[asterisk-users] Friday Dec 19th at Noon ET: Jazinga pbx appliance
Hi all, Get your questions ready as tomorrow's VUC call will feature Shidan Gouran, CTO of Jazinga, makers of a new Asterisk appliance. Jazinga have developed a web 2.0 GUI for their embedded Asterisk appliance. We all love GUIs, right? They want to make it easy for a non-techie to setup a small office Asterisk solution. For details about the Jazinga product you can see Michael Graves' review at: http://www.smallnetbuilder.com/content/view/30660/80/ Be certain to look at the slide show of GUI screen shots: http://www.smallnetbuilder.com/content/view/30661/217/ Conference details, as usual: Info: http://VoipUsersConference.org Follow IRC.freenode.net #voip-users-conference SIP: talks...@vuc.onsip.com PSTN: (724) 444-7444 When connected, enter 22622# 1# (or your PIN in the place of the 1) ___ -- 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] Conference with an AGI inside Queue for password change
2008/12/19 Rajkumar S rajkum...@gmail.com Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password of caller. One possible solution to this is for the agent to call an agi into conference with the call after caller has been verified. The agi will prompt for the password which the caller will type in his keypad. Although the agent will hear the password prompt, he cannot overhear the DTMF digits typed by caller. Can this be implemented in asterisk? I have looked but did not find any hints. Is there a better solution to the problem I am having? Thanks for reading and any replies. raj ___ -- 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 maybe a simpler solution is set some variables to the caller channel trasfer to extencion where asterisk ask for the password put it in the data base and then transfer back to the agent. this is not so dificult to implement. you can use the mysql function or you can make a webservice and use CURL where you just put a url whit all the info. the variables in the caller channel are for tell asterisk where tos end the call back and the caller user to use in the mysql or webservices. David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Conference with an AGI inside Queue for password change
maybe a simpler solution is set some variables to the caller channel trasfer to extencion where asterisk ask for the password put it in the data base and then transfer back to the agent. this is not so dificult to implement. you can use the mysql function or you can make a webservice and use CURL where you just put a url whit all the info. the variables in the caller channel are for tell asterisk where tos end the call back and the caller user to use in the mysql or webservices. David 2008/12/19 Rajkumar S rajkum...@gmail.com Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password of caller. One possible solution to this is for the agent to call an agi into conference with the call after caller has been verified. The agi will prompt for the password which the caller will type in his keypad. Although the agent will hear the password prompt, he cannot overhear the DTMF digits typed by caller. Can this be implemented in asterisk? I have looked but did not find any hints. Is there a better solution to the problem I am having? Thanks for reading and any replies. raj ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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
[asterisk-users] realtime queue change ring strategy
Hello, I'm using asterisk 1.6.0.1 and realtime queues. But when I make changes in database (for example: change strategy from ringall to random), but asterisk shows old strategy, doesn't update this parameter. My question is, how I can dynamically change ring strategy. Thanks in advance. -- Pagarbiai / Best Regards, Giedrius Augys ___ -- 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
[asterisk-users] Asterisk 1.6.1-beta4 released
The Asterisk.org development team has created the fourth beta release for Asterisk 1.6.1. 1.6.1-beta4 is available for immediate download from http://downloads.digium.com/. This beta release contains fixes for multiple issues since 1.6.1-beta3 including crashes and a problem in chan_sip that would cause incorrect user and peer matching. For a full list of the changes in this release, please see the ChangeLog: http://svn.digium.com/view/asterisk/tags/1.6.1-beta4/ChangeLog?view=markup Thank you for your continued support of Asterisk! ___ -- 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
[asterisk-users] Dynamic Feature Playback acting on *both* channels?
I'd like to be able to playback a file to *both* channels in a call as a result of a DTMF feature. Can anyone suggest how I might do this? I thought of using a DYNAMIC_FEATURE to call a macro that starts a dynamic meetme but the macro only gets to control the 'caller' or 'callee' :-( Failing that I'm trying to provide a simple means of playing back a recorded message during a phone call controled by someone's phone (the actual message will have been pre-selected by an external app). Any suggestions? -- Regards, Russell | Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ -- 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] Increase DTMF Tone Duration
I suspect you have a DTMF mode mismatch. If Asterisk is expecting RFC2833 or INFO DTMF and the phones are sending inband DTMF then Asterisk won't detect it and won't regenerate the DTMF (and so toneduration would have no effect). Unfortunately this is not the case. The issue only happens with certain IVRs and we have a analyer in the PRI link that clearly shows the correct DTMF tones being sent. The problem is simply the duration is too short (120ms), and the remote IVR seems to not detect them when they are that short. If I press the digits longer (180ms), it works fine. In Asterisk 1.4 and later there are some DTMF debug options, as well as SIP and RTP debug options. You should start out by making sure that Asterisk is detecting the tones as DTMF and not simply passing the raw audio thru. I don't know what specific DTMF and RTP debug commands are in 1.4+ (you should be able to look them up in the CLI), as my customers have chosen to skip 1.4 and go directly to 1.6 once they have become comfortable with it. Thats right. The CLI (when you have dtmf defined in logger.conf), will also show the correct DTMF tones being detected and sent. Good luck with this. DTMF issues can be hell to diagnose and fix sometimes. I went into the code on channel.c and added a 'ast_safe_sleep(chan,100);' before the tone was being ended and it worked like a charm. Now we can send DTMF tones to those IVRs and they are working 100% of the time. Thanks, Andres, http://www.telesip.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 ___ -- 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] Authorize Microsoft SQL
Greg's question is this: - Does anybody has a sample on how to open and query a Microsoft SQL database from the dialplan?(and which are the correct drivers/addons to install?) Thanks CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Wofford Sent: Thursday, December 18, 2008 11:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL There is some code somewhere on the Asterisk/Linux box getting the SQL data, be it a program, script or batch file. There is something initiating the T-SQL code... SELECT * FROM supportcases WHERE id = 123456789 This code comes from the client, not the server. The Asterisk box will have the database drivers (ODBC...), but that just allows a connection, there is something that tells the server to return data (via the query). You are going to have to write the script (middleman) and pass it on from SQL to Asterisk. I don't know of anything like this ready-made. 1. DialPlan collect @number from caller 2. Call script, program etc and use the @number as a parameter 3. The script, program etc will the create the SQL Query to query the database: SELECT COUNT(*) FROM supportcases WHERE @number = 123456789 4. The script, program etc will then get the number of rows returned, hopefully 1 or 0 and assign it as a variable. 5. Your script, program etc with then use the following logic: If @variable = 0 Then Play enter your case again Voice Prompt ElseIf @variable = 1 Then Connect to Agent... HTH, Steve Wofford www.uctrlit.com P.(949)743-0233 Ext. 200 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gregory Malsack Sent: Thursday, December 18, 2008 20:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL Steve, my friends setup does not utilize perl/php code. His communication is directly between asterisk and mysql, there is no middle man. This is what I was hoping for with ms sql. But it doesn't sound like that will be the case. Thanks for everything! Greg -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Wofford Sent: Thursday, December 18, 2008 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL This is exactly what you need. Get your friends perl/php script and the SQL code will be near identical, or at least you will have no problem changing it yourself even if you don't know SQL. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gregory Malsack Sent: Thursday, December 18, 2008 20:13 To: f...@teamforrest.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL This much I already know. This information is easily found through a simple google search. What I'm looking for is if anyone knows what a dialplan would look like that would perform an ODBC query to an ODBC database. I've seen minuet documentation on ODBCget, which is what I'm thinking will do the trick, but as I said the documentation on this is so vague that I'm not quite understanding it. There's also the possibility that there is another option here that I'm not seeing. One idea Steve gave me, was to create a perl/php script that does the query and returns a result code. Basically acting like a middle man between asterisk and the MS SQL database. Greg -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred Posner Sent: Thursday, December 18, 2008 9:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL All you need is odbc and freetds. Then it will integrate very smoothly. Fred Posner f...@teamforrest.com Direct: +1 (503) 914-0999 -Original Message- From: Steve Wofford s...@uctrlit.com Date: Thu, 18 Dec 2008 19:46:36 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Authorize Microsoft SQL ___ -- 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 ___ -- 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 No virus
