Re: [Asterisk-Users] GSM sound player for windows?
Chuck Bunn napisał(a): Is there a way to play .gsm sound files on Windows. Is there an extension for Windows Media Player or Real Player to allow playing of these files? http://www.voip-info.org/wiki/view/Asterisk+sound+files Section Playing GSM files on Windows. -- Best regards, Bartosz Piec ___ --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] GSM sound player for windows?
Apple Quicktime can do it (Is a part of itunes now) Or look on google for winamp gsm plugin. Joachim. Bartosz Piec wrote: Chuck Bunn napisał(a): Is there a way to play .gsm sound files on Windows. Is there an extension for Windows Media Player or Real Player to allow playing of these files? http://www.voip-info.org/wiki/view/Asterisk+sound+files Section Playing GSM files on Windows. ___ --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] Festival Sound Quality
Hi! I installed Asterisk and Festival on a Intel Pentium 4 CPU 1.70GHz and 500 MB Ram. Whenever Asterisk calls the Festival(..) Application, it seems that a lot of UDP packets get lost or are corrupt (although the festival server is running on the same machine). (ast_rtp_read: RTP: Received packet with bad UDP checksum). The resulting sound quality is really poor. Did anyone encounter the same problem and knows a workaround for it? I tried to pre-recording the Festival-Text and to do just a simple playback. Unfortunately, the Asterisk-Text itself is clicky and noise. A web site tells me, I can get good results if I use a multiband EQ to filter out the undesired freqs. Did anyone do this before, so that he can recommend me a program and the freqs? (a simple command-line instruction for sox or audacity 'filter -f blah in out' would be great... ). Greetings, Marcus ___ --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] Meetme Conference-reg
I don't have app_meetme.so file neither in /usr/lib/asterisk/modules, nor /usr/src/asterisk/apps. How to get it? -- Best regards, Bartosz Piec ___ --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] Re: Meetme Conference-reg
In article [EMAIL PROTECTED], Bartosz Piec [EMAIL PROTECTED] wrote: I don't have app_meetme.so file neither in /usr/lib/asterisk/modules, nor /usr/src/asterisk/apps. How to get it? You need to get, build and install zaptel on your system, and then rebuild Asterisk. Asterisk won't build app_meetme if it doesn't find zaptel on your system. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] DNS Server Failure wreaks havoc
Using IPs only and no domains didn't do the trick for me, but then I may have missed one or two without realising. I did find that putting a DNS server in the asterisk box works perfectly for asterisk, but the grandstream budgetone 101 phones i had relied on a dns (even if i replaced the ntp server with an IP) and would not work without them. the symptom would be the call comes in (over ISDN), rings , but the phones don't pickup. This was not an issue for SPA-841s or SNOM 190s, both of which are perfectly happy with just IPs. Funny things was that they weren't happy using the asterisk box's dns either - even though i could dig @192.168.1.254 domain.com from my laptop without problems. ___ --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] Re: Meetme Conference-reg
Tony Mountifield napisał(a): You need to get, build and install zaptel on your system, and then rebuild Asterisk. ztdummy is enough? Will building Asterisk break something in my working installation? :) -- Best regards, Bartosz Piec ___ --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] Re: Meetme Conference-reg
On Mon, 2005-11-07 at 10:07 +0100, Bartosz Piec wrote: Tony Mountifield napisał(a): You need to get, build and install zaptel on your system, and then rebuild Asterisk. ztdummy is enough? it is for app_meetme, you just need a timing source that is external. app_conference doesnt require this however, but there are some limitations, such as no dtmf while in the conference (exiting, admin menu, etc). -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ --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] newbie questions
Hello Hiu, Monday, November 7, 2005, 4:51:35 AM, you wrote: HYO i am pretty new to asterisk. hope to learn more. HYO i have this notice from the console. when i was doing the echo testing HYO by putting the context=default. then, i called out 600 to get the echo HYO test, i can hear the operator talking, but i cant really hear the playback. HYO i am trying to dig around from info from the log files. HYO what does it mean? HYO RFC3389 support incomplete. Turn off on client if possible HYO hope to help..thanks That means that you have to turn off silence suppression in your softphone (in xlite is named transmit silence). Hope it helps! -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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] What's the purpose of the username= line?
[EMAIL PROTECTED] wrote: After some experimentation and posting, I have concluded that in the file sip.conf, the line: username = irrelevant Please read sip.conf.sample in your distribution for updates on configuration parameters. The username parameter has nothing at all to do with a username for registrations... It should be named defaultuser since it is used in combination with defaultip to construct an URI to user if we want to call a peer with host=dynamic before registration or when registration expired. It is by far the most misunderstood parameter in sip.conf, but also a hint that you should not guess, but read available documentation ;-) /Olle ___ --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] Re: Meetme Conference-reg
In article [EMAIL PROTECTED], Bartosz Piec [EMAIL PROTECTED] wrote: Tony Mountifield napisa³(a): You need to get, build and install zaptel on your system, and then rebuild Asterisk. ztdummy is enough? ztdummy is only a device driver. You also need the zaptel module. Will building Asterisk break something in my working installation? :) Not if you do it properly and with understanding. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] Dropping last digit from phone number
Hello,
{$EXTEN:1} is used for dropping the first digit. But hot to get rid of
the last digit? Is it possible?
--
Best regards,
Bartosz Piec
___
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Re: [Asterisk-Users] Re: Meetme Conference-reg
No - just make sure you DO NOT type make samples. regards, Jenn - Original Message - From: Bartosz Piec [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, November 07, 2005 8:07 PM Subject: Re: [Asterisk-Users] Re: Meetme Conference-reg Tony Mountifield napisał(a): You need to get, build and install zaptel on your system, and then rebuild Asterisk. ztdummy is enough? Will building Asterisk break something in my working installation? :) -- Best regards, Bartosz Piec ___ --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] Re: Meetme Conference-reg
Tony Mountifield napisał(a): ztdummy is only a device driver. You also need the zaptel module. And this is this: http://ftp.digium.com/pub/zaptel/zaptel-1.0.9.2.tar.gz, right? I'm using 1.0.9 version. Will building Asterisk break something in my working installation? :) Not if you do it properly and with understanding. Backuping /etc/asterisk is enough? -- Best regards, Bartosz Piec ___ --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] SJphone Awaiting ACK after updating Asterisk to CVS-HEAD of September
Hi, sometimes I cant answer calls with SJphone because of an Awaiting ACK error. The problem has come after I updated Asterisk from CVS HEAD of August to HEAD of September. I had no other changes in my configuration, so I think it must be related to something in Asterisk. FYI, Asterisk is now updated to the latest CVS HEAD and the problem is still there. Did anyone already have this kind of problem, please? Full debug and tcpdump are available if needed. Thanks, Alex ___ --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] Asterisk Addons linker's error....
Hi Everybody, I have spent last 3 days in trying to compile Addons. 1.0.9 I have succeded in reaching linker's phase. The linker is unable to find both z lib and mysqlclient lib. Many tests in trying to change z to zlib.so(available on /usr/lib) and libmysqlclient.so (existing on /usr(lib/mysql). Still the problem is there! (going nuts...) I'm using Suse 9.2 distro of Linux, while I downloaded sources for mysql... Thank you everybody Mauro Zanin ___ --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] Re: Meetme Conference-reg
In article [EMAIL PROTECTED], Bartosz Piec [EMAIL PROTECTED] wrote: Tony Mountifield napisa³(a): ztdummy is only a device driver. You also need the zaptel module. And this is this: http://ftp.digium.com/pub/zaptel/zaptel-1.0.9.2.tar.gz, right? I'm using 1.0.9 version. Yes. Will building Asterisk break something in my working installation? :) Not if you do it properly and with understanding. Backuping /etc/asterisk is enough? So long as you just do make clean and make install, and DON'T do make samples, then rebuilding won't disturb your /etc/asterisk directory; you just stop and restart asterisk and it will use the existing /etc/asterisk files. Of course, keeping a backup of /etc/asterisk is a good idea anyway... Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --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] queues in 1.2-beta2
after we upgraded to beta 2 incoming queues does not work properly. This simple test lines in extensions.conf creates the problem: exten = s,1,Answer exten = s,2,Queue(250|r|||30) exten = t,1,Hangup Evrything is working except that the callee can not here the music, I have also tried to use the r to use ringing instead of music with the same result. This worked in beta 1, I have read the upgrade documentation and can't find any thing related to this. Anyone having the same experience or have I missed something? I have set the autofallthrough option to no in extensions.conf just as a test with the same result. It's not NAT related, it has been working before and as soon as an agent answer the call the audio is working in booth directions... cheers urban ___ --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 Addons linker's error....
