[asterisk-users] AGI script exits non-zero when running system command
Hey guys I was hoping I could get a few pointers on a problem I have been trying to debug for the last couple of months regarding asterisk AGI scripts and unexpected termination. I have this agi script that accepts incoming faxes using RxFax on the latest asterisk 1.4 branch. Its written with perl and it works fine except for one line that causes the entire script to terminate unexpectedly. The script always terminates at the point where I use the 'system' command or backticks to run a system command. Example: system( "/usr/bin/tiff2pdf -f -p letter -o $faxpath/$unique.pdf $faxpath/$unique.tiff" ); The asterisk log with agi debugging on is pasted below I have tried everything I can think of over the past few months, taking a break every so often obviously, but now I feel like I really need outside eyes. Its worth noting that the script runs fine without running the system command, and it does not matter which system command I run. I tried just doing a simple copy of the file and it failed in the same place. Asterisk leaves me with little help, just explaining that the script returned non-zero. Are there any issues I should be aware of when running system commands from an AGI script? I did check permissions and made sure my asterisk user can write to /tmp and use the converting commands. I did a lot more testing of course but that is probably the biggest face-palm error there could be. Asterisk log: -- Launched AGI Script /var/lib/asterisk/agi-bin/fax.agi AGI Tx >> agi_request: fax.agi AGI Tx >> agi_channel: SIP/trunk-0035 AGI Tx >> agi_language: en AGI Tx >> agi_type: SIP AGI Tx >> agi_uniqueid: 1296624119.53 AGI Tx >> agi_callerid: anonymous AGI Tx >> agi_calleridname: Anonymous AGI Tx >> agi_callingpres: 32 AGI Tx >> agi_callingani2: 0 AGI Tx >> agi_callington: 0 AGI Tx >> agi_callingtns: 0 AGI Tx >> agi_dnid: XX AGI Tx >> agi_rdnis: unknown AGI Tx >> agi_context: from-trunk AGI Tx >> agi_extension: XX AGI Tx >> agi_priority: 3 AGI Tx >> agi_enhanced: 0.0 AGI Tx >> agi_accountcode: AGI Tx >> AGI Rx << GET VARIABLE EXTEN AGI Tx >> 200 result=1 (XX) AGI Rx << GET VARIABLE CALLERID(num) AGI Tx >> 200 result=1 (anonymous) AGI Rx << VERBOSE "DEBUG: EXTEN - XX CID - anonymous" 1 fax.agi: DEBUG: EXTEN - XX CID - anonymous AGI Tx >> 200 result=1 AGI Rx << GET VARIABLE UNIQUEID AGI Tx >> 200 result=1 (1296624119.53) AGI Rx << VERBOSE "RxFAX XX: /tmp/1296624119.53.tiff" 1 fax.agi: RxFAX XX: /tmp/1296624119.53.tiff AGI Tx >> 200 result=1 AGI Rx << EXEC RxFAX "/tmp/1296624119.53.tiff" -- AGI Script Executing Application: (RxFAX) Options: (/tmp/1296624119.53.tiff) Really destroying SIP dialog '6327EDB3@XXX' Method: OPTIONS [Feb 1 23:22:18] ERROR[13753]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:214 phase_e_handler: [FaxReceived ERROR] result (13) Unexpected message received. [FaxReceived ERROR] result (13) Unexpected message received. [Feb 1 23:22:18] WARNING[13753]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:478 fax_run: RXFAX: t30_call_active is FALSE. AGI Tx >> 200 result=0 AGI Rx << EXEC RxFAX "/tmp/1296624119.53.tiff" -- AGI Script Executing Application: (RxFAX) Options: (/tmp/1296624119.53.tiff) Really destroying SIP dialog '132f38cb284eef837df0038477511f55@XXX' Method: OPTIONS REGISTER attempt 1 to XX@trunk Really destroying SIP dialog '33dff0b60f7ce29944351e446c2e7b5b@XXX' Method: REGISTER Really destroying SIP dialog 'AE6C429F@XXX' Method: OPTIONS [Feb 1 23:23:17] NOTICE[13753]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:230 phase_d_handler: [RXFAX NEW PAGE]: Channel: SIP/trunk-0035 Pages: -1224970700 Speed: 14400 [Feb 1 23:23:17] NOTICE[13753]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:231 phase_d_handler:Bad rows: 0 - Longest bad row run: 0 - Compression type: T.4 2-D [Feb 1 23:23:17] NOTICE[13753]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:232 phase_d_handler:Image size bytes: 86071 - Image size: 1728 x 2156 - Image resolution: 8031 x 7700 -- [RXFAX NEW PAGE]: Channel: SIP/trunk-0035 Pages: -1224970700 Speed: 14400 [Feb 1 23:23:18] NOTICE[13752]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:230 phase_d_handler: [RXFAX NEW PAGE]: Channel: SIP/trunk-0034 Pages: -1225599608 Speed: 14400 [Feb 1 23:23:18] NOTICE[13752]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:231 phase_d_handler:Bad rows: 0 - Longest bad row run: 0 - Compression type: T.4 2-D [Feb 1 23:23:18] NOTICE[13752]: /usr/src/asterisk/agx-ast-addons/app-spandsp/app_fax.c:232 phase_d_handler:Image size bytes: 86072 - Image size: 1728 x 2156 - Image resolution: 8031 x 7700 -- [RXFAX NEW PAGE]: Channel: SIP/trunk-0034 Pages: -1225599608 Speed: 14400 Really destroying SIP dialog '439a2cca2a745a565a4e0aab56a054b8@XXX' Method: OPTIONS Really destroying SIP dialog '49515A3F@XXX' Method:
