similar to: Context restrictions

On Mon, Aug 12, 2024 at 11:23:44AM +0200, Daniel Vetter wrote: > On Mon, Aug 12, 2024 at 10:28:22AM +0200, Thomas Zimmermann wrote: > > Amdgpu and nouveau call vga_switcheroo_process_delayed_switch() from > > their lastclose callbacks. Call it from drm_lastclose(), so that the > > driver functions can finally be removed. Only PCI devices with enabled > > switcheroo do

Hi Am 12.08.24 um 12:18 schrieb Daniel Vetter: > On Mon, Aug 12, 2024 at 11:23:44AM +0200, Daniel Vetter wrote: >> On Mon, Aug 12, 2024 at 10:28:22AM +0200, Thomas Zimmermann wrote: >>> Amdgpu and nouveau call vga_switcheroo_process_delayed_switch() from >>> their lastclose callbacks. Call it from drm_lastclose(), so that the >>> driver functions can finally be

I want to include a new context in my exensions.conf I have read this page http://www.voip-info.org/wiki-Asterisk+howto+dial+plan and I can sort of follow it?! I have a context [local] that I know zapata.conf points to, I have edited extensions.conf and put in my phone, sip and iax extensions. I want to add an sms context. I understand that all calls go through my [local] context and I have

On Mon, Aug 12, 2024 at 12:41:39PM +0200, Thomas Zimmermann wrote: > Hi > > Am 12.08.24 um 12:18 schrieb Daniel Vetter: > > On Mon, Aug 12, 2024 at 11:23:44AM +0200, Daniel Vetter wrote: > > > On Mon, Aug 12, 2024 at 10:28:22AM +0200, Thomas Zimmermann wrote: > > > > Amdgpu and nouveau call vga_switcheroo_process_delayed_switch() from > > > > their

Hi all I have installed AAH 2.2 in my P4 PC following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp and made as per the guide says and downloaded SJ Phone, and registered user and when i try to dial the 19197543700 i get message that, all circuits are busy now, please try your call later and when i see in the console i get this mesage any help Called easycall/19197543700

Hi, Few months back I configured Asterisk 11.6.0 for an outbound fax using T.38 protocol as listing down the flow below; Asterisk Fax server -> (IP) -> Cisco VGW ->(IP) -> Carrier The issue I'm currently getting when Asterisk receives warnings as listed below, it is overloading the Cisco VGW, therefore need to restart Asterisk service or sometimes reboot VGW to clear these

I constantly get the following error messages in /var/log/asterisk/messages: Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 3324 (iax_ack_registry): Received unsolicited registry ack from '192.168.0.1' Dec 8 10:52:57 WARNING[1009521664]: File chan_iax.c, Line 4181 (socket_read): Registration failure Where 192.168.0.1 is another asterisk server. Below are the local and

I have an interesting dilemma with sequencing in the dialplan. Up to now, I have assumed that the extensions in the dial plan were tested in the order that they appear in extensions.conf. In other words, I have the following fragment which was designed to dial toll free on the PSTN and all other long distance on VoIP: [longdistance] include => local

Has anyone seen this before? This channel is consistently present on both of my asterisk servers. Sometimes they disappear for a few seconds and then come back. It always has the same Call ID. voip1*CLI> sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Format 192.168.0.102 (None) df92fb1b-8a 00101/03059 00000ms 0000ms UNKN

I'm trying to create several contexts for extentions with different levels of access to features and I'm wondering how the heck do I include all the contexts so that you can call internal to any extention in another context without giving the features of the higher level context to the lower level context? ie..... [admin] include => local include => longdistance include =>

I appologize for the long, new-ish question, but after a few days of trying to work a solution by reading through the list archives and WIKI and coming up with what I thought would work, I think I'm just not getting a fine detail. I titled this thread "Setting up Security Groups" because I'm trying to set up some sip user groups with certain calling rights, e.g., one group of

Here is my Configuration PSTN -> Cisco AS5350 -> SIP -> ASTERISK -> SIP -> ATA186 When I call from the pstn to the ATA, the ATA rings but I don't hear anything on the calling side until the call is picked up. When I call from the ATA, everything seems to work fine. When I bypassed ASTERISK, everything seems to work fine. Anyone know what I might have configured wrong?

Upgraded from 16.2.14 to 1.6.2.15 on Fedora 13, with aastra 9133i and 57i. On 9133i and 57i: #<extension># works for a blind transfer. Xfer<extension>Xfer doesn't! All this worked on 1.6.2.14. Nothing useful on cli, verbose 3, DEBUG. Here extension 169 answers an outside call, and tries to transfer it to 145 using the Xfer button: -- SIP/169-0000009c answered

I am seeing strange behavior that I don't understand. Voicemail2 and voicemailmain2 work fine if I call from a sip phone directly connected to *, but if I call either of them from an analog line on the other side of a sip gateway, voicemail seems to ignore digits. If I am recording a message and press #, nothing happens except that it records the tone onto the message and I can't specify

Still trying to upgrade to 1.6.0.9 for 1.4. It worked - it worked all day yesterday, but this morning: -- Executing [646xxxyyyy at longdistance:1] Answer("SIP/172-08276a60", "") in new stack .......... -- Executing [646xxxyyy at longdistance:6] Dial("SIP/172-08276a60", ""DAHDI/g2"/1646xxxyyyy") in new stack May 27 09:56:57]

Howdy, My first forray into using res_mysql.conf for realtime access of sip users and extensions. I have the following relevant section of extensions.conf: --- [trunklocal] exten => _NXXXXXX,1,Dial(${GLOBAL(TRUNK)}/${EXTEN:${GLOBAL(TRUNKMSD)}}) [local] include => trunklocal include => trunktollfree [longdistance] include => local include => trunkld [international] include

All, Thank you in advance for any help. I have a customer in need of a conferencing system. A requirement is for users to each have their own PIN for the same bridge. So, I put the list of users, PINs bridges into a MYSQL DB and used an ODBC connector to parse the table. Asterisk is connected and reads the rows as expected. The problem is that if a user enters a PIN that is NOT in the table,

HI, I dont understand the syntax of the dial application when used like this: Dial(Local/${EXTEN}@longdistance) i want to know what is this "Local" doing instead of Tech like SIP, IAX, H323? -- Regards Rizwan Hisham Software Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL:

On Mon, Aug 12, 2024 at 10:28:22AM +0200, Thomas Zimmermann wrote: > Amdgpu and nouveau call vga_switcheroo_process_delayed_switch() from > their lastclose callbacks. Call it from drm_lastclose(), so that the > driver functions can finally be removed. Only PCI devices with enabled > switcheroo do the delayed switching. The call has no effect on other > hardware. > > v2: >

I'm trying to use the FollowMe app with Asterisk 1.4.17. I've followed the WIKI page on setting it up but I can't seem to get it to work. Here is my Asterisk version: pbx1*CLI> core show version Asterisk 1.4.17 built by root @ pbx1 on a i686 running Linux on 2008-01-10 12:08:48 UTC Here is a log of when the FollowMe is being called: NOTE: I've tried to use the AstDB as