Hello Mauro, Monday, November 7, 2005, 11:21:25 AM, you wrote: MZ Hi Everybody, MZ I have spent last 3 days in trying to compile Addons. 1.0.9 MZ I have succeded in reaching linker's phase. The linker is unable to find MZ both z lib and mysqlclient lib. MZ Many tests in trying to change z to zlib.so(available on /usr/lib) and MZ libmysqlclient.so (existing on /usr(lib/mysql). MZ Still the problem is there! (going nuts...) I'm using Fedora, normally I use ldconfig after installing libraries. http://www.die.net/doc/linux/man/man8/ldconfig.8.html Hope it helps! P.S. italiano ? :) -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ --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] meetme conference getting error using codec g729
Hi all when i try to the conference the number i am getting the following error in asterisk console. i am using the g729 codec in asterisk and my sip devices but i can able make the call between the device. error: Nov 7 16:07:49 NOTICE[3190]: channel.c:1703 ast_set_write_format: Unable to find a path from gsm to g729 Nov 7 16:07:49 WARNING[3190]: file.c:787 ast_streamfile: Unable to open conf-onlyperson (format g729): No such file or directory regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.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
Re: [Asterisk-Users] meetme conference getting error using codec g729
You will need to buy some g729 licences... PaulH - Original Message - From: nr k [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, November 07, 2005 9:42 PM Subject: [Asterisk-Users] meetme conference getting error using codec g729 Hi all when i try to the conference the number i am getting the following error in asterisk console. i am using the g729 codec in asterisk and my sip devices but i can able make the call between the device. error: Nov 7 16:07:49 NOTICE[3190]: channel.c:1703 ast_set_write_format: Unable to find a path from gsm to g729 Nov 7 16:07:49 WARNING[3190]: file.c:787 ast_streamfile: Unable to open conf-onlyperson (format g729): No such file or directory regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.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
[Asterisk-Users] Outgoing and incoming call of LineJack
Hi all, I'm testing Quicknet LineJack with Asterisk. I tried SIP to LineJack call. It is OK. No problem. I have problem on the LineJack to SIP call. Also I want to routeincoming call(PSTN-LineJack) to SIP user. Please help me Thanks Regards Ganbaa ___ --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] meetme conference getting error using codec g729
Hi, Seem to be a G729 licences issue. Have you buy G729 licences ? Regards, Le lun 07/11/2005 à 11:42, nr k a écrit : Hi all when i try to the conference the number i am getting the following error in asterisk console. i am using the g729 codec in asterisk and my sip devices but i can able make the call between the device. error: Nov 7 16:07:49 NOTICE[3190]: channel.c:1703 ast_set_write_format: Unable to find a path from gsm to g729 Nov 7 16:07:49 WARNING[3190]: file.c:787 ast_streamfile: Unable to open conf-onlyperson (format g729): No such file or directory regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.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
[Asterisk-Users] queue_log on MySQL
Hello list, we are in the process of releasing a new version of QueueMetrics that will be able to analyze queue_log data stored on MySQL table, with no need to change your table definition. If you currently host queue_log data on MySQL and would like to help us testing it, please drop us a line. QueueMetrics is a call center monitoring system and is free (as in beer) for smaller installations. Yours l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] asterisk as SIP gateway
I want to enable access to some context in asterisk without authentication. The only limitation is ip number. All possible extension can make calls. How can i do that? greetings mk ___ --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] Dropping last digit from phone number
Bartosz Piec wrote:
Hello,
{$EXTEN:1} is used for dropping the first digit. But hot to get rid of
the last digit? Is it possible?
-1
:)
--
Cheers,
Matt Riddell
___
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Re: [Asterisk-Users] Dropping last digit from phone number
Matt Riddell napisał(a): -1 I've tried it. It just leaves the last 1 digit and drops the rest. -- Best regards, Bartosz Piec ___ --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] Voicemail
Hi, I'm trying to translate the voicemail application to my local language. I want to translate the notification email which Asterisk send when you have new massages. Where I can find this file ?? Cheers to all Andrew ___ --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] h323 nat externip
HI All, Is there some way of telling h323 / oh323 to use an externip in the rtp streams and 323 data similar to how you can configure sip on asterisk? We have a system that we need to be able to send outbound calls via sip (working fine) and receive calls via h323. The machine is on a private lan behind nat, which we have managed to get working fine with the voice gateway via sip, but when ever calls come in via h323, the asterisk server sends accepts the call then starts sending rtp packets back to the voice gateway, with the local lan ip address as the address for the gateway to respond to (the voice gateway is a Quintum tenor DX, we did some logging on it to figure out that it was trying to respond to 192.168.1.2 instead of our external ip). In a sip environment, we would simply set externip and the rtp / sip traffic would have our correct external address in them for the the downstream gateway to communicate back with, but there does not apear to be any such option in the h323.conf file and i have done a quick browse of the source and cannot seem to find anything (my c++ us not the best thou) - is there such an option to force h323 to use an external ip instead of the local private one? Or does anyone know of a patch that would allow it to do so? If not, could anyone point me in the right direction for hakcing the code to use either a configured ecternal ip like in sip.conf, or even where to hardcode the external ip in the code (yep, am getting that desperate) so that calls will work. Currently we can call into the gateway, asterisk answers the call, then we get the good old 1 way audio problem that plagues this industry (thou we have sip working fine with a bit of port and firewall tweaking) Cheers, Ben ___ --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] compiling problems
The problem is the 2.6. I know that there is compability also with that kernel, but in my small experience, I've got not these problems with 2.4. Now, I've got to migrate Asterisk into a Dual Xeon 3.0 4Gb RAM. What distro would you use? Until now, I've tested CentOS 3.4 Server with no problem, but not on this kind of server. With Fedora 3, too many problems, concerning the kernel 2.6. Suggestions? Thanks 2005/11/6, Tzafrir Cohen [EMAIL PROTECTED]: On Sat, Nov 05, 2005 at 07:29:18PM +0100, FaberK wrote: Fedora Core 3 kernel-0-2.6.9-1.667 and kernel-2.6.12-1.1380 (same results) Sangoma 102 Concerning udev, I've read that it uses hotplug and if I'm not wrong, I remember that zaptel got conflicts with hotplug. But maybe I'm confusing (terrible headache!) Thanks a lot! zaptel should not conflict with hotplug if the specific hardware driver module is well-written (e.g: declares PCI IDs it will identify). This will mean that hotplug will try using it automatically. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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 -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] queues in 1.2-beta2
No. I've not had the problem you've mentioned. You can post your relevant extensions.conf, queues.conf, and agents.conf either here or in the bugs.digium.com Bug Tracker and someone will take a look at your problem. On 11/7/05, Urban [EMAIL PROTECTED] wrote: after we upgraded to beta 2 incoming queues does not work properly. This simple test lines in extensions.conf creates the problem: exten = s,1,Answer exten = s,2,Queue(250|r|||30) exten = t,1,Hangup Evrything is working except that the callee can not here the music, I have also tried to use the r to use ringing instead of music with the same result. This worked in beta 1, I have read the upgrade documentation and can't find any thing related to this. Anyone having the same experience or have I missed something? I have set the autofallthrough option to no in extensions.conf just as a test with the same result. It's not NAT related, it has been working before and as soon as an agent answer the call the audio is working in booth directions... cheers urban ___ --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] One Touch Record in 1.2
Tim Litwiller wrote: Well, I'd like them to drop in my voicemail when done recording - maybe in a separate recordings folder but I'd like to use the same interface to play them back. I would like that, too. Is anyone working on it? If not, I will put it on my TODO list. ___ --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] Voicemail
Andrew Nowrot napisał(a): I'm trying to translate the voicemail application to my local language. I want to translate the notification email which Asterisk send when you have new massages. Where I can find this file ?? I think that this can be set in 'emailbody' variable in voicemail.conf: http://www.voip-info.org/wiki/index.php?page=Asterisk+config+voicemail.conf -- Best regards, Bartosz Piec ___ --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] Dropping last digit from phone number
Bartosz Piec wrote:
Matt Riddell napisał(a):
-1
I've tried it. It just leaves the last 1 digit and drops the rest.
You could try ${EXTEN:-LEN(${EXTEN:1})}
--
Cheers,
Matt Riddell
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[Asterisk-Users] FXS problems
Hi all I've got a digium Wildcard TDM400P REV E/F Board 1, I seem to be having some problems with the FXS modules on i, for example when i dial 90044117XX tail -f /var/log/asterisk/full gives me Nov 7 13:01:01 VERBOSE[2516]: -- Starting simple switch on 'Zap/1-1' Nov 7 13:01:05 DEBUG[2516]: DTMF digit: 9 on Zap/1-1 Nov 7 13:01:07 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:07 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:08 DEBUG[2516]: DTMF digit: 6 on Zap/1-1 Nov 7 13:01:12 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:12 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:12 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:13 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:13 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:13 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:14 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Which indicates to me that the FXS module is not getting all the signalling, as numbers are missing I have added relaxdtmf=yes to my zapata.conf but this seems not to help atall. Could this be a hardware failure? Thanks in advance Bails ___ --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 as SIP gateway
Drop the authentication elements from the SIP stanza that refers to the devices you want to use. Miloš Kocbek wrote: I want to enable access to some context in asterisk without authentication. The only limitation is ip number. All possible extension can make calls. How can i do that? greetings mk ___ --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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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
Re: [Asterisk-Users] Caller ID How does it get setup?
Most SIP based VoIP providers will not allow you to change your CID. Especially if you have a phone number from them. If I could get an all you can eat IAX based provider I'd buy it in a heartbeat. Then I could make my outbound number change according to which phone I was using or which line I was on. Jason Brashear wrote: OK I am exhausted. I can't seem to figure out how to send a caller ID along with a Outbound call. Can you believe that I got Vonage to reset my Cisco ATA for $15.00 I then canceled my account! Well I was with them for over two years, now I am running Asterisk like the big boys! LOL... Anyway, Outbound Caller ID Hos is this done? I now use VoicePulse as my provider. -Jason ___ --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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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
Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200
sip.conf: [general] allowguest=no bindaddr=0.0.0.0 bindport=5060 callevents=yes defaultexpirey=300 externip=204.74.89.12 externip=204.74.89.13 localnet=10.0.10.0/255.255.255.0 maxexpirey=3600 relaxdtmf=yes srvlookup=yes tos=lowdelay videosupport=no ; global channel settings disallow=all allow=ulaw allow=gsm canreinvite=no dtmfmode=rfc2833 language=en [100074] type = friend secret = mysecret qualify = yes nat = never host = dynamic callerid = Waldo Rubinstein 211 context = test-context mailbox = [EMAIL PROTECTED] The phone is at IP 10.0.10.236, so it's within the localnet. Thanks, Waldo On Nov 6, 2005, at 11:11 PM, C F wrote: can you post the sip.conf for that uip200? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: When I dial the extension, I get this: -- Executing Dial(IAX2/gateway0-16386, SIP/100074|20) in new stack == Everyone is busy/congested at this time (1:0/0/1) When I do a sip show peer 100074, everything it shows matches the results of executing the same sip show peer on * 1.0.9 and 1.2b1, except: Status : UNREACHABLE However, I can make any type of calls from them phone. I can ping the phone from the * server. It's just that * 1.2b2 can't reach it, for some reason. Thanks, Waldo On Nov 6, 2005, at 1:37 PM, C F wrote: Whats the exact CLI output you are getting when calling that extension? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Nope. It isn't active. I even factory reseted the phone but still the same. One more piece of information: it works just fine in 1.2b1. I beginning to think it could be a bug in 1.2b2. Any other ideas/suggestions? Thanks, Waldo On Nov 5, 2005, at 9:10 PM, C F wrote: You sure that the DND (Do Not Disturb) button is not active on the UIP200? On 11/4/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro phones. All phones register fine with * and I can place outbound calls with no problem. I can call from the X-Pro to any other X-Pro. I can call from UIP200 to any other X-Pro. However, the UIP200 cannot receive calls. Every time I call the UIP200, the CLI says Everyone is Busy/ Congested and sends the call to voicemail. Everything is in the same network. I have in sip.conf localnet=10.0.10.0/24 and in each UIP200 sip profile nat=never What's wrong? I have the same configuration in * 1.0.9 and it works just fine. Could the SIP protocol be broken in 1.2b2? Thanks, Waldo ___ --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 ___ --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 ___ --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] One Touch Record in 1.2
That would be great - Waldo On Nov 7, 2005, at 7:36 AM, Warren Burstein wrote: Tim Litwiller wrote: Well, I'd like them to drop in my voicemail when done recording - maybe in a separate recordings folder but I'd like to use the same interface to play them back. I would like that, too. Is anyone working on it? If not, I will put it on my TODO list. ___ --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] Dropping last digit from phone number
Matt Riddell napisał(a):
You could try ${EXTEN:-LEN(${EXTEN:1})}
When I have 61* number, it isn't the same as ${EXTEN:-2}?
