Re: [Asterisk-Users] Busy not jumping n + 101 anymore
Andrew Kohlsmith wrote: On Friday 14 October 2005 20:50, Eric ManxPower Wieling wrote: Oddly enough, I believe it's mentioned in UPGRADE.txt. Care to tell us where? I just checked my CVS HEAD copy of UPGRADE.txt. Sorry, it's in asterisk/configs/extensions.conf.sample ___ --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] Disconnecting after 1 min while Communicating Clarent class 5 call manager
Hi List I installed asterisk server and tried to transfer calls from asterisk to Clarent class 5 call manager. The calls are passing through with out any problem but after 60 seconds the call get disconnected automatically. Please help me to sort out this problem. Attaching here with my sip configuration file Sip.conf. [general] rtpholdtimeout=300 rtptimeout=300 defaultexpirey=20 context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes [clarent] register = test:[EMAIL PROTECTED]/123456 type=friend secret=1234 username=testuser host=192.168.10.150 fromuser=test1234 insecure=very fromdomain=192.168.10.150 canreinvite=no nat=no disallow=all allow=g723 context=default Thanks in advance. Anil ___ --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] Attended Call Transfer
Title: Message Hi, We're tryingto setupattended call transfer, but we have not been able to findthe required configuration. Blind transfer works fine using the # key, but we don't like the fact that thetransferring extension does not have any info on what happened to the call. Any pointers to setting up proper attended transfer? Thanks, DenisThe information contained in this email is confidential and may be privileged. It is intended for the addressee only, if you are not the intended recipient please notify the sender and delete the email immediately. The contents of this email must not be disclosed or copied without the senders consent. We cannot accept any responsibility for viruses. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Philip Toledo Limited ___ --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] Hints and Call Waiting
Hi! We have a big problem in our call center: when an agent does an outgoing call it can receive calls from the queues. The same happens if one agent transfer a call for another agent... and the ringing tone while in a call is puting the agents like crazy... We have the hints working with lines like this in extensions.conf: exten = 101,hint,SIP/101 If we set incominglimit to 1 the agent cannot do another call (to do attended transfers) We are using Beta1 Can anyone help? Thanks, Joao Antunes ___ --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] Quad BRI with Fedora, anyone?
We have a QuadBRI ISDN card from Digium. We would like to make it work with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of bristuff from the Digium homepage fails, both the stable version with asterisk 1.0.9, and the experimental version with asterisk CVS-HEAD. Has anybody here succeeded to make this work? Or could we even be so lucky that somebody made RPMs for this? Lars Dybdahl. ___ --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] Busy not jumping n + 101 anymore
Eric ManxPower Wieling wrote: Sorry, it's in asterisk/configs/extensions.conf.sample And the default is supposed to be 'on', so that it is backwards compatible unless you turn it off (which is in the sample config file so that new users will learn to build their dialplans with it turned off). If it is defaulting to off for some reason, that is a bug. ___ --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] Busy not jumping n + 101 anymore
Kevin P. Fleming wrote: Eric ManxPower Wieling wrote: Sorry, it's in asterisk/configs/extensions.conf.sample And the default is supposed to be 'on', so that it is backwards compatible unless you turn it off (which is in the sample config file so that new users will learn to build their dialplans with it turned off). If it is defaulting to off for some reason, that is a bug. ___ --Bandwidth and Colocation sponsored by Easynews.com -- When exactly was THAT change? I upgraded recently, but did not see that behaviour: Asterisk CVS HEAD built by [EMAIL PROTECTED] on a x86_64 running Linux on 2005-10-11 01:01:53 UTC bye Ronald Wiplinger ___ --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] Problem with '#' key recognition
Hi, I seem to be unable to get Asterisk to recognise the '#' key being pressed to acknowledge an incoming call from a queue. No matter how many times I press the key to acknowledge, the Asterisk server acts as if I have not. I have installed the ztdummy module, and it seems that Asterisk is picking it up OK (mainly since it's not moaning about permissions for IAX timing any more). I am using the Asterisk version that is packaged in Debian Stable (Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k) and the call that goes out expecting a '#' goes over an IAX provider to a UK mobile/landline. I have Googled on the subject but can't seem to find anything useful. I was wondering if anyone had any ideas or places to look before I start tearing my hair out. Thanks, Colin ___ --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] Maintenance panel
Where can I get the maintenance panel for AMP? I have searched all over and cannot seem to find it. Thank you in advance, Tommy ___ --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] Maintenance panel
thats an Asterisk At Home mod.On 10/15/05, Tommy Denton [EMAIL PROTECTED] wrote: Where can I get the maintenance panel for AMP? I have searched all over and cannot seem to find it. Thank you in advance, Tommy ___--Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --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] AGI Variable problem
kind of difficutl to help you if you dont provide the script or
relevant info about your php.ini configuration, stuff related to the
output buffer.
I used to do programming with phpagi class, but then i came up with something more simple and usefull for my purposes.
http://galileo.ivsol.net/scripts/AgiPhp5.php
only works with php, but with small modifications can work with php4.
best regardsOn 10/13/05, René Enskat [Teamware GmbH] [EMAIL PROTECTED] wrote:
The (/var/lib/asterisk/agi-bin/phpagi.php) is the newest form the siteupdated today, and i wrote the script like other examples and i can't find asyntax mistake inside extension.conf and the php script .(
On Thu, 13 Oct 2005 09:32:14 -0500Moises Silva [EMAIL PROTECTED] wrote: for some reason your script is not executing the get_var correctly, as you
can see in the output, asterisk is saying: invalid or unknown command. check the internals of your script, the most common reason is that you are mispelling the command.
best regards On 10/13/05, René Enskat [Teamware GmbH] [EMAIL PROTECTED] wrote: Hello all, I try to use a agi script to get a variable from * und put them into a
script which gives me another variablke and put this in *. My problem is now it seems the var ID is empty coz i always jump into the result 0 loop. The $MSN should be in the SetCIDNum.
