Re: [asterisk-users] originate , callerid
On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten = 6003,n,Originate(SIP/6003@asterisk,app,meetme,6003,x) I use this patch https://messinet.com/rpms/browser/asterisk/asterisk-12-app_originate_callerid.patch because of https://issues.asterisk.org/jira/browse/ASTERISK-23016 -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] originate , callerid
On Thursday, December 25, 2014 03:53:44 PM Dmitry Melekhov wrote: 25.12.2014 15:46, Anthony Messina пишет: On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten = 6003,n,Originate(SIP/6003@asterisk,app,meetme,6003,x) I use this patch https://messinet.com/rpms/browser/asterisk/asterisk-12-app_originate_calleri d.patch Thank you! I'll try it. because of https://issues.asterisk.org/jira/browse/ASTERISK-23016 Unfortunately , get The issue you are trying to view does not exist. on this link :-( Sorry for referencing the wrong issue. The correct one is here https://issues.asterisk.org/jira/browse/ASTERISK-22992 -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Voicemail ODBC Storage
On Saturday, October 25, 2014 09:09:57 PM Dan Journo wrote: Is there any reason why ODBC voicemail storage requires varchar for most fields? For example, is there anything stopping me using a BIGINT or similar for origtime or INT for duration? It may cause you trouble when using PostgreSQL: https://issues.asterisk.org/jira/browse/ASTERISK-24441 -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] On kernel 3.16.2 : dahdi_rec: Invalid argument
On Wednesday, September 17, 2014 04:35:14 PM Russ Meyerriecks wrote: Patch for this has been committed to master here: http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=b9a8000bbd1 b6120f22627c105a2c2194dcc793d I expect to release a v2.10.1 for this soon. Thanks for the report. Thanks for the quick turnaround. It is much appreciated. -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] On kernel 3.16.2 : dahdi_rec: Invalid argument
On Saturday, September 13, 2014 03:15:57 PM sean darcy wrote: On 09/13/2014 01:52 PM, sean darcy wrote: On 09/13/2014 12:09 PM, sean darcy wrote: On Fedora 20, just updated to kernel 3.16.2. Rebuilt dahdi 2.9.2 against it. dahdi show channels works fine, but when I try to place a call: chan_dahdi.c:9345 dahdi_read: dahdi_rec: Invalid argument Any help appreciated. sean Updated to dahdi-2.10.0. No joy. Went back to kernel 3.15.10 - it works. sean FWIW, asterisk-11.10.2. I can report a similar issue using DAHDI 2.10.0 and Asterisk 13.0.0-beta1 where Asterisk overwhelms the log system, repeating the following until Asterisk and syslog consume all available CPU time, bringing the system to it's knees. chan_dahdi.c:11556 do_monitor: Read failed with -1: Invalid argument Unfortunately, I haven't found a solution, but reverting to kernel-3.15.10 resolves this issue as well. -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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 on CentOS7
On Wednesday, August 13, 2014 12:11:42 PM Carlos Chavez wrote: I installed CentOS 7 on a spare server along with all our Asterisk configuration system and the only thing that failed is the asterisk startup script included in the asterisk tarball. I guess because the startup system has changed so much that script will have to be updated. Everything else worked fine as far as I can tell but obviously I did not stress test that installation. You can use the systemd unit file I have here: https://messinet.com/rpms/browser/asterisk/asterisk.service?rev=2ce57c334633881bb4d1baaeb6ae1e63c032abdc It's what Fedora uses as well. This should work properly in EL7. Hopefully in not too long, I'll have Asterisk 13 builds for EL7, though I need to figure out a few dependency issues: https://messinet.com/rpms/ -A -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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 on CentOS7
On Thursday, August 14, 2014 03:15:16 AM Paul Greenberg wrote: Hi Anthony, That script does not work. My guess is that it is related to the way asterisk interacts with CentOS environment. Best Regards, Paul Greenberg, Esq. On Wednesday, August 13, 2014 12:11:42 PM Carlos Chavez wrote: I installed CentOS 7 on a spare server along with all our Asterisk configuration system and the only thing that failed is the asterisk startup script included in the asterisk tarball. I guess because the startup system has changed so much that script will have to be updated. Everything else worked fine as far as I can tell but obviously I did not stress test that installation. You can use the systemd unit file I have here: https://messinet.com/rpms/browser/asterisk/asterisk.service?rev=2ce57c334633 881bb4d1baaeb6ae1e63c032abdc It's what Fedora uses as well. This should work properly in EL7. Hopefully in not too long, I'll have Asterisk 13 builds for EL7, though I need to figure out a few dependency issues: https://messinet.com/rpms/ I do know that the Fedora EPEL project provides Asterisk for EL6, and I believe they will support it for EL7 as well, once EPEL 7 comes out of beta status. EL7 uses systemd, so I'm not sure that the regular init file will work properly without some tweaking, which is why I pointed you to the systemd unit file that is used by the Asterisk RPMs from Fedora 20, and the one I use with the RPM builds I make myself. How are you installing Asterisk on CentOS 7? Are you doing a regular make/install from source, or using RPM packages? -- Anthony - https://messinet.com/ - https://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Unable to build DAHDI-Linux in mock chroot
On Sunday, March 30, 2014 02:24:35 PM Anthony Messina wrote: On Sunday, March 30, 2014 07:07:47 PM Tzafrir Cohen wrote: On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote: On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote: Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc 1c 1fb1 2cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 https://issues.asterisk.org/jira/browse/DAHLIN-337 Thanks for your report. I hope to get it fixed soon. I should note that this specific target does not belong in a proper chroot build, as it downloads from outside. How can I get those firmware files properly included? This is the spec file I use: https://messinet.com/rpms/browser/dahdi-linux/dahdi-linux.spec DAHDI-Linux-2.9.1.1 fixes this issue. Thank you. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Unable to build DAHDI-Linux in mock chroot
On Sunday, March 30, 2014 07:07:47 PM Tzafrir Cohen wrote: On Fri, Mar 28, 2014 at 07:57:54PM -0500, Anthony Messina wrote: On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote: Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c 1fb1 2cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 https://issues.asterisk.org/jira/browse/DAHLIN-337 Thanks for your report. I hope to get it fixed soon. I should note that this specific target does not belong in a proper chroot build, as it downloads from outside. How can I get those firmware files properly included? This is the spec file I use: https://messinet.com/rpms/browser/dahdi-linux/dahdi-linux.spec -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Unable to build DAHDI-Linux in mock chroot
Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb12cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Unable to build DAHDI-Linux in mock chroot
On Friday, March 28, 2014 07:43:48 PM Anthony Messina wrote: Unfortunately, after http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commitdiff;h=6cebc1c1fb1 2cc0661f3810ef47ad33206b2e398 I am unable to build DAHDI-Linux in a mock chroot for packaging purposes. I believe this is related to the Makefile calling install_firmware with only 2 args, where install_firmware is a shell script with DESTDIR set to $3, which is empty. In this case, the DESTDIR evaluates to /usr/lib/hotplug/firmware, rather than buildroot_destdir/usr/lib/hotplug/firmware. make -C drivers/dahdi/firmware hotplug-install DESTDIR=/builddir/build/BUILDROOT/dahdi-linux-2.9.1-1.fc20.x86_64 HOTPLUG_FIRMWARE=yes make[1]: Entering directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/usr/lib/hotplug/firmware mkdir -p /builddir/build/BUILDROOT/dahdi- linux-2.9.1-1.fc20.x86_64/lib/firmware Installing dahdi-fw-oct6114-032.bin to hotplug firmware directories install: cannot create regular file '/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 https://issues.asterisk.org/jira/browse/DAHLIN-337 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] SIP Simple support on Asterisk 11
On Wednesday, June 19, 2013 11:11:17 AM Matthew J. Roth wrote: Eloi Bail wrote: I am trying to enable SIP SIMPLE communication in my test environment. I use the following which semi-enables message broadcasting to multiple devices so a user who receives a message can reply from any of the devices. http://messinet.com/trac/wiki/Asterisk/Message -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] RPM updates
On Monday, January 28, 2013 08:06:38 AM Anthony Messina wrote: On Monday, January 28, 2013 01:55:09 PM Steven Howes wrote: Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. I'm not sure which CentOs you're using, but I' build them for CentOS/EL 6: See http://messinet.com/rpms/ Of course, if you're looking for the latest possible build, it might take me a few days: https://admin.fedoraproject.org/updates/rpm-4.10.2-2.fc18 As a side note, I've been working out how to move forward with kernel module signing in Koji, as I've upgraded to Fedora 18. So far, the prospects for signed kernel modules are looking good. Though I wish Digium would just get DAHDI into the upstream kernel already :/ -A As of the update to rpm: https://admin.fedoraproject.org/updates/FEDORA-2013-2107, I'm now able to build EL6 packages again. I should have builds for dahdi-linux and dahdi- tools in the repos within an hour or so. (http://messinet.com/rpms). Also, for Fedora 18, and those interested in testing UEFI/Secure Boot and third-party kernel module signing, I've been working out the signed kernel module buildsystem integration thing and will post my public kernel module signing key to http://messinet.com/rpms sometime tonight. Fedora 18 DAHDI- Linux versions greater than dahdi-linux-2.6.2-0.2.rc1 will have the kernel modules signed. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] RPM updates
