes [zonemessages] eastern=Australian/Sydney|'vm-received' Q 'digits/at' IMp [default] ;mailboxs follow here. ___ 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] Voicemail timeouts after 30sec's everytime.
Brian Dingman wrote: This is just a guess, but try an Answer before sending it to VM. Hmm ok not sure what that would do but I am willing to try anything at the moment. Here is the incomming from Extensions.conf [default] exten = 61290071091,1,Wait,1 exten = 61290071091,n,Answer exten = 61290071091,n,DigitTimeout,3 exten = 61290071091,n,ResponseTimeout,5 exten = 61290071091,n,Dial(SIP/800,60) exten = 61290071091,n,Waitexten exten = 61290071091,n,Playback,voicemail/default/801/unavail exten = 61290071091,n,Voicemail,801 exten = 61290071091,n,Goto,t|1 I wouldn't put another answer in there before the VM or would I? Thanks David On Tue, 08 Feb 2005 11:34:30 +1100, David Uzzell [EMAIL PROTECTED] wrote: Ok I have a challange that I can't seem to find a way to fix it. My Voicemail in * timesout after 30secs without fail everytime no matter what I do. I have incomming calls comming in through Freshtel IAX2, if it goes to SIP extension when it is online it can hang on for what ever time the call goes for. If however it goes to the Voicemail it will timeout at 30sec and I can't seem to figure out why it just hangs up the call. From below is my voicemail.conf were I have tried everything I can think of from making the maxmessage large and turning off silencesuppresion but nothing changines the fact that it will time out at 30sec's. Does any have any ideas as to why this would do this and how I could go about correcting the issue? Thanks. David [general] format=wav49|gsm|wav [EMAIL PROTECTED] attach=yes maxmessage=360 minmessage=1 maxgreet=60 maxsilence=0 maxlogins=3 sendvoicemail=yes review=yes [zonemessages] eastern=Australian/Sydney|'vm-received' Q 'digits/at' IMp [default] ;mailboxs follow here. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speex codec problem (unresolved ?)
Walter Klomp wrote: Hi, I'm sorry to bring this up again, but I have been googling forever and whatever solutions are offered don't work for me. I am using x-lite (the latest build) and trying to use Speex. When I do call from the x-lite to another SIP phone or PSTN (through Cisco gateway) My asterisk fills up with this message: WARNING[1007]: codec_speex.c:196 speextolin_framein: Out of buffer space The x-lite client can hear the remote end (SIP or PSTN call) quite clearly, but what comes from the X-Lite is completely garbled and mixed with DTMF tones. I had tried the registry fix (which only changes the magic number from 97 to 110 and apparently didn't do anything else), didn't work. After looking at the source I had also tried to increase the buffer size from 8000 to 16000, but that made other codecs (like lin_to_g729) choke, and I still had the problem... I like speex and would like to use it (as I find ilbc a bit too scratchy) I am running Asterisk CVS-HEAD-11/16/04-17:19:53 and speex-1.0.4 libraries on Gentoo Linux. The best sugestion that I can offer is that I saw the same problem and could not resolve it but after upgrading * to CVS after the 12/10 it went away. Never did find a solution and gave up looking as it solved it. It also fixed some SIP issues I had and they went away aswell. Sorry that might not be the answer you are looking for but thats what worked for me. David Can anybody help me further on how to resolve this problem ? Thanks Walter ___ 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] FXO to IAX on ethernet. or FXO to SIP on Ethernet
Now I have searched around and not seen anything to do this. I want to in remote locations were we need to have single or 2 PSTN lines for in dial as little hardware as possible and as stable as possible so that they will operate without user intervention. What I want to do is be able to take a single PSTN line in and go out through adsl for the Inet link. These would be in VERY remote locations like smaller towns so they would need to be simple, stable and require little to no user intervention after they are installed. Does anyone know of any hardware that will do this or a way that this could be done or ?? Thanks David ___ 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] parking.conf
mohammad wrote: Hi; I downloded asterisk CVS-HEAD 12/20/04 but I canot see parking.conf in /etc/asterisk. There is no longer any parking.conf. It is now know as features.conf and you can find all the info about it at http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20parking.conf David Appreciate any help Mohammad ___ 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] Hmm something strange.
I have been tring to send email to the List for the last day or so and have not seen it come back on the list and have not seen any reponse's to my email so I am unsure if it is making it to the list. And I have also been seeing some of the same emails over and over again. The list does not seem to have got back to 100% since it was down :( My mail server works fine I am not having those problems with other list's I am on like Hylafax list and others. :( Does anyone have anyideas? I can hope that this email gets to the list. David ___ 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] Lets try this again then! Q: SIP error from dialplan I suspect!
I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial(IAX2/[EMAIL PROTECTED]/3, SIP/800|5) in new stack Dec 21 00:15:57 NOTICE[3922]: app_dial.c:800 dial_exec: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time -- Executing WaitExten(IAX2/[EMAIL PROTECTED]/3, ) in new stack The Extensions.conf file for that section is exten = s,1,Wait,1 exten = s,n,Answer exten = s,n,DigitTimeout,3 exten = s,n,ResponseTimeout,5 exten = s,n,Dial(SIP/800,5) exten = s,n,Waitexten exten = s,n,Playback,voicemail/default/801/unavail exten = s,n,Voicemail,801 exten = s,n,Goto,t|1 and I have in sip.conf [800] type=friend regexten=800 username=800 secret=password callerid=800 host=dynamic ;dtmfmode=inband mailbox=800 nat=yes canreinvite=no qualify=yes disallow=all allow=gsm allow=speex allow=ilbc allow=ulaw allow=alaw ___ 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] Lets try this again then! Q: SIP error from dialplan I suspect!
