Re: [asterisk-users] stop log/debug messages into /var/log/messages
Did you look at logger.conf? From: [EMAIL PROTECTED] [EMAIL PROTECTED] On Behalf Of bilal ghayyad [EMAIL PROTECTED] Sent: Sunday, September 16, 2007 5:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] stop log/debug messages into /var/log/messages Dear Benjamin; OK friend, things are clear. But now I came to the same original issue that you asked about it, which is the ability to stop the log/debug messages into /var/log/messages. Same like your situation, the messages is comment (;) and even the logges are written to the /var/log/messages, so why that is happening? Did u find answer for that? Regards Bilal --- Benjamin Jacob [EMAIL PROTECTED] wrote: Hello Bilal, You have to do quite some reading mate, before you post your questions(like your nat and canreinvite questions). Anyway, look into /etc/asterisk/manager.conf for the required directories where Asterisk stores its various files/directories. Then read up logger.conf and look at some examples on the net as well. cheerz - Ben. bilal ghayyad wrote: Hi Benjamin; I am also interested in the same issue, but I would like to know how you can know where these logs are stored (in which file and path)? I readed that syslog, can you please help me about that? Regards Bilal Ghayad Mobile: 00965 9849460 --- When you access the A*k console, is this via a tty connection (ssh/telnet), or actually on the physical console of the server? I don't think it's A*k that's directly logging to the console - the config doesn't show that... I'm guessing, that you're accessing A*k via the local terminal, and that your syslog config for the server is configured to log this to messsages Maybe.. hmmm. interesting. need to investigate syslog now. Even me thinks, as far as I've read(abt logger and the existing configuration), it shouldn't be writing to any syslogs. btw, am accessing the * console via ssh. thanks for ur help. - Benjamin Jacob. Got a little couch potato? Check out fun summer activities for kids. http://search.yahoo.com/search?fr=oni_on_mailp=summer+activities+for+kidscs=bz EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. Yahoo! oneSearch: Finally, mobile search that gives answers, not web links. http://mobile.yahoo.com/mobileweb/onesearch?refer=1ONXIC ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Single sign on PC + phone?
This is an interesting idea, did you come up with anything? Are your users logging into an AD domain? A script to interact with the Asterisk server could be run after login which adds an extension mapping the user to the phone. One set of extensions for the users (which is published) and another set of real extensions for the phones and when a user extension is dialed it rings the phone extension. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Patrick Sent: Monday, March 12, 2007 8:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Single sign on PC + phone? Hi all, Does anyone have any experience with creating a Single sign on (SSO) concept where if someone logs in on their PC the phone next to that PC is also automatically assigned to that user? TIA, Patrick ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.9/719 - Release Date: 3/12/2007 8:41 AM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.10/720 - Release Date: 3/12/2007 7:19 PM ___ --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] What happend to voip-info?
I would be willing to mirror it also…. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Wednesday, March 14, 2007 9:39 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] What happend to voip-info? I have not been able to get to it for a few days. I offered to mirror it several times when it was up and down a few years ago and was declined. So much good info, bits and pieces that have saved me over and over. Let’s hope it comes back up. Thanks, Steve Totaro HYPERLINK http://www.asteriskhelpdesk.comhttp://www.asteriskhelpdesk.com KB3OPB _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nir Simionovich Sent: Wednesday, March 14, 2007 6:49 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What happend to voip-info? Anyone has an idea what happend to voip-info? it stopped working about 24 hours ago. Nir S -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.10/720 - Release Date: 3/12/2007 7:19 PM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.446 / Virus Database: 268.18.10/720 - Release Date: 3/12/2007 7:19 PM ___ --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] max tnt pri voice channels 56k or 64k, does it matter, selection parameter?
If it's using RBS then 56k is the right number. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of JR Richardson Sent: Saturday, January 27, 2007 12:55 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] max tnt pri voice channels 56k or 64k,does it matter, selection parameter? Hi All, We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the voice channels connect at 56K. Does anyone have the DS0 channels connecting at 64K for voice, if so what is the parameter to select 56k or 64k channels? I'm not having any issues that I know of, just wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the Masses ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.12/653 - Release Date: 1/26/2007 11:11 AM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.14/657 - Release Date: 1/29/2007 9:04 AM ___ --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] Only secretary can call the boss, all others only reach the secretary when dial the boss extension
Why don't you just give the secretary the boss' REAL extension and give a different extension to the world that just rings the secretary? -jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Ricardo Carvalho Sent: Friday, January 26, 2007 12:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Only secretary can call the boss, all others only reach the secretary when dial the boss extension Dear all, How may I configure my extensions.conf so that only the boss's secretary can call the boss through his extension, all others when dial his extension only makes the boss's secretary phone ring, not his. If she wants, she can transfer the incoming call to the boss dialling his extension. I've tried the following, but it doesn't work: exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension) exten = _boss_extension,1,Dial(SIP/secretary_extension) This doesn't work because when the secretary tries to transfer the call to the boss (using her phone's transfer key, not #), one REFER SIP message is sent back to the caller's phone providing him the new address for whom the next INVITE should be sent. That INVITE is sent, but when reaches Asterisk, that INVITE matches this line: exten = _boss_extension,1,Dial(SIP/secretary_extension) and not this one: exten = _boss_extension/callerid_secretary,1,Dial(SIP/boss_extension) Any ideas of how may I solve this issue? Regards, Ricardo. ___ --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 -- No virus found in this incoming message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.12/653 - Release Date: 1/26/2007 11:11 AM -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.432 / Virus Database: 268.17.12/653 - Release Date: 1/26/2007 11:11 AM ___ --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 limit IAX calls
A demonstration:
exten = _X.,1,Set(GROUP()=${CALLERID(num))
exten = _X.,n,Set(CDR(AccountCode)=${CALLERID(num))
exten = _X.,n,GotoIf($[${GROUP_COUNT(${CALLERID(num))} 2]?103)
exten = _X.,n,Macro(trunk,${EXTEN},residential)
exten = _X.,n,Hangup
exten = _X.,103,Playback(allison7/all-circuits-busy-now)
exten = _X.,n,Hangup
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco
Mouta
Sent: Friday, January 19, 2007 6:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to limit IAX calls
Take a look on:
Dialplan applications:
GetGroupMatchCount([EMAIL PROTECTED])
SetGroup([EMAIL PROTECTED])
Using this two applications you can deploy a max calls control inside
your dialplan!
check this too: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup
Hope it helps
On 1/19/07, Barzilai Spinak [EMAIL PROTECTED] wrote:
The SIP channels have a call-limit parameter (which is badly
documented and I haven't tested yet)
How can I have the same behaviour for IAX channels? I can't see anything
related to it.
Ah, I'm using Asterisk 1.2.13... maybe there is something in the 1.4
versions... but I can't change to 1.4 right now because of MFC/R2
BarZ
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RE: [asterisk-users] Polycom 601 Contacts List
There is an index in the configuration file which I believe it will obey. I'll try and find it later if you haven't found it by the time I get to the office. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Wednesday, December 27, 2006 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] Polycom 601 Contacts List I don't think that's possible. We have the same issue. -Original Message- From: Dovid B [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 27, 2006 8:04 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Polycom 601 Contacts List Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the -directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in the order that I specified and NOT in alphabedical order. Thanks a lot. Dovid ___ --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] Searching the list
I'm not sure if there is a more official method but Google has always been my friend when searching the lists. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Greene Sent: Wednesday, December 27, 2006 12:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Searching the list Hey guys. I am new to the list and would like to know how to search it so that I do not post any questions that have already been answered (like this one) - Mark ___ --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] AstManProxy - Manager
I don't use many of the features of astmanproxy but it does work. I use it to capture events from several servers. Some of these are running the 1.4 beta releases. -Jonahtan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Wednesday, December 20, 2006 9:21 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AstManProxy - Manager On Wed, Dec 20, 2006 at 02:57:17PM +0100, Olivier wrote: Hi, Is AstManProxy an alive project ? It seems to me that no development are ongoing. Will AstManProxy comply with Asterisk 1.4 ? Last release seems to be from 3 monthes ago. 1.4 has not been released yet, as you recall. Anyway, latest astmanproxy seems to have a basic support for the manager over HTTP protocol of 1.4. But maybe this is just me reading the docs wrong. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --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] Parsing Area Code from CallerID
${Number:-10:3} if I recall correctly would give you 3 characters
starting at the 10th from the end.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of John French
Sent: Tuesday, December 19, 2006 10:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Parsing Area Code from CallerID
How would I parse the area code from this variable? Number=2515551212
Sorry for the dense question, I don't seem to be able to find an
appropriate function for parsing left to right.
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RE: [asterisk-users] Parsing Area Code from CallerID
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bruce Ferrell
Sent: Tuesday, December 19, 2006 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Parsing Area Code from CallerID
John French wrote:
How would I parse the area code from this variable?
Number=2515551212
Sorry for the dense question, I don't seem to be able to find an
appropriate function for parsing left to right.
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NPA=${NUMBER:0:3}
--
One day at a time, one second if that's what it takes
That works if the number is always NPA-NXX-. If you end up with
+1NPANXX or 1NPANXX then you don't have the right data.
-Jonathan
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RE: [asterisk-users] asterisk to asterisk - to zap
I may be making this easier than it is but something like this should
work:
A:
DIAL(IAX2/${ASTERISKB}/[EMAIL PROTECTED])
B:
[context]
exten = EXTEN,1,DIAL(Zap/${EXTEN})
I have this scenario also except we have numerous A servers connecting
via the PRI lines on B servers.
-Jonathan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Pryakhin
Dimitry
Sent: Monday, December 18, 2006 8:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] asterisk to asterisk - to zap
Hello
that might would be an easy question for someone, but im in doubt
Is there any possibility to pass a call from one asterisk to another and
then to ZAP channel.
For instance
I have
A asterisk with numbering 45670
B asterisk with numbering 45680
second asterisk has TE110P card with single PRI port connected to
Siemens EWSD.
When I originate call from asterisk B I reach the world thru ZAP,
when I call from asterisk A I reach numbering of asterisk B but cant
get to the PSTN network.
ASTERISK---ASTERISK-ZAP-PSTN
Should I have OpenSER for that and terminate my call on CISCO AS5350 or
something?