[asterisk-users] Cut Through DTMF caller ID on SIP phone
Hi Setup : Asterisk 1.6 on Fedora Core 9 with TE410P.. 1. I;ve noticed that whenever during background(menu-filename) method - i try to press any key for selection like 1 for some prompt, 2 for another prompt etc...Asterisk takes a while before it takes me to the respective option..Is that normal behaviour ? by the time the caller waits to listen to the appropriate prompt on selecting 1 - he thinks nothing is happening for 2-3 seconds .. fyi, I used to use Trixbox prev. and didnt find any such problem ... 2. Is there any way to block the caller id from appearing on the SIP Phone ? my trunk is E1 PRI while i used softphones internally - i dont want my agents to see the caller id - is their any way to block caller ids from appearing on softphones ? Thanks Sriram ___ -- 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] Authorize Microsoft SQL
This isn't what you're specifically looking for, but if you get an odbc connection to the database, you can use that logic to do this. Try a google on pgsql odbc connection asterisk. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Friday, December 19, 2008 10:51 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Authorize Microsoft SQL Greg's question is this: - Does anybody has a sample on how to open and query a Microsoft SQL database from the dialplan?(and which are the correct drivers/addons to install?) Thanks CS -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Wofford Sent: Thursday, December 18, 2008 11:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL There is some code somewhere on the Asterisk/Linux box getting the SQL data, be it a program, script or batch file. There is something initiating the T-SQL code... SELECT * FROM supportcases WHERE id = 123456789 This code comes from the client, not the server. The Asterisk box will have the database drivers (ODBC...), but that just allows a connection, there is something that tells the server to return data (via the query). You are going to have to write the script (middleman) and pass it on from SQL to Asterisk. I don't know of anything like this ready-made. 1. DialPlan collect @number from caller 2. Call script, program etc and use the @number as a parameter 3. The script, program etc will the create the SQL Query to query the database: SELECT COUNT(*) FROM supportcases WHERE @number = 123456789 4. The script, program etc will then get the number of rows returned, hopefully 1 or 0 and assign it as a variable. 5. Your script, program etc with then use the following logic: If @variable = 0 Then Play enter your case again Voice Prompt ElseIf @variable = 1 Then Connect to Agent... HTH, Steve Wofford www.uctrlit.com P.(949)743-0233 Ext. 200 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gregory Malsack Sent: Thursday, December 18, 2008 20:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL Steve, my friends setup does not utilize perl/php code. His communication is directly between asterisk and mysql, there is no middle man. This is what I was hoping for with ms sql. But it doesn't sound like that will be the case. Thanks for everything! Greg -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Wofford Sent: Thursday, December 18, 2008 10:21 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL This is exactly what you need. Get your friends perl/php script and the SQL code will be near identical, or at least you will have no problem changing it yourself even if you don't know SQL. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gregory Malsack Sent: Thursday, December 18, 2008 20:13 To: f...@teamforrest.com; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL This much I already know. This information is easily found through a simple google search. What I'm looking for is if anyone knows what a dialplan would look like that would perform an ODBC query to an ODBC database. I've seen minuet documentation on ODBCget, which is what I'm thinking will do the trick, but as I said the documentation on this is so vague that I'm not quite understanding it. There's also the possibility that there is another option here that I'm not seeing. One idea Steve gave me, was to create a perl/php script that does the query and returns a result code. Basically acting like a middle man between asterisk and the MS SQL database. Greg -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Fred Posner Sent: Thursday, December 18, 2008 9:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Authorize Microsoft SQL All you need is odbc and freetds. Then it will integrate very smoothly. Fred Posner f...@teamforrest.com Direct: +1 (503) 914-0999 -Original Message- From: Steve Wofford s...@uctrlit.com Date: Thu, 18 Dec 2008 19:46:36 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Authorize Microsoft SQL
Re: [asterisk-users] Cut Through DTMF caller ID on SIP phon
set(CALLERID(number)=000) David 2008/12/19 Sriram d_r_sri...@hotmail.com Hi Setup : Asterisk 1.6 on Fedora Core 9 with TE410P.. 1. I;ve noticed that whenever during background(menu-filename) method - i try to press any key for selection like 1 for some prompt, 2 for another prompt etc...Asterisk takes a while before it takes me to the respective option..Is that normal behaviour ? by the time the caller waits to listen to the appropriate prompt on selecting 1 - he thinks nothing is happening for 2-3 seconds .. fyi, I used to use Trixbox prev. and didnt find any such problem ... 2. Is there any way to block the caller id from appearing on the SIP Phone ? my trunk is E1 PRI while i used softphones internally - i dont want my agents to see the caller id - is their any way to block caller ids from appearing on softphones ? Thanks Sriram ___ -- 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 set(CALLERID(number)=000) David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. ___ -- 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] Application Layer Gateway for SIP protocol
Asterisk certainly supports NAT traversal, and is SIP-aware. However, Asterisk is a user agent; it can act as a SIP endpoint, not as an ALG. Olfa Echi wrote: Hello everybody, I want to know if Asterisk can provide any solution to perform NAT traversal for SIP protocol which means that it implements the functions of an ALG (Application Layer Gateway). Thanks. ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Increase DTMF Tone Duration
The problem is simply the duration is too short (120ms), and the remote IVR seems to not detect them That sounds like an IVR issue. I've worked on some traditional PABXs and even designed some DTMF receivers. Any decent DTMF receiver should be able to reliably decode 80 ms tones, and a really good one can decode 40 ms. 120 ms should be a very generous duration. I shipped 80 ms duration to COs 20 years ago. Wilton ___ -- 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] Cut Through DTMF caller ID on SIP phone
Sriram wrote: Setup : Asterisk 1.6 on Fedora Core 9 with TE410P.. 1. I;ve noticed that whenever during background(menu-filename) method - i try to press any key for selection like 1 for some prompt, 2 for another prompt etc...Asterisk takes a while before it takes me to the respective option..Is that normal behaviour ? by the time the caller waits to listen to the appropriate prompt on selecting 1 - he thinks nothing is happening for 2-3 seconds .. fyi, I used to use Trixbox prev. and didnt find any such problem ... This typically happens when you have overlapping extensions. i.e. a menu option 1 and extensions starting with 1. Don't forget to look in include'd contexts, and don't forget wildcards. ___ -- 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] Application Layer Gateway for SIP protocol
On Fri, Dec 19, 2008 at 2:56 AM, Olfa Echi olfae...@yahoo.fr wrote: Hello everybody, I want to know if Asterisk can provide any solution to perform NAT traversal for SIP protocol which means that it implements the functions of an ALG (Application Layer Gateway). Thanks. Look at the SIP connection tracking module for iptables/netfilter: http://www.calivia.com/iptables-sip-conntrack-nat -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com ___ -- 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
[asterisk-users] OpenSer and MYSQL Lookup Queries!
