Re: [asterisk-users] [asterisk-user] Confbridge Kick Action
reolved On Wed, Oct 22, 2014 at 10:28 AM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Here, I attached CLI log for above dialplan ... -- Executing [8484@conf-bridge:1] NoOp(SIP/8484-, Confbridge application) in new stack -- Executing [8484@conf-bridge:2] Answer(SIP/8484-, ) in new stack 0xb76309d8 -- Probation passed - setting RTP source address to 172.18.100.73:8000 -- Executing [8484@conf-bridge:3] Set(SIP/8484-, CONFBRIDGE(user,template)=default_user) in new stack -- Executing [8484@conf-bridge:4] ConfBridge(SIP/8484-, 1234786) in new stack -- SIP/8484- Playing 'conf-onlyperson.gsm' (language 'en') -- SIP/8484- Playing 'confbridge-join.gsm' (language 'en') -- CBAnn/1234786-;1 Playing 'confbridge-join.gsm' (language 'en') -- Channel CBAnn/1234786-;2 joined 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 -- Channel CBAnn/1234786-;2 left 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 -- Channel SIP/8484- joined 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 == Using SIP RTP CoS mark 5 -- Executing [8484@conf-bridge:1] NoOp(SIP/8484-0001, Confbridge application) in new stack -- Executing [8484@conf-bridge:2] Answer(SIP/8484-0001, ) in new stack 0xb76468f0 -- Probation passed - setting RTP source address to 172.18.100.73:8002 -- Executing [8484@conf-bridge:3] Set(SIP/8484-0001, CONFBRIDGE(user,template)=default_user) in new stack -- Executing [8484@conf-bridge:4] ConfBridge(SIP/8484-0001, 1234786) in new stack -- SIP/8484-0001 Playing 'confbridge-join.gsm' (language 'en') -- Channel CBAnn/1234786-;2 joined 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 -- CBAnn/1234786-;1 Playing 'confbridge-join.gsm' (language 'en') -- Channel CBAnn/1234786-;2 left 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 -- Channel SIP/8484-0001 joined 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 == Using SIP RTP CoS mark 5 -- Executing [8484@conf-bridge:1] NoOp(SIP/8484-0002, Confbridge application) in new stack -- Executing [8484@conf-bridge:2] Answer(SIP/8484-0002, ) in new stack 0xb765c4c0 -- Probation passed - setting RTP source address to 172.18.100.73:8004 -- Executing [8484@conf-bridge:3] Set(SIP/8484-0002, CONFBRIDGE(user,template)=default_user) in new stack -- Executing [8484@conf-bridge:4] ConfBridge(SIP/8484-0002, 1234786) in new stack -- SIP/8484-0002 Playing 'confbridge-join.gsm' (language 'en') -- CBAnn/1234786-;1 Playing 'confbridge-join.gsm' (language 'en') -- Channel CBAnn/1234786-;2 joined 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 -- Channel CBAnn/1234786-;2 left 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 -- Channel SIP/8484-0002 joined 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 chandrakant*CLI confbridge list 1234786 ChannelFlags User Profile Bridge Profile Menu CallerID == == SIP/8484- default_user default_bridge8484 SIP/8484-0001 default_user default_bridge8484 SIP/8484-0002 default_user default_bridge8484 == Client from 127.0.0.1, failed to authenticate in 30 seconds == Connect attempt from '127.0.0.1' unable to authenticate == Manager 'sabsebolo' logged on from 127.0.0.1 -- Channel SIP/8484-0001 left 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 -- Channel CBAnn/1234786-;2 joined 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 -- CBAnn/1234786-;1 Playing 'confbridge-leave.gsm' (language 'en') -- Channel CBAnn/1234786-;2 left 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 -- SIP/8484-0001 Playing 'conf-kicked.gsm' (language 'en') -- Executing [8484@conf-bridge:5] Set(SIP/8484-0001, CONFBRIDGE(user,marked)=yes) in new stack -- Executing [8484@conf-bridge:6] ConfBridge(SIP/8484-0001, 1234786) in new stack -- SIP/8484-0001 Playing 'confbridge-join.gsm' (language 'en') -- CBAnn/1234786-;1 Playing 'confbridge-join.gsm' (language 'en') -- Channel CBAnn/1234786-;2 joined 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 -- Channel CBAnn/1234786-;2 left 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 -- Channel SIP/8484-0001 joined 'softmix' base-bridge ed689de5-4d1b-4c40-8f59-ce2378d65542 chandrakant*CLI confbridge list
[asterisk-users] [asterisk-user] Confbridge Kick Action
Hi All, I am working on Asterisk 12.6.0 with ConfBridge module, when there are multiple user like admin and normal participant running with conference. When I try to kicked 2 user (Normal User), it play file conf-kicked and again join conference My scenario in confbridge like. 1] Admin User (e.g. SIP/8484-) 2] Normal User (e.g. SIP/8484-0001) 3] Admin User (e.g. SIP/8484-0002) When I try to execute confbridge kick using below AMI. Action: ConfbridgeKick Conference: 1701414 Channel: SIP/8484-0001 User kicked successfully and joined same conference again. Here is some asterisk CLI. *CLI confbridge list 1701414 ChannelFlags User Profile Bridge Profile Menu CallerID == == SIP/8484- Am conf-adminmenu 8484 SIP/8484-0001 m conf-menu 8484 SIP/8484-0002 Am conf-adminmenu 8484 -- Channel SIP/8484-0001 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel CBAnn/1701414-;2 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- CBAnn/1701414-;1 Playing 'confbridge-leave.gsm' (language '') -- Channel CBAnn/1701414-;2 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- SIP/8484-0001 Playing 'conf-kicked.gsm' (language 'en') -- Executing [s@conference-room:27] ConfBridge(SIP/8484-0001, 1701414,,user,conf-menu) in new stack -- SIP/8484-0001 Playing 'confbridge-join.gsm' (language 'en') -- CBAnn/1701414-;1 Playing 'confbridge-join.gsm' (language '') -- Channel CBAnn/1701414-;2 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel CBAnn/1701414-;2 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel SIP/8484-0001 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Chandrakant Solanki -- _ -- 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] [asterisk-user] Confbridge Kick Action
exten = 8484,1,noop(Confbridge application) same = n,Answer() same = n,Set(CONFBRIDGE(user,template)=default_user) same = n,Set(CONFBRIDGE(user,admin)=yes) same = n,ConfBridge(1701414) I am toggling user,admin option enable/disable. On Tue, Oct 21, 2014 at 1:56 PM, Shishir Pokharel shishir.pokha...@on24.com wrote: Can you share us your extensions.conf or the dialplan logic for this call? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Chandrakant Solanki *Sent:* Monday, October 20, 2014 11:19 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] [asterisk-user] Confbridge Kick Action Hi All, I am working on Asterisk 12.6.0 with ConfBridge module, when there are multiple user like admin and normal participant running with conference. When I try to kicked 2 user (Normal User), it play file conf-kicked and again join conference My scenario in confbridge like. 1] Admin User (e.g. SIP/8484-) 2] Normal User (e.g. SIP/8484-0001) 3] Admin User (e.g. SIP/8484-0002) When I try to execute confbridge kick using below AMI. Action: ConfbridgeKick Conference: 1701414 Channel: SIP/8484-0001 User kicked successfully and joined same conference again. Here is some asterisk CLI. *CLI confbridge list 1701414 ChannelFlags User Profile Bridge Profile Menu CallerID == == SIP/8484- Am conf-adminmenu 8484 SIP/8484-0001 m conf-menu 8484 SIP/8484-0002 Am conf-adminmenu 8484 -- Channel SIP/8484-0001 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel CBAnn/1701414-;2 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- CBAnn/1701414-;1 Playing 'confbridge-leave.gsm' (language '') -- Channel CBAnn/1701414-;2 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- SIP/8484-0001 Playing 'conf-kicked.gsm' (language 'en') -- Executing [s@conference-room:27] ConfBridge(SIP/8484-0001, 1701414,,user,conf-menu) in new stack -- SIP/8484-0001 Playing 'confbridge-join.gsm' (language 'en') -- CBAnn/1701414-;1 Playing 'confbridge-join.gsm' (language '') -- Channel CBAnn/1701414-;2 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel CBAnn/1701414-;2 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel SIP/8484-0001 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Chandrakant Solanki -- _ -- 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] [asterisk-user] Confbridge Kick Action