Matt Hess wrote: is this current cvs or something? It looks completely abnormal for stable.. Ah sorry it is CVS 12/12/04 seems you are doing a lot of extra stuff you don't need to.. I'd see if just this works for you.. exten = 800,1,Dial(SIP/800,60) exten = 800,2,VoiceMail(800) Cool thanks for that. It creates the same error. :( -- Executing Dial(IAX2/[EMAIL PROTECTED]/3, SIP/800|60) in new stack Dec 22 17:22:10 NOTICE[6568]: app_dial.c:800 dial_exec: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time I have X-lite running and can call from one to the another sip so thats not a problem :( also.. why disallow all and then allow most everything? seems like you are trying to over think things.. no offense. None taken. The only reason I am doing it that way and I know it is not great for a productions system is that I am using a few different SIP phones both soft and hard to do testing. why not slim down your peer entry a bit? ie: [800] type=friend username=800 secret=password callerid=800 host=dynamic dtmfmode=inband mailbox=800 nat=yes canreinvite=no David Uzzell wrote: I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial(IAX2/[EMAIL PROTECTED]/3, SIP/800|5) in new stack Dec 21 00:15:57 NOTICE[3922]: app_dial.c:800 dial_exec: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time -- Executing WaitExten(IAX2/[EMAIL PROTECTED]/3, ) in new stack The Extensions.conf file for that section is exten = s,1,Wait,1 exten = s,n,Answer exten = s,n,DigitTimeout,3 exten = s,n,ResponseTimeout,5 exten = s,n,Dial(SIP/800,5) exten = s,n,Waitexten exten = s,n,Playback,voicemail/default/801/unavail exten = s,n,Voicemail,801 exten = s,n,Goto,t|1 and I have in sip.conf [800] type=friend regexten=800 username=800 secret=password callerid=800 host=dynamic ;dtmfmode=inband mailbox=800 nat=yes canreinvite=no qualify=yes disallow=all allow=gsm allow=speex allow=ilbc allow=ulaw allow=alaw ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lets try this again then! Q: SIP error from dialplan I suspect!
Sorry for replying to my own mesg but I have more info. David Uzzell wrote: Matt Hess wrote: is this current cvs or something? It looks completely abnormal for stable.. Ah sorry it is CVS 12/12/04 I have just this min downloaded the latest CVS as of about 20mins ago and compiled etc and the error is now gone. I am unsure what the error was but it must have been something to do with that CVS version. Thanks, now I just have to figure out the default dialplan :) David seems you are doing a lot of extra stuff you don't need to.. I'd see if just this works for you.. exten = 800,1,Dial(SIP/800,60) exten = 800,2,VoiceMail(800) Cool thanks for that. It creates the same error. :( -- Executing Dial(IAX2/[EMAIL PROTECTED]/3, SIP/800|60) in new stack Dec 22 17:22:10 NOTICE[6568]: app_dial.c:800 dial_exec: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time I have X-lite running and can call from one to the another sip so thats not a problem :( also.. why disallow all and then allow most everything? seems like you are trying to over think things.. no offense. None taken. The only reason I am doing it that way and I know it is not great for a productions system is that I am using a few different SIP phones both soft and hard to do testing. why not slim down your peer entry a bit? ie: [800] type=friend username=800 secret=password callerid=800 host=dynamic dtmfmode=inband mailbox=800 nat=yes canreinvite=no David Uzzell wrote: I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial(IAX2/[EMAIL PROTECTED]/3, SIP/800|5) in new stack Dec 21 00:15:57 NOTICE[3922]: app_dial.c:800 dial_exec: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time -- Executing WaitExten(IAX2/[EMAIL PROTECTED]/3, ) in new stack The Extensions.conf file for that section is exten = s,1,Wait,1 exten = s,n,Answer exten = s,n,DigitTimeout,3 exten = s,n,ResponseTimeout,5 exten = s,n,Dial(SIP/800,5) exten = s,n,Waitexten exten = s,n,Playback,voicemail/default/801/unavail exten = s,n,Voicemail,801 exten = s,n,Goto,t|1 and I have in sip.conf [800] type=friend regexten=800 username=800 secret=password callerid=800 host=dynamic ;dtmfmode=inband mailbox=800 nat=yes canreinvite=no qualify=yes disallow=all allow=gsm allow=speex allow=ilbc allow=ulaw allow=alaw ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Extensions SIP problems.