Hi! Can OpenSer perform some database lookup queries based on dialed number like we can do with Asterisk. Asterisk Can do it and there is MYSQL Function available which allow us to open connection and execute any query to get required results from database, Can we do same with OpenSer or OpenSIP etc.? Thanks Regards, Muhammad Zulqarnain ___ -- 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] Increase DTMF Tone Duration
Wilton Helm wrote: The problem is simply the duration is too short (120ms), and the remote IVR seems to not detect them That sounds like an IVR issue. I've worked on some traditional PABXs and even designed some DTMF receivers. Any decent DTMF receiver should be able to reliably decode 80 ms tones, and a really good one can decode 40 ms. 120 ms should be a very generous duration. I shipped 80 ms duration to COs 20 years ago. If you shipped detectors requiring an 80ms burst of tone to a telco 20 years ago, they would have sent it back. They have never accepted more than a 50ms minimum tone burst, and many demand detection with just a 40ms tone burst. Demanding 120ms is crazy. People just don't hold down the keys that long. You'd have horrible failure rates. Regards, Steve ___ -- 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
[asterisk-users] Pre-routing manipulation of calls
This is concerning an Asterisk 1.4.18 server. We have approximately 70 DID numbers. Incoming calls are placed into the incoming context (by zapata.conf) and are routed based on the dialed number. I want to do some manipulation (CallerID name override) to all incoming calls before they are routed. I would prefer to avoid duplicating the necessary code in each DID extension stanza, even if it's just a call to a macro. 1. Can I set up a catch-all extension in incoming, do my processing, and then have the calls fall through to the existing extension stanzas? 2. Or, should I use a separate pre-incoming context to do the manipulation and then jump to the real incoming context containing the specific extension stanzas? 3. Or, is there another method that would be more elegant? Thanks. -- Kevin DeGraaf ___ -- 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] Pre-routing manipulation of calls
#1 is your best bet. Use Goto(). On Dec 19, 2008, at 14:03, Kevin DeGraaf ke...@kdegraaf.net wrote: This is concerning an Asterisk 1.4.18 server. We have approximately 70 DID numbers. Incoming calls are placed into the incoming context (by zapata.conf) and are routed based on the dialed number. I want to do some manipulation (CallerID name override) to all incoming calls before they are routed. I would prefer to avoid duplicating the necessary code in each DID extension stanza, even if it's just a call to a macro. 1. Can I set up a catch-all extension in incoming, do my processing, and then have the calls fall through to the existing extension stanzas? 2. Or, should I use a separate pre-incoming context to do the manipulation and then jump to the real incoming context containing the specific extension stanzas? 3. Or, is there another method that would be more elegant? Thanks. -- Kevin DeGraaf ___ -- 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 ___ -- 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] Pre-routing manipulation of calls
Kevin DeGraaf wrote: a macro. 1. Can I set up a catch-all extension in incoming, do my processing, and then have the calls fall through to the existing extension stanzas? I use number 1 with a Gosub(get_name,s,1) It jumps to a mysql lookup against the number and sets the name and continues on. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Pre-routing manipulation of calls
I use number 1 with a Gosub(get_name,s,1)
It jumps to a mysql lookup against the number and sets the name and
continues on.
Based on the ambiguity of the documentation with respect to extension
sorting order [0], I ended up going with the pre-incoming context
idea. It worked fine.
[pre-incoming]
exten =
_X.,1,Set(CALLERID(name)=${IF($[${DB(cidname/${CALLERID(num)})} = ]
?${CALLERID(name)}:${DB(cidname/${CALLERID(num)})})})
exten = _X.,n,Goto(incoming,${EXTEN},1)
By the way, I'm using the AstDB for CallerID overrides, which seems like
it would be more reliable than using an external database. Is there
some advantage (e.g. scalability) to using MySQL?
Thanks.
[0] http://tinyurl.com/3jn62a
--
Kevin DeGraaf
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[asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem
I am experiencing a 606 not Acceptable error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. I tried putting stunaddr=stun.ekiga.net into the sip.conf file under [ekiga]. I also tried externip=71.xxx.xxx.xxx as shown below. Neither works. My junction client set up is working behind the same firewall. Can anybody suggest a fix? Thanks, Bud Roth P.S. Debug output and sip.conf file pasted below: P.P.S. I posted this problem once before, but it looked like I forwarded it from another list/person. Actually, I have multiple emails and sent it from the wrong address the first time. I didn't get any responses, so thought I'd try again. Any help would be greatly appreciated. Sip set debug output: --- SIP read from 86.64.162.35:5060 --- SIP/2.0 606 Not Acceptable Via: SIP/2.0/UDP 10.1.1.40:5060;branch=z9hG4bK2f1127c2;rport=9288;received=71.178.235.241 From: sip:budzh...@ekiga.net;tag=as7506851f To: sip:budzh...@ekiga.net;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.ddfa Call-ID: 7ed69d2f12f88039420ab10a32604...@127.0.0.1 CSeq: 124 REGISTER Server: Kamailio (1.4.0-notls (i386/linux)) Content-Length: 0 --- --- victoria*CLI --- SIP read from 86.64.162.35:5060 --- SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP 10.1.1.40:5060;branch=z9hG4bK176cc789;rport=9288;received=71.178.235.241 From: asterisk sip:aster...@10.1.1.40;tag=as183adb5e To: sip:ekiga.net;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.d07f Call-ID: 43e457266aaf90c115d37a47156c1...@10.1.1.40 CSeq: 102 OPTIONS Server: Kamailio (1.4.0-notls (i386/linux)) Content-Length: 0 - relevant portions of sip.conf: [general] context=default srvlookup=yes videosupport=yes echocancelwhenbridged=yes dtmfmode=rfc2833 disallow=all ; First disallow all codecs allow=ulaw allow=alaw ; Allow codecs in order of allow=ilbc ; preference allow=gsm allow=h261 register = budzhaus:secreth...@ekiga.net register = obitori:secreth...@jnctn.net SNIP [jnctn] type=peer host=sip.jnctn.net username=obitori secret=SECRETHERE fromdomain=jnctn.net insecure=very nat=yes qualify=yes context=incoming [101] type=friend secret=SECRETHERE qualify=yes; Qualify peer is not more than 2000 mS away nat=no ; This phone is not natted host=dynamic ; This device registers with us canreinvite=yes ; Asterisk by default tries to redirect context=home ;in/outgoing to ekiga.net [ekiga] type=friend username=budzhaus secret=SECRETHERE host=ekiga.net canreinvite=no qualify=yes insecure=very fromdomain=ekiga.net nat=yes context=incoming externip=71.xxx.xxx.xxx localnet=10.1.1.0/255.255.255.0 - Bud Roth Lake Barcroft, VA public PGP key available at: https://www.cacert.org/gpg.php?id=3cert=9272 For Northern Virginia's best Judo, visit: http://sportjudo.org ___ -- 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] Pre-routing manipulation of calls