On Monday, January 28, 2013 01:55:09 PM Steven Howes wrote: Who do I need to poke to get the yum repository / RPM files updated? The dahdi RPMs are not up to date with the CentOS kernel versions any more, it's making doing an installation a bit tricky due to dependancies, I'd rather not roll back / remove new kernels if I don't have to.. I'm not sure which CentOs you're using, but I' build them for CentOS/EL 6: See http://messinet.com/rpms/ Of course, if you're looking for the latest possible build, it might take me a few days: https://admin.fedoraproject.org/updates/rpm-4.10.2-2.fc18 As a side note, I've been working out how to move forward with kernel module signing in Koji, as I've upgraded to Fedora 18. So far, the prospects for signed kernel modules are looking good. Though I wish Digium would just get DAHDI into the upstream kernel already :/ -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] dahdi 2.6.1+2.6.1 compile fails
On Saturday, November 03, 2012 09:32:37 PM Eric Smith wrote: How would I apply the patch included in the above url? [eric@pepper ~/src/asterisk-complete/asterisk/dahdi/2.6.1+2.6.1] $ patch DAHTOOL-60-f17.diff can't find file to patch at input line 5 Perhaps you should have used the -p or --strip option? You'll need to use.the -p or --strip option^^ But in your case, both you and DAHTOOL-60-f17.diff will need to be in the 2.6.1+2.6.1/tools/ directory before you issue: patch -p1 DAHTOOL-60-f17.diff -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Receiving and processing unsolicited XMPP messages with Asterisk 11
On Friday, August 31, 2012 06:48:46 PM Noah Engelberth wrote: I’m trying to set up a way that our users can send an XMPP message to Asterisk (unsolicited) to request information, such as voicemail status or the like. No matter what I set for the dialplan, I’m only seeing Asterisk execute the s,1 priority in the context defined in xmpp.conf for incoming messages, and then the “call” hangs up without executing further instructions. Anything I’ve tried to accomplish in that first priority has worked, but it never continues to an additional priority. This might be a separate, but related issue, as I am not using XMPP messaging yet, but I found that at least with SIP messaging in Asterisk 11, if I had a Hangup() in the dialplan for message routing, every message sent AFTER the first would fail just as you describe, since the first message routed through the dialplan hung up the channel. This did not happen to me in Asterisk 10. After removing the traditional Hangup() at the end, and restarting Asterisk, the messages route properly for me. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] [SOLVED] Re: CSipSimple audio issue with DAHDI/IAX2 calls
On 12/02/2011 11:37 AM, Anthony Messina wrote: I've just connected my new Android (Motorola RAZR) phone to Asterisk using CSipSimple and have discovered that on any call between CSipSimple and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will hear a rhythmic tapping as if my voice stream is being chopped up in equal parts about every 500ms or so. I can always hear the remote party without issue, regardless of the channel type. The issue occurs only on connections to DAHDI channels (even those that don't pass through the PSTN), and IAX2 connections to remote Asterisk servers. This issue occurs whether I am using WiFi, 3G or 4G connections on the Android. This does NOT occur on any SIP channels, local to my Asterisk box, or to others. I've investigated changing just about every setting on the Android with no resolution. It seems like some sort of timing issue and is strange to me that this issue is confined to DAHDI and IAX2 channels, but I'm no expert. I have tested using only res_timing_dadhi.so since I have the card, but that did not help either. Would anyone be willing to point me in the right direction for resolving this issue? Please let me know if any more information is required. Thanks in advance. -A Enabling the jitterbuffer=yes on the iax channel and setting Set(JITTERBUFFER(fixed)=default) prior to any calls to DAHDI channels seems to resolve the issue for now. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] CSipSimple audio issue with DAHDI/IAX2 calls
I've just connected my new Android (Motorola RAZR) phone to Asterisk using CSipSimple and have discovered that on any call between CSipSimple and an Asterisk DAHDI or IAX2 channel, the 'other' end of the call will hear a rhythmic tapping as if my voice stream is being chopped up in equal parts about every 500ms or so. I can always hear the remote party without issue, regardless of the channel type. The issue occurs only on connections to DAHDI channels (even those that don't pass through the PSTN), and IAX2 connections to remote Asterisk servers. This issue occurs whether I am using WiFi, 3G or 4G connections on the Android. This does NOT occur on any SIP channels, local to my Asterisk box, or to others. I've investigated changing just about every setting on the Android with no resolution. It seems like some sort of timing issue and is strange to me that this issue is confined to DAHDI and IAX2 channels, but I'm no expert. I have tested using only res_timing_dadhi.so since I have the card, but that did not help either. Would anyone be willing to point me in the right direction for resolving this issue? Please let me know if any more information is required. Thanks in advance. -A I am currently using the following on a Fedora 15 x86_64 system: Asterisk 1.8.7.1 built by mockbuild @ x86-13.phx2.fedoraproject.org on a x86_64 running Linux on 2011-10-17 21:42:11 UTC ]# cat /proc/dahdi/* Span 1: WCTDM/4 Wildcard TDM400P REV I Board 5 (MASTER) 1 WCTDM/4/0 FXOKS (In use) (EC: OSLEC - INACTIVE) 2 WCTDM/4/1 FXOKS 3 WCTDM/4/2 FXSKS (In use) (EC: OSLEC - INACTIVE) *CLI module show like timing Module Description Use Count res_timing_dahdi.soDAHDI Timing Interface 0 res_timing_pthread.so pthread Timing Interface 0 res_timing_timerfd.so Timerfd Timing Interface 1 *CLI core show settings PBX Core settings - Version: 1.8.7.1 Build Options: LOADABLE_MODULES Maximum calls: Not set Maximum open file handles: Not set Verbosity: 3 Debug level: 0 Maximum load average:0.00 Minimum free memory: 0 MB Startup time:10:23:07 Last reload time:10:23:07 System: Linux/2.6.32-131.2.1.el6.x86_64 built by mockbuild on x86_64 2011-10-17 21:42:11 UTC Default language:en Language prefix: Enabled User name and group: / Executable includes: Disabled Transcode via SLIN: Enabled Internal timing: Enabled Transmit silence during rec: Disabled Generic PLC: Enabled -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] Sytem Commands not executing
On 08/20/2011 07:00 AM, Tim King wrote: exten = h,n,System(/usr/bin/php /var/lib/asterisk/bin/faxnotify.php do you need the -f option to php? exten = h,n,System(/usr/bin/php -f /var/lib/asterisk/bin/faxnotify.php -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] Dahdi does not build against Kernel 3.0.0
On 08/06/2011 09:49 PM, Bruce Ferrell wrote: Errors follow: http://lists.digium.com/pipermail/asterisk-users/2011-July/264993.html -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] Looking for Email to Fax Solutions
On 06/08/2011 01:09 AM, Paddy Grice wrote: Hi All I am looking for a small scale Email to fax solution Searches seem to throw up AsterFax http://sourceforge.net/projects/asterfax/ which seems to go to http://www.noojee.com.au/products/noojee-fax/fax-overview/ email12fax http://wpkg.org/email2fax/index.php/Main_Page I would appreciate any comments on these or other solutions I am running asterisk 1.4 and I am looking for a small scale solution say 10 lines (ddis) While I designed it with Asterisk 1.6 or 1.8 in mind, you may try this: http://messinet.com/trac/wiki/AsteriskFAXGateway I have some time next week if it needs some tweaks to work with Asterisk 1.4. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] Faxing with Asterisk 1.8.4 T.38
On 05/24/2011 01:07 PM, e...@erols.com wrote:
I have tried faxing to the DID from 2 different fax machines connected to
different POTS lines. One fax machine is a Xerox Workcentre, and the other
is a Brother Intellifax. Can you provide some more information about your
setup? If you wouldn't mind sharing your sip.conf settings, and maybe any
other FaxForAsterisk related dialplan settings I would be greatly
appreciative. I feel like we must have *something* really stupid set
incorrectly. The faxes usually attempt to send, and appear to be properly
switching to T.38, but usually end up failing with a receive partial. We
are currently using the Digium fax driver, but have also tried it with
spandsp.
sip.conf peer:
[ipcomms]
type=peer
host=64.154.41.100
canreinvite=nonat
context=ipcomms
insecure=port
sendrpid=yes
trustrpid=yes
t38pt_udptl=yes
videosupport=no
contactdeny=0.0.0.0/0.0.0.0
deny=0.0.0.0/0.0.0.0
permit=64.154.41.100/255.255.255.255
disallow=all
allow=ulaw
extensions.conf:
[ipcomms]
exten = your_ipcomms_number_here,1,Goto(receivefax,s,1)
[receivefax]
exten =
s,1,Set(ARRAY(CALLERID(DNID),FAXOPT(headerinfo),FAXOPT(localstationid),to_email)=${EXTEN},Asterisk
FAX Gateway,+1 NXX NXX ,amessina)
same = n,ReceiveFAX(/var/spool/asterisk/fax-gw/archive/${UNIQUEID}.tif)
same = n,Hangup()
exten = h,1,AGI(fax-gw/fax-gw.agi,${CONTEXT})
exten = h,n,Hangup()
And I use my own Asterisk FAX Gateway program:
http://messinet.com/trac/wiki/AsteriskFAXGateway
--
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Re: [asterisk-users] Faxing with Asterisk 1.8.4 T.38
On 05/20/2011 01:20 PM, e...@erols.com wrote: #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38 DIDs from our current ITSP, but we have been unable to make it work. I am more than happy to purchase new DIDs from a different provider if they will consistently work and are fairly priced. I use http://www.ipcomms.net/ with a free inbound DID for faxes. I always receive T.38. I use http://www.gafachi.com/ for outbound T.38. I have had excellent service from both. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] Echocancellation OSLEC vs MG2 ?
On 04/27/2011 02:06 PM, satish patel wrote: Which echo cancellation is good between OSLEC and MG2. Dahdi by default use MG2 echo cancellation on channel. If i want to use OSLEC then what should i need to do ? Do i need to recompile dahdi with OSLEC ? Yes, you would need to compile the OSLEC kernel module. Or, if you are using a RedHat/Fedora based distro, you're welcome to use the dahdi-linux and dahdi-linux-kmod RPMS I build here. I include OSLEC with the dahdi-linux-kmod build. http://messinet.com/rpms/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: OpenPGP digital signature -- _ -- 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] [1.4] Failed callfile doesn't jumptofailedextension
On 03/18/2011 05:43 PM, Gilles wrote:
On Fri, 18 Mar 2011 10:08:52 -0700 (PDT), Steve Edwards
asterisk@sedwards.com wrote:
On Fri, 18 Mar 2011, Danny Nicholas wrote:
I believe you will achieve the desired result by replacing ${REASON}
with ${HANGUP_CAUSE}.