Hi, I have also added hangup priority as well but same result. [conf-bridge] exten = 8484,1,noop(Confbridge application) same = n,Answer() same = n,Set(CONFBRIDGE(user,template)=default_user) same = n,ConfBridge(1234786) exten = h,1,Hangup() -- Chandrakant Solanki On Tue, Oct 21, 2014 at 9:55 PM, Shishir Pokharel shishir.pokha...@on24.com wrote: After you kicked user from the conference it will continue to its dial plan. From your logs it indicates the call went to context conference-room s extension. Check your dialplan. Or hangup the call after confbridge application. Sent from my iPhone On Oct 21, 2014, at 2:36, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: exten = 8484,1,noop(Confbridge application) same = n,Answer() same = n,Set(CONFBRIDGE(user,template)=default_user) same = n,Set(CONFBRIDGE(user,admin)=yes) same = n,ConfBridge(1701414) I am toggling user,admin option enable/disable. On Tue, Oct 21, 2014 at 1:56 PM, Shishir Pokharel shishir.pokha...@on24.com wrote: Can you share us your extensions.conf or the dialplan logic for this call? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Chandrakant Solanki *Sent:* Monday, October 20, 2014 11:19 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] [asterisk-user] Confbridge Kick Action Hi All, I am working on Asterisk 12.6.0 with ConfBridge module, when there are multiple user like admin and normal participant running with conference. When I try to kicked 2 user (Normal User), it play file conf-kicked and again join conference My scenario in confbridge like. 1] Admin User (e.g. SIP/8484-) 2] Normal User (e.g. SIP/8484-0001) 3] Admin User (e.g. SIP/8484-0002) When I try to execute confbridge kick using below AMI. Action: ConfbridgeKick Conference: 1701414 Channel: SIP/8484-0001 User kicked successfully and joined same conference again. Here is some asterisk CLI. *CLI confbridge list 1701414 ChannelFlags User Profile Bridge Profile Menu CallerID == == SIP/8484- Am conf-adminmenu 8484 SIP/8484-0001 m conf-menu 8484 SIP/8484-0002 Am conf-adminmenu 8484 -- Channel SIP/8484-0001 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel CBAnn/1701414-;2 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- CBAnn/1701414-;1 Playing 'confbridge-leave.gsm' (language '') -- Channel CBAnn/1701414-;2 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- SIP/8484-0001 Playing 'conf-kicked.gsm' (language 'en') -- Executing [s@conference-room:27] ConfBridge(SIP/8484-0001, 1701414,,user,conf-menu) in new stack -- SIP/8484-0001 Playing 'confbridge-join.gsm' (language 'en') -- CBAnn/1701414-;1 Playing 'confbridge-join.gsm' (language '') -- Channel CBAnn/1701414-;2 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel CBAnn/1701414-;2 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel SIP/8484-0001 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Chandrakant Solanki -- _ -- 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
Re: [asterisk-users] [asterisk-user] Confbridge Kick Action
== == SIP/8484- default_user default_bridge8484 SIP/8484-0002 default_user default_bridge8484 SIP/8484-0001 M default_user default_bridge8484 -- Chandrakant Solanki On Wed, Oct 22, 2014 at 10:24 AM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hi, I have also added hangup priority as well but same result. [conf-bridge] exten = 8484,1,noop(Confbridge application) same = n,Answer() same = n,Set(CONFBRIDGE(user,template)=default_user) same = n,ConfBridge(1234786) exten = h,1,Hangup() -- Chandrakant Solanki On Tue, Oct 21, 2014 at 9:55 PM, Shishir Pokharel shishir.pokha...@on24.com wrote: After you kicked user from the conference it will continue to its dial plan. From your logs it indicates the call went to context conference-room s extension. Check your dialplan. Or hangup the call after confbridge application. Sent from my iPhone On Oct 21, 2014, at 2:36, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: exten = 8484,1,noop(Confbridge application) same = n,Answer() same = n,Set(CONFBRIDGE(user,template)=default_user) same = n,Set(CONFBRIDGE(user,admin)=yes) same = n,ConfBridge(1701414) I am toggling user,admin option enable/disable. On Tue, Oct 21, 2014 at 1:56 PM, Shishir Pokharel shishir.pokha...@on24.com wrote: Can you share us your extensions.conf or the dialplan logic for this call? *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Chandrakant Solanki *Sent:* Monday, October 20, 2014 11:19 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] [asterisk-user] Confbridge Kick Action Hi All, I am working on Asterisk 12.6.0 with ConfBridge module, when there are multiple user like admin and normal participant running with conference. When I try to kicked 2 user (Normal User), it play file conf-kicked and again join conference My scenario in confbridge like. 1] Admin User (e.g. SIP/8484-) 2] Normal User (e.g. SIP/8484-0001) 3] Admin User (e.g. SIP/8484-0002) When I try to execute confbridge kick using below AMI. Action: ConfbridgeKick Conference: 1701414 Channel: SIP/8484-0001 User kicked successfully and joined same conference again. Here is some asterisk CLI. *CLI confbridge list 1701414 ChannelFlags User Profile Bridge Profile Menu CallerID == == SIP/8484- Am conf-adminmenu 8484 SIP/8484-0001 m conf-menu 8484 SIP/8484-0002 Am conf-adminmenu 8484 -- Channel SIP/8484-0001 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel CBAnn/1701414-;2 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- CBAnn/1701414-;1 Playing 'confbridge-leave.gsm' (language '') -- Channel CBAnn/1701414-;2 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- SIP/8484-0001 Playing 'conf-kicked.gsm' (language 'en') -- Executing [s@conference-room:27] ConfBridge(SIP/8484-0001, 1701414,,user,conf-menu) in new stack -- SIP/8484-0001 Playing 'confbridge-join.gsm' (language 'en') -- CBAnn/1701414-;1 Playing 'confbridge-join.gsm' (language '') -- Channel CBAnn/1701414-;2 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel CBAnn/1701414-;2 left 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Channel SIP/8484-0001 joined 'softmix' base-bridge 485fcffc-49ad-4e86-8d1b-4655631a232a -- Chandrakant Solanki -- _ -- 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
[asterisk-users] Asterisk Crash 1.8.13.0
