[asterisk-users] priority problem
In dial plan, we can set the priority. However, I find that the
priority count is a global value. It will continue to increment no
matter in different context. Below is what I have tried.
In extension.conf
[testflow]
exten = _X.,1,NoOp(testflow,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow,2=${CALLERID(NUM)})
include = testflow1
include = testflow2
include = testflow3
[testflow1]
exten = _X.,1,NoOp(testflow1,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow1,2=${CALLERID(NUM)})
exten = _X.,3,NoOp(testflow1,3=${CALLERID(NUM)})
[testflow2]
exten = _X.,1,NoOp(testflow2,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow2,2=${CALLERID(NUM)})
exten = _X.,3,NoOp(testflow2,3=${CALLERID(NUM)})
exten = _X.,4,NoOp(testflow2,4=${CALLERID(NUM)})
[testflow3]
exten = _X.,1,NoOp(testflow3,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow3,2=${CALLERID(NUM)})
exten = _X.,3,NoOp(testflow3,3=${CALLERID(NUM)})
exten = _X.,4,NoOp(testflow3,4=${CALLERID(NUM)})
exten = _X.,5,NoOp(testflow3,5=${CALLERID(NUM)})
---
result in the log
-- Executing NoOp(SIP/871966760539-4467,
testflow|1=871966760539) in new stack
-- Executing NoOp(SIP/871966760539-4467,
testflow|2=871966760539) in new stack
-- Executing NoOp(SIP/871966760539-4467,
testflow1|3=871966760539) in new stack
-- Executing NoOp(SIP/871966760539-4467,
testflow2|4=871966760539) in new stack
-- Executing NoOp(SIP/871966760539-4467,
testflow3|5=871966760539) in new stack
As you can see in context of testflow1 and afterward, the priority
start is 3 instead of 1. Is the the correct sequence of priority
execution? Can I reset it to run the priority 1 in every context?
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[asterisk-users] CDRTools please help
Hi users I have been trying to install CDRTool but due to lack of documents i cannot do it properly so any body please respond me :-( here what i did is installed CDRTool in /var/www/ and create database cdrtool and there are two more files in that setup directory 1) create_tables.mysql 2) create_data.mysqlbut when i gave command #mysqladmin -uroot -px -hlocalhost ./setup_mysql.sh create cdrtoolit is creating cdrtool database in mysql server but with no tables and nothing just creating databse and then i tried to run the commands like #mysqladmin -uroot -px -hlocalhost ./create_tables.mysql but no use can any body please help meI know this is not a right place but for me there no other option (CDRTool mail archive has no messages inside) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7912 IP Phone - Convert SIP to SCCP
Hi all,I converted a Cisco 7912 IP phone from SCCP to SIP version and successfully registered with asterisk. And now I want to install back the SCCP firmware.I managed to convert both SCCP to SIP and SIP to SCCP in Cisco 7940 IP phone but I cannot do it in 7912?Is there anyone knows how to do it?Thanks. Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Channel Redirect
Hi i need to know how can i redirect a channel to a conference. The scenario is: im talking to a person, we decide to join a conference room and invite another person by calling his/her extension, the caller disconnect from conversation , the callee automatically is redirected to a conference room, the caller then dials another person's exten, as the person answere the phone hw,she automatically redirected to the conference room. So is it possible to redirect channels to conference without hangingup, if yes, HOW.-- RegardsRizwan HishamSoftware Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 vs TDMoE
Hello List We are having some load problems, and they are impacting IAX2 performance the most, with large amounts of jitter and lost packets. I'm currently thinking about using TDMoE for internal communication between our Asterisk servers. Does anybody know how load problems impact TDMoE? We are not having quality problems on our E1 connections, so I would guess that performance should be the same for TDMoE. Kind Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.10/387 - Release Date: 12-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Martin Joseph wrote: On Jul 12, 2006, at 7:16 AM, Rich Adamson wrote: Andrea Spadaccini wrote: Hello, I need to install Asterisk on a test machine that will soon become a production environment. Do you think that 1.2.9.1 is reliable? I read some posts that say it isn't as good as the previous versions. Should I install 1.2.8 or 1.2.7.1? I've had no issues at all with 1.2.9.1, however there have been several patches applied to the svn which I don't believe are part of the distro packages as yet. (My system is very basic with no need for queues, etc.) Same here, about three weeks of uptime on 1.2.9.1 with no issues, but a very simple setup. YMMV. On 2 setups, 1 very simple, 1 with queues and a hardware timing source, the latter crashed 3 times in the first week (stable since then), and has been dropping calls. The former has been rock solid, and fine. Looking at the bugtracker, there have been a lot of fixes since 1.2.9.1 was released, I wonder when 1.2.10 will be here (or 1.4.0b). ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra phones - disable call waiting
On 7/13/06, I T [EMAIL PROTECTED] wrote: Actually Aastra phones do support disabling Call Waiting on the their phones. Just add the following to your configuration file: call waiting tone: 0 It's in the release notes for the 1.4 release http://www.aastratelecom.com/downloads/RN-001024-00-08%20Aastra%20SIP%20Phone%201.4%20RN.pdf Actually, all that setting does is disable the tone, and not the call waiting - You still get a visual indication on the phone's screen of the waiting call. I contacted Aastra, who were very helpful. The upshot was that you need to use the call-limit setting in your SIP configuration of Asterisk. BTW, there's a separate Google Group for Aastra phones called Aastra 480i Users. Thanks for the pointer :). Shame it is focused on the one phone, and does not cover the other models :( Regards, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] where the bottleneck lies ? (was: Serverredundancy)
Douglas Garstang ha scritto: -Original Message- From: Simone Cittadini [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 12, 2006 10:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] where the bottleneck lies ? (was: Serverredundancy) unplug ha scritto: I feel interested about you can support 16,000 users of your system. As I have tested using sipp in a dual CPU Xeon with 2G Ram, the maximum number of current call is about 160. In some forums, most of ppl claim the maximum current call is about 100-200. What do you expect the number of current call to handle in 16,000 users? I'm curious about what was limiting the number of calls in your tests. For every system I have in production/testing I see the only bottleneck is system load, cpu and memory usage is well beyond limits when things starts to fall apart. The unexplicable (at least by me) thing is that system load seems to be only partially influenced by the number of calls, for example sometimes there are 100/150 calls and the load is around 0.70, sometimes it skyrockets to 2.00 / 2.50 (when it is 2 calls quality is crippled, I think because of too many dropped packets). I see this behaviour no matter how simple/complex the system is, from just a terminator with a couple of digium in it and a five-lines extension to the central server with fastagi doing mysql queries and taking hundreds of concurrent calls in both sip and iax. Can it be something related to asterisk itself ? I'm thinking about installing oprofile on the various servers, someone by chance already did it ? Another consideration is if the phones have performed reinvites, and removed Asterisk from the RTP stream. If you can live without call recording, and other features where Asterisk has to remain in the RTP path, then I imagine that this would significanlty reduce load on the Asterisk systems. Could some of your phones be reinviting? This may explain the variation in load. Doug. no, all the traffic has to pass from the machine (and all the codec is g711 so no differences in transcoding either) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] priority problem
On 7/13/06, unplug [EMAIL PROTECTED] wrote:
In dial plan, we can set the priority. However, I find that the
priority count is a global value. It will continue to increment no
matter in different context. Below is what I have tried.
This is exactly how it is designed, and is a very useful feature.
You need to understand the matching order properly to appreciate to
power of this type of evaluation, but I believe the solution to your
particular question is Goto, and not include.
[testflow]
exten = _X.,1,NoOp(testflow,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow,2=${CALLERID(NUM)})
exten = _X.,3,Goto(testflow1,${EXTEN},1)
...
Each context has its own exten = lines checked first, and then each
include = is checked in the order specified after that. Consider
using Macros too, they will often shorten a dialplan.
Cheers,
Steve
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Re: [asterisk-users] priority problem
Thanks. Could you tell me how useful it is? As I think if the
priority is reset in every context. I can design something like this.
My default context is myflow and your default context is yrflow. So
I can easy to delete a context in the context myflow if necessary.
[myflow]
include = testflow1
include = testflow2
include = testflow3
[yrflow]
include = testflow1
include = testflow3
[testflow1]
exten = _X.,1,NoOp(testflow1,1=${CALLERID(NUM)})
[testflow2]
exten = _X.,1,NoOp(testflow2,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow2,2=${CALLERID(NUM)})
[testflow3]
exten = _X.,1,NoOp(testflow3,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow3,2=${CALLERID(NUM)})
exten = _X.,3,NoOp(testflow3,3=${CALLERID(NUM)})
On 7/13/06, Steve Davies [EMAIL PROTECTED] wrote:
On 7/13/06, unplug [EMAIL PROTECTED] wrote:
In dial plan, we can set the priority. However, I find that the
priority count is a global value. It will continue to increment no
matter in different context. Below is what I have tried.
This is exactly how it is designed, and is a very useful feature.
You need to understand the matching order properly to appreciate to
power of this type of evaluation, but I believe the solution to your
particular question is Goto, and not include.
[testflow]
exten = _X.,1,NoOp(testflow,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow,2=${CALLERID(NUM)})
exten = _X.,3,Goto(testflow1,${EXTEN},1)
...
Each context has its own exten = lines checked first, and then each
include = is checked in the order specified after that. Consider
using Macros too, they will often shorten a dialplan.