REASON is documented as being valid in the 'failed' extension. If it is
not working as you expect it to, maybe you could read through the source
(/usr/src/asterisk-x.x.x.x/main/pbx.c) to understand why.
You could always submit a patch...
HANGUP_CAUSE should be HANGUPCAUSE.
Thanks guys. In which case does Asterisk jump to the failed
extension?
You need to define the 'failed' extension in your context to have the
${REASON} variable set (I've found).
exten = failed,1,NoOp(Failure reason is: ${REASON})
--
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Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Tuesday, October 26, 2010 01:16:29 pm Stephen Reese wrote: http://messinet.com/trac/wiki/AsteriskGVGateway (AGI script) Is your .agi and .git the same script? I do not have a git client on this host to see for myself. I keep the AGI in Git as a version control system. But, you can view the AGI source here: http://messinet.com/trac/browser/gv/gv.agi And at the very bottom of that page is a link to download it as an individual file here: http://messinet.com/trac/export/b3229dbba3e01c887b3bdf6b0e0d93e897bd8a59/gv/gv.agi This is not the same thing as what is in the Changelog. I am using Asterisk 1.6 with this AGI. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Monday, October 25, 2010 07:30:22 am Stephen Reese wrote: Does the AGI have to be used? In this example http://www.davidvossel.com/?p=28 I see mention of a script, but not in this one: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ I believe I missing the connection in how the whole process actually works therefore making troubleshooting a little difficult. I was hoping with the release of 1.6.0 there wouldn't be a lot of bandage work to get it to play nicely with Google Voice. Since Google Voice (GV) doesn't let us connect diretly via SIP, IAX2, etc., for outbound calls, it acts basically like a fancy click-to-call application. So... You need Asterisk to login into GV, and initiate the call. GV will dial the number you tell it to, then connect it to one of your GV numbers. In my case, the AGI is what connects to GV and initiates the call. GV, then dials the number I told it to dial, then connects it with my ipKall number (which I have as one of my GV numbers). In Asterisk, the outbound call runs the AGI and places the channel in the DB, then waits for an incoming call via my inbound ipKall trunk. Once the ipKall comes into Asterisk, the Bridge command is used to bridge the original (with the matching DB entry) call-- the call that is coming in from GV through ipKall. I suppose you don't need that AGI and could probably do this using Curl in the dialplan. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Integrating Asterisk 1.8 with Google Talk and Google Voice
On Sunday, October 24, 2010 05:23:13 pm Stephen Reese wrote:
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
For Google Voice, I use an ipKall number for the inbound trunk. Here are the
relevant sections of my extensions.conf:
; inbound ipKall trunk (to which Google Voice makes the connection)
[ipkall]
exten = ipKall-number,1,GotoIf($[${DB_EXISTS(gv/channel)} = 1]?gv)
same = n,Goto(default,s,1)
same = n(gv),Bridge(${DB_DELETE(gv/channel)})
same = n,AGI(gv/gv.agi,hangup)
same = n,Hangup()
; outbound Google Voice initiation
[gv-out]
exten = _X.,1,AGI(gv/gv.agi,call)
same = n,While($[${DB_EXISTS(gv/channel)} = 1])
same = n,Wait(0.3)
same = n,EndWhile()
same = n,Hangup()
And the AGI (written in Bash) is here:
http://messinet.com/trac/wiki/AsteriskGVGateway
http://messinet.com/trac/browser/gv/gv.agi
--
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Re: [asterisk-users] channel variables in AGI
On Saturday, August 21, 2010 02:19:00 pm Steve Edwards wrote: Wow. I thought I knew a bit about bash. I made notes on 19* different lines I have no clue what they do. It's going to take me hours to figure these out so I can add them to my repertoire. *) I'm sure there's more nuggets in there but my eyes are glazing ove Believe me, I've glazed over the Bash man page for quite some time to get that interface going ;) If you're interested in mail to fax (and back), give it a shot. I could use some testers. Have a good night. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Click2call from an OpenOffice document
On Friday, August 20, 2010 10:35:10 am Olivier wrote: Yes, adding this kind of link should do it but I'm looking for a solution which automatically insert whatever is needed to launch a call. wouldn't it be difficult to know exactly which applications are available on the system which has the document open? the solution might be different for every reader of that document. the previously proposed web link-based solution would provide you with the greatest reach. perhaps we aren't exactly sure what you are trying to accomplish. what is your end goal? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] channel variables in AGI
On Wednesday, August 11, 2010 11:08:37 am Tino wrote: #!/bin/bash -x T=$agi_uniqueid I want to save value of 'agi_uniqueid' channel variable into a variable called 'T' in my script When executing and AGI from the dialplan, it will dump out it's variables immediately, so you need to tell Bash to read them in and write them to whatever variables you want. For example, see: http://messinet.com/trac/asterisk-fax-gw/browser/fax-gw.agi#L622 Here, I set the variable name from Asterisk to the variable value from Asterisk. So I end up with: agi_uniqueid=123456... (or whatever the uniqueid was) Then I could go on to say T=$agi_uniqueid -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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 app_fax, T.30, weird received faxes
On Monday, July 26, 2010 09:55:38 am Tzafrir Cohen wrote: I suppose I should make a list of known good packages, and put it on that FAQ page. GIMP is useless for FAX. Not only does it get the shape of the images wrong, it can only display the first page of a FAX. I am not familiar with gqview or feh. The package I usually use to display FAXes on Linux/BSD machines is okular. That seems to behave very well, unless you have a really old version. convert and the rest of imagemagick should handle multi-page tiff (e.g. convert it to PDF). libTIFF's tiff2pdf works well also. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Distinctive ring for INTERNAL calls only? How to do it?
On Monday, July 19, 2010 01:03:57 am Peter Childs wrote: One of the problems with Distinctive Ring tones is that its not consistent, between different phones so if you have a mix of phone types you have a problem. Agreed. I only mentioned what I did since I, along with the OP use Aastra phones. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Distinctive ring for INTERNAL calls only? How to do it?
On Wednesday, July 14, 2010 01:44:54 pm bruce bruce wrote: Using Elastix (FreePBX + Asterisk 1.4.2x combination) with Aastra phones, how can one receive distinctive ring tones for INTERNAL calls ONLY? Using Aastra 4801 CT phones... [external-context] ; Calls entering from outside the system exten = 1234,1,SIPAddHeader(Alert-Info: info=Bellcore-dr2) ; Double Ring same = n,Dial(SIP/... [internal-context] ; Calls routed from within the system exten = 1234,1,Dial(SIP/... ; No special ring -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Lookup ${EXTEN} in database, update context/route if found... AGI?
On Tuesday 11 May 2010 01:25:30 pm Tim Nelson wrote: I have a handful of Asterisk 1.4.x installations where users dial 'outbound calls' to the PSTN even though the destination is on the same Asterisk box or on another Asterisk box on the same network. Instead of paying twice for the call to go out to the PSTN on one channel and back in on another channel, I'd like the ability to lookup the destination number in a MySQL database and if found, change the way the call is routed. The call routing update could be as simple as issuing a Goto() to change contexts or priorities in the current context. you could use DUNDi for this and avoid external DB and/or AGI. -a -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] dahdi-2.2.1 kernel-2.6.32: working for anyone?
On Sunday 07 March 2010 09:16:55 am sean darcy wrote: Well, I've figured it out, at least for me. Another driver was grabbing the TDM400P: netjet. added netjet to /etc/modprobe.d/blacklist.conf. I think you can do this by: cat /lib/modules/`uname -r`/modules.pcimap | grep 00e159 e159 is the vendorid for the TDM400P. You'll see all the drivers that use e159. Then lsmod | grep those drivers other than wctdm. If you see one loaded, blacklist it. sean thanks, sean! that worked for me: http://messinet.com/trac/rpms/changeset/141 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] dahdi-2.2.1 kernel-2.6.32: working for anyone?
On Sunday 07 March 2010 05:10:02 pm sean darcy wrote: Good. Glad it we figured it out. BTW, is your src.rpm for dahdi-linux available? sean Here you go. -A http://messinet.com/pub/fedora/linux/updates/12/SRPMS/dahdi- linux-2.2.1-2.fc12.src.rpm -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] dahdi-2.2.1 kernel-2.6.32: working for anyone?
On Saturday 06 March 2010 09:18:13 pm sean darcy wrote: I have a TDM400. Just updated Fedora 12 to kernel 2.6.32. Rebuilt and installed dahdi-2.2.1. kernel modules loaded. lsmod | grep wctdm wctdm 37233 0 dahdi 194985 1 wctdm lsmod | grep dahdi dahdi 194985 1 wctdm crc_ccitt 1549 2 dahdi,isdnhdlc dmesg: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.1 . dahdi_dummy: Trying to load High Resolution Timer dahdi_dummy: Initialized High Resolution Timer dahdi_dummy: Starting High Resolution Timer dahdi_dummy: High Resolution Timer started, good to go which is much less dmesg on 2.6.31: dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.2.1 ACPI: PCI Interrupt Link [APC1] enabled at IRQ 16 wctdm :01:05.0: PCI INT A - Link[APC1] - GSI 16 (level, low) - IRQ 16 Freshmaker version: 73 Freshmaker passed register test Module 0: Installed -- AUTO FXS/DPO Module 1: Installed -- AUTO FXS/DPO Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (3 modules) and dahdi_cfg fails: dahdi_cfg -vv DAHDI Tools Version - 2.2.1 DAHDI Version: 2.2.1 Echo Canceller(s): Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01) Channel 02: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02) Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04) 3 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) I tried dahdi svn r8255 from today. Same result. If I reboot with 2.6.31, all's well. Am I missing something? Amazing, I just finished the same thing with, unfortunately, the same result as you on both i686 and x86_64. I'll keep googling :) -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Losing local SIP phones when internet goes down?