But it isn't work anyway...
--
Best regards,
Bartosz Piec
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Re: [Asterisk-Users] FXS problems
On Monday 07 November 2005 08:03, bails wrote: I seem to be having some problems with the FXS modules on i, for example when i dial 90044117XX Nov 7 13:01:01 VERBOSE[2516]: -- Starting simple switch on 'Zap/1-1' Nov 7 13:01:05 DEBUG[2516]: DTMF digit: 9 on Zap/1-1 Nov 7 13:01:07 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:07 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:08 DEBUG[2516]: DTMF digit: 6 on Zap/1-1 Nov 7 13:01:12 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:12 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:12 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:13 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:13 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:13 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:14 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Which indicates to me that the FXS module is not getting all the signalling, as numbers are missing I have added relaxdtmf=yes to my zapata.conf but this seems not to help atall. Don't play with relaxdtmf. Could this be a hardware failure? Perhaps, but before you do that please post your relevant parts of zaptel.conf, zapata.conf and also the the output of the following little stanza: rmmod wctdm zaptel dmesg -c [ ignore any output until this point, I want the output from this point downward ] modprobe wctdm ztcfg -v dmesg -c That will tell me how the module's loading. What country are you in? -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropping last digit from phone number
Matt Riddell napisał(a):
What does that result in?
I have this in extensions.conf:
exten = _XX*,1,NoOp(${EXTEN:-LEN(${EXTEN:1})})
And when I dial 61*, the result in Asterisk console is:
-- Executing NoOp(SIP/65-aad1, 61*) in new stack
just like with ${EXTEN}.
--
Best regards,
Bartosz Piec
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Re: [Asterisk-Users] Timestamps in Console?
[EMAIL PROTECTED] wrote on 11/03/2005 11:49:12 AM: Use 'timestamp=yes' in asterisk.conf instead of -T. This is exactly what I was looking for. Thank you very much! Tim Massey ___ --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 as SIP gateway
How can i do that? 2005/11/7, Mark Phillips [EMAIL PROTECTED]: Drop the authentication elements from the SIP stanza that refers to the devices you want to use. Miloš Kocbek wrote: I want to enable access to some context in asterisk without authentication. The only limitation is ip number. All possible extension can make calls. How can i do that? greetings mk ___ --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 -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.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] Dropping last digit from phone number
Matt Riddell napisał(a): What do you want 61* to become? Checking the voicemail of 61 number. -- Best regards, Bartosz Piec ___ --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] Dropping last digit from phone number
exten = _XX*,1,NoOp(${EXTEN:0:-1})
?
Bartosz Piec wrote:
Matt Riddell napisał(a):
What does that result in?
I have this in extensions.conf:
exten = _XX*,1,NoOp(${EXTEN:-LEN(${EXTEN:1})})
And when I dial 61*, the result in Asterisk console is:
-- Executing NoOp(SIP/65-aad1, 61*) in new stack
just like with ${EXTEN}.
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Re: [Asterisk-Users] Dropping last digit from phone number
Erik napisał(a):
exten = _XX*,1,NoOp(${EXTEN:0:-1})
exten = _XX*,1,NoOp(${EXTEN:0:2})
:)
It works, thanks.
--
Best regards,
Bartosz Piec
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Re: [Asterisk-Users] One Touch Record in 1.2
You're going to need to do more than just putting the recorded media file into the voicemail folder hierarchy if you want the apps to recognize them. You will need to accompany them with their respective .txt file so the voicemail system and various web interface tools recognize them as files that are associated with voicemail. On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: That would be great - Waldo On Nov 7, 2005, at 7:36 AM, Warren Burstein wrote: Tim Litwiller wrote: Well, I'd like them to drop in my voicemail when done recording - maybe in a separate recordings folder but I'd like to use the same interface to play them back. I would like that, too. Is anyone working on it? If not, I will put it on my TODO list. ___ --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] Use of Queues and agents to provide office phone coverage.
Hi,
I have a small office that is one multiple stories with people doing
multiple jobs. Since people come and go all day long I was thinking of
implementing a Queue and agent scheme for handling the incoming calls.
Basically as people come in to the office they would log into the queue
(actually automatic login and logout would be preferred but I do not
know if this can be done without shutting the PC off - we are using SIP
clients on the PC, SJPhone) and as they leave they would be logged off.
The queue would randomly transfer calls to any open user and if no user
is available it would go to an automated operator (so that voice mail
can be left in the appropriate place.) My question is is this an
appropriate use of queues and is there any limitation in version 1.0.9
Asterisk to using this? What do I need to do in the extensions file to
make this happen??
Here is my Agent code:
[agents]
ackcall=yes
wrapuptime=0
musiconhold = default
updatecdr=yes
;Operator - Home
group=1
agent = 1300,1300,Name
agent = 1301,1301,Name
agent = 1302,1302,Name
agent = 1303,1303,Name
agent = 1304,1304,Name
agent = 1305,1305,Name
agent = 1306,1306,Name
;Operator - Spa
group = 1
agent = 1400,1400,Name
;Operator - Rest
group=2
agent = 1500,1500,Name
Here is my Queue code:
[general]
[default]
;Operator Home
[Q100]
music=default
strategy=ringall
maxlen=0
context=internal-home
member = Agent/@1
;Operator Resturant
[Q110]
music=default
strategy=ringall
maxlen=0
context=internal-rest
member = Agent/@2
A default file that is included in the extension.conf file:
[default]
exten = s,1,Goto(default,100,1)
exten = t,1,Goto(default,100,1)
exten = 1,1,Goto(default,100,1)
;Operator queue, Operator Console, and Receptionist Phone
exten = 100,1,Answer
exten = 100,2,Queue(Q100|trn|||120)
;Office Personnel
exten = _30[0-6],1,Macro(stdexten,${EXTEN},SIP/${EXTEN})
;Voicemail Main
exten = 800,1,Answer
exten = 800,2,VoicemailMain
;Agent Login
exten = 801,1,AgentCallbackLogin
;Voice Conferencing
exten = _85X,1,Answer
exten = _85X,2,MeetMe(${EXTEN})
THANKS
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RE: [Asterisk-Users] how to conferencd in Asterisk
Nrk, Do some googling and try to find all this info on the Wiki site for Asterisk. SK -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of nr k Sent: Sunday, November 06, 2005 5:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] how to conferencd in Asterisk Hi all How ro enable conference in asterisk and also how to make call between sccp device and sip device is there any special config in asterisk. regards ramakrishnan.n __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.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 NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --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] bristuff-0.2.0-RC8n problems and kernel panic
Hi, I had some problems to with a quadBRI with a 2.6 kernel debian distro. Have you tried to insmod the zaptel.ko module instead of modprobing? It worked for me, hope it will work for you too. Giorgio Incantalupo Remco Barende wrote: Hi list! On a newly installed RHEL 4 box I'm trying to install bristuff-0.2.0-RC8n. Everything did compile but I am running into some problems with the zaphfc driver. First of all when I load zaphfc *before* zaptel (yes I know I shouldn't do that) I get a kernel panic and the box hangs. Not so nice, especially when you are trying to fix stuff from remote locations. But ok. Now for the real trouble, when I do make load in zaphfc I get this: make -C /usr/src/linux-2.6 SUBDIRS=/tmp/bristuff-0.2.0-RC8n/zaphfc ZAP=-I/tmp/bristuff-0.2.0-RC8n/zaptel-1.0.9 modules make[1]: Entering directory `/usr/src/kernels/2.6.9-11.EL-x86_64' Building modules, stage 2. MODPOST *** Warning: zt_register [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_receive [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_transmit [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_ec_chunk [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! *** Warning: zt_unregister [/tmp/bristuff-0.2.0-RC8n/zaphfc/zaphfc.ko] undefined! make[1]: Leaving directory `/usr/src/kernels/2.6.9-11.EL-x86_64' modprobe zaptel insmod ./zaphfc.ko ztcfg -v Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) 3 channels configured. Notice: Configuration file is /etc/zaptel.conf line 8: Unable to open master device '/dev/zap/ctl' I guess (hope) the warnings are nothing serious but the message about /dev/zap/ctl is. (I did read README.udev and added the lines.) Rebooting the box didn't help. And when I try to start asterisk: Aug 15 23:25:51 WARNING[6454]: chan_zap.c:933 zt_open: Unable to specify channel 1: No such device or address Aug 15 23:25:51 ERROR[6454]: chan_zap.c:6484 mkintf: Unable to open channel 1: No such device or address here = 0, tmp-channel = 1, channel = 1 Aug 15 23:25:51 ERROR[6454]: chan_zap.c:10329 setup_zap: Unable to register channel '1-2' Aug 15 23:25:51 WARNING[6454]: loader.c:345 ast_load_resource: chan_zap.so: load_module failed, returning -1 == Unregistered channel type 'Tor' == Unregistered channel type 'Zap' Aug 15 23:25:51 WARNING[6454]: loader.c:440 load_modules: Loading module chan_zap.so failed! Ideas anyone? Thanks! ___ 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] Cisco 7970
Hello I have a Cisco 7970 phone that when I was trying to reset it to factory defaults it rebooted and now is stuck in a constant loop of the lights flashing by going down the line pool one light at a time in a constant rotation. I have the firmware for the phone, but have no idea on how to load or it how to get this phone functioning again. I would definitely be willing to pay someone to help me get this thing back online, if someone can contact me either here or offlist to get this resolved I would appreciate it tremendously. Thanks Dan - Dan Levine [EMAIL PROTECTED] 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.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
Re: [Asterisk-Users] Cisco 7970
The 7970 when reset to factory will delete the firmware load leaving
just the bootloader.