Kevin DeGraaf wrote: By the way, I'm using the AstDB for CallerID overrides, which seems like it would be more reliable than using an external database. Is there some advantage (e.g. scalability) to using MySQL? I manage 5 systems. Each has a slave database against the master Mysql. Each systems basic features are hard coded in the dial plan. If any of the Mysqls fail, it will not prevent the phone systems from functioning. All the 'fluff' is in the databases. Caller-ID name/number matches Black listed numbers Fax2Email Conferencing Access restrictions for after hours. Makes it easier to administrate, especially for those in my group that don't have experience with *nix. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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] Install app_rxfax and app_txfax in 1.4 with Lenny
Hi, I tried agx-addons with different version. I got it working till asterisk version 1.4.21 included on ubuntu with libtiff4. Starting from asterisk 1.4.22 it did not longer work. Loic On Wed, 2008-12-17 at 17:12 +0100, Olivier wrote: Hi, I've read README file in agx-ast-addons-1.4.17.5.tar.bz2 It says Install libTiff =3.8 and 4.0 Should you really use this agx-ast-addons to get app_rxfax and app-_txfax running with latest 1.4.22 ? If positive, should you take this libtiff warning into account ? If positive, where can you find such libtiff version as Debian repository (I didn't check alternate distrib) includes libtiff4 but no libtiff3 not libtiff. Cheers ___ -- 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 -- Loïc DIDELOT MIXvoip S.a. Tel: +352 20 20 Fax: +352 20 90 ldide...@mixvoip.com http://www.mixvoip.com ___ -- 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] Problem configuring Asterisk as client for ekiga.net -- NAT problem
obitori junk wrote: I am experiencing a 606 not Acceptable error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. In my experience Not Acceptable errors happen because the two endpoints cannot agree on a codec. Try allowing all the codecs in your softphone and in Asterisk sip.conf [general] do a disallow=all and an allow=ulaw.I suggest you do this in [general] when testing because it can sometimes be hard to make sure that a peer/friend/user entry is actually matching the incoming call. Once you get it working you can refine it. You could be having a NAT issue too, but I don't think it is related to your 606 error. ___ -- 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] Install app_rxfax and app_txfax in 1.4 with Lenny
On Fri, Dec 19, 2008 at 09:32:57PM +0100, Loic Didelot wrote: Hi, I tried agx-addons with different version. I got it working till asterisk version 1.4.21 included on ubuntu with libtiff4. Starting from asterisk 1.4.22 it did not longer work. Just updated my backport. Originally intended to be in a Debian package but now I see that it won't make it. A patch vs. recent apps/app_fax.c (from 1.6.0) http://svn.debian.org/viewsvn/pkg-voip/asterisk-spandsp-plugins/trunk/debian/patches/app_fax_14?rev=6561view=log app_fax.c could be found oon the same area. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- 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] Increase DTMF Tone Duration
Steve Underwood wrote: Wilton Helm wrote: The problem is simply the duration is too short (120ms), and the remote IVR seems to not detect them That sounds like an IVR issue. I've worked on some traditional PABXs and even designed some DTMF receivers. Any decent DTMF receiver should be able to reliably decode 80 ms tones, and a really good one can decode 40 ms. 120 ms should be a very generous duration. I shipped 80 ms duration to COs 20 years ago. Demanding 120ms is crazy. People just don't hold down the keys that long. You'd have horrible failure rates. Regards, Steve You can edit the duration of the tones in app_senddtmf.c and then rebuild asterisk. -- Thank you and have any kind of day you want, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 v...@rockynet.com ___ -- 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] Problem configuring Asterisk as client for ekiga.net -- NAT problem
I tried both disallow all /// allow ulaw and allow all. Neither worked. Ekiga.net is not accepting asterisk as a client. I keep getting the 600 not acceptable error. What seems odd to me is that the NAT column shows an n for both ekiga.net and jnctn.net, but I have nat=yes for both. Thanks for the suggestion...I'm still open to ideas. Regards, Bud Roth sip.conf contains: allow all OUTPUT-- Name/username HostDyn Nat ACL Port Status ekiga/budzhaus 86.64.162.35 N 5060 OK (102 ms) 110(Unspecified)D 0 UNKNOWN 109(Unspecified)D 0 UNKNOWN 108(Unspecified)D 0 UNKNOWN 107(Unspecified)D 0 UNKNOWN 106(Unspecified)D 0 UNKNOWN 105(Unspecified)D 0 UNKNOWN 104(Unspecified)D 0 UNKNOWN 103(Unspecified)D 0 UNKNOWN 102/10210.1.1.20D 5061 OK (1 ms) 101(Unspecified)D 0 UNKNOWN jnctn/obitori 66.227.100.20N 5060 OK (22 ms) 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 offline] [Dec 19 17:20:31] NOTICE[2478]: chan_sip.c:7517 sip_reg_timeout:-- Registration for 'budzh...@ekiga.net' timed out, trying again (Attempt #2) -- Got SIP response 606 Not Acceptable back from 86.64.162.35 sip.conf contains: disallow all allow ulaw OUTPUT-- victoria*CLI sip show peers Name/username HostDyn Nat ACL Port Status ekiga/budzhaus 86.64.162.35 N 5060 OK (97 ms) 110(Unspecified)D 0 UNKNOWN 109(Unspecified)D 0 UNKNOWN 108(Unspecified)D 0 UNKNOWN 107(Unspecified)D 0 UNKNOWN 106(Unspecified)D 0 UNKNOWN 105(Unspecified)D 0 UNKNOWN 104(Unspecified)D 0 UNKNOWN 103(Unspecified)D 0 UNKNOWN 102/10210.1.1.20D 5061 OK (1 ms) 101(Unspecified)D 0 UNKNOWN jnctn/obitori 66.227.100.20N 5060 OK (23 ms) 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 offline] victoria*CLI On Fri, 2008-12-19 at 15:18 -0600, Eric ManxPower Wieling wrote: obitori junk wrote: I am experiencing a 606 not Acceptable error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. In my experience Not Acceptable errors happen because the two endpoints cannot agree on a codec. Try allowing all the codecs in your softphone and in Asterisk sip.conf [general] do a disallow=all and an allow=ulaw.I suggest you do this in [general] when testing because it can sometimes be hard to make sure that a peer/friend/user entry is actually matching the incoming call. Once you get it working you can refine it. You could be having a NAT issue too, but I don't think it is related to your 606 error. ___ -- 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 ___ -- 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] Authorize Microsoft SQL
I'm doing something similar to validate employees for DISA access. I
built Asterisk with ODBC support by installing unixODBC and FreeTDS
before I built Asterisk. I have a couple of stored procedures on the MS
SQL box that do the heavy lifting and hide the database details from the
Asterisk system. Really, the backend could be any ODBC compliant
datasource that supports stored procs. (I use the stored procedure to
expose a consistent interface regardless of the database schema behind
it)
Here is the relevant portion of my dialplan: (You can also see I use
ODBC to push CDR records back to the database for logging purposes)
exten = s,1,NoOp() ; Validate the employee's id number. Give
them MAX_ID_TRIES to get it right.
