[asterisk-users] DIALSTATUS Values
Hello, I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) SIP/10036-0096AGI Rx VERBOSE Status 1 SIP/10036-0096AGI Tx 200 result=1 Please help me in this. Thanks, Kamlesh -- _ -- 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] DIALSTATUS Values
Hi, How are you calling this AGI in your dialplan !!? Regards, Sammy. On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 *SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) SIP/10036-0096AGI Rx VERBOSE Status 1 SIP/10036-0096AGI Tx 200 result=1* ** Please help me in this. Thanks, Kamlesh ** * * -- _ -- 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] DIALSTATUS Values
Hello, in /etc/extension.conf [privoip] exten = _00X.,n,AGI(isdcall.php) Regards, Kamlesh Date: Fri, 2 Dec 2011 16:16:27 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS Values Hi, How are you calling this AGI in your dialplan !!? Regards, Sammy. On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) SIP/10036-0096AGI Rx VERBOSE Status 1 SIP/10036-0096AGI Tx 200 result=1 Please help me in this. Thanks, Kamlesh -- _ -- 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 -- _ -- 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] DIALSTATUS Values
Can you also paste the Asterisk Console logs around the part where AGI is dialing and after the dialing part ! make sure AGi debug is enabled as well. On Fri, Dec 2, 2011 at 4:24 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, in /etc/extension.conf [privoip] exten = _00X.,n,AGI(isdcall.php) Regards, Kamlesh -- Date: Fri, 2 Dec 2011 16:16:27 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS Values Hi, How are you calling this AGI in your dialplan !!? Regards, Sammy. On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.comwrote: Hello, I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 *SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) SIP/10036-0096AGI Rx VERBOSE Status 1 SIP/10036-0096AGI Tx 200 result=1* ** Please help me in this. Thanks, Kamlesh ** * * -- _ -- 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 -- _ -- 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] DIALSTATUS Values
Here it is: SIP/10036-00a8AGI Tx agi_request: isdcall.php SIP/10036-00a8AGI Tx agi_channel: SIP/10036-00a8 SIP/10036-00a8AGI Tx agi_language: en SIP/10036-00a8AGI Tx agi_type: SIP SIP/10036-00a8AGI Tx agi_uniqueid: 1322853473.198 SIP/10036-00a8AGI Tx agi_version: 1.6.2.7 SIP/10036-00a8AGI Tx agi_callerid: 10036 SIP/10036-00a8AGI Tx agi_calleridname: 10036 SIP/10036-00a8AGI Tx agi_callingpres: 0 SIP/10036-00a8AGI Tx agi_callingani2: 0 SIP/10036-00a8AGI Tx agi_callington: 0 SIP/10036-00a8AGI Tx agi_callingtns: 0 SIP/10036-00a8AGI Tx agi_dnid: 0012127773456 SIP/10036-00a8AGI Tx agi_rdnis: unknown SIP/10036-00a8AGI Tx agi_context: privoip SIP/10036-00a8AGI Tx agi_extension: 0012127773456 SIP/10036-00a8AGI Tx agi_priority: 3 SIP/10036-00a8AGI Tx agi_enhanced: 0.0 SIP/10036-00a8AGI Tx agi_accountcode: 10036 SIP/10036-00a8AGI Tx agi_threadid: -1220478064 SIP/10036-00a8AGI Rx VERBOSE 10036 1 SIP/10036-00a8AGI Tx 200 result=1 SIP/10036-00a8AGI Rx VERBOSE 0012127773456 1 SIP/10036-00a8AGI Tx 200 result=1 SIP/10036-00a8AGI Rx VERBOSE 10036 1 SIP/10036-00a8AGI Tx 200 result=1 SIP/10036-00a8AGI Rx VERBOSE Dialling 1 SIP/10036-00a8AGI Tx 200 result=1 SIP/10036-00a8AGI Tx 200 result=1 SIP/10036-00a8AGI Rx EXEC Dial SIP/202.89.78.21/12127773456 SIP/10036-00a8AGI Tx 200 result=-1 SIP/10036-00a8AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00a8AGI Tx 200 result=1 (ANSWER) SIP/10036-00a8AGI Rx VERBOSE Status 1 SIP/10036-00a8AGI Tx 200 result=1 Regards, Kamlesh Date: Fri, 2 Dec 2011 16:26:50 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS Values Can you also paste the Asterisk Console logs around the part where AGI is dialing and after the dialing part ! make sure AGi debug is enabled as well. On Fri, Dec 2, 2011 at 4:24 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, in /etc/extension.conf [privoip] exten = _00X.,n,AGI(isdcall.php) Regards, Kamlesh Date: Fri, 2 Dec 2011 16:16:27 +0500 From: govoi...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DIALSTATUS Values Hi, How are you calling this AGI in your dialplan !!? Regards, Sammy. On Fri, Dec 2, 2011 at 3:18 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: Hello, I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) SIP/10036-0096AGI Rx VERBOSE Status 1 SIP/10036-0096AGI Tx 200 result=1 Please help me in this. Thanks, Kamlesh -- _ -- 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 -- _ -- 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] DIALSTATUS Values