I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial(IAX2/[EMAIL PROTECTED]/3, SIP/800|5) in new stack Dec 21 00:15:57 NOTICE[3922]: app_dial.c:800 dial_exec: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time -- Executing WaitExten(IAX2/[EMAIL PROTECTED]/3, ) in new stack The Extensions.conf file for that section is exten = s,1,Wait,1 exten = s,n,Answer exten = s,n,DigitTimeout,3 exten = s,n,ResponseTimeout,5 exten = s,n,Dial(SIP/800,5) exten = s,n,Waitexten exten = s,n,Playback,voicemail/default/801/unavail exten = s,n,Voicemail,801 exten = s,n,Goto,t|1 and I have in sip.conf [800] type=friend regexten=800 username=800 secret=password callerid=800 host=dynamic ;dtmfmode=inband mailbox=800 nat=yes canreinvite=no qualify=yes disallow=all allow=gsm allow=speex allow=ilbc allow=ulaw allow=alaw ___ 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] On Australian News Sites : Open source software set to influence VoIP
http://www.itnews.com.au/newsstory.aspx?CIaNID=17357eid=1edate=20041221 Did not know how far and wide this was but I thought it might be of intrest to people on here. QUOTE The Asterisk PBX runs on Linux and provides three VoIP protocols. The software PBX provides voicemail services with directory, call conferencing and a host of additional telephony calling services. Its developers maintain that Asterisk can merge voice and data traffic seamlessly across disparate networks. Once it overcomes the stigma of being free, it should take off, said Pulver. END QUOTE David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Martin List-Petersen wrote: On Sun, 2004-12-19 at 02:11, David Uzzell wrote: Then the other thing if mem serves me you are running 2.6 kernel so why not run ztdummy? With the 2.6 kernel this does not require any specialist Hardware or anything! Sorry, but maybe you should have read his posts more thoroughly. ztdummy is not an option because of his chipset. He has usb-ohci. ztdummy requires usb-uhci. Umm yes it does on 2.4 kernel but on a 2.6 kernel it doesn't cause I am running it on a 2.6 kernel and I don't have that hardware. Quoted from http://www.voip-info.org/tiki-index.php?page=Asterisk%20timer On kernel version 2.6 it uses internal high-resolution kernel timer and do not require any additional hardware. Now in the original post he says that he is using FC2 so I am not 100% sure if it is 2.6 or 2.4 but FC2 is only one step away from FC3 which does run a 2.6 kernel. I don't know on FC2 as I have never run it. And yes to answer the original poster it did solve my IAX problems. With the demo I would sugest that maybe the SMP kernel on a single CPU server could be a partial cause. I have seen strange things on Dual CPU servers running SMP kernels were 1 CPU has been removed. Hope that helps. David Slán leat, Martin List-Petersen Dublin, Eire (contact info on -- http://www.marlow.dk/) ___ 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-Dev] RE: [Asterisk-biz] Asterisk training andcertification :: AstriconTraining
Brian West wrote: I feel this is a slap in the face for those of us that have been here and I don't feel I should HAVE to pay to be certified... I think me and MANY others are about to walk out of the project over this. I have already spoken with many people that are close to the project. You're hurting US and our ability to make money. I still know the code better than most of the people that will be paying to be certified. You're pushing it here. Well from a newbies point of view I hope you don't pull out cause I still need help and you guys that have been around and know it backwards are a great help with setup and problems. I REFUSE TO PAY!!! I know you guys mean well but you didn't take any of us into account that know this software and know it well. I would have thought that it would be a great idea if in the process of setting this idea up they would need worldwide transer and people on the dev and long timer helpers on users list would have been prime place to find those people. Might have been an idea to come up with a Testing course first for those who think they are good enough and if they are they can pass and become the support/trainers for * in the future. Just my thoughts. And as I said above Please don't leave you guys are way to mucch support for us newbies! David bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-biz- [EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Sunday, December 19, 2004 1:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-biz] Asterisk training and certification :: AstriconTraining *** AsteriskT Open Source Linux PBX Training and Certification Huntsville, AL and Kansas City, MO, December 20, 2004: Digium, Inc., Edvina AB and Sokol Associates today released a new program for training and certification of Asterisk professionals. Asterisk is the leading Open Source PBX for Linux, with support for both PSTN connectivity and many VoIP protocols. The first class in the Astricon Training product line is the five-day bootcamp Introduction to Asterisk. This class will be held in the US and Europe six times during 2005. The organizers and teachers is the same team that set up the Astricon 2004 conference and expo in September this year, an event that gathered over 450 Asterisk users and developers in Atlanta, GA. The new Asterisk certification is named dCAP, Digium Certified Asterisk Professional. To get the certification, one has to go through a 150 question exam as well as a practical exam, where the student builds and configures a PBX. The certification will be given by the Astricon team under license from Digium. This is an important step towards greater acceptance of Asterisk in the enterprise, says Olle E. Johansson of Edvina in Sweden. With a professional training and certification, you can ensure that your staff or your consultants has the required skills to setup and manage a mission-critical PBX platform based on Asterisk. The Asterisk Open Source project is building a professional business ecosystem, says Mark Spencer, the founder of Digium and creator of Asterisk. Many companies are now selling Asterisk-based solutions. With the 1.0 stable release in September, the Digium hardware that ranges from the IAXy end-user device to carrier-class quad-T1 cards and the Digium commercial support we have a