Cheers,
Steve
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Re: RE: RE: RE: [Asterisk-Users] Very bad quality withAVMFritz!cardPCIandchan_capi
- Original Message - From: Dave Cotton [EMAIL PROTECTED] Date: Tuesday, July 11, 2006 9:55 am Subject: RE: RE: RE: [Asterisk-Users] Very bad quality withAVMFritz!cardPCIandchan_capi On Tue, 2006-07-11 at 16:38 +1000, James Harper wrote: So I've just had the time to swap and disable usb in my bios and it changed nothing the quality is still the same (which means horrible). How could I check where the problem comes from? I had absolutely awful sound on my AVM Fritz! with chan_capi until someone pointed me in the direction of the codec setting in capi.conf ;ulaw=yes;set this, if you live in u-law world instead of a- law I thought I am in the u-law world but evidently I am not. -- Dave Cotton [EMAIL PROTECTED] THanks for the hint I tried to switch to ulaw and the quality is even worse, unfortunately I have to find something else... Ben ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: RE: RE: [Asterisk-Users] Very bad quality withAVMFritz!cardPCIandchan_capi
- Original Message - From: James Harper [EMAIL PROTECTED] Date: Tuesday, July 11, 2006 8:38 am Subject: RE: RE: RE: [Asterisk-Users] Very bad quality withAVMFritz!cardPCIandchan_capi So I've just had the time to swap and disable usb in my bios and it changed nothing the quality is still the same (which means horrible). How could I check where the problem comes from? Ben Hmmm... that's a shame. Apologies if you have already specified this, but what are the versions of chan_capi and the Linuux kernel? Have you tried mISDN? (again, apologies if you've already mentioned this). James chan_capi version: chan_capi-cm-0.6.5 My asterisk is running on ubuntu: [EMAIL PROTECTED]:/usr/src# uname -a Linux sofia 2.6.12-9-386 #1 Mon Oct 10 13:14:36 BST 2005 i686 GNU/Linux I'd like to try misdn but I don't feel like recompiling my kernerl (asterisk is used in my company as main and only PBX). Maybe there is a way to use it without? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX2 vs TDMoE
We have used TDMoE before across 3 servers connected to each other over a dedicated network. It does work, but there were several issues that caused us to stop using it and we switched to IAX and/or crossover T1/E1s from server to server. At it's peak we were using two TDMoE E1s(over a single ethernet card) to have upto 60 concurrent channels going over TDMoE to a single Asterisk server from two other Asterisk servers with good results. One thing to consider about TDMoE is that Digium doesn't really support it or test for it anymore. There are still bugs in it that cause load issues if the connection is unexpectely terminated. Also, you need to use TDMoE on it's own dedicated network and on it's own dedicated Ethernet devices not connected to your normal network. As for your problem, TDMoE is a Zap channel so you will probably not see the issues that you are currently seeing with IAX2. It is something you should try to see if it will work for you, but keep in mind it might be killed off in a future version of Asterisk because of neglect. The one supporter of TDMoE's survival is RedFone which makes an add-on quad-T1/E1 box that communicates to an Asterisk server through TDMoE. They are about the only company that has an interest in keeping TDMoE going. http://www.red-fone.com/ MATT--- On 7/13/06, Jon Schøpzinsky [EMAIL PROTECTED] wrote: Hello List We are having some load problems, and they are impacting IAX2 performance the most, with large amounts of jitter and lost packets. I'm currently thinking about using TDMoE for internal communication between our Asterisk servers. Does anybody know how load problems impact TDMoE? We are not having quality problems on our E1 connections, so I would guess that performance should be the same for TDMoE. Kind Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.9.10/387 - Release Date: 12-07-2006 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] priority problem
On 7/13/06, unplug [EMAIL PROTECTED] wrote:
Thanks. Could you tell me how useful it is? As I think if the
priority is reset in every context. I can design something like this.
My default context is myflow and your default context is yrflow. So
I can easy to delete a context in the context myflow if necessary.
[myflow]
include = testflow1
include = testflow2
include = testflow3
[yrflow]
include = testflow1
include = testflow3
[testflow1]
exten = _X.,1,NoOp(testflow1,1=${CALLERID(NUM)})
[testflow2]
exten = _X.,1,NoOp(testflow2,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow2,2=${CALLERID(NUM)})
[testflow3]
exten = _X.,1,NoOp(testflow3,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow3,2=${CALLERID(NUM)})
exten = _X.,3,NoOp(testflow3,3=${CALLERID(NUM)})
To get what you want to do, I would rewrite it something like this
(There may be better ways, I just spent 5 seconds on this ;) ):
[myflow]
exten = _X.,1,Macro(testflow1)
exten = _X.,2,Macro(testflow2)
exten = _X.,3,Macro(testflow3)
[yrflow]
exten = _X.,1,Macro(testflow1)
exten = _X.,2,Macro(testflow3)
[macro-testflow1]
exten = s,1,NoOp(testflow1,1=${CALLERID(NUM)})
[macro-testflow2]
exten = s,1,NoOp(testflow2,1=${CALLERID(NUM)})
exten = s,2,NoOp(testflow2,2=${CALLERID(NUM)})
[macro-testflow3]
exten = s,1,NoOp(testflow3,1=${CALLERID(NUM)})
exten = s,2,NoOp(testflow3,2=${CALLERID(NUM)})
exten = s,3,NoOp(testflow3,3=${CALLERID(NUM)})
include = is a continue-until-matched type of contruct. Macro()
is a re-use-of-code construct.
Steve
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RE: RE: RE: RE: [Asterisk-Users] Very bad qualitywithAVMFritz!cardPCIandchan_capi
I'd like to try misdn but I don't feel like recompiling my kernerl (asterisk is used in my company as main and only PBX). Maybe there is a way to use it without? You shouldn't need to recompile the kernel, just follow the instructions on the voip wiki. James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ISDN BRI NetJet
I'm trying to use a Teles (netjet) ISDN BRI card with asterisk 1.2.9.1 Anyone was able to use this card with asterisk? I couldn't find much information about it. Any help? FYI, netjet is now supported under misdn cvs. James ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] priority problem
Thanks again. But I want to ask what is the usage of include if it is
a continue-until-matched type of contruct.
On 7/13/06, Steve Davies [EMAIL PROTECTED] wrote:
On 7/13/06, unplug [EMAIL PROTECTED] wrote:
Thanks. Could you tell me how useful it is? As I think if the
priority is reset in every context. I can design something like this.
My default context is myflow and your default context is yrflow. So
I can easy to delete a context in the context myflow if necessary.
[myflow]
include = testflow1
include = testflow2
include = testflow3
[yrflow]
include = testflow1
include = testflow3
[testflow1]
exten = _X.,1,NoOp(testflow1,1=${CALLERID(NUM)})
[testflow2]
exten = _X.,1,NoOp(testflow2,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow2,2=${CALLERID(NUM)})
[testflow3]
exten = _X.,1,NoOp(testflow3,1=${CALLERID(NUM)})
exten = _X.,2,NoOp(testflow3,2=${CALLERID(NUM)})
exten = _X.,3,NoOp(testflow3,3=${CALLERID(NUM)})
To get what you want to do, I would rewrite it something like this
(There may be better ways, I just spent 5 seconds on this ;) ):
[myflow]
exten = _X.,1,Macro(testflow1)
exten = _X.,2,Macro(testflow2)
exten = _X.,3,Macro(testflow3)
[yrflow]
exten = _X.,1,Macro(testflow1)
exten = _X.,2,Macro(testflow3)
[macro-testflow1]
exten = s,1,NoOp(testflow1,1=${CALLERID(NUM)})
[macro-testflow2]
exten = s,1,NoOp(testflow2,1=${CALLERID(NUM)})
exten = s,2,NoOp(testflow2,2=${CALLERID(NUM)})
[macro-testflow3]
exten = s,1,NoOp(testflow3,1=${CALLERID(NUM)})
exten = s,2,NoOp(testflow3,2=${CALLERID(NUM)})
exten = s,3,NoOp(testflow3,3=${CALLERID(NUM)})
include = is a continue-until-matched type of contruct. Macro()
is a re-use-of-code construct.