Hi, I have tried to start asterisk 1.8.13.0 using asterisk -vgc and service asterisk start. Every time I found below kinds of error. Please help me out, if anyone have idea Reading symbols from /usr/lib/libpq.so.5...(no debugging symbols found)...done. Loaded symbols for /usr/lib/libpq.so.5 Reading symbols from /lib/libldap_r-2.4.so.2...(no debugging symbols found)...done. Loaded symbols for /lib/libldap_r-2.4.so.2 Core was generated by `asterisk -gc'. Program terminated with signal 11, Segmentation fault. #0 dbt_data2str (cb=0x80d99a0 db_gettree_cb, data=0xb5fa623c, filter=0xb5fa613c /dundi/cache, sync=0) at db.c:155 155data[dbt-size - 1] = '\0'; Missing separate debuginfos, use: debuginfo-install cyrus-sasl-lib-2.1.23-13.el6_3.1.i686 glibc-2.12-1.107.el6_4.5.i686 keyutils-libs-1.4-4.el6.i686 krb5-libs-1.10.3-10.el6_4.6.i686 libcom_err-1.41.12-14.el6_4.2.i686 libcurl-7.19.7-37.el6_4.i686 libgcc-4.4.7-4.el6.i686 libidn-1.18-2.el6.i686 libjpeg-turbo-1.2.1-1.el6.i686 libselinux-2.0.94-5.3.el6_4.1.i686 libssh2-1.4.2-1.el6.i686 libstdc++-4.4.7-4.el6.i686 libtool-ltdl-2.2.6-15.5.el6.i686 libxml2-2.7.6-14.el6.i686 mysql-connector-odbc-5.1.5r1144-7.el6.i686 mysql-libs-5.5.36-21.el6.art.i686 mysqlclient16-5.1.59-2.el6.art.i686 ncurses-libs-5.7-3.20090208.el6.i686 nspr-4.10.2-1.el6_5.i686 nss-3.15.3-6.el6_5.i686 nss-softokn-freebl-3.14.3-3.el6_4.i686 nss-util-3.15.3-1.el6_5.i686 openldap-2.4.23-32.el6_4.1.i686 openssl-1.0.1e-15.el6.i686 postgresql-libs-8.4.18-1.el6_4.i686 unixODBC-2.2.14-12.el6_3.i686 zlib-1.2.3-29.el6.i686 (gdb) bt #0 dbt_data2str (cb=0x80d99a0 db_gettree_cb, data=0xb5fa623c, filter=0xb5fa613c /dundi/cache, sync=0) at db.c:155 #1 dbt_data2str_full (cb=0x80d99a0 db_gettree_cb, data=0xb5fa623c, filter=0xb5fa613c /dundi/cache, sync=0) at db.c:163 #2 process_db_keys (cb=0x80d99a0 db_gettree_cb, data=0xb5fa623c, filter=0xb5fa613c /dundi/cache, sync=0) at db.c:196 #3 0x080dac04 in ast_db_gettree (family=0x7fd54e dundi/cache, keytree=0x0) at db.c:595 #4 0x007e85a2 in process_clearcache (ignore=0x0) at pbx_dundi.c:2204 #5 0x08196e6b in dummy_start (data=0x9a46f70) at utils.c:1004 #6 0x00781a49 in start_thread () from /lib/libpthread.so.0 #7 0x006bdaae in clone () from /lib/libc.so.6 (gdb) quit -- Chandrakant Solanki -- _ -- 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] 2 PRI Card - Interrupt Problem
Hello All, I have 2 Digium card configure on Single machine, which can't share interrupt across all CPUs and sometimes asterisk reach 100% CPU usage. Here is system details and /proc/interrupt o/p. OS: CentOS 6.4 Kernel: 2.6.32-431.11.2.el6.x86_64 Dahdi Version: DAHDI Version: 2.7.0.2 Echo Canceller: HWEC Asterisk Version: 1.8.13.0 Output: /proc/interrupts cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 ... 37:1132730 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp 39:1132831 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp ... Thanks. -- Chandrakant Solanki -- _ -- 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] 2 PRI Card - Interrupt Problem
Thanks for reply, I am interested to see patch, but I don't find any link for the same. -- Chandrakant Solanki On Wed, May 14, 2014 at 2:09 PM, Thorsten Göllner t...@ovm-group.com wrote: Look for irqbalancer for your distribution: http://www.tutorialspoint.com/unix_commands/irqbalance.htm Am 14.05.2014 09:00, schrieb Chandrakant Solanki: Hello All, I have 2 Digium card configure on Single machine, which can't share interrupt across all CPUs and sometimes asterisk reach 100% CPU usage. Here is system details and /proc/interrupt o/p. OS: CentOS 6.4 Kernel: 2.6.32-431.11.2.el6.x86_64 Dahdi Version: DAHDI Version: 2.7.0.2 Echo Canceller: HWEC Asterisk Version: 1.8.13.0 Output: /proc/interrupts cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 ... 37:1132730 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp 39:1132831 0 0 0 0 0 0 0 IR-IO-APIC-fasteoi wct4xxp ... Thanks. -- Chandrakant Solanki -- _ -- 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] Meetme Show Activity in Minus
Hello All, Asterisk: 1.8.13.0 Dahdi : 2.6.2 Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux OS : CentOS 6.4 When I show meetme room details using meetme list command it shows Minus in activity column. Any Idea. meetme list Conf Num PartiesMarked Activity Creation Locked 54682 0002 N/A00:01:31 Dynamic No 62649 0003 N/A00:04:14 Dynamic No *52633 0002 N/A-6:-56:-48 Dynamic No 89737 0001 N/A-6:-40:-42 Dynamic No 89932 0002 N/A-6:-39:-20 Dynamic No 65393 0002 N/A-6:-33:-17 Dynamic No * -- Chandrakant Solanki -- _ -- 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] Meetme Show Activity in Minus
Solved On Wed, Jan 22, 2014 at 12:44 PM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hello All, Asterisk: 1.8.13.0 Dahdi : 2.6.2 Kernel : 2.6.32-431.3.1.el6.i686 #1 SMP Fri Jan 3 18:53:30 UTC 2014 i686 i686 i386 GNU/Linux OS : CentOS 6.4 When I show meetme room details using meetme list command it shows Minus in activity column. Any Idea. meetme list Conf Num PartiesMarked Activity Creation Locked 54682 0002 N/A00:01:31 Dynamic No 62649 0003 N/A00:04:14 Dynamic No *52633 0002 N/A-6:-56:-48 Dynamic No 89737 0001 N/A-6:-40:-42 Dynamic No 89932 0002 N/A-6:-39:-20 Dynamic No 65393 0002 N/A-6:-33:-17 Dynamic No * -- Chandrakant Solanki -- _ -- 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] Asterisk 'n Dahdi on Sun Solaris
Hello All, I am trying to install Asterisk 1.8.13.0 dahdi-complete 2.5.1 libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. I am using SunOS solaris-server 5.11 11.1 i86pc i386 i86pc on Virtual Box. -- Regards, Chandrakant Solanki -- _ -- 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] Asterisk 'n Dahdi on Sun Solaris
Actually I am trying for meetme module. On Wed, Jun 12, 2013 at 3:09 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Wed, Jun 12, 2013 at 12:32:40PM +0530, Chandrakant Solanki wrote: Hello All, I am trying to install Asterisk 1.8.13.0 dahdi-complete 2.5.1 libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. Maybe. But dahdi-complete you're trying to install includes dahdi-linux which is drivers for Linux. What do you need DAHDI for? I am using SunOS solaris-server 5.11 11.1 i86pc i386 i86pc on Virtual Box. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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 -- Regards, Chandrakant Solanki -- _ -- 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] Asterisk 'n Dahdi on Sun Solaris
There is some changes, which I made. If anybody knows ... please share knowledge for compilation of dahdi-complete and asterisk 1.8.13.0 On Wed, Jun 12, 2013 at 3:44 PM, Johan Wilfer li...@jttech.se wrote: 2013-06-12 11:42, Chandrakant Solanki skrev: Actually I am trying for meetme module. On Wed, Jun 12, 2013 at 3:09 PM, Tzafrir Cohen tzafrir.co...@xorcom.com mailto:tzafrir.cohen@xorcom.**com tzafrir.co...@xorcom.com wrote: On Wed, Jun 12, 2013 at 12:32:40PM +0530, Chandrakant Solanki wrote: Hello All, I am trying to install Asterisk 1.8.13.0 dahdi-complete 2.5.1 libpri 1.4.13 version. Is it possible to install dahdi on Sun Solaris? I have searched so many, but don't found any help. If it's a new application you are building - Why not test asterisk 11 + confbridge? This way you won't need DAHDI. -- Johan Wilfer -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Chandrakant Solanki -- _ -- 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] ODBC Connection Problem