[asterisk-users] regarding sip.conf and extensions.conf
Hi all, My experiment scenario is like this: SIPp Uac -> ASTERISK SERVER-->SIPp uas 1. when i had registered bob with this command ./sipp -sf register_client.xml -inf register1.csv -i 192.168.1.6:5060 192.168.1.6 -p 5061 -m 1 it has registered If i want to register another client alice with same command but with few changes in port number ...etc. The first registered client is erased and showing second one when i have checked with sip show peers and sip show registry... Why is this happening? 2. if i want to register 4 clients at same time how can i give sip.conf and extensions.conf.. ? i mean how can i write 4 sip.conf in same sip.conf and extensions.conf? and finally i need some thing about 3. When i send register messages from 5 clients at a time how can the asterisk server handle them ? suppose if we have written many sip.confs how asterisk server will take ? i mean diagram like this 1st clientxml+.csv file-> 2st clientxml+.csv file->asterisk 3st clientxml+.csv file-> in sip.conf if i have written 3 sip.confs which one will be taken by server first and which one by second i mean like sequence? Best R viswavardhan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how to get Current Calls details
Hi everyone How can I get the current calls details in asterisk.if I use cli commad core show channels,there is two channels of each call.But the requirement is, need to get caller ,calee,starttime ,duration of the current calls.This value should be proper for call forward,call transfer ,and scenarios.Please help me on this. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Performance
That's quite possible. We handle around 100 similtaneous calls(PRI + SIP) with a decent dell server with only 4gb ram. On Wed, Feb 2, 2011 at 6:22 AM, Juan David Diaz wrote: > Hi Asterisk Users, > I would like to handle about 250 simultaneous (calls & agents only) calls > with PRI or a SIP trunk with the following configuration > > Dell R710 > > Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single > Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz > > Memory 12GB, 1333MHz > > RAID 1 - 1 Tb X 2 > > Is that possible?? > Kind Regards > Juan. > Linux User #441131 > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Musiconhold priority
On Tue, Feb 1, 2011 at 10:20 AM, Danny Nicholas wrote: > Not sure how queues factor into this equation; guess that’s a “try and see” > thing. > > >From my experience, the explicitly defined Set(CHANNEL(musicclass)=blah) takes precedence over a queue's defined moh class. -- Thanks, --Warren Selby, dCAP http://www.selbytech.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Performance
Tuesday, February 1, 2011, 11:22:30 PM, Juan wrote: > I would like to handle about 250 simultaneous (calls & agents only) calls > with PRI or a SIP trunk with the following configuration > Dell R710 > Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single > Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz > Memory 12GB, 1333MHz > RAID 1 - 1 Tb X 2 > Is that possible?? This is an overkill machine for that :) -- Best regards, Gergomailto:csi...@gmail.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Performance
On 11-02-01 05:22 PM, Juan David Diaz wrote: I would like to handle about 250 simultaneous (calls& agents only) calls with PRI or a SIP trunk with the following configuration Dell R710 Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz Memory 12GB, 1333MHz RAID 1 - 1 Tb X 2 Is that possible?? While it's certainly hard to accurately determine (without testing) what a system can handle, that certainly seems like an adequate machine to handle that kind of load, especially if you're not transcoding, doing much call recording, etc... (While that doesn't mean it can't handle all that, conferencing, recording calls (disk I/O), and transcoding are the heaviest uses of resources in my experience.) Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Performance