Steve
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[asterisk-users] RE: Possible polycom_acd_functions BUG
Has anybody had this issue before? -Original Message- From: Dean @ INKnBITs [mailto:[EMAIL PROTECTED] Sent: 12 July 2006 11:35 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Possible polycom_acd_functions BUG I have noticed a couple of issues, unless I'm doing something wrong? I pulled with svn the svn.digium.com/svn/asterisk/team/bweschke/polycom_acd_functions/ which got release 37416 This complies fine, in particular the meetme app. If I setup a sip device in the sip.conf with a username and password, I get: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.1.3.103' - Username/auth name mismatch with out the username and password, I get a registration. (This doesn't help as I need the password field for the ACD function to work) I have gone backwards through the releases, 30432 complies fine, except it will not compile the meetme app, but the username and passwords works fine. Does anybody know a release in the middle that works with both features? Thanks, Dean Bath. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sending out fax using asterisk
Hi all, I am using CentOS 4.3, asterisk 1.2.9.1 with rx_fax and tx_fax I am having problem sending out fax from fax using an ATA connected to the asterisk. Below is the message pop up in asterisk -vvvgc: - -- Executing Set(SIP/60005-1c18, FAXFILE=/var/spool/asterisk/fax/mydocument.tif) in new stack -- Executing Set(SIP/60005-1c18, LOCALHEADERINFO=Company name and department) in new stack -- Executing Set(SIP/60005-1c18, LOCALSTATIONID=Company name) in new stack -- Executing TxFAX(SIP/60005-1c18, /var/spool/asterisk/fax/mydocument.tif|caller) in new stack Segmentation fault (core dumped) Regards, rootlinux __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR DTMF
Dear I want to make a billing recharge through receiving digits from IVR through dtmf and store it on a text file , How can todo that ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using DUNDi with TrixBox mini HOWTO
Hi all,I wanted to tie a few trixboxes together but to there wasn't much documetation out there. After a few problems I eventually figured out to do it and off course I want to share it with all of you. Here's a mini howto where I'll explain how to use DUNDi to tie 3 Trixboxes together and make local extensions available to all the Trixboxes.You can find the document here: http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdf Have fun!Regards,Tijmen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] H323 implementation
I have a requirement to set up an Asterisk server that will handle H323. In the end this is used for video conferencing but it will be transitioning other H323 devices to SIP at some point. My question is this: Does anyone know of or have good documentation that explains how this configuration might work or should work. I understand that the implementation of H323 in Asterisk is for a gateway only. I have put GnuGK on the same box to handle the gatekeeper role and they appear to work individually but I have not tested interoperability yet (I will be later this morning). I am supposing that I just point the Asterisk gateway to the gatekeeper (which happens to be on the same box) and it should be able to handle the number mapping. The other problem I have is MCU. I did not have much luck with openMCU yet, so I am in need of that as well. I suppose this turned into a multipoint question, sorry. Has anyone done anything like this out there that was a completely capable unit that will handle (PBX functionality, PSTN connection, and MCU functionality)? Thanks Curt ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using DUNDi with TrixBox mini HOWTO
This is great! ThanxOn 7/13/06, tijmen van den brink [EMAIL PROTECTED] wrote: Hi all,I wanted to tie a few trixboxes together but to there wasn't much documetation out there. After a few problems I eventually figured out to do it and off course I want to share it with all of you. Here's a mini howto where I'll explain how to use DUNDi to tie 3 Trixboxes together and make local extensions available to all the Trixboxes.You can find the document here: http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdf Have fun!Regards,Tijmen ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk dual servers through iax: Accepting UNAUTHENTICATED call
Hi, I managed to connect two asterisk servers and now I can call from one to the other but when I make a call (no matter from which one) the called asterisk console always shows: *Accepting UNAUTHENTICATED call* I tried with registered users but it is the same. I do not consider it a problem ...but maybe I could be wrong. What does that message mean? TIA Giorgio Incantalupo ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CDRTools please help
Von: ravi reddy [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 13. Juli 2006 10:03 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] CDRTools please help ... but when i gave command #mysqladmin -uroot -px -hlocalhost ./setup_mysql.sh create cdrtool it is creating cdrtool database in mysql server but with no tables and nothing just creating databse and then i tried to run the commands like #mysqladmin -uroot -px -hlocalhost ./create_tables.mysql I don't know much about CDRTool, but I think you've a typo in your mysqladmin statement. Perhaps try this: mysqladmin -u root -px -h localhost ./setup_mysql.sh create cdrtool With your command you try to export something out of your database: mysqladmin -uroot -px -hlocalhost ./setup_mysql.sh create cdrtool Also, have a look at your setup_mysql.sh script, it is broken after the above command. This happens normaly when it's too late at night ;-) Hope it helps... Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Transfer Application
Hi!Where can I find more informations about the Transfer() application in a All-SIP environment? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP To: header
Is there a way to access the actual SIP To: header? I know the URI is easily accessible, and is handy for a multitude of things, but in a scenario in which a call has been forwarded from one URI to another, it's handy to know whence the forward was initiated (which would only be in the To: header presumably). Ideally, I need this via AGI, but if it can be accessed anywhere at all, I can code something up. N. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] SIP To: header
Isnt SIP_HEADER(TO) enough?
e.g.
exten = ,1,Answer
exten = ,2,Set(TO_HEADER=${SIP_HEADER(TO)})
exten = ,3,NoOp(TO_HEADER)
exten = ,4,Hangup
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of sip
Sent: 13 July 2006 12:12
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP To: header
Is there a way to access the actual SIP To: header? I know
the URI is easily
accessible, and is handy for a multitude of things, but in a
scenario in which
a call has been forwarded from one URI to another, it's handy
to know whence
the forward was initiated (which would only be in the To:
header presumably).
Ideally, I need this via AGI, but if it can be accessed
anywhere at all, I can
code something up.
N.
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RE: [asterisk-users] SIP To: header
Oops, line 3 of the example should have read:
exten = ,3,NoOp(${TO_HEADER})
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Steve
Langstaff
Sent: 13 July 2006 12:21
To: sip; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] SIP To: header
Isnt SIP_HEADER(TO) enough?
e.g.
exten = ,1,Answer
exten = ,2,Set(TO_HEADER=${SIP_HEADER(TO)})
exten = ,3,NoOp(TO_HEADER)
exten = ,4,Hangup
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+sip_header
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of sip
Sent: 13 July 2006 12:12
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] SIP To: header
Is there a way to access the actual SIP To: header? I know
the URI is easily
accessible, and is handy for a multitude of things, but in a
scenario in which
a call has been forwarded from one URI to another, it's handy
to know whence
the forward was initiated (which would only be in the To:
header presumably).
Ideally, I need this via AGI, but if it can be accessed
anywhere at all, I can
code something up.
N.
___
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Re: [asterisk-users] SIP To: header
You can pull anything from the header with SIP_HEADERI'll often just pass them into a Perl AGI as $ARGV[0] $ARGV[1] with this line:exten = myapp,2,AGI(myapp.agi|${SIP_HEADER(From)}|${SIP_HEADER(To)})
Note also you can get *anything* in the SIP header SIP_HEADER(Mumblefratz) etc.
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RE: [asterisk-users] Polycom compatible phone for Asterisk
I'm going to have to echo everyone else, the 301s are ok but the lack of full duplex speakerphone sucks, but they have a 430 now. I have a ton of 501s and 601s at clients and they are great. I do have some 300s (like the 301 without as much memory I guess) that did crash during a power outage and lost their configs but the x01 models have been fantastic and rock solid even during the same power outages. Bill -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of (AstATN) Sent: Wednesday, July 12, 2006 10:21 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom compatible phone for Asterisk Hi all, Can some one provide me the infor about polycom phones model that compatible and stable to work with Asterisk? I intend to purchase IP 300, and IP 501 models. Tq ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Play sound to called party ...
Hi List, When a call comes in on a specific number (via ISDN) we would like the callee ( SIP Phone ) to get an audio message before the caller is put through. Is this possible? To clarify: Caller calls a specific number Callee answers the phone as normal Callee hears a message Caller is seamlessly put through to callee It would also help if the caller continues to hear ringing until the caller is put through to the callee. Hope this makes sense. Thanks in advance Phil.___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Play sound to called party ...
I used Dial() with A(filename.sgm) option - somethnig like
Dial(SIP/trunk/${number}|60|A(callrecorded.gsm))
when calee answered then he/she was 'announced' by message in file then
the conversation starts.
When file was playing i hear nothnig that calee says.
http://www.voip-info.org/wiki-Asterisk+cmd+dial
Użytkownik [EMAIL PROTECTED] napisał:
Hi List,
When a call comes in on a specific
number
(via ISDN) we would like the callee ( SIP Phone ) to get an audio
message
before the caller is put through. Is this possible?
To clarify:
Caller calls a specific number
Callee answers the phone as normal
Callee hears a message
Caller is seamlessly put through to
callee
It would also help if the caller
continues
to hear ringing until the caller is put through to the callee.
Hope this makes sense.
Thanks in advance
Phil.