1. Hold down the # key
2. Power it on
3. Keep holding the power key until the line keys blink orange down the
tree
4. Have the firmware files on your tftpserver when it boots
5. Put the load into the config file like so:
/devicePool
loadInformationTERM70.7-0-2-0S/loadInformation
versionStamp{21ECCF08-13DB-4EC5-8BCE-B177569C489B}/versionStamp
userLocale
nameEnglish_United_States/name
It will retrieve the firmware and boot.
-Greg
On Mon, 2005-11-07 at 09:50 -0500, Dan Levine wrote:
Hello
I have a Cisco 7970 phone that when I was trying to reset it to factory
defaults it rebooted and now is stuck in a constant loop of the lights
flashing by going down the line pool one light at a time in a constant
rotation.
I have the firmware for the phone, but have no idea on how to load or it
how to get this phone functioning again.
I would definitely be willing to pay someone to help me get this thing
back online, if someone can contact me either here or offlist to get
this resolved I would appreciate it tremendously.
Thanks
Dan
-
Dan Levine
[EMAIL PROTECTED]
877.CYTEXONE x 810
212.477.0990 x 810
212.208.6889 FAX
502 Laguardia Place, Suite 2B
New York, NY 10012
http://www.cytexone.com
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Re: [Asterisk-Users] One Touch Record in 1.2
BJ Weschke wrote: You're going to need to do more than just putting the recorded media file into the voicemail folder hierarchy if you want the apps to recognize them. You will need to accompany them with their respective .txt file so the voicemail system and various web interface tools recognize them as files that are associated with voicemail. Well, I'd like them to drop in my voicemail when done recording - maybe in a separate recordings folder but I'd like to use the same interface to play them back I would be happy with just having the recording emailed to the appropriate user. I'm guessing that should be able to be done in the dial plan. Anyone have an example doing this? Thanks, Darrick -- Darrick Hartman DJH Solutions, LLC ___ --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] upgrade to 1.2 beta 2 issue
[EMAIL PROTECTED] wrote: Ever since I upgraded to beta2, the console is littered with these kind of messages: NOTICE[206]: chan_iax2.c:5654 update_registry: Restricting registration for peer 'kkai13' to 60 seconds (requested 0) Any way to suppress this? Of course! Fix your IAX2 client to stop requesting a registration expiry interval of zero seconds, since that's obviously a silly thing to request. ___ --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] Voicemail
The text sent on this notificationscan be found in voicemail.conf Hope this helps. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Andrew Nowrot |Sent: Monday, November 07, 2005 5:54 AM |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Voicemail | |Hi, | |I'm trying to translate the voicemail application to my local |language. I want to translate the notification email which |Asterisk send when you have new massages. Where I can find this file ?? | |Cheers to all | |Andrew |___ |--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] TDM400 FXO vs FXS Interrupt performance
[EMAIL PROTECTED] wrote: I was pretty unhappy to see that the new cards had RJ12 sockets - you can put RJ12 into RJ45, but not the other way round... But I do know that a lot of people would ask if RJ12 would fit, so it might have been to cut down on support calls. Definitely not :-) It was done to appease the certification officials in a couple of places, including (IIRC), an unnamed carrier in the land 'down under' G ___ --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] Re: Asterisk-Users Digest, Vol 16, Issue 44
The 104d has been available fora fewweeks. I've had one for 4 weeks working with Sangoma on the driver side. My echo issues are a thing of the past. I had a few issues configuring, but it turned out to be Asterisk configuration- not Sangoma configuration. You must download their latest drivers. Their tech support is a 9.9 out of 10. These guys know what they are doing. I'm trying to run a business here. Why should I compromise my business with Digium's 406P solution if they take 4 days to get back to me, or sell me soemthing that doesn't work in the machine Ichose to use? What I really want is the Sangoma analog FXS/FXO hardware!!! Pat Yahoo! FareChase - Search multiple travel sites in one click. ___ --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] FXS problems
Andrew Kohlsmith wrote: On Monday 07 November 2005 08:03, bails wrote: I seem to be having some problems with the FXS modules on i, for example when i dial 90044117XX Nov 7 13:01:01 VERBOSE[2516]: -- Starting simple switch on 'Zap/1-1' Nov 7 13:01:05 DEBUG[2516]: DTMF digit: 9 on Zap/1-1 Nov 7 13:01:07 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:07 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:08 DEBUG[2516]: DTMF digit: 6 on Zap/1-1 Nov 7 13:01:12 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:12 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:12 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:13 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:13 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:13 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Nov 7 13:01:14 DEBUG[2516]: DTMF digit: 0 on Zap/1-1 Which indicates to me that the FXS module is not getting all the signalling, as numbers are missing I have added relaxdtmf=yes to my zapata.conf but this seems not to help atall. Don't play with relaxdtmf. OK taken out Could this be a hardware failure? Perhaps, but before you do that please post your relevant parts of zaptel.conf, zapata.conf and also the the output of the following little /etc/zaptel.conf # Span 1: WCTDM/0 Wildcard TDM400P REV E/F Board 1 fxoks=1 fxsks=2 # channel 3, WCTDM, inactive. # channel 4, WCTDM, inactive. # Global data loadzone= uk defaultzone = uk /etc/asterisk/zapata.conf ; Zapata telephony interface ; ; Configuration file [trunkgroups] [channels] language=en context=from-pstn signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes ;relaxdtmf=yes usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=5 txgain=5 group=0 callgroup=1 pickupgroup=1 immediate=no cidsignalling=v23 cidstart=polarity ;faxdetect=both faxdetect=incoming ;faxdetect=outgoing ;faxdetect=no ;Include AMP configs #include zapata_additional.conf ;Include genzaptelconf configs #include zapata-auto.conf stanza: rmmod wctdm zaptel dmesg -c [ ignore any output until this point, I want the output from this point downward ] modprobe wctdm ztcfg -v dmesg -c Zaptel Configuration == 2 channels configured. dmesg -c Freed a Wildcard PCI: Found IRQ 12 for device 00:0a.0 PCI: Sharing IRQ 12 with 00:10.1 Freshmaker version: 71 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXO (FCC mode) Module 2: Not installed Module 3: Not installed Found a Wildcard TDM: Wildcard TDM400P REV E/F (2 modules) Registered tone zone 4 (United Kingdom) Registered tone zone 4 (United Kingdom) That will tell me how the module's loading. What country are you in? UK, this card was working correctly for over 1 year. Thanks in advance Bails -A. ___ --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] Cisco 7970
Hi Greg,
Would you mind a telephone call to help me with the final steps?
-
Dan Levine
[EMAIL PROTECTED]
877.CYTEXONE x 810
212.477.0990 x 810
212.208.6889 FAX
502 Laguardia Place, Suite 2B
New York, NY 10012
http://www.cytexone.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Greg
Oliver
Sent: Monday, November 07, 2005 10:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Cisco 7970
The 7970 when reset to factory will delete the firmware load leaving
just the bootloader.
1. Hold down the # key
2. Power it on
3. Keep holding the power key until the line keys blink orange down the
tree
4. Have the firmware files on your tftpserver when it boots
5. Put the load into the config file like so:
/devicePool
loadInformationTERM70.7-0-2-0S/loadInformation
versionStamp{21ECCF08-13DB-4EC5-8BCE-B177569C489B}/versionStamp
userLocale
nameEnglish_United_States/name
It will retrieve the firmware and boot.
-Greg
On Mon, 2005-11-07 at 09:50 -0500, Dan Levine wrote:
Hello
I have a Cisco 7970 phone that when I was trying to reset it to
factory
defaults it rebooted and now is stuck in a constant loop of the lights
flashing by going down the line pool one light at a time in a constant
rotation.
I have the firmware for the phone, but have no idea on how to load or
it
how to get this phone functioning again.
I would definitely be willing to pay someone to help me get this thing
back online, if someone can contact me either here or offlist to get
this resolved I would appreciate it tremendously.