On Thursday 04 February 2010 23:22:27 Alex Samad wrote: What I have seen on my asterisk box when I had a up/down adsl line was that the asterisk box couldn't do dns resolution and would hang( well no other internal calls could be made, seemed like some sort of semaphore was stuck) when the adsl came up and dns could be done, everything worked fine again I can confirm that exact same behavior: 1.6.1.12 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. -- _ -- 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] Dahdi and oslec
On Monday 04 January 2010 07:16:49 Joseph L. Casale wrote: Looking at the source in the rpms from the asterisk package site appears that oslec is not built and enabled for the kmod rpms. Anyone know an existing repo or have direction on how to enable this to built for those rpms? I build them for the kernels I use (Fedora 12 x86_64 and i686.PAE) but you can check http://messinet.com/trac/rpms/ and checkout the svn-build-rpm tool to build from an svn checkout if you already have a build setup configured. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Dahdi and oslec
On Tuesday 05 January 2010 12:21:15 Joseph L. Casale wrote: So this script builds them with the dahdi-tools-libs package requirement, I thought the fedora spec built all of these? Any idea? Fedora packages the dahdi-tools* suff, but can't include the kernel modules. I did not realize you were using CentOS. You'll need to change some of the definitions at the top of the file to match whatever version of dahdi-tools you have installed (if CentOS has them). If not, the Fedora specs and patches are here: http://cvs.fedoraproject.org/viewvc/rpms/dahdi-tools/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Dahdi and oslec
On Tuesday 05 January 2010 17:09:32 Joseph L. Casale wrote: From what I can tell so far, I can continue to use his user tools unchanged but I need to apply this patch to the tar file in the dahdi-linux-2.2.0.2-1_centos5.src.rpm and rebuild it, but that , `dahdi-linux` pulls in atrpms.net also provides packages for RHEL5, if those would work. http://atrpms.net/dist/el5/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Dahdi and oslec
On Tuesday 05 January 2010 17:30:31 Joseph L. Casale wrote: Just on my way to work on this server now, this would be great! That way I don't have to work all night:) Does the atrpms ones finally do oslec? I don't use them myself, but I was thinking that the RHEL5 spec files might be another place to look for what you need to build with OSLEC included, more specifically for CentOS. I just tried taking a look at ATrpms, but the site is having some connection issues at the moment. How about this -- another CentOS repo: http://www.zultron.com/2009/03/dahdi-rpms/ Otherwise I'm afraid you'll need to patch and compile. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] ASTERISK and SNMP
original message- From: mickael ropars mrop...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Fri, 27 Nov 2009 11:18:30 +0100 - Hi Michal, thanks a lot for you quick answer I appreciate. I run your commands and I have the following answer [localhost snmp]# snmpwalk -c local -v 1 localhost asterisk no answer [localhost snmp]# snmpwalk -c local -v 2c localhost asterisk ASTERISK-MIB::asterisk = No Such Object available on this agent at this OID you may need to do export MIBS=+ASTERISK-MIB snmpwalk ... -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery ___ -- 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] OriginateResponse Event
On Monday 05 October 2009 12:33:47 Danny Nicholas wrote: What are the limitations of ActionID? In all of the examples I see, it is usually 1 or some integer. Can it be a timestamp like uniqueid? I use AMI as part of an external bash application and I usually specify the ActionID to the something unique outside of Asterisk itself, such as as the external bash process id $$ or the process id combined with the date in nanoseconds. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] 1.6.0.5: I need a really simple analog SendFax dialplan
On Wednesday 23 September 2009 01:44:31 sean darcy wrote:
Does anyone use SendFax for analog faxing?
Yes. I have two contexts as follows:
[outbound]
exten = _X.,1,Dial(DAHDI/G2/${EXTEN})
[sendfax]
exten = s,1,SendFAX(${FAXFILE})
exten = h,n,Hangup()
When I want to send a fax, I initiate a call from a call file or the AMI using
a local channel.
Channel: Local/s...@sendfax
Exten: number to be dialed
Context: outbound
Priority: 1
--
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Re: [asterisk-users] Older Aastra phones and Asterisk 1.6
On Monday 07 September 2009 16:27:30 Carlos Chavez wrote: It seems that older Aastra phones (9112i, 9133i, 480i, 480i CT) have a problem with the new SIP implementation in Asterisk 1.6.X that makes them unable to dial. They can receive calls but when you attempt to dial the phone remains silent. You can see in core show channels that the first channel is active and it is impossible to kill it without restarting Asterisk. The solution I found for this is to set session-timers=refuse in sip.conf and now I am able to send calls. I suppose this is a problem with the firmware of those phones as newer versions of Aastra phones (5Xi) work without the modification. I have several Aastra 480i CT phones on three separate Asterisk 1.6.1.6 on Fedora 11 (asterisk-1.6.1.6-1.fc11.x86_64) and do not see this problem. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] features.conf : feature map == getting feature to work
On Monday 07 September 2009 13:40:16 jonas kellens wrote: [applicationmap] opnemencallee = #3,self/callee,Monitor,wav,/var/samba/profiles/jonaskl/recording,m FeatureName = DTMF_sequence,ActivateOn[/ActivatedBy],Application[,AppArguments[,MOH_Class]] it looks like /var/samba/profiles/jonaskl/recording is in the spot for [,MOH_Class] -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Asterisk + CDRTool
On Wednesday 12 August 2009 08:30:33 am harry R wrote: Or maybe can suggest another CDR GUI ? i began work on this a while ago... http://messinet.com/trac/webcdr+/ it's what i use now, though i'd like to add more features, etc. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] e164.org and tollfree ENUM records
Recently, I've been having issues with the URIs returned from e.164.org and
toll free calls. It seems that the URIs that are returned from ENUMQUERY and
ENUMRESULT are no longer the proper numbering schemes that the poviders use.
I've been using the following [enum] template in my outbound route for quite
some time with great success until recently.
[enum](!)
exten = _X.,n,Set(ARRAY(i,id)=1,${ENUMQUERY(+${EXTEN},ALL,e164.org)})
exten = _X.,n,Set(max=${ENUMRESULT(${id},getnum)})
exten = _X.,n,While($[${i} = ${max}])
exten = _X.,n,Set(uri=${ENUMRESULT(${id},${i})})
exten = _X.,n,Exec(${IF($[${uri:0:3} =
sip]?Dial(SIP/${uri:4},40,KL(720:12)T):NoOp(ENUM URI is not of type
SIP))})
exten = _X.,n,Exec(${IF($[${uri:0:4} =
iax2]?Dial(IAX2/${uri:5},40,KL(720:12)T):NoOp(ENUM URI is not of
type IAX2))})
exten = _X.,n,Set(i=${MATH(${i}+1,i)})
exten = _X.,n,EndWhile()
The console results are as follows. Each of sip-happens, siptollfreegateway,
and voipmich return either a 404 or 403 error.
I'm wondering if their ENUM records are old and no longer represent how
callers should reach their servers.
== ast_get_enum(num='+18002662278', tech='ALL', suffix='e164.org',
options='', record=1
== ENUM options(): pos=1, options='0'
== ast_get_enum() profiling: FAIL, 8.7.2.2.6.6.2.0.0.8.1.e164.org, 405 ms
-- Executing [18002662...@outbound:3] Set(SIP/aastra-sip1-0c004d98,
ARRAY(i,id)=1,0) in new stack
-- Executing [18002662...@outbound:4] Set(SIP/aastra-sip1-0c004d98,
max=3) in new stack
-- Executing [18002662...@outbound:5] While(SIP/aastra-sip1-0c004d98,
1) in new stack
-- Executing [18002662...@outbound:6] Set(SIP/aastra-sip1-0c004d98,
uri=sip:164164180018002662...@sip.tollfreegateway.com) in new stack
-- Executing [18002662...@outbound:7] Exec(SIP/aastra-sip1-0c004d98,
Dial(SIP/164164180018002662...@sip.tollfreegateway.com,40,KL(720:12)T))
in new stack
-- Limit Data for this call:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 164164180018002662...@sip.tollfreegateway.com
-- Got SIP response 480 Temporarily Unavailable back from 204.8.45.222
-- SIP/sip.tollfreegateway.com-140f2228 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [18002662...@outbound:8] Exec(SIP/aastra-sip1-0c004d98,
NoOp(ENUM URI is not of type IAX2)) in new stack
-- Executing [18002662...@outbound:9] Set(SIP/aastra-sip1-0c004d98,
i=2) in new stack
-- Executing [18002662...@outbound:10] EndWhile(SIP/aastra-
sip1-0c004d98, ) in new stack
-- Executing [18002662...@outbound:5] While(SIP/aastra-sip1-0c004d98,
1) in new stack
-- Executing [18002662...@outbound:6] Set(SIP/aastra-sip1-0c004d98,
uri=sip:164164180018002662...@tollfree.sip-happens.com) in new stack
-- Executing [18002662...@outbound:7] Exec(SIP/aastra-sip1-0c004d98,
Dial(SIP/164164180018002662...@tollfree.sip-
happens.com,40,KL(720:12)T)) in new stack
-- Limit Data for this call:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 164164180018002662...@tollfree.sip-happens.com
-- SIP/tollfree.sip-happens.com-140f3668 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [18002662...@outbound:8] Exec(SIP/aastra-sip1-0c004d98,
NoOp(ENUM URI is not of type IAX2)) in new stack
-- Executing [18002662...@outbound:9] Set(SIP/aastra-sip1-0c004d98,
i=3) in new stack
-- Executing [18002662...@outbound:10] EndWhile(SIP/aastra-
sip1-0c004d98, ) in new stack
-- Executing [18002662...@outbound:5] While(SIP/aastra-sip1-0c004d98,
1) in new stack
-- Executing [18002662...@outbound:6] Set(SIP/aastra-sip1-0c004d98,
uri=sip:180018002662...@tf.voipmich.com) in new stack
-- Executing [18002662...@outbound:7] Exec(SIP/aastra-sip1-0c004d98,
Dial(SIP/180018002662...@tf.voipmich.com,40,KL(720:12)T)) in new
stack
-- Limit Data for this call:
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 4
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
-- Called 180018002662...@tf.voipmich.com
-- SIP/tf.voipmich.com-140f2228 is circuit-busy
--
Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
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Re: [asterisk-users] DUNDi Errors (ENCREJ)
On Tuesday 30 June 2009 05:01:42 am srinivas Antarvedi wrote: - To resolve this i tried to remove all keys in all servers and once again created and distributed the loaded in each system with keys init command but stilll i am getting the same error can anybody help me out??? Thanks and regards srinivas antarvedi try module reload res_crypto.so or restart your asterisk servers. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] AMI and Originate on 1.6.0.5
On Friday 29 May 2009 11:20:31 am David Backeberg wrote: On Fri, May 29, 2009 at 4:22 AM, DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: i cannot originate call from AMI interface here are my Originate action Packet Channel: SIP/111 where 111 Is my SIP phone number which registered with my asterisk server I can login with this manager User and while trying with above action i got Response: Error Message: Channel Not Specified You need a destination. SIP/111 needs an @destination to be a complete channel name. i apologize for not being able to get to the right bug # right now, but there was a manager bug that was fixed in following versions of asterisk. the patch that does the fix is simple: http://cvs.fedoraproject.org:80/viewvc/rpms/asterisk/F-10/0016-Fix-a-reversed- logic-ast_strlen_zero.patch?revision=1.1view=markup -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] howto store local exchange prefixes ?