Thanks
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RE: [asterisk-users] Hardware TDM Switching
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, December 14, 2006 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hardware TDM Switching [EMAIL PROTECTED] wrote: Do anybody know, if there is a way to connect 2 zap-channels with Hardware TDM Switching? It's called DACS. See the /etc/zapata.conf config file sample. He means /etc/zaptel.conf I thinkright? ___ --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] Hardware TDM Switching
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Eric ManxPower Wieling Sent: Thursday, December 14, 2006 4:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hardware TDM Switching [EMAIL PROTECTED] wrote: Do anybody know, if there is a way to connect 2 zap-channels with Hardware TDM Switching? It's called DACS. See the /etc/zapata.conf config file sample. Is there a way to do this dynamically? Something in the dialplan that would trigger this? I have calls coming in on one PRI and depending on the DID they go out on a second PRI (going to a dialup pool). I had hoped the Zaptel drivers would do a bridge of these channels but that doesn't happen. ___ --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 Manager
CLIPPED I would have some kind of user 1010 (the actual extension and username too) Let's say that in manager.conf i would have again some user 1010 but i would like that this user can only see the events associated to the extension 1010 ... CLIPPED I am pretty sure that using the proxy, astmanproxy, you can achieve this goal. It is recommended to use the proxy so that there is only one connection to the server and all the other applications will connect to the proxy. -Jonathan ___ --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 Manager
Not meaning to argue with you but the proxy replaces the manager interface so it could most likely be a seamless replacement to your application. It was for all but one of my applications and the problem there was in the way I parsed the startup string. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Daniel Gradecak Sent: Tuesday, December 12, 2006 1:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Manager Hello Jonathan, thank you for answering ... I read about astmanproxy but it cannot help me. I am using asterisk-java all my application is written in java too. I already have a kind of proxy ad I am not doing several connection to the asterisk manager. I am afraid this is not helping me much. Anyway, I have done this in my proxy but i thought i could avoid things like that in my code... I did not test the asterisk manager contexts and dial plan, so I wonder if I make a call via astman from 1010 to a GSM and that 1010 is in a context that is not allowing calls to GSM would astman execute it anyway or would it look also in the 1010 context? I am asking that because my system guys are not available until friday ... Jonathan k. Creasy wrote: CLIPPED I would have some kind of user 1010 (the actual extension and username too) Let's say that in manager.conf i would have again some user 1010 but i would like that this user can only see the events associated to the extension 1010 ... CLIPPED I am pretty sure that using the proxy, astmanproxy, you can achieve this goal. It is recommended to use the proxy so that there is only one connection to the server and all the other applications will connect to the proxy. -Jonathan ___ --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] CLI History
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Monday, December 11, 2006 1:37 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] CLI History -Original Message- From: Dave Cotton [mailto:[EMAIL PROTECTED] Sent: Monday, December 11, 2006 10:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] CLI History On Mon, 2006-12-11 at 10:31 -0700, Douglas Garstang wrote: What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Nothing wrong here. Can you possibly be a little more specific on why it isn't a problem? Doug. ___ Sounds like it is working as intended if that is the last command you executed. I'd say be more careful when executing commands. ___ --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] CLI History
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Carla Schroder Sent: Monday, December 11, 2006 2:17 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] CLI History On Monday 11 December 2006 9:31 am, Douglas Garstang wrote: What's wrong with the Asterisk CLI history? When I exit the CLI, and re-enter, the last command in the history always defaults to 'stop now'. This is very bad, and it's caused accidental shutdowns more than once. Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI A No such command 'A' (type 'help' for help) hera*CLI B No such command 'B' (type 'help' for help) hera*CLI C No such command 'C' (type 'help' for help) hera*CLI D No such command 'D' (type 'help' for help) hera*CLI E No such command 'E' (type 'help' for help) hera*CLI [10:[EMAIL PROTECTED](pbx3):~]# asterisk -trv Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. = Connected to Asterisk 1.2.9.1 currently running on hera (pid = 17399) Verbosity is at least 3 hera*CLI stop now -- I pressed the UP arrow upon re-entering the console! Mine appears to work: ##Connected to Asterisk and execute stop now: dragon*CLI stop now dragon*CLI Disconnected from Asterisk server ## Restarted Asterisk: [EMAIL PROTECTED] ~]# asterisk -p ## Connected to Asterisk then ran exit: [EMAIL PROTECTED] ~]# asterisk -r Asterisk 1.4.0-beta3, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistributeit under certain conditions. Type 'show license' for details. == === Connected to Asterisk 1.4.0-beta3 currently running on dragon (pid = 32521) dragon*CLI exit ## Connected to Asterisk Again and hit the up arrow: [EMAIL PROTECTED] ~]# asterisk -r Asterisk 1.4.0-beta3, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer [EMAIL PROTECTED] Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistributeit under certain conditions. Type 'show license' for details. == === Connected to Asterisk 1.4.0-beta3 currently running on dragon (pid = 32521) dragon*CLI exit Exit is displayed not stop now. If you hit A and it's an invalid command...maybe that is your problem... ___ --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] Polycom 501 config questions
Title: RE: [asterisk-users] Polycom 501 config questions Dumb question here: Why the need to dial 9 for an outside line? If your extensions are less than 7 digits long then you know anything "XXX." is an outside call Maybe this isn't true everywhere, just curious. -Jonathan From: [EMAIL PROTECTED] on behalf of MikeSent: Thu 8/31/2006 10:46 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] Polycom 501 config questions I was expecting a more elegant answer to the "9 to dial out" problem withthe Polycom 501. Sure I can change my dialplan, but that means I have toadapt my dialplan to the phone, while the opposite seems like the way to go.Thanks for the answer,Mike-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of Jerry JonesSent: August 30, 2006 6:03 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Polycom 501 config questionsOn Aug 30, 2006, at 2:58 PM, Mike wrote: Hi, I have a few questions on the Polycom 501. I am using latest firmware. 1) When I press the "Call List" button (on the left row of buttons), I get the call lists (as expected). When I press the "Directory" button, I get the choice between Directory and Call lists. How can I make this button go to Directory immediately? 2) I have 2 extensions on my 501. (let's say 101 and 102). Because of my dialplan, it actually matters which one I dial out with. When I pick a contact out of the directory, it calls automatically using line 101. How can I make it call with 102?Pick up 102, then select contact 3) In call lists, my numbers are listed as 555-555-. Yet my asterisk dial plan requires me (by design) to press 9 first. How can I make the phone put the 9 by itself?It will not.either add to your contact entries, or alternatively have your dialplan add 9 to any exten longer than say 3 digits Thank you for any help you may give me, 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___--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 ___ --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] quintum Calling Card
Ive only used a Quintum a few times,sorry. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Abdul Sent: Friday, August 25, 2006 6:49 AM To: Asterisk-Users@lists.digium.com Subject: RE: [asterisk-users] quintum Calling Card Hello Jonathan, I tried in quintum to route my server with any dialed number. but i am not agble to get in quintum FXO line configuration, so i can route the call to my asterisk. do u have any about quintum how i can route calls to server once FXO line will be called? Abdul Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail. ___ --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] quintum Calling Card
Abdul, it doesnt sound like you need to do anything to the Quintum. I would recommend making your dial plan execute the AGI script of your choice no matter what number is dialed from the context where the quantum users land. -Jonathan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Abdul Sent: Thursday, August 24, 2006 8:12 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] quintum Calling Card Hi all, Could anyone provide me some usefull link or some idea, how to configure quintum as calling card purpose with Asterisk. Already i created AGI script which working with SIPURA well. But i do not have the idea about quintum how to configure so quintum will dial our asterisk calling card number. i have add [EMAIL PROTECTED] server, so if some one trying to call FXO line of quintum then quintum should dial automatically this URI and rest my AGI will do. even i don't wnat to use quintum IVR. I will be appriciate for your helps. Regards Stay in the know. Pulse on the new Yahoo.com. Check it out. ___ --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] failed calls
I am trying to track down a problem which is occurring on about 1% of the phone calls through a customers system. Layout looks like this: PSTN PRI Asterisk A IAX Trunk over point to point T1 Asterisk B SIP over LAN Polycom IP501 1) The user on the Polycom IP501 phone dials a number. 2) It is routed across the LAN to an Asterisk PBX 3) The call is then routed across the T1 via IAX to another Asterisk Server 4) This server drops the call on a PRI line 5) The callee will hear their phone ring 6) On the Polycom you hear 5-10 seconds of silence then a fast busy. 7) The callee answers but no one is there. I see the following in my debug log (on Asterisk B) but Im not sure if any of these messages are abnormal: Aug 21 08:39:18 DEBUG[16560] channel.c: Didn't get a frame from channel: SIP/101-40c4 Aug 21 08:39:18 DEBUG[16560] channel.c: Bridge stops bridging channels SIP/101-40c4 and IAX2/ROUTING-6 Aug 21 08:39:18 DEBUG[16560] chan_iax2.c: We're hanging up IAX2/ROUTING-6 now... Aug 21 08:39:18 DEBUG[16560] app_dial.c: Exiting with DIALSTATUS=ANSWER. Aug 21 08:39:18 DEBUG[16560] chan_sip.c: update_call_counter(101) - decrement call limit counter Anyone have any ideas on this? -Jonathan Jonathan Creasy Network Engineer BluegrassNet Development www.bgnd.com www.bluegrass.net o. 502-589-4638 c. 502-889-5567 h. 812-206-1830 ___ --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] Quick One - PHP Script to restart Asterisk
Title: Re: [asterisk-users] Quick One - PHP Script to restart Asterisk ?php execute "asterisk -rx 'restart when convienent"; ? Not the exact syntax but should be enough to get you going. From: [EMAIL PROTECTED] on behalf of Paul HalesSent: Fri 8/11/2006 3:51 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] Quick One - PHP Script to restart Asterisk We did this for a customer completely in the dialplan, with the Asteriskinternal database.I don't have the coding here, but I know it involved the read command,followed by putting the number keyed into the internal database.(as something like ah/mobile)later,PaulHOn Fri, 2006-08-11 at 12:56 +1000, Corporate IT Solutions - MichaelDunne wrote: I have spent the best part of half the morning googling a solution to this but nothing has jumped out at me. Is there a simple method of allowing dynamic changes to the extensions via a web interface without having to go the @home method. All I want is to make a webpage to select which person gets the call redirections after hours, then reload the extensions/pbx_config. Basically, a MySQL database will contain the phone numbers of individuals. Another table will have the "Time Of Day" configuration in it. The extensions.conf will have all the configurations for the extensions for after hours redirection. All I want is to dump out a new "timeofday.conf" into /etc/asterisk with the updated refereces, then a last command like reload pbx_config. A quick link with example code would be nice. ___ --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 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] Hotels...
2) Phone activation at check-in/phone de-activation and billing at check-out. Are there GUI tools for this, or should I write my own back/front end? The integration with the hotel systems for the activation/deactivation and billing can be tricky. Check the archives for some discussions on this topic. ___ --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: Polycom compatible phone for Asterisk
Very happy with the 501 and 601. So far, like the 430 as well. The 301 is good for what it is but the display and lack of speakerphone are annoying to me. They are all very stable and compatible though. The provisioning on these phones is excellent as well. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of andrew matthews Sent: Wednesday, July 12, 2006 10:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Re: Polycom compatible phone for Asterisk I use the 301, 501, and 601 with asterisk daily. 301's are kind of cheesy and feel cheap to me but the 501 is rock solid. On 7/12/06, (AstATN) [EMAIL PROTECTED] wrote: 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 ___ --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] H.264 and Asterik?
Haven't read this whole thread (got way behind in this list :) ) Polycom has a softphone with video support also. Not sure if it is good or not, just downloaded the trial version to test it out. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Erick Weber V. Sent: Saturday, May 20, 2006 2:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] H.264 and Asterik? Kevin: Thanks for the info, I think I will buy the video phones Erick W. - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, May 19, 2006 6:18 PM Subject: Re: [Asterisk-Users] H.264 and Asterik? Erick Weber V. wrote: Dose someone know if the latest version of asterisk support H.264? Asterisk SVN trunk (which will become Asterisk 1.4) supports H.264, and I have a Grandstream H.264 phone on my desk right now which I am testing with it (and it works fine!). ___ --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] best hardphone for Asterisk?
