[asterisk-users] Asterisk 13 / realtime voicemail creation
Hello, I'm having a problem when trying to create realtime stuff for asterisk 13 in mysql DB. Following https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic, I've successfully created the main tables using config schema. Update |script_location|to the schema to update. Asterisk currently supports two sets of schemas: 1. |config|- the set of schemas for Asterisk Realtime databases 2. |voicemail|- the schema for ODBC VoiceMail But when trying to use voicemail schema, I get the following error : [root@centrex5 ast-db-manage]# alembic -c config.ini upgrade head INFO [alembic.migration] Context impl MySQLImpl. INFO [alembic.migration] Will assume non-transactional DDL. ERROR [alembic.util] No such revision or branch 'a541e0b5e89' FAILED: No such revision or branch 'a541e0b5e89' [root@centrex5 ast-db-manage]# Any tip ? I'm using latest asterisk 13 sources. Regards, Cédric -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Return
Hi Aj
Can you perhaps show me an example as to how you would do it as I have tried
setting it very early but still doesn’t work
Kind Regards
Andrew Colin
Converged Telecoms (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Switchboard: +27 (0)10 591 4600
Email: and...@convergedgroup.net
Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
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-Original Message-
From: A J Stiles [mailto:asterisk_l...@earthshod.co.uk]
Sent: Thursday, July 9, 2015 10:03 AM
To: Andrew Colin; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Call Return
On Wednesday 08 Jul 2015, Andrew Colin wrote:
Hi Guys
I am trying to write a macro for a call return so for example
Anyone in the company transfers a call to another extension and it is
not answered etc it must return to the person who did the transfer
I have got it working but if the call originates externally for
example someone calls in to the switchboard and they transfer it then
it tries to return to the outside caller.
As doing a return to ${EXTEN}) wont work as that is the external party.
How do I declare a variable from the extension dialed?
So for example when 200 dials 201 I can capture the calling party(in
this case 200) and declare it as a variable?
You need to set a variable quite early in your extension logic, using a Set
command;
Set(dialled=${EXTEN})
and then later you can retrieve it as ${dialled} . This variable will
persist across context jumps, even although ${EXTEN} may have changed.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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Re: [asterisk-users] 11.18.0 patch against 11.17.0 running version failed to apply
On Thu, Jul 09, 2015 at 12:28:15AM +0200, Administrator TOOTAI wrote: zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p1 ../asterisk-11.18.0-patch patching file .version Hunk #1 FAILED at 1. 1 out of 1 hunk FAILED -- saving rejects to file .version.rej patching file ChangeLog Hunk #1 FAILED at 1. 1 out of 1 hunk FAILED -- saving rejects to file ChangeLog.rej The next patch would delete the file asterisk-11.18.0-rc1-summary.html, which does not exist! Assume -R? [n] Apply anyway? [n] Skipping patch. 1 out of 1 hunk ignored The next patch would delete the file asterisk-11.18.0-rc1-summary.txt, which does not exist! Assume -R? [n] Apply anyway? [n] Skipping patch. 1 out of 1 hunk ignored patching file asterisk-11.18.0-summary.html patching file asterisk-11.18.0-summary.txt As you can see, patch is against -rc1 not 11.17.0 ... The content of files has changed. patch refuses to change from an unfamiliar content. Either edit the patch file and remove .version (edit the version manually) or edit the patch file and edit the version form 11.7.0 to 11.8.0 . The content of the files you refer to is normally insignificant to the behaviour of Asterisk. Just remove them from the patch and be done with it. -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.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
Re: [asterisk-users] Asterisk 13 / realtime voicemail creation