original message- From: Jimmy Godbout s...@inbox.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 25 May 2009 18:01:11 -0800 - Check on www.localcallingguide.com. You'll find all npanxx that are local to your exchange. Jimmy -Original Message- From: seandar...@gmail.com Sent: Mon, 25 May 2009 21:39:30 -0400 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] howto store local exchange prefixes ? Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 sean darcy wrote: I've looked at the Berkeley DB. That works pretty well, if the exchanges are all stored. But it looks like the exchanges have to be entered 1 by 1 from the CLI. And can only be reviewed, corrected, or deleted from the CLI. I haven't found any simple frontend for the DB. I do this be writing a dialplan which adds those entries. The first entry checks to see if the DB has been initialized and if so, skips to the lookup. Otherwise it loads each into the database before the lookup. It's very easy to write a quick script to generate the dialplan code. Barry Maybe I've not explained this correctly. I know, or can look up, the 40+ local exchanges that are local. I can parse the dial EXTEN to determine the exchange. I can check the exchange against a DB. I want to determine which exchanges are local. I do not want to store an exchange dialed by a user. How can I store a lot of 3 digit numbers which I then can check against an EXTEN to determine a local number? in addition to localcallingguide, if your pstn connection is from att, you can take a look at the script i made to grab only the local calls (incurring no local-toll or long distance charges) which areband a and band b. https://messinet.com/trac/telephony-tools/wiki/LocalCallingAreaGrabber -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery ___ -- 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] change AGI script return result
On Friday 15 May 2009 03:49:05 pm Hristo Benev wrote: I came up to this solution, but is there a way to change the AGISTATUS variable to FAILURE - We have it always SUCCESS if the script you use exits successfully (without an error), AGISTATUS will always be SUCCESS even if it didn't do what you wanted. you need to have your script exit with something other than 0 if you'd like to have AGISTATUS not be SUCCESS. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Connection to non-human numbers
On Thursday 16 April 2009 09:52:45 Danny Nicholas wrote: I've got 1.4.21.2 using Polycom 501 phones and Zap lines. Most of my calls come in and go out fine with the exception of Mechanized answering devices. When I call my 401K plan (1-800-777-401K) the call will last exactly one minute. The call never bridges, so even though the connection is made, Asterisk hangs up at the end of the Dial command. Any suggestions? are you using progress detection on your zap lines? callprogress=yes progzone=us this may be the problem. i have the same issue when i dial into my work voicemail out of my asterisk box at home. try setting callprogress=no by the way, for anyone else, might there be a way to enable/disable callprogress from the dialplan? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Phone Directories/Asterisk/SIP/directory.html
On Tuesday 10 March 2009 13:32:37 Elizabeth Steinke wrote: Greetings! We are using cisco 7940 phone with SIP and asterisk. We would like to be able to have phone directories available on the phones that are sourced from active directory. Are their any scripts that can connect to the AD server via LDAP and then create the directory.html file for the phones? Thanks! Liz i made something for Aastra phones and LDAP that probably wouldn't be too hard to adapt: http://messinet.com/AastraDirectory -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Asterisk 1.6.1-rc1 with OpenAIS and different subnets
On Thursday 05 March 2009 07:10:59 Kevin P. Fleming wrote: Peter Mueller wrote: Has anybody set up such an installation and/or is OpenAIS able to transfer the devstates over different subnets? Haven't found docs and hints for this use case. The method OpenAIS uses to communicate between nodes is designed for a very low latency local connection; it is not designed to work across routed connections. Russell Bryant has spent some time talking to the OpenAIS developers about this, but so far there doesn't seem to be a good solution. true, that's why i'm hoping that distributed presence via dundi comes about sooner, rather than later :) -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Continue in dialplan on hangup
On Friday 27 February 2009 14:03:19 Doug Lytle wrote: Daniel Hazelbaker wrote: Specifically, I am trying to play around with setting up a fax server. I can receive the fax, but sometimes the sending fax hangs up If your looking into setting up a reliable fax server and your not doing it over IP, then your best results will be using HylaFAX+ and iaxmodem with Asterisk. HylaFAX+ handles the printing/re-faxing/fax2email of all inbound/outbound faxes via it's FaxDispatch script. It's a 'Set and forget (tm)' package. I absolutely love it. Doug Or, if you're using Asterisk 1.6 and looking to try something new, take a look at http://messinet.com/AsteriskFAXGateway -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Continue in dialplan on hangup
On Friday 27 February 2009 17:02:16 Daniel Hazelbaker wrote: Or, if you're using Asterisk 1.6 and looking to try something new, take a look at http://messinet.com/AsteriskFAXGateway I'll take a look at both packages. I hadn't given HylaFAX(+) any thought as when I searched initially I found just the old version of HylaFAX that last had a release in 2007, which makes me a bit nervous. :) hylafax is excellent. i used it myself. i was just looking to try and start something simpler. also, hylafax uses the concept of modems, so even if you're doing an all-software solution, you'll still need iaxmodem or t38modem. in asterisk 1.6, the SendFAX and ReceiveFAX applications do all of that work for you and all you need is a way to streamline getting faxes into and out of asterisk. in my case, with the AsteriskFAXGateway, it's e-mail. but if you look at the script, it's just a bash script and rather than having incoming faxes be e-mailed, just have them go to a printer. here is the current script: http://messinet.com/viewvc/asterisk-fax-gw/trunk/fax-gw?view=markup on a side note, i'dlove some folks who are willing to test and help me work out scenarios that i've not thought of. there has been interest on digium's side of getting this to be part of the default tarball, but i'd like to get some testing and feedback (and devel help) before i do that. -a -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] MINNESOTA: TwinCities Asterisk Users Group Meeting, this Saturday. Valentine's Day Feb/14/2009 11:30am
On Friday 13 February 2009 07:54:48 Jeff LaCoursiere wrote: Is there a Chicago area users group? If not is there any interest in creating one? there is: http://groups.google.com/group/asterisk-chicago though it's fairly inactive. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Asterisk 1.6.0.5 and Aastra phones...
On Friday 13 February 2009 11:39:07 Carlos Chavez wrote: Anybody here is able to use Aastra phones with Asterisk 1.6.0.5? Making calls is not a problem but when you receive a call it always drops at 1:45 minutes, always! I use 1.6.0.5 with 3 Aastra 480i CT phones and have no issues with any of them dropping calls from either the base station or the wireless handsets. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Allison Smith, Music-on-Hold Parody--outstanding.
On Wednesday 31 December 2008 18:07:26 Karl Fife wrote: Allison Smith just created a hysterical parody music on hold Parody. Whatever you were doing, stop, and dial this number to listen to it: 360-519-5689. 2 minutes. I just gave her a few ideas, but she took it and ran with it--she chose the audio and did the mix-down and everything. Really funny!! -Karl hah! that's great! it's nice to be in chicago. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] outging ---asterisk -bug
On Tuesday 23 December 2008 01:05:40 jordan pan wrote: Hi everyone, when i use the automated dial out,I found that once the zap answerd,the contex will be exectued, but i don't hope do it ,i hope when extern phone answered ,then ,the context will be exectued. Anyone can help me solve the problem! the call file is: Channel: Zap/g0/15015895665 Context: myivr RetryTime: 60 MaxRetries: 2 Waittime: 60 Extension: 808 Priority: 1 Callerid: 15015895665 [myivr] exten = s,1,Background(test) exten = s,n,WaitExten Thanks in advance! you could try using progress detection in zapata.conf. i had the same issue, until i enabled progress detection. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Setup ReceiveFax(), fax2mail, mime-construct - but now Sendmail :(
On Tuesday 23 December 2008 05:00:10 Tzafrir Cohen wrote: And this is a reminder: they don't queue mail. Hence if they fail to deliver once, the mail is lost. May not be the best idea for sending mail over the internet. This is a great reminder as many systems use graylisting. Emails to those systems will fail with a send once type system. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Setup ReceiveFax(), fax2mail, mime-construct - but now Sendmail :(
On Friday 19 December 2008 20:24:11 sean darcy wrote:
Using 1.6 on Fedora Core 9 I'm trying to receive faxes. I've got this far:
[incoming-fax]
exten =
s,1,Set(FAXFILE=/var/spool/asterisk/fax/${STRFTIME(${EPOCH},,%Y%m%d%H%M)}-0
${CALLERIDNUM}) exten = s,2,ReceiveFAX(${FAXFILE}.tif)
exten = s,3,Hangup()
exten=h,1,System(/usr/local/bin/fax2mail --cid-number 0${CALLERIDNUM}
--cid-name home fax --dest-name admin --dest-email ${admin_email}
-f ${FAXFILE})
which all seems work well on the CLI. No errors.
fax2mail uses mime-contruct to send the fax by sendmail. That didn't work.
