Re: [asterisk-users] Getting 603 Declined after AGI execution
Hi Alejandro, As per my a2billing knowledge it's termination issue. Might be your call balance is finished please check and then test again. On Thu, May 19, 2011 at 8:40 PM, Alejandro Mejia Evertsz ame...@gua.netwrote: Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=in extensions.conf== [from-customerX] exten = _X.,1,Set(CDR(accountcode)=) ;Here I change the accountcode depending on each customer exten = _X.,2,Set(CALLERID(dnid)=${EXTEN}) exten = _X.,3,Goto(a2billing|${EXTEN}|1) [from-customerY] exten = _X.,1,Set(CDR(accountcode)=) exten = _X.,2,Set(CALLERID(dnid)=${EXTEN}) exten = _X.,3,Goto(a2billing|${EXTEN}|1) [a2billing] exten = _X.,1,DeadAGI(a2billing.php|1) exten = _X.,2,Hangup(34) ;= A2Billing authenticates and routes the call properly, but when the termination gateway for the destination dialed by the customer rejects the call, my Asterisk box sends 603 Declined to the customer. It also happens when A2Billing doesn't find any route for that destination, in which it should return 404 Not Found, but returns 603 Declined instead. I tried to force every rejected attempt with 503 Service Unavailable putting the Hangup(34) you see on my config, but it never seems to get there. The last thing I see on CLI running in verbose is: -- AGI Script a2billing.php completed, returning 0 Is there anything I could do to return a different cause than 603 Declined? I posted the same question on A2Billing's forum, but had no luck. Thanks in advance, Alejandro Mejia -- _ -- 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 -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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] difference between SIP peer and SIP user ?
Hi list, I am confuse about these CLI commands *sip show users sip show peers* Can someone clear my doubt . what are the difference between them? - Thanks and regards Virendra Bhati +91-9172341457 -- _ -- 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 user SIP realtime option
Hi List, After read the link http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip. I changes all information in below conf files *res_mysql.conf* [mpathsala] dbhost = localhost dbname = mpathsala dbuser = root dbpass = dbport = 3306 *extconfig.conf* sipusers = mysql,mpathsala,sip_buddies sippeers = mysql,mpathsala,sip_buddies *mysql* make same table sip_buddies in mpathsala database. insert 1 row into database [image: Full Texts]http://192.168.193.69/phpmyadmin/sql.php?db=mpathsalatable=sip_buddiessql_query=SELECT+%2A+FROM+%60sip_buddies%60goto=sql.php%3Fdb%3Dmpathsala%26amp%3Btable%3Dsip_buddies%26amp%3Btoken%3D0ba5913eba31b6657e85f0a1ffec66de%26amp%3Bsql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560dontlimitchars=1token=0ba5913eba31b6657e85f0a1ffec66de id name host nat type accountcode amaflags call-limit callgroup callerid cancallforward canreinvite context defaultip dtmfmode fromuser fromdomain insecure language mailbox md5secret deny permit mask musiconhold pickupgroup qualify regexten restrictcid rtptimeout rtpholdtimeout secret setvar disallow allow fullcontact ipaddr port regserver regseconds lastms username defaultuser subscribecontext useragent [image: Edit]http://192.168.193.69/phpmyadmin/tbl_change.php?db=mpathsalatable=sip_buddiestoken=0ba5913eba31b6657e85f0a1ffec66deprimary_key=+%60sip_buddies%60.%60id%60+%3D+1sql_query=SELECT+%2A+FROM+%60sip_buddies%60goto=sql.php [image: Delete]http://192.168.193.69/phpmyadmin/sql.php?db=mpathsalatable=sip_buddiestoken=0ba5913eba31b6657e85f0a1ffec66desql_query=DELETE+FROM+%60sip_buddies%60+WHERE+%60sip_buddies%60.%60id%60+%3D+1+LIMIT+1zero_rows=The+row+has+been+deletedgoto=sql.php%3Fdb%3Dmpathsala%26table%3Dsip_buddies%26token%3D0ba5913eba31b6657e85f0a1ffec66de%26sql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560%26zero_rows%3DThe%2Brow%2Bhas%2Bbeen%2Bdeleted%26goto%3Dsql.php%3Fdb%3Dmpathsala%26amp%3Btable%3Dsip_buddies%26amp%3Btoken%3D0ba5913eba31b6657e85f0a1ffec66de%26amp%3Bsql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560 1 300 dynamic no friend *NULL* *NULL* *NULL* *NULL* 100 yes yes *NULL* *NULL * *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* * NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* all g729;ilbc;gsm;ulaw;alaw 0 *NULL* 0 0 *NULL* *NULL* *sip.conf * [general] rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=yes After all when I check on CLI then I will get *cent70*CLI sip show peers* Name/username HostDyn Nat ACL Port Status Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] *cent70*CLI sip show users* Username Secret Accountcode Def.Context ACL NAT Why SIP/300 is not display here ?? Please help me I want to learn asterisk real-time concept to make my server real-time. - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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 know how many calls are on hold