#!/usr/bin/php -q ?php include(/var/lib/asterisk/agi-bin/phpagi.php); $agi = new AGI(); $ID = $agi-get_variable(SIPUSER);
if ($ID['result'] == 0) { $agi-verbose(SIPUSER not set -- nothing to do); exit(1); } $agi-set_variable(MSN, exec(/var/lib/asterisk/agi-bin/msn4sip 111
222 333 .$ID['data'])); ? Output from asterisk: -- Executing SetVar(SIP/31-79e2, SIPUSER=31) in new stack -- Executing AGI(SIP/31-79e2,
msn4sip.agi) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/msn4sip.agi msn4sip.agi: Arrayn(n [code] = 510n [result] = n [data] = Invalid or unknown commandn)n
msn4sip.agi: SIPUSER not set -- nothing to do -- AGI Script msn4sip.agi completed, returning 0 -- Executing SetLanguage(SIP/31-79e2, de) in new stack -- Executing SetCIDNum(SIP/31-79e2, ) in new stack
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Re: [Asterisk-Users] Problem with '#' key recognition
On Sat, Oct 15, 2005 at 03:21:38PM +0100, Colin Martin wrote: Hi, I seem to be unable to get Asterisk to recognise the '#' key being pressed to acknowledge an incoming call from a queue. No matter how many times I press the key to acknowledge, the Asterisk server acts as if I have not. I have installed the ztdummy module, and it seems that Asterisk is picking it up OK (mainly since it's not moaning about permissions for IAX timing any more). I am using the Asterisk version that is packaged in Debian Stable (Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k) and the call that goes out expecting a '#' goes over an IAX provider to a UK mobile/landline. What phone do you use? How is it connected to Asterisk? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with '#' key recognition
Tzafrir Cohen wrote: On Sat, Oct 15, 2005 at 03:21:38PM +0100, Colin Martin wrote: Hi, I seem to be unable to get Asterisk to recognise the '#' key being pressed to acknowledge an incoming call from a queue. No matter how many times I press the key to acknowledge, the Asterisk server acts as if I have not. I have installed the ztdummy module, and it seems that Asterisk is picking it up OK (mainly since it's not moaning about permissions for IAX timing any more). I am using the Asterisk version that is packaged in Debian Stable (Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k) and the call that goes out expecting a '#' goes over an IAX provider to a UK mobile/landline. What phone do you use? How is it connected to Asterisk? I have tried all the following phones: Cisco 7940 (via SIP) UK Mobile, Orange, Samsung D500 (via IAX to outbound provider) UK Landline (via IAX to outbound provider) It just seems that Asterisk totally ignores the '#' key. I found the following on http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20AgentCallbackLogin (Note: Without a valid zaptel timing source, the '#' acknowledgement will not happen.) Although I have installed the ztdummy module, is there any way to get Asterisk to tell me if it is working or not? Many thanks for the response. Colin ___ --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] Attended Call Transfer
Search voip-info.org or google for features.conf.TomOn Oct 15, 2005, at 4:58 AM, Denis Vella wrote: Hi, We're trying to setup attended call transfer, but we have not been able to find the required configuration. Blind transfer works fine using the # key, but we don't like the fact that the transferring extension does not have any info on what happened to the call. Any pointers to setting up proper attended transfer? Thanks, Denis The information contained in this email is confidential and may be privileged. It is intended for the addressee only, if you are not the intended recipient please notify the sender and delete the email immediately. The contents of this email must not be disclosed or copied without the senders consent. We cannot accept any responsibility for viruses. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Philip Toledo Limited___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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] Hints and Call Waiting
We use Cisco phones and we simply disabled call-waiting for those lines. Don't know if that will help, but whatever soft/hardphone you are using probably has a way to disable call-waiting. Tom On Oct 15, 2005, at 5:38 AM, João Paulo Antunes wrote: Hi! We have a big problem in our call center: when an agent does an outgoing call it can receive calls from the queues. The same happens if one agent transfer a call for another agent... and the ringing tone while in a call is puting the agents like crazy... We have the hints working with lines like this in extensions.conf: exten = 101,hint,SIP/101 If we set incominglimit to 1 the agent cannot do another call (to do attended transfers) We are using Beta1 Can anyone help? Thanks, Joao Antunes ___ --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 Vip-150T
Hi All, I'm having problem with this phone. Problems are regarding voicemail message alert on the phone. --- handle_response: Host 'xxx.xxx.xxx.xxx' does not implement 'NOTIFY' --- Can somebody help? On the phone manual, is written that it can acept MWI, but... not mine!!! Thanks! -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad BRI with Fedora, anyone?
Actually, the QuadBRI card is not from Digium, but manufactured by Junghanns.net Michael 2005/10/15, Lars Dybdahl [EMAIL PROTECTED]: We have a QuadBRI ISDN card from Digium. We would like to make it work with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of bristuff from the Digium homepage fails, both the stable version with asterisk 1.0.9, and the experimental version with asterisk CVS-HEAD. Has anybody here succeeded to make this work? Or could we even be so lucky that somebody made RPMs for this? Lars Dybdahl. ___ --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] Direct Dial In - second try
Finally(!), the answer is in /etc/asterisk/zapata.conf add 'overlapdial=yes' hope that helps someone with the same problem. many thanks to [EMAIL PROTECTED] and gerold On Mon, 3 Oct 2005 23:55:21 +0200 ChB [EMAIL PROTECTED] wrote: Hi all, I have an asterisk-server (cvs-head from august) connected to a carrier's switch (DMS/Euroisdn) via a te410p, and I am having problems with DDI (standard 'official pstn' number plus extra digits for 'internal' use) Basically, when the entire number (including the extra digits) is dialled via a redial or a programmed key, I see the entire called party number (including the extra numbers) in * debug. However if I dial the number manually (digit for digit) the carrier puts it through before I've finished dialling (after 10 digits), and I never see the whole DDI dialed. Since I want to be able to use numbers with and without DDIs, I can't tell the carrier to raise the minimum level to more than 10 since numbers dialled without DDIs would never be passed. The carrier is telling me to wait a second or two before seizing the call, and that any additional digits received would be passed in the isdn protocol. My understanding is that the called party number would be retransmitted, including the DDI. (Wait in the dialplan doesn't work, as the call is already taken) Does anyone have any idea why this is not working? Is the carrier right? This doesn't seem to be such an odd feature that nobody else would use it ;-), so please leave a comment, even if it works for you out of the box without fuss. Your help is highly appreciated, thanks! I have included two traces. The first is the number dialled with a speed dial key, the second is exactly the same number dialed manually. Best regards Christian trace: with speeddial(sent as one block): mgw1*CLI Protocol Discriminator: Q.931 (8) len=35 Call Ref: len= 2 (reference 1549/0x60D) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [6c 02 00 a1] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation prohibited, user number passed network screening (33) '' ] [70 0e a1 30 37 32 30 30 30 33 34 35 36 37 38 39] Called Number (len=16) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '07xx6789' ] -- Making new call for cr 1549 -- Processing Q.931 Call Setup -- Processing IE 4 (cs0, Bearer Capability) -- Processing IE 24 (cs0, Channel Identification) -- Processing IE 108 (cs0, Calling Party Number) -- Processing IE 112 (cs0, Called Party Number) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 1549/0x60D) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] -- Accepting call from '' to '07xx6789' on channel 0/2, span 1 -- Executing GotoIf(Zap/2-1, 0?100:2) in new stack snip dialed manually digit by digit: Protocol Discriminator: Q.931 (8) len=32 Call Ref: len= 2 (reference 1543/0x607) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a1 83 82] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Preferred Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 2 ] [6c 02 00 a1] Calling Number (len= 4) [ Ext: 0 TON: Unknown Number Type (0) NPI: Unknown Number Plan (0) Presentation: Presentation prohibited, user number passed network screening (33) '' ] [70 0b a1 30 37 32 30 30 30 33 34 35 36] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '072xx6' ] -- Making new
Re: [Asterisk-Users] ASTCC -- semantic note of 'callstart' in cdrs?