No email. /var/log/maillog:
Dec 19 21:04:53 asterisk sendmail[2628]: mBH2mWvQ006043:
to=ad...@myco.com, ctladdr=r...@localhost.localdomain (0/0),
delay=2+23:16:09, xdelay=00:00:00, mailer=esmtp, pri=6640305,
relay=mx01.1and1.com., dsn=4.0.0, stat=Deferred: Connection timed out
with mx01.1and1.com.
Dec 19 21:04:53 asterisk sendmail[2628]: mBH2mWvS006043:
to=ad...@myco.com, ctladdr=r...@localhost.localdomain (0/0),
delay=2+23:16:09, xdelay=00:00:00, mailer=esmtp, pri=6640312,
relay=mx00.1and1.com., dsn=4.0.0, stat=Deferred: Connection timed out
with mx00.1and1.com.
I've avoided MTA's like sendmail for a _long_ time. So I need help.
1. Is this the right list to try to resolve this? If not, which list?
2. postfix seems to considered much easier to configure than sendmail.
Do I install postfix? If so, will this work out of the box?
3. If sendmail, what's the magic configuration?
i'm still working on this, but take a look at
http://messinet.com/viewvc/asterisk-fax-gw/trunk/
currently, i use postfix, which seems easier to me to configure than sendmail
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Re: [asterisk-users] docs for rxfax in 1.4 or app_fax in 1.6?
On Saturday 13 December 2008 11:34:07 Loic Didelot wrote: Hello, which package are you using to get the application ReceiveFAX under asterisk 1.4? since this is still in development, for me, i'm working with asterisk 1.6. this package is mostly independent of the fax system within asterisk (just change the dialplan to use tx_fax instead of sendfax, basically. as far as getting app_txfax and app_rxfax to work in 1.4, i never tried that. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] docs for rxfax in 1.4 or app_fax in 1.6?
On Friday 12 December 2008 12:08:55 sean darcy wrote:
I just want to pdf and email faxes coming in over pstn on a TDM400P.
Outgoing faxes would just go out over pstn, not through asterisk.
All the voipinfo , etc, howto's are quite complicated. And most use
third party apps like Hylafax.
I thought there was a rxfax and txfax in 1.4. And 1.6 had app_fax. I'm
now using 1.4.22, but I'd go to 1.6 if it made this easier.
But I've found no docs or sample configs for either 1.4 or 1.6. In fact,
1.4.22 ( nor addons nor 1.4.23 rc2 ) have no rx{tx}fax.c files that I'd
expect.
I do have spandsp installed, FWIW.
i'm working on a email - fax gateway right now to do just that. it works
well, but is unpolished and basically undocumented at this point. you can see
the work in svn here: http://messinet.com/viewvc/asterisk-fax-gw/trunk/
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Re: [asterisk-users] MSet()
On Friday 12 December 2008 15:41:54 Mark Michelson wrote: would result in a variable called FOO being set to the value hello,BAR=world. The MSet application was added to facilitate being able to set multiple variables in a single application call. If using MSet, the above would instead result in a variable called FOO being set to the value hello and a variable called BAR being set to world. what about Set(ARRAY(var1,var2)=value1,value2) ? is the MSet() app a better/quicker way to do this? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Any 1.6 SendFAX example ?
On Thursday 27 November 2008 05:03:00 Olivier wrote: Hi, Do you have any example showing how to use SendFAX ? I can see several examples of ReceiveFAX but not a single one showing SendFAX. i'm working on a script to incorporate e-mail - fax gatewaying with asterisk using programs that are already available in linux. there are simple examples here: http://messinet.com/viewvc/asterisk-fax-gw/trunk/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] force channel hangup
On Thursday 27 November 2008 20:25:49 Kelvin Chan wrote: Hi guys, I have 1 zap channel in my house shared among couple people. If someone dials 911, I want that zap channel to be disconnected right away to make way for the 911 call. I dug through voip-info.org and didn't find much. Any hints? i use this: http://messinet.com/index.php?page_name=Asteriskwikipage=Asteriske911 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Setting up to reveive faxes.
On Saturday 22 November 2008 09:10:39 am Gordon Henderson wrote: On Sat, 22 Nov 2008, Noah Miller wrote: Hi Ken - Hey, all. When I last was heavily into Asterisk (1.0.x), setting up to receive faxes was, well, a PITA, what with having to patch the Asterisk install with various driver patches and this, that, and the other. Is that still true? Is there a fax HOWTO out there that reflects Asterisk 1.4.x? Not sure if you mean IP faxing or TDM faxing, but I don't think you'll need to do any patching. In general check out: http://www.voip-info.org/wiki-Asterisk+fax For IP faxes, check out the wiki here: http://www.voip-info.org/wiki/view/Asterisk+T.38 AFAIK: 1.4.x can't terminate IP faxes - only pass through, while 1.6.x can pass through and terminate with the help of spandsp. Does this mean that 1.4 lost an ability that 1.2 currently has? Right now, with 1.2, I use spandsp (or rather the RxFAX application which uses spandsp) to terminate faxes and save them as TIF files, then outside asterisk email them to the appropriate destination... I'm currently looking at moving to 1.4 and this is something that's essential for me... I guess it's time to look deeper into this. with 1.6, i'm using this: http://messinet.com/?page_name=AsteriskFAXGateway to do email-fax gatewaying in both directions. i suppose it could also work with 1.4's tx_fax and rx_fax which i think are in the agx-ast-addons package available at http://sourceforge.net/projects/agx-ast-addons/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Old mantis e-mails
On Thursday 30 October 2008 11:57:29 am Daniel Hazelbaker wrote: Is it just me or has mantis been holding onto old e-mail and finally sending it? i'm getting them too. even the original your license agreement is accepted email. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] fax / t38 gateway
On Friday 24 October 2008 11:49:15 am Wilton Helm wrote: 1. Why would anyone originate a FAX via VoIP? If it has to go through a bunch of translation steps at both ends, it would seem better to simply scan the document (assuming it isn't in electronic form to begin with) and attach it to an E-Mail. fax is a legally accepted form of document transport in many states. signing and returning mortgage/lease/title/contract papers during the sale of a house for example. theoretically fax has two legally defined and assignable endpoints--they can be identified with a person or an organization. though i do wish fax would go away in favor of real (affordable) electronic signatures or universally accepted gpg -- we all get one with our social sec. numbers or something, but this is nation-dependent. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] setup for fax machine
On Monday 13 October 2008 01:25:12 am Gordon Henderson wrote: On Sun, 12 Oct 2008, sean darcy wrote: Becasue of all the issues with fax over voip, we want to use pstn for our fax machine, but not dedicate a line just to fax. I'm thinking of having asterisk answer the pstn line, check for fax tones, and route appropriately. In zapata ( chan_dahdi ) set faxdetect=incoming then the dial plan would have [incoming-pstn] exten = fax,1,Dial(DAHDI/1) ; the fax machine exten = fax,2,Hangup() exten = s,1,Answer() exten = s,2,Dial(DAHDI/2) ; internal extension . Would this work? I'll need another TDM410 card to do this, so I'd like some reassurance before I go purchase it. You need to Answer() the call first, then insert a Wait(2). During that time, asterisk will be listning for fax tone and jump to the fax extension if it hears them. So: exten = s,1,Answer() exten = s,n,Wait(2) Ringing() Dial(DAHDI/2) Hangup() and exten = fax,1,Dial(DHADI/1) exten = fax,n,Hangup() would you also be able to detect fax tones during the Backgound app? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] DAHDI aaaaaaaaaaaaaaarrrrrrrrrghhhhhhhhh :((((
On Thursday 09 October 2008 09:57:30 pm Steve Totaro wrote: Now I have not touched any of that code, but to me, it would have been much simpler to change names, then change functionality later. Make DAHDI a drop in replacement for Zaptel, in fact, if memory serves me correctly that is what someone at Digium explained, it was merely a find and replace operation. i agree with the idea that a drop in should have been created, and functionality built from there. i'm not sure i feel as strongly as the OP suggested in the subject subject line he created. for those users of 1.6, you're now in a corner: go any higher than 1.6 beta 9 and you need dahdi. no overlap with zaptel was created here. perhaps zaptel could have been kept in until 1.6.1, giving the 0.0.1 overldap :) i've been trying to follow the devel of dahdi closely, even to the point of building kernel modules for fedora that *should* work with jeffrey ollie's tools packages, just approved by the package reviewers. still, there are some concerning things that have been lingering, namely for me: http://bugs.digium.com/view.php?id=13443 well, anyway, just my two cents. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: Address already in use
On Wednesday 17 September 2008 09:18:58 pm hugolivude wrote: I think it's better to find out what is listening on port 4520. CentOS 5 Asterisk 1.4.20 Presumably my other Asterisk server is listening on 4520. The problem here is that I can change the port, and it will work... until I reboot. When I reboot the problem reappears and I can fix it by changing the port again. Any other thoughts? what else on THIS machine is uusing port 4520? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Wi-SIP 802.11f - Inter Access Point Protocol HANDOFF
On Saturday 30 August 2008 11:51:49 am Karl Fife wrote: Let's say a 'YES' only counts if you had a bona-fide handoff. In other words, you began in place 'A' (within range of AP#1 but OUTSIDE the range of AP#2), AND THEN MOVED to place 'B' (in range of AP#2, but completely outside the range of AP#1) WITYOUT dropping the call. wouldn't the ap ranges have to have *some* overlap, lest the basic network connection be dropped, whereby dropping the voip call? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Wi-SIP 802.11f - Inter Access Point Protocol HANDOFF
On Saturday 30 August 2008 01:35:10 pm Karl Fife wrote: Indeed you're right. You'd have area covered by AP 'A' only, AP 'B' only and area of AB overlap, Picture a venn diagram: http://upload.wikimedia.org/wikipedia/commons/5/56/Venn-diagram-AB.png right. it's just your inital description made it sound as if there was NO overlap. is the delay from the switchover too much to cover the call without dropping? i was thinking, on a laptop, where you are not streaming realtime, a few seconds you might not notice, but on a voip phone during a call... -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Digium Coffee anyone? PCI Expresso? WTF?