In article snt142-w45a64e4743de653da591...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com wrote: I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); Shouldn't that be: $dialstatus=$agi-get_variable(DIALSTATUS); Having DIALSTATUS as a bare word might work in some versions of php, but is likely to produce a warning. Although in your case, it does appear to have worked. $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) This shows that AGI is indeed returning the value of DIALSTATUS, which is ANSWER. SIP/10036-0096AGI Rx VERBOSE Status 1 But you are not picking it up. SIP/10036-0096AGI Tx 200 result=1 Please help me in this. I'm not familiar with php-agi (I usualy write my AGI in C), but it looks like $dialstatus[data] is not the correct way to retrieve the returned value. Or else there is a bug in php-agi. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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] DIALSTATUS Values
I believe the syntax is correct because, If I use $dd=$dialstatus[code]; $agi-verbose(Status.$dd); it gives me: SIP/10036-00b2AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b2AGI Tx 200 result=1 (ANSWER) SIP/10036-00b2AGI Rx VERBOSE Status200 1 If I use $dd=$dialstatus[result]; $agi-verbose(Status.$dd); it gives me: SIP/10036-00b4AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b4AGI Tx 200 result=1 (CANCEL) SIP/10036-00b4AGI Rx VERBOSE Status1 1 but if I use $dd=$dialstatus[data]; $agi-verbose(Status.$dd); SIP/10036-00b6AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b6AGI Tx 200 result=1 (CANCEL) SIP/10036-00b6AGI Rx VERBOSE Status 1 Regards, Kamlesh To: asterisk-users@lists.digium.com From: t...@softins.co.uk Date: Fri, 2 Dec 2011 11:44:34 + Subject: Re: [asterisk-users] DIALSTATUS Values In article snt142-w45a64e4743de653da591...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com wrote: I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); Shouldn't that be: $dialstatus=$agi-get_variable(DIALSTATUS); Having DIALSTATUS as a bare word might work in some versions of php, but is likely to produce a warning. Although in your case, it does appear to have worked. $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) This shows that AGI is indeed returning the value of DIALSTATUS, which is ANSWER. SIP/10036-0096AGI Rx VERBOSE Status 1 But you are not picking it up. SIP/10036-0096AGI Tx 200 result=1 Please help me in this. I'm not familiar with php-agi (I usualy write my AGI in C), but it looks like $dialstatus[data] is not the correct way to retrieve the returned value. Or else there is a bug in php-agi. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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
Re: [asterisk-users] DIALSTATUS Values
In addition to my reply: I used to fetch the value using print_r function but that also tells that there is no value in data section. $dialstatus=$agi-get_variable(DIALSTATUS); print_r($dialstatus); SIP/10036-00b8AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b8AGI Tx 200 result=1 (CANCEL) SIP/10036-00b8AGI Rx Array SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx ( SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [code] = 200 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [result] = 1 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [data] = Regards, Kamlesh From: kamlesh_...@hotmail.com To: asterisk-users@lists.digium.com Subject: RE: [asterisk-users] DIALSTATUS Values Date: Fri, 2 Dec 2011 11:58:26 + I believe the syntax is correct because, If I use $dd=$dialstatus[code]; $agi-verbose(Status.