This will actually be easy to fix. I'll post a patch along with
someother stuff shortly.
Darren
Darren Wiebe wrote:
That is true. It's just one of those things that is easier to leave
alone to avoid breakage in upgrades. It would be nice to get fixed
though
Darren Wiebe
[EMAIL PROTECTED]
Eric Lyons wrote:
Looking at the code, it would appear that the 'callstart' column of
the cdrs table should really be called 'callend':
$dialstr = IAX2/$res-{path}/$phone|30|HL( .
($maxtime * 60 * 1000) . :6:3);
$res = $AGI-exec(DIAL $dialstr);
$answeredtime = $AGI-get_variable(ANSWEREDTIME);
$dialstatus = $AGI-get_variable(DIALSTATUS);
$callstart = localtime();
return $dialstatus;
No?
Eric.
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[Asterisk-Users] sip show peers
Hi, I have 3 SIP extensions, setup as follows: # Device Location options 200 Sipura local 210 Sipura remote nat=yes qualify=yes 310 eyebeam remote nat=yes qualify=yes This is the result of sip show peers: Name/user Host Dyn Nat Status 200/200 192.168.1.150 D Unmonitored 210/210 84.36.36.140 D N OK (305 ms) 310/310 71.180.126.60 D N Unmonitored Does anyone know why ext 210 the only one has a ping status OK (305 ms) and the others are Unmonitored Regards __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad BRI with Fedora, anyone?
On Sat, 2005-10-15 at 14:29 +0200, Lars Dybdahl wrote: We have a QuadBRI ISDN card from Digium. We would like to make it work with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of bristuff from the Digium homepage fails, both the stable version with asterisk 1.0.9, and the experimental version with asterisk CVS-HEAD. Has anybody here succeeded to make this work? Or could we even be so lucky that somebody made RPMs for this? Perhaps if you stated more clearly the actual errors some one might be able to help you. What worries me, is you don't seem to know where you got the source from. -- Dave Cotton [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons license, allowing the book in its entirity to be freely distributed. Asterisk: The Future of Telephony is now freely available, for download in PDF form, from the Asterisk Documentation Project website located at http://www.asteriskdocs.org. On the left hand side, click on Read the book online! for a copy. The authors would like to thank O'Reilly Media for having the vision to understand how significant it is for the Asterisk community to have a book freely available, thereby lowering the barrier of entry for those new to Asterisk, and to give back to a project that has given us all so much. I would personally like to thank Jared Smith, Jim van Meggelen, Michael Loukides (our editor) and the entire O'Reilly Media staff. The book is currently shipping, and should be available at all major book stores in paperback, and also online from http://www.oreilly.com/catalog/asterisk/ and other online outlets. Thanks, and we hope you enjoy reading it as much as we enjoyed writing it! PS: If the Asterisk Documentation Project website becomes slow due to the number of people accessing it at once, we appoligize and appreciate your patience. For those of you who are able to obtain the full copy, please consider helping us out by creating mirrors and torrents and posting them to the list by replying to this thread. Thanks! -- Leif Madsen - http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --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] DID on analog line
Can someone tell me how I can test DID on analog lines ? I have the WCfxo clone and would like to try DID on it. Can my telco (qwest) provide that service ? -apu ___ --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] Attempted to delete nonexistent schedule entry...
I'm also having this issue. Everything seems to work, but it's an unnerving error. Any thoughts? Jimmy wrote: I just upgraded my test Asterisk box to the latest CVS HEAD. show version only shows Asterisk CVS HEAD built by rootetc, with no date or version number. I downloaded this version on Monday, Oct 3. About once every minute, I get this while at the CLI prompt: sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 1! This only appeared after updating. All functions seem normal, other than these messages. Phones work, auto-attendant works, voicemail works, etc. What's going on? ___ --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] Busy not jumping n + 101 anymore
Eric ManxPower Wieling wrote: Andrew Kohlsmith wrote: On Friday 14 October 2005 20:50, Eric ManxPower Wieling wrote: Oddly enough, I believe it's mentioned in UPGRADE.txt. Care to tell us where? I just checked my CVS HEAD copy of UPGRADE.txt. Sorry, it's in asterisk/configs/extensions.conf.sample Which isn't even produced if one doesn't make samples What backwards thinking put the information there, and in addition changed the way jumps used to work as the default? If more time were spent on fixing things that were broken, and making the interface to the existing PSTN analog lines work smoother there might be more acceptance. JMO JN ___ --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] sip show peers
Are the devices at 200 and 310 set up to register with your asterisk? On Sat, 2005-10-15 at 11:42 -0700, Samy Antoun wrote: Hi, I have 3 SIP extensions, setup as follows: # Device Location options 200 Sipura local 210 Sipura remote nat=yes qualify=yes 310 eyebeam remote nat=yes qualify=yes This is the result of sip show peers: Name/user Host Dyn Nat Status 200/200 192.168.1.150 D Unmonitored 210/210 84.36.36.140 D N OK (305 ms) 310/310 71.180.126.60 D N Unmonitored Does anyone know why ext 210 the only one has a ping status OK (305 ms) and the others are Unmonitored Regards __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --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] sip show peers
--- Sergey Okhapkin [EMAIL PROTECTED] wrote: Are the devices at 200 and 310 set up to register with your asterisk? Yes, they are registered and I can call them __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Quad BRI with Fedora, anyone?