Ø From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Oceanet - Cédric BASSAGET Sent: Thursday, July 09, 2015 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk 13 / realtime voicemail creation Ø Hello, I'm having a problem when trying to create realtime stuff for asterisk 13 in mysql DB. Following https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic, I've successfully created the main tables using config schema. Ø Update script_location to the schema to update. Asterisk currently supports two sets of schemas: Ø config - the set of schemas for Asterisk Realtime databases Ø voicemail - the schema for ODBC VoiceMail Ø But when trying to use voicemail schema, I get the following error : [root@centrex5 ast-db-manage]# alembic -c config.ini upgrade head INFO [alembic.migration] Context impl MySQLImpl. INFO [alembic.migration] Will assume non-transactional DDL. ERROR [alembic.util] No such revision or branch 'a541e0b5e89' FAILED: No such revision or branch 'a541e0b5e89' [root@centrex5 ast-db-manage]# Any tip ? I'm using latest asterisk 13 sources. Regards, Cédric Cedric, I’m not sure what the RIGHT setup is, but in 11 and so far in 13 I’ve been using the voicemail_users table and it seems to work just fine. I’m also using the MySQL direct driver, and NOT odbc for that. Travis -- _ -- 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 13 / realtime voicemail creation
I'm trying to make a new and clean install of asterisk 13 (I'm using 1.4/6/8 for many years). So I'm trying to follow the wiki, and install realtime tables through alembic. If nobody is able to help me, I'll use the old way, but I wish I won't have to. And I'm using ODBC too ;) Regards, Cédric OCEANET --- [AGENCE DU MANS] 7, rue des Frênes ZAC de la Pointe 72190 SARGE LES LE MANS [t] +33 (0)2.43.50.26.50 [f] +33 (0)2.43.72.21.14 [AGENCE D'ANGERS] 5, rue Fleming Angers Technopole 49066 ANGERS [t] +33 (0)2.41.19.28.65 [f] +33 (0)2.52.19.22.00 On 09/07/2015 17:34, Ryan, Travis wrote: Ø*From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Oceanet - Cédric BASSAGET *Sent:* Thursday, July 09, 2015 11:29 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Asterisk 13 / realtime voicemail creation ØHello, I'm having a problem when trying to create realtime stuff for asterisk 13 in mysql DB. Following https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic, I've successfully created the main tables using config schema. ØUpdate |script_location|to the schema to update. Asterisk currently supports two sets of schemas: Ø|config|- the set of schemas for Asterisk Realtime databases Ø|voicemail|- the schema for ODBC VoiceMail Ø But when trying to use voicemail schema, I get the following error : [root@centrex5 ast-db-manage]# alembic -c config.ini upgrade head INFO [alembic.migration] Context impl MySQLImpl. INFO [alembic.migration] Will assume non-transactional DDL. ERROR [alembic.util] No such revision or branch 'a541e0b5e89' FAILED: No such revision or branch 'a541e0b5e89' [root@centrex5 ast-db-manage]# Any tip ? I'm using latest asterisk 13 sources. Regards, Cédric Cedric, I’m not sure what the RIGHT setup is, but in 11 and so far in 13 I’ve been using the voicemail_users table and it seems to work just fine. I’m also using the MySQL direct driver, and NOT odbc for that. Travis -- _ -- 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 install gmime22
No one could assist? Could someone please tell me on which repository I can find Gmime22-devel for 64-bit Centos6.5? Is gmime-devel good or do I need to have gmime22-devel? What will happen if I don't install gmime22? Thank you... Harel Message: 3 Date: Mon, 6 Jul 2015 02:53:51 +0200 From: Harel Cohen ha...@mayorcom.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can't install gmime22 Message-ID: 00a601d0b786$3a7bffd0$af73ff70$@mayorcom.com Content-Type: text/plain; charset=us-ascii Hello list, I'm trying to install gmime22 package which is one of the packages reported as required by ./contrib/scripts/install_prereq test. Whatever I do I'm getting to a dead end. On the regular yum repositories that I use (centos, epel, rpmforge, asterisk, digium) it is not found. I've found it on Fedora repositories however trying to use those I get all sorts of errors: On fedora17 repository: ERROR You need to update rpm to handle: rpmlib(X-CheckUnifiedSystemdir) is needed by filesystem-3-2.fc17.x86_64 When I try to update rpm I'm getting a conflict between some systemd package to kernel On fedora21 repository: Not found On fedora20 repository it is reported as installed but with these errors: Error unpacking rpm package