/etc/odbc.ini [telco-ops] Description = Asterisk realtime and other FUNC_ODBC access Driver = MySQL Server = 172.18.100.18 Socket = /var/lib/mysql/data3306/mysql.sock User= dba Password= c3podb@2012 Database= mytelcoexample Port= 3306 Option = 3 On Mon, Dec 10, 2012 at 4:34 PM, Thorsten Göllner t...@ovm-group.com wrote: Am 10.12.2012 06:37, schrieb Chandrakant Solanki: Hi All, OS : CentOS 5 64bit OS Machine Asterisk: 1.8.13.0 ODBC Packages: unixODBC-2.2.11-7.1 mysql-connector-odbc-3.51.12-2.2 unixODBC-devel-2.2.11-7.1 res_odbc.conf [telco-ops] enabled = yes dsn = telco-ops username = dba password = c3podb@2012 pre-connect = yes sanitysql = select 1 idlecheck = 15 ;isolation = repeatable_read pooling = yes limit = 3600 connect_timeout = 10 negative_connection_cache = 30 Above is my installation package and configuration file (res_odbc.conf), when I try to execute odbc show all it always gives below output. *CLI odbc show all ODBC DSN Settings - Name: telco-ops DSN:telco-ops Last connection attempt: 1970-01-01 00:00:00 Pooled: Yes Limit: 3600 Connections in use: 1 - Connection 1: connected When Insert/Update/Select query will be executed, it can't update last connection attempt field. In result, ODBC stuck after few minutes, and in this case I also need to restart asterisk, because I can't type any command, it can't give any command's output. Also updated asterisk with 10.9.0, but same result. Please show us /etc/odbc.ini too. -- Regards, Chandrakant Solanki -- _ -- 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] tcptls ssl connection error
Hello All, Anyone have idea regarding below error. After applying all patch, still faced the same issue. -- Regards, Chandrakant Solanki On Fri, Nov 9, 2012 at 11:39 AM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hello All, I am using asterisk 1.8.13.0 and which is running on TLS port and my request forwarded from opensips which is also run tls port. On both end my certificate is same. During search about this error, I found below blog and apply patch, then also found below error. https://issues.asterisk.org/jira/browse/ASTERISK-18345 https://issues.asterisk.org/jira/browse/ASTERISK-20559 Also applied r375023 [Nov 8 21:57:34] ERROR[16357]: tcptls.c:89 ssl_close: SSL_shutdown() failed: 5 [Nov 8 21:57:36] ERROR[16001]: tcptls.c:89 ssl_close: SSL_shutdown() failed: 5 [Nov 8 21:57:37] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:37] WARNING[19274]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! [Nov 8 21:57:39] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:39] WARNING[19356]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! [Nov 8 21:57:49] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:49] WARNING[19357]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! -- Regards, Chandrakant Solanki -- _ -- 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] tcptls ssl connection error
Hello All, I am using asterisk 1.8.13.0 and which is running on TLS port and my request forwarded from opensips which is also run tls port. On both end my certificate is same. During search about this error, I found below blog and apply patch, then also found below error. https://issues.asterisk.org/jira/browse/ASTERISK-18345 https://issues.asterisk.org/jira/browse/ASTERISK-20559 Also applied r375023 [Nov 8 21:57:34] ERROR[16357]: tcptls.c:89 ssl_close: SSL_shutdown() failed: 5 [Nov 8 21:57:36] ERROR[16001]: tcptls.c:89 ssl_close: SSL_shutdown() failed: 5 [Nov 8 21:57:37] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:37] WARNING[19274]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! [Nov 8 21:57:39] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:39] WARNING[19356]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! [Nov 8 21:57:49] == Problem setting up ssl connection: error::lib(0):func(0):reason(0) [Nov 8 21:57:49] WARNING[19357]: tcptls.c:251 handle_tcptls_connection: FILE * open failed! -- Regards, Chandrakant Solanki -- _ -- 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] [asterisk-user] INTERNAL_OBJ error in asterisk 1.8.13
Hi All, Asterisk Version: 1.8.13.0 CentOs : 6.3 Continues getting this error while submitting cdr record. [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 [Sep 11 22:36:13] ERROR[12307]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x1 for 0xb61ddeb8 -- Regards, Chandrakant Solanki -- _ -- 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] Interrupt error
Hello, Asterisk : asterisk-1.6.0.5 Dahdi: dahdi-linux-complete-2.5.1 Kernel Version: 2.6.18-128.el5xen AS_1 kernel: Uhhuh. NMI received for unknown reason 00 on CPU 0. Message from syslogd@ at Tue Sep 4 11:46:57 2012 ... AS_1 kernel: Do you have a strange power saving mode enabled? Message from syslogd@ at Tue Sep 4 11:46:57 2012 ... AS_1 kernel: Dazed and confused, but trying to continue Message from syslogd@ at Tue Sep 4 11:49:39 2012 ... AS_1 kernel: Uhhuh. NMI received for unknown reason 00 on CPU 0. Message from syslogd@ at Tue Sep 4 11:49:39 2012 ... AS_1 kernel: Do you have a strange power saving mode enabled? Message from syslogd@ at Tue Sep 4 11:49:39 2012 ... AS_1 kernel: Dazed and confused, but trying to continue Message from syslogd@ at Tue Sep 4 11:52:17 2012 ... AS_1 kernel: Uhhuh. NMI received for unknown reason 00 on CPU 0. Message from syslogd@ at Tue Sep 4 11:52:17 2012 ... AS_1 kernel: Do you have a strange power saving mode enabled? Message from syslogd@ at Tue Sep 4 11:52:17 2012 ... AS_1 kernel: Dazed and confused, but trying to continue Message from syslogd@ at Tue Sep 4 11:52:27 2012 ... AS_1 kernel: Uhhuh. NMI received for unknown reason 00 on CPU 0. Message from syslogd@ at Tue Sep 4 11:52:27 2012 ... AS_1 kernel: Do you have a strange power saving mode enabled? Message from syslogd@ at Tue Sep 4 11:52:27 2012 ... AS_1 kernel: Dazed and confused, but trying to continue -- Regards, Chandrakant Solanki -- _ -- 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] AMR - Segmentation Fault
On Mon, Jul 9, 2012 at 7:32 AM, Patrick Lists asterisk-l...@puzzled.xs4all.nl wrote: On 04-07-12 06:45, Chandrakant Solanki wrote: So, is http://sourceforge.net/**projects/aterisk-amr/files/http://sourceforge.net/projects/aterisk-amr/files/same patch also works in 1.8.13.0?? I don't know about 1.8.13 but it did work with 1.8.11. Just manually apply the patch if it does not apply automagically. Regards, Patrick -- __**__**_ -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users Hi @Patrick, are you using which AMR source, will you please provide me link, I also tried with 1.8.11 but didn't found success. I am using sourceforge one http://sourceforge.net/projects/aterisk-amr/files/ -- Regards, Chandrakant Solanki -- _ -- 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] call file and NFS server
Hello,
I have 3 server, 2 running with asterisk and another one generate call
files say some directory callfile/serverA and callfile/serverB (NFS
Sharing) and mounted this directory to respectively on Server A (Asterisk)
and Server B(Asterisk) on /var/spool/asterisk/outgoing.
Server A has Asterisk 1.8.0-rc2 and Server B has asterisk version 1.8.9.0,
and both asterisk compile ./configure --without-inotify
Callfile will execute call successfully on both machine, but got the
following problem
*[Jul 6 16:15:04] WARNING[26921]: pbx_spool.c:278 safe_append: Unable to
set utime on /var/spool/asterisk/outgoing/15.call: Operation not
permitted
*
I have set the folder (callfile/Server{A/B}) permission to 777 as well as
call file permission to 777.
--
Regards,
Chandrakant Solanki
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Re: [asterisk-users] call file and NFS server
Hi,
I have 100+ call file generated in other directory, and by using program, I
have moved 10-10 files in /var/spool/asterisk/outgoing, and call made
successfully.
Once all call completed, I found following error for all files...