I'll second that. I really like the provisioning features. My customers prefer the 501 because they like the layout and speaker phone functionality. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joshua West Sent: Friday, June 23, 2006 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] best hardphone for Asterisk? I find the Polycom Soundpoint 301 and 501 models to be great phones. Christian Victor wrote: Crazy Boy schrieb: We have implemented Asterisk in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost? I would say a Swissvoice IP 10S, a Snom 300 or - if you want better quality - a Polycom 300. The Snom looks good and is solid, the Swissvoice is similar plus it supports PoE, the Polycom is a bit more expensive but worth the additional cost. Chris ___ --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 -- Joshua West Linux Infrastructure Engineer Boston Engineering Corporation http://www.boston-engineering.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] Forwarded Calls crash the system on 64 bit
I have a strange problem. I have a central server with my PRI on it. There are three peripheral servers connected via IAX. I have a 64bit system for my central server and the backup system is a 32bit system. If I have forwarding (sip redirect) turned on and forwarding to an outside number (i.e. my cell phone) when the 64bit system is in the middle it will crash. The Asterisk process doesn't actually crash so there is no backtrace. It uses about 99% of the CPU and all IAX channels go down and IAX will no longer accept connections. SIP calls continue as if nothing had happened although audio quality is compromised due to the CPU being used heavily. A quick restart now fixes it right up. If I bring the 32 bit system up and have it doing the routing then there will be one-way audio on the forwarded call (termination point can here the originator but the called cannot be heard). It does not crash and all the other calls are unaffected. Both systems are CentOS 4.2 fully updated and running Asterisk 1.2.7. The peripheral system is Fedora Core 3 running Asterisk 1.2.7 also. -Jonathan ___ --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] hardware
I am not by any means recommending this to anyone but I wanted to publish this for reference. I have an Asterisk system connected to a provider via IAX trunks. There are 32 phones on our network and we have about 400 calls per day to/from our system. The hardware running this is a Pentium Pro 400mhz with 256MB ram and a 9GB scsi hard drive. Everything is working great even on such meager hardware. Our other systems are Dual Xeon servers with 1 or 2GB of ram each handling our PRI's and customer systems. -Jonathan ___ --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] Interesting Dial-Plan Question
Just appending the area code variable is not always going to be correct.
You will need to lookup (google local calling guide) the proper NPA for
the NXX you are dialing. For example, in Louisville, Ky if you dial
948-1592 you will actually reach 812-948-1592 instead of 502-948-1592
even though your callerid number would be 502-NXX-.
I have a script I'm working on that does this via an agi script, it
looks up the 7 digit dialing rules for a NPA-NXX combination and caches
the results so you don't have to do a lookup for every call. I'll post
it on the wiki when I'm done. It works but I still need to test it and
document it a little better. It is based on some scripts by other people
that I combined together.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Matt
Sent: Thursday, April 27, 2006 8:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Interesting Dial-Plan Question
Eric,
Yes.. I am setting calleridnum to be their phone number. And your
example is peachy... except for the fact that it assumes I want to go
out ZAP/g1!!
My problem is I have a very intricite routing plan that routes that
call out several different carriers depending on what you dialed.
(Long Distance, international, local, etc).
The way it works now is the dialplan just looks at the number you
dialed and routes based on that. I guess what I am asking is in
theory I should be able to do:
Look at origination number. Take first 3 digits and put into
variable. So 5705551212 becomes 570 in ${AREACODE}.
Now, look at the number we dialed. If it is (and this is where I am a
little unclear on what to do) 7 digits long then we append the
${AREACODE} variable. Else, we send it through to the dialplan as
is.
exten = _NXX,1,Dial(Zap/g1/${CALLERIDNUM::0:3}${EXTEN})
This assumes that you set the user's Caller*ID number to be their
telephone number. It takes the first 3 digits of their CALLERIDNUM
and
prepends it to the number they dialed.
See README.variables.
--
Now accepting new clients in Birmingham, Atlanta, Huntsville,
Chattanooga, and Montgomery.
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[Asterisk-Users] PRI caller ID
Below is a snipped debug on our PRI. We are getting number only for the CallerID but the telco says they are sending us Name and Number. We are getting the Name in a second frame but Asterisk is not passing it to the device it rings. The message below says Presenation allowed of network provided number which leads me to believe Asterisk thinks it should not be displaying it. Can anyone interpret this for me and maybe shed some light on why I am not getting the caller ID name displayed? I have asreceived in my Zapata.conf file. Facility (len=23, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x0f, 0x02, 0x01, 0x01, 0x06, 0x07, 0x2a, 0x86, 'H', 0xce, 0x15, 0x00, 0x04, 0x0a, 0x01, 0x00 ] [6c 0c 21 83 35 30 32 38 38 39 35 35 36 37] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '5028895567' ] [70 08 c1 33 31 35 30 35 36 36] Called Number (len=10) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3150566' ] Handle Q.932 ROSE Invoke component Don't know what to do if second ROSE component is of type 0x6 Sending Receiver Ready (84) Message type: FACILITY (98) [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 43 65 6c 6c 20 50 68 6f 6e 65 20 20 20 4b 59] Facility (len=31, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'Cell', 0x20, 'Phone', 0x20, 0x20, 0x20, 'KY' ] -Jonathan Jonathan Creasy Network Engineer BluegrassNet Development www.bgnd.com www.bluegrass.net o. 502-589-4638 c. 502-889-5567 h. 502-541-0566 ___ --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] PRI caller ID
I searched through the archives and the wiki...don't be so pissy...i missed it I guess, my bad -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Wednesday, April 19, 2006 9:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] PRI caller ID Pleaase read the archives or the wiki - you will shortly find you need a wait in your dialplan On Apr 19, 2006, at 8:10 AM, Jonathan k. Creasy wrote: Below is a snipped debug on our PRI. We are getting number only for the CallerID but the telco says they are sending us Name and Number. We are getting the Name in a second frame but Asterisk is not passing it to the device it rings. The message below says Presenation allowed of network provided number which leads me to believe Asterisk thinks it should not be displaying it. Can anyone interpret this for me and maybe shed some light on why I am not getting the caller ID name displayed? I have asreceived in my Zapata.conf file. Facility (len=23, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x0f, 0x02, 0x01, 0x01, 0x06, 0x07, 0x2a, 0x86, 'H', 0xce, 0x15, 0x00, 0x04, 0x0a, 0x01, 0x00 ] [6c 0c 21 83 35 30 32 38 38 39 35 35 36 37] Calling Number (len=14) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation allowed of network provided number (3) '5028895567' ] [70 08 c1 33 31 35 30 35 36 36] Called Number (len=10) [ Ext: 1 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '3150566' ] Handle Q.932 ROSE Invoke component Don't know what to do if second ROSE component is of type 0x6 Sending Receiver Ready (84) Message type: FACILITY (98) [1c 1d 9f 8b 01 00 a1 17 02 01 01 02 01 00 80 0f 43 65 6c 6c 20 50 68 6f 6e 65 20 20 20 4b 59] Facility (len=31, codeset=0) [ 0x9f, 0x8b, 0x01, 0x00, 0xa1, 0x17, 0x02, 0x01, 0x01, 0x02, 0x01, 0x00, 0x80, 0x0f, 'Cell', 0x20, 'Phone', 0x20, 0x20, 0x20, 'KY' ] -Jonathan Jonathan Creasy Network Engineer BluegrassNet Development www.bgnd.com www.bluegrass.net o. 502-589-4638 c. 502-889-5567 h. 502-541-0566 ___ --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] polycom blind transfer button
I could be wrong but off the top of my head I think that it is in the features section of the config file. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anton Krall Sent: Tuesday, April 18, 2006 4:47 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] polycom blind transfer button Guys, this is a weird question but has anybody disabled the blind button that appears on polycoms or know if you can disable the use of blind transfers on polycoms to make any transfer attended? Thx! ___ --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] Cannot dial out with Polycom 501 after upgrade
I can dial other extensions internally, and can get to voicemail, but when I try an outside number, I hear dial tone, the digits dialed, yet nothing happens when I press Send. Nothing appears on the Asterisk CLI screen. Did the upgrade modify the dialplan setting on your phone? This sounds suspiciously like trying to dial a number that is not matched or allowed by the dialplan. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom TOS
Does anyone know the format for the TOS element in the Polycom config? -Jonathan Jonathan Creasy Network Engineer BluegrassNet Development www.bgnd.com www.bluegrass.net o. 502-589-4638 c. 502-889-5567 h. 502-541-0566 ___ --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] OT: local calling guide
Anyone know what has happened to the local calling guide? http://members.dandy.net/~czg/search.html -Jonathan ___ --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 restrict simultaneous phone registrations
I apologize if this information is posted elsewhere. Unfortunately I haven't found it yet if it is. I'm not familiar with the channel counting features could you please explain? Also, how are you tagging the phones to account codes? You can limit calls using the set/check group commands. http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SetGroup Account codes are set either by using the Set function or the accountcode= property in the SIP/IAX conf files. -Jonathan ___ --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: Monitor or mixmonitor
Title: Re: [Asterisk-Users] Blocked channels,according to our telco... leading to CONGESTION status Id say try it out and see what the CPU load is. Its not that hard to drop it in your dialplan and give it a try. Its much easier than figuring out all the possible variables in your setup that might also affect the performance. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wai Wu Sent: Monday, April 03, 2006 10:30 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Monitor or mixmonitor Hi all, I am setting up a script to record all the call. There are two app for recording. Monitor and Mixmonitor, one mixing the audio on the fly and one mixing it at the end but also allow a option not to mixing the audio at all. If mixing the audio on the fly is not that taxing on the CPU, I would like to use 'mixmonitor' app. My question is, what is penalty on the CPU when mixing the audio on the fly? I know this is the better option, but I don't really need the 'in' and 'out' audio mixed until it's played back, and which happens less than 5% of the time. What are your thoughts? ___ --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] Hinting
I have had this working but not reliably. It seemed to work like this: Phone A watched B and C. Phone B watched A and C and Phone C watched A and B. I could see on Phone A (601) when phone B (501) was on the phone. Phone C never saw the status of either and Phone B would show the status of C. B would never show the status of A and A would never show the proper status of C. Phone C never showed the proper status of A or B. I didn't spend any more time on it but I'll try and get a chance to day to set the phones back up and give it a little more scientific testing. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Monday, April 03, 2006 10:32 AM To: Asterisk Users List Subject: [Asterisk-Users] Hinting Of the people in here that have hinting working with the polycom 601's (or any phone for that matter)... do you have it working so that the shared line appearance shows that there's someone on the phone? If so, any hints on how to do it? -- Aaron Daniel Computer Systems Technician Sam Houston State University [EMAIL PROTECTED] (936) 294-4198 ___ --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] Quintum Tenor DX4060
You have to use H323 the last time I did anything with their equipment. It has been almost a year but I think it went fairly smoothly. Do you have a specific question? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen Arulraj Sent: Friday, March 31, 2006 5:48 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Quintum Tenor DX4060 Has anyone linked the Asterisk to the Quintum Tenor DX4060? If yes I would appreciate any valuable information to do this in anyway. Cheers Stephen ___ --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 and Hylafax, on the same box
I agree we have this working also. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Boris Bakchiev Sent: Friday, March 31, 2006 8:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk and Hylafax, on the same box That's not entirely correct :) Fax and voice on the same DID is not possible when using a second application like hylafax. Because how should the two applications decide which one accepts the call? With the help of iaxmodem (which works really well) its easily done! Just detect the incoming call is fax and the route it to iaxmodem on fax extension. ___ --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] Dumb question - reaching the PSTN
This is not a dumb question. Most of the other replies I have read mentioned various ways to connect to the pstn. I wanted to mention why it makes sense to do that. Many of the companies I have installed asterisk for didn't even have their system on a network with a gateway. They have dedicated networks built for the phones and the Asterisk server acts as a dhcp, ntp and ftp server as well as the PBX. The only devices on the networks were phones. They use it as a really nice phone system and use old fashioned termination. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Charles Marcus Sent: Wednesday, March 29, 2006 5:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Dumb question - reaching the PSTN Hi everyone, I am fairly new to the idea of VoIP, although I've been reading about it off and on for the last few years. Now it is starting to look mature enough to consider implementing it, but there is one thing that I haven't been able to get a clear answer on... With Vonage, you are using the Vonage network - it is their responsibility to route your call to the endpoint, which is more than likely on the old fashined PSTN. If I install Asterisk, how do my calls actually get completed? How do they get 'bridged' over to the PSTN? I attended a Seminar today hosted by Dynasis, and one of the issues was VoIP. ShoreTel was there, and the said I had to have phone lines, whether they were POTS lines, chennels from a T-1, whatever, we still had to have phone lines. Now I'm confused. If I implement an Asterisk based system (yes, I'd be paying a consultant to help), will I still have to maintain phone lines and pay full price for Long Distance? Simple pointers to White Papers on this issue will be sufficient. Many thanks, -- Best regards, Charles ___ --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] Avoiding initial deadlock on iax?