Thanks
Dan
-
Dan Levine
[EMAIL PROTECTED]
877.CYTEXONE x 810
212.477.0990 x 810
212.208.6889 FAX
502 Laguardia Place, Suite 2B
New York, NY 10012
http://www.cytexone.com
___
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http://lists.digium.com/mailman/listinfo/asterisk-users
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[Asterisk-Users] Help needed for Onhold calls
Good Day list, I have read wiki pages I have googled to death and am getting no closer to understanding the methodology of onhold music. Maybe I am trying to do something that is just not possible: Here is my desire. 1) Call comes in to the asterisk box via Zap channel 2) call is answered by SIP/100 3) call is parked 1) Sip/200 unparks the call and places the caller on hold (by pressing hold button on the SIP Phone) I would like to have any callers that have been placed on hold from this extension to hear musicclass SALES Repeat steps 1-3 above 1) Sip/300 unparks the call and places the caller on hold (by pressing hold button on the SIP Phone) I would like to have any callers that have been placed on hold from this extension to hear musicclass SUPPORT I have found discrepancy in the source code between using musicclass and musiconhold therefore I have tried both of them individually and simultaneously. PS I know the different classes of music are working because I can specify them to be used in the queues I have set up. Bottom line is Can I specify the music class that a caller hears based upon WHO puts them on hold?: Thanks for your assistance Musiconhold.conf ; ; Music on hold class definitions ; [classes] default = quietmp3:/var/lib/asterisk/mohmp3 general = quietmp3:/var/lib/asterisk/mohmp3/general Support = quietmp3:/var/lib/asterisk/mohmp3/Support Sales = quietmp3:/var/lib/asterisk/mohmp3/Sales Sip.conf [general] port = 5060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) disallow=all allow=ulaw allow=alaw context = from-sip-external ; Send unknown SIP callers to this context callerid = Unknown musicclass=general [200] . . . musiconhold=Sales musicclass=Sales [300] . . musiconhold=Support musicclass=Support oledata.mso Description: Binary data oledata.mso Description: Binary data oledata.mso Description: Binary data ___ --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 1.2beta2 and UIP200
Additionally: *CLI sip show peer 100074 * Name : 100074 Secret : Set MD5Secret: Not set Context : qa Subscr.Cont. : Not set Language : en AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 0 Call limit : 0 Dynamic : Yes Callerid : Waldo Rubinstein 211 Expire : 11077 Insecure : no Nat : No ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.0.10.236 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 100074 SIP Options : (none) Codecs : 0x6 (gsm|ulaw) Codec Order : (ulaw,gsm) Status : UNREACHABLE Useragent: Uniden SIP Phone p2 Ver BS4.63 Reg. Contact : sip:[EMAIL PROTECTED]:5060 Thanks, Waldo On Nov 6, 2005, at 11:11 PM, C F wrote: can you post the sip.conf for that uip200? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: When I dial the extension, I get this: -- Executing Dial(IAX2/gateway0-16386, SIP/100074|20) in new stack == Everyone is busy/congested at this time (1:0/0/1) When I do a sip show peer 100074, everything it shows matches the results of executing the same sip show peer on * 1.0.9 and 1.2b1, except: Status : UNREACHABLE However, I can make any type of calls from them phone. I can ping the phone from the * server. It's just that * 1.2b2 can't reach it, for some reason. Thanks, Waldo On Nov 6, 2005, at 1:37 PM, C F wrote: Whats the exact CLI output you are getting when calling that extension? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Nope. It isn't active. I even factory reseted the phone but still the same. One more piece of information: it works just fine in 1.2b1. I beginning to think it could be a bug in 1.2b2. Any other ideas/suggestions? Thanks, Waldo On Nov 5, 2005, at 9:10 PM, C F wrote: You sure that the DND (Do Not Disturb) button is not active on the UIP200? On 11/4/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro phones. All phones register fine with * and I can place outbound calls with no problem. I can call from the X-Pro to any other X-Pro. I can call from UIP200 to any other X-Pro. However, the UIP200 cannot receive calls. Every time I call the UIP200, the CLI says Everyone is Busy/ Congested and sends the call to voicemail. Everything is in the same network. I have in sip.conf localnet=10.0.10.0/24 and in each UIP200 sip profile nat=never What's wrong? I have the same configuration in * 1.0.9 and it works just fine. Could the SIP protocol be broken in 1.2b2? Thanks, Waldo ___ --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 ___ --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 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com
Re: Re: [Asterisk-Users] call from asterisk to SIP cisco 5300
sorry, i didnt write i have voip peer
so i have sloved thy problem, nubder like
#00#7091222
*00*7091222
*777
doesnt work
Cisco says
dpMatchPeersMoreArg: Match Dest. pattern; called ()
and when i tries to dial *777*777
it says
dpMatchPeersMoreArg: Match Dest. pattern; called (777)
But I cant understand why CISCO cant understand this MAGIC # and * :)
I think you should set dial-peer voice 21 voip with incoming called number
#00#..\* too, this catch this call and the dial peer 22 send it.
Adam
Cytowanie Ivan Vershigora [EMAIL PROTECTED]:
i dial on my phone to to 8091222
and convert it on asterisk to #00#7091222
But Cisco says 404
cisco peer=
!
dial-peer voice 22 pots
huntstop
preference 5
destination-pattern #00#..\*
translate-outgoing calling 1
direct-inward-dial
port 0:D
prefix 810
!
peer in sip.conf==
[krdvox]
context=from-sip
type=peer
host=123.123.123.123
canreinvite=yes
dtmfmode=inband
extensions.conf==
exten = _.,1,SetCallerID(861273 8612731107[|a])
exten = _.,2,Dial(SIP/#00#7${EXTEN:[EMAIL PROTECTED],60)
exten = _.,3,Congestion
Asterisk says===
-- Executing Dial(SIP/201-2966, SIP/[EMAIL PROTECTED]|60) in
new stack
-- Called [EMAIL PROTECTED]
-- Got SIP response 404 Not Found back from XXX.XXX.XXX.XXX
-- SIP/krdvox-3910 is circuit-busy
== Everyone is busy/congested at this time
===
==CISCO debug ccsip ===
Nov 3 16:10:03.516: Received:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6697eb34
From: 861273 sip:[EMAIL PROTECTED];tag=as74db268c
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: CSCO/6
Date: Thu, 03 Nov 2005 13:10:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 235
.
Nov 3 16:10:03.524: MatchNextPeer: Peer 999 matched
Nov 3 16:10:03.524: Using Voice Class Codec, tag=1
.
Disconnect Cause (SIP) : 404
===
Nov 3 16:10:03.524: MatchNextPeer: Peer 999 matched
Peer 999- wrong one !!!
why he cant find dial-peer voice 22
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Pozdrawiam,
Adam Rybak
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Re: [Asterisk-Users] Dropping last digit from phone number
Here is what I do:
${EXTEN:0:$[${LEN(${EXTEN})} - 1]}
that should give you for the following
exten = 123456789,1,Noop(${EXTEN:0:$[${LEN(${EXTEN})} - 1]})
12345678
Hope this helps.
On 11/7/05, Bartosz Piec [EMAIL PROTECTED] wrote:
Erik napisał(a):
exten = _XX*,1,NoOp(${EXTEN:0:-1})
exten = _XX*,1,NoOp(${EXTEN:0:2})
:)
It works, thanks.
--
Best regards,
Bartosz Piec
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Re: [Asterisk-Users] Asterisk 1.2beta2 and UIP200
The unreachable is the problem. Try adding a qualify=no to that sip entry. On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Additionally: *CLI sip show peer 100074 * Name : 100074 Secret : Set MD5Secret: Not set Context : qa Subscr.Cont. : Not set Language : en AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 0 Call limit : 0 Dynamic : Yes Callerid : Waldo Rubinstein 211 Expire : 11077 Insecure : no Nat : No ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.0.10.236 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 100074 SIP Options : (none) Codecs : 0x6 (gsm|ulaw) Codec Order : (ulaw,gsm) Status : UNREACHABLE Useragent: Uniden SIP Phone p2 Ver BS4.63 Reg. Contact : sip:[EMAIL PROTECTED]:5060 Thanks, Waldo On Nov 6, 2005, at 11:11 PM, C F wrote: can you post the sip.conf for that uip200? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: When I dial the extension, I get this: -- Executing Dial(IAX2/gateway0-16386, SIP/100074|20) in new stack == Everyone is busy/congested at this time (1:0/0/1) When I do a sip show peer 100074, everything it shows matches the results of executing the same sip show peer on * 1.0.9 and 1.2b1, except: Status : UNREACHABLE However, I can make any type of calls from them phone. I can ping the phone from the * server. It's just that * 1.2b2 can't reach it, for some reason. Thanks, Waldo On Nov 6, 2005, at 1:37 PM, C F wrote: Whats the exact CLI output you are getting when calling that extension? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Nope. It isn't active. I even factory reseted the phone but still the same. One more piece of information: it works just fine in 1.2b1. I beginning to think it could be a bug in 1.2b2. Any other ideas/suggestions? Thanks, Waldo On Nov 5, 2005, at 9:10 PM, C F wrote: You sure that the DND (Do Not Disturb) button is not active on the UIP200? On 11/4/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro phones. All phones register fine with * and I can place outbound calls with no problem. I can call from the X-Pro to any other X-Pro. I can call from UIP200 to any other X-Pro. However, the UIP200 cannot receive calls. Every time I call the UIP200, the CLI says Everyone is Busy/ Congested and sends the call to voicemail. Everything is in the same network. I have in sip.conf localnet=10.0.10.0/24 and in each UIP200 sip profile nat=never What's wrong? I have the same configuration in * 1.0.9 and it works just fine. Could the SIP protocol be broken in 1.2b2? Thanks, Waldo ___ --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 ___ --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] asterisks talking to asterisks
I have a request. I have a server in Texas And one in NJ. Is it possible for the system in Texas to log into the system in NJ so that Extensions can call each other? -J ___ --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 1.2beta2 and UIP200
Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1. Very strange. Anyway, thanks. - Waldo On Nov 7, 2005, at 10:57 AM, C F wrote: The unreachable is the problem. Try adding a qualify=no to that sip entry. On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Additionally: *CLI sip show peer 100074 * Name : 100074 Secret : Set MD5Secret: Not set Context : qa Subscr.Cont. : Not set Language : en AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 0 Call limit : 0 Dynamic : Yes Callerid : Waldo Rubinstein 211 Expire : 11077 Insecure : no Nat : No ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.0.10.236 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 100074 SIP Options : (none) Codecs : 0x6 (gsm|ulaw) Codec Order : (ulaw,gsm) Status : UNREACHABLE Useragent: Uniden SIP Phone p2 Ver BS4.63 Reg. Contact : sip:[EMAIL PROTECTED]:5060 Thanks, Waldo On Nov 6, 2005, at 11:11 PM, C F wrote: can you post the sip.conf for that uip200? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: When I dial the extension, I get this: -- Executing Dial(IAX2/gateway0-16386, SIP/100074|20) in new stack == Everyone is busy/congested at this time (1:0/0/1) When I do a sip show peer 100074, everything it shows matches the results of executing the same sip show peer on * 1.0.9 and 1.2b1, except: Status : UNREACHABLE However, I can make any type of calls from them phone. I can ping the phone from the * server. It's just that * 1.2b2 can't reach it, for some reason. Thanks, Waldo On Nov 6, 2005, at 1:37 PM, C F wrote: Whats the exact CLI output you are getting when calling that extension? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Nope. It isn't active. I even factory reseted the phone but still the same. One more piece of information: it works just fine in 1.2b1. I beginning to think it could be a bug in 1.2b2. Any other ideas/suggestions? Thanks, Waldo On Nov 5, 2005, at 9:10 PM, C F wrote: You sure that the DND (Do Not Disturb) button is not active on the UIP200? On 11/4/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro phones. All phones register fine with * and I can place outbound calls with no problem. I can call from the X-Pro to any other X-Pro. I can call from UIP200 to any other X-Pro. However, the UIP200 cannot receive calls. Every time I call the UIP200, the CLI says Everyone is Busy/ Congested and sends the call to voicemail. Everything is in the same network. I have in sip.conf localnet=10.0.10.0/24 and in each UIP200 sip profile nat=never What's wrong? I have the same configuration in * 1.0.9 and it works just fine. Could the SIP protocol be broken in 1.2b2? Thanks, Waldo ___ --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 ___ --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