$dd); it gives me: SIP/10036-00b2AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b2AGI Tx 200 result=1 (ANSWER) SIP/10036-00b2AGI Rx VERBOSE Status200 1 If I use $dd=$dialstatus[result]; $agi-verbose(Status.$dd); it gives me: SIP/10036-00b4AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b4AGI Tx 200 result=1 (CANCEL) SIP/10036-00b4AGI Rx VERBOSE Status1 1 but if I use $dd=$dialstatus[data]; $agi-verbose(Status.$dd); SIP/10036-00b6AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b6AGI Tx 200 result=1 (CANCEL) SIP/10036-00b6AGI Rx VERBOSE Status 1 Regards, Kamlesh To: asterisk-users@lists.digium.com From: t...@softins.co.uk Date: Fri, 2 Dec 2011 11:44:34 + Subject: Re: [asterisk-users] DIALSTATUS Values In article snt142-w45a64e4743de653da591...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com wrote: I tried to search the answer of my problem but unable to get resolution so sending to you guys. I'm using asterisk 1.6.2.7 and writing the AGI scripts using PHP. I'm unable to retrieve the DIALSTATUS value, during execution of AGI script, I get empty value. Extracts from AGI Script: #!/usr/bin/php -q #!/bin/bash ?php include_once (phpagi-2.14/phpagi.php); $agi = new AGI(); some codes for dial out $dialstatus=$agi-get_variable(DIALSTATUS); Shouldn't that be: $dialstatus=$agi-get_variable(DIALSTATUS); Having DIALSTATUS as a bare word might work in some versions of php, but is likely to produce a warning. Although in your case, it does appear to have worked. $dd=$dialstatus[data]; $agi-verbose(Status.$dd); In AGI debug, I get: SIP/10036-0096AGI Tx agi_channel: SIP/10036-0096 SIP/10036-0096AGI Tx agi_language: en SIP/10036-0096AGI Tx agi_type: SIP SIP/10036-0096AGI Tx agi_uniqueid: 1322848927.172 SIP/10036-0096AGI Tx agi_version: 1.6.2.7 SIP/10036-0096AGI Tx agi_callerid: 10036 SIP/10036-0096AGI Tx agi_calleridname: 10036 SIP/10036-0096AGI Tx agi_dnid: 0012127773456 SIP/10036-0096AGI Tx agi_rdnis: unknown SIP/10036-0096AGI Tx agi_context: privoip SIP/10036-0096AGI Tx agi_extension: 0012127773456 SIP/10036-0096AGI Rx GET VARIABLE DIALSTATUS SIP/10036-0096AGI Tx 200 result=1 (ANSWER) This shows that AGI is indeed returning the value of DIALSTATUS, which is ANSWER. SIP/10036-0096AGI Rx VERBOSE Status 1 But you are not picking it up. SIP/10036-0096AGI Tx 200 result=1 Please help me in this. I'm not familiar with php-agi (I usualy write my AGI in C), but it looks like $dialstatus[data] is not the correct way to retrieve the returned value. Or else there is a bug in php-agi. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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:
Re: [asterisk-users] DIALSTATUS Values
In article snt142-w54267269808afd17bccd5891...@phx.gbl, Kamlesh Kumar kamlesh_...@hotmail.com wrote: In addition to my reply: I used to fetch the value using print_r function but that also tells that there is no value in data section. $dialstatus=$agi-get_variable(DIALSTATUS); print_r($dialstatus); SIP/10036-00b8AGI Rx GET VARIABLE DIALSTATUS SIP/10036-00b8AGI Tx 200 result=1 (CANCEL) SIP/10036-00b8AGI Rx Array SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx ( SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [code] = 200 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [result] = 1 SIP/10036-00b8AGI Tx 510 Invalid or unknown command [Dec 3 01:18:47] ERROR[24839]: utils.c:1128 ast_carefulwrite: write() returned error: Broken pipe SIP/10036-00b8AGI Rx [data] = Well since the AGI return string does indeed contain the value, shown above as (CANCEL), that suggests there is definitely a bug in php-agi. It appears to be creating a ['data'] element, but not setting it. You will have to study the source code and work out how to fix it. I did a quick google for php agi get variable and found other reports of it not working properly, but I didn't see anyone offer a solution. It's only programming, so it shouldn't be hard to fix. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.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] Skype For Asterisk (SFA): any replacement?