I think there is a bug related to this. I haven't been able to track it down or really recreate it with any certainty yet. When I do I'll post something to Mantis. If you have any info to share with me about your situation when this occurs let me know. I have noticed that I can get it to occur if I suddenly power down one of the 4 asterisk servers that peer with my primary server via IAX. The main server will do this and then the other servers will sometimes do this also. They are setup in a Spoke and Hub type setup. If I take one of the spokes down suddenly about three out of five times the main box will do this and about one of two times one or more of the other spokes will do this. It happens at other times also when there are no boxes down. I have not been able to recreate it in that case. For several weeks, we had a problem with one-way audio coming off Level3's network and whenever we got a call from them that had one way audio and we redirected that call in and then back out to another server (on the weekends) the box would do this. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of stevanus Sent: Wednesday, March 29, 2006 4:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Avoiding initial deadlock on iax? Hi, My asterisk sometimes stop responding to iax calls. In the log, I've found this: Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[13002] chan_sip.c: update_call_counter(1409) - decrement call limit counter Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 WARNING[8386] channel.c: Avoided initial deadlock for '0x81d9530', 10 retries! It happens unpredictably and all I can do just killall -9 asterisk :S. When I execute iax2 show channels on CLI, I got messages that indicate many iax channel hung and I cannot do soft hangup to them :(. Here is my iax.conf: [general] bindport = 4569 ; Port to bind to (IAX is 4569) bindaddr = 0.0.0.0; Address to bind to (all addresses on machine) tos=0x68 ; bandwidth=low jitterbuffer=yes dropcount=2 disallow=all allow=ilbc ;allow=g723.1 ;allow=g729 ;allow=ulaw ;allow=alaw ;allow=gsm mailboxdetail=yes the other settings on iax.conf are just iax2 account for trunk and personal use. So I cut them in order to save spaces... Perhaps it's a bug? I've found this http://bugs.digium.com/view.php?id=4045 , but from the link I read that it is just for H323 not for iax. Will that patch cure my asterisk problem since the symptom are the same? Anyone has any ideas? Thanks Regards, Stevanus ___ --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] registration with different username
I have found this to be true also. [whatever] has to match username= It appears that it ignores the username field for IAX users. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Tomas Komarek Sent: Monday, March 27, 2006 8:43 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] registration with different username Well, I did, but the reason is still the same, if the username is different from the phone number, asterisk rejects the registration :-( Dovid Bender napsal(a): --- Tomas Komarek [EMAIL PROTECTED] wrote: Hello, I am trying to register to the asterisk with different phone number, login and password. This is my setting in the sip.conf: [246079011] type=friend context=cisco secret=XXX host=dynamic username=tomas allow=alaw nat=yes canreinvite=no mailbox=246079011 but I get this reply: Mar 27 13:17:00 NOTICE[5144]: chan_sip.c:10889 handle_request_register: Registration from '246079011sip:[EMAIL PROTECTED]' failed for '195.122.204.149' - Username/auth name mismatch Double check the user id and pass. Seems that asterisk is rejecting for that reason. __ 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 ___ --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] Polycom IP 301 is slow
I haven't read every message in this thread so I apologize if this is a repeat. Have you considered using the cfg files and an ftp server to configure the phones? I have found it to be very convenient as a way to manage many phones spread out across several locations as well as maintaining one or two phones. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Hoffman Sent: Saturday, March 25, 2006 3:06 AM To: asterisk-users Mailing List Subject: [Asterisk-Users] Polycom IP 301 is slow Hi guys, I've been using a Polycom IP 301 for a couple of weeks now and find that it's extremely slow for configuring. For instance, it takes several minutes to boot up, apply any changes via the web interface takes at least a minute, etc. Is this normal behaviour? Is there anything that can be done about it? Thanks, -- Nick e: [EMAIL PROTECTED] p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. ___ --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] Multiple commands per priority
Do you want to dial an outgoing line as well as the SIP line?
Dial(SIP/${OUTGOING}/${EXTEN}) ?
I can't say obviously without more info but it sounds to me like you are
looking for the wrong solution
-Jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Jason Frisch
Sent: Tuesday, March 21, 2006 11:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Multiple commands per priority
Hi everybody.
I have been searching and trying for an answer, but no luck, so here I
go..
Is there anyway to execute multiple commands on a single priority in
extensions.conf?
eg:
exten = X.,1,Dial(SIP/) somefunction(${EXTEN})
I need the dial command to dial internal extensions, and the
somefunction to
kick of our own outgoing system for redirection to outside lines; it
has
to go through our system for billing purposes.
Hope someone can help.
Regards,
Jason
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RE: [Asterisk-Users] Re: OT: Unblocking bloced CID
It's a toll free number. You can call it from anywhere and the costs of the call go on the callee not the caller. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of C F Sent: Wednesday, March 22, 2006 7:50 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: OT: Unblocking bloced CID It's a type of shoe you can get at any Macys On 3/22/06, Tomislav Parčina [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 2. If you receive it because you have an 800 number, you are not allowed to use it for anything else (read marketing) but billing. Can you please tell me what is 800 number? -- Tomislav Parcina tparcina#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] FAX over PRI
We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Michael Gaudette Sent: Tuesday, March 21, 2006 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] FAX over PRI Hmmm, Im not so sure I can apply this to me though. I just want to do Fax-To-Email using PRI channels as the incoming lines. Not so much transfer to a real fax. I am assuming that this is easily done with Asterisk? (I did it before with Asterisk SIP, but it only worked once every 10 tries or so) Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: March 21, 2006 3:25 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] FAX over PRI On Tuesday 21 March 2006 15:09, Michael Gaudette wrote: How should I consider Fax over PRI channels with Asterisk? Is the quality and reliability good, or should I be prepared for alot of grief? I'm having good success doing fax over PRI using a TE405; one span to the PRI, the other to an FXS channel bank that is almost obscenely underutilized (3 channels). I also have channel bank - T100P - IAX2 - TE405 - PRI, where the IAX2 link is a 1-hop SDSL (VOIP only) data link. This works well too. -A. ___ --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] Problem with intermittent one-way audio
I am having this problem also. I have 2 systems running 1.2.5. I had the problem and one system was running 1.2.4 and the other was running a CVS HEAD from October so I upgraded them both to 1.2.5 with no success. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Barry Flanagan Sent: Monday, March 20, 2006 2:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Problem with intermittent one-way audio Hi, I have a 1.2.4 asterisk box at a remote location, which is using IAX2 to connect to a 1.2.5 box for PSTN. There are 15 users on the remote server, all connecting via SIP softphones. For some reason, there is an increasing number of calls where the callee does not get any audio although the caller can hear them perfectly. This happens between 5% and 10% of the time. If they hang up and call again, it usually works. I have tried both with trunk=yes, and trunk=no but they are still having the problem. The debug log has a lot of the following, but not much else to go on. Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received VNAK: resending outstanding frames Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received VNAK: resending outstanding frames Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received VNAK: resending outstanding frames Mar 20 19:26:37 DEBUG[26754] chan_iax2.c: Received iseqno 18 not within window 19-19 Any help much appreciated. -- -Barry Flanagan ___ --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] OT: Unblocking bloced CID
Well for one thing, on a PRI it is usually still transmitted with a bit set that tells the system to hide it. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Thursday, March 16, 2006 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OT: Unblocking bloced CID If it is blocked how are you 'getting it' If it is ANI over an 800 number then you as the person paying for the call have the 'right' to use this information. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, March 16, 2006 9:39 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: Unblocking bloced CID Hello list, I know this has been brought up before but I dont think there was ever a final answer. Is it legal in the US to modify asterisk to show the CID information that was received as blocked ? Thanks. Dovid p.s. Sorry for the poor typing format, it was written from a mobile phone. __ 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 ___ --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] OT: Unblocking bloced CID
Well, whether I SHOULD get it or not may be totally irrelevant to whether I CAN or DO get it. The caller ID info is most definitely there and it shows up in my CDR records. However, it is not displayed on the device because only the number is allowed on our PRI. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Thursday, March 16, 2006 10:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] OT: Unblocking bloced CID On 3/16/06, Jonathan k. Creasy [EMAIL PROTECTED] wrote: Well for one thing, on a PRI it is usually still transmitted with a bit set that tells the system to hide it. I'm almost sure you are wrong. One shouldn't get it on a PRI either. Only on SS7, or toll free. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Thursday, March 16, 2006 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] OT: Unblocking bloced CID If it is blocked how are you 'getting it' If it is ANI over an 800 number then you as the person paying for the call have the 'right' to use this information. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, March 16, 2006 9:39 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OT: Unblocking bloced CID Hello list, I know this has been brought up before but I dont think there was ever a final answer. Is it legal in the US to modify asterisk to show the CID information that was received as blocked ? Thanks. Dovid p.s. Sorry for the poor typing format, it was written from a mobile phone. __ 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 ___ --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
[Asterisk-Users] Hung IAX Channels
I have a problem where my Asterisk server stops answering new TCP requests and begins to use 99.9% of the CPU on my box. The server is a 64bit Xeon with 2GB of ram. I haven't been able to recreate the problem but it occurs sometimes when there is a call coming from my provider (via IAX) to a customer (via IAX). The customer and the provider systems are running asterisk also. The call will end and the provider and the client's systems will show that it has ended (confirming with show channels). On my server however, it is still showing the active channels. This happened about once a week and since upgrading to 1.2.5 this morning from 1.2.4 it has happened twice. Does anyone know what is happening here? -Jonathan ___ --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: SOLVED: Re: [Asterisk-Users] Polycom IP501 with Asterisk -distinctive ring?