Re: [Asterisk-Users] asterisks talking to asterisks
Yes. Most certainly. Take a look at IAX (Inter Asterisk eXchange) protocol to enable this functionality for you with minimal impact on your firewall/NAT setups. On 11/6/05, Jason Brashear [EMAIL PROTECTED] wrote: I have a request. I have a server in Texas And one in NJ. Is it possible for the system in Texas to log into the system in NJ so that Extensions can call each other? -J ___ --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] Change Asterisk User
Hi all, I would like to start asterisk with a different user than asterisk in order to use the same than my apache server. I have tried to change it in /etc/init.d/asterisk but when I change USER, asterisk doesnt start. Has someone already start asterisk under other user that asterisk? Thanks Amaury ___ --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] Change Asterisk User
there is a lot more to changing the user than just su'ing you need to change the permissions on a lot of files too. On Mon, 7 Nov 2005, Amaury BOSSE wrote: Hi all, I would like to start asterisk with a different user than asterisk in order to use the same than my apache server. I have tried to change it in /etc/init.d/asterisk but when I change USER, asterisk doesn't start. Has someone already start asterisk under other user that asterisk? Thanks Amaury -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- - - - Jason Pyeron PD Inc. http://www.pdinc.us - - Partner Sr. Manager 7 West 24th Street #100 - - +1 (443) 269-1555 Baltimore, Maryland 21218 - - - -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, purge the message from your system and notify the sender immediately. Any other use of the email by you is prohibited. ___ --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] compiling problems
Some problems happened with precompiled kernels. If you compile your own vanilla kernel, I'm sure that you haven't this issues. Remember, if you use 2.6 kernel, you can need udev and hotplug systems to better performance. I allways use Debian with vanilla kernel that I compile, and I haven't problems neither 2.4 nor 2.6 kernels on single or dual procesors (32 or 64bits) I hope that it helps you. FaberK wrote: The problem is the 2.6. I know that there is compability also with that kernel, but in my small experience, I've got not these problems with 2.4. Now, I've got to migrate Asterisk into a Dual Xeon 3.0 4Gb RAM. What distro would you use? Until now, I've tested CentOS 3.4 Server with no problem, but not on this kind of server. With Fedora 3, too many problems, concerning the kernel 2.6. Suggestions? Thanks 2005/11/6, Tzafrir Cohen [EMAIL PROTECTED]: On Sat, Nov 05, 2005 at 07:29:18PM +0100, FaberK wrote: Fedora Core 3 kernel-0-2.6.9-1.667 and kernel-2.6.12-1.1380 (same results) Sangoma 102 Concerning udev, I've read that it uses hotplug and if I'm not wrong, I remember that zaptel got conflicts with hotplug. But maybe I'm confusing (terrible headache!) Thanks a lot! zaptel should not conflict with hotplug if the specific hardware driver module is well-written (e.g: declares PCI IDs it will identify). This will mean that hotplug will try using it automatically. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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 -- .:FaberK:. ___ --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] Change Asterisk User
If you give us more info it is easier to help. For example, if you are using a standard debian sarge setup I could help you and be sure to give you the right advice. However you might want to think carefully about this type of change. There are other approaches such as setting ownership and permissions for files and directories the webserver needs access to. Amaury BOSSE wrote: Hi all, I would like to start asterisk with a different user than “asterisk” in order to use the same than my apache server. I have tried to change it in /etc/init.d/asterisk but when I change USER, asterisk doesn’t start. Has someone already start asterisk under other user that “asterisk”? Thanks Amaury ___ --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] bristuff-0.2.0-RC8n problems and kernel panic
On Mon, Nov 07, 2005 at 03:43:03PM +0100, gincantalupo wrote: Hi, I had some problems to with a quadBRI with a 2.6 kernel debian distro. Have you tried to insmod the zaptel.ko module instead of modprobing? It worked for me, hope it will work for you too. Giorgio Incantalupo Could you please give more details? One thing you should try to do is remove the automatic run of ztcfg at module load time. Practically: rem-out all the lines in /etc/modprobe.d/zaptel . There is some black-magic claim that if you un ztcfg more than once it may cause a problem to a configured zaphfc module. Don't forget to run ztcfg manually (or in an init.d script) later. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] Change Asterisk User
On Mon, Nov 07, 2005 at 05:27:00PM +0100, Amaury BOSSE wrote: Hi all, I would like to start asterisk with a different user than asterisk in order to use the same than my apache server. Hmmm, you basically need to run apache's user to the Asterisk group. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --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] References?
Matt, Sorry for the response off-list... Would you be willing to talk to the powers that be for about 30 minutes about your experiences with Asterisk? I don't know what questions they're planning to ask, but they're likely to be centered around reliability and supportability as those are their major paranoia points. -Chad On Nov 3, 2005, at 1:08 PM, Matt wrote: I can not say that we are using it for a call center, as we use a NorHell switch for that.. but we will be migrating to Asterisk. However, we do use it to provide VoIP to all of our customers, and even customers on other broadband networks. On 11/3/05, Chad Scott [EMAIL PROTECTED] wrote: All, I've been pushing hard for the use of Asterisk for the corporate phone solution at the company I work for. Unfortunately, this decision is completely out of my hands, although I've been applying gentle influence and pressure where I can. The management for this project would like reference accounts that utilize Asterisk for their telephony solution and are happy with it. Ideally, the reference accounts would be around 500 seats in size and have some sort of call center and/or outbound sales calling. Anyone want to volunteer for this? If I can get Asterisk in here it would be HUGE but this is currently standing in my way. I know there *must* be installations out there this size and larger, I just don't know who they are... help me out! Thanks, -Chad ___ --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
Re: [Asterisk-Users] References?
Hrm. Perhaps I should have actually responded off-list... DOH! :D On Nov 7, 2005, at 9:11 AM, Chad Scott wrote: Matt, Sorry for the response off-list... Would you be willing to talk to the powers that be for about 30 minutes about your experiences with Asterisk? I don't know what questions they're planning to ask, but they're likely to be centered around reliability and supportability as those are their major paranoia points. -Chad ___ --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] Help with dialplan to allow breakout to DISA
I'm trying to set-up a dialplan for incoming calls that allows a breakout by pressing something like *. Users would then be able to get an inside dial tone for voicemail, outgoing calls, etc. I've been struggling with Waitexten(), Disa() in the dialplan but not having much luck. Are there any good documents out there to assist me in this? Frank ___ --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] Help with dialplan to allow breakout to DISA
I do it this way: exten = *, 1, Authenticate(PASSWORD) exten = *, 2, DISA(no-password|DESTINATION_CONTEXT) exten = *, 3, Hangup It seems to work fine... -Rusty On 11/7/05, Frank Tarczynski [EMAIL PROTECTED] wrote: I'm trying to set-up a dialplan for incoming calls that allows a breakoutby pressing something like *.Users would then be able to get an insidedial tone for voicemail, outgoing calls, etc.I've been struggling with Waitexten(), Disa() in the dialplan but not having much luck.Are there any good documents out there to assist me in this?Frank___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://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] sill looking for a provider
On 11/06/05 02:31 Dustin Goodwin said the following: Of course it's hard for me to see the return route with traceroute. I assume the return path from their host takes on some bizarre route that adds a lot of latency. try a traceroute with lft. lft gives you the different AS/BGP routers your packet will pass thru, and is a good tool to isolate latency problems. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0)http://www.alphaque.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --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] TDM400 FXO vs FXS Interrupt performance
Most modern installations/buildings are wired with RJ45, as are the patch panels. RJ12 is a real pain - I had to chop up patch leads and put RJ12 sockets on the end. Very messy and a waste of time. On Sun, 6 Nov 2005 22:04:48 -0500, Andrew Kohlsmith wrote: On Sunday 06 November 2005 21:46, [EMAIL PROTECTED] wrote: I was pretty unhappy to see that the new cards had RJ12 sockets - you can put RJ12 into RJ45, but not the other way round... You've gotta be shitting me. Why on earth do you want RJ45 jacks for POTS connections? Sure it fits but it's a loose fit to start and you get absolutely zero advantages unless you count being able to make a screwy cable a good thing. :-) ___ --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] CentOS vs. Vanilla Kernel
HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off). OS: CentOS 4.2 Dual Embedded NIC enabled USB disabled serial disabled printer disabled 2x73GB SCSI in HW Raid 1 What is the opinion of this fine list - should I use the default CentOS kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable (2.6.14) Anyone got any clues / hints / tips on what should go into the kernel ? All views and comments appreciated :) Julian. ___ --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] AGI environment dump callerid
Hi,
Since * 1.2-beta1 (incl CVS HEAD) there is a change in the
callerid's output to STDERR when an AGI environment
dump is requested:
Asterisk CVS HEAD built by root @ chick on a i686 running
Linux on 2005-11-06 16:35:14 UTC
AGI Environment Dump:
-- accountcode =
-- callerid = 1234689
-- calleridname = Callee Name
-- callingani2 = 0
-- callingpres = 0
-- callingtns = 0
-- callington = 0
-- channel = SIP/22-f55e
-- context = default
-- dnid = 19147858756
-- enhanced = 0.0
-- extension = 19147858756
-- language = en
-- priority = 1
-- rdnis = unknown
-- request = dump.agi
-- type = SIP
-- uniqueid = 1131381756.13
but ... Connected to Asterisk 1.0.9 currently running on dog
(pid = 28360)
AGI Environment Dump:
-- accountcode =
-- callerid = Callee Name 1234689
-- channel = SIP/22-9351
-- context = default
-- dnid = 19147858756
-- enhanced = 0.0
-- extension = 19147858756
-- language = en
-- priority = 1
-- rdnis = unknown
-- request = dump.agi
-- type = SIP
-- uniqueid = 1131381457.0
Thus my question was which is the future-to-be callerid
format?