On Thursday 01 December 2011, Hans Witvliet wrote: On Thu, 2011-12-01 at 14:02 +, A J Stiles wrote: On Thursday 01 December 2011, gincantalupo wrote: Hi all, any idea about how to replace Skype For Asterisk? Thank You. Giorgio 1. Migrate your Skype users over to a better product which supports proper open standards. perhaps you missed it, but the installed base of skype is unfortunately slightly (,,,) larger than the amount of peope that are using a decent product. Alas Then it's simply a bigger job than the original suggestion made it seem. When -- not if -- Skype give up supporting their anti-telecommunications product altogether, every single one of those users is going to be left in the lurch. And that might be the critical mass that brings on the revolution. We can only hope :) 2. Write to your elected representatives asking that they order Skype to release documentation on their protocols to allow third party interoperability (as is already required under EU law). 3. make it a offence to use any closed source products like skype. ;-) Huge fines, jail centences or worse. [How about an appendice to the Thora, Quran or Bible, even better, forbid it by the sharia] You may jest, but now you are seeing *EXACTLY* why closed, proprietary standards are a bad idea -- something I have been saying almost ever since Skype was first launched. Note, not necessarily closed *source*, but closed *standards*. The two are easily confused, but not quite the same. An Open Source program can only ever implement open standards, since the Source Code implicitly documents the standards. But Closed Source programs can, and often do, implement open standards. And wherever they do, then there are usually alternative, Open Source programs that do the same job. Every aspect of a program's interaction with the outside world -- communications protocols, save file formats and similar -- must be documented to the point where any competent programmer could write a program which interacts seamlessly with the application that originally generated them. That documentation may well be the Source Code for the program itself, of course; or it could just be something like the RFCs -- in which case, the will is surely out there for someone within the Open Source community to do the rest. Anything less is just blatant anti-competitive behaviour. -- AJS Answers come *after* questions. -- _ -- 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] A new hack?
On 11/26/2011 5:00 PM, C F wrote: On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 26 Nov 2011, Terry Brummell wrote: Install Configure Fail2Ban then the host will be blocked from connecting. And no, it's not new. I don't need Fail2Ban, thank you. But your advice might be useful to others. Why is that? Even if they don't compromise an account they are still using your bandwidth and resources on your machine. How is using Fail2Ban less resource intensive then me writing (by hand) iptable rules? Also, since both methods involve the use of iptables, where exactly is the bandwidth savings? -- Jim Lucas -- _ -- 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] A new hack?
Fail2ban assumes that #1 your environment is (wide) open and #2 you will need to update iptables on an instant response to attack basis. If you are open enough, even fail2ban isn't going to really help. If you have a sufficiently written set of iptables rules (or you aren't allowing external SIP/TCP/UDP traffic) you shouldn't (just my opinion) need fail2ban at all. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jim Lucas Sent: Friday, December 02, 2011 10:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] A new hack? On 11/26/2011 5:00 PM, C F wrote: On Sat, Nov 26, 2011 at 7:50 AM, Gordon Henderson gordon+aster...@drogon.net wrote: On Sat, 26 Nov 2011, Terry Brummell wrote: Install Configure Fail2Ban then the host will be blocked from connecting. And no, it's not new. I don't need Fail2Ban, thank you. But your advice might be useful to others. Why is that? Even if they don't compromise an account they are still using your bandwidth and resources on your machine. How is using Fail2Ban less resource intensive then me writing (by hand) iptable rules? Also, since both methods involve the use of iptables, where exactly is the bandwidth savings? -- Jim Lucas -- _ -- 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] CSipSimple audio issue with DAHDI/IAX2 calls
I've just connected my new Android (Motorola RAZR) phone to Asterisk using CSipSimple and have discovered that on any call between CSipSimple and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will hear a rhythmic tapping as if my voice stream is being chopped up in equal parts about every 500ms or so. I can always hear the remote party without issue, regardless of the channel type. The issue occurs only on connections to DAHDI channels (even those that don't pass through the PSTN), and IAX2 connections to remote Asterisk servers. This issue occurs whether I am using WiFi, 3G or 4G connections on the Android. This does NOT occur on any SIP channels, local to my Asterisk box, or to others. I've investigated changing just about every setting on the Android with no resolution. It seems like some sort of timing issue and is strange to me that this issue is confined to DAHDI and IAX2 channels, but I'm no expert. I have tested using only res_timing_dadhi.so since I have the card, but that did not help either. Would anyone be willing to point me in the right direction for resolving this issue? Please let me know if any more information is required. Thanks in advance. -A I am currently using the following on a Fedora 15 x86_64 system: Asterisk 1.8.7.1 built by mockbuild @ x86-13.phx2.fedoraproject.org on a x86_64 running Linux on 2011-10-17 21:42:11 UTC ]# cat /proc/dahdi/* Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) 1 WCTDM/4/0 FXOKS (In use) (EC: OSLEC - INACTIVE) 2 WCTDM/4/1 FXOKS 3 WCTDM/4/2 FXSKS (In use) (EC: OSLEC - INACTIVE) *CLI module show like timing Module Description Use Count res_timing_dahdi.soDAHDI Timing Interface 0 res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 *CLI core show settings PBX Core settings - Version: 1.8.7.1 Build Options: LOADABLE_MODULES Maximum calls: Not set Maximum open file handles: Not set Verbosity: 3 Debug level: 0 Maximum load average:0.00 Minimum free memory: 0 MB Startup time:10:23:07 Last reload time:10:23:07 System: Linux/2.6.32-131.2.1.el6.x86_64 built by mockbuild on x86_64 2011-10-17 21:42:11 UTC Default language:en Language prefix: Enabled User name and group: / Executable includes: Disabled Transcode via SLIN: Enabled Internal timing: Enabled Transmit silence during rec: Disabled Generic PLC: Enabled -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] A new hack?