BOFH told me he uses it to listen to his co-workers -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Thursday, February 09, 2006 12:27 PM To: asterisk-users@lists.digium.com Subject: SOLVED: Re: [Asterisk-Users] Polycom IP501 with Asterisk - distinctive ring? On Thursday 09 February 2006 11:28, Kevin P. Fleming wrote: There are examples (IIRC) of making the phone auto-answer for specific types of calls; those should get you started, since they demonstrate how to have the phone choose a different 'alerting' configuration on a call-by-call basis. Yup I just found some possibilities (with some help from people on the Asterisk-Ontario list): http://www.voip-info.org/wiki-Polycom+auto-answer+config and after some more googling: http://lists.digium.com/pipermail/asterisk-users/2004- September/061116.html Now I'm just trying to see what else I can do with ALERT INFO and the IP501. :-) -A. ___ --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: SOLVED: Re: [Asterisk-Users] Polycom IP501 with Asterisk -distinctive ring?
This feature also works on the IP301 phones. The obvious caveat is that it is one-way only. Still nice for an all-page though. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Thursday, February 09, 2006 12:27 PM To: asterisk-users@lists.digium.com Subject: SOLVED: Re: [Asterisk-Users] Polycom IP501 with Asterisk - distinctive ring? On Thursday 09 February 2006 11:28, Kevin P. Fleming wrote: There are examples (IIRC) of making the phone auto-answer for specific types of calls; those should get you started, since they demonstrate how to have the phone choose a different 'alerting' configuration on a call-by-call basis. Yup I just found some possibilities (with some help from people on the Asterisk-Ontario list): http://www.voip-info.org/wiki-Polycom+auto-answer+config and after some more googling: http://lists.digium.com/pipermail/asterisk-users/2004- September/061116.html Now I'm just trying to see what else I can do with ALERT INFO and the IP501. :-) -A. ___ --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 can I configure to call from the consolebymeans of a sip phone,
It's something like exten = 15,1,Dial(Console/DSP) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Azzopardi Sent: Saturday, February 04, 2006 2:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] How can I configure to call from the consolebymeans of a sip phone, I can call from the console by means of the 'dial' command, now I need to know how to call the console itself. Anthony. ___ --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 on laptop connected to POTS line
Do people not use the Grandstream ATA's because they are cheap or because there is actually a problem with them? They have a 2 line version for around $50 that I have used in various locations. I have about 8 or so. They seem to do an excellent job. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Saturday, February 04, 2006 11:56 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS line Not a chance, they sell SPA3000's by the truckload. If you only need one line, then go with the SPA3000, if you need more, I would go with the Mediatrix 1204. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Saturday, February 04, 2006 8:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS line I thought they stopped selling the spa3000 ? --- Damon Estep [EMAIL PROTECTED] wrote: Sipura SPA-3000 will give you 1 fxs and 1 fxo so you can connect to a POTS line AND a analog phone at the same time with one small box. Makes a great demo system. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 02, 2006 6:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid ___ --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 __ 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 ___ --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] CallerID popup
I have been planning to do the same thing but never got around to it, I actually did write a nice class to wrap the interface to the manager but it isn't complete. Would you be willing to share your work? -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mimmus Sent: Friday, February 03, 2006 5:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] CallerID popup Hi, I'm trying to write a small Visual Basic app to throw a popup with CallerIDNum when a call center agent answers a queue call. Does anyone know what is the right manager event to intercept? Thanks Mimmus ___ --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 on laptop connected to POTS line
The Grandstream ATA (480 I think...) does this and usually costs less than the Sipura. It has 1 FXS and 1 FXO. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Thursday, February 02, 2006 8:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Asterisk on laptop connected to POTS line Sipura SPA-3000 will give you 1 fxs and 1 fxo so you can connect to a POTS line AND a analog phone at the same time with one small box. Makes a great demo system. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dovid Bender Sent: Thursday, February 02, 2006 6:20 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk on laptop connected to POTS line Anyone know of any equipment that I can use to connect a laptop running asterisk to a POTS line (RJ11) ? Regards, Dovid ___ --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] Dundi key Problem
I am getting the following message when trying to lookup up a number via Dundi: Feb 1 13:39:24 NOTICE[20146]: pbx_dundi.c:1309 update_key: No such key 'office.pbx.bluegrass.net.pub' for creating RSA encrypted shared key for '00:a0:c9:55:91:89'! I have created keys on each box with astgenkey -n office.pbx.bluegrass.net using the host name for each box of course. I then copied the .pub files to the /var/lib/asterisk/keys folder from each box to the other box. What am I missing? -Jonathan ___ --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] winnipeg canada
Anyone in Winnipeg Canada? ___ --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] Notifications when host fails qualify
I am looking to be notified via email when a host fails it's qualify (is unreachable). I found this patch (http://bugs.digium.com/view.php?id=5372) but I wasn't sure if I could get that from it. Anyone else tried this? -Jonathan ___ --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] Polycom IP301 time changing
I have 13 Polycom IP301's where the clock keeps resetting to a +5 offset. I can change the config file to show -5, change it to -5 on the phone and after an hour or so the phone will update itself back to +5. Anyone have any ideas? The other 70+ phones are not exhibiting this behavior. -Jonathan ___ --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] anybody getting No authority found with teliaxnow?
This is an authentication problem. Check the username, password, number and context being sent across to see if they are correct. Post your iax debug info for the call if you can. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Thomas Miller Sent: Thursday, December 22, 2005 8:58 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] anybody getting No authority found with teliaxnow? Everything was working great until last night. All calls since last night are getting No Authority Found message. I am using IAX2 Is anybody else having this problem? Thx, Tom __ 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 ___ --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 get received digits from console channel
I am going to step out on a limb and guess that you need to hangup the call when the digits are received which would move you on to the next priority where you could then enter your loop. This is an autoattendant I made at some point when I was playing around with how to do it. Maybe it will help you get what you want done. [funstuff] Exten = s,1,Answer Exten = s,2,Playback(you-are-caller-num) Exten = s,3,SayNumber(9233,c) Exten = s,4,DigitTimeout,25 Exten = s,5,ResponseTimeout,10 Exten = s,6,Background(press) Exten = s,7,Background(digits/1) Exten = s,8,Background(if-maint-contract-or-emergency) Exten = s,9,Background(infuriate-tech-staff) Exten = s,10,Background(digits/2) Exten = s,11,Background(hear-odd-noise) Exten = s,12,Background(digits/3) exten = t,1,goto(s,4) exten = i,1,Hangup Exten = 2,1,Playback(why-no-answer-mystery) Exten = 2,2,Playback(lines-complaining-customers) Exten = 2,3,Playback(for-quality-purposes) Exten = 2,4,Playback(gambling-drunk) Exten = 2,5,Goto(s,4) Exten = 3,1,Playback(you-sound-cute) Exten = 3,2,Playback(what-are-you-wearing) Exten = 3,3,Goto(s,4) Exten = 6,1,Playback(because-paranoid) Exten = 6,2,Playback(all-your-base) Exten = 6,3,Playback(hang-on-a-second-angry) exten = 6,4,Goto(s,4) Exten = 4,1,Playback(asterisk-friend) Exten = 4,2,Playback(computer-friend1) Exten = 4,3,Goto(s,4) Exten = 5,1,Playback(step-in-stream) Exten = 5,2,Goto(s,4) Exten = 1,1,Playback(go-away2) Exten = 1,2,Goto(s,4) Exten = 7,1,Playback(i-dont-understand3) Exten = 7,2,Goto(s,4) Exten = 8,1,Playback(could-lose-a-few-pounds) Exten = 8,2,Goto(s,4) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Phuong Nguyen Sent: Tuesday, December 20, 2005 8:06 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to get received digits from console channel Importance: High Hi, I need to develop a project in which the user can phone a number, say something and the voice will be output to a speaker, if the user want to select other actions, he could just press a number on the keypad, e.g.: press 1. I did it with the following: 1. make a incoming context, looks like: [incoming] exten = s,1,Answer() exten = s,2,Background(/var/lib/asterisk/tgsounds/greeting) exten = 1,1,Dial(console/dsp,10,G(loop^s^1)) [loop] exten = s,1,Background(/var/lib/asterisk/tgsounds/waiting) exten = s,2,Goto(loop,s,1) exten = 2,1,Goto (othercontext,s,1) After the Dial(console/dsp) command, the user speak and his voice is output to the speaker as desired. However, it seems that the channel (Zap/4) with the connected telephone line, did not go to the loop context. Therefore, all the keypad input from the calling phone are just printed out to the console like Console received digits 2. What I actually want to do is that; when the user press 2, I will shutdown the console (so if the user speak to the phone, his sound will not output to the speaker) and just go to another context (say: playing a music file). The input are received by Asterisk as shown above, but I don't know how to get this digits and redirect the Zap channel to other context (at that moment is still connected with the console ). Thanks in advance for any suggestion. Regards, Phil -- 10 GB Mailbox, 100 FreeSMS/Monat http://www.gmx.net/de/go/topmail +++ GMX - die erste Adresse für Mail, Message, More +++ ___ --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] IAX error message
What causes this? Dec 13 15:16:06 NOTICE[2660]: chan_iax2.c:1561 iax2_destroy: Avoiding IAX destroy deadlock Something occurs and I get a flood of these then the box quits taking calls and asterisk wont die. -Jonathan The contents of this email message and any attachments are confidential and are intended solely for addressee. The information may also be legally privileged. This transmission is sent in trust, for the sole purpose of delivery to the intended recipient. If you have received this transmission in error, any use, reproduction or dissemination of this transmission is strictly prohibited. If you are not the intended recipient, please immediately notify the sender by reply email and delete this message and its attachments, if any. ___ --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 Dial Failover
I chose this method and have been happy with the results. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 09, 2005 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Dial Failover Your other option is to setup the OpenSER boxes in a truly redundant configuration using Linux HA (www.linux-ha.org). That way you setup all your PSTN calls to forward to one shared virtual IP between the boxes. One of the boxes is the Master, the other is the Slave. There is a heartbeat between the boxes that goes at a configurable rate. If the Master fails then the Slave will take over and it can even be configured for sub-second failover. I think there is a article on voip-info.org about this, but don't have time to look it up. Good luck and let us know what you choose to do. Ryan All, I have an Asterisk system that sends PSTN calls to an OpenSER system to be routed. I have a command like this in my extensions.conf: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) There's actually two OpenSER systems for redundancy. I'm trying to find a way to have Asterisk attempt to route the call to one OpenSER system, and if it's down, fallback to another. Any first thoughts on how to achieve this? I can't have Asterisk do a DNS SRV lookup because Asterisks SRV lookups are broken. If I issue a series of Dial commands, such as this: exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr) ... what seems to happen is that when proxy1 is down, Asterisk waits the full 20 seconds before returning control. Also, This 20s includes the time is takes for the other end to answer, so if I put a small value of say 5s in there, the dial command will probably give up before someone answers at the other end. Neither is workable. Asterisk SHOULD be able to distinguish between a TRYING and no response. In the event it gets no TRYING response to a dial command within a specified timeout it should return control and flag an error. If on the other hand it does get a TRYING response (and maybe a RINGING too) it should continue to wait until the 20s has expired. I can't use dynamic DNS (ie putting two A records for a hostname in DNS) because Asterisk reads the extensions.conf on startup and also seems to cache what the host maps to on startup. Subsequent calls to the host always return the same IP address. But... in general... how have people implemented this? Help appreciated! Doug ___ --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] video phones
Anyone using any H.263+ video phones and want to relay their experiences? -Jonathan ___ --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] harry's project