[Jul 7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to
open /var/spool/asterisk/outgoing/100097_172.18.100.72.call: No such
file or directory, deleting
[Jul 7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to
open /var/spool/asterisk/outgoing/100098_172.18.100.72.call: No such
file or directory, deleting
[Jul 7 10:54:03] WARNING[7884]: pbx_spool.c:407 scan_service: Unable to
open /var/spool/asterisk/outgoing/100099_172.18.100.72.call: No such
file or directory, deleting
On Fri, Jul 6, 2012 at 8:47 PM, Steve Edwards asterisk@sedwards.comwrote:
On Friday 06 July 2012, Chandrakant Solanki wrote:
I have set the folder (callfile/Server{A/B}) permission to 777 as well
as call file permission to 777.
On Fri, 6 Jul 2012, A J Stiles wrote:
(By the way, you should have permissions 666 for a callfile, not 777.
Callfiles should not be executable.)
Whenever I see 777 (or it's Satanic cousin, 666) I see 'I don't really
understand ownership and permissions so let's just allow everything and
hope for the best.'
Do you really intend to allow every user and exploited program to be able
to create call files? (And if you've done this, you've probably created
other holes in your system's security.)
While 'opening the flood gates' is (IMO) a valid temporary debugging
technique to identify the source of the problem, the directories and files
should be owned by the user executing Asterisk and permissions should limit
reading to only users and groups that need reading and limit writing
to only users and groups that need writing.
I don't have any need or experience with call files on my production
boxes, but I suspect a successful implementation would include NTP and
creating the call file in another directory on the shared device and then
moving the call file to the outgoing spool directory.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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--
Regards,
Chandrakant Solanki
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[asterisk-users] AMR - Segmentation Fault
Hi All, OS : Cent OS 5 64Bit Asterisk : 1.8.0-rc2 AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/ When I tried to call or start asterisk, I found Segmentation Fault. Below I paste same for AMR Loaded symbols for /usr/lib/asterisk/modules/app_db.so Core was generated by `asterisk -qg'. Program terminated with signal 11, Segmentation fault. #0 D_plsf_3 (st=value optimized out, mode=value optimized out, bfi=value optimized out, indice=value optimized out, lsp1_q=0x7fff11d05df0) at sp_dec.c:567 567 tmp = ( ( cos_table[ind+1]-cos_table[ind] )*offset ) 1; (gdb) br Breakpoint 1 at 0x2aaab57093f1: file sp_dec.c, line 567. (gdb) bt #0 D_plsf_3 (st=value optimized out, mode=value optimized out, bfi=value optimized out, indice=value optimized out, lsp1_q=0x7fff11d05df0) at sp_dec.c:567 #1 0x2aaab570df95 in Decoder_amr (st=0x2aaad6147d00, mode=MR515, parm=0x7fff11d06a40, frame_type=value optimized out, synth=0x7fff11d060a0, A_t=0x7fff11d06730) at sp_dec.c:4717 #2 0x2aaab5712e6a in Speech_Decode_Frame (st=0x2aaad613e200, mode=80, parm=0x2aaab5725400, frame_type=4294949091, synth=0x2aaad6142ba0) at sp_dec.c:5676 #3 0x2aaab56efb25 in Decoder_Interface_Decode (st=0x2aaad613e1e0, bits=value optimized out, synth=0x2aaad6142ba0, bfi=value optimized out) at interf_dec.c:816 #4 0x2aaab56ee6f9 in amrtolin_framein (pvt=0x2aaad613e5c0, f=value optimized out) at codec_amr.c:263 #5 0x00528244 in framein (pvt=0x2aaad613e5c0, f=0x2aaab5942e40) at translate.c:178 #6 0x00529538 in calc_cost (t=0x2aaab593ff40, seconds=1) at translate.c:397 #7 0x0052a00c in __ast_register_translator (t=0x2aaab593ff40, mod=value optimized out) at translate.c:835 #8 0x2aaab56ee37b in load_module () at codec_amr.c:490 #9 0x004c29e3 in start_resource (mod=0xdf) at loader.c:785 #10 0x004c3308 in load_resource_list (load_order=0x7fff11d07000, global_symbols=0, mod_count=0x7fff11d0701c) at loader.c:973 #11 0x004c3727 in load_modules (preload_only=0) at loader.c:1126 #12 0x0043c2c4 in main (argc=value optimized out, argv=0x7fff11d095e8) at asterisk.c:3794 (gdb) quit -- Regards, Chandrakant Solanki -- _ -- 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] AMR - Segmentation Fault
So, is http://sourceforge.net/projects/aterisk-amr/files/ same patch also works in 1.8.13.0?? On Wed, Jul 4, 2012 at 3:18 AM, Hans Witvliet aster...@a-domani.nl wrote: On Tue, 2012-07-03 at 17:13 +0530, Chandrakant Solanki wrote: Hi All, OS : Cent OS 5 64Bit Asterisk : 1.8.0-rc2 AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/ When I tried to call or start asterisk, I found Segmentation Fault. Without trying to be pedantic, but 1.8.0-rc2 Ever considered upgrading? To 1.8.13.0 or so.. hans -- _ -- 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 -- Regards, Chandrakant Solanki -- _ -- 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] VP8 Codec integration in Asterisk
Hi All, Anybody have idea that how to add VP8 codec into Asterisk 1.8 and from where to download. Please share if anybody has idea or related document. -- Regards, Chandrakant Solanki -- _ -- 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] CLI command 'database deltree' doesn't remove family with space in its name
Hi Satish Try to do something like this way CLI database deltree 18-05-2011 00:00:0052011175221575 TESTDATE I have done like this way hope it works for you. -- Regards, Chandrakant Solanki On Mon, May 30, 2011 at 2:53 PM, Satish Barot satish4aster...@gmail.comwrote: While playing with DB function in Dialplan, I have added some garbage in AstDB. These are some family names with space in them. See this, demo*CLI database show /18-05-2011 00:00:0052011175221575/TESTDATE: 2011-05-14 21:33:46 /18-05-2011 00:00:0052011175221575/TEST1 : 410 /18-05-2011 00:00:0052011175221575/TEST2 : 155 /18-05-2011 00:00:0052011182614252/TEST3 : 157 I treid to remove it from CLI through database deltree and database del commands, but no hope. demo*CLI database deltree 18-05-2011 00:00:0052011175221575 0 database entries removed. demo*CLI database deltree 18-05-2011 00:00:0052011175221575 TESTDATE 0 database entries removed. demo*CLI database del 18-05-2011 00:00:0052011175221575 TESTDATE 0 database entries removed. Some more variations... demo*CLI database deltree '18-05-2011 00:00:0052011175221575' 0 database entries removed. demo*CLI database deltree `18-05-2011 00:00:0052011175221575` 0 database entries removed. Any suggestions to remove them? Thanking you, [SATISH] -- _ -- 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] environment variable + res_mysql.conf
Hi All. I have export some db parameter in /etc/bashrc as follows ... export DB_NAME=xyz export DB_IP=1x.1x.1x.1x export DB_PWD=dkjfaoi Now, I want use these all environment variable into /etc/asterisk/res_mysql.conf file. Is there any way to do this..?? -- Regards, Chandrakant Solanki -- _ -- 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] [Asterisk-Users] How do I install speex for asterisk?
On Fri, Aug 6, 2010 at 5:29 PM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi, I have followed steps which were mentioned on forum and given below. Still couldn’t get speex working. On test calls getting error “chan_sip.c: sip_call: No audio format found to offer.” # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not showing speex translation on “core show translation recalc 10”. Can anybody please tell if missing some step in this. --- Kind Regards, *Deepika Nijhawan* *VoIP Engineer* * * Hi Go For asterisk top directory. And follow below steps to check whether speex function module is enable or not. ./configure make menuselect = Go for Dialplan Function = Then func_speex. if func_speex shows [XXX] this symbol that means func_speex module is not enable. And if you select func_speex then it shows dependency below of module list. -- Regards, Chandrakant Solanki -- _ -- 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] [Asterisk-Users] How do I install speex for asterisk?