exten = s,n,Set(TIMEOUT(digit)=5)
exten = s,n,Set(TIMEOUT(response)=10)
exten = s,n,Set(ID_TRIES=0) ; Set the max number of login attempts
exten = s,n,Set(MAX_ID_TRIES=3)
exten = s,n(get_id),NoOp()
exten = s,n,Set(ID_TRIES=$[${ID_TRIES} + 1])
exten =
s,n,Read(ID_ENTERED,/var/lib/asterisk/sounds/custom/disa_greet1,5)
exten = s,n,Set(ID_RESULT=${ODBC_INFO(ClockID,${ID_ENTERED})})
exten = s,n,GotoIf($[${ISNULL(${ID_RESULT})}]?:valid_id,1)
exten = s,n,Playback(/var/lib/asterisk/sounds/custom/disa_badempnum)
exten = s,n,GotoIf($[${ID_TRIES}
${MAX_ID_TRIES}]?get_id:login_fail,1)
exten = valid_id,1,NoOp() ; Validate the employee's pin number. Give
them MAX_PIN_TRIES to get it right.
exten = valid_id,n,Set(PIN_TRIES=0) ; Set the max number of login
attempts
exten = valid_id,n,Set(MAX_PIN_TRIES=3)
exten = valid_id,n(get_pin),NoOp()
exten = valid_id,n,Set(PIN_TRIES=$[${PIN_TRIES} + 1])
exten =
valid_id,n,Read(PIN_ENTERED,/var/lib/asterisk/sounds/custom/disa_greet2,
4)
exten =
valid_id,n,Set(PIN_RESULT=${ODBC_PIN(ClockID,${ID_ENTERED},${PIN_ENTERED
})})
exten = valid_id,n,GotoIf($[${ISNULL(${PIN_RESULT})}]?:valid_login,1)
exten =
valid_id,n,Playback(/var/lib/asterisk/sounds/custom/disa_badpincode)
exten = valid_id,n,GotoIf($[${PIN_TRIES}
${MAX_PIN_TRIES}]?get_pin:login_fail,1)
exten = login_fail,1,NoOp() ; They suck. They couldn't get either the
pin number or the emp id right.
exten =
login_fail,n,Playback(/var/lib/asterisk/sounds/custom/disa_faillogin)
exten = login_fail,n,Hangup()
exten = valid_login,1,NoOp()
exten = valid_login,n,Set(CALLDATE=${STRFTIME(${EPOCH},GMT+5,%x %X)})
exten = valid_login,n,Set(CLID=${CALLERID(num)})
exten = valid_login,n,Set(UNID=${CDR(uniqueid)})
exten = valid_login,n,Set(DBINS =
${ODBC_DISA(${CALLDATE},${CLID},${ID_ENTERED},${UNID})})
exten =
valid_login,n,Playback(/var/lib/asterisk/sounds/custom/disa_greet3)
exten = valid_login,n,DISA(no-password,from-disa,CID Name
xx)
exten = valid_login,n(end),Goto(valid_login,s,1)
With unixODBC you need a couple of config files...
Here is my /etc/odbc.ini:
[OHSQL_ELABOR]
Driver = FreeTDS
Description = Connection to eLabor database on OHSQL - LIVE
Trace = No
Server = ohsql.ohio..xxx
Database= eLabor
Port= 1870
TDS_Version = 8.0
ReadOnly= Yes
[OHSQL_ASTERISK]
Driver = FreeTDS
Description = Connection to Asterisk Database
Trace = No
Server = ohsql.ohio.x.xxx
Database= Asterisk
Port= 1870
TDS_Version = 8.0
Here is my /etc/odbcinst.ini:
(The FileUsage=1 is important when working against MS SQL... the driver
doesn't support multiple connections)
[FreeTDS]
Description = FreeTDS Driver (MS-SQL access)
Driver = /usr/local/freetds/lib/libtdsodbc.so
Setup = /usr/local/freetds/lib/libtdsS.so
FileUsage = 1
Here is /etc/asterisk/func_odbc.conf
; We define two DSNs for database function access:
; - eLaborSQL which provides access the eLabor database
;(Could be testing or live... depends on res_odbc.conf)
; - AsteriskSQL which provides access to the Asterisk database
[INFO]
; This is a general grab statement to allow us to access any column in
the employee table
; by clock ID
dsn=eLaborSQL
read=SELECT ${ARG1} FROM Employee WHERE ClockID = ${ARG2} and Terminated
= 0
[PIN]
; This will return a given column based on the clock ID PIN passed in
dsn=eLaborSQL
read=SELECT ${ARG1} FROM Employee WHERE ClockID = ${ARG2} and PIN =
${ARG3} and Terminated = 0
[DISA]
;This will insert a new record into the DISA database to allow for cdr
match-ups
dsn=AsteriskSQL
read=INSERT INTO Asterisk_DISA (calldate, src, empID, uniqueid) VALUES
('${ARG1}','${ARG2}','${ARG3}','${ARG4}')
And finally... here is /etc/asterisk/res_odbc.conf
[eLaborSQL]
enabled = yes
dsn = OHSQL_ELABOR
pooling = yes
limit = 1
username = x
password = xx
pre-connect = yes
; Many databases have a default of '\' to escape special characters. MS
SQL
; Server does not.