filesystem-3.2-19.fc20.x86_64 error: unpacking of archive failed on file /bin: cpio: rename Cleanup: glibc-common-2.12-1.149.el6_6.9.x86_64 7/10 Cleanup: bash-4.1.2-29.el6.x86_64 8/10 Non-fatal POSTUN scriptlet failure in rpm package bash warning: %postun(bash-4.1.2-29.el6.x86_64) scriptlet failed, exit status 127 Cleanup: glibc-2.12-1.149.el6_6.9.x86_64 9/10 warning: /etc/localtime saved as /etc/localtime.rpmsave ...and also the system hang on shutdown and won't boot again Could you please advise how to properly install this package? I'm on CentOS 6.5 with updates, running on AMD Athlon +4400 and 64bit Thank you... -- _ -- 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 install gmime22
On Thu, Jul 9, 2015 at 1:18 PM, Harel Cohen ha...@mayorcom.com wrote: No one could assist? Could someone please tell me on which repository I can find Gmime22-devel for 64-bit Centos6.5? Is gmime-devel good or do I need to have gmime22-devel? What will happen if I don't install gmime22? Thank you... Harel Message: 3 Date: Mon, 6 Jul 2015 02:53:51 +0200 From: Harel Cohen ha...@mayorcom.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can't install gmime22 Message-ID: 00a601d0b786$3a7bffd0$af73ff70$@mayorcom.com Content-Type: text/plain; charset=us-ascii Hello list, I'm trying to install gmime22 package which is one of the packages reported as required by ./contrib/scripts/install_prereq test. Whatever I do I'm getting to a dead end. On the regular yum repositories that I use (centos, epel, rpmforge, asterisk, digium) it is not found. I've found it on Fedora repositories however trying to use those I get all sorts of errors: On fedora17 repository: ERROR You need to update rpm to handle: rpmlib(X-CheckUnifiedSystemdir) is needed by filesystem-3-2.fc17.x86_64 When I try to update rpm I'm getting a conflict between some systemd package to kernel On fedora21 repository: Not found On fedora20 repository it is reported as installed but with these errors: Error unpacking rpm package filesystem-3.2-19.fc20.x86_64 error: unpacking of archive failed on file /bin: cpio: rename Cleanup: glibc-common-2.12-1.149.el6_6.9.x86_64 7/10 Cleanup: bash-4.1.2-29.el6.x86_64 8/10 Non-fatal POSTUN scriptlet failure in rpm package bash warning: %postun(bash-4.1.2-29.el6.x86_64) scriptlet failed, exit status 127 Cleanup: glibc-2.12-1.149.el6_6.9.x86_64 9/10 warning: /etc/localtime saved as /etc/localtime.rpmsave ...and also the system hang on shutdown and won't boot again Could you please advise how to properly install this package? I'm on CentOS 6.5 with updates, running on AMD Athlon +4400 and 64bit Thank you... gmime is only required for the res_http_post module. If you don't need that module, you really don't need that dependency. -- Matthew Jordan Digium, Inc. | Director of Technology 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call Return
On Wednesday 08 Jul 2015, Andrew Colin wrote:
Hi Guys
I am trying to write a macro for a call return so for example
Anyone in the company transfers a call to another extension and it is not
answered etc it must return to the person who did the transfer
I have got it working but if the call originates externally for example
someone calls in to the switchboard and they transfer it then it tries to
return to the outside caller.
As doing a return to ${EXTEN}) wont work as that is the external party.
How do I declare a variable from the extension dialed?
So for example when 200 dials 201 I can capture the calling party(in this
case 200) and declare it as a variable?
You need to set a variable quite early in your extension logic, using a Set
command;
Set(dialled=${EXTEN})
and then later you can retrieve it as ${dialled} . This variable will persist
across context jumps, even although ${EXTEN} may have changed.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_
-- 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] What Dial Plan function can access the contents of the SDP ?
Dear ASTERISK-users, What Dial Plan function can access the contents of the SDP ? If there is no Dial Plan Function for that, is there some another way to access contents of the SDP? Maybe via ARI ou AGI? If there is, how to access the SDP that comes with the SIP 183 response? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 (Brasil) -- _ -- 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