1. -- callerid = 1234689
-- calleridname = Callee Name
OR
2. -- callerid = Callee Name 1234689
Nothing wrong with that in general since clid, as
${CDR(clid)}, is still being written correctly in 1.0.7,
1.0.9,
1.2-beta12 and CVS HEAD in the usual cdr database/table,
and in any custom table through
$dbh-quote($callerid).
However, since * 1.2-beta1 (incl CVS HEAD), when
AGI(perl) script try $callerid=$input{callerid} it results
to $dbh-quote($callerid) calleridnum(by
default it appears eq to callerid), only.
/* Obviously, because in res_agi.c $Revision: 1.53 $:
fdprintf(fd, agi_callerid: %s\n, chan-cid.cid_num ?
chan-cid.cid_num : unknown);
fdprintf(fd, agi_calleridname: %s\n, chan-cid.cid_name ?
chan-cid.cid_name : unknown); */
Changing to $callerid=$input{calleridname} is inserted as
requested.
Trying to group both callerid attributes results in an empty
string.
Playing with the dilaplan yet damages ${CDR(clid)}
record.
Any thoughts?
benchev
-
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Re: [Asterisk-Users] TDM400 FXO vs FXS Interrupt performance
On Monday 07 November 2005 12:57, George Gardiner wrote: Most modern installations/buildings are wired with RJ45, as are the patch panels. RJ12 is a real pain - I had to chop up patch leads and put RJ12 sockets on the end. Very messy and a waste of time. We just moved in to a new building. While you're right in the sense that there's cat5 and rj45 everywhere, *every* phone port is RJ11. I've never seen it otherwise. Up in the equipment room the telco is all terminated to BIX, and there are special BIX strips that have BIX on the back and 12 (I think) RJ11 on the front. There are also similar BIX strips that do 6 RJ45 on the front but our data termination is all done on 19 patch panels with BIX on the back and 24 (I think) RJ45 on the front. -A. ___ --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] Speex codec problems
I am trying to tweak my Asterisk servers to talk to each other using Speex codec. I downloaded and installed speex and speex devel libraries, recompiled asterisk (including make clean), did set up speex codec as only one allowed on both sides. Sounds enough. However, conversations are not Speex encoded!!! It is codec 64 (16 bit Signed Linear PCM) all the time. Any clue as to why Asterisk don't want to kick in Speex into play? BTW One asterisk (initiator) is HEAD version, another is asterisk-1.0.9. Any help is wappreciated. ___ --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] asterisk-1.2-bêta2 | pres ence/subscription support in the SIP channel driver
Hello, I configure Polycom ip300 for presence but when status change notify is no sent to subscriber !? Why ? Regards 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
[Asterisk-Users] Re: Help with dialplan to allow breakout to DISA
I have my dialplan setup the same, only with 0 instead of * as the extension. What would the reason be, after authenticating, that I get a dialtone, as expected, but no response to any DTMF tones I input? It is as if the DISA works, gives me tone, but is unresponsive? The destination context is exactly the same as any of my internal extensions, too... I do it this way: exten = *, 1, Authenticate(PASSWORD) exten = *, 2, DISA(no-password|DESTINATION_CONTEXT) .exten = *, 3, Hangup.. It seems to work fine... -Rusty Sincerely, Brent A. Torrenga [EMAIL PROTECTED] Torrenga Engineering, Inc. 907 Ridge Road Munster, Indiana 46321-1771 219.836.8918 Voice 219.836.1138 Facsimile ___ --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 1.2beta2 and UIP200
I guess that somewhere in your settings you have a qualify on, or that 1.2 has it on by default. Do the following: cd /etc/asterisk grep .*qualify.* ./* and see the output, if the only line that has qualify is that qualify=no, then this looks like a bug to me. Please report back. On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1. Very strange. Anyway, thanks. - Waldo On Nov 7, 2005, at 10:57 AM, C F wrote: The unreachable is the problem. Try adding a qualify=no to that sip entry. On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Additionally: *CLI sip show peer 100074 * Name : 100074 Secret : Set MD5Secret: Not set Context : qa Subscr.Cont. : Not set Language : en AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : [EMAIL PROTECTED] VM Extension : asterisk LastMsgsSent : 0 Call limit : 0 Dynamic : Yes Callerid : Waldo Rubinstein 211 Expire : 11077 Insecure : no Nat : No ACL : No CanReinvite : No PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.0.10.236 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Def. Username: 100074 SIP Options : (none) Codecs : 0x6 (gsm|ulaw) Codec Order : (ulaw,gsm) Status : UNREACHABLE Useragent: Uniden SIP Phone p2 Ver BS4.63 Reg. Contact : sip:[EMAIL PROTECTED]:5060 Thanks, Waldo On Nov 6, 2005, at 11:11 PM, C F wrote: can you post the sip.conf for that uip200? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: When I dial the extension, I get this: -- Executing Dial(IAX2/gateway0-16386, SIP/100074|20) in new stack == Everyone is busy/congested at this time (1:0/0/1) When I do a sip show peer 100074, everything it shows matches the results of executing the same sip show peer on * 1.0.9 and 1.2b1, except: Status : UNREACHABLE However, I can make any type of calls from them phone. I can ping the phone from the * server. It's just that * 1.2b2 can't reach it, for some reason. Thanks, Waldo On Nov 6, 2005, at 1:37 PM, C F wrote: Whats the exact CLI output you are getting when calling that extension? On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Nope. It isn't active. I even factory reseted the phone but still the same. One more piece of information: it works just fine in 1.2b1. I beginning to think it could be a bug in 1.2b2. Any other ideas/suggestions? Thanks, Waldo On Nov 5, 2005, at 9:10 PM, C F wrote: You sure that the DND (Do Not Disturb) button is not active on the UIP200? On 11/4/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro phones. All phones register fine with * and I can place outbound calls with no problem. I can call from the X-Pro to any other X-Pro. I can call from UIP200 to any other X-Pro. However, the UIP200 cannot receive calls. Every time I call the UIP200, the CLI says Everyone is Busy/ Congested and sends the call to voicemail. Everything is in the same network. I have in sip.conf localnet=10.0.10.0/24 and in each UIP200 sip profile nat=never What's wrong? I have the same configuration in * 1.0.9 and it works just fine. Could the SIP protocol be broken in 1.2b2? Thanks, Waldo ___ --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 ___ --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
Re: [Asterisk-Users] asterisk as SIP gateway
Miloš Kocbek wrote: I want to enable access to some context in asterisk without authentication. In sip.conf: [username] type=friend host=x.x.x.x context=context_for_this_user -- Regards, Peter Petrov [EMAIL PROTECTED] ___ --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] CentOS vs. Vanilla Kernel
my $0.02 if you are going w/ RHEL use one of the kernel rpms provided. You can always add a module rpm to supplement it. Once you roll your own there might be better distros for you, since you are going to break the rpm/up2date features that make RHEL a desirable product. On Mon, 7 Nov 2005, Julian Lyndon-Smith wrote: HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off). OS: CentOS 4.2 Dual Embedded NIC enabled USB disabled serial disabled printer disabled 2x73GB SCSI in HW Raid 1 What is the opinion of this fine list - should I use the default CentOS kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable (2.6.14) Anyone got any clues / hints / tips on what should go into the kernel ? All views and comments appreciated :) Julian. ___ --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 -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- - - - Jason Pyeron PD Inc. http://www.pdinc.us - - Partner Sr. Manager 7 West 24th Street #100 - - +1 (443) 269-1555 Baltimore, Maryland 21218 - - - -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, purge the message from your system and notify the sender immediately. Any other use of the email by you is prohibited. ___ --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] CentOS vs. Vanilla Kernel
On Mon, 2005-11-07 at 18:17 +, Julian Lyndon-Smith wrote: What is the opinion of this fine list - should I use the default CentOS kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable (2.6.14) Anyone got any clues / hints / tips on what should go into the kernel ? All views and comments appreciated :) Depends. Do you want to spend your time using the system and working on Asterisk, or do you want to spend your time tracking kernel changes, patching security fixes, tracking down kernel bugs, breaking rpm deps and working around that, etc, etc, etc... Red Hat puts a lot of work into making sure their kernel is solid and secure. They backport security fixes and bug fixes into their stable tree, 2.6.9. In my opinion, I'd rather let the folks that know the kernel work on it rather than spend my limited time on it. -- Jesse Keating GameHouse -- Systems Engineer ___ --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] Re: Double DTMF with tdm card
