similar to: Logging registration/unregistration of peers/extensions in database

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

We tried to use the CURL fn to POST json, but it's sent as form data and there seems no support for changing the Content-Type header. We switched to invoking curl in the shell. All the documentation I could find says there is just one parameter for the url and an optional second for POST body. Is there an undocumented way to set Content-Type? -------------- next part -------------- An HTML

When I try to use 1.6.1.1 with ODBC and MySQL, I get these: [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_sip at asterisk: column type (-9) unrecognized for column 'name' [Jun 19 17:19:22] WARNING[5882] res_config_odbc.c: Realtime table supporten_sip at asterisk: column type (-9) unrecognized for column 'ipaddr' [Jun 19 17:19:22] WARNING[5882]

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hi all, From the web, I can find a table scheme of sipusers for ARA using. However, I can't find any meaning of each field, especially for the field regserver which is new in the table. Can any tell me more detail about the usage of each field? CREATE TABLE `sip_buddies` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `host` varchar(31) NOT NULL

Hi everybody, >From asterisk CLI we can add extensions in dial-plan dynamically using "dialplan add extension" command. but how we can dynamically create a context in dialplan. is that possible? Nasir Iqbal

I read on the Wiki that Asterisk Realtime requires CVS HEAD, but I've also discovered that not everything on the Wiki is 100% accurate (that's not a knock, but with a program that is changing as fast as Asterisk, it's impossible for the documentation to keep up). Is it true that Realitme requires CVS HEAD? TIA, Jeff Heath

Hi guys, I'm trying to use Asterisk with LDAP integration. I created some schemas and it seems to work fine for sip.conf replacement. When I try to register a softphone to test the service, it seems ok from the softphone point of view (user registred) but when I do a "sip show peers", no one is registered (nor sip show subrscriptions, users...) I put my Asterisk on full debug and I

Hi Olivior, We have recently worked on a WebRTC based agent panel. As based on my experience I think that WebRTC based phones are far better and cheaper then those soft / sip phone. the big plus is that they are easy to customize and developer can use the power of browser and web to build / offer features which are not possible with regular phones. Regarding your concern about BLF or call

I'm trying to get sipusers working with a realtime odbc database on Asterisk 1.6. We have sippeers working from the database, but need sipusers to be in a separate table for other implementation reasons. sip show user test load returns results from the database. CLI> sip show user test load * Name : test Secret : <Set> MD5Secret : <Not set>

From: Sjur Br?ndeland <sjur.brandeland at stericsson.com> Fix panic in virtio.c when CONFIG_DEBUG_SLAB is set. Use device_del() and put_device() instead of device_unregister(), and access device data before calling put_device(). Signed-off-by: Sjur Br?ndeland <sjur.brandeland at stericsson.com> cc: Guzman Lugo, Fernadndo <fernando.lugo at ti.com> cc: Michael S. Tsirkin <mst

From: Sjur Br?ndeland <sjur.brandeland at stericsson.com> Fix panic in virtio.c when CONFIG_DEBUG_SLAB is set. Use device_del() and put_device() instead of device_unregister(), and access device data before calling put_device(). Signed-off-by: Sjur Br?ndeland <sjur.brandeland at stericsson.com> cc: Guzman Lugo, Fernadndo <fernando.lugo at ti.com> cc: Michael S. Tsirkin <mst

On Sat, Nov 09, 2019 at 03:54:17PM +0800, Chuhong Yuan wrote: > drm_put_dev also calls drm_dev_unregister, so dev will be unregistered > twice. > Replace it with drm_dev_put to fix it. > > Signed-off-by: Chuhong Yuan <hslester96 at gmail.com> Nice catch, I'll apply. Since this is so confusing, we actually have a todo to remove drm_put_dev completely from the codebase

drm_put_dev also calls drm_dev_unregister, so dev will be unregistered twice. Replace it with drm_dev_put to fix it. Signed-off-by: Chuhong Yuan <hslester96 at gmail.com> --- drivers/gpu/drm/virtio/virtgpu_drv.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/virtio/virtgpu_drv.c b/drivers/gpu/drm/virtio/virtgpu_drv.c index 0fc32fa0b3c0..fccc24e21af8

drm_put_dev also calls drm_dev_unregister, so dev will be unregistered twice. Replace it with drm_dev_put to fix it. Signed-off-by: Chuhong Yuan <hslester96 at gmail.com> --- drivers/gpu/drm/virtio/virtgpu_drv.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/virtio/virtgpu_drv.c b/drivers/gpu/drm/virtio/virtgpu_drv.c index 0fc32fa0b3c0..fccc24e21af8

I am trying to have asterisk update the system name in my realtime peers, but it does not seem to be working. Here is what I've done so far. - added systemname => mysystemname in asterisk.conf - set rtsavesysname=yes in sip.conf. - created a table called "sysname" in my peers table in mysql - restarted asterisk - rebooted my phone to force a re-register Is there something

Though asterisk support AMD which is based on silence detection but I did not found support of tone / beep detection in asterisk to record a voice message for answering machines after detecting tone Will appreciate any help in this regard Best Regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT Unified Communication Telemarketing

I understand that Asterisk 1.4 should support T.38 pass-through, but I need Asterisk (or something on the Asterisk box) to act as a T.38 endpoint. Judging from the unclaimed $12,000USD bounty, it doesn't appear that Asterisk itself can do this. http://www.voip-info.org/wiki-Asterisk+T.38+Bounty Does anyone have any experience with this, or are able to point to an example of this working?

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this