professional platform for partnering with major enterprises. The Astricon training and dCAP certification enables us to build a network of consultants that we know will and are able to assist us working on the continued success of Asterisk. The first training class will be held in Kansas City, MO, January 17-21 2005. The cost for a five-day bootcamp with certification is $3,275 USD. Details can be found on http://www.astricon.net AsteriskT is the leading open source PBX, used all over the world. Since it is Linux-based, it inherits all of the power and stability of the operating system. Linux provides open source alternatives to proprietary applications. Asterisk is the first package to fit all telecommunication needs in a broad variety of environments. DigiumT is the creator and primary developer of Asterisk, the industry's first open source PBX. Used in combination with Digium's PCI telephony interface cards, Asterisk offers a strategic, highly cost-effective approach to voice and data transport over TDM, switched, IP, and Ethernet architectures. Digium provides a highly refined selection of quality hardware and software products, developed and implemented using innovative engineering techniques (primarily open source development). A full range of professional services complement these product lines, including consulting, technical support, and custom software development services. The open source communications revolution is here, and Digium is leading the way. Contacts: . Olle E. Johansson, Edvina AB, Phone +46 8 594 788 10, http://www.astricon.net . Steven M. Sokol, Sokol Associates, Phone:
Re: [Asterisk-Users] Caller ID - TE405P - Telstra Onramp 10 - Australia
Nathan Alberti wrote: I am having problems getting incoming caller id to work on a Telstra Onramp 10. I have changed /DEFAULT_CIDRINGS 2/ Is there something i'm missing ? My Cisco 7960 just shows asterisk Thanks, Nathan SNIP linux*CLI show channel Zap/2-1 -- General -- Name: Zap/2-1 Type: Zap UniqueID: 1103473308.3 Caller ID: (N/A) DNID Digits: (N/A) State: Up (6) Rings: 1 NativeFormat: 72 WriteFormat: 8 ReadFormat: 8 1st File Descriptor: 12 Frames in: 1224 Frames out: 1986 Time to Hangup: 0 Elapsed Time: 0h0m24s I know this might be a basic answer, but have you confirmed that CID is enabled and working on the onramp? I know when I dealt with T for an OnRamp 30 18months ago it was ordered with CID enabled but did not work for weeks when it should have. When T was chalanged about the problem it was found out that it was not enabled :( They enabled it and all the problems went away. Might be worth a thought anyway. David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality
Keith O'Brien wrote: Thanks. On a related note. The problem that I am troubleshooting has to do with an IAX connection to TELIAX. Outgoing calls are perfect. When I have incoming calls they are very crackly and break up. I have checked the jitter buffer and it is not overrunning so it doesn't appear to be a jitter or packet loss problem. I am beginning to suspect that since I don't have a ZAPtel card in my machine, * is losing sync with the incoming stream. From what I understand, if there isn't a zap timing source * uses the incoming data stream to derive timing. I have been watching the to and fro of this over the last day or so and being a fairly newbie myself, Just looking plainly at what you have running, You have a SMP kernel running on a Dual Capable server but with only 1 cpu. Why Don't you run a NONE SMP kernel, one which would be suited to the fact that you only have one CPU in the server. Then the other thing if mem serves me you are running 2.6 kernel so why not run ztdummy? With the 2.6 kernel this does not require any specialist Hardware or anything! I have * running on Mandrake 10.1 and I to had similar problems in the begging but as soon as I had ztdummy configured correctly everything seemed to just fall into place and work with IAX and *, not that I have got a perfect dialplan as that confuse's me but hey thats another subject. David Since the incoming stream is using VAD, my assumption is that it is losing the timing during the pauses in the speech. Does anyone know of a way to just turn off VAD in *? This would have multiple benefits (if you have the bandwidth). Turning off VAD will improve voice quality by eliminating and front end clipping during talk spurts and I am assuming will also minimize the impact of not having a ZAP timing source. Is there a way to disable VAD in *? Thanks again. -Original Message- From: Martin List-Petersen [mailto:[EMAIL PROTECTED] Sent: Saturday, December 18, 2004 8:20 PM To: Keith O'Brien Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Re: Asterisk Crackly Bad quality Citat Keith O'Brien [EMAIL PROTECTED]: The URL you are looking for is: http://www.voip-info.org/wiki-Asterisk+timer Thanks. After reading through the notes I checked my server (Dell 1750) and noted that it uses a USB OHCI interface so the first option doesn't appear to be an option. Also it indicates that the second option of using zaprtc http://www.voip-info.org/wiki-Asterisk+zaprtc won't work on SMP systems. The 1750 is a SMP system and I am running a 2.4 SMP kernel but do not actually have a second processor installed. Can I still use zaprtc with a SMP kernel if the second processor isn't actually installed?? zaprtc should work indeed if you only have one CPU in the system. Slán Leat, Martin List-Petersen Dublin, Eire (contact info == http://www.marlow.dk) -- We Klingons believe as you do -- the sick should die. Only the strong should live. -- Kras, Friday's Child, stardate 3497.2 ___ 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] Totally LOST with dialplan and Extensions.
Ok I have spent the last week working on getting my small PBX to work.
I will in the end only have 4 SIP extensions being either softphones of
IP phones. Currently only 1 SIP config for testing.
And at the this point it should be all fairly easy with all inbound and
outbound to PSTN will be going Via Firefly/Freshtel.net in Australia via
IAX. Inbound does work in it's current basic state.
There will be NO ZAP devices, so I have ztdummy running.
I would say that for the outbound dialing I have either missed out
something plainly obvious or a simple typo which would be the challange.
