similar to: asterisk.org blocked - rejecting connections

CVS HEAD/Asterisk 1.2: Is there a way to have the entire extensions.conffile coming from the realtime? It appears that RealTime for the extensions.conf file is on a context by context basis, but you have to create each new context in the extensions.conf file then add a "switch => Realtime" line (then reload). I want to be able to add phones without having to edit any files.

Dear all, Has anyone seen this before and can suggest a solution? I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. Could