Hi Asterisk Users, I would like to handle about 250 simultaneous (calls & agents only) calls with PRI or a SIP trunk with the following configuration Dell R710 Dual Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz or Single Intel® Xeon® X5650, 2.66Ghz, 12M Cache,Turbo, HT, 1333MHz Memory 12GB, 1333MHz RAID 1 - 1 Tb X 2 Is that possible?? Kind Regards Juan. Linux User #441131 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use Monitor() in Python AGI
On Tue, 1 Feb 2011, Felix Dong wrote:
How can I use the application Monitor() in the Python AGI skripts?
Use the exec AGI command.
I use C so it looks something like this:
exec_agi("exec MONITOR wav|%s/%02d-prompt|m"
, recording_path
, idx
);
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Upgrade and recompilation
On 02/01/2011 12:34 PM, Harel Cohen wrote: > As one with theoretical knowledge in programing, but never on Linux, I > can understand terms and code structure but I don’t know: > > 1. What shell commands (e.g. ./configure, make, make install etc.) > should I run to recompile Asterisk (same version)? > > 2. What shell commands should I run if I want to apply a change to > source code? > > 3. Is there a general guide on how to upgrade Asterisk? Read the "README" file included with the source. Barry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Return variables from func_odbc calls?
On 11-02-01 01:21 PM, Tilghman Lesher wrote: > Assuming you were using a MySQL backend that supported transactions, > you could use the transaction layer in Asterisk 1.6.2 and greater to ensure > that each channel got a serialized view. That would make this approach > work. > Ya, I think I'm going to use this approach for the test. I was able to find some limited information on the wiki, let me see if I can get it working. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Return variables from func_odbc calls?
On Tuesday 01 February 2011 12:36:46 Jose P. Espinal wrote: > Paul Belanger wrote: > > On 11-01-26 02:59 PM, Tilghman Lesher wrote: > >> On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote: > >>> [CREATECALL] > >>> dsn=Example > >>> writesql=INSERT INTO x (y) VALUES (z) > >>> readsql=SELECT LAST_INSERT_ID(); > >> > >> That assumes you have only one call in existence at a time. If two > >> calls came in and executed the query at about the same time, it's > >> possible for both reads to return the same value. > > > > Yup, didn't even think of that. My testing of ODBC was a single > > channel. Guess I need another method to return the last ID of the > > record that was just inserted. > > In this case, does the Asterisk connection to MySQL through odbc counts > as a unique 'client', or does each call to a function will count as a > 'client'? The first. But you need to also understand that unless you use transactions, and specifically the transaction support in Asterisk, each channel is not guaranteed to be using the same connection on the second query. Or even if they all use the same connection, the queries are not serialized in the way that you might otherwise expect. The transaction support introduced in Asterisk 1.6.2 allows a connection to be reserved exclusively to a single channel, thus ensuring that the second query on a channel really was the very next query on the connection. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3?
On Tue, Feb 1, 2011 at 12:30 PM, Danny Nicholas wrote: > Now that my “smart” answer is out of the way, did you try > > - srtpcapable=no > > - in sip.conf? > > > > reference: http://www.voip-info.org/wiki/view/Asterisk+SRTP I've been looking at the trunk (1.8.+) code recently wrt srtp configuration. 'srtpcapable' is not a parsed string in sip.conf. The string does not even appear in the source code. I would recommend that you check for all occurances of encryption=... in sip.conf and comment them out, though encryption=no should also work. Can you show your sip.conf and a trace of the call with 'sip set debug on' that shows the a=crypto: line in the INVITE SDP? It doesn't happen for me. I am able to send INVITEs with or without the a=crypto: line by setting encryption=[yes|no] since 1.8.0-beta2. HTH -- -Bob Beers -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Return variables from func_odbc calls?