I would think that all the problems are in the extensions.conf file
which really has me confused and totally lost.
I don't expect answers just pointers in the correct direction so that I
can get it to work for the outbound calling to work, I have the inbound
working which was a task but I was able with some pointers to get it
working.
I would like to thank you all for your casting experianced eyes to look
over this. What ever is worked out I will make sure the info gets onto
the Wiki for Freshtel and for a SIP to IAX to PSTN config so that others
can look up the basic configs to do this type of setup. There does not
seem to be from what I can find this basic configs for IAX without FXS
FXO devices.
cheers
David
SIP.CONF
[general]
context=default
realm=monitor.diversified.com.au
bindaddr=203.29.98.221
srvlookup=yes
maxexpirey=180
defaultexpirey=160
disallow=all
allow=speex
allow=gsm
allow=ilbc
allow=ulaw
allow=ilbc
[801]
type=friend
regexten=801
username=801
secret=password
callerid=801
host=dynamic
nat=yes
canreinvite=no
qualify=yes
disallow=all
allow=gsm
allow=speex
allow=ulaw
allow=alaw
IAX.CONF
[general]
tos=lowdelay
jitterbuffer=no
disallow=all
allow=speex
allow=ilbc
allow=gsm
allow=adpcm
allow=alaw
register = 89280250:[EMAIL PROTECTED]
register = 89280250:[EMAIL PROTECTED]
[guest]
type=user
context=default
auth=none
;inbound
[firefly]
type=friend
host=cts-au.freshtel.net
context=default
; outbound
; Firefly (Freshtel)
[89280250] ; Firefly
context=89280250
qualify=no
username=89280250
secret=password
auth=md5
type=friend
host=gateway.freshtel.net
EXTENSIONS.CONF
[general]
static=yes
writeprotect=no
[globals]
SpeakingClock=123
[default]
exten = s,1,Wait,1
exten = s,n,Answer
exten = s,n,DigitTimeout,5
exten = s,n,ResponseTimeout,10
exten = s,n,WaitExten
exten = s,n,Dial(SIP/801)
exten = 13,1,DateTime()
exten = 13,2,Wait(1)
exten = 13,3,DateTime()
exten = 13,4,Hangup
exten = t,1,Goto(#,1)
exten = i,1,Playback(invalid)
exten = 600,1,Playback(demo-echotest)
exten = 600,n,Echo
exten = 600,n,Playback(demo-echodone)
exten = 600,n,Goto(s,6)
exten = ${SpeakingClock},1,Wait(1)
exten = ${SpeakingClock},2,setvar(FutureTime=$[${EPOCH} + 10])
exten = ${SpeakingClock},3,Wait,3
exten = ${SpeakingClock},4,SayUnixTime(${FutureTime},,R)
exten = ${SpeakingClock},5,playback(vm-and)
exten = ${SpeakingClock},6,SayUnixTime(${FutureTime},,S)
exten = ${SpeakingClock},7,playback(seconds)
exten = ${SpeakingClock},8,playback(beep)
exten = ${SpeakingClock},9,wait(2)
exten = ${SpeakingClock},10,goto(1)
exten = _394.,1,SetCallderId(89280250)
exten =
_394.,2,Dial(IAX2/89280250:[EMAIL PROTECTED]/${EXTEN:3},60,r)
[outgoing-firefly-peers]
exten = _62,1,Macro(outgoingfirefly,${EXTEN:2},70) ; Firefly
[macro-outgoingfirefly]
exten = s,1,SetCallerID(89280250 89280250)
exten =
s,2,Dial(IAX2/89280250:[EMAIL PROTECTED]/${ARG1},${ARG2},r)
exten = s,3,Congestion
[macro-outgoingfreshtel]
exten = s,1,SetCallerID(89280250 89280250)
exten =
s,2,Dial(IAX2/89280250:[EMAIL PROTECTED]/${ARG1},${ARG2},r)
exten = s,3,Congestion
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Re: [Asterisk-Users] How to demo the Power of Asterisk
Jean-Michel Hiver wrote: I've been setting * at home just to train myself with it. Here is what I have: - IVR menu - music on hold / transfer - voicemail - transparent Zap or IAX routing - I can call home, dial a pin and make long distance call through IAX It would be great if you could share with the rest of us newbie type people some of your extensions.conf and iax.conf to do things especially like the last one were you can dial in and pin and make long distance calls. This does very much intrest me especially :) Cheers David I have just been *scratching* the surface of * but I'm already puzzled with its power... Cheers, Jean-Michel. ___ 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] List's quiet or down?
Is it just me or are there problems? The list has just shutdown over the last 24 hours :( David ___ 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] Users list.
Does this sudden rush of email mean we are all back online? David ___ 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] Voicemail Codec challanges.