On Tuesday 26 August 2008 11:44:42 pm Karl Fife wrote: I'll be that none of the other coffee makers can handle anywhere NEAR 60 voice channels, and don't get me started about HPEC! http://www1.shopzilla.com/8N_-_cat_id--13050802__oid--680459759 Good find! Does it grind it's own beans? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Global VoIP Calls?
On Saturday 23 August 2008 03:56:15 am Gavin Henry wrote: What setup would you recommend for making VoIP calls whilst bringing latency down between offices at: * Edinburgh * Kuala Lumpur * Singapore * Tokyo * Seoul * Beijing * San Francisco Some of the Asia offices are 300ms some 200ms. Are the calls to be within company offices, or E.164 numbers? Either way, you could set up DUNDi nodes at each location. Have primary peering routes between Edinburgh, San Francisco and Tokyo (or whichever is your least latent Asian peer), for example. Then peer the Asian peers with Tokyo. DUNDi caching will reduce route lookup times. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] OT - Asterisk-Stats - Billsec instead of Duration
On Thursday 21 August 2008 08:26:47 am Olivier wrote: Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use i started working from that software to come up that was maybe simpler and css-based. i'm still messing around with it, but you can look at https://messinet.com/svn/projects/asterisk-stat/trunk/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] OT - Asterisk-Stats - Billsec instead of Duration
On Friday 22 August 2008 07:54:43 am Anthony Messina wrote: On Thursday 21 August 2008 08:26:47 am Olivier wrote: Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use i started working from that software to come up that was maybe simpler and css-based. i'm still messing around with it, but you can look at https://messinet.com/svn/projects/asterisk-stat/trunk/ sorry, i forgot to mention that all you'd need to change is the 'formatDuration' function in include/config.inc.php i currently have the billing duration contained in the abbr tag and it is displayed on hover over the duration. but that is easily switched. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] USA Lata AreaCode Database
On Thursday 14 August 2008 03:09:42 pm roberto wrote: I'm looking for some free LATA X Area Code database. Anyone have any idea where can i found? this site has lots of info: http://www.localcallingguide.com/ -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] [Dundi] Looking for new peers/limited time only
On Thursday 31 July 2008 11:36:18 am Anthony Messina wrote: For a limited time only, Messinet Secure Services (me) will be offering DUNDi E.164 termination to the entire +1 country code. I'd like to encourage more peering within the US, but peering is open to anyone. See http://messinet.com/?page_name=DUNDi for peering information. During the past 12 days, Messinet Secure Services routed over 2000 free VoIP calls to the +1 country code using DUNDi. We managed to log over 60 hours of call time served and many of our existing peers were able to find new peers willing to join the growing DUNDi cloud. Now that the free calls have come to an end, I want to thank all those who participed in this event. I hope that it has been a success for all involved. See http://messinet.com/?page_name=DUNDi for peering information. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] [Dundi] Looking for new peers/limited time only
On Thursday 31 July 2008 11:36:18 am Anthony Messina wrote: For a limited time only, Messinet Secure Services (me) will be offering DUNDi E.164 termination to the entire +1 country code. I'd like to encourage more peering within the US, but peering is open to anyone. See http://messinet.com/?page_name=DUNDi for peering information. Between 11AM yesterday and 11AM today, Messinet Secure Services serviced over 430 free calls totaling over 18 hour and 15 minutes of free calling time to any number using the +1 country code. All this was accomplished via DUNDi E.164 peering. Judging by the numbers for yesterday, today will probably be the last day this offer, so peer up now! See http://messinet.com/?page_name=DUNDi for peering information. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Looking for a more robust Click to Dial/Web Dial solution than AsteriDex (potential for a bounty!)
On Tuesday 22 July 2008 02:58:38 pm Jason Lixfeld wrote: I was looking for a Click to Dial/Web Dial solution and I found AsteriDex. I'm looking for something I can use on the road where I can hit an internal Click to Dial/Web Dial page from my cell, tap on a number and have it bridge a call between my cell and the other number so I can use my office PBX for company LD, clients see my company's CallerID etc. AsteriDex seems to have almost everything that I'm looking for, but I need something with a few more enhancements and I'm wondering if such a thing exists or if I need this to be custom made. - I need something that can import a phone book from vcards and/or pull names and numbers from an LDAP directory, not just MySQL (I don't even really care about keeping my numbers in AsteriDex's MySQL database). - I need something that, when I hit it with a web browser (specifically, Mobile Safari on my iPhone 3G), will also have a field where I can enter a number manually, incase a number I need to dial isn't in the directory. - I need something that has hooks to customize the CallerID fields. It should have configuration hooks somewhere where I can set a couple of different the CallerID Names and Numbers, then have the option to select which CallerID gets set when the outbound call to the client is made. I have control over the CallerID that gets sent to the Telco. Please advise, and if someone is looking for a few extra bucks, let me know how much you will charge to develop something like this. I can provide a deposit if you are credible. you could look at: http://messinet.com/?page_name=MessinetSecureDirectory it's just my toying around with the concepts, but the basics are there. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Best Linux distribution to use in Asterisk server
On Friday 09 May 2008 10:19:23 am equis software wrote: Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? Thanks I have used Fedora 7 8 on both i386 x86_64. I have used the RPMs from atrpms.net in the past and now I use the RPMs from Fedora. I have never had any trouble with Asterisk, though Zaptel has been quirky now and then, but that was always a bug that was resolved upstream in the next version of Zaptel. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Is Asterisk really good??
On Thursday 10 April 2008 02:14:17 pm Steve Edwards wrote: 1 - Can you really make free outgoing calls from let's say Portland OR, to Frankfurt Germany? No. There is no free lunch. It takes electricity, bandwidth, and depending on who you want to call in Germany, termination. though you may want to look into DUNDi and http://asterisk.li/peeringgraf.htm there seems to be a fairly strong DUNDi network in Europe -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] How to configure Voice mail for multi users.
On Thursday 20 March 2008 05:06:29 am Mian M Asif wrote:
Hi eric,
can you please tell me how can i save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1),
thanks for you help,
regards,
Asif
Message: 14
Date: Wed, 19 Mar 2008 10:39:22 -0500
From: Eric Wieling [EMAIL PROTECTED]
Subject: Re: [asterisk-users] How to configure Voice mail for multi
users.
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Mian M Asif wrote:
Hi All,
i want to configure voice mail on Asterisk 1.4 for multiple users. let
me explain you the scenario.
i have 10 users with the name of
1000,2000,3000,4000,5000,6000,...and these user can call to each
other. Now i want to configure separate voice mail box for separate
user.
my extensions.conf . settings below..
[voicemail]
exten = _X.,1,Dial(SIP/${EXTEN})
exten = _X.,n,NoOp(Dial Status: ${DIALSTATUS})
exten = _X.,n,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Background(vm-nobodyavail)
exten = s-NOANSWER,n,VoiceMail([EMAIL PROTECTED])
exten = s-NOANSWER,n,Hangup()
As I'm sure you know, ${EXTEN} is the value of the currently executing
extension, in the example above your line would be parsed as:
exten = s-NOANSWER,n,VoiceMail([EMAIL PROTECTED]) You would have
seen this if you were watching the Asterisk console when a call failed
to go to Voicemail.
Find some other way. You could save the value of EXTEN in a different
variable before the Goto(s-${DIALSTATUS},1), but there are many, many,
many other ways.
the variable setting i'm not helpful with, but how about:
[context]
exten = 2200,n,Dial(${DEVICE},20,kKotTwW)
exten = 2200,n,Goto(vm,${EXTEN},1)
[vm]
exten = _X.,1,Exec(${IF($[${DIALSTATUS}
= BUSY]?VoiceMail(${EXTEN},b):VoiceMail(${EXTEN},u))})
exten = _X.,n,Playback(vm-goodbye)
exten = _X.,n,Hangup()
the only part that gets repeated for each exten are the two lines in [context]
--
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Re: [asterisk-users] What replaces Macro() now? And how do you do the equivalent?
On Sunday 09 March 2008 09:59:32 pm Philip Prindeville wrote:
http://bugs.digium.com/view.php?id=11969
If Macro()/MacroExit() is deprecated, how does one go about achieving
the same functionality with Gosub()/Return()?
i agree--an excellent question. Since Macro is depreciated and I am using
1.4, planning on upgrading to 1.6, I started to mess around a bit with this.
Currently, I use:
[incoming]
; Calls from outside
exten = 2202,1,Dial(${F1000G},30,kKotTwW)
exten = 2202,n,Goto(vm-external,${EXTEN},1)
[internal-extens]
; Calls from inside
exten = 2202,1,Dial(${F1000G},30,kKotTwW)
exten = 2202,n,Goto(vm-internal,${EXTEN},1)
And the Goto directs you to this context:
[vm](!)
exten = _X.,1,Answer(500)
exten = _X.,n,Exec(${IF($[${DIALSTATUS}
= BUSY]?VoiceMail(${EXTEN},b):VoiceMail(${EXTEN},u))})
exten = _X.,n,Playback(vm-goodbye)
exten = _X.,n,Hangup()
[vm-internal](vm)
[vm-external](vm)
I use two separate, but equal contexts so that in voicemail, i could
distinguish between whether a call was from-extension or from-phonenumber
I didn't need to use Gosub since I don't need a return.