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Re: [asterisk-users] Using DUNDi with TrixBox mini HOWTO
The same key used across all boxes? *Grumbles about security*... In all seriousness though, that's very well written and looks quite nice. You have a knack for that type of document it would seem. Great job!Alex On 7/13/06, tijmen van den brink [EMAIL PROTECTED] wrote: Hi all,I wanted to tie a few trixboxes together but to there wasn't much documetation out there. After a few problems I eventually figured out to do it and off course I want to share it with all of you. Here's a mini howto where I'll explain how to use DUNDi to tie 3 Trixboxes together and make local extensions available to all the Trixboxes.You can find the document here: http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdf Have fun!Regards,Tijmen ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXS adapters and Polycom phones
Mike wrote: I`m looking for a SIP-PSTN adapter, basically to switch a customer from a cheap PBX to mine, but resuing their own Norstar PSTN phones. They have 10 phones. From a price point of view, it seems that 10 individual GrandStream SIP adapters is the best way to go, but it seems so inelegant to me. What is recommended ? Not sure about the others, but I've had decent experiences with the Linksys PAP2 series, and they aren't that expensive. Second question: I have a GrandStream GXP-2000, that despite what everybody says I love. I am still looking for a replacement, if only because it doesn`t look as good and it does have a few quirks. I was looking at Polycoms, but some questions are unanswered by looking at their datasheet. - Does the Polycom 501 have an integrated router (like the GXP-2000, latest firmware, does) Not entirely sure what you're asking here. If you're wondering if it has a two NIC interfaces (a pass-through for the PC) then yes. - Can you have more than one SIP/account on the phone, each ringing in a way that lets the user know which account is ringing? (GXP2000 does it by making it possible to have each line linked to a separate SIP account) The 501 is capable of having 3 different line appearances, each of which can have a primary and secondary server configured for them. -- Jamin W. Collins ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connect to 'agi://blablabla' failed: Operation now in progress
I get a lot of this warnings in my logs. Connect to 'agi://blablabla' failed: Operation now in progress What exactly 'operation now in progress means' ? is asterisk still trying so the call isn't lost ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using DUNDi with TrixBox mini HOWTO
Yeah I'm sorry about that security issue, perhaps someone can add a post to make this a more secure solution?TijmenOn 7/13/06, Alex Robar [EMAIL PROTECTED] wrote:The same key used across all boxes? *Grumbles about security*... In all seriousness though, that's very well written and looks quite nice. You have a knack for that type of document it would seem. Great job! Alex On 7/13/06, tijmen van den brink [EMAIL PROTECTED] wrote: Hi all,I wanted to tie a few trixboxes together but to there wasn't much documetation out there. After a few problems I eventually figured out to do it and off course I want to share it with all of you. Here's a mini howto where I'll explain how to use DUNDi to tie 3 Trixboxes together and make local extensions available to all the Trixboxes.You can find the document here: http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdf Have fun!Regards,Tijmen ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- T. van den Brink BEWilhelminaweg 463441 XC WoerdenTel: 0878706429GSM: 0651623080 == NIEUW!!! MSN: [EMAIL PROTECTED]Skype: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] an operational scenario
Bruce Ferrell wrote: the problem I'm seeing is one way audio between extensions. I've splpit up the numbering plan internal/external. All are in the same range. I'll try splitting them and see what happens. By one way audio between extensions are you talking about calls between extensions where one side is on the internal network and one side is on the external network? If so you might look at disabling reinvite and/or making sure the external party's RTP connection is able to make it through any firewall you might have in place. -- Jamin W. Collins ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Using DUNDi with TrixBox mini HOWTO
Great document, Ive added the url to my intro to asterisk web page. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of tijmen van den brink Sent: Thursday, 13 July 2006 9:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Using DUNDi with TrixBox mini HOWTO Yeah I'm sorry about that security issue, perhaps someone can add a post to make this a more secure solution? Tijmen On 7/13/06, Alex Robar [EMAIL PROTECTED] wrote: The same key used across all boxes? *Grumbles about security*... In all seriousness though, that's very well written and looks quite nice. You have a knack for that type of document it would seem. Great job! Alex On 7/13/06, tijmen van den brink [EMAIL PROTECTED] wrote: Hi all, I wanted to tie a few trixboxes together but to there wasn't much documetation out there. After a few problems I eventually figured out to do it and off course I want to share it with all of you. Here's a mini howto where I'll explain how to use DUNDi to tie 3 Trixboxes together and make local extensions available to all the Trixboxes. You can find the document here: http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdf Have fun! Regards, Tijmen ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- T. van den Brink BE Wilhelminaweg 46 3441 XC Woerden Tel: 0878706429 GSM: 0651623080 == NIEUW!!! MSN: [EMAIL PROTECTED] Skype: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Database
Hi list! I'm planning do use LookupCIDName application. TO use it I need to input CID data to internal asterisk DB. Question is, how much data can I store to Asterisk DB? Is there any maximum? Does outing to much (how much is too much?) data in DB effects work of * in any way? Please share your experience. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using DUNDi with TrixBox mini HOWTO
I made a little DUNDi on [EMAIL PROTECTED] tutorial that includes how to make keys on each system. It's not nearly as nice looking as yours, but you might merge the section into your doc if you think it fits: http://blog.thegoldfish.net/dundi-tutorial-for-asteriskhome/AlexOn 7/13/06, tijmen van den brink [EMAIL PROTECTED] wrote:Yeah I'm sorry about that security issue, perhaps someone can add a post to make this a more secure solution? TijmenOn 7/13/06, Alex Robar [EMAIL PROTECTED] wrote:The same key used across all boxes? *Grumbles about security*... In all seriousness though, that's very well written and looks quite nice. You have a knack for that type of document it would seem. Great job! Alex On 7/13/06, tijmen van den brink [EMAIL PROTECTED] wrote: Hi all,I wanted to tie a few trixboxes together but to there wasn't much documetation out there. After a few problems I eventually figured out to do it and off course I want to share it with all of you. Here's a mini howto where I'll explain how to use DUNDi to tie 3 Trixboxes together and make local extensions available to all the Trixboxes.You can find the document here: http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdf Have fun!Regards,Tijmen ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- T. van den Brink BEWilhelminaweg 463441 XC WoerdenTel: 0878706429 GSM: 0651623080 == NIEUW!!! MSN: [EMAIL PROTECTED]Skype: [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Database
Why not use mysql?Do something like this: exten = s,1,MYSQL(SELECT * FROM whatever)bpOn 7/13/06, Tomislav Parčina [EMAIL PROTECTED] wrote: Hi list!I'm planning do use LookupCIDName application. TO use it I need to input CID data to internal asterisk DB. Question is, how much data can I store to Asterisk DB? Is there any maximum? Does outing to much (how much is too much?) data in DB effects work of * in any way? Please share your experience.--Tomislav ParčinaLama Computers SplitStinice 12, 21000 SplitTel.: +385(21)495148Mob.: +385(91)1212148SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hrhttp://www.lama.hr___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: Channel Redirect
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi i need to know how can i redirect a channel to a conference. The scenario is: im talking to a person, we decide to join a conference room and invite another person by calling his/her extension, the caller disconnect from conversation , the callee automatically is redirected to a conference room, the caller then dials another person's exten, as the person answere the phone hw,she automatically redirected to the conference room. So is it possible to redirect channels to conference without hangingup, if yes, HOW. Hi Rizwan, Maybe I didn't get it right, but why don't you transfer that person to meetme extension? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [EMAIL PROTECTED]
since when is this a google list ? - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Commercial and Business-Oriented Asterisk Discussion asterisk-biz@lists.digium.com Sent: Wednesday, July 12, 2006 6:32 PM Subject: [asterisk-users] [EMAIL PROTECTED] To keep the Asterisk mailing list free of Voip provider complaints: VoIP is a growing business area. We all find days of problems. Some companies can handle problems. Some VoIP providers create problems. In this group we can discuss and learn how to handle conflicts. What to do and what not to do in this group: 1. Report cases and your impression. 2. Try to word it polite, even it is sometimes hard to do so. 3. Do not use any words you would not say also to your own 12 year old child. 4. Accept advices. Caution and remember, this is a Google list. All messages are UNREMOVABLE in the Search engine, Good luck! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1000s of extensions in one context?
I wouldnt think why not and its a lot easier to program as oposed to going thru a conf file over and over, reloading etc. - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Wednesday, July 12, 2006 8:32 PM Subject: Re: [asterisk-users] 1000s of extensions in one context? On Wed, Jul 12, 2006 at 08:07:30PM -0400, Dovid Bender wrote: i would go with realtime for that Does the real-time configuration engine handle 1000-s of extensions in the same context more efficiently? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: asterisk dual servers through iax: Accepting UNAUTHENTICATED call
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I managed to connect two asterisk servers and now I can call from one to the other but when I make a call (no matter from which one) the called asterisk console always shows: *Accepting UNAUTHENTICATED call* I tried with registered users but it is the same. I do not consider it a problem ...but maybe I could be wrong. What does that message mean? Hi Giorgio, Asterisk will always try to make unauthenticated call, and if you allow it it will established it as unauthenticated. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using DUNDi with TrixBox mini HOWTO
Hi Alex,I forgot to mention your document in mine but I can say it was very usefull for me to! So when I'm making some changes on the document I will add the keys part to my document. Thanks very much!Tijmen On 7/13/06, Alex Robar [EMAIL PROTECTED] wrote: I made a little DUNDi on [EMAIL PROTECTED] tutorial that includes how to make keys on each system. It's not nearly as nice looking as yours, but you might merge the section into your doc if you think it fits: http://blog.thegoldfish.net/dundi-tutorial-for-asteriskhome/AlexOn 7/13/06, tijmen van den brink [EMAIL PROTECTED] wrote:Yeah I'm sorry about that security issue, perhaps someone can add a post to make this a more secure solution? TijmenOn 7/13/06, Alex Robar [EMAIL PROTECTED] wrote:The same key used across all boxes? *Grumbles about security*... In all seriousness though, that's very well written and looks quite nice. You have a knack for that type of document it would seem. Great job! Alex On 7/13/06, tijmen van den brink [EMAIL PROTECTED] wrote: Hi all,I wanted to tie a few trixboxes together but to there wasn't much documetation out there. After a few problems I eventually figured out to do it and off course I want to share it with all of you. Here's a mini howto where I'll explain how to use DUNDi to tie 3 Trixboxes together and make local extensions available to all the Trixboxes.You can find the document here: http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdf Have fun!Regards,Tijmen ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can I register multiple TERMINATORS to a single account on IAX?