Just working on Configing up Voicemail and now that I have got it working and configed and answering the way it should be I have another challange. on the * CLI I get this -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/6001/INBOX/msg format: wav49, 0x8133390 -- x=1, open writing: /var/spool/asterisk/voicemail/default/6001/INBOX/msg format: gsm, 0x8132f48 -- x=2, open writing: /var/spool/asterisk/voicemail/default/6001/INBOX/msg format: wav, 0x8157988 Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space Dec 6 20:13:36 WARNING[24187]: codec_speex.c:196 speextolin_framein: Out of buffer space -- Recording automatically stopped after a silence of 10 seconds -- Playing 'auth-thankyou' (language 'en') -- Recording was 0 seconds long but needs to be at least 3 - abandoning -- Playing 'vm-opts' (language 'en') == Spawn extension (default, 8500, 1) exited non-zero on 'SIP/6001-8e4e' when I go to record a voicemail mesg. Anyone got any idea as to which way I would turn? It is likely to be a Config issue but I am unsure were it is to look for it. Thanks for advice in advance. David ___ 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] DTMF via PSTN to * to IAX to * challanges.
Ok I have an * server finally setup and acepting calls from freshtel and I am VERY impressed at how well the Freshtel.net service works but thats another subject :) I have it all setup so that I can Dial my DID number on freshtel and that gets set to my * via IAX. At the moment I have the demo configured so that I can test it all and make sure it is all working. The problem is that I can't use the phones Keypad to dial anything. If I use linphone or x-lite softphones I can DTMF on my * server. I have my * server setup as http://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20FireFly which is what is required to get it this far. Thanks to Adam Hart on the list. I have my * server answering and playing the demo setup but I can't select an extn or anything that requires DTMF from the phone. Does anyone have any ideas as to why this would be and were I would look to solve it? I would like to be able to have it so that people can select an Extn to be connected to or other things. Thanks David ___ 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] IAX to freshtel
Well here is something simple, well I think it is for the smarty's out there :) I got a connection to freshtel and want to get the iax working. I have config'ed up iax.conf with the register line and get in return in the cli -- Registered to '202.168.7.130', who sees us as 203.29.98.221:4569 So that appears to be connected. When I call the DID number I get the Voicemail on Freshtel's service which means that freshtel does not see my * server as being online and active. How would I go about Debuging this type of challange? I am sure it is something at my end but I am not sure were I would start to go through debuging an IAX challange. Thanks David ___ 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] www.voip-info.org
Has the wiki died or is it just my routing to the wiki from Australia? I have not been able to connect to it for the last hour or more :( David ___ 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: SV: [Asterisk-Users] www.voip-info.org
Thorben G. Jensen wrote: It dead from Denmark too :-( Well I think yes it is! :( All I get on traceroute from me! traceroute to www.voip-info.org (66.151.54.101), 30 hops max, 38 byte packets 1 192.168.2.1 (192.168.2.1) 0.377 ms 0.366 ms 0.189 ms 2 rns02-kent-syd.comindico.com.au (203.194.30.201) 30.771 ms 25.099 ms 26.116 ms snip to save bandwidth 14 unknown.Level3.net (63.208.234.134) 178.782 ms 180.669 ms 179.576 ms 15 border17.ge3-0-bbnet2.lax.pnap.net (216.52.255.85) 178.842 ms 180.493 ms 180.459 ms 16 commp-2.border17.lax.pnap.net (216.52.253.50) 185.969 ms 190.263 ms 185.715 ms 17 * * * 18 * * * 19 * * * Me hopes it is not down to long am in the middle on tring to config my now working * server :( Oh well at least it is not just me. Thanks David -Oprindelig meddelelse- Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af David Uzzell Sendt: 2. december 2004 07:32 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: [Asterisk-Users] www.voip-info.org Has the wiki died or is it just my routing to the wiki from Australia? I have not been able to connect to it for the last hour or more :( David ___ 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: SV: [Asterisk-Users] www.voip-info.org
James H. Thompson wrote: Commpartners (who provides hosting for voip-info.org) is doing a network upgrade tonight. AH That would explain it :) Just happens to be when I want to work how to config * :( Oh well will just have to wait. To bad we could not mirror the wiki around the world. I am sure there are people who would help but not sure if that is even possible. Anyway thats off topic. Thanks for the info guys. David Jim James H. Thompson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] - Original Message - *From:* Luki mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:[EMAIL PROTECTED] *Sent:* Wednesday, December 01, 2004 8:53 PM *Subject:* Re: SV: [Asterisk-Users] www.voip-info.org http://www.voip-info.org Dead for me too.. I am in the US.. Dead here too and I am in LA, next door to it (last hop commp-2.border17.lax.pnap.net). Maybe there are doing an upgrade... I recall their DB server was spitting out too many connection errors yesterday... --Luki ___ Asterisk-Users mailing list [EMAIL PROTECTED] mailto:[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] OS Choice ?