On Fri, 2 Dec 2011, Jim Lucas wrote: How is using Fail2Ban less resource intensive then me writing (by hand) iptable rules? It depends on how you define resources and how much of those resources you have. Gordon (based on my understanding of his posts) does a lot of Asterisk systems on very limited hardware hosts. His approach uses iptables features to limit the number of SIP INVITES and REGISTERS per second per IP address. Thus, Gordon's approach is more responsive (since it doesn't require periodic log file scanning) and requires less hardware resources (since it doesn't depend on running relatively 'slothish' resource intensive script interpreters like Perl or PHP periodically). If you have limited admin skills and more hardware resources, F2B makes sense. If you have more admin skills and limited hardware resources, Gordon's approach makes more sense. Personally, I find any approach that tracks log files 'hackish' but if you centralize your logging (which I always do) it does allow you to detect patterns of abuse across multiple hosts. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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] A new hack?
On 12/2/2011 12:44 PM, Steve Edwards wrote: On Fri, 2 Dec 2011, Jim Lucas wrote: How is using Fail2Ban less resource intensive then me writing (by hand) iptable rules? It depends on how you define resources and how much of those resources you have. Gordon (based on my understanding of his posts) does a lot of Asterisk systems on very limited hardware hosts. His approach uses iptables features to limit the number of SIP INVITES and REGISTERS per second per IP address. Thus, Gordon's approach is more responsive (since it doesn't require periodic log file scanning) and requires less hardware resources (since it doesn't depend on running relatively 'slothish' resource intensive script interpreters like Perl or PHP periodically). If you have limited admin skills and more hardware resources, F2B makes sense. If you have more admin skills and limited hardware resources, Gordon's approach makes more sense. Personally, I find any approach that tracks log files 'hackish' but if you centralize your logging (which I always do) it does allow you to detect patterns of abuse across multiple hosts. Now this, I would say was very well put. As always, just my opinion. JohnM -- _ -- 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] IAX - An informative question
Hello all, I recently found this when looking an IAX trunk: context=* Does it have a special meaning or is it the same like 'default'? Thanks, Elder -- _ -- 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] Where to download sample video file for Asterisk 1.8x playback?
Hello, I have been trying to playback a video file via Playback() in Asterisk 1.8.7.1 but the file format seems to fail. [2011-12-02 18:46:24] WARNING[7665]: file.c:653 ast_openstream_full: File /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 does not exist in any format [2011-12-02 18:46:24] WARNING[7665]: file.c:959 ast_streamfile: Unable to open /etc/asterisk/cp-10fps-QCIF-20Kbps.h263 (format 0x4 (ulaw)): No such file or directory The file of course exists and it's chowned to asterisk.asterisk. I think it's a file format issue. So, I appreciate it someone can give me a link to a file or maybe point me a universal convertor (open-source or linux based software) that can convert my videos to Asterisk readable format. Thanks, -- _ -- 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 can I decipher password in SIP Packet?
