Hi, I'm getting errors while originating a call through AMI. [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() returned error: Broken pipe Asterisk version 11.0.1 OS:?CentOS release 5.8 (Final) //manager.conf settings [faheem] secret =f at xxxxxx permit=127.0.0.1/255.255.255.255 read = system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan write = system,call,agent,user,config,command,reporting,originate ///AMI script <?php $sys_ip = "127.0.0.1"; $User_str = "faheem"; $Secret_str = "f at h33m112xxxxxx"; $phoneNumb = 1234; $dialNumb = ?4567; $spoofNumb = 786; $context = "xxxxx-xxxxx"; $oSocket = fsockopen($sys_ip, 5038, $errnum, $errdesc) or die("Connection to host failed"); fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Username: $User_str\r\n"); fputs($oSocket, "Secret: $Secret_str\r\n\r\n"); fputs($oSocket, "Events: off\r\n\r\n"); fputs($oSocket, "Action: originate\r\n"); fputs($oSocket, "Channel: SIP/testTrunk/$phoneNumb\r\n"); fputs($oSocket, "Exten: $dialNumb\r\n"); fputs($oSocket, "Context: $context\r\n"); fputs($oSocket, "Priority: 1\r\n\r\n"); fputs($oSocket, "Timeout: 10000\r\n"); fputs($oSocket, "CallerId: $spoofNumb\r\n"); fputs($oSocket, "Async: false\r\n"); fputs($oSocket, "Action: Logoff\r\n\r\n"); echo "originate executed"; fclose($oSocket); ?> Can any one please help me over it. Thank you! ? Muhammad Faheem -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121212/713031dc/attachment.htm>
Christopher Harrington
2012-Dec-12 21:22 UTC
[asterisk-users] Asterisk 11 originate errors
I've observed the same behavior. This is what happens when you close the socket before logoff is completed. You need to wait until the logoff action is completed before closing the socket (your fclose() call). Alternately, use a proxy or a daemon that will sit between your script and the AMI and keep the socket open. On Wed, Dec 12, 2012 at 12:44 PM, Faheem <faheem_imt at yahoo.com> wrote:> Hi, > I'm getting errors while originating a call through AMI. > [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() > returned error: Broken pipe > [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() > returned error: Broken pipe > [Dec 12 21:18:35] ERROR[8661]: utils.c:1236 ast_careful_fwrite: fwrite() > returned error: Broken pipe > Asterisk version 11.0.1 > OS: CentOS release 5.8 (Final) > > //manager.conf settings > [faheem] > secret =f at xxxxxx > permit=127.0.0.1/255.255.255.255 > read > system,call,log,verbose,agent,user,config,dtmf,reporting,cdr,dialplan > write = system,call,agent,user,config,command,reporting,originate > > ///AMI script > <?php > > $sys_ip = "127.0.0.1"; > $User_str = "faheem"; > $Secret_str = "f at h33m112xxxxxx"; > $phoneNumb = 1234; > $dialNumb = 4567; > $spoofNumb = 786; > $context = "xxxxx-xxxxx"; > > $oSocket = fsockopen($sys_ip, 5038, $errnum, $errdesc) or die("Connection > to host failed"); > fputs($oSocket, "Action: login\r\n"); > fputs($oSocket, "Username: $User_str\r\n"); > fputs($oSocket, "Secret: $Secret_str\r\n\r\n"); > fputs($oSocket, "Events: off\r\n\r\n"); > fputs($oSocket, "Action: originate\r\n"); > fputs($oSocket, "Channel: SIP/testTrunk/$phoneNumb\r\n"); > fputs($oSocket, "Exten: $dialNumb\r\n"); > fputs($oSocket, "Context: $context\r\n"); > fputs($oSocket, "Priority: 1\r\n\r\n"); > fputs($oSocket, "Timeout: 10000\r\n"); > fputs($oSocket, "CallerId: $spoofNumb\r\n"); > fputs($oSocket, "Async: false\r\n"); > fputs($oSocket, "Action: Logoff\r\n\r\n"); > echo "originate executed"; > fclose($oSocket); > > ?> > > > Can any one please help me over it. > Thank you! > > Muhammad Faheem > > > > -- > _____________________________________________________________________ > -- 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 >-- -Chris Harrington ACSDi Office: 763.559.5800 Mobile Phone: 612.326.4248 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20121212/0e1a55c8/attachment.htm>