Hi Can you tell me which Linux OS are you used and what is speex / speex-devel version. Can you give details for above? -- Regards, Chandrakant Solanki On Fri, Aug 6, 2010 at 6:22 PM, Nasir Iqbal na...@ictinnovations.comwrote: Hi, May you also need to install *speex-tools* . if problem retain then let us know about your Linux distribution and Asterisk version. Regards On Fri, Aug 6, 2010 at 4:59 PM, Deepika Nijhawan deepika.nijha...@oxygen8.com wrote: Hi, I have followed steps which were mentioned on forum and given below. Still couldn’t get speex working. On test calls getting error “chan_sip.c: sip_call: No audio format found to offer.” # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not showing speex translation on “core show translation recalc 10”. Can anybody please tell if missing some step in this. --- Kind Regards, *Deepika Nijhawan* *VoIP Engineer* * * *Oxygen8* Communications -- _ -- 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 -- Nasir Iqbal ICT Innovations http://www.ictinnovations.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
[asterisk-users] dialog module count
Hello
I need *count* for number of active calls on kamailio server. I have done
following configuration in my kamailio.cfg file
...
loadmodule dialog.so
modparam(dialog,profiles_with_value,caller)
modparam(dialog, dlg_flag, 4)
route[0] {
...
if(is_method(INVITE))
{
get_profile_size(caller, $fu, $var(SIZE));
xlog(L_INFO, == $var(SIZE) \n);
if( $var(SIZE) 1 ){
set_dlg_profile(caller,$fu);
xlog(L_INFO, M IN server1 ONLY \n);
use_media_proxy();
record_route();
rewritehostport(server.pbx.com:5060);
route(8);
}
else {
set_dlg_profile(caller,$fu);
xlog(L_INFO, M GOING TO SERVER2);
sl_send_reply(100, Trying);
ds_select_domain(1, 4);
forward();
}
xlog(L_INFO, +++ == $var(SIZE) \n);
exit;
}
...
}
Is anything missing in above configuration or something goes wrong.?
--
Regards,
Chandrakant Solanki
--
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Re: [asterisk-users] Can't compile DAHDI - wrong kernel source
Hello What will be your exact kernel version. Give me output uname -a command. -- Regards, Chandrakant Solanki On Thu, Jul 15, 2010 at 6:58 AM, Thermal Wetland thermalwetl...@gmail.comwrote: On Wed, Jul 14, 2010 at 4:55 AM, bruce bruce bruceb...@gmail.com wrote: I am stuck with the same problem but I have used asterisk yum repository and it worked by itself without me worrying for kernel stuff. However, I need to install speex codec and now I am stuck as it doesn't get picked up by the yum asterisk install somehow. I have lib speex and speex already installed and when doing yum install asterisk16 I don't see speex in core show translation Is there anything specific I have to do? Do I have to build from source as well? -Sorry, didn't mean to hijack the thread. Thanks, Bruce On Wed, Jul 14, 2010 at 5:08 AM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hi If you install rpm from any location it goes to its default location. You just go for above steps. For kernel you can go for http://kernel.org -- Regards, Chandrakant Solanki On Wed, Jul 14, 2010 at 2:06 PM, liuxin nyliuxin...@gmail.com wrote: Hi. The best easy way is: copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm 2010/7/14 Gareth Blades list-aster...@skycomuk.com Thermal Wetland wrote: I have a virtual server with godaddy but can not compile DAHDI as it complains that I do not have the correct kernel source. The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686: Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest version Nothing to do uname -a returns: Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net http://ip-XXX-XXX-XXX-XXX.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux When I try to compile DAHDI it fails with: make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-028stab064.7 kernel installed. Is there a way to trick DAHDI to use the installed kernel? Thanks for the help! -- -Thermal What kernel versions do you have installed? If you are currently running an older kernel but installed a newer kernel and sources but havent rebooted to activate the new one yet then it may still be trying to locate the source for the older running kernel. I was able to download the rpm's and install them: [r...@ip-97-74-119-59 src]# rpm -ivh ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm warning: ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm: Header V3 DSA signature: NOKEY, key ID a7a1d4b6 Preparing...### [100%] package ovzkernel-2.6.18-128.2.1.el5.028stab064.7.i686 is already installed [r...@ip-97-74-119-59 src]# rpm -ivh ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm warning: ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686.rpm: Header V3 DSA signature: NOKEY, key ID a7a1d4b6 Preparing...### [100%] package ovzkernel-devel-2.6.18-128.2.1.el5.028stab064.7.i686 is already installed [r...@ip-97-74-119-59 src]# cd - /usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0 [r...@ip-97-74-119-59 dahdi-linux-complete-2.3.0.1+2.3.0]# make all make -C linux all make[1]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make -C drivers/dahdi/firmware firmware-loaders make[2]: Entering directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-028stab064.7 kernel installed. make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux' make: *** [all] Error 2 The directories in /usr/src/kernels is: [r...@ip-97-74-119-59 kernels]# ls -l total 51328 drwxr-xr-x 20 root root 4096 Jul 14 18:04 2.6.18-128.2.1.el5.028stab064.7-i686 drwxr-xr-x 19 root root 4096 Jul 13 20:25 2.6.18-164.11.1.el5-i686 drwxrwxr-x 19 root root 4096 Feb 23 2007 linux-2.6.18.8 I tried to install the kernel from source but couldn't find the exact kernel, I installed linux-2.6.18.8 as I was the closest. Both of the directories in /usr/src/kernels/ have the -i686 suffix, is that the issue? -- -Thermal -- _ -- 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
Re: [asterisk-users] Can't compile DAHDI - wrong kernel source
Hi Following steps to do... 1] # cd /usr/src/kernels/ 2] # ln -s 2.6.18-128.2.1.el5.028stab064.7-i686 2.6.18-028stab064.7 Try this 'n let me know... Hope this will work fine... -- Regards, Chandrakant Solanki On Thu, Jul 15, 2010 at 12:00 PM, Thermal Wetland thermalwetl...@gmail.comwrote: On Wed, Jul 14, 2010 at 8:09 PM, Chandrakant Solanki solanki.chandrak...@gmail.com wrote: Hello What will be your exact kernel version. Give me output uname -a command. -- Regards, Chandrakant Solanki Thank you for the help! Here is the output: [r...@ip-97-74-119-59 ~]# uname -a Linux ip-97-74-119-59.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux -Thermal -- _ -- 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] Can't compile DAHDI - wrong kernel source
Hi Check your kernel version using *uname -r *and then try to download tar.gz setup for that version. And extract it into /usr/src/kernels directory , then try to compile. -- Regards, Chandrakant Solanki On Wed, Jul 14, 2010 at 1:46 PM, Gareth Blades list-aster...@skycomuk.comwrote: Thermal Wetland wrote: I have a virtual server with godaddy but can not compile DAHDI as it complains that I do not have the correct kernel source. The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686: Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest version Nothing to do uname -a returns: Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net http://ip-XXX-XXX-XXX-XXX.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux When I try to compile DAHDI it fails with: make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-028stab064.7 kernel installed. Is there a way to trick DAHDI to use the installed kernel? Thanks for the help! -- -Thermal What kernel versions do you have installed? If you are currently running an older kernel but installed a newer kernel and sources but havent rebooted to activate the new one yet then it may still be trying to locate the source for the older running kernel. -- _ -- 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] Can't compile DAHDI - wrong kernel source
Hi If you install rpm from any location it goes to its default location. You just go for above steps. For kernel you can go for http://kernel.org -- Regards, Chandrakant Solanki On Wed, Jul 14, 2010 at 2:06 PM, liuxin nyliuxin...@gmail.com wrote: Hi. The best easy way is: copy kernel-devel-2.6.18-028stab064.7.rpm to /usr/src then run rpm -ivh kernel-devel-2.6.18-028stab064.7.rpm 2010/7/14 Gareth Blades list-aster...@skycomuk.com Thermal Wetland wrote: I have a virtual server with godaddy but can not compile DAHDI as it complains that I do not have the correct kernel source. The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686: Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest version Nothing to do uname -a returns: Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.nethttp://ip-xxx-xxx-xxx-xxx.ip.secureserver.net/ http://ip-XXX-XXX-XXX-XXX.ip.secureserver.nethttp://ip-xxx-xxx-xxx-xxx.ip.secureserver.net/ 2.6.18-028stab064.7 #1 SMP Wed Aug 26 13:11:07 MSD 2009 i686 i686 i386 GNU/Linux When I try to compile DAHDI it fails with: make[2]: Leaving directory `/usr/src/asterisk/dahdi-linux-complete-2.3.0.1+2.3.0/linux/drivers/dahdi/firmware' You do not appear to have the sources for the 2.6.18-028stab064.7 kernel installed. Is there a way to trick DAHDI to use the installed kernel? Thanks for the help! -- -Thermal What kernel versions do you have installed? If you are currently running an older kernel but installed a newer kernel and sources but havent rebooted to activate the new one yet then it may still be trying to locate the source for the older running kernel. -- _ -- 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] Asterisk Crashes - Segmentation Fault