I am receiving requests to register to my Asterisk extensions. I have the full SIP packets. I also do see what extension is being tried to be registered. Is there ANY WAY to know what password is being attempted? I think the appropriate term would be decode the base64 response I get from the client. Here is what I get in the SIP packet from the client: * * *Authorization: Digest username=4456678, realm=asterisk, nonce=67461340, uri=sip:mailbox, response=5a9a5f2b527ca9687c8f75705e6a2d25, algorithm=MD5* Using a base64 decoder I get this:* *å¯Zåý›çnÜkÞ¼íÏ ïžôåîšÙݹ from the response above. Of course, that is not the plain password. So, is that encrypted? How can I can I decrypt it? Thanks, On Mon, Nov 28, 2011 at 12:48 AM, asterisk jobs asteriskcod...@gmail.comwrote: Hello, I am receiving requests to register to my Asterisk extensions. I have the full SIP packets. I also do see what extension is being tried to be registered. Is there ANY WAY to know what password is being attempted? Thanks, Peter -- _ -- 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] DHCP Option 43 and pfSense + Asterisk
Hi, Has anyone succeded using DHCP Option 43 and Aastra phones to set the configuration server from a pfSense router or any other router? Sorry, if not directly related to Asterisk but I am sure the collective knowledge will pay off. I am specifically wondering what the Number, Type and Value should be in Additional BOOTP/DHCP Options under pfSense 2.0 Thanks a lot -- _ -- 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] Max channel analyser from asteriskcdrdb?
Hello, Is there a php or any other program to analyse Asterisk CDR which is stored in asteriskcdrdb. I want to know outbound and inbound channels and not the internal calls channels as well which is what CDR Stats does currently. It doesn't differentiate between those. Someone might have done a custom script to find out their monthly inbound / outbound peak lines? I appreciate a guide on this. *FreePBX reporting and CDR Stats from Areski is not the answer to my issue. Thanks, -- _ -- 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 can I decipher password in SIP Packet?
On 12/02/2011 05:24 PM, asterisk jobs wrote: I am receiving requests to register to my Asterisk extensions. I have the full SIP packets. I also do see what extension is being tried to be registered. Is there ANY WAY to know what password is being attempted? I think the appropriate term would be decode the base64 response I get from the client. Here is what I get in the SIP packet from the client: * * *Authorization: Digest username=4456678, realm=asterisk, nonce=67461340, uri=sip:mailbox, response=5a9a5f2b527ca9687c8f75705e6a2d25, algorithm=MD5* Using a base64 decoder I get this:**å¯Zåý›çnÜkÞ¼íÏ ïžôåîšÙݹ from the response above. Of course, that is not the plain password. So, is that encrypted? How can I can I decrypt it? As the Authorization header clearly states, this value is created using an MD5 Digest (hash). Since it is a digest function, it is not reversible. It is impossible to recover the password that was used during the calculation of the response value (although given enough time and CPU resources, it is possible go through a massive list of possibilities and try each one until you find one that matches). -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.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] skype connect early media
hi folks. when i use regular PSTN(sip phone - asterisk - PRI) to call certain numbers and when that number is unavailable. i usually hear an early media message saying blahblah is unavailable, please try again. but when i use skype connect(sip phone - asterisk - skype connect). i just hear ring back tone for about 20 seconds and then become fast busy. is there any setting i'm unaware of when setting up sip w/ skype connect? any suggestions would be appreciated. -- Edwin Lam edwin@officegeneral.com Systems Engineer, OfficeWyze, Inc. Ph: +1 415 439 4988 Fax: +1 415 283 3370 http://pgpkeys.mit.edu:11371/pks/lookup?op=getsearch=0xD6506D20 -- _ -- 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 can I decipher password in SIP Packet?
As the Authorization header clearly states, this value is created using an MD5 Digest (hash). Since it is a digest function, it is not reversible. It is impossible to recover the password that was used during the calculation of the response value (although given enough time and CPU resources, it is possible go through a massive list of possibilities and try each one until you find one that matches). Thanks. Based on above, I am getting that Asterisk also runs MD5 algorithm on the password and then matches the two hash digests to see if they are good or not. Is that all happens? or is there an encryption involved as well? Chance of collision of 1^128? 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] A new hack?