http://www.automated.it/guidetoasterisk.htm I don't think you even require SER in that case. That will be $100. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac Sent: Thursday, November 24, 2005 7:11 PM To: users@openser.org; asterisk-users@lists.digium.com Subject: [Asterisk-Users] harry's project Hello, here is an other diagram for people who don't yet understand what i expect to do. Look at sip_call_flow.png file i wish to substitute ondo sip server with ser and ondo pbx with asterisk . ondo sip server is able to do far-end near-end nat I guess ser too. I do hope i will find some people who help me to configure that . Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk 1.0/1.2 on cobalt Raq2-4
I've thought about doing that as I have a few spare also. I would use the raq4 I think. Let me know if you have any trouble with it. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of bram kortleven Sent: Monday, November 14, 2005 5:14 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk 1.0/1.2 on cobalt Raq2-4 Anyone ever tried to install on a Cobalt Raq device? This is a 1U 19 rack computer, the Raq2 using a Mips processor, the Raq4 using a K6-2/3 processor. As I do have a few of these as spares, I was wondering if I could use them as my pbx system, because of their low power-system and dence system box. I simply need the pbx to serve 2 phones in my appartment, a SIP- connection for 4 external internet devices (my brother, living in the USA, my parents, living a few miles from here, and my nefew living in France and his mother, living here in Belgium too) Has anyone done this setup on a Raq2? Or do I need to use the extra power of the Raq4 (faster cpu and mem, bigger faster disk, ...) Anyone having a pkg-installer for the raq devices, as they are used for updates etc...? Thanks Bram ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Mission-Critical Deployments
I'm just throwing this out here, not dissing anyone. Someone asking these types of questions may want to seek some professional consultancy with regards to the network before building a mission critical deployment. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson Sent: Sunday, November 20, 2005 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Mission-Critical Deployments 2. What do we need to do for our data network to make VOIP reliable? QoS, basic traffic prioritization on the switch, vlan, ??? If you are not running a bandwidth hungy network, then you might be able to work with just one vlan, if you don't want to take the chance, then yes you need: QOS, and VLANs. Vlan's are certainly not a tool used to improve performance, and in many cases, will cause more issues then what they solve. Part of the reason for that is that there isn't any realistic way to manage switch-to-switch trunks when multiple vlans traverse it. Eg, If the offered instantanous traffic is greater then the port speed, packets will be dropped. Identifying the root-cause of such issues is no where near as easy as one might believe. QoS on a switch will have zero impact _unless_ the offered traffic is greater then the port speed, or, port speed differences (eg, traffic from 100 meg ports heading outbound on a 10 meg port). Not likely to be the case in environments outside larger corporate networks. Can you elaborate a bit on that? I've never used VLANS, nor QoS on the switch level. Do we really get more reliability by using both, or would QoS alone be enough? QoS would be enough if your existing traffic is congested. Congestion can be seen in the form of dropped packets on individual switch ports. If you're not dropping any packets, then QoS will not do any good at all on the switch. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Provisioning server
For which equipment? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of xcel Sent: Friday, November 18, 2005 11:53 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Provisioning server Can any one help me in setting up Provisioning sever ?? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Context restrictions for long distance access, examples not clear?
What context are your phones in? (context= in sip or iax config) If your phones are in the local-users context, they will be able to dial numbers found in local-users, extensions and local. If your phones are in the long-users context, they will be able to dial numbers in long-users, local, long-distance and extensions. Extensions in a context are handled in the order they are listed. In this case, I would remove the entries which are also in extensions from the local-users and long-users extensions. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Friday, November 18, 2005 12:00 PM To: Asterisk - Users Subject: [Asterisk-Users] Context restrictions for long distance access,examples not clear? Hi, I am trying to limit access to long distance in my dial plan but I am really confused by the examples I am seeing (perhaps I am misunderstanding how context work). The following example was given in a previous posting. [extensions] exten = 8478414198,1,Dial(SIP/8478414198) exten = 8478414198,2,Hangup exten = 8478414199,1,Dial(SIP/8478414199) exten = 8478414199,2,Hangup [local] exten = _XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _XX,2,Congestion [long-distance] exten = _1NXXNXX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _1NXXNXX,2,Congestion [local-users] exten = 8478414198,1,Dial(SIP/8478414198) exten = 8478414198,2,Hangup include = local include = extensions [long-users] exten = 8478414199,1,Dial(SIP/8478414199) exten = 8478414199,2,Hangup include = local include = long-distance include = extensions What I do not understand is how this restricts access. Since the context 'extensions' is included in both would that not give all users access to local and long distance??? Or is there some sort of order of entry thing with context??? I supposed that zapata.conf would include a reference to extensions - that would be the only reason for having the extension context... Also since the extensions appear under local-users and long-users followed by the include 'extensions' wouldn't this generate an error since the extension already exist (ie in local users has the extension 8478414198 with a priority of 1 and the include statement means that another extension 8478414198 with a priority of 1 in the same context 'local-users') Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Provisioning server
Im not familiar with provisioning on those, sorry. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of xcel Sent: Friday, November 18, 2005 1:02 PM To: asterisk-users@lists.digium.com; asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Provisioning server initially for Sipura SPA 1001. Regards, *** REPLY SEPARATOR *** On 11/18/2005 at 12:44 PM Jonathan k. Creasy wrote: For which equipment? From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of xcel Sent: Friday, November 18, 2005 11:53 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Provisioning server Can any one help me in setting up Provisioning sever ?? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dialing out with FXO
FXO ports are an interface between your system and a phone carrier. FXS
ports are an interface between your system and a phone station (or
handset).
You can send outbound calls on an FXO port as well as receive them.
To dial on the TDM card you would dial : DIAL(Zap/g1/${number}) or
Dial(Zap/1/${number} depending on how you have configured your card.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Dulmandakh Sukhbaatar
Sent: Wednesday, November 16, 2005 3:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dialing out with FXO
[EMAIL PROTECTED] wrote:
FXO ports are for dialling outunless you use them for dialling
in.
PaulH
- Original Message -
From: Dulmandakh Sukhbaatar [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 16, 2005 5:57 PM
Subject: [Asterisk-Users] Dialing out with FXO
I got TDM card with 4 FXO ports. But I need to dial out? How I can
do
this? Is it possible? Please help
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FXO seems only to receive voice call? Isn't it?
http://www.digium.com/index.php?menu=fxsvfxo shows this. I have no FXS
card.
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[Asterisk-Users] ATT Merlin Communications System 6102 Cartridge Music on Hold and Paging
I am trying to replace the overhead paging function of an old phone system. There is a device with an RJ11 connection connected to two screws on the phone system. The two screws are on a cartridge labeled as the subject of this message. I thought the other device was probably a station andthat I could plug it into an ATA and send it a call to come out on the speakers. I was wrong...anyone know anything about this and how that box might be wired? More importantly, does anyone know how to make it work with an asterisk system? -Jonathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] receive fax with asterisk
I can't seem to compile IAXmodem. sh build: iaxmodem.c: In function `cleanup': iaxmodem.c iaxmodem-cfg.ttyIAX lib README termpkg-ttydforfax.patch TODO iaxmodem.c:90: error: too many arguments to function `iax_register' iaxmodem.c: In function `main': iaxmodem.c:705: error: `IAX_EVENT_CNG' undeclared (first use in this function) iaxmodem.c:705: error: (Each undeclared identifier is reported only once iaxmodem.c:705: error: for each function it appears in.) iaxmodem.c:754: error: too many arguments to function `iax_register' I am using the libiax2 that I just got out of CVS with cvs checkout libiax2 -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lee Howard Sent: Wednesday, November 16, 2005 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] receive fax with asterisk Anton Krall wrote: Sounds like a good setup. Will this replace spandsp or how does this setup integrate with that? IAXmodem uses spandsp (the library) but does not use txfax/rxfax. I suppose it works on sending and receveing right? Yes. Lee. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Message waiting notification
mailbox= in the sip.conf From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sixto Diaz Sent: Tuesday, November 15, 2005 9:33 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Message waiting notification Hi i want to notify a user that he has an unreadvoicemail waiting to be read. How can i do this for sip users? THanks Sixto ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Message waiting notification
On my phones (polycoms) its an option in the configuration to change the tone, etc. -Jonathan From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sixto Diaz Sent: Tuesday, November 15, 2005 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Message waiting notification i want to ring the phone user or change the tone is this posible with mailbox= ? - Original Message - From: Jonathan k. Creasy To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, November 15, 2005 11:58 AM Subject: RE: [Asterisk-Users] Message waiting notification mailbox= in the sip.conf From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sixto Diaz Sent: Tuesday, November 15, 2005 9:33 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Message waiting notification Hi i want to notify a user that he has an unreadvoicemail waiting to be read. How can i do this for sip users? THanks Sixto ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Editing Asterisk config files with WORD Pad
Use notepad if you must edit them on a windows box. Nano/Pico/Joe are pretty user friendly editors for the *nix environment. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Chuck Bunn Sent: Tuesday, November 15, 2005 11:19 AM To: Asterisk - Users Subject: [Asterisk-Users] Editing Asterisk config files with WORD Pad Hi, I have tried editing some Asterisk config files (ie sip.conf) in MS Word Pad and I have saved the files as 'Unicode Text Document' with quotes around the full file name = sip.conf and then uploaded the files to a Linux server using FileZilla. When I do this the config files fail to work. Although I am somewhat proficient at using 'vi' I find it easier to cut and paste with Word Pad (I know I can cut and paste with 'vi'... ). Is this possible to do or am I all wet... Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Planet Network - VIP-153
Anyone used a sip from from Planet Network? VIP-153 http://www.planetnw.com/ http://www.planetstoresite.com/Merchant2/merchant.mvc?Screen=PRODStore_Code=PNIProduct_Code=VIP-152TCategory_Code=VOIP ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hiss
Is the ambient noise in the room high? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Tuesday, November 08, 2005 8:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Hiss Paul wrote: I get the hiss and noise with softphones using all headsets I have tried so far. I don't get it with grandstream budgetone 101 phones or phones connected to ata's. Then it's likely to be your sound card. Try using a nice usb headset (not the cheapest you can find) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension Ring on Multiple Phones
Title: Extension Ring on Multiple Phones EXTEN= 100,1,DIAL(SIP/ONESIP/TWOSIP/THREE) From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Morrow Sent: Tuesday, November 08, 2005 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Extension Ring on Multiple Phones Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] http://www.autodata.net Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension Ring on Multiple Phones
I guess I should have read up further before I posted a response. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mojo with Horan Company, LLC Sent: Tuesday, November 08, 2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Ring on Multiple Phones Like instead of exten = s,1,Dial(SIP/110,20,tr) you must mean exten = s,1,Dial(SIP/110SIP/112,20,tr) ? Just append all extensions you wish to ring, separated by ampersands (). The first one to answer will be winner. That's what I think you're asking, at least. Moj Dave Morrow wrote: Hi all. I wonder if anyone out there has a dial-plan which will ring an extension on multiple phones. David A. Morrow Technical Systems Lead Autodata Solutions Company [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] _http://www.autodata.net_ Tel: (519) 951-6079 Fax: (519) 451-6615 Poor planning on your part does not necessarily constitute an emergency on my part! This message has originated from Autodata Solutions. The attached material is the Confidential and Proprietary Information of Autodata Solutions. This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. If you have received this email in error please delete this message and notify the Autodata system administrator at_ [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]_ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Cisco 7970