I had a typo in my original mail - of course it's from junghanns.net, not from digium, and so is the bristuff, that I downloaded. I'm not searching for solutions on how to make it compile - I'm just trying to find out if anybody succeeded in having it work on FC3 and FC4, and if yes, if there are RPMs available. It seems that the bristuff source from junghanns.com wasn't written for gcc 4, which is the one included in FC4, and I have seen some descriptions on making zaphfc compile, but there are more problems than just that one. Also, RPMs would reduce the amount of time spent on making this work significantly. I hope you can forgive my typo in my original mail and give me a hint or two :-) Lars Dybdahl On 10/15/05, Dave Cotton [EMAIL PROTECTED] wrote: Perhaps if you stated more clearly the actual errors some one might be able to help you. What worries me, is you don't seem to know where you got the source from. ___ --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] Attempted to delete nonexistent schedule entry...
On 10/15/2005, J. Iddings [EMAIL PROTECTED] wrote: I'm also having this issue. Everything seems to work, but it's an unnerving error. Any thoughts? Jimmy wrote: I just upgraded my test Asterisk box to the latest CVS HEAD. show version only shows Asterisk CVS HEAD built by rootetc, with no date or version number. I downloaded this version on Monday, Oct 3. About once every minute, I get this while at the CLI prompt: sched.c:296 ast_sched_del: Attempted to delete nonexistent schedule entry 1! This only appeared after updating. All functions seem normal, other than these messages. Phones work, auto-attendant works, voicemail works, etc. What's going on? OK - I been wrong so many times this week - it ain't funny... But - I think - this part of the scheduling change to the registration stuff. In one update, when a remote phone/system stopped responding to qualify attempts, the system would stop trying to verify the connection. Forever. Not exactly a 'good thing'. It would tell you that by saying Forever but still not good. Then an update added some stuff to ?iax.conf? like: ;qualify=yes ;qualifysmoothing = yes ;qualifyfreqok = 12 ;qualifyfreqnotok = 3 to modify how and when the system would retry these connections. During the time between the first and second update, I would get these messages when I did an iax2 reload. It had stopped trying to qualify the connection - and then the reload would start it backup. It would 'inform' me with the 'attempted to delete nonexistant schedule entry' because the time of the next scheduled event was no longer active. So in essence - it is a warning and not an error. 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
Re: [Asterisk-Users] Busy not jumping n + 101 anymore
On 10/15/2005, John Novack [EMAIL PROTECTED] wrote: Eric ManxPower Wieling wrote: Andrew Kohlsmith wrote: On Friday 14 October 2005 20:50, Eric ManxPower Wieling wrote: Oddly enough, I believe it's mentioned in UPGRADE.txt. Care to tell us where? I just checked my CVS HEAD copy of UPGRADE.txt. Sorry, it's in asterisk/configs/extensions.conf.sample Which isn't even produced if one doesn't make samples What backwards thinking put the information there, and in addition changed the way jumps used to work as the default? If more time were spent on fixing things that were broken, and making the interface to the existing PSTN analog lines work smoother there might be more acceptance. JMO Well - number 1 - it IS - CVS HEAD. Next - I always run make samples. In the /etc/asterisk/ directory it renames all your old config files that have changed to *.old. So long as you don't stop and restart Asterisk - its fine. Just diff the two files and you get to see the differences. Now that seems a 'funny' thing to do as you are thinking - heck ALL my configs are 'different' - as are mine. But I keep a subdirectory full of MY config files (just in case) and I also keep a subdirectory full of the LAST updates config files just to compare against. After seeing what's new - fixing MY config files - moving the NEW ones to the latest subdirectory - moving MY config files back to the working directory - THEN I restart Asterisk. It's called system administration. One day I'll set it up with a script, but right now it isn't bad or too much work. But I do update almost everyday. 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
Re: [Asterisk-Users] Quad BRI with Fedora, anyone?
On Sat, 2005-10-15 at 21:35 +0200, Lars Dybdahl wrote: I had a typo in my original mail - of course it's from junghanns.net, not from digium, and so is the bristuff, that I downloaded. I'm not searching for solutions on how to make it compile - I'm just trying to find out if anybody succeeded in having it work on FC3 and FC4, and if yes, if there are RPMs available. It seems that the bristuff source from junghanns.com wasn't written for gcc 4, which is the one included in FC4, and I have seen some descriptions on making zaphfc compile, but there are more problems than just that one. Also, RPMs would reduce the amount of time spent on making this work significantly. Well all I can say is that I've compiled and loaded it without problem on Mandriva Cooker with gcc4 and also on a 64 bit system. -- Dave Cotton [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] Quad BRI with Fedora, anyone?
On 10/15/05, Lars Dybdahl [EMAIL PROTECTED] wrote: It seems that the bristuff source from junghanns.com wasn't written for gcc 4, which is the one included in FC4, and I have seen somedescriptions on making zaphfc compile, but there are more problemsthan just that one. Also, RPMs would reduce the amount of time spenton making this work significantly. What kernel are you using? What version of bristuff are you using? (latest is bristuff-0.2.0-RC8p.tar.gz) Where do you encounter errors, and wich errors do you get? ___ --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] You ASKED for an Asterisk book, you GOT an Asterisk book!
On 15 Oct 2005, at 19:58, Leif Madsen wrote: Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons license, allowing the book in its entirity to be freely distributed. Asterisk: The Future of Telephony is now freely available, for download in PDF form, from the Asterisk Documentation Project website located at http://www.asteriskdocs.org. On the left hand side, click on Read the book online! for a copy. The authors would like to thank O'Reilly Media for having the vision to understand how significant it is for the Asterisk community to have a book freely available, thereby lowering the barrier of entry for those new to Asterisk, and to give back to a project that has given us all so much. I would personally like to thank Jared Smith, Jim van Meggelen, Michael Loukides (our editor) and the entire O'Reilly Media staff. The book is currently shipping, and should be available at all major book stores in paperback, and also online from http://www.oreilly.com/catalog/asterisk/ and other online outlets. Thanks, and we hope you enjoy reading it as much as we enjoyed writing it! PS: If the Asterisk Documentation Project website becomes slow due to the number of people accessing it at once, we appoligize and appreciate your patience. For those of you who are able to obtain the full copy, please consider helping us out by creating mirrors and torrents and posting them to the list by replying to this thread. Thanks! No, thank you! I've read the book and it is very good. Just what asterisk needed. I've mirrored it on our website at http://www.westhawk.co.uk/resources/AsteriskTFOT.zip Tim -- Leif Madsen - http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --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] You ASKED for an Asterisk book, you GOT anAsterisk book!