Paul Belanger wrote: On 11-01-26 02:59 PM, Tilghman Lesher wrote: On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote: [CREATECALL] dsn=Example writesql=INSERT INTO x (y) VALUES (z) readsql=SELECT LAST_INSERT_ID(); That assumes you have only one call in existence at a time. If two calls came in and executed the query at about the same time, it's possible for both reads to return the same value. Yup, didn't even think of that. My testing of ODBC was a single channel. Guess I need another method to return the last ID of the record that was just inserted. In this case, does the Asterisk connection to MySQL through odbc counts as a unique 'client', or does each call to a function will count as a 'client'? I ask because of this: "For LAST_INSERT_ID(), the most recently generated ID is maintained in the server on a per-connection basis. It is not changed by another client. It is not even changed if you update another AUTO_INCREMENT column with a nonmagic value (that is, a value that is not NULL and not 0). Using LAST_INSERT_ID() and AUTO_INCREMENT columns simultaneously from multiple clients is perfectly valid. Each client will receive the last inserted ID for the last statement that client executed." at: http://dev.mysql.com/doc/refman/5.0/en/getting-unique-id.html -- Jose P. Espinal http://www.eSlackware.com IRC: Khratos @ #asterisk / -doc / -bugs -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Return variables from func_odbc calls?
On Tuesday 01 February 2011 11:49:51 Paul Belanger wrote: > On 11-01-26 02:59 PM, Tilghman Lesher wrote: > > On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote: > >> [CREATECALL] > >> dsn=Example > >> writesql=INSERT INTO x (y) VALUES (z) > >> readsql=SELECT LAST_INSERT_ID(); > > > > That assumes you have only one call in existence at a time. If two > > calls came in and executed the query at about the same time, it's > > possible for both reads to return the same value. > > Yup, didn't even think of that. My testing of ODBC was a single > channel. Guess I need another method to return the last ID of the > record that was just inserted. Assuming you were using a MySQL backend that supported transactions, you could use the transaction layer in Asterisk 1.6.2 and greater to ensure that each channel got a serialized view. That would make this approach work. -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3?
According to chapter 7 (Outside Connectivity) of the excellent "Asterisk: The Definitive Guide" (review version online at http://ofps.oreilly.com/titles/9780596517342/index.html), the following enables secure signaling and media paths: exten => 1234,1,Set(CHANNEL(secure_bridge_signaling)=1) same => n,Set(CHANNEL(secure_bridge_media)=1) I would assume that setting these channel variables to 0 will disable secure media and signaling. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
On Tuesday 01 February 2011 02:24:20 Benny Amorsen wrote: > Tilghman Lesher writes: > > Correct; and Asterisk needs to be started as root, even if it will > > drop privileges after startup. Do this, and there should be no > > problems. > > Starting as root + dropping privileges is fine. Running configure as > root is not so fine; that basically makes building RPMS impossible. Alternatively, if you can set "ulimit -n 32768" in your RPM build environment (this needs to be set as a login requirement), you can sidestep the need for configure to run as root. The only reason it needs root is to expand the file descriptor limit so it can test using a file descriptor beyond 1023 (the usual limit). -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] B410P: DAHDI BRI PTMP HDLC Abort (6) on Primary D-channel of span 1 (TEI Errors)
2011/1/24 Matt Riddell > Hi all, > > So, we reverted the LibPRI version and tested it, and then tried with the > latest version of everything. Still no changes. > > The BRI line is in PTMP. If we set the configs to PTMP in the > genconf_parameters and try it, we get the following: > > [Jan 21 17:32:20] ERROR[20341]: chan_dahdi.c:12645 dahdi_pri_error: Unable > to receive TEI from network! > > If we set it to PTP (which it is not) we get the following message: > > [Jan 21 17:33:42] ERROR[20418]: chan_dahdi.c:12645 dahdi_pri_error: > Received MDL/TEI managemement message, but configured for mode other than > PTMP! > > So, with PTMP it says we don't get a TEI message and without it, it says we > do! > > :) > > Either way we also get the following message non stop in the console: > > [Jan 21 17:33:42] NOTICE[20419]: chan_dahdi.c:12946 pri_dchannel: PRI got > event: HDLC Abort (6) on Primary D-channel of span 2 > [Jan 21 17:33:42] NOTICE[20419]: chan_dahdi.c:12946 pri_dchannel: PRI got > event: HDLC Abort (6) on Primary D-channel of span 1 > > If we change the hardhdlc to dchannel instead the message goes away, but > obviously it doesn't work :) > > So, anyone have any ideas? > > -- > Cheers, > > Matt Riddell > ___ > > Hi Matt, Too bad I can't be more helpful on this but could work around this issue ? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Return variables from func_odbc calls?