I thought there was a sip image for that phone? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Reynolds Sent: Tuesday, November 08, 2005 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: Cisco 7970 Jeremiah, You say you have your 7970 working great with * ... The 7970 only supports SCCP, so are you using the chan_skinny modules that come with *, or are you using the chan_sccp modules? Thanks for any response. JR On 11/8/05, Jeremiah Millay [EMAIL PROTECTED] wrote: I ran into this same problem the other day. What you need to do is put all firmware files in the tftp root directory. The trick with the files is you need to match the case of the filename that the phone is looking for. My XmlDefault.cnf.xml needed to have the proper case. If you do a tcpdump on your server you can see what file its getting stuck on. This is how I figured out what it is looking for: tcpdump -i eth1 port tftp -vv It will output what file the phone is looking for. Have my 7970 working great with *. Hope this helps. Jeremiah On Nov 7, 2005, at 10:24 AM, [EMAIL PROTECTED] wrote: Hello I have a Cisco 7970 phone that when I was trying to reset it to factory defaults it rebooted and now is stuck in a constant loop of the lights flashing by going down the line pool one light at a time in a constant rotation. I have the firmware for the phone, but have no idea on how to load or it how to get this phone functioning again. I would definitely be willing to pay someone to help me get this thing back online, if someone can contact me either here or offlist to get this resolved I would appreciate it tremendously. Thanks Dan - Dan Levine [EMAIL PROTECTED] 877.CYTEXONE x 810 212.477.0990 x 810 212.208.6889 FAX 502 Laguardia Place, Suite 2B New York, NY 10012 http://www.cytexone.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: [Asterisk-biz] Asterisk as a VoiceConferenceServer
Any IRQ or duplex problem with your NIC? Any collisions or errors? I have had similar results to others here in that conferences with 50-100 users are just fine even on fairly outdated hardware. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, November 02, 2005 5:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: [Asterisk-biz] Asterisk as a VoiceConferenceServer SIP On 11/2/05, Tom Hayden [EMAIL PROTECTED] wrote: Can I ask what kind of trunking you are using for the calls? Zap/SIP/IAX? -- Tom On 11/2/05, Cullin J. Wible [EMAIL PROTECTED] wrote: We used to run a conference server on a PII 400Mhz with 512MB of RAM. We had 2 separate conference rooms with 15 users each (30 simultaneous) calls with no problem. We have since upgraded it to a P4 2.0Ghz with 1GB of RAM (just because it was getting old) and it still works just fine with even higher call volumes. No degradation of quality either that we can see. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kanuri, Seshu (Company IT) Sent: Wednesday, November 02, 2005 2:51 PM To: Iain Barker Cc: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] RE: [Asterisk-biz] Asterisk as a Voice ConferenceServer Iain Barker Wrote: - Our experience with over 10 or more participants in a single Asterisk conference was that quality degraded quite rapidly. Is this really true as there were many in this list who had confirmed that they have used the conference bridge for a lot more connections than what you have Suggested as the upper limit. Logically the conference bridge should work at the same capacity as the number of calls Asterisk can handle in a given configuration. Though your solution looks impressive and probably is the best for upto 30 simultaneous calls, I am more interested in knowing what it takes for Asterisk to be able to handle the 100 channels I need to run Simultaneously. Seshu Kanuri -Original Message- From: Iain Barker [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 02, 2005 1:41 PM To: Kanuri, Seshu (Company IT) Subject: Re: [Asterisk-biz] Asterisk as a Voice Conference Server Seshu, Our experience with over 10 or more participants in a single Asterisk conference was that quality degraded quite rapidly. The solution was a dedicated hardware bridge for conference mixing http://www.aastra.com/enterpriseip/pro_238.asp Kanuri, Seshu (Company IT) wrote: I am working on a bid for a New York State requirement where we need to provide access to 100 Simulataneous Investors to get into a conference with the Pension Funds Officer for discussions. As you might have guessed it, I am presenting an Asterisk enabled Conference solution. One of the Bid requirement is to provide three verifiable references who have implemented a similar voice conference solution for more or less 100 simultaneous calls, with a possible recording of the entire call. If anyone has implemented this on a commercial scale, I am looking for referrals at this time, and a possible co-operation in future. I would appreciate if you can send me your name, contact Info, company name and a one para description of the solution and the name/type of client whom/where this solution is running at this time. A couple of minutes of your time is needed when the guys at Albany may like to speak to you for a confirmation that Asterisk is real and it can do the 100 people conference, what they are looking for. I do thousands of conferences a day using asterisk as the backend, most are in the 5-50 user range, but many are in the 150+ range. (but, I use app_conference, not app_meetme for them). I can give you my contact information off-list if you want it. -SteveK NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To
[Asterisk-Users] shared lines
Has anyone figured out how to make the shared line appearance thing work with asterisk? From wiki: http://www.voip-info.org/wiki/view/Polycom+Phones Supports shared lines (but asterisk does not) - Anyone having details on the specifications used for Shared Call / Bridged Line Appearances (SIP-B), Please post details!! -Jonathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Shared Lines
Can a Polycom IP601 with the addon modules be setup to work like an attendant console showing the status of other lines? How does that sort of thing work with Asterisk? -Jonathan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] anyone using these?
Voicetronix OpenSwitch6 http://www.telephonyware.com/telephonyware/tw3.html ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Geneys
Anyone using the Genesys framework with an Asterisk PBX? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip not working suddenly
Anyone know what's causing this: -- SIP read from x.x.x.x:56800: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKba5c00f818387D67 From: Xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B To: sip:[EMAIL PROTECTED];user=phone;tag=as57402fc2 CSeq: 1 ACK Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' lou01*CLI -- SIP read from x.x.x.x:56800: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0 From: Xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B To: sip:[EMAIL PROTECTED];user=phone CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username=user1, realm=asterisk, nonce=07b9f9a3, uri=sip:[EMAIL PROTECTED]:5060;user=phone, response=a8f005540682f07a88e023d50135cce0, algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 253 v=0 o=- 1130439113 1130439113 IN IP4 192.168.200.16 s=Polycom IP Phone c=IN IP4 192.168.200.16 t=0 0 a=sendrecv m=audio 2228 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Reliably Transmitting (NAT) to x.x.x.x:56800: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0;received=x.x.x.x;rport=568 00 From: Xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B To: sip:[EMAIL PROTECTED];user=phone;tag=as71adaedb Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=56bff437 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from x.x.x.x:56800: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0 From: xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B To: sip:[EMAIL PROTECTED];user=phone CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username=user1, realm=asterisk, nonce=07b9f9a3, uri=sip:[EMAIL PROTECTED]:5060;user=phone, response=a8f005540682f07a88e023d50135cce0, algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 253 v=0 o=- 1130439113 1130439113 IN IP4 192.168.200.16 s=Polycom IP Phone c=IN IP4 192.168.200.16 t=0 0 a=sendrecv m=audio 2228 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines)--- Ignoring this INVITE request Transmitting (NAT) to x.x.x.x:56800: SIP/2.0 488 Not Acceptable Here (codec error) Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0;received=x.x.x.x;rport=568 00 From: xxx xxx sip:[EMAIL PROTECTED];tag=69375B3E-ACF6C78B To: sip:[EMAIL PROTECTED];user=phone;tag=as71adaedb Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] user name
I dont get it. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Richmond Sent: Thursday, October 20, 2005 9:46 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] user name I am geting e-mail but asterisk doesn't know my user name or password. My user name has always Been Jerry Richmond, my e-mail address [EMAIL PROTECTED] I need a password of some kind. thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom IP501 and record on demand
I probably can't provide any better information for you, however, have you looked through the Polycom configuration files. The button mappings are there. I haven't spent much time with it so I can not attest to what you can map them to do. Hope this helps you a little. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew T. O'Connor Sent: Wednesday, October 19, 2005 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom IP501 and record on demand Matt Gibson wrote: You could also take a look at features.conf, and use ** for blind transfers, ## for attended transfers, *0 for recording, and *1 to hangup. I haven't tried mapping them to polycom buttons, but there was recently a discussion about that, just this week you can search the archives. There was a discussion (of which I was a part of) however there was no resolution. I have not found any good documentation on how to remap Polycom buttons. At this point I'm willing to pay for some help. Anybody got some better info on this? Thanks, Matthew O'Connor ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] One phone ringing, one phone flashing ?
You can do it with a Polycom (and probably a Cisco) by setting an Alert var and it will handle the call using a defined class. Search for paging. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jason Pyeron Sent: Tuesday, October 18, 2005 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] One phone ringing, one phone flashing ? Working on it right now, its called shared line appearance, and it is a feature of both. If anyone has patches, ideas and comments, here is your chance to get some work and not get billed for it. On Tue, 18 Oct 2005, Stefan-Michael. Guenther (in-put GbR) wrote: Hi, well, some clients have strange ideas and wishes (at least to my mind). Yesterday I gave a presentation about asterisk to a CEO. At the end he asked me whether asterisk is able to do the following: When a call for the CEO comes in, the calling number should be shown on the display of his phone and the phone of his secretary. The secretary's phones should ring, but at his phone only a light should flash. ;-)) No, turning off the sound isn't the solution. This restriction should e.g. only apply, when it is an external call, internal calls should result in ringing both phones. I'm not quite sure, whether this could be a feature of asterisk or the phone or both together. Does anything of you successfully set up something like this or could recommend a phone that would help/support it? Thanks a lot in advance, Stefan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- - - - Jason Pyeron PD Inc. http://www.pdinc.us - - Partner Sr. Manager 7 West 24th Street #100 - - +1 (443) 921-0381 Baltimore, Maryland 21218 - - - -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- This message is for the designated recipient only and may contain privileged, proprietary, or otherwise private information. If you have received it in error, purge the message from your system and notify the sender immediately. Any other use of the email by you is prohibited. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Audio from Console but mpg123fromshellworksfine.
Thanks. I was only loading OSS. I installed the alsa development libraries and then loaded alsa instead of oss and everything is working now. Thanks! -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Sunday, October 16, 2005 9:00 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] No Audio from Console but mpg123fromshellworksfine. On 10/16/2005, Jonathan k. Creasy [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Sunday, October 16, 2005 2:59 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] No Audio from Console but mpg123 from shellworksfine. Do you use ALSA or OSS for sound? What kernel version? ALSA. I used alsactl to reset the mixer controls as it was muted by default. I'm running CentOS 4.1, I don't remember the kernel version right off and I don't have access to that box here, I'll check it from work tomorrow. [chan_oss.so] = (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found == Registered channel type 'Console' (OSS Console Channel Driver) Asterisk grabs /dev/dsp . I figure you can't play anything at this point. Though you should get stuck at trying to open it. Sigh... From modules.conf ; ; Load either OSS or ALSA, not both ; By default, load OSS only (automatically) and do not load ALSA ; noload = chan_alsa.so ;noload = chan_oss.so Bet the problem is around here. Brett ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Audio from Console but mpg123 from shellworksfine.