backslash_is_escape = no
[AsteriskSQL]
enabled = yes
dsn = OHSQL_ASTERISK
pooling = yes
limit = 1
username =
Re: [asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem
Have you done a sip set debug, then sip reload? Do you have a range of 1-2 open in your firewall? Asterisk will poke out through 5060 but has to get a random response back in the 10-20K range (you can narrow this) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of obitori junk Sent: Friday, December 19, 2008 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem I tried both disallow all /// allow ulaw and allow all. Neither worked. Ekiga.net is not accepting asterisk as a client. I keep getting the 600 not acceptable error. What seems odd to me is that the NAT column shows an n for both ekiga.net and jnctn.net, but I have nat=yes for both. Thanks for the suggestion...I'm still open to ideas. Regards, Bud Roth sip.conf contains: allow all OUTPUT-- Name/username HostDyn Nat ACL Port Status ekiga/budzhaus 86.64.162.35 N 5060 OK (102 ms) 110(Unspecified)D 0 UNKNOWN 109(Unspecified)D 0 UNKNOWN 108(Unspecified)D 0 UNKNOWN 107(Unspecified)D 0 UNKNOWN 106(Unspecified)D 0 UNKNOWN 105(Unspecified)D 0 UNKNOWN 104(Unspecified)D 0 UNKNOWN 103(Unspecified)D 0 UNKNOWN 102/10210.1.1.20D 5061 OK (1 ms) 101(Unspecified)D 0 UNKNOWN jnctn/obitori 66.227.100.20N 5060 OK (22 ms) 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 offline] [Dec 19 17:20:31] NOTICE[2478]: chan_sip.c:7517 sip_reg_timeout:-- Registration for 'budzh...@ekiga.net' timed out, trying again (Attempt #2) -- Got SIP response 606 Not Acceptable back from 86.64.162.35 sip.conf contains: disallow all allow ulaw OUTPUT-- victoria*CLI sip show peers Name/username HostDyn Nat ACL Port Status ekiga/budzhaus 86.64.162.35 N 5060 OK (97 ms) 110(Unspecified)D 0 UNKNOWN 109(Unspecified)D 0 UNKNOWN 108(Unspecified)D 0 UNKNOWN 107(Unspecified)D 0 UNKNOWN 106(Unspecified)D 0 UNKNOWN 105(Unspecified)D 0 UNKNOWN 104(Unspecified)D 0 UNKNOWN 103(Unspecified)D 0 UNKNOWN 102/10210.1.1.20D 5061 OK (1 ms) 101(Unspecified)D 0 UNKNOWN jnctn/obitori 66.227.100.20N 5060 OK (23 ms) 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 offline] victoria*CLI On Fri, 2008-12-19 at 15:18 -0600, Eric ManxPower Wieling wrote: obitori junk wrote: I am experiencing a 606 not Acceptable error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. In my experience Not Acceptable errors happen because the two endpoints cannot agree on a codec. Try allowing all the codecs in your softphone and in Asterisk sip.conf [general] do a disallow=all and an allow=ulaw.I suggest you do this in [general] when testing because it can sometimes be hard to make sure that a peer/friend/user entry is actually matching the incoming call. Once you get it working you can refine it. You could be having a NAT issue too, but I don't think it is related to your 606 error. ___ -- 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 ___ -- 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] Problem configuring Asterisk as client for ekiga.net -- NAT problem
Danny: I restarted asterisk both times after making changes to the sip.conf in response to Eric's suggestions. I'll send you my original post with sip debug info. I ran sip debug again. The logged info is identical to what I was getting before I changed the codec allow/disallow settings... I think this is an Asterisk behind a NAT configuration problem, but if I could be wrong. Bud On Fri, 2008-12-19 at 16:38 -0600, Danny Nicholas wrote: Have you done a sip set debug, then sip reload? Do you have a range of 1-2 open in your firewall? Asterisk will poke out through 5060 but has to get a random response back in the 10-20K range (you can narrow this) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of obitori junk Sent: Friday, December 19, 2008 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem I tried both disallow all /// allow ulaw and allow all. Neither worked. Ekiga.net is not accepting asterisk as a client. I keep getting the 600 not acceptable error. What seems odd to me is that the NAT column shows an n for both ekiga.net and jnctn.net, but I have nat=yes for both. Thanks for the suggestion...I'm still open to ideas. Regards, Bud Roth sip.conf contains: allow all OUTPUT-- Name/username HostDyn Nat ACL Port Status ekiga/budzhaus 86.64.162.35 N 5060 OK (102 ms) 110(Unspecified)D 0 UNKNOWN 109(Unspecified)D 0 UNKNOWN 108(Unspecified)D 0 UNKNOWN 107(Unspecified)D 0 UNKNOWN 106(Unspecified)D 0 UNKNOWN 105(Unspecified)D 0 UNKNOWN 104(Unspecified)D 0 UNKNOWN 103(Unspecified)D 0 UNKNOWN 102/10210.1.1.20D 5061 OK (1 ms) 101(Unspecified)D 0 UNKNOWN jnctn/obitori 66.227.100.20N 5060 OK (22 ms) 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 offline] [Dec 19 17:20:31] NOTICE[2478]: chan_sip.c:7517 sip_reg_timeout:-- Registration for 'budzh...@ekiga.net' timed out, trying again (Attempt #2) -- Got SIP response 606 Not Acceptable back from 86.64.162.35 sip.conf contains: disallow all allow ulaw OUTPUT-- victoria*CLI sip show peers Name/username HostDyn Nat ACL Port Status ekiga/budzhaus 86.64.162.35 N 5060 OK (97 ms) 110(Unspecified)D 0 UNKNOWN 109(Unspecified)D 0 UNKNOWN 108(Unspecified)D 0 UNKNOWN 107(Unspecified)D 0 UNKNOWN 106(Unspecified)D 0 UNKNOWN 105(Unspecified)D 0 UNKNOWN 104(Unspecified)D 0 UNKNOWN 103(Unspecified)D 0 UNKNOWN 102/10210.1.1.20D 5061 OK (1 ms) 101(Unspecified)D 0 UNKNOWN jnctn/obitori 66.227.100.20N 5060 OK (23 ms) 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 offline] victoria*CLI On Fri, 2008-12-19 at 15:18 -0600, Eric ManxPower Wieling wrote: obitori junk wrote: I am experiencing a 606 not Acceptable error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. In my experience Not Acceptable errors happen because the two endpoints cannot agree on a codec. Try allowing all the codecs in your softphone and in Asterisk sip.conf [general] do a disallow=all and an allow=ulaw.I suggest you do this in [general] when testing because it can sometimes be hard to make sure that a peer/friend/user entry is actually matching the incoming call. Once you get it working you can refine it. You could be having a NAT issue too, but I don't think it is
Re: [asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem
Check you codecprobe settings http://lists.digium.com/pipermail/asterisk-dev/2006-October/024101.html -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of obitori junk Sent: Friday, December 19, 2008 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem Danny: I restarted asterisk both times after making changes to the sip.conf in response to Eric's suggestions. I'll send you my original post with sip debug info. I ran sip debug again. The logged info is identical to what I was getting before I changed the codec allow/disallow settings... I think this is an Asterisk behind a NAT configuration problem, but if I could be wrong. Bud On Fri, 2008-12-19 at 16:38 -0600, Danny Nicholas wrote: Have you done a sip set debug, then sip reload? Do you have a range of 1-2 open in your firewall? Asterisk will poke out through 5060 but has to get a random response back in the 10-20K range (you can narrow this) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of obitori junk Sent: Friday, December 19, 2008 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem I tried both disallow all /// allow ulaw and allow all. Neither worked. Ekiga.net is not accepting asterisk as a client. I keep getting the 600 not acceptable error. What seems odd to me is that the NAT column shows an n for both ekiga.net and jnctn.net, but I have nat=yes for both. Thanks for the suggestion...I'm still open to ideas. Regards, Bud Roth sip.conf contains: allow all OUTPUT-- Name/username HostDyn Nat ACL Port Status ekiga/budzhaus 86.64.162.35 N 5060 OK (102 ms) 110(Unspecified)D 0 UNKNOWN 109(Unspecified)D 0 UNKNOWN 108(Unspecified)D 0 UNKNOWN 107(Unspecified)D 0 UNKNOWN 106(Unspecified)D 0 UNKNOWN 105(Unspecified)D 0 UNKNOWN 104(Unspecified)D 0 UNKNOWN 103(Unspecified)D 0 UNKNOWN 102/10210.1.1.20D 5061 OK (1 ms) 101(Unspecified)D 0 UNKNOWN jnctn/obitori 66.227.100.20N 5060 OK (22 ms) 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 offline] [Dec 19 17:20:31] NOTICE[2478]: chan_sip.c:7517 sip_reg_timeout:-- Registration for 'budzh...@ekiga.net' timed out, trying again (Attempt #2) -- Got SIP response 606 Not Acceptable back from 86.64.162.35 sip.conf contains: disallow all allow ulaw OUTPUT-- victoria*CLI sip show peers Name/username HostDyn Nat ACL Port Status ekiga/budzhaus 86.64.162.35 N 5060 OK (97 ms) 110(Unspecified)D 0 UNKNOWN 109(Unspecified)D 0 UNKNOWN 108(Unspecified)D 0 UNKNOWN 107(Unspecified)D 0 UNKNOWN 106(Unspecified)D 0 UNKNOWN 105(Unspecified)D 0 UNKNOWN 104(Unspecified)D 0 UNKNOWN 103(Unspecified)D 0 UNKNOWN 102/10210.1.1.20D 5061 OK (1 ms) 101(Unspecified)D 0 UNKNOWN jnctn/obitori 66.227.100.20N 5060 OK (23 ms) 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 offline] victoria*CLI On Fri, 2008-12-19 at 15:18 -0600, Eric ManxPower Wieling wrote: obitori junk wrote: I am experiencing a 606 not Acceptable error trying to set up an Asterisk server as an ekiga.net client. My server is behind a firewall with NAT routing. I have googled this problem and read about Asterisk feeding its local ip address to ekiga.net. That seems to be my problem. In my experience Not Acceptable errors happen because the two
Re: [asterisk-users] Increase DTMF Tone Duration
You can edit the duration of the tones in app_senddtmf.c and then rebuild asterisk. That would be true if we were using app_senddtmf, but these are DTMF tones sent directly from the phone once the call is up. What we did was to edit channel.c and added 100ms to the tone duration using 'ast_safe_sleep(chan,100);'. Andres http://www.telesip.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] Authorize Microsoft SQL
On Thursday 18 December 2008 21:37:29 Gregory Malsack wrote:
I have an installation where the client has a Microsoft SQL database that
holds all of their case information. They would like the asterisk system to
require users to enter a valid case number when making an outgoing call.
