[Asterisk-Users] Asterisk out of Media Path - Call Park
Hi all, Can i make Asterisk stay out of the media path forcall park feature?In the 'sip.conf' i made canreinvite=yes in the general sectionbut it does not seem to take effect. I don't see any reason for Asterisk to withhold sending re-invite. I am testing the call park in the single LAN,both on caller side and reciever side i am using X-Lite phones. Any suggestions?? Thanks, Sharath Chandra ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Oneway Audio
Hi all, I did not get this error in Asterisk 1.2.5 release. I am testing on Asterisk SVN-trunk-r15187 to avail the PARKEDAT variable. - I park the call using ParkAndAnnounce - plays moh. - accept the call using ParkedCall The following errors are coming on the console and there is oneway audio - no audio after Music-On-Hold at caller's side. Please advice. I am testing using cisco 7902 phones and using cisco 2800 router. Codec is g711ulaw regards, -- Executing ParkedCall(SIP/192.168.50.2-09cbd610, 366) -- Channel SIP/192.168.50.2-09cbd610 connected to parked call 366Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:16 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 29 17:59:17 WARNING[13774]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WARNINGS For SIP call
I am getting the following warnings on the Asterisk when i try a call parking scenario. I use Ciso 7920 phones and Cisco2800 Executing ParkedCall(SIP/192.168.50.2-088cde00, 366) in new stack -- Channel SIP/192.168.50.2-088cde00 connected to parked call 366Mar 28 17:07:36 WARNING[10027]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 28 17:07:36 WARNING[10027]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 28 17:07:36 WARNING[10027]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) Mar 28 17:07:36 WARNING[10027]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4)Mar 28 17:07:36 WARNING[10027]: chan_sip.c:2692 sip_write: Asked to transmit frame type 64, while native formats is 4 (read/write = 4/4) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] On ParkAndAnnounce and parking lot
I am using ParkAndAnnounce to Park the call and explicitly retrieving using ParkedCall app in the dial plan. I am trying to guess the parking lot being used in a particular call by incrementing a counter just before the ParkAndAnnounce and decrement the counter just before the ParkedCall. I am not sure if this is the right way to do. What i want to know is when is the parking lot released for recycling. Is is a safe assumption to decrement just beforeParkedCall. Thanks, Sharath ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial Out IVR
How can i configure the following scenario, - User 'A' dials into Asterisk, - Asterisk puts user 'A' on hold - Dials Out to User 'B' - Consults user B' if he wants to take the call (Press 1)or divert to voicemail (press 2) - Depending on the option chosen,either user A' call is bridged with the out call or transfered to voicemail. Thanks, Sharath Chandra ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1
Thanks Moj. But i need to connect to MySQL. Could this be a problemwith C libraries that i am using. Regards, Sharath On 3/8/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: This may not be the applicable solution, but if you're not using themysql config capabilities, add noload = res_config_mysql.so to modules.confMojSharath Chandra wrote: Hi all, I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red Hat linux box( Linux version 2.4.20-8smp). I was able to compile both the software but when i start Asterisk, it exits with the following dump. Error Text Start= [res_crypto.so] = (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup'[res_config_mysql.so]Mar6 05:18:23 WARNING[12779]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: __stack_chk_fail Mar6 05:18:23 WARNING[12779]: loader.c:554 load_modules: Loading module res_config_mysql.so failed! End=== Can someone suggest a solution. Regards, Sharath Chandra ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users --Mojo [EMAIL PROTECTED]Office Manger, Horan Company, LLC(907) 747- x112___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to start Asterisk 1.2.5 with Asterisk-Addons 1.2.1
Hi all, I installed the Asterisk 1.2.5 and asterisk-addons 1.2.1 of a new Red Hat linux box( Linux version 2.4.20-8smp). I was able to compile both the software but when i start Asterisk, it exits with the following dump. Error Text Start= [res_crypto.so] = (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup'[res_config_mysql.so]Mar 6 05:18:23 WARNING[12779]: loader.c:325 __load_resource: /usr/lib/asterisk/modules/res_config_mysql.so: undefined symbol: __stack_chk_fail Mar 6 05:18:23 WARNING[12779]: loader.c:554 load_modules: Loading module res_config_mysql.so failed!End=== Can someone suggest a solution. Regards, Sharath Chandra ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help needed on setting up realtime
I installed Asterisk CVS-NHEAD-05/13/05-01:59:30 and placed few call in and through successfully. I was trying to set up the Realtime - picking the sip.conf and extensions.conf from mysql. I was going through some wiki pages, but what i don't understand is - which configuration change makes asterisk stop looking at extensions.conf and sip.conf for sip peers and pick the same from database. Please suggest. Thank you. Sharath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 2800 with Asterisk
Hi, Has anyone used Cisco 2800 Integrated services router to intiate SIP call to Asterisk. I would like to use it as gateway on to which T1 terminates and make Asterisk as my session target for few lines. Please let me know if there are any issues. Thanks, Sharath ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Softswitch
Hi, I am new to Asterisk. Can i use Asterisk as a session target from softswitch/Call Agent. I mean, is it possible to initiate a SIP call to Asterisk. My PRI terminates onto Cisco 2800 and i want to send few numbers to Asterisk to do some application related call control. Please advice ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