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Sunday, October 16, 2005 2:59 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] No Audio from Console but mpg123 from shellworksfine. One possibility is that the volume is set to 0. aumix can be handy here. Does asterisk have a volume control? Like I said Asterisk is the only thing not playing. Try simpler things, like playing wav files with 'play' of sox. Asterisk won't play anything to the cli console but anything else on the box can play fine. Do you use ALSA or OSS for sound? What kernel version? ALSA. I used alsactl to reset the mixer controls as it was muted by default. I'm running CentOS 4.1, I don't remember the kernel version right off and I don't have access to that box here, I'll check it from work tomorrow. [chan_oss.so] = (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found == Registered channel type 'Console' (OSS Console Channel Driver) Asterisk grabs /dev/dsp . I figure you can't play anything at this point. Though you should get stuck at trying to open it. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] No Audio from Console but mpg123 from shell worksfine.
Anyone have anything on this? (I'm sure someone will complain about me bringing it up again, chill out...) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jonathan k. Creasy Sent: Friday, October 14, 2005 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] No Audio from Console but mpg123 from shell worksfine. I get audio from mpg123 at the command line but when I load up asterisk and try to get audio from the console it looks like it's working, and even pauses like it is playing the file but there is no audio coming from the speakers. I have searched and looked through the archives and tried to fix this but I have had no success. This is an onboard Intel card (AC'97) and I also tried an SB Live card with the same result. -Jonathan * Asterisk startup: (asterisk -vvvc) * [chan_oss.so] = (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found == Registered channel type 'Console' (OSS Console Channel Driver) * Dial 100: * *CLI -- Executing Answer(OSS/dsp, ) in new stack Console call has been answered -- Executing Playback(OSS/dsp, tones-that-follow-are-for-the-deaf) in new stack -- Playing 'tones-that-follow-are-for-the-deaf' (language 'en') * *** pause while it plays but no audio *** * -- Executing Hangup(OSS/dsp, ) in new stack == Spawn extension (default, 100, 3) exited non-zero on 'OSS/dsp' Hangup on console * Exit asterisk: (ctrl-c which normally I wouldn't do) * Beginning asterisk shutdown Executing last minute cleanups == Destroying musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Asterisk cleanly ending (2). * Run mpg123 (Plays fine): (don't need the /dev/dsp, I was just trying to make mpg123 not work to hopefully find out why asterisk doesn't) * [EMAIL PROTECTED] ~]# mpg123 -d /dev/dsp /var/lib/asterisk/mohmp3/TristeAlegriaPromo.mp3 High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. Uses code from various people. See 'README' for more! THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK! Title : 10 - Track 10 Artist: Unknown Album : PROMO Year : Comment: Genre : Club Directory: /var/lib/asterisk/mohmp3/ Playing MPEG stream from TristeAlegriaPromo.mp3 ... Junk at the beginning 49443303 MPEG 1.0 layer III, 128 kbit/s, 44100 Hz joint-stereo [0:02] Decoding of TristeAlegriaPromo.mp3 finished. [EMAIL PROTECTED] ~]# * Extensions.conf * exten = 100,1,Answer exten = 100,2,Playback(tones-that-follow-are-for-the-deaf) exten = 100,3,Hangup * oss.conf * ; ; Open Sound System Console Driver Configuration File ; [general] ; ; Automatically answer incoming calls on the console? Choose yes if ; for example you want to use this as an intercom. ; autoanswer=yes ; ; Default context (is overridden with @context syntax) ; context=default ; ; Default extension to call ; extension=s ; ; Default language ; ;language=en ; ; Silence supression can be enabled when sound is over a certain threshold. ; The value for the threshold should probably be between 500 and 2000 or so, ; but your mileage may vary. Use the echo test to evaluate the best setting. ;silencesuppression = yes ;silencethreshold = 1000 ; ; On half-duplex cards, the driver attempts to switch back and forth between ; read and write modes. Unfortunately, this fails sometimes on older hardware. ; To prevent the driver from switching (ie. only play files on your speakers), ; then set the playbackonly option to yes. Default is no. Note this option has ; no effect on full-duplex cards. ;playbackonly=yes ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] No Audio from Console but mpg123 from shell works fine.
I get audio from mpg123 at the command line but when I load up asterisk and try to get audio from the console it looks like it's working, and even pauses like it is playing the file but there is no audio coming from the speakers. I have searched and looked through the archives and tried to fix this but I have had no success. This is an onboard Intel card (AC'97) and I also tried an SB Live card with the same result. -Jonathan * Asterisk startup: (asterisk -vvvc) * [chan_oss.so] = (OSS Console Channel Driver) == Parsing '/etc/asterisk/oss.conf': Found == Registered channel type 'Console' (OSS Console Channel Driver) * Dial 100: * *CLI -- Executing Answer(OSS/dsp, ) in new stack Console call has been answered -- Executing Playback(OSS/dsp, tones-that-follow-are-for-the-deaf) in new stack -- Playing 'tones-that-follow-are-for-the-deaf' (language 'en') * *** pause while it plays but no audio *** * -- Executing Hangup(OSS/dsp, ) in new stack == Spawn extension (default, 100, 3) exited non-zero on 'OSS/dsp' Hangup on console * Exit asterisk: (ctrl-c which normally I wouldn't do) * Beginning asterisk shutdown Executing last minute cleanups == Destroying musiconhold processes Yuck! Error in buffer handling...: Connection reset by peer Asterisk cleanly ending (2). * Run mpg123 (Plays fine): (don't need the /dev/dsp, I was just trying to make mpg123 not work to hopefully find out why asterisk doesn't) * [EMAIL PROTECTED] ~]# mpg123 -d /dev/dsp /var/lib/asterisk/mohmp3/TristeAlegriaPromo.mp3 High Performance MPEG 1.0/2.0/2.5 Audio Player for Layer 1, 2 and 3. Version 0.59r (1999/Jun/15). Written and copyrights by Michael Hipp. Uses code from various people. See 'README' for more! THIS SOFTWARE COMES WITH ABSOLUTELY NO WARRANTY! USE AT YOUR OWN RISK! Title : 10 - Track 10 Artist: Unknown Album : PROMO Year : Comment: Genre : Club Directory: /var/lib/asterisk/mohmp3/ Playing MPEG stream from TristeAlegriaPromo.mp3 ... Junk at the beginning 49443303 MPEG 1.0 layer III, 128 kbit/s, 44100 Hz joint-stereo [0:02] Decoding of TristeAlegriaPromo.mp3 finished. [EMAIL PROTECTED] ~]# * Extensions.conf * exten = 100,1,Answer exten = 100,2,Playback(tones-that-follow-are-for-the-deaf) exten = 100,3,Hangup * oss.conf * ; ; Open Sound System Console Driver Configuration File ; [general] ; ; Automatically answer incoming calls on the console? Choose yes if ; for example you want to use this as an intercom. ; autoanswer=yes ; ; Default context (is overridden with @context syntax) ; context=default ; ; Default extension to call ; extension=s ; ; Default language ; ;language=en ; ; Silence supression can be enabled when sound is over a certain threshold. ; The value for the threshold should probably be between 500 and 2000 or so, ; but your mileage may vary. Use the echo test to evaluate the best setting. ;silencesuppression = yes ;silencethreshold = 1000 ; ; On half-duplex cards, the driver attempts to switch back and forth between ; read and write modes. Unfortunately, this fails sometimes on older hardware. ; To prevent the driver from switching (ie. only play files on your speakers), ; then set the playbackonly option to yes. Default is no. Note this option has ; no effect on full-duplex cards. ;playbackonly=yes ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Don't know what to do if second ROSE componentis of type 0x6
I have been getting that message also. I have been using various versions of CVS head since Feb. 2005. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeremy Gault Sent: Friday, October 14, 2005 10:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Don't know what to do if second ROSE componentis of type 0x6 FWIW, we are also seeing this message each time we receive a call. I also went the Google route and found only questions, not answers. We are running a PRI from US LEC (channels 1-10 are B-channels, with channel 24 being the D-channel, and we are only running voice on the PRI.) The PRI is connected directly to our Digium TE110P card, and obviously we are using the zaptel drivers. We did not see this message when running */zaptel/libpri 1.0.9. However, after upgrading (we went to 1.2.0-beta1 and now to CVS) we started seeing the message. (I don't remember exactly if we saw it in the beta, but we do in the CVS.) In our case, it does not seem to affect the stability of our * machine. (However, bear in mind that you may be using parts of * that we do not, and the problem could lie in those parts.) We're handling all PSTN calls via the PRI, except outbound to toll-free which are handed off to an IAX gateway on the Internet. Our employees' desks are connected via the LAN (using Polycom 500/501 SIP phones.) I have a remote extension at home (also SIP) using a Sipura SPA-2000. We did have some stability issues (Asterisk would segfault) when we first moved to CVS. Of course, safe_asterisk handled this and a couple of days later we updated again from CVS and it seemed to fix the stability issue we were having. If you are using CVS (but not the latest one) you may want to try upgrading. I wouldn't worry about that message, though. However, I would also be interested in knowing what it means/what causes it. :) Jeremy Tom Rymes wrote: Has anyone figured out what this message means: Don't know what to do if second ROSE component is of type 0x6 We are running a PRI through a Sangoma card that is handling the D-channel natively at this point, but we go the error when zaptel was handling the D-channel, too. I have googled, but all of the messages are like this one with no answers that I can find. It's probably a non-issue, but we have been having issues with stability of our * install and I'd like to figure this out! Tom ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Jeremy Gault, KD4NED[EMAIL PROTECTED] Network Administrator, WinWorld Corporation Member: Bradley County ACS/RACES/SkyWarn voice: +1.423.473.8084 fax: +1.423.472.9465 Want a free GMail invite? E-Mail me and let me know! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 2 POTS to
I dont think the Quintum hardware supports SIP devices (just SIP trunks). -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Claudio Canseco Sent: Friday, October 14, 2005 4:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 2 POTS to Hi all, Im trying to build an small home system. I have 2 pots lines, and i need to make 8 extensions and be able to use my old analog phones. What would you recommend to use asthe 8FXS switch? I saw some equipment from quintum, they have a Tenor AS that offer 4 FXS ports. But i don't know if it is the best solution. Does anyone have a better solution to build this system? If an analog switch for 2 incoming POTS to 8 POTS is a better solution, i would appreciate ifyou could point me to posibles solutions. But I would prefer not to lose the IP option, so later i could ad some ip phones, or softphones, and be able to make calls to FWD numbers, etc, through my internet connection. Regards, tia Claudio ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Changing IP on Asterisk
I have changed the IP. It would only have an affect on your system if you have a specific bind x.x.x.x in your config files. I use bind 0.0.0.0 to use all addresses on the machine so I had no problems. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Shaikh Jallaluddin Sent: Thursday, October 06, 2005 3:29 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Changing IP on Asterisk Hi List, I would like to change the IP Address of my Asterisk PBX, Would change of IP has any effect on Asterisk or its dependent application. Has any one done this before. If so please let me know if there is any procedure to change the IP. Thanks Shaikh ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