On 11-01-26 02:59 PM, Tilghman Lesher wrote: > On Wednesday 26 January 2011 07:01:12 Paul Belanger wrote: >> [CREATECALL] >> dsn=Example >> writesql=INSERT INTO x (y) VALUES (z) >> readsql=SELECT LAST_INSERT_ID(); > > That assumes you have only one call in existence at a time. If two calls > came in and executed the query at about the same time, it's possible for > both reads to return the same value. > Yup, didn't even think of that. My testing of ODBC was a single channel. Guess I need another method to return the last ID of the record that was just inserted. -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Upgrade and recompilation
Hello All, As one with theoretical knowledge in programing, but never on Linux, I can understand terms and code structure but I don't know: 1. What shell commands (e.g. ./configure, make, make install etc.) should I run to recompile Asterisk (same version)? 2. What shell commands should I run if I want to apply a change to source code? 3. Is there a general guide on how to upgrade Asterisk? Kind Regards, Harel Cohen -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Baptista Sent: Tuesday, February 01, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3? Hi again, Nobody knows how to disable it? Can at least someone pinpoint me where can I check the latest documentation regarding SRTP. Maybe something might have change in the meanwhile 'Cause so far it looks like there is a bug in asterisk. Well, maybe I should report this bug then. - Miguel Baptista On 28.01.2011 18:22, Miguel Baptista wrote: Hi all, I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the encryption=no in sip.conf and the _SIPSRTP_CRYPTO=disable on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have noload=res_srtp.so on my modules.conf otherwise I cannot place SIP calls (cause other ends don't support it) Regards, Miguel Baptista Now that my "smart" answer is out of the way, did you try - srtpcapable=no - in sip.conf? reference: http://www.voip-info.org/wiki/view/Asterisk+SRTP -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Baptista Sent: Tuesday, February 01, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3? Hi again, Nobody knows how to disable it? Can at least someone pinpoint me where can I check the latest documentation regarding SRTP. Maybe something might have change in the meanwhile 'Cause so far it looks like there is a bug in asterisk. Well, maybe I should report this bug then. - Miguel Baptista On 28.01.2011 18:22, Miguel Baptista wrote: Hi all, I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the encryption=no in sip.conf and the _SIPSRTP_CRYPTO=disable on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have noload=res_srtp.so on my modules.conf otherwise I cannot place SIP calls (cause other ends don't support it) Regards, Miguel Baptista Not to be smart (because I am not) but you answered your own question - noload=res_srtp.so disables the module. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3?
I am as well - Original Message - From: asterisk-users-boun...@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tue Feb 01 11:22:41 2011 Subject: Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3? Hi again, Nobody knows how to disable it? Can at least someone pinpoint me where can I check the latest documentation regarding SRTP. Maybe something might have change in the meanwhile 'Cause so far it looks like there is a bug in asterisk. Well, maybe I should report this bug then. - Miguel Baptista On 28.01.2011 18:22, Miguel Baptista wrote: Hi all, I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I compiled it with SRTP support. Everything seems to work OK but I am having a weird issue. I cannot disable SRTP. I tried the encryption=no in sip.conf and the _SIPSRTP_CRYPTO=disable on my dailplan and it keeps trying to use the SRTP. Well, right now I have to have noload=res_srtp.so on my modules.conf otherwise I cannot place SIP calls (cause other ends don't support it) Regards, Miguel Baptista -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to disable srtp in asterisk 1.8.2.3?