Just wanted to let the group know this problem is fixed (for me). Mark log-on to my system and found a bug in chan_zap.c on Saturday night and made the correction - I believe the change is available for download by now at zaptel 1.0.9.2, or CVS Head. He stated that recent changes unmask the bug and the change will slightly improve TE410P performance Thanks for you help! Bart - Original Message - From: Bart Fisher [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 6:20 PM Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card I just heard back from Mark. I volunteered my system to used for testing. From Mark: Generally, issues which involve Digium hardware should go through technical support, even if it's a newly introduced problem, because they can help narrow down the nature of the failure, what might have changed, etc. If you or a representative of this group want to fill this role instead, I'm happy to work with you, but I need the situation labbed up in an environment where the problem can be demonstrated, where I can remotely log in, and where I can edit, recompile, and test in real time (i.e. not on a production server). If you want to set all this up and contact me with login details and a number where I can see the problem occur, then when it's ready, I can work with you directly. Mark Bart - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 2:41 PM Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card If in fact it is the exact same issue, then I'd suggest creating a feature request to add disable dtmf detection after answer supervision and post it to the -dev list (which is what Kevin is suggesting now). You will need to be explain the wanted functionality in terms that non-telephone technical folks can understand. I'd suggest a zapata.conf configuration option that is something like ignore-dtmf-after-answersup with a default value of however it works today (=no). Think about that carefully as the option set to =yes will disable dtmf from interacting with your internal * ivr (assuming you have one). What you want is kind of related to a pass-thru connection and not necessarily for a connection terminating within *. There might be other ways to handle your objective. This same issue comes up in other cases where interaction with an external ivr is needed, some airlines automated systems, etc. I honestly believe the exact same thing should apply to iax2 incoming trunks as well. Not so sure about sip trunks. I'd agree with your statement relative to digium support being contacted, but if the boss-man suggests it, there might be an unstated reason for that. If properly worded (and with the supporting documentation that you heard the problem with a T1 analyzer), they might be able to help support the need for some kind of option. This is exactly what is happening... It's bad news... In my case the T1 is connected to a PBX Voice Mail. So, double dialing really messes up thing like when entering a passcode. Where passcode 1234 arrives as 11223344 - no good. This would always be an issue in cases where the call is Tandem thru Asterisk. In fact, I can't see any reason to repeat the digits when the signal is inband and/or Zap Bridged call. - And why was it changed from 1.0.9? Makes no sense. It seems an easy fix, maybe a digit time-out parameter or disable sending after answer supervision has been achieved. Given what you say, Digium Support won't be able to fix without code changes - I don't know what Mark is thinking here. Bart - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 03, 2005 1:17 PM Subject: Re: [Asterisk-Users] Re: Double DTMF with tdm card I might be able to shed a little light on this... Asterisk is constantly listening for dtmf tones on most channels. Its either listening for inband or rfc-out-of-band, depending upon how the attached device is defined and how asterisk def's for that device is defined. For pstn interfaces, the cards don't listen for any dtmf, but rather the zap sutff is listening. If a call is generated from some external source (coming into *), the dtmf will be inband once a channel is answered. For commercial telephone equipment, once a channel is answered, the telephone equipment no longer listens for dtmf (its simply passed inband). Not so with asterisk, and this point has been argued with Mark some time ago; asterisk still listens and trys to handle the dtmf, translating to rfc2833 as it thinks is necessary. So, it sounds like you have an answered T1 call where * is still
RE: [Asterisk-Users] Change Asterisk User
Thanks for your answer, I am working on Debian Sarge but I have compiled Astersik 1.0.9 myself without .deb Packages. I need to access to voicemail and sound files from my web-interface (php and cgi/perl) but I can't change Apache user because of others applications. Asterisk creates files under Asterisk user and I have to access them from www-data user. Do you have other solution? I have tried using sudo but it doesn't seem to work. Regards, Amaury ___ --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-1.2-bêta2 | presen ce/subscription support in the SIP channel driver
There could be 1 of 100 reasons that's causing this not to work. Let's start out by you posting your relevant sections of sip.conf and extensions.conf and then do a sip show subscriptions from the CLI and give us the results of that as well. On 11/7/05, harry gaillac [EMAIL PROTECTED] wrote: Hello, I configure Polycom ip300 for presence but when status change notify is no sent to subscriber !? Why ? Regards 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] CentOS vs. Vanilla Kernel
The default CentOS kernel has worked fine for me. Just an FYI; CentOS uses the RedHat EL kernel source to build... It's pretty heavily patched so if you want to use the latest stable, download the SRPMs from RedHat/CentOS and patch in the kernel.org patches. But yeah, stick with the CentOS kernel unless you have problems. -Ryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian Lyndon-Smith Sent: Monday, November 07, 2005 12:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] CentOS vs. Vanilla Kernel HW: HP DL360 1GB Ram Single 3GHz Pentium (with Hyper-threading turned off). OS: CentOS 4.2 Dual Embedded NIC enabled USB disabled serial disabled printer disabled 2x73GB SCSI in HW Raid 1 What is the opinion of this fine list - should I use the default CentOS kernel (2.6.9-22.0.1.EL) or download from kernel.org the latest stable (2.6.14) Anyone got any clues / hints / tips on what should go into the kernel ? All views and comments appreciated :) Julian. ___ --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] MP3 or OGG
You're probably not going to be violating any patent protections by using OGG instead of MP3. As far as compression goes, I've found the difference between the two of them to be negligible. I've always used OGG when possible to stay IP safe. On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: I'm trying to archive out call recordings and would appreciate some feedback as to which audio compression is more recommended MP3 or OGG. In the past, I've use lame to convert to MP3, but I noticed the audio volume drops significantly. Is it just a setting on the command line of lame or is OGG better? Which achieves higher compression rates while maintaining call quality? Thanks, Waldo ___ --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] Getting ztdummy to load on startup for X100P
I'm not sure where in your startup process asterisk gets loaded. I load my asterisk from my rc.local file, so I can of course control when ztdummy would be loaded in relation to asterisk. Tzafrir Cohen wrote: On Fri, Nov 04, 2005 at 11:43:37AM -0900, Mojo with Horan Company, LLC wrote: Try putting a line at the very bottom of /etc/rc.d/rc.local like /sbin/modprobe ztdummy Which means ztdummy gts loaded only after asterisk is run? -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --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] Change Asterisk User
Use group permissions. Add the apache user to the asterisk group and give the group the appropriate read and/or write access. IMO this is the easiest way to get around the apache permissions thing, and probably the Right Way (tm) -Ryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of amaury BOSSE Sent: Monday, November 07, 2005 12:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Change Asterisk User Thanks for your answer, I am working on Debian Sarge but I have compiled Astersik 1.0.9 myself without .deb Packages. I need to access to voicemail and sound files from my web-interface (php and cgi/perl) but I can't change Apache user because of others applications. Asterisk creates files under Asterisk user and I have to access them from www-data user. Do you have other solution? I have tried using sudo but it doesn't seem to work. Regards, Amaury ___ --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] Stopping Asterisk from forwarding calls?
The first time I asked this to the list I didn't do a great job of it so I'm posting again with more details. Problem: when ringing multiple extensions, if one user has their phone forwarded directly to voicemail, it stops the whole group from ringing because the voicemail picks up immediately. Also, after hours incoming calls are to ring all extensions so anyone can pickup. But if one person in the office has their phone forwarded the same problem occurs. What we need is for asterisk, when ringing multiple extensions, to completely ignore the forward requests and just ring the remaining phones. Reading the source code I see there are two parameters for channels, allowredir_in allowredir_out. These offer me some hope that Asterisk has the ability but I couldn't figure out what these do or how to make use of them (I'm not a C programmer so maybe its just a red herring?). -- John Lange ___ --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] CentOS vs. Vanilla Kernel
Ryan Amos [EMAIL PROTECTED] wrote: The default CentOS kernel has worked fine for me. Just an FYI; CentOS uses the RedHat EL kernel source to build... It's pretty heavily patched so if you want to use the latest stable, download the SRPMs from RedHat/CentOS and patch in the kernel.org patches. It would be easier to patch in those patches already merged in the Rawhide (Fedora Development) kernels. Especially if you rebuild from SRPM proper. Just a clarification, I'm not advocating using the Rawhide kernels. If there is one place where Fedora Development/Core/Legacy differ heavily with Red Hat Enterprise Linux, it's at the kernel. But the patches from Rawhide kernels would probably be a far better fit for the RHEL kernels. But yeah, stick with the CentOS kernel unless you have problems. Agreed. Way too much is added/removed/changed. -- Bryan J. Smith| Sent from Yahoo Mail mailto:[EMAIL PROTECTED] | (please excuse any http://thebs413.blogspot.com/ | missing headers) ___ --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] asterisk 1.2b2 compiling problem
I just checked out asterisk 1.2b2 for zaptel, libpri, asterisk and asterisk-sounds. Zaptel and libpri compile fine with a 'make clean' and 'make install'. However even after a make clean, the asterisk 'make install' does not finish on my redhat 7.3 system. CVS-D2005.09.12.05.00.00-09/14/05-02:05:11 is currently running. Here are the last few lines before erroring out. chan_agent.c:1684: parse error before `char' chan_agent.c:1701: `agent_goodbye' undeclared (first use in this function) chan_agent.c:1701: (Each undeclared identifier is reported only once chan_agent.c:1701: for each function it appears in.) chan_agent.c:1708: `tmpoptions' undeclared (first use in this function) chan_agent.c:1714: `update_cdr' undeclared (first use in this function) chan_agent.c:1732: `context' undeclared (first use in this function) chan_agent.c:1737: `play_announcement' undeclared (first use in this function) chan_agent.c:1864: `filename' undeclared (first use in this function) make[1]: *** [chan_agent.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 Any ideas? Don Pobanz ___ --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