I´m a new user.I´m reading it seams to be very good. Great job. It´s realy the book i as tryng to find! Regards! Cumprimentos, André Rodrigues Grupo Paulo Serra Irmãos, Lda. Direcção de Sistemas de Informação Tel.: +351 25237 (ext: 296) Fax: +351 252313483 Telem : +351 964245524 E-mail :[EMAIL PROTECTED] Website: www.cheyenne-pt.com -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Mr. James W. Laferriere Enviada: sábado, 15 de Outubro de 2005 20:46 Para: Leif Madsen; Asterisk Users Mailing List - Non-Commercial Discussion Cc: Discussions regarding The Asterisk Documentation Project Assunto: Re: [Asterisk-Users] You ASKED for an Asterisk book, you GOT anAsterisk book! Hello Leif , The appendices A B are missing from the zip file available at the location mentioned below . Is there some reason of copyright that is not mentioned here ? Tia , JimL On Sat, 15 Oct 2005, Leif Madsen wrote: Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons license, allowing the book in its entirity to be freely distributed. Asterisk: The Future of Telephony is now freely available, for download in PDF form, from the Asterisk Documentation Project website located at http://www.asteriskdocs.org. On the left hand side, click on Read the book online! for a copy. The authors would like to thank O'Reilly Media for having the vision to understand how significant it is for the Asterisk community to have a book freely available, thereby lowering the barrier of entry for those new to Asterisk, and to give back to a project that has given us all so much. I would personally like to thank Jared Smith, Jim van Meggelen, Michael Loukides (our editor) and the entire O'Reilly Media staff. The book is currently shipping, and should be available at all major book stores in paperback, and also online from http://www.oreilly.com/catalog/asterisk/ and other online outlets. Thanks, and we hope you enjoy reading it as much as we enjoyed writing it! PS: If the Asterisk Documentation Project website becomes slow due to the number of people accessing it at once, we appoligize and appreciate your patience. For those of you who are able to obtain the full copy, please consider helping us out by creating mirrors and torrents and posting them to the list by replying to this thread. Thanks! -- Leif Madsen - http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --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 -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --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] sip show peers
Hmm.. What is the output of sip show users and sip show peers? On Sat, 2005-10-15 at 12:30 -0700, Samy Antoun wrote: --- Sergey Okhapkin [EMAIL PROTECTED] wrote: Are the devices at 200 and 310 set up to register with your asterisk? Yes, they are registered and I can call them __ Start your day with Yahoo! - Make it your home page! http://www.yahoo.com/r/hs ___ --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] Quad BRI with Fedora, anyone?
You can find RPMS with bristuff included at: http://www.laimbock.com/asterisk/ the are compiled for centos. rebuilding the SRPMS under FC3 work without a problem. We have a QuadBRI ISDN card from Digium. We would like to make it work with Fedora Core 4 (maybe FC3), but haven't succeeded. Compilation of bristuff from the Digium homepage fails, both the stable version with asterisk 1.0.9, and the experimental version with asterisk CVS-HEAD. Has anybody here succeeded to make this work? Or could we even be so lucky that somebody made RPMs for this? Lars Dybdahl. ___ --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] Busy not jumping n + 101 anymore
On Saturday 15 October 2005 16:18, [EMAIL PROTECTED] wrote: Well - number 1 - it IS - CVS HEAD. Agreed. Next - I always run make samples. In the /etc/asterisk/ directory it renames all your old config files that have changed to *.old. So long as you don't stop and restart Asterisk - its fine. Just diff the two files and you get to see the differences. A diff over my config files and the samples would be useless. Better to just read over the samples to see if there's anything obvious. HOWEVER. There's a changelog that should have this data in it, and the CVS logs should also have it, but more importantly, the upgrade.txt should have had this. It's just an oversight, one that I hope Digium'll fix shortly. It's called system administration. One day I'll set it up with a script, but right now it isn't bad or too much work. But I do update almost everyday. Nice passive-agressive attack there. System administration has nothing to do with what making samples and diff'ing. Keeping a log of changes you've made... now that is indicative of good administration. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip show peers
--- Sergey Okhapkin [EMAIL PROTECTED] wrote: Hmm.. What is the output of sip show users and sip show peers? sip show users Username Def.Context ACL NAT 200 from-internalNo No 210 from-internalNo Always 310 from-internalNo Always sip show peers Name/user Host Dyn Nat Status 200/200 192.168.1.150 D Unmonitored 210/210 84.36.36.140 D N OK (305 ms) 310/310 71.180.126.60 D N Unmonitored __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] You ASKED for an Asterisk book, you GOT an Asterisk book!
Is appendix A and B missing? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jean-Michel Hiver Sent: Saturday, October 15, 2005 5:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Discussions regarding The Asterisk Documentation Project Subject: Re: [Asterisk-Users] You ASKED for an Asterisk book,you GOT an Asterisk book! Leif Madsen a écrit : Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons license, allowing the book in its entirity to be freely distributed. Kudos for this. Brilliant! ___ --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] res_perl - Compiling error
Having trouble running make on res_perl: [EMAIL PROTECTED] res_perl]# make perl -MExtUtils::Embed -e xsinit gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/ -I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk \-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR= \/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE -c -o perlxsi.o perlxsi.c gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/ -I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk \-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR= \/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE -c -o res_perl.o res_perl.c gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/ -I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk \-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR= \/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE -c -o apihelp.o apihelp.c gcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/ -I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk \-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR= \/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DMULTIPLICITY -D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE -c -o AstAPIBase.o AstAPIBase.c gcc -Wall -fPIC -shared -Xlinker -x -o res_perl.so perlxsi.o res_perl.o apihelp.o -Wl,-E -L/usr/local/lib /usr/local/lib/perl5/5.8.7/-linux-thread-multi/auto/DynaLoader/DynaLoader.a -L/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE -lperl -lnsl -ldl -lm -lcrypt -lutil -lpthread -lc -lnsl -lndbm -lgdbm -ldl -lm -lcrypt -lutil -lc AstAPIBase.o /usr/bin/ld: cannot find -lndbm collect2: ld returned 1 exit status make: *** [res_perl.so] Error 1 [EMAIL PROTECTED] res_perl]# ___ --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] res_perl - Compiling error