On Fri, Dec 2, 2011 at 12:44 PM, Steve Edwards asterisk@sedwards.com wrote: Gordon (based on my understanding of his posts) does a lot of Asterisk systems on very limited hardware hosts. His approach uses iptables features to limit the number of SIP INVITES and REGISTERS per second per IP address. A very narrow solution to a fairly narrow attack surface and surely isn't applicable to any medium to large scale solutions. Thus, Gordon's approach is more responsive (since it doesn't require periodic log file scanning) and requires less hardware resources (since it doesn't depend on running relatively 'slothish' resource intensive script interpreters like Perl or PHP periodically). So Fail2Ban is inefficient on how it reads log files? If so, that could be an informed criticism of Fail2Ban. Personally, I find any approach that tracks log files 'hackish' but if you centralize your logging (which I always do) it does allow you to detect patterns of abuse across multiple hosts. Others would say that not using IPS/IDS/adaptive sec appliances is hackish but I'm not one of those. There are very efficient ways to read log files even with Perl on hardware no bigger than my Dockstar when coded properly, so reading log files isn't hackish. Looking at advanced threats that are encrypted or otherwise located within legitimately large streams of UDP and TCP traffic are not going to lend themselves to some simpleton IP/port/rate iptables rule or even more complex iptables view into the data. The application log might be the ONLY place to correlate events. Good luck doing that with iptables alone. -- _ -- 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] google voice calling dial plan question.
When a caller calls my google voice phone number, I must answer, wait and
press one to accept. Sometimes even that does not work.
I have tried a few different things to get asterisk to place the call in an
answered state and send the DTMF 1 with the Dial macro.
I found Malcom Davenports wiki page regarding Google calling which has been
very helpful in troubleshooting the issue.
https://wiki.asterisk.org/wiki/display/AST/Calling+using+Google?focusedCommentId=18415969#comment-18415969
I'm sure that I'm close to getting things working properly.
Here's my config.
##jabber.conf##
[general]
debug=no
autoprune=no
autoregister=yes
[whitehat238]
type=client
serverhost=talk.google.com
username=whitehat...@gmail.com/Talk
secret=password
port=5222
usetls=yes
usesasl=yes
status=Available
statusmessage=No Information Available
timeout=100
keepalive=yes
##gtalk.conf##
[general]
allowguest=yes
context=googlein
stunaddr=stun01.sipphone.com
[guest]
disallow=all
allow=ulaw
connection=whitehat238
context=googlein
##extensions_custom.conf##
exten = whitehat...@gmail.com
,1,Set(CALLERID(name)=${CUT(CALLERID(name),@,1)})
exten = whitehat...@gmail.com,n,GotoIf($[${CALLERID(name):0:2} !=
+1]?notrim)
exten = whitehat...@gmail.com,n,Set(CALLERID(name)=${CALLERID(name):2})
exten = whitehat...@gmail.com
,n(notrim),Set(CALLERID(number)=${CALLERID(name)})
exten = whitehat...@gmail.com,n,Answer
exten = whitehat...@gmail.com,n,Wait(1)
exten = whitehat...@gmail.com,n,SendDTMF(1)
exten = whitehat...@gmail.com,n,Goto(from-trunk,5025551212,1)
[gvoice-whitehat238]
exten = _X.,1,Dial(Gtalk/whitehat238/+${EXTEN}@voice.google.com)
exten = _X.,n,Noop(GVoice Call to ${EXTEN} failed)
exten = h,1,Macro(hangupcall,)
I have a working inbound route which rings an internal extension (7008)
when calling the GV number. I can also make outbound calls to any number
using the GV trunk.
I found this page (Link to Michigan telephone blog) which helped me get
everything setup initially and included a shell script that made it easy to
generate the configuration.
http://michigantelephone.wordpress.com/2011/01/20/a-bash-script-to-assist-asterisk-1-8freepbx-2-8-users-in-adding-new-google-voice-accounts/
The author explains the config in more detail and why he choose to write it
the way he did.
I have tried using the alternative method of sending the DTMF 1 tone by
changing the last block as follows:
[gvoice-whitehat238]
exten = _X.,1,Dial(Gtalk/whitehat238/+${EXTEN}@voice.google.com,D(:1))
exten = _X.,n,Noop(GVoice Call to ${EXTEN} failed)
exten = h,1,Macro(hangupcall,)
However, that did not work.
I just need a little advice on how to write the dial plan. I still have
much to learn about asterisk, and appreciate any advice.
Thanks,
--
_
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