I'm sure there are plenty of ways that are simpler, etc. They have not yet
arrived at my brain's doorstep.
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Re: [asterisk-users] TDM400P dialout problem
On Thursday 28 February 2008 05:41:55 pm Al Baker wrote: Is this only on the _64 zaptel or will affect ALL zpatel 1.4.9 ? -Original Message- From: Russell Bryant [EMAIL PROTECTED] Sent: Feb 28, 2008 6:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] TDM400P dialout problem Anthony Messina wrote: Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: This should be fixed in Zaptel 1.4.9.2. thanks russell. in reply to al: with 1.4.7.1, i had no problems with either x86_64 or i386. with 1.4.8, i386 worked, but x86_64 did not. with 1.4.9 and 1.4.9.1, neither worked. i use the rpms from atrpms.net for fedora 7 i'm looking forward to 1.4.9.2, but am also concerned about http://bugs.digium.com/view.php?id=12099 as i saw this error with 1.4.9 and 1.4.9.1 on both platforms. unfortunately, due to my work schedule, i did not have time to debug the differences between the platforms. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] TDM400P dialout problem
Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [EMAIL PROTECTED]:1] Dial(Zap/1-1, Zap/3/8801234) in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing '8801234' [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:2030 zt_call: Deferring dialing... -- Called 3/8801234 [Feb 25 02:37:00] WARNING[7194]: chan_zap.c:3835 zt_handle_event: Detected alarm on channel 3: No Alarm -- Hungup 'Zap/3-1' == Everyone is busy/congested at this time (1:0/0/1) [Feb 25 02:37:00] NOTICE[7082]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 3 So the call fails and if I weren't using a test extension: exten = 2111,1,Dial(Zap/3/8801234) it would proceed in the dialplan. asterisk]# cat /proc/zaptel/1 Span 1: WCTDM/0 Wildcard TDM400P REV I Board 1 (MASTER) 1 WCTDM/0/0 FXOKS (In use) 2 WCTDM/0/1 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 Where do I go with this? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] DUNDi with two servers
On Sunday 24 February 2008 02:01:10 am arkda wrote: Hi, I'm having difficulties with using DUNDi between two servers. If it were three I think I could control looping by limiting TTL, but with two I'm not sure how to prevent a loop causing bad things to happen. I've tried ttl=1 but things still blow up. The DUNDi configurations are pretty simple and work just fine in both directions as long as only one of them is using the switch = DUNDi/context. dundi lookup number@dundified works great as well as test calls. What is the proper method of handling DUNDi between only two servers? Should I be using a dummy context on one server to handle this? I'm listing the relevant files below for only one server for brevities sake. --- dundi.conf [general] department=Test Lab organization=My Test lab locality=Anywhere stateprov=CA country=US [EMAIL PROTECTED] phone=+55 entityid=00:11:22:33:44:55 cachetime=5 ttl=1 autokill=yes [mappings] dundified = internal,0,SIP,[EMAIL PROTECTED][EMAIL PROTECTED] ,nopartial [55:44:33:22:11:00] model=symmetric host=server2.domain.com inkey=dundikey outkey=dundikey include=dundified permit=dundified qualify=yes order=primary --- extensions.conf [general] static = yes writeprotect = no clearglobalvars = no [globals] [default] include = internal include = parkedcalls [internal] include = external include = parkedcalls switch = DUNDi/dundified exten = 300,1,Dial(SIP/300) exten = 300,n,Hangup() exten = 5551234567,1,Goto(300,1) exten = 301,1,Dial(SIP/301) exten = 301,n,Hangup() exten = 8885551212,1,Goto(301,1) exten = _NXXNXX,1,Dial([EMAIL PROTECTED]) exten = _NXXNXX,n,Hangup() [external] exten = 5551234567,1,Goto(internal,300,1) --- sip.conf [dundified] type=friend dbsecret=dundi/secret context=internal [voipprovider] type=friend host=voipprovider.web dtmfmode=rfc2833 insecure=port,invite disallow=all allow=g729 context=external [300] type=peer callerid=300 username=300 secret=secret host=dynamic context=internal [EMAIL PROTECTED] notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 [301] type=peer callerid=301 username=301 secret=secret host=dynamic context=internal [EMAIL PROTECTED] notifyringing=yes notifyhold=yes limitonpeers=yes call-limit=2 Thanks in advance! i believe that you don't want to have your mappings point to a context that includes the switch = DUNDi/... statement. The switch is what a server uses to interface to the rest of the DUNDi world. your mappings should point to what a particular host is serving up, not including the rest of the DUNDi world. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Variable setting in AMI Originate
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
Anthony Messina wrote:
Working with asterisk 1.4; using the AMI Originate command, it is
possible to do something like:
Variable: CDR(accountcode)123456
Or must the variable names be var[n] where n is a number?
I'd like to set the accountcode for a Local channel that originates a
call.
Thanks. -A
Anthony,
I may not understand your question, but setting variables from the AMI is
easy enough:
Action: Originate
Channel: local/[EMAIL PROTECTED]
Context: to_meetme
Exten: s
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
That was exactly my question (even though I forgot the =sign). However, I am
not able to get that to work for reason. I'm trying to set the
CDR(accountcode) on the first leg of the call and am using Channel: Local/...
I am able to get it to work if I use Variable: var1=12345 then, use
CDR(accountcode)=${var1} in the dialplan, but I was hoping to avoid this
hack.
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Re: [asterisk-users] Variable setting in AMI Originate
On Friday 15 February 2008 01:49:46 pm Richard Lyman wrote:
Anthony Messina wrote:
On Friday 15 February 2008 10:21:33 am Lee Jenkins wrote:
*snipped
Priority: 1
Variable: CALLERID(num)=${DEV_NAME}|CALLERID(name)=Conference Waiting
Async: true
That was exactly my question (even though I forgot the =sign). However,
I am not able to get that to work for reason. I'm trying to set the
CDR(accountcode) on the first leg of the call and am using Channel:
Local/...
I am able to get it to work if I use Variable: var1=12345 then, use
CDR(accountcode)=${var1} in the dialplan, but I was hoping to avoid this
hack.
why not just add
Account: 12345
to the originate?
(side note: you can also have multiple Variable: lines (some versions of
asterisk have issue with the | from what i hear)
so the above would look like
...
Variable: CALLERID(num)=${DEV_NAME}
Variable: CALLERID(name)=Conference Waiting
those are bad examples as you should just use CallerID:
Callerid: Conference Waiting DEVNUMBER
i hope this helps.
that does work like a charm--it sets the accountcode, except that, for some
reason, i can't access the CDR(accountcode) value during call time.
i CAN see it in channel variables, etc. but ${CDR(accountcode)} evaluates to
nothing--it's blank. it even show up in the CDR after the call is over.
my dialplan basically says, set the callerid to the accountcode (which is my
real pstn number). since i have some users for which i need to block
outbound callerid on the pstn line, this was a convenient way to distinguish
between my devices with my accountcode and those without.
now that i'm trying to originate calls from a secured webpage using the
manager, it seems like my method isn't working well :(
i'd like to keep the callerid=Name internal exten settings in my devices
so internally, we see the extensions instead of a full pstn number.
how else would i be able to set the outbound/external callerid per device?
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[asterisk-users] Variable setting in AMI Originate
Working with asterisk 1.4; using the AMI Originate command, it is possible to do something like: Variable: CDR(accountcode)123456 Or must the variable names be var[n] where n is a number? I'd like to set the accountcode for a Local channel that originates a call. Thanks. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] ATA with pulse dialing support over FXS
On Saturday 02 February 2008 02:45:09 am Alberto Pastore wrote: I need a bunch of them to convert some old fashioned rotary phones into VoIP ones (I'd like to disassemble the ATAs to remove the boards from the plastic case and to fit them into the phones after making the appropriate changes to the phones' exterior to add holes for rj-45 socks and dc power input) look into Rotatone: http://www.oldphoneworks.com/antique-phone-parts/by-type/part.asp?currency=USDPhonePart=729PhonePartType=138 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Problem with DTMF dialing
On Thursday 31 January 2008 11:52:09 pm Ian wrote: Sorry for taking so long to reply, This email got lost in translation, again. Ian Ian said the following on 30-Jan-08 03:57 PM Thaks for the speedy reply Tzafrir Cohen said the following on 30-Jan-08 12:37 PM: On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote: Hi all I have a small problem here. I asked this question on another asterisk mailing list, but nobody seemed to be able to help me there. We are running * Asterisk 1.4.17 * Libpri 1.4.3 * Zaptel 1.4.8 did you use zaptel-1.4.7 prior to this? did it work then? if so, it may be related to http://bugs.digium.com/view.php?id=11855 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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] Conditional Dial
On Friday 04 January 2008 10:19:07 pm Tilghman Lesher wrote:
On Friday 04 January 2008 17:52:52 bilal ghayyad wrote:
Is there a command that can let me execute the
Dial(.) if {CALLERIDNUM}= ..? Without using
GotoIf?
Exec(${IF($[${CALLERID(num)} = 12345]?Dial(foo):NoOp)}
are there any advantages/disadvantages/differences of using the above instead
of the ExecIf application?
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Re: [asterisk-users] 1.4.17 - Breaks park announce?
On Thursday 03 January 2008 02:39:44 pm Brent Torrenga wrote: Upgraded to 1.4.17 and found that the parking slot is not announced. Reverted back and all is well. Anyone else notice this behavior? it seems to work fine here. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ --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