Hi again, Nobody knows how to disable it? Can at least someone pinpoint me where can I check the latest documentation regarding SRTP. Maybe something might have change in the meanwhile 'Cause so far it looks like there is a bug in asterisk. Well, maybe I should report this bug then. - Miguel Baptista On 28.01.2011 18:22, Miguel Baptista wrote: > Hi all, > > I upgraded one of our servers running asterisk 1.6.X to 1.8.2.3. I > compiled it with SRTP support. > Everything seems to work OK but I am having a weird issue. I cannot > disable SRTP. I tried the /encryption=no/ in /sip.conf /and the > /_SIPSRTP_CRYPTO=disable/ on my dailplan and it keeps trying to use > the SRTP. > Well, right now I have to have/ noload=res_srtp.so/ on my > /modules.conf /otherwise I cannot place SIP calls (cause other ends > don't support it) > > Regards, > > Miguel Baptista > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to load new musiconhold classes ?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, February 01, 2011 10:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to load new musiconhold classes ? Hello, I've defined some new musiconhold classes in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [908001] mode=files directory=/var/lib/asterisk/moh/908001 random=yes ; [101001-1] mode=files directory=/var/lib/asterisk/moh/101001/1 random=yes ; [101001-2] mode=files directory=/var/lib/asterisk/moh/101001/2 random=yes But the new classes never show up : asterisk16*CLI> moh show classes Class: default Mode: files Directory: /var/lib/asterisk/moh I've done the following : asterisk16*CLI> module reload res_musiconhold.so asterisk16*CLI> moh reload asterisk16*CLI> core restart now How to load new musiconhold classes ?? Using asterisk 1.6.2.10 Kind regards, Jonas. I researched this in 1.4.37 and the new classes didn't show up in the "moh show classes" until they had a valid file in the directory to play. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback in uplink and recording in downlink
Yes, you can use the Mixmonitor command. But if you want to have only one party on the recording, you should use the Monitor command without the 'm' option. http://www.astblog.com/2011/02/01/asterisk-mixmonitor-cmd/ -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger Sent: 2011-02-01 09:41 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Playback in uplink and recording in downlink On 11-02-01 04:02 AM, Felix Dong wrote: > I got a question to asterisk 1.6. Is it possible to playback a > Audiofile in uplink and to record the downlink channel in another > Audifile at the same time? > Yes, look at MixMonitor. *CLI> core show application MixMonitor -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to use Monitor() in Python AGI
How can I use the application Monitor() in the Python AGI skripts? Thanks a lot. best regards, Feilx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to load new musiconhold classes ?
Hello, I've defined some new musiconhold classes in musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [908001] mode=files directory=/var/lib/asterisk/moh/908001 random=yes ; [101001-1] mode=files directory=/var/lib/asterisk/moh/101001/1 random=yes ; [101001-2] mode=files directory=/var/lib/asterisk/moh/101001/2 random=yes But the new classes never show up : asterisk16*CLI> moh show classes Class: default Mode: files Directory: /var/lib/asterisk/moh I've done the following : asterisk16*CLI> module reload res_musiconhold.so asterisk16*CLI> moh reload asterisk16*CLI> core restart now How to load new musiconhold classes ?? Using asterisk 1.6.2.10 Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Connecting to Cisco Iad2430 to Asterisk
Is it possible to SIP trunk to this Cisco device so that phones connected to the Cisco box can dial extensions on the Asterisk box? What I would like to be able to do is dial some extension(s) on phones connected to the Cisco box and have the call routed into extension(s) on the Asterisk box. One of our clients has a call center with 65 analog phones connected to the Cisco box. We would like to be able add our dialer appliance into their operation without having to replace any more equipment than needed. We need an easy way for the agents to connect to an extension on our appliance that basically does an agentlogin. Ideas as to how to best accomplish this would be appreciated. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Musiconhold priority