Ok, Here is my scenario I have one (1) server that does my termination. (SERVER A) I have three (3) call centers that I want to terminate outbound calls to the terminator (SERVER A). Assuming I don't care about knowing what call came from which server, can I register all 3 call centers to SERVER-A under the same account? Or do I need to have two different accounts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [EMAIL PROTECTED]
It's not a Google list but Google are indexing the posts so.making a fool of yourself here will live on in perpetuity. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, 13 July 2006 9:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]unhappy-about-VoIP- [EMAIL PROTECTED] since when is this a google list ? - Original Message - From: Ronald Wiplinger [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com; Commercial and Business-Oriented Asterisk Discussion asterisk-biz@lists.digium.com Sent: Wednesday, July 12, 2006 6:32 PM Subject: [asterisk-users] [EMAIL PROTECTED] To keep the Asterisk mailing list free of Voip provider complaints: VoIP is a growing business area. We all find days of problems. Some companies can handle problems. Some VoIP providers create problems. In this group we can discuss and learn how to handle conflicts. What to do and what not to do in this group: 1. Report cases and your impression. 2. Try to word it polite, even it is sometimes hard to do so. 3. Do not use any words you would not say also to your own 12 year old child. 4. Accept advices. Caution and remember, this is a Google list. All messages are UNREMOVABLE in the Search engine, Good luck! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: $3,000 server
Yeah, but in a soap opera both sides are equally annoying... Nick On Wed, Jul 12, 2006 at 10:55:55PM -0400, William Piper wrote: Man, with a thread like this... who needs a soap opera? ;-) bp ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 1000s of extensions in one context?
On Thu, Jul 13, 2006 at 09:43:45AM -0400, Dovid Bender wrote: I wouldnt think why not and its a lot easier to program as oposed to going thru a conf file over and over, reloading etc. Hint: a config file can be generated. Part of it can. Anyway, The OP asked about performance. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: Channel Redirect
Ok, i'll try to make it cclear. i think you r saying that i use this: exten=1,1,Dial(SIP/2000,,Tt) exten=2,1,MeetMe(1234||) by pressing # and then the extension 2 the caller will be transferred to conferenece room 1234 and the called person will be disconnected. This is partly what i want. i dont want the called person to get disconnected, rather he should also follow and join the conference. and if they want to invite a third person or maybe fourth person, they should be able to do that by exiting the conference but remaining in the dilaplan where they should do something to invite some more people to conference. during all this process, no one should get disconnected or hangup. I hope im clear now..On 7/13/06, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi i need to know how can i redirect a channel to a conference. The scenario is: im talking to a person, we decide to join a conference room and invite another person by calling his/her extension, the caller disconnect from conversation , the callee automatically is redirected to a conference room, the caller then dials another person's exten, as the person answere the phone hw,she automatically redirected to the conference room. So is it possible to redirect channels to conference without hangingup, if yes, HOW.Hi Rizwan,Maybe I didn't get it right, but why don't you transfer that person to meetme extension? --Tomislav ParčinaLama Computers SplitStinice 12, 21000 SplitTel.: +385(21)495148Mob.: +385(91)1212148SIP: [EMAIL PROTECTED]e-mail: tparcina#lama.hr http://www.lama.hr___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- RegardsRizwan HishamSoftware Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Connect to 'agi://blablabla' failed: Operation now in progress
AFAIK operation now in progress is a common status when you open a socket connection. When you use blocking sockets usually you dont see this because the connect call does not return until the connection is done. But when using non-blocking sockets, the connect call returns immediatly and if you try to connect again, you will get the operation now in progress message. I have seen this in my PHP Manager Proxy, but not sure what implications may have in FastAGI. May be it only tells that the connection stablishment takes a little longer, network congestion may be? Regards On 7/13/06, Simone Cittadini [EMAIL PROTECTED] wrote: I get a lot of this warnings in my logs. Connect to 'agi://blablabla' failed: Operation now in progress What exactly 'operation now in progress means' ? is asterisk still trying so the call isn't lost ? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Transfer Application
Goto voip-info.org and search it.On 7/13/06, Benjamin Stocker [EMAIL PROTECTED] wrote:Hi!Where can I find more informations about the Transfer() application in a All-SIP environment? ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- RegardsRizwan HishamSoftware Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [Solved]Re: Exclude a certain route from using a trunk
This is how i did it.
Solution 1;
Local Outbound Calls Trunk
0+N
011254|N
International Outbound Trunk
011.
011254|NZXXX
1NXXNXX ;incase someone dials US/CA 1 (10 digit #)
011|NXXNXX;incase someone dials US/CA 011 1 (10 digit #)
01|NXXNXX;incase someone dials US/CA 011 (10 digit #)Solution2:Someone else suggested;Although nobody uses or dials 011 for local
calls but if you have to enter it to dial locally you can have the
first non-local digits excluded before getting passed unto dial. The
${extension:[EMAIL PROTECTED] used below is used to exclude the
digits not needed to dial so you can adjust the number'6' to whatever
digit that matches the amount of numbers to be excluded before dialing
through your local trunk entered after the '@' sign.
exten=011254NXX,1,Dial,${extension:[EMAIL PROTECTED]KOn 7/12/06, Levis Kimotho
[EMAIL PROTECTED] wrote:Hi,In my Outbound routes i have created International Local Calls. I have 2 trunks for both ITL and LC. All calls are dialed using 011.but all 011254, 01125473, 01125472 should use the local trunk. NB Local Route is 1st priority in my list or routes. Everyone has to dial 011(number) to make a call whether Local or international but all 011254* number should use my local trunk. How do i achieve this? This is what i have so far;
Outbound Route - (International Calls) **Ive put the same in the trunksDial Pattern 011.Outbound Route - (Local Calls) **Ive put the same in the trunksDial Pattern 011254|072XXX011254|073XXX
011254|K
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[asterisk-users] Asterisk Console Colorization Question
Hi All, Im not sure if this is more a linux question than an Asterisk question. Either way, any advice would be appreciated. The issue I have is on the server the console for asterisk is showing up on like console 8 or 9. (What ever the default is). When viewing that console however, any word that is colorized is all jumbled letters. If I reconnect to asterisk on a console that I have logged in from, the colorized words are fine. One way Ive solved this is to modify the asterisk startup script in my /etc/init.d to load the asterisk daemon with the n parameter. However, If I do this then all the other consoles I open when actually logged in are also in a Non-Colorized format. Im assuming that once the asterisk daemon starts with n there is no way to re-enable color without stopping and restarting the asterisk process without the n. Is that correct? Is there something I can do to make the colorized version work ok on Console 8 or 9(which ever one its at)? Thanks, --Shawn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to do load balancing (1:1) with IAX and two different ISPs
Ken Dresdell a écrit : Hello folks, Does anyone have an idea how I could setup a load balancing (1:1) solution with IAX and two different Internet service providers. The idea is to increase the bandwidth between offices with cheap Internet access (DSL/Cable). Do you want to load balance all your LAN / WAN traffic or just VoIP traffic? Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail CallerID
I've got a question about voicemail and callerid and I can't quite
figure it out. I've got extensions 100, 101 and 102. For outbound
callerID (calls from the phones to the PSTN), I want the callerid to say
100 on all phones, so under sip.conf, I added:
callerid=Bill 100
The problem is that when they go to check voicemail, it looks at their
callerID and it drops them into mailbox 100 (calls to them still go into
their own specific mailbox, it is just when they hit their messages
button). Any idea how to get around that? Or do I just have to send
them to voicemail without having it automatically enter their extension?
This is what my voicemail does:
exten = 3299,1,VoicemailMain(${CALLERIDNUM})
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Re: [asterisk-users] IVR DTMF
hiring some one to do it :) sorry, i couldnt avoid to tell it, but your question is so generic that the response will be generic, unless some kind sould takes several minutes of their time to explain it to you. First i would recommend you this document: http://www.catb.org/~esr/faqs/smart-questions.html Now, about your question. You can create such an application in several ways, one of them is using AGI () and GET DATA command Regards On 7/13/06, Khaled Chehab [EMAIL PROTECTED] wrote: Dear I want to make a billing recharge through receiving digits from IVR through dtmf and store it on a text file , How can todo that ? Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cdr functions change between * 1.2.4 and 1.2.9.1 (agi)
Hi,I was using cdr-csv through phpagi with asterisk 1.2.4 and it works great.I had:$variable = CDR(accountcode)= . $_SESSION['username']; // Add CDR account codeand it works great. It create a new csv file for each user + the Master.csv with everything in it.I juste upgraded to asterisk 1.2.9.1, everything is ok exept that I only get a Master.csv without the username and no csv file per user.Does the cdr function changed? Is the CDR(accountcode)=username depricated? I can't find shy it doesn't work. Where should I hunt for informations/logs/hints ?Thank you for your help.benq ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to do load balancing (1:1) with IAX and twodifferent ISPs
Just our VoIP traffic Ken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: 13 juillet 2006 10:48 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to do load balancing (1:1) with IAX and twodifferent ISPs Ken Dresdell a écrit : Hello folks, Does anyone have an idea how I could setup a load balancing (1:1) solution with IAX and two different Internet service providers. The idea is to increase the bandwidth between offices with cheap Internet access (DSL/Cable). Do you want to load balance all your LAN / WAN traffic or just VoIP traffic? Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: asterisk dual servers through iax: Accepting UNAUTHENTICATED call
Thanks! Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Hi, I managed to connect two asterisk servers and now I can call from one to the other but when I make a call (no matter from which one) the called asterisk console always shows: *Accepting UNAUTHENTICATED call* I tried with registered users but it is the same. I do not consider it a problem ...but maybe I could be wrong. What does that message mean? Hi Giorgio, Asterisk will always try to make unauthenticated call, and if you allow it it will established it as unauthenticated. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Voicemail CallerID
Would this work?
exten = 3299,1,VoicemailMain(${EXTEN})
This way it would check the voicemail of the extension doing the dialing?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peder @
NetworkOblivion
Sent: Thursday, July 13, 2006 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Voicemail CallerID
I've got a question about voicemail and callerid and I can't quite
figure it out. I've got extensions 100, 101 and 102. For outbound
callerID (calls from the phones to the PSTN), I want the callerid to say
100 on all phones, so under sip.conf, I added:
callerid=Bill 100
The problem is that when they go to check voicemail, it looks at their
callerID and it drops them into mailbox 100 (calls to them still go into
their own specific mailbox, it is just when they hit their messages
button). Any idea how to get around that? Or do I just have to send
them to voicemail without having it automatically enter their extension?