Maybe you don't have libc6-dev installed? On 10/15/05, Brent August Torrenga [EMAIL PROTECTED] wrote: Having trouble running make on res_perl:[EMAIL PROTECTED] res_perl]# makeperl -MExtUtils::Embed -e xsinitgcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\-DMULTIPLICITY-D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE-D_FILE_OFFSET_BITS=64-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE -c -o perlxsi.operlxsi.cgcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk- 1.0.9/-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\\-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\ -DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\-DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\ -DMULTIPLICITY-D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS-fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE-D_FILE_OFFSET_BITS=64-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE -c -o res_perl.o res_perl.cgcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\\-DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk \-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\-DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR= \/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\-DMULTIPLICITY-D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS-fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE -c -o apihelp.oapihelp.cgcc -Wall -fPIC -DHAVE_AST_CUST_CONFIG -I/usr/src/asterisk-1.0.9/-I/usr/src/asterisk-1.0.9//include -I. -DAST_INSTALL_PREFIX=\\ -DASTETCDIR=\/etc/asterisk\ -DASTLIBDIR=\/usr/lib/asterisk\-DASTVARLIBDIR=\/var/lib/asterisk\ -DASTVARRUNDIR=\/var/run\-DASTSPOOLDIR=\/var/spool/asterisk\ -DASTLOGDIR=\/var/log/asterisk\ -DASTCONFPATH=\/etc/asterisk/asterisk.conf\ -DASTMODDIR=\/usr/lib/asterisk/modules\ -DASTAGIDIR=\/var/lib/asterisk/agi-bin\-DMULTIPLICITY-D_REENTRANT -D_GNU_SOURCE -DTHREADS_HAVE_PIDS -fno-strict-aliasing -pipe -I/usr/local/include -D_LARGEFILE_SOURCE-D_FILE_OFFSET_BITS=64-I/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE -c -oAstAPIBase.o AstAPIBase.cgcc -Wall -fPIC -shared -Xlinker -x -o res_perl.so perlxsi.o res_perl.oapihelp.o -Wl,-E-L/usr/local/lib /usr/local/lib/perl5/5.8.7/-linux-thread-multi/auto/DynaLoader/DynaLoader.a -L/usr/local/lib/perl5/5.8.7/-linux-thread-multi/CORE -lperl -lnsl -ldl -lm -lcrypt -lutil -lpthread -lc -lnsl -lndbm -lgdbm -ldl -lm -lcrypt -lutil -lcAstAPIBase.o/usr/bin/ld: cannot find -lndbmcollect2: ld returned 1 exit statusmake: *** [res_perl.so] Error 1[EMAIL PROTECTED] res_perl]#___ --Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com Phone: 518-631-2855 x205Fax: 518-631-2856 ___ --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] You ASKED for an Asterisk book, you GOT an Asterisk book!
The appendices are in the book. :) Apparently only missing in the pdf. Is appendix A and B missing? -Original Message- Leif Madsen a écrit : Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons license, allowing the book in its entirity to be freely distributed. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail 2
Hi list, I'm trying, as usual, to set up voicemail. It works, but signaling to phones, doesn't. Into XLite logs, I have: -- Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 1/0 (0/0) -- but nothing appear on the XLite screen. So, I understand that I'm able to send the right signal, but something is still wrong. Ideas? Thanks in advance -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail 2
FaberK wrote: Hi list, I'm trying, as usual, to set up voicemail. It works, but signaling to phones, doesn't. Into XLite logs, I have: -- Messages-Waiting: yes Message-Account: sip:[EMAIL PROTECTED] Voice-Message: 1/0 (0/0) -- but nothing appear on the XLite screen. So, I understand that I'm able to send the right signal, but something is still wrong. Ideas? Thanks in advance -- .:FaberK:. I have something similar. I have the little mail envelope on the screen of xten-xlite, but can't figure out how to clear it off. -- -Linc Fessenden In the Beginning there was nothing, which exploded - Yeah right... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail 2
I've got it too until yesterday!!!
Now no more envelope either.
This is from extensions.conf:
---
exten = 221,1,Dial(SIP/221,20,tr)
exten = 221,2,Voicemail(u${EXTEN})
exten = 221,102,Voicemail(b${EXTEN})
exten = 221,103,Hangup
---
this is from sip.conf:
---
[221]
type=friend
username=221
secret=221
callerid=221 221
fromuser=221
accountcode=221
context=local
host=dynamic
dtmfmode=rfc2833
nat=yes
qualify=yes
Port=5060
[EMAIL PROTECTED]
Disallow=all
Allow=gsm
Allow=ulaw
Allow=alaw
---
Some ideas?
2005/10/16, Linc Fessenden [EMAIL PROTECTED]:
FaberK wrote:
Hi list,
I'm trying, as usual, to set up voicemail.
It works, but signaling to phones, doesn't.
Into XLite logs, I have:
--
Messages-Waiting: yes
Message-Account: sip:[EMAIL PROTECTED]
Voice-Message: 1/0 (0/0)
--
but nothing appear on the XLite screen.
So, I understand that I'm able to send the right signal, but something
is still wrong.
Ideas?
Thanks in advance
--
.:FaberK:.
I have something similar. I have the little mail envelope on the screen
of xten-xlite, but can't figure out how to clear it off.
--
-Linc Fessenden
In the Beginning there was nothing, which exploded - Yeah right...
___
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--
.:FaberK:.
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Re: [Asterisk-Users] Callerid on t1 lines
What is the adit 600 doing? FXO? FXS? how you connected to the PSTN? I got an Adit 600 with both FXO and FXS as well as a PRI and I'm getting CallerID on all three. On 10/14/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello All, Just a question, I have an adit600 and I am looking for a way to pull the incoming cid into asterisk. Does anyone know if this is just not possible via t1? Or is it only available on PRI? Thanks, Greg ___ --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 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
Re: [Asterisk-Users] Voicemail 2
FaberK wrote: [EMAIL PROTECTED] --- Some ideas? Only thing I have that even looks different is [EMAIL PROTECTED] -- -Linc Fessenden In the Beginning there was nothing, which exploded - Yeah right... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail 2
First answer:
the envelope came back to me!
Yesterday I've added the line:
notifymimetype=text/plain
into sip.conf, so no more envelope just text into logs.
Is just an answer to you, Linc.
Nothing for me, yet.
2005/10/16, FaberK [EMAIL PROTECTED]:
I've got it too until yesterday!!!
Now no more envelope either.
This is from extensions.conf:
---
exten = 221,1,Dial(SIP/221,20,tr)
exten = 221,2,Voicemail(u${EXTEN})
exten = 221,102,Voicemail(b${EXTEN})
exten = 221,103,Hangup
---
this is from sip.conf:
---
[221]
type=friend
username=221
secret=221
callerid=221 221
fromuser=221
accountcode=221
context=local
host=dynamic
dtmfmode=rfc2833
nat=yes
qualify=yes
Port=5060
[EMAIL PROTECTED]
Disallow=all
Allow=gsm
Allow=ulaw
Allow=alaw
---
Some ideas?
2005/10/16, Linc Fessenden [EMAIL PROTECTED]:
FaberK wrote:
Hi list,
I'm trying, as usual, to set up voicemail.
It works, but signaling to phones, doesn't.
Into XLite logs, I have:
--
Messages-Waiting: yes
Message-Account: sip:[EMAIL PROTECTED]
Voice-Message: 1/0 (0/0)
--
but nothing appear on the XLite screen.
So, I understand that I'm able to send the right signal, but something
is still wrong.
Ideas?
Thanks in advance
--
.:FaberK:.
I have something similar. I have the little mail envelope on the screen
of xten-xlite, but can't figure out how to clear it off.
--
-Linc Fessenden
In the Beginning there was nothing, which exploded - Yeah right...
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
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--
.:FaberK:.
--
.:FaberK:.
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Re: [Asterisk-Users] Voicemail 2
well, is just the context. You could call it as you prefer, mickeymouse??? ;o) Bye 2005/10/16, Linc Fessenden [EMAIL PROTECTED]: FaberK wrote: [EMAIL PROTECTED] --- Some ideas? Only thing I have that even looks different is [EMAIL PROTECTED] -- -Linc Fessenden In the Beginning there was nothing, which exploded - Yeah right... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ACD calls to busy agents
One of my friends is facing this problems and I could not find any solution to that. Hence this post. In her Asterisk PBX, she has programmed about 10 agents, and strategy is rrmemory. Everything works fine. When an agent has received an ACD call, another call is not presented to him as long as he is on the ACD call. However when an agent has made an outgoing call, he is still presented another ACD call when his turn comes. This results in unnecessary delay in answering that call. Taking out call waiting is not an option, as an agent can also get a direct dialed call, and he should be able to pick up that call even when he is on another call. Is there a way so that a busy agent (whether busy because of an incoming call, or outgoing call) is not presented another ACD call? Thanks, -- jt ___ --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] Looking for Info on OH323
I have compiled the OH323 module for my system. When can I find some info on how to properly configure it? I haven't read any info for its configuration, and I need some starting info. Were do I start? Obelix This message was sent using IMP, the Internet Messaging Program. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail 2
Correct - but is the context defined in voicemail.conf? As mickeymouse? Or whatever...? ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Saturday, October 15, 2005 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail 2 well, is just the context. You could call it as you prefer, mickeymouse??? ;o) Bye 2005/10/16, Linc Fessenden [EMAIL PROTECTED]: FaberK wrote: [EMAIL PROTECTED] --- Some ideas? Only thing I have that even looks different is [EMAIL PROTECTED] -- -Linc Fessenden In the Beginning there was nothing, which exploded - Yeah right... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] ACD calls to busy agents
Have you tried the incominglimit parameter (or did she)? I have found this to work pretty well when limiting the number of calls. After monitoring the full log, I saw that incoming calls where incrementing or decrementing the active call parameter for SIP agents. By limiting the number of calls that the phone extension/user can accept at one time limited the calls going to an agent. I am still trying to figure out how to jump out of the dialplan when a call comes into queue -- if anyone has any suggestions for that, it would be greatly appreciated. But in any event, for similar situations, limiting the number of calls for a SIP agent seems to help in the calls coming in on top of another. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of J Thomas Sent: Saturday, October 15, 2005 7:04 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] ACD calls to busy agents One of my friends is facing this problems and I could not find any solution to that. Hence this post. In her Asterisk PBX, she has programmed about 10 agents, and strategy is rrmemory. Everything works fine. When an agent has received an ACD call, another call is not presented to him as long as he is on the ACD call. However when an agent has made an outgoing call, he is still presented another ACD call when his turn comes. This results in unnecessary delay in answering that call. Taking out call waiting is not an option, as an agent can also get a direct dialed call, and he should be able to pick up that call even when he is on another call. Is there a way so that a busy agent (whether busy because of an incoming call, or outgoing call) is not presented another ACD call? Thanks, -- jt ___ --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] ACD calls to busy agents
Setting incominglimit = 1 does not really solve the problem as I had already mentioned. That practically takes away the call waiting and will block all incoming calls including direct dialed calls. She does not want that. Moreover, incominglimit is deprecated too. -- jt On Sat, 2005-10-15 at 22:05, Jason Walker wrote: Have you tried the incominglimit parameter (or did she)? I have found this to work pretty well when limiting the number of calls. After monitoring the full log, I saw that incoming calls where incrementing or decrementing the active call parameter for SIP agents. By limiting the number of calls that the phone extension/user can accept at one time limited the calls going to an agent. I am still trying to figure out how to jump out of the dialplan when a call comes into queue -- if anyone has any suggestions for that, it would be greatly appreciated. But in any event, for similar situations, limiting the number of calls for a SIP agent seems to help in the calls coming in on top of another. -Original Message- ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Voicemail 2
Yes, is defined in voicemail.conf too. The context is the glue of the system, this is what I understood, the way to follow, the arrow that show the direction to every application. But the problems, still remain. 2005/10/16, Jason Walker [EMAIL PROTECTED]: Correct - but is the context defined in voicemail.conf? As mickeymouse? Or whatever...? ;) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of FaberK Sent: Saturday, October 15, 2005 6:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail 2 well, is just the context. You could call it as you prefer, mickeymouse??? ;o) Bye 2005/10/16, Linc Fessenden [EMAIL PROTECTED]: FaberK wrote: [EMAIL PROTECTED] --- Some ideas? Only thing I have that even looks different is [EMAIL PROTECTED] -- -Linc Fessenden In the Beginning there was nothing, which exploded - Yeah right... ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- .:FaberK:. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ACD calls to busy agents
I don't know how to make this happen, and I don't even think it is really possible given the current Queue app, but this would be a very nice feature to have. The queue shouldn't pass a call to an agent if they are already on a call from the queue, but an incoming call from another internal extension, or even a DID ought to be able to get through. Consider this a feature request? Tom On Oct 15, 2005, at 10:04 PM, J Thomas wrote: One of my friends is facing this problems and I could not find any solution to that. Hence this post. In her Asterisk PBX, she has programmed about 10 agents, and strategy is rrmemory. Everything works fine. When an agent has received an ACD call, another call is not presented to him as long as he is on the ACD call. However when an agent has made an outgoing call, he is still presented another ACD call when his turn comes. This results in unnecessary delay in answering that call. Taking out call waiting is not an option, as an agent can also get a direct dialed call, and he should be able to pick up that call even when he is on another call. Is there a way so that a busy agent (whether busy because of an incoming call, or outgoing call) is not presented another ACD call? Thanks, -- jt ___ --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] What would cause a high memory usage in pbx_spool.c ?