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens Sent: Tuesday, February 01, 2011 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Musiconhold priority Hello list, what musiconhold class has priority : - field "musiconhold" of the SIPaccount and field "musiconhold" of a queue or - Set(CHANNEL(musicclass)=) ?? Kind regards, Jonas. According to the internal documentation of res/res_musiconhold.c (source of 1.4.37), this is the "pecking order" /* The following is the order of preference for which class to use: * 1) The channels explicitly set musicclass, which should *only* be *set by a call to Set(CHANNEL(musicclass)=whatever) in the dialplan . * 2) The mclass argument. If a channel is calling ast_moh_start() as the *result of receiving a HOLD control frame, this should be the *payload that came with the frame. * 3) The interpclass argument. This would be from the mohinterpret *option from channel drivers. This is the same as the old musicclass *option. * 4) The default class. */ Not sure how queues factor into this equation; guess that's a "try and see" thing. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Musiconhold priority
Hello list, what musiconhold class has priority : - field "musiconhold" of the SIPaccount and field "musiconhold" of a queue or - Set(CHANNEL(musicclass)=) ?? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback in uplink and recording in downlink
On 11-02-01 04:02 AM, Felix Dong wrote: > I got a question to asterisk 1.6. Is it possible to playback a Audiofile in > uplink and to record the downlink channel in another Audifile at the same > time? > Yes, look at MixMonitor. *CLI> core show application MixMonitor -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com & http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] end a call after a specific time period
exten => _9944NX,1,Answer()
exten => _9944NX,2,Noop(GOING FOR THE AGI)
exten => _9944NX,3,Noop(XX)
exten => _9944NX,4,Noop()
exten => _9944NX,5,AGI(//Some script here it works perfectly fine)
exten => _9944NX,6,Noop(AGI ENDED)
exten => _9944NX,7,Set(TIMEOUT(absolute)=${TIME_OF_CALL_SECONDS})
exten =>
_9944NX,8,Dial(${VICITRUNK2}/${EXTEN:2},35,goL(${TIME_OF_CALL_SECONDS}))
exten => _9944NX,9,Hangup
I have tried many options but in vain. Thanks in advance
On Tue, Feb 1, 2011 at 3:40 AM, C F wrote:
>>
>> Channel Location State Application(Data)
>> SIP/NTT00- 99449046902115@vicid Down AppDial((Outgoing Line))
>> Local/99449046902115 99449046902115@defau Up
>> Dial(SIP/NTT00/449046902115||o
>> Local/99449046902115 8302@default:2 Up Playback(conf)
>>
>>
>> After a few seconds it shows
>>
>> 0*CLI> core show channels
>> Channel Location State Application(Data)
>> SIP/NTT00- 8302@default:2 Up Playback(conf)
>> 1 active channel
>> 1 active call
>>
>>
>> Now the conf file is not that long that it keeps on playing.
>>
>> I dont know what else to use to end the call.
>
>
> Post your dialplan please
>
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Best Regards
Shakeel Abbas
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[asterisk-users] How to Change The Caller Position in Queue
Dear Mr/Ms; web have some Queues and our Call Center and put caller in Queue Based on some regional decisions. by the way, after the Caller placed on Queues, we like to be able to reorder them on our rules. as an example: there is a queue which have 10 caller in waiting stage right now, one with the no:7 is VIP! so we need to change her place to no:2. ** again: i don't need to just make decision on incoming calls! I am about the callers which has been Queue before now! they are some where in the middle of the Queue. *what is the best way to do such a things, or alike...* -- Regards, Ali R. Taleghani 0936 322 4069 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Playback in uplink and recording in downlink
Hallo everybody, I got a question to asterisk 1.6. Is it possible to playback a Audiofile in uplink and to record the downlink channel in another Audifile at the same time? If it is possible, how should I do it? Please explain it. Thank you for your help to my thesis! best regards, Felix -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] exceeds the maximum size of ast_fdset error on Asterisk-1.8.0
Tilghman Lesher writes: > Correct; and Asterisk needs to be started as root, even if it will drop > privileges after startup. Do this, and there should be no problems. Starting as root + dropping privileges is fine. Running configure as root is not so fine; that basically makes building RPMS impossible. /Benny -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