Hi Install mysql 'n mysql-devel which includes /usr/lib/mysql/libmysqlclient.so.15 library. And also insert /usr/lib/mysql into /etc/ld.so.conf and then execute ldconfig command on terminal. -- Regards, Chandrakant Solanki On Fri, Jul 9, 2010 at 2:32 PM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Hi, My Web-MeetMe_v4.0.1, i followed the instructions in the README File in the same package. Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? i didnt get this actually what do i need to check here? Please dont mind but m not so good in opensource world. I try to read and understand and on trial n error basis try to implement things. Though had very much interest in learning things. The GDB output is huge on, Following are my GDB errors. [r...@linuxtest tmp]# gdb asterisk core.LinuxTest-2010-07-07T21:13:15+0400 | more GNU gdb (GDB) Red Hat Enterprise Linux (7.0.1-23.el5_5.1) Copyright (C) 2009 Free Software Foundation, Inc. License GPLv3+: GNU GPL version 3 or later http://gnu.org/licenses/gpl.html This is free software: you are free to change and redistribute it. There is NO WARRANTY, to the extent permitted by law. Type show copying and show warranty for details. This GDB was configured as i386-redhat-linux-gnu. For bug reporting instructions, please see: http://www.gnu.org/software/gdb/bugs/... Reading symbols from /usr/sbin/asterisk...done. warning: .dynamic section for /usr/lib/libidn.so.11 is not at the expected address warning: difference appears to be caused by prelink, adjusting expectations [New Thread 3212] SOME OF THE LINES IN the end of GDB Error: Reading symbols from /usr/lib/asterisk/modules/cdr_manager.so...done. Loaded symbols for /usr/lib/asterisk/modules/cdr_manager.so Reading symbols from /usr/lib/asterisk/modules/res_config_mysql.so...(no debugging symbols found)...done. Loaded symbols for /usr/lib/asterisk/modules/res_config_mysql.so Reading symbols from /usr/lib/asterisk/modules/chan_phone.so...done. Loaded symbols for /usr/lib/asterisk/modules/chan_phone.so Core was generated by `/usr/sbin/asterisk -f -vvvg -c'. Program terminated with signal 11, Segmentation fault. #0 0x01027d9d in mysql_fetch_row () from /usr/lib/mysql/libmysqlclient.so.15 --Manmohan Singh. On Thu, Jul 8, 2010 at 11:21 PM, Dan Austin dan_aus...@phoenix.comwrote: Manmohan wrote: I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i provide the conference number, asterisk crashes giving segmentation fault. I have been trying to google up and checked lot of forums but didnt get any solution for this yet. Which instructions did you follow for the integration? Are you using RealTime enabled app_meetme or app_cbmysql from the WMM package? Which exact version of WMM? Dan -- _ -- 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 Regards Manmohan Singh Jandu -- _ -- 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] Asterisk Crashes - Segmentation Fault
On Thu, Jul 8, 2010 at 12:21 PM, Manmohan Singh Jandu manmoha...@gmail.comwrote: Hello Team, I was looking for audio conferencing solution where i got Web-meetme. I had installed Asterisk 1.6.2.9 on Centos 5.4. Its perfecting working fine. I tried using Meetme even meetme app is working perfectly fine. I installed Webmeetme 4.0 and integrated with my asterisk. When i try to dial the conference number it take me to an IVR wherein it asks for the conference number. The time i provide the conference number, asterisk crashes giving segmentation fault. I have been trying to google up and checked lot of forums but didnt get any solution for this yet. Kernel version -- 2.6.18-194.3.1.el5PAE -- Thanks Regards Manmohan Singh Jandu -- _ -- 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 Hi If you get Segmentation fault. One of core.$ file is created. Try to use # gdb asterisk core.$ and use bt command. And then paste error here. -- Regards, Chandrakant Solanki -- _ -- 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] [asterisk-user] gsmtolin_framein: Invalid GSM data
Hi I have created meetme with 3 user. When i going to mute user it gives following error.. *Asterisk Version : 1.6.2.6* -- SIP/52987-0040 Playing 'conf-muted.gsm' (language 'en') [Jul 2 22:46:51] WARNING[10823]: codec_gsm.c:103 gsmtolin_framein: Invalid GSM data (1) [Jul 2 22:46:51] WARNING[10823]: translate.c:204 framein: gsmtolin did not update samples 0 [Jul 2 22:46:51] WARNING[10823]: codec_gsm.c:103 gsmtolin_framein: Invalid GSM data (1) [Jul 2 22:46:51] WARNING[10823]: translate.c:204 framein: gsmtolin did not update samples 0 [Jul 2 22:46:51] WARNING[10823]: codec_gsm.c:103 gsmtolin_framein: Invalid GSM data (1) [Jul 2 22:46:51] WARNING[10823]: translate.c:204 framein: gsmtolin did not update samples 0 [Jul 2 22:46:51] WARNING[10823]: codec_gsm.c:103 gsmtolin_framein: Invalid GSM data (1) [Jul 2 22:46:51] WARNING[10823]: translate.c:204 framein: gsmtolin did not update samples 0 Any Idea..? -- Regards, Chandrakant Solanki -- _ -- 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] MeetMe Options with S(10)L(100)
Hi I have set MeetMe options like *sdMS(10)L(1000)* in dialplan. But when i print this value in c file using ast_log.. I am getting only *sdMS(10 *this options. Is there any special way to set option in dialplan with *sdMS(10)L(1000) *in dialplan -- Regards, Chandrakant Solanki -- _ -- 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] SIP / Echo Cancellation
Hello I have successfully compiled OSLEC for echo cancellation for DAHDI channel. Is there any way to do echo cancellation for SIP Channel. Is any, please suggest me.?? Thanks in advance.. -- Regards, Chandrakant Solanki -- _ -- 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] [asterisk-user] SIP / Echo Cancellation
Hello I have successfully compiled OSLEC for echo cancellation for DAHDI channel. Is there any way to do echo cancellation for SIP Channel. Is any, please suggest me.?? Thanks in advance.. -- Regards, Chandrakant Solanki -- _ -- 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] dahdi and oslec
Hi All, I have followed below steps to enable echo cancellation. # cd /usr/src # wget http://kernel.org/pub/linux/kernel/v2.6/linux-2.6.28.tar.bz2 # tar xjf linux-2.6.28.tar.bz2 # tar zxvf dahdi-linux-2.1.0.4.tar.gz # ln -s /usr/src/dahdi-linux-2.1.0.4 /usr/src/dahdi # mkdir /usr/src/dahdi/drivers/staging # cp -fR /usr/src/linux-2.6.28/drivers/staging/echo /usr/src/dahdi/drivers/staging # sed -i s|#obj-m += dahdi_echocan_oslec.o|obj-m += dahdi_echocan_oslec.o| /usr/src/dahdi/drivers/dahdi/Kbuild # sed -i s|#obj-m += ../staging/echo/|obj-m += ../staging/echo/| /usr/src/dahdi/drivers/dahdi/Kbuild # echo 'obj-m += echo.o' /usr/src/dahdi/drivers/staging/echo/Kbuild # cd /usr/src/dahdi # make # make install # cd /usr/src # tar zxvf dahdi-tools-2.1.0.2.tar.gz # cd /usr/src/dahdi-tools-2.1.0.2 # ./configure # make # make install # wget http://www.rowetel.com/ucasterisk/downloads/oslec-0.2.tar.gz # tar xvzf oslec-0.2.tar.gz # cd oslec-0.2 # make # insmod kernel/oslec.ko when i restart /etc/init.d/dahdi service it gives me following error in /var/log/message Mar 3 11:06:37 server1 kernel: echo: exports duplicate symbol oslec_hpf_tx (owned by oslec) Mar 3 11:06:37 server1 modprobe: WARNING: Error inserting echo (/lib/modules/2.6.18-92.1.22.el5/staging/echo/echo.ko): Invalid module format Mar 3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol oslec_create Mar 3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol oslec_update Mar 3 11:06:37 server1 kernel: dahdi_echocan_oslec: Unknown symbol oslec_free Mar 3 11:06:37 server1 modprobe: FATAL: Error inserting dahdi_echocan_oslec (/lib/modules/2.6.18-92.1.22.el5/dahdi/dahdi_echocan_oslec.ko): Unknown symbol in module, or unknown parameter (see dmesg) # cat /etc/dahdi/system.conf loadzone= in defaultzone = in span=1,1,7,ccs,hdb3 bchan=1-15 dchan=16 bchan=17-31 echocanceller=oslec,1-15,17-31 Is there anything missing or i am going wrong.. Help me out. Thanks in advance... -- Regards, Chandrakant Solanki -- _ -- 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] Dial Plan configuration in asterisk
On Fri, Feb 19, 2010 at 12:21 PM, Gopalakrishnaiyer Venugopal-Q16770
venui...@motorola.com wrote:
Hi experts,
The extensions.conf has the dial plan set as
exten == _988XXX,1,Dial(DAHDI/g1/${EXTEN},20)
I want to modify this so that i can dial numbers with more than 10 digits
for example like accessing an IVR menu.