This is what my voicemail does:
exten = 3299,1,VoicemailMain(${CALLERIDNUM})
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Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Thomas Kenyon wrote: On 2 setups, 1 very simple, 1 with queues and a hardware timing source, the latter crashed 3 times in the first week (stable since then), and has been dropping calls. The former has been rock solid, and fine. I guess I am a 'me to'. I am running 1.2.9.1 and did not have any problems until setting up queues. Within a day of doing queue logins/logouts our T1 DID trunks (not PRI) stopped accepting calls from the local telco. Internal calls though channel banks continued to function properly. A restart would clear the situation. This happened on three separate occasions. After that, I got smarter and stopped doing anything with queues. ;) We will implement our queues at a later date! Don Pobanz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Don Pobanz wrote: Thomas Kenyon wrote: On 2 setups, 1 very simple, 1 with queues and a hardware timing source, the latter crashed 3 times in the first week (stable since then), and has been dropping calls. The former has been rock solid, and fine. I guess I am a 'me to'. I am running 1.2.9.1 and did not have any problems until setting up queues. Within a day of doing queue logins/logouts our T1 DID trunks (not PRI) stopped accepting calls from the local telco. Internal calls though channel banks continued to function properly. A restart would clear the situation. This happened on three separate occasions. After that, I got smarter and stopped doing anything with queues. ;) We will implement our queues at a later date! For others to better understand the issues, did you install asterisk as a distro or download v1.2 svn code and compile? If you installed source via svn, did you try make update to pick up any patches? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New York city Asterisk consultants
If there are any freelance asterisk consultants (or small companies) who live in the NY on this list looking for additional work please email me with some background reference installations and contact details. Unfortunately the client is going to be very very picky about who they brief on this large scale project and who is eventually accepted as its a non-standard asterisk installation with high net commercial value. You must have the follow capabilities Reside in NY city or be able to commute. (youll be working in person with a team). Have a rudimentary understanding of web html and java. Have demonstrable asterisk installations with multi server load balancing. Have demonstrable experience with multi-tenant and associate billing applications. Regards, Dean Collins Cognation ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Rich Adamson wrote: For others to better understand the issues, did you install asterisk as a distro or download v1.2 svn code and compile? I downloaded the 1.2.9.1 release from the www.asterisk.org website and compiled it. If you installed source via svn, did you try make update to pick up any patches? I have not used svn. This is the 1.2.9.1 release with no patches applied. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New York city Asterisk consultants
Dean,You'll probably get better results posting this to the business group. This group is for non-commercial discussion only.AlexOn 7/13/06, Dean Collins [EMAIL PROTECTED] wrote: If there are any freelance asterisk consultants (or small companies) who live in the NY on this list looking for additional work please email me with some background reference installations and contact details. Unfortunately the client is going to be very very picky about who they brief on this large scale project and who is eventually accepted as it's a non-standard asterisk installation with high net commercial value. You must have the follow capabilities Reside in NY city or be able to commute. (you'll be working in person with a team). Have a rudimentary understanding of web html and java. Have demonstrable asterisk installations with multi server load balancing. Have demonstrable experience with multi-tenant and associate billing applications. Regards, Dean Collins Cognation ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I register multiple TERMINATORS to a single account on IAX?
Assuming I understand what you are trying to do, just put accountcode=whatever in your iax.conf for each user.bpOn 7/13/06, Matt [EMAIL PROTECTED] wrote:Ok,Here is my scenario I have one (1) server that does my termination. (SERVER A)I have three (3) call centers that I want to terminate outbound callsto the terminator (SERVER A).Assuming I don't care about knowing what call came from which server, can I register all 3 call centers to SERVER-A under the same account?Or do I need to have two different accounts?___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I register multiple TERMINATORS to a single account on IAX?
Yes that solves the issue completed.. and that's what I did... but that got me to wondering if you could just register multiple servers to a single user-name (probably not the best way to do things hehe but wondered if it would work since you aren't pushing anything back to them). At any rate, I did just go with the accountcode=blah for each account. On 7/13/06, William Piper [EMAIL PROTECTED] wrote: Assuming I understand what you are trying to do, just put accountcode=whatever in your iax.conf for each user. bp On 7/13/06, Matt [EMAIL PROTECTED] wrote: Ok, Here is my scenario I have one (1) server that does my termination. (SERVER A) I have three (3) call centers that I want to terminate outbound calls to the terminator (SERVER A). Assuming I don't care about knowing what call came from which server, can I register all 3 call centers to SERVER-A under the same account? Or do I need to have two different accounts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Ditto here.. we were running 1.2.6.. decided to upgrade to 1.2.9.1... crash crash crash crash... so we downgraded back to 1.2.6 and have been up for weeks at a time now without issues. On 7/12/06, j [EMAIL PROTECTED] wrote: I personally have had some issues with 1.2.9.1 in production and had to revert to an older version. We are using 1.2.6 which has proven to be pretty stable. Others might have different experiences. j On Wed, 2006-07-12 at 15:31 +0200, Andrea Spadaccini wrote: Hello, I need to install Asterisk on a test machine that will soon become a production environment. Do you think that 1.2.9.1 is reliable? I read some posts that say it isn't as good as the previous versions. Should I install 1.2.8 or 1.2.7.1? Please give me an advice! Thanks in advance, ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Don Pobanz wrote: Rich Adamson wrote: For others to better understand the issues, did you install asterisk as a distro or download v1.2 svn code and compile? I downloaded the 1.2.9.1 release from the www.asterisk.org website and compiled it. If you installed source via svn, did you try make update to pick up any patches? I have not used svn. This is the 1.2.9.1 release with no patches applied. Okay. There has been about 20 or so (pure guess on the actual number) patches applied to the v1.2 code in svn in the last few weeks. I don't have a clue whether any of those patches related to queues, but would have to guess that some do. For those that are running v1.2, it certainly is not difficult to execute make install from within the asterisk source directory and pick up those updates/patches. As I understand it, the 1.2.9.1 distro is a snapshot of the svn v1.2 source code (on some specific date/time), and that executing the make update simply applies those patches that will be going into 1.2.10 (or whatever the next stable release number happens to be). So, by running 1.2.9.1 code, you're running something that is known to contain bugs. And, by not doing an update, its essentially suggesting that bug fixes are not important enough to apply them. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How do you harden an Asterisk install?
I remember reading a small write up somewhere. I think it was on the Asterisk Wiki. I can't find it anymore. It's probably a bit dated by now but some of it would still be relevant. Can anyone recommend a good guide or even some of their own suggestions. For clarity, what I mean by hardening is to make an Asterisk Server or network appliance or embedded server or whatever you want to call it, as fail safe, stable, and reliable as possible. Just like an expensive traditional PBX. This is for a small business application of 50 extensions or less. It can't be too crazy like redundant servers or anything like that. I am looking for ideas like RAID 1, redundant power supply, cron job to reboot every night (yuck!), disable caching(?), Astlinux on embedded with CF, yada yada! Anyway to set up automatic failover to a second Network Card with same IP if primary network card fails? That is one point of failure I haven't found a way around yet. Failure of the managed switch is another one I get a bit paranoid about. Switches generally don't fail but I'd like to have some sort of fail safe plan. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mediatrix 1204 and Asterisk 1.2.9 stops working intermittently
I have a mediatrix 1204 which is connecting with asterisk 1.2.9. I have created a howto as well, but I am now encountering one interesting problem that did not occur with asterisk 1.0.8. Every so often the mediatrix will nothandoff a call to the asterisk box, until I change the login name on both the mediatrix box and asterisk for sip authentication. After that everything runs fine..for a while. I have scanned the logs but the onlyerror I seeevery once in awhile is "chan_sip.c:10988 in handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 1877XXX in context default" 1877XXX is our 800 number Has anyone had a similar issue? If I were to add a hint for the mediatrix how would I do this? Thanks Julian Date: Thu, 13 Jul 2006 12:38:31 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk version: 1.2.9.1 or older? Dittohere..wewererunning1.2.6..decidedtoupgradeto1.2.9.1... crashcrashcrashcrash...sowedowngradedbackto1.2.6andhave beenupforweeksatatimenowwithoutissues.On7/12/06,j[EMAIL PROTECTED]wrote: Ipersonallyhavehadsomeissueswith1.2.9.1inproductionandhadto reverttoanolderversion. Weareusing1.2.6whichhasproventobeprettystable. Othersmighthavedifferentexperiences. j OnWed,2006-07-12at15:31+0200,AndreaSpadacciniwrote: Hello, IneedtoinstallAsteriskonatestmachinethatwillsoonbecomea productionenvironment. Doyouthinkthat1.2.9.1isreliable?Ireadsomepoststhatsayit isn'tasgoodasthepreviousversions.ShouldIinstall1.2.8or 1.2.7.1? Pleasegivemeanadvice! Thanksinadvance, ___ --BandwidthandColocationprovidedbyEasynews.com-- asterisk-usersmailinglist ToUNSUBSCRIBEorupdateoptionsvisit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --BandwidthandColocationprovidedbyEasynews.com-- asterisk-usersmailinglist ToUNSUBSCRIBEorupdateoptionsvisit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you harden an Asterisk install?