dean collins wrote: Michael, Xorcom can allow ssh. You didn't read the instructions properly (lord knows I didn't the first few dozen times). When you insert the disk for the first time instead of typing linux or pressing enter to start the install type expert This will halt the installation at each section to ask you various questions (most of which you can ignore) but it will allow you to install ssh and then you can continue the rest of the installation remotely. And the other option is to let it install as it would standard, Go into the Menu and under Maintaince I think from mem and then install packages then you find ssh down under net and install it. Done. With regards to being out of date on * I can understand that but I would say that if it was a normal install it should be fairly simple to upgrade to current version. This is what I am going to try today. Hopefully they have not stripped out all the compile stuff from the install cause that will make it pointless. Anyway if anyone is intrested I will let them know how I go. David Cheers, Dean (yeh one question I was able to answer). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Monday, November 29, 2004 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OS Choice ? On Mon, 29 Nov 2004 10:09:26 +0200, Gilad Ben-Yossef wrote: Alex Brecher wrote: Which Distro is the most commonly used distro with Asterisk please ? I don't know which is most commonly used, but I can tell you which is the easiest to install if you're going to install the OS from scratch anyway and plan to use it with Asterisk: Xorcom Rapid is a Debian/Asterisk distribution program that includes an auto-install and special auto-configuration features. It quickly and effortlessly converts any PC to a functioning Asterisk PBX... Since I had to rebuild my * server over the weekend I had a go with this Xorcom thingy. It pretty much did as it promised, with minimal user interaction it created a working * installation with a handy text mode shell. However, being a Linux newbie I found that it lacked a few basic things that I needed to make it work for me...most significantly the ability to use SSH to connect from my desktop transfer config files and otherwise and administer *. Had I been able to do that I would probably have tried it out for a while. Oh, also the version of * it installed was quite old...CVS 5/11/04 if I recall. That was also a major concern. If I have to build a new server for my home office some time in the future I'll try the AstLinux ISO which is an embedded version of Gentoo with Asterisk 1.0. Runs on PC Engines WRAP boards. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list [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] OS Choice ?
Michael Graves wrote: On Mon, 29 Nov 2004 10:09:26 +0200, Gilad Ben-Yossef wrote: Alex Brecher wrote: Which Distro is the most commonly used distro with Asterisk please ? I don't know which is most commonly used, but I can tell you which is the easiest to install if you're going to install the OS from scratch anyway and plan to use it with Asterisk: Xorcom Rapid is a Debian/Asterisk distribution program that includes an auto-install and special auto-configuration features. It quickly and effortlessly converts any PC to a functioning Asterisk PBX... Since I had to rebuild my * server over the weekend I had a go with this Xorcom thingy. It pretty much did as it promised, with minimal user interaction it created a working * installation with a handy text mode shell. However, being a Linux newbie I found that it lacked a few basic things that I needed to make it work for me...most significantly the ability to use SSH to connect from my desktop transfer config files and otherwise and administer *. Had I been able to do that I would probably have tried it out for a while. Oh, also the version of * it installed was quite old...CVS 5/11/04 if I recall. That was also a major concern. This is the only thing that worried me. And now from what I have been able to find is that there are limited to none development software on the install. I was hopeful that there would have been enough development software installed so that I could run an Upgrade :( which there appears not to be!. So as you I think I will be rebuilding the server again :( I need to ability to be able to update the * versions as bugs are fixed and features are added. It is a Great start and will allow newbies to get started with little problem but i can see it being an issue with the mailling list being users asking questions about out of date CVS versions which may well have the bugs fixed. David If I have to build a new server for my home office some time in the future I'll try the AstLinux ISO which is an embedded version of Gentoo with Asterisk 1.0. Runs on PC Engines WRAP boards. Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list [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] GNUGK + Asterisk consultant requiered
Please 1 is enough. Upto 4 so far :( David Voip Business wrote: Hello Lists I need help to setup a GNUGK with an Asterisk my needs are simple I have: billing system compatible with GNUGK SIP ATA devices for origination h323 and SIP termination services 1 linux box (can be 2 in same datacenter) I need: Sip traffic from ATAS point to asterisk then tranlsate to h323 and bill that call from the GNUGK (Radius to my billing system) May be some SIP termination to other Service providers so call back to asterisk in h323 then asterisk terminates in SIP. MUST HAVE: G729 (paying licences) AAA from the gnugk Please Help in this regard.contact me off-list HA ___ 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] 4 port ISDN BRI pci card
I am after something similar. I want to be able to use 2 bonded ISDN BRI's and I am not sure what hardware will run with asterisk? Anyone got any ideas? Cheers David Miroslav Nachev wrote: Dear Bartosz, Try this: http://www.junghanns.net/asterisk/page17.html quadBRI PCI ISDN EUR 600,- Best Regards, Miroslav Nachev BJ Hello, BJ I am looking for 4 port ISDN BRI card. BJ I have checked wiki and found one, but they do not show prices BJ for that card. Can somebody advise me which ISDN 4 port card works good BJ with Asterisk, BJ Thank you in advance. BJ Bartosz BJ ___ BJ Asterisk-Users mailing list BJ [EMAIL PROTECTED] BJ http://lists.digium.com/mailman/listinfo/asterisk-users BJ To UNSUBSCRIBE or update options visit: BJhttp://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] ISDN BRI VoIP Internet
Ok I have searched the Archives and can't seem to find anything that would give me a hint on if it is possible or not. What I want to look at having a Small Home Office setup were I can use the 1 BRI for both DID and Internet at the same time. Is it possible to use something like a FRITZ! ISDN BRI card to have full time Internet on one of the B channels and have the other B channel for * for both incomming and outgoing calls? and even to be able to have 2 BRI's in the future and have them split between internet and external lines. Thanks for the advice in advance :) David ___ 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] ISDN BRI VoIP Internet
Thanks for your advice also Colin, I would like to stay away from POTS cards in the server but it is an option if all else fails. Philipp von Klitzing wrote: Hi! What I want to look at having a Small Home Office setup were I can use the 1 BRI for both DID and Internet at the same time. Is it possible to use something like a FRITZ! ISDN BRI card to have full time Internet on one of the B channels and have the other B channel for * for both incomming and outgoing calls? Should work. However chan_capi apparently cannot tell if a B channel is already in use by another application (that manages your dial-up). So that means you either need to have devices=1 (instead of 2) in /etc/asterisk/capi.conf to permanently assign one B channel for Internet dialup only, or you live with the fact that once in a while chan_capi will return a busy when you try to use two (outgoing) B channels while Internet traffic is on. Not sure how i4l works in this respect. Cool thanks for that. Would the FRITZ! card be best suited to this or would something else be better suited? Thanks David and even to be able to have 2 BRI's in the future and have them split between internet and external lines. Should work, although stability *might* be an issue. There are even reports about 3 cards in one system. Philipp ___ 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] Small office requirements - Can this be done ?