Warm Regards
Venugopal G
*
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Hi
Try this
exten = _X.,1,Dial(DAHDI/g1/${EXTEN},20)
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Regards,
Chandrakant Solanki
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[asterisk-users] announce prompt to user
Hi I am using asterisk 1.6.0.5. I have one conference say 1234786 and in this conference 25 users are talking with each other.. In this 25 users, 5 is admin/marked and 20 are normal.. Admin user has rights to mute/unmute all user by executing action: meetmemuteall with meetme number. While executing MeetmeMuteAll action, this action will mute all 20 normal users but not admin.. This thing work fine but I want to play conf-muted prompt file to all these 20 users simultaneously.. Is there any way to do this...?? -- Regards, Chandrakant Solanki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] how apps/enter.h
Hello I want to know that how apps/enter.h data can be generated... I want to do same for conf-muted / conf-unmuted but not getting idea how data is generated for muted/unmuted same like apps/enter.h Help me out... -- Regards, Chandrakant Solanki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Meetme 'o' - what actually it does..??
Hi Can someone explain me what is the purpose for MeetMe Option 'o'.. If I defined 'o' with MeetMe option or If not defined with MeetMe option... What is the difference between these two if defined or not defined MeetMe 'o' option... -- Regards, Chandrakant Solanki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] error - sources for the 2.6.18-92.1.22.el5xen kernel
Hi Just download tar.gz of your kernel version and extract into /usr/src/kernels/ directory ! -- Regards, Chandrakant Solanki On Wed, Oct 21, 2009 at 1:34 PM, PATRICK KANGETHE patricemb...@yahoo.comwrote: while compiling zaptel drivers for my yeaster TDM800 hardware, I get this error; make[3]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect/mxml' gcc -o menuselect menuselect.o strcompat.o menuselect_curses.o mxml/libmxml.a mxml/libmxml.a -lncurses make[2]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect' make[1]: Leaving directory `/usr/src/zaptel-1.4.12/menuselect' make[1]: Entering directory `/usr/src/zaptel-1.4.12' echo You do not appear to have the sources for the 2.6.18-92.1.22.el5xen kernel installed. You do not appear to have the sources for the 2.6.18-92.1.22.el5xen kernel installed. exit 1 make[1]: *** [modules] Error 1 make[1]: Leaving directory `/usr/src/zaptel-1.4.12' make: *** [all] Error 2 i understand i have to install 2.6.18-92.1.22.el5xen kernel installed. How do i do this? Any help or guide will be highly appreciated. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MeetMe + SLA
Hello In app_meetme.c, there are two configuration file loaded i.e. meetme.conf and sla.conf.. I want to know that if i removed whole code of sla_* and sla.conf from app_meetme.c file.. Is this create problem for MeetMe application and register action/event... -- Regards, Chandrakant Solanki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDRs on call forward
Hi r u forwarding call using Originate action.. Which version of asterisk u used. On Thu, Sep 24, 2009 at 12:44 PM, John Fawcett john...@erba.tv wrote: In some circumstances I am transferring incoming calls to an external number (cell phone). Whenever this happens at the end of the call I get a single CDR representing the outgoing leg. There is no CDR for the incoming leg and no trace of incoming caller id in the CDR for outgoing leg. Is this expected behaviour? Is there a way to generate two CDRs one for the incoming and for the outgoing leg of forwarded calls? thanks John ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Regards, Chandrakant Solanki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPP + Duration
Hello How can I park call for 1 hour using sipp... Below command and xml file I am using... *# ./sipp -s 8600 -sf uac.xml -sn uac_pcap 127.0.0.1 -l 1 -r 1 -rp 5000* XML File === ?xml version=1.0 encoding=ISO-8859-1 ? scenario name=UAC with media send retrans=500 ![CDATA[ INVITE sip:[servi...@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp sip:sipp@ [local_ip]:[local_port];tag=[pid]SIPpTag09[call_number] To: sut sip:[servi...@[remote_ip]:[remote_port] Call-ID: [call_id] CSeq: 1 INVITE Contact: sip:s...@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: [len] v=0 o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip] s=- c=IN IP[local_ip_type] [local_ip] t=0 0 m=audio [auto_media_port] RTP/AVP 8 101 a=rtpmap:8 PCMA/360 a=rtpmap:101 telephone-event/360 a=fmtp:101 0-11,16 ]] /send recv response=100 optional=true /recv recv response=180 optional=true /recv recv response=200 rtd=true crlf=true /recv send ![CDATA[ ACK sip:[servi...@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp sip:sipp@ [local_ip]:[local_port];tag=[pid]SIPpTag09[call_number] To: sut sip:[servi...@[remote_ip]:[remote_port][peer_tag_param] Call-ID: [call_id] CSeq: 1 ACK Contact: sip:s...@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]] /send nop action exec play_pcap_audio=pcap/g711a.pcap/ /action /nop pause milliseconds=360/ nop action exec play_pcap_audio=pcap/dtmf_2833_1.pcap/ /action /nop pause milliseconds=360/ send retrans=500 ![CDATA[ BYE sip:[servi...@[remote_ip]:[remote_port] SIP/2.0 Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch] From: sipp sip:sipp@ [local_ip]:[local_port];tag=[pid]SIPpTag09[call_number] To: sut sip:[servi...@[remote_ip]:[remote_port][peer_tag_param] Call-ID: [call_id] CSeq: 2 BYE Contact: sip:s...@[local_ip]:[local_port] Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ]] /send recv response=200 crlf=true /recv ResponseTimeRepartition value=10, 20, 30, 40, 50, 100, 150, 200/ CallLengthRepartition value=10, 50, 100, 500, 1000, 5000, 1, 3600/ /scenario Is anything wrong with XML or what... -- Regards, Chandrakant Solanki ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