For the NIC setup you can bond 2 cards together for redundency. Take a look here for some more info on bonding. http://www.redhat.com/docs/manuals/enterprise/RHEL-4-Manual/ref-guide/s1-networkscripts-interfaces.html#S2-NETWORKSCRIPTS-INTERFACES-CHAN On 7/13/06, shadowym [EMAIL PROTECTED] wrote: I remember reading a small write up somewhere. I think it was on the Asterisk Wiki. I can't find it anymore. It's probably a bit dated by now but some of it would still be relevant. Can anyone recommend a good guide or even some of their own suggestions. For clarity, what I mean by hardening is to make an Asterisk Server or network appliance or embedded server or whatever you want to call it, as fail safe, stable, and reliable as possible. Just like an expensive traditional PBX. This is for a small business application of 50 extensions or less. It can't be too crazy like redundant servers or anything like that. I am looking for ideas like RAID 1, redundant power supply, cron job to reboot every night (yuck!), disable caching(?), Astlinux on embedded with CF, yada yada! Anyway to set up automatic failover to a second Network Card with same IP if primary network card fails? That is one point of failure I haven't found a way around yet. Failure of the managed switch is another one I get a bit paranoid about. Switches generally don't fail but I'd like to have some sort of fail safe plan. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection and Sangoma cards
On 7/12/06, El Flynn [EMAIL PROTECTED] wrote: Are you only having this problem for call parking? Any issues when the caller is navigating an IVR? We're not running an IVR on this particular system. Here's the strange thing: the DTMF is not coming from the inbound caller but rather, the agents who are using Polycom SIP phones and trying to park the calls. I'm not sure how Asterisk's DTMF detection works but in this instance, despite what I said earlier, I'm starting to think that the Sangoma card is not involved. It gets even stranger: when I call into the system using a Polycom phone (over POTS, on a different PBX in a different state), the agent can park my call. When I call in with my cell phone, the agent cannot park the call. Strange! Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inc.com Names Mark Spencer of Digium to its “30 Under 30: America’ s Coolest Young Entrepreneurs”
Very cool, congratulations Mark. Regards Josué 2006/7/13, Randall H. [EMAIL PROTECTED]: Congrats Mark !___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DTMF detection and Sangoma cards
I have not read through this entire thread, but I used to experience an issue on Polycom phones where if you were on a call, interacting with an IVR menu, if a call came in on the second line, you could not interact with the IVR via DTMF, until the call coming in on the second line went away. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Snell Sent: Thursday, July 13, 2006 1:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF detection and Sangoma cards On 7/12/06, El Flynn [EMAIL PROTECTED] wrote: Are you only having this problem for call parking? Any issues when the caller is navigating an IVR? We're not running an IVR on this particular system. Here's the strange thing: the DTMF is not coming from the inbound caller but rather, the agents who are using Polycom SIP phones and trying to park the calls. I'm not sure how Asterisk's DTMF detection works but in this instance, despite what I said earlier, I'm starting to think that the Sangoma card is not involved. It gets even stranger: when I call into the system using a Polycom phone (over POTS, on a different PBX in a different state), the agent can park my call. When I call in with my cell phone, the agent cannot park the call. Strange! Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] quad T1 pri
I am connecting my Quad T1 card to the Siemens HIpath 4000 with pri. Port 1 is working just fine. But port 2 is not working. I think I have my configuration correct (see below). Is there something special about configuring 2 PRI? I have done it with dual T1 no problem. I am using asterisk 1.2.9.1 libpri-1.2.3 and zaptel 1.2.6. THanks, Jerry -- zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=1,1,0,esf,b8zs bchan=25-47 dchan=48 zapata.conf: signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=smvoice-incoming group=1 channel = 1-23 signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=smvoice-incoming group=1 channel = 25-47 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New VEGASTREAM 400 Devices Cheap
Michael Workman wrote: The Vega 400 connects [-snip-] Did you not catch the name of the list? What part of Non-Commercial Discussion did you not get? Post this over on the -biz list where it belongs. W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inc.com Name s Mark Spencer of Digium to its “30 U nder 30: America’s Coolest Young Entrepre neurs”
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Randall H. wrote: Congrats Mark ! Link and excerpt here: http://www.sineapps.com/news.php?rssid=1366 - -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFEtonJS6d5vy0jeVcRAn7hAJ9hubkLGxQk/fnSiw6o6iw1y6mHLACfYWYy fri5OteCHuztxlTdlrm6sSY= =ZYPu -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP adapters questions
I have a question on SIP adapters : I understand they translate SIP voice, callerID and such to PSTN, but what about phone functionalities like3-way conferences and transferts? How do they handle that? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] quad T1 pri
Should be span 1 for the for T1 and span 2 for the second T1 in your config. They are both span 1. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Geis Sent: Thursday, July 13, 2006 12:47 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] quad T1 pri I am connecting my Quad T1 card to the Siemens HIpath 4000 with pri. Port 1 is working just fine. But port 2 is not working. I think I have my configuration correct (see below). Is there something special about configuring 2 PRI? I have done it with dual T1 no problem. I am using asterisk 1.2.9.1 libpri-1.2.3 and zaptel 1.2.6. THanks, Jerry -- zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=1,1,0,esf,b8zs bchan=25-47 dchan=48 zapata.conf: signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=smvoice-incoming group=1 channel = 1-23 signalling=pri_cpe switchtype=national echocancel=yes echocancelwhenbridged=yes echotraining=400 callerid=asreceived context=smvoice-incoming group=1 channel = 25-47 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] quad T1 pri
Jerry Geis wrote: zaptel.conf: span=1,1,0,esf,b8zs bchan=1-23 dchan=24 span=1,1,0,esf,b8zs bchan=25-47 dchan=48 you don't have span 2 listed. span=2,2,0,esf,b8zs Don Pobanz ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
On Jul 13, 2006, at 9:39 AM, Rich Adamson wrote: snip Okay. There has been about 20 or so (pure guess on the actual number) patches applied to the v1.2 code in svn in the last few weeks. I don't have a clue whether any of those patches related to queues, but would have to guess that some do. For those that are running v1.2, it certainly is not difficult to execute make install from within the asterisk source directory and pick up those updates/patches. I don't think that will add the patches, will it? I thought this builds from the already present sources? As I understand it, the 1.2.9.1 distro is a snapshot of the svn v1.2 source code (on some specific date/time), and that executing the make update simply applies those patches that will be going into 1.2.10 (or whatever the next stable release number happens to be). Doesn't make update look at CVS? So, by running 1.2.9.1 code, you're running something that is known to contain bugs. And, by not doing an update, its essentially suggesting that bug fixes are not important enough to apply them. I would love to see a simple explanation of how to update to the latest, including patches. Although I am not using queues, I have wondered about this ever since the change over to SVN, and this seems a good place to ask. Thanks for any help explaining, Marty ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Martin Joseph wrote: I would love to see a simple explanation of how to update to the latest, including patches. Although I am not using queues, I have wondered about this ever since the change over to SVN, and this seems a good place to ask. The latest release is 1.2.9.1 Anything in SVN is development code. -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inc.com Name s Mark Spencer of Digium to its “30 U nder 30: America’s Coolest Young Entrepre neurs”
Hi everyone, I'm new in this list. I've seen the docs about agi commands, CHANNEL STATUS especifically. The format of channelname is supposed to be one of the show channel's output , Zap/1-1 is fine but for a sip or iax device, it's attached an id number to the call. how can i verify it then? and if the device is a multiline phone, i'd like to know if they have an active phone call at least. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Eric ManxPower Wieling wrote: Martin Joseph wrote: I would love to see a simple explanation of how to update to the latest, including patches. Although I am not using queues, I have wondered about this ever since the change over to SVN, and this seems a good place to ask. The latest release is 1.2.9.1 Anything in SVN is development code. Not true. If one has installed the source code for v1.2, a make update applies only those changes that have been committed to the v1.2 svn branch. Been doing it for a long time. :) R. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can I register multiple TERMINATORS to a single account on IAX?
