Re: [asterisk-users] Call Transfer not working
Thank you so much. OK, I understood that to transfer the call t is usually used, is it right? And I should write so in my last mail. t and T are described with same sentences in official wiki... Regards, Takehiro Matsushima 2012/4/9 Chris Bagnall aster...@lists.minotaur.cc: On 9/4/12 3:04 am, Takehiro Matsushima wrote: // I don't know what's difference t and T. T allows the caller to transfer. t allows the called user to transfer. You very rarely want Tt - since I doubt you want an incoming caller to be able to transfer their call all over the place. You usually want t on incoming calls and T on outgoing calls. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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] Call Transfer not working
Hi. Maybe you forgotten specify to allow the transferring a call. Try with tT options in Dial() in extensions.conf. // I don't know what's difference t and T. -- Takehiro Matsushima takehiro.dream...@gmail.com 2012/4/7 Rizwan Hisham rizwanhas...@gmail.com: Hi All, I am using asterisk 1.8.11 on centos 5. I have realtime sip peers with dtmf setting rfc2833 and inband. I have also enabled blind and attended transfer features in features.conf but still call transfers dont work. I have setup transfer feature in past but i dont think i am missing anything this time. I just dont have any clue why its not working. I have tried using ATAs and softphones but cant make it to work. Can anyone help? Am I missing anything? features show output: === Builtin Feature Default Current --- --- --- Pickup *8 *8 Blind Transfer # #1 Attended Transfer *2 One Touch Monitor Disconnect Call * * Park Call One Touch MixMonitor == -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.com W: www.axvoice.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 -- _ -- 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] Replicating SIP registration Info between active to standby
Hi. How about place backend DB on shared disk, or make replication between them? 2012/02/24 13:58 Muro, Sam resea...@businesstz.com: I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a time. How can I copy sip registration information from Active Server to Standby Server Sam -- _ -- 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] Replicating SIP registration Info between active to standby
Hi, Sam Yes, I’m understanding that the backend of AstDB is bdb or sqlite(since asterisk10). So, I suggested to place files of them on shared disk (like DRBD). regards, takehiro 2012/2/24 Muro, Sam resea...@businesstz.com: Hi Takehiro Are you suggesting sharing the AstDB ? Sam Takehiro Matsushima wrote: Hi. How about place backend DB on shared disk, or make replication between them? 2012/02/24 13:58 Muro, Sam resea...@businesstz.com: I have a scenario whereby two servers are acting in active-standby mode. In case the active server fail, the shared IP is activated on standby server for continuity. However, SIP phones (all are Polycom) takes quite a long time to register to the Standby Server (up to 1-10min). While Polycom allow double registration, we would like to make it simple by provision only one registration server at a time. How can I copy sip registration information from Active Server to Standby Server Sam -- _ -- 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] Realtime Registration
Hi. I'd configured realtime registration, but configuration was not applied when I changed a row of sippeers table. To apply, 'sip reload' was needed (in Asterisk 1.8.0). (On 12/08/2011 03:42), Andrew O. Zhukov wrote: No secrets :) SELECT * FROM sippeers WHERE name = '105680' AND host = 'dynamic' name|type|username|secret|fromuser|fromdomain|nat|context|canreinvite|disallow|allow|host|insecure|port|ipaddr|outboundproxy 105680|peer|testbutton2|XXX||button.ipshka.com:5060|no|button|no|all|speex;ulaw;alaw;g729;ilbc;g726;g726aal2;slin;lpc10;adpcm;g723|dynamic|port,invite|5060 ||ipshka.com On 12/07/2011 08:04 PM, Jonathan Rose wrote: [Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer '105680' is trying to register, but not configured as host=dynamic Going on this, I'd say you probably tried to specify the host with a static IP address or a host name. If that's the case, you can't register, because that would be against the whole point of registering in the first place. You should probably post the DB entry for this peer to this thread to make things simpler... if it doesn't contain sensitive data. Of course, you can censor that out too. - Original Message - From: Andrew O. Zhukovgn...@telegroup.com.ua To: asterisk-users@lists.digium.com Sent: Wednesday, December 7, 2011 11:56:20 AM Subject: [asterisk-users] Realtime Registration [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:821 pgsql_reconnect: Postgresql RealTime: Everything is fine. [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:201 realtime_pgsql: Postgresql RealTime: Result=0x12e4edd0 Query: SELECT * FROM sippeers WHERE name = '105680' AND host = 'dynamic' [Dec 7 19:31:18] DEBUG[4217]: res_config_pgsql.c:209 realtime_pgsql: Postgresql RealTime: Found 1 rows. [Dec 7 19:31:18] ERROR[4217]: chan_sip.c:9940 register_verify: Peer '105680' is trying to register, but not configured as host=dynamic Any suggestions??? Asterisk 1.4.42 -- _ -- 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