Low, Adam wrote: I've done a fare amount of analysis on codec bandwidth requirements and you should remember that you typically will require more bandwidth over ADSL than you would over any other technology. I estimate a requirement of around 108Kb (on the wire) per G.711 channel rather than 86kb over straight PPP/HDLC based connections. Why I hear you ask ? The following calculations are based on G.711 PCM running at 20ms samples resulting in 200 byte packets (default for most codec implementations). 200 bytes G.711 packet + 8 bytes AAL5 overhead = 208 bytes 208 bytes fit in 5 cells of 48 bytes payload 5 cells are 265 bytes. VoIP over ATM AAL5MUX thus has an overhead of 21.51% VoIP G.711 conversation sends 50 packets per second. This uses 250 cells per second. This causes approximately 10 OAM5 cells to be sent over the duration. The total bitrate is thus (250 + 10) * 53 bytes * 8 bits = 110240 bits/second = 107.66Kbit/s So for us Dummies out here :) who just know it works. This would mean that if you had a 512/256 aDSL and a 256 ISDN connection you would be able to have more channels over the ISDN? David Steve Kennedy wrote: On Mon, Mar 01, 2004 at 07:08:29PM -0600, Michael Graves wrote: I have disagree about the ADSL. I have a1.5 Mbit/512kbit service from Covad (in the US Southwest) and I have sustained 4 calls without a problem. I prefer to use GSM over G.711to squeeze it down, but that is my choice. I don't feel that call quality is substandard. That's the crunch (1.5/512) ... it's actually the 512 which is relevent. Virtually all DSL in the UK is a wholesale product from BT (they have about 2 million customers, Easynet who local loop unbundle may have 20,000, the rest of the providers maybe another 10,000 between them). All BT ADSL is 256K upstream, all BT DSL is contented (in theory 20:1 and 50:1, but actually a lot less than that), there are a few providers doing their own contention over BT's product. However the 256K upstream is still the limiting factor, so you can get one, and MAYBE two VoIP lines over it. If BT would up the upstream to 512, you could probaly get 4 out of it Steve On the UK DSL using G.711 you should easily get 2 concurrect calls, G.711 uses about 84k(incl overhead) in each direction, so 2 calls would be 168K (of the 256k) If you switch o GSM or iLBC you should get 6 concurrent calls, and if you were to use IAX2 trunking you could *maybe* squeeze another one.. Other codecs could offer even more but I haven't tested them.. Later.. * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ 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] PBX (Asterisk) - Cellular Phone Network
Dan wrote: Hi, ... I won't bother with any of that - purchase a Nokia Premicell (or other manufacturers similar item). This device takes a normal GSM SIM card and then presents a normal PSTN line interface - plug that into your normal Asterisk PSTN line card - job done. A PCI and/or USB device, Asterisk compatible, able to accept a SIM card and talking digital only (inclusiv audio) will be a great thing, even internally it can be based on a cheaper GSM phone... What about a PCMCIA GSM card connected through a PCI/PCMCIA adapter? There is only needed an Asterisk supported driver. :-) Dan That sounds like a really GREAT idea, I thought I would put my foot in. A great result in this would be some way to drive the Nokia phones through the serial Port and Computer cable! This would allow us in Australia to be able to use CDMA it would also mean that any cheap Nokia phone could be used as long as it has the computer cable link. Anyway thats my 2c worth in the Pot! David ___ 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] Couple of Questions for Australian Users!
Just a couple of questions for Aussie users/resellers! I have only just started to look at asterisk a couple of weeks ago and have found some very intresting discussions and some useful info on what can and can´t be done with it and technology. The questions i have is, Are there people using it is Aussie?( I would say yes so prob answered my own question ) :) Hardware, is the digium hardware OK in australia? or does it require changes/mod´s or is just a pain in the butt? Alternate hardward for Aussie if above is an issues? Resellers/Importers of the Digium hardware? I have looked at the website and see that there are aussie resellers but they all seem to have their own agenda and the one I spoke to was not intrested in selling me hardware alone they wanted to do me a whole service deal! Also IP phones? What type/recomendation? I am going to be using Phone Software as well on laptops but I need an IP phone option to do phones over WAN. I really only know of the cisco and avaya types, I have not had any contact with any other types of phones. Intrested in Aussie suppliers for what ever phone you would recomened unless they are great phones then I would have to find an USA reseller. OK think thats all! Thanks Guys doing some great work with what looks to be some very GREAT software! Thanks in advance for your help! David Uzzell ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