I don't know about IAX, but what you are trying to do should work in the SIP world. Obviously, it won't work for inbound... as it will send the call the box that sent the latest registration.There is always the ole, test it and see, approach. I guess the question comes to mind... why on earth don't you you just create seperate entries in IAX.conf?bpOn 7/13/06, Matt [EMAIL PROTECTED] wrote:Yes that solves the issue completed.. and that's what I did... but that got me to wondering if you could just register multiple serversto a single user-name (probably not the best way to do things hehe butwondered if it would work since you aren't pushing anything back to them).At any rate, I did just go with the accountcode=blah for each account.On 7/13/06, William Piper [EMAIL PROTECTED] wrote: Assuming I understand what you are trying to do, just put accountcode=whatever in your iax.conf for each user. bp On 7/13/06, Matt [EMAIL PROTECTED] wrote: Ok, Here is my scenario I have one (1) server that does my termination. (SERVER A) I have three (3) call centers that I want to terminate outbound calls to the terminator (SERVER A). Assuming I don't care about knowing what call came from which server, can I register all 3 call centers to SERVER-A under the same account? Or do I need to have two different accounts? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NuFone, please send the log file
Ronald Wiplinger wrote: Kevin P. Fleming wrote: Can we please keep the discussions about carriers, money, jobs, work, etc. off of this list? This is not the place to discuss your experiences with _any_ company, it's a place to talk about Asterisk and using Asterisk. Please move flamewars and similar discussions to some other forum. I agree with you! Which place is in your opinion the right place? As long there is no other place, such messages will always pop up. How about the Asterisk-biz list? W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Email notification of voicemail
Try having nothing after the name in your voicemail.conf: 1234 = 1234,The Marquis de Sade Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jul 12, 2006, at 11:17 AM, Kevin Savoy wrote: I have attach=no in my voicemail.conf so that can't be doing it. Not sure where that sendmail command is. Don't see it in voicemail.conf or any other config in the asterisk directory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Street Sent: Wednesday, July 12, 2006 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Email notification of voicemail Kevin Savoy wrote: Asterisk is trying to send an email to users when they receive a voicemail. Can this be shut off? I have not entered any email addresses in voicemail.conf so it tries to send to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]. This of course gets rejected since the user does not exist and the root users mailbox on linux gets full of these rejection notices. I can't seem to find anywhere to tell Asterisk to stop notifying people they have voicemails. I'm using 1.2.9.1 of Asterisk. Thanks _ **Kevin Savoy** **Business Unit Telecom Analyst** 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc --- - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You could try commenting out: attach=yes Also, if you don't want any emails sent ever for any voice mail users you could probably uncomment the following line and give it a bogus path to the mailer. ;mailcmd=/usr/sbin/sendmail -t There is probably a better way to do this but we have never needed to turn it off so I am not sure. Hope this helps. -- VoIP Street Origination/Termination with SUPERIOR customer service! http://www.VoIPstreet.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Getting Cut Off after 5 seconds
I'm having a problem with voicemail messages (sometimes) getting cut off after 4 or 5 seconds. Here's what show's up in my the logs: Jul 13 14:48:17 hermes asterisk[3125]: VERBOSE[4614]: -- Playing '/var/spool/asterisk/voicemail/DLRVM/1060/unavail' (language 'en') Jul 13 14:48:26 hermes asterisk[3125]: VERBOSE[4614]: -- Playing 'vm-intro' (language 'en') Jul 13 14:48:31 hermes asterisk[3125]: NOTICE[4614]: sched.c:224 in ast_sched_add_variable: Scheduled event in 0 ms? Jul 13 14:48:31 hermes asterisk[3125]: VERBOSE[4614]: -- Playing 'beep' (language 'en') Jul 13 14:48:32 hermes asterisk[3125]: VERBOSE[4614]: -- Recording the message Jul 13 14:48:32 hermes asterisk[3125]: VERBOSE[4614]: -- x=0, open writing: /var/spool/asterisk/voicemail/DLRVM/1060/tmp/x5z59O format: wav, 0x81e1c80 Jul 13 14:48:36 hermes asterisk[3125]: WARNING[4614]: file.c:180 in ast_writestream: Natural write failed Jul 13 14:48:36 hermes asterisk[3125]: WARNING[4614]: app.c:638 in __ast_play_and_record: Error writing frame Running current (or pretty near) asterisk trunk, but I have similar issues with trunk code from the last 4 weeks or so. Asterisk SVN-trunk-r37380 built by root @ hermes on a i686 running Linux on 2006-07-11 19:28:30 UTC There's plenty of disk space on the partition and permissions are OK. I'm a little stumped. Any ideas? Thanks, -Josh C. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New York city Asterisk consultants
Dean Collins wrote: If there are any freelance asterisk consultants (or small companies) who Does anyone pay attention to the non-commercial part of the list name any longer?!? HELLO PEOPLE - THERE IS A -biz LIST FOR STUFF LIKE THIS!! W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
So let's cut to the chase here... If you want to run a production server with queues, which version should you be running to get 30+ days of uptime without needed a reset? W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Email notification of voicemail
Thanks for replying. Have tried that. If I don't specify an email address it then takes the first name and last name and then the domain of the pbx. For example 1234 = 1234,Bob Smith I then get: [EMAIL PROTECTED] Which of course fails because that address doesn't exist. Any other ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Thursday, July 13, 2006 2:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Email notification of voicemail Try having nothing after the name in your voicemail.conf: 1234 = 1234,The Marquis de Sade Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jul 12, 2006, at 11:17 AM, Kevin Savoy wrote: I have attach=no in my voicemail.conf so that can't be doing it. Not sure where that sendmail command is. Don't see it in voicemail.conf or any other config in the asterisk directory. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of VoIP Street Sent: Wednesday, July 12, 2006 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Email notification of voicemail Kevin Savoy wrote: Asterisk is trying to send an email to users when they receive a voicemail. Can this be shut off? I have not entered any email addresses in voicemail.conf so it tries to send to [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]. This of course gets rejected since the user does not exist and the root users mailbox on linux gets full of these rejection notices. I can't seem to find anywhere to tell Asterisk to stop notifying people they have voicemails. I'm using 1.2.9.1 of Asterisk. Thanks _ **Kevin Savoy** **Business Unit Telecom Analyst** 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc --- - ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You could try commenting out: attach=yes Also, if you don't want any emails sent ever for any voice mail users you could probably uncomment the following line and give it a bogus path to the mailer. ;mailcmd=/usr/sbin/sendmail -t There is probably a better way to do this but we have never needed to turn it off so I am not sure. Hope this helps. -- VoIP Street Origination/Termination with SUPERIOR customer service! http://www.VoIPstreet.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How do you harden an Asterisk install?
shadowym wrote: I remember reading a small write up somewhere. I think it was on the Asterisk Wiki. I can't find it anymore. It's probably a bit dated by now but some of it would still be relevant. Can anyone recommend a good guide or even some of their own suggestions. For clarity, what I mean by hardening is to make an Asterisk Server or network appliance or embedded server or whatever you want to call it, as fail safe, stable, and reliable as possible. Just like an expensive traditional PBX. This is for a small business application of 50 extensions or less. It can't be too crazy like redundant servers or anything like that. I am looking for ideas like RAID 1, redundant power supply, cron job to reboot every night (yuck!), disable caching(?), Astlinux on embedded with CF, yada yada! Anyway to set up automatic failover to a second Network Card with same IP if primary network card fails? That is one point of failure I haven't found a way around yet. Failure of the managed switch is another one I get a bit paranoid about. Switches generally don't fail but I'd like to have some sort of fail safe plan. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users You are talking about 2 things: (1) How to harden a linux box (2) How to do failover. for (1), be sure telnet, ftp and any other service you do not need is off. Move standard services to non-standard ports, especially web and ssh. Do not run a name server on the box. For (2): You need to have a secondary box that runs a mirror copy of Asterisk and mysql and pretty much has everything else configured the same. mysql should be replicated to the second box. You then run a program on the second box that pings the first box. If the first box fails the second takes over the first box's IP and runs with it. There are heartbeat programs that can help out with this. W ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extensions not busy showing as busy
I ran into an odd problem last night. I had to walk someone through troubleshooting a problem and they didnt have a lot of knowledge, so I couldnt garner a lot of info. So my question is two fold regarding the issue described below: What could I have asked to troubleshoot this better? Any idea what the problem might have been? Issue: extensions registered on the Polycom 601s were showing as busy. Do not disturb wasnt set, nor any forwarding conditions. However, if you forwarded to a different extension in a busy condition, it would forward. (That tipped me off that Asterisk thought the extension was busy.) So then we did a sip show channels, and sure enough, those extensions were listed. At this point I didnt want to frustrate the user much, so I had her do a reboot. Sure enough, the reboot solved the problem. (Im sure just restarting Asterisk would have done it as well, but the reboot command is easier for her to type.) Extremely dumb, but effective solution. I wished I could have fixed it differently. Also, the extensions in question had been in that condition for a few days. We had already rebooted the phones and had them register (and they did register.) That didnt fix the issue. --- Brian Vincent Copper Mountain Telecom [EMAIL PROTECTED] __ ConfidentialityWarning:Thismessageandanyattachmentsareintendedonlyfortheuseoftheintendedrecipient(s), areconfidential,andmaybeprivileged.Ifyouarenottheintendedrecipient,youareherebynotifiedthatanyreview, retransmission,conversiontohardcopy,copying,circulationorotheruseofthismessageandanyattachmentsisstrictly prohibited.Ifyouarenottheintendedrecipient,pleasenotifythesenderimmediatelybyreturne-mail,anddeletethis messageandanyattachmentsfromyoursystem.Thankyou. __ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
Warren (mailing lists) wrote: So let's cut to the chase here... If you want to run a production server with queues, which version should you be running to get 30+ days of uptime without needed a reset? Someone else needs to reply to this one as I don't run queues. R. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Email notification of voicemail
Can you send me (or pastebin) your voicemail.conf? A. On Jul 13, 2006, at 12:35 PM, Kevin Savoy wrote: Thanks for replying. Have tried that. If I don't specify an email address it then takes the first name and last name and then the domain of the pbx. For example 1234 = 1234,Bob Smith I then get: [EMAIL PROTECTED] Which of course fails because that address doesn't exist. Any other ideas? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Rodgers Sent: Thursday, July 13, 2006 2:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Email notification of voicemail Try having nothing after the name in your voicemail.conf: 1234 = 1234,The Marquis de Sade ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk version: 1.2.9.1 or older?
We have just come through our busy tax season for our tax line queue on 1.2.1 with zero problems :-) Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web: http://www.dnv.org RSS Feed: http://www.dnv.org/rss.asp On Jul 13, 2006, at 12:41 PM, Rich Adamson wrote: Warren (mailing lists) wrote: So let's cut to the chase here... If you want to run a production server with queues, which version should you be running to get 30+ days of uptime without needed a reset? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