I’m seeing some documentation regarding people using Microsoft SQL for CDR
storage, however nothing regarding validating authentication using a
Microsoft SQL database.
Install tdsodbc and UnixODBC. Configure UnixODBC:
odbcinst.ini:
[TDS]
Driver = /usr/lib/odbc/libtdsodbc.so
odbc.ini:
[windows]
Driver=TDS
tds_version=8.0
Server=192.168.1.150
Database=asterisk
Port=1433
res_odbc.conf:
[sqlserver]
dsn=windows
pooling=yes
limit=100
username=oscar
password=thegrouch
pre-connect = yes
sanitysql = select count(*) from systables
func_odbc.conf:
[WHATEVER]
dsn=sqlserver
read=SELECT foo FROM bar WHERE baz='${SQL_ESC(${ARG1})}'
write=UPDATE bar SET foo='${SQL_ESC(${VAL1})}' WHERE baz='${SQL_ESC(${ARG1})}'
extensions.conf:
exten = 123,1,Set(foo=${ODBC_WHATEVER(${CALLERID(num)})})
exten = 456,1,Set(ODBC_WHATEVER(${CALLERID(num)})=someotherfoo)
--
Tilghman
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[asterisk-users] New mailing list: digium-announce
Recently it was brought to our attention that while we announce new releases of Asterisk and Asterisk-Addons on the asterisk-announce mailing list (and others), and we publish security advisories on the asterisk-announce and asterisk-security mailing lists (and others), there are frequently changes that are made in our policies, procedures, and products that do not get announced on any widely-read mailing list. To help with this situation, we've created a new mailing list called 'digium-announce', located on the lists.digium.com list server. This will be a low-volume, read-only (no discussion) list that will be used for various announcements, including: 1) Changes to Digium's services for the Asterisk community, like the bugs.digium.com issue tracker, the downloads.digium.com download server and others 2) Changes to and new releases of Digium's commercial software products used by the Asterisk community like the G.729 codec, the HPEC echo canceler and others 3) Improvements to the asterisk.org web site Please feel free to subscribe to this mailing list by visiting http://lists.digium.com/mailman/listinfo/digium-announce Thanks for using Asterisk! -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- 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] Pre-routing manipulation of calls
Quoth Kevin DeGraaf ke...@kdegraaf.net
I want to do some manipulation (CallerID name override) to all incoming
calls before they are routed. I would prefer to avoid duplicating the
necessary code in each DID extension stanza, even if it's just a call to
a macro.
I do it like so:
[isdn_in]
exten = _X.,1,Set(CALLERID(name)=Banana)
;
exten = _X.,n,Goto(real_isdn_in,${EXTEN},1)
;
[real_isdn_in]
;
; DDI planning
;
exten = _871800,1,Goto(groups,200,1)
exten = _871802,1,Macro(stdexten|112|Sip/112)
exten = _871803,1,Macro(stdexten|113|Sip/113)
exten = _871804,1,Macro(stdexten|114|Sip/114)
;...
HtH
--
Regards,
Russell
| Russell Brown | MAIL: russ...@lls.com PHONE: 01780 471800 |
| Lady Lodge Systems | WWW Work: http://www.lls.com |
| Peterborough, England | WWW Play: http://www.ruffle.me.uk |
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[asterisk-users] Setup ReceiveFax(), fax2mail, mime-construct - but now Sendmail :(
Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got this far:
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0${CALLERIDNUM})
exten = s,2,ReceiveFAX(${FAXFILE}.tif)
exten = s,3,Hangup()
exten=h,1,System(/usr/local/bin/fax2mail --cid-number 0${CALLERIDNUM}
--cid-name home fax --dest-name admin --dest-email ${admin_email}
-f ${FAXFILE})
which all seems work well on the CLI. No errors.
fax2mail uses mime-contruct to send the fax by sendmail. That didn't work.
No email. /var/log/maillog:
Dec 19 21:04:53 asterisk sendmail[2628]: mBH2mWvQ006043:
to=ad...@myco.com, ctladdr=r...@localhost.localdomain (0/0),
delay=2+23:16:09, xdelay=00:00:00, mailer=esmtp, pri=6640305,
relay=mx01.1and1.com., dsn=4.0.0, stat=Deferred: Connection timed out
with mx01.1and1.com.
Dec 19 21:04:53 asterisk sendmail[2628]: mBH2mWvS006043:
to=ad...@myco.com, ctladdr=r...@localhost.localdomain (0/0),
delay=2+23:16:09, xdelay=00:00:00, mailer=esmtp, pri=6640312,
relay=mx00.1and1.com., dsn=4.0.0, stat=Deferred: Connection timed out
with mx00.1and1.com.
I've avoided MTA's like sendmail for a _long_ time. So I need help.
1. Is this the right list to try to resolve this? If not, which list?
2. postfix seems to considered much easier to configure than sendmail.
Do I install postfix? If so, will this work out of the box?
3. If sendmail, what's the magic configuration?
sean
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Re: [asterisk-users] Setup ReceiveFax(), fax2mail, mime-construct - but now Sendmail :(
On Friday 19 December 2008 20:24:11 sean darcy wrote:
Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got this far:
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0
${CALLERIDNUM}) exten = s,2,ReceiveFAX(${FAXFILE}.tif)
exten = s,3,Hangup()
exten=h,1,System(/usr/local/bin/fax2mail --cid-number 0${CALLERIDNUM}
--cid-name home fax --dest-name admin --dest-email ${admin_email}
-f ${FAXFILE})
which all seems work well on the CLI. No errors.
fax2mail uses mime-contruct to send the fax by sendmail. That didn't work.
No email. /var/log/maillog:
Dec 19 21:04:53 asterisk sendmail[2628]: mBH2mWvQ006043:
to=ad...@myco.com, ctladdr=r...@localhost.localdomain (0/0),
delay=2+23:16:09, xdelay=00:00:00, mailer=esmtp, pri=6640305,
relay=mx01.1and1.com., dsn=4.0.0, stat=Deferred: Connection timed out
with mx01.1and1.com.