Hi Yes it show none until we accept the call but after receive the call and then we check the status is too complected. 1) before answer the call **CLI core show channels concise* SIP/200-0007!office!200!1!Ringing!AppDial!(Outgoing Line)!200!!3!11!(None)!1305982734.7 SIP/300-0006!office!200!2!Up!Dial!SIP/200,45,tTrkK!300!!3!12!(None)!1305982734.6 SIP/200-0005!office!!1!Up!AppDial!(Outgoing Line)!200!!3!88!SIP/100-0004!1305982658.5 SIP/100-0004!office!200!2!Up!Dial!SIP/200,45,tTrkK!100!!3!88!SIP/200-0005!1305982658.4 2) After answer the call (info is different due to different log info) SIP/200-0003!office!!1!Up! AppDial!(Outgoing Line)!200!!3!35!SIP/300-0002!1305981842.3 SIP/300-0002!office!200!2!Up!Dial!SIP/200,45,tTrkK!300!!3!36!SIP/200-0003!1305981841.2 SIP/200-0001!office!!1!Up!AppDial!(Outgoing Line)!200!!3!77!SIP/100-!1305981800.1 SIP/100-!office!200!2!Up!Dial!SIP/200,45,tTrkK!100!!3!77!SIP/200-0001!1305981800.0 Please help me on that. On Wed, May 18, 2011 at 1:47 PM, Tiago Geada tiago.ge...@gmail.com wrote: core show channels concise Those with '(None)' haven't been briged yet. On 17 May 2011 15:16, virendra bhati virbh...@gmail.com wrote: hi list, please help me how to know how many calls are on hold. -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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 -- - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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 user SIP realtime option
Hi Trying exclude rtcachefriends from your sip.conf and include the field rtchachefriends in table sip_buddies. And exclude the field qualify from sip_buddies. Set YES in field rtcachefriends. Att, Rafael Saraiva 2011/5/21 virendra bhati virbh...@gmail.com Hi List, After read the link http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip . I changes all information in below conf files *res_mysql.conf* [mpathsala] dbhost = localhost dbname = mpathsala dbuser = root dbpass = dbport = 3306 *extconfig.conf* sipusers = mysql,mpathsala,sip_buddies sippeers = mysql,mpathsala,sip_buddies *mysql* make same table sip_buddies in mpathsala database. insert 1 row into database [image: Full Texts]http://192.168.193.69/phpmyadmin/sql.php?db=mpathsalatable=sip_buddiessql_query=SELECT+%2A+FROM+%60sip_buddies%60goto=sql.php%3Fdb%3Dmpathsala%26amp%3Btable%3Dsip_buddies%26amp%3Btoken%3D0ba5913eba31b6657e85f0a1ffec66de%26amp%3Bsql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560dontlimitchars=1token=0ba5913eba31b6657e85f0a1ffec66de id name host nat type accountcode amaflags call-limit callgroup callerid cancallforward canreinvite context defaultip dtmfmode fromuser fromdomain insecure language mailbox md5secret deny permit mask musiconhold pickupgroup qualify regexten restrictcid rtptimeout rtpholdtimeout secret setvar disallow allow fullcontact ipaddr port regserver regseconds lastms username defaultuser subscribecontext useragent [image: Edit]http://192.168.193.69/phpmyadmin/tbl_change.php?db=mpathsalatable=sip_buddiestoken=0ba5913eba31b6657e85f0a1ffec66deprimary_key=+%60sip_buddies%60.%60id%60+%3D+1sql_query=SELECT+%2A+FROM+%60sip_buddies%60goto=sql.php [image: Delete]http://192.168.193.69/phpmyadmin/sql.php?db=mpathsalatable=sip_buddiestoken=0ba5913eba31b6657e85f0a1ffec66desql_query=DELETE+FROM+%60sip_buddies%60+WHERE+%60sip_buddies%60.%60id%60+%3D+1+LIMIT+1zero_rows=The+row+has+been+deletedgoto=sql.php%3Fdb%3Dmpathsala%26table%3Dsip_buddies%26token%3D0ba5913eba31b6657e85f0a1ffec66de%26sql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560%26zero_rows%3DThe%2Brow%2Bhas%2Bbeen%2Bdeleted%26goto%3Dsql.php%3Fdb%3Dmpathsala%26amp%3Btable%3Dsip_buddies%26amp%3Btoken%3D0ba5913eba31b6657e85f0a1ffec66de%26amp%3Bsql_query%3DSELECT%2B%252A%2BFROM%2B%2560sip_buddies%2560 1 300 dynamic no friend *NULL* *NULL* *NULL* *NULL* 100 yes yes *NULL* * NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL * *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* *NULL* all g729;ilbc;gsm;ulaw;alaw 0 *NULL* 0 0 *NULL* *NULL* *sip.conf * [general] rtcachefriends=yes rtsavesysname=yes rtupdate=yes rtautoclear=yes After all when I check on CLI then I will get *cent70*CLI sip show peers* Name/username HostDyn Nat ACL Port Status Realtime 0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline] *cent70*CLI sip show users* Username Secret Accountcode Def.Context ACL NAT Why SIP/300 is not display here ?? Please help me I want to learn asterisk real-time concept to make my server real-time. - Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -- _ -- 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 -- Att, Rafael Saraiva -- _ -- 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 know how many calls are on hold
On Tue, May 17, 2011 at 10:16 AM, virendra bhati virbh...@gmail.com wrote: hi list, please help me how to know how many calls are on hold. If they are SIP channels you can use: sip show inuse The last column are calls on hold. Ryan -- _ -- 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] Polycom IP335 3.3.1 Call Waiting
On Thu, May 19, 2011 at 1:24 PM, Ryan Wagoner rswago...@gmail.com wrote: I updated my phones to the UCS 3.3.1 firmware a few months back. The scenario is I place a call and receive an incoming call. With 3.3.1 the screen will show call 1/2 and I have to press the down arrow to see the caller name / number. Has anybody else noticed this with 3.3.1? I had thought with 3.2.4 it would automatically show call waiting name and number without pressing any keys. It could be possible I missed a setting, but I didn't see anything in the admin guide. Ryan For those wondering it appears to be a bug in 3.2.5 and later versions. I downgraded to 3.2.4 and the caller id for the incoming call waiting call will show for 10 seconds as described in the Polycom user guide. This only effects those with IP33x model phones. Ryan -- _ -- 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