Hi, After only 4 days I have 107472352 bytes in 46007 allocations in file 'pbx_spool.c' asterisk*CLI show memory summary 180 bytes in 2 allocations in file 'netsock.c' 12 bytes in 1 allocations in file 'devicestate.c' 2268 bytes in 1 allocations in file 'jitterbuf.c' 8160 bytes in 1 allocations in file 'localtime.c' 1232 bytes in 1 allocations in file 'app_queue.c' 1 bytes in 1 allocations in file 'res_features.c' 132376 bytes in 2 allocations in file 'res_musiconhold.c' 3672 bytes in18 allocations in file 'file.c' 1032 bytes in 2 allocations in file 'enum.c' 378 bytes in 2 allocations in file 'iax2-parser.c' 752 bytes in 2 allocations in file 'res_crypto.c' 21 bytes in 1 allocations in file 'cli.c' 1348 bytes in 4 allocations in file 'cdr.c' 12984 bytes in19 allocations in file 'chan_iax2.c' 960 bytes in40 allocations in file 'manager.c' 17647 bytes in18 allocations in file 'app_voicemail.c' 1456 bytes in 2 allocations in file 'dsp.c' 48456 bytes in 3 allocations in file 'frame.c' 2203 bytes in15 allocations in file 'channel.c' 42224 bytes in 8 allocations in file 'rtp.c' 150040 bytes in86 allocations in file 'chan_sip.c' 230 bytes in 9 allocations in file 'chanvars.c' 25736 bytes in19 allocations in file 'io.c' 4736 bytes in 177 allocations in file 'asterisk.c' 4872 bytes in 144 allocations in file 'sched.c' 513542 bytes in 124 allocations in file 'chan_zap.c' 11907 bytes in 403 allocations in file 'logger.c' 40320 bytes in 140 allocations in file 'loader.c' 4262 bytes in 265 allocations in file 'res_indications.c' 50839 bytes in 850 allocations in file 'pbx.c' 7653 bytes in 585 allocations in file 'pbx_config.c' 2345 bytes in 147 allocations in file 'app_dial.c' 107472352 bytes in 46007 allocations in file 'pbx_spool.c' 5218074 bytes in 110920 allocations in file 'config.c' 113784270 bytes allocated 160019 units total And top shows me that asterisk is using 153m virtual memory and 141 res memory... Only completely stopping asterisk and starting again will free up the memory. What could be causing this? I did read something about call files and variables in the past, but that was a long time ago. I do have to process .call files but not that many... Could there be a memory leak in pbx_spool.c, how can I help troubleshoot that? Thanks Walter. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM400 not working -- SOLVED
Sorry, I was pretty busy and did not work on my *. Problem was in zaptel not properly registering driver with udev. Manually updating udev rules fixed the problem. Thanks, Rudolf - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, October 11, 2005 6:20 AM Subject: Re: [Asterisk-Users] TDM400 not working On Mon, Oct 10, 2005 at 07:25:59PM +1000, Rudolf Ladyzhenskii wrote: Hi, all I have installed TDM400 card. I can see it is there (lspci). But Asterisk does not find it. phonebox2*CLI zap show status No Zaptel interface found. I assume that driver is not loaded, but I am sure I have installed it (I compiled zaptel and then re-build asterisk and did make install for both zaptel and asterisk). When asterisk is started I get: Jan 2 06:28:08 WARNING[3473]: chan_zap.c:872 zt_open: Unable to open '/dev/zap/channel': No such file or directory The device file does not exast Jan 2 06:28:08 ERROR[3473]: chan_zap.c:6572 mkintf: Unable to open channel 2: No such file or directory here = 0, tmp-channel = 2, channel = 2 Jan 2 06:28:08 ERROR[3473]: chan_zap.c:9927 setup_zap: Unable to register channel '2' Jan 2 06:28:08 WARNING[3473]: loader.c:402 __load_resource: chan_zap.so: load_module failed, returning -1 Jan 2 06:28:08 WARNING[3473]: loader.c:523 load_modules: Loading module chan_zap.so failed! Ok, I look in the /dev and I could not find /dev/zap at all! But, there is a /dev/zapchannel character device. Is that a typo? It should be /dev/zap/channel . Do you use udev? If so, see README.udev . If not: you need to generate those device files. Anyway: could you please post the output of: lsmod | grep zaptel Any ideas what can be wrong? And last question. Does zaptel driver reads configuration file on startup? If so, how do I force the driver to update if config file was changed? ztcfg loads the configuration to the zaptel module from /etc/zaptel.conf . -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What would cause a high memory usage in pbx_spool.c ?
Walter Klomp wrote: Hi, After only 4 days I have 107472352 bytes in 46007 allocations in file 'pbx_spool.c' I seem to recall a memory leak in pbx_spool being fixed a few days ago. check the asterisk-cvs mailing list archive on lists.digium.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] You ASKED for an Asterisk book, you GOT an Asterisk book!
Hello Leif , The appendices A B are missing from the zip file available at the location mentioned below . Is there some reason of copyright that is not mentioned here ? Tia , JimL On Sat, 15 Oct 2005, Leif Madsen wrote: Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons license, allowing the book in its entirity to be freely distributed. Asterisk: The Future of Telephony is now freely available, for download in PDF form, from the Asterisk Documentation Project website located at http://www.asteriskdocs.org. On the left hand side, click on Read the book online! for a copy. The authors would like to thank O'Reilly Media for having the vision to understand how significant it is for the Asterisk community to have a book freely available, thereby lowering the barrier of entry for those new to Asterisk, and to give back to a project that has given us all so much. I would personally like to thank Jared Smith, Jim van Meggelen, Michael Loukides (our editor) and the entire O'Reilly Media staff. The book is currently shipping, and should be available at all major book stores in paperback, and also online from http://www.oreilly.com/catalog/asterisk/ and other online outlets. Thanks, and we hope you enjoy reading it as much as we enjoyed writing it! PS: If the Asterisk Documentation Project website becomes slow due to the number of people accessing it at once, we appoligize and appreciate your patience. For those of you who are able to obtain the full copy, please consider helping us out by creating mirrors and torrents and posting them to the list by replying to this thread. Thanks! -- Leif Madsen - http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --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 -- +--+ | James W. Laferriere | SystemTechniques | Give me VMS | | NetworkEngineer | 3542 Broken Yoke Dr. | Give me Linux | | [EMAIL PROTECTED] | Billings , MT. 59105 | only on AXP | +--+ ___ --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] You ASKED for an Asterisk book, you GOT an Asterisk book!
Leif Madsen a écrit : Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons license, allowing the book in its entirity to be freely distributed. Kudos for this. Brilliant! ___ --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