Dec 19 21:04:53 asterisk sendmail[2628]: mBH2mWvS006043:
to=ad...@myco.com, ctladdr=r...@localhost.localdomain (0/0),
delay=2+23:16:09, xdelay=00:00:00, mailer=esmtp, pri=6640312,
relay=mx00.1and1.com., dsn=4.0.0, stat=Deferred: Connection timed out
with mx00.1and1.com.
I've avoided MTA's like sendmail for a _long_ time. So I need help.
1. Is this the right list to try to resolve this? If not, which list?
2. postfix seems to considered much easier to configure than sendmail.
Do I install postfix? If so, will this work out of the box?
3. If sendmail, what's the magic configuration?
i'm still working on this, but take a look at
http://messinet.com/viewvc/asterisk-fax-gw/trunk/
currently, i use postfix, which seems easier to me to configure than sendmail
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
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[asterisk-users] how to get /var/run/asteris/asterisk.ctl
Hello there everyone, Well I have set up Asteriks 6.0 and almost have Freepbx working too. However, freepbx is showing me that /var/run/asterisk/asterisk.ctl is not found. I confirmed that by going to the directory. How do I get /var/run/asterisk/asterisk.ctl put in correctly? I am using a Ubuntu 8.10 system. Thanks much. ___ -- 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
[asterisk-users] Needs more cpu usage
Hi, I am running * on centos5 using 4core cpu. When it is busy, * uses 99.9% of cpu max. How can I make * to use more cpu power? Thanks. ___ -- 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] Problem configuring Asterisk as client for ekiga.net -- NAT problem
Danny: I added codecprobe=yes did a restart and got the same error. I changed it to codecprobe=no and got the same error. Is it possible that the error is similar to this: http://www.nabble.com/ekiga-registration-in-asterisk-td15816700.html The reason I ask this is that my sip debug logs show: From: asterisk sip:aster...@10.1.1.40;tag=as183adb5e To: sip:ekiga.net;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.d07f Asterisk is putting itself out as 10.1.1.40 which obviously is my internal LAN IP and not something that can be reached by ekiga.net. I've used both stunaddr=stun.ekiga.net and externip=71.xxx.xxx.xxx (my router's IP address). I double checked my router. I'd not updated my IP address for the UDP port forwarding since installing my new server (that is running asterisk). But, even after making sure 1-2 is being forwarded to my box, I don't see a single packet being logged by my firewall (either accepted or blocked--I fiddled with logging to try both) from ekiga.net's ip address. Also, junctn.net's sip connection is working no problem. I'm stumped. :( Any ideas out there? Thanks for the help so far...Still plugging away... Bud On Fri, 2008-12-19 at 17:00 -0600, Danny Nicholas wrote: Check you codecprobe settings http://lists.digium.com/pipermail/asterisk-dev/2006-October/024101.html -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of obitori junk Sent: Friday, December 19, 2008 4:55 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem configuring Asterisk as client for ekiga.net -- NAT problem Danny: I restarted asterisk both times after making changes to the sip.conf in response to Eric's suggestions. I'll send you my original post with sip debug info. I ran sip debug again. The logged info is identical to what I was getting before I changed the codec allow/disallow settings... I think this is an Asterisk behind a NAT configuration problem, but if I could be wrong. Bud On Fri, 2008-12-19 at 16:38 -0600, Danny Nicholas wrote: Have you done a sip set debug, then sip reload? Do you have a range of 1-2 open in your firewall? Asterisk will poke out through 5060 but has to get a random response back in the 10-20K range (you can narrow this) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of obitori junk Sent: Friday, December 19, 2008 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problem configuring Asterisk as client for86.83.214.233 ekiga.net -- NAT problem I tried both disallow all /// allow ulaw and allow all. Neither worked. Ekiga.net is not accepting asterisk as a client. I keep getting the 600 not acceptable error. What seems odd to me is that the NAT column shows an n for both ekiga.net and jnctn.net, but I have nat=yes for both. Thanks for the suggestion...I'm still open to ideas. Regards, Bud Roth sip.conf contains: allow all OUTPUT-- Name/username HostDyn Nat ACL Port Status ekiga/budzhaus 86.64.162.35 N 5060 OK (102 ms) 110(Unspecified)D 0 UNKNOWN 109(Unspecified)D 0 UNKNOWN 108(Unspecified)D 0 UNKNOWN 107(Unspecified)D 0 UNKNOWN 106(Unspecified)D 0 UNKNOWN 105(Unspecified)D 0 UNKNOWN 104(Unspecified)D 0 UNKNOWN 103(Unspecified)D 0From: asterisk sip:aster...@10.1.1.40;tag=as183adb5e To: sip:ekiga.net;tag=c64e1f832a41ec1c1f4e5673ac5b80f6.d07f UNKNOWN 102/10210.1.1.20D 5061 OK (1 ms) 101(Unspecified)D 0 UNKNOWN jnctn/obitori 66.227.100.20N 5060 OK (22 ms) 12 sip peers [Monitored: 3 online, 9 offline Unmonitored: 0 online, 0 offline] [Dec 19 17:20:31] NOTICE[2478]: chan_sip.c:7517 sip_reg_timeout:-- Registration for 'budzh...@ekiga.net' timed out, trying again (Attempt #2) -- Got SIP response 606 Not Acceptable back from 86.64.162.35 sip.conf contains: disallow all allow ulaw OUTPUT-- victoria*CLI sip show peers Name/username HostDyn Nat ACL Port Status ekiga/budzhaus 86.64.162.35 N 5060 OK (97 ms)
Re: [asterisk-users] how to get /var/run/asteris/asterisk.ctl
Scott Berry wrote: Hello there everyone, Well I have set up Asteriks 6.0 and almost have Freepbx working too. However, freepbx is showing me that /var/run/asterisk/asterisk.ctl is not found. I confirmed that by going to the directory. How do I get /var/run/asterisk/asterisk.ctl put in correctly? I am using a Ubuntu 8.10 system. Thanks much. When you start asterisk, it creates the control socket. Have a look at /etc/asterisk/asterisk.conf to see if your install has put it somplace other than the stated default location. If you think asterisk is started and the control socket isn't there, look at /var/log/asterisk/full to see why it didn't start ___ -- 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] app directory error: libc-client undefined symbol
Hi Sean - On Wed, Dec 3, 2008 at 7:36 PM, sean darcy seandar...@gmail.com wrote: Installing 1.4.23-rc2, I actually looked at the startup and saw this warning: WARNING[10730]: loader.c:359 load_dynamic_module: Error loading module 'app_directory.so': /usr/lib/libc-client.so.2007: undefined symbol: mm_dlog I'm running Fedora Core 9, with libc-client 2007d. googling didn't help, so what's the problem? Do I need a more recent (different) libc-client? I've got the same problem here with CentOS 5.2 and Asterisk 1.6.0.3-rc1. I tried rebuilding the libc-client rpm's from the source rpm, but the problem is still there. I'm trying to build libc-client from UW's source, but it seems to be a non-trivial thing. I'll let you know. Thanks, Noah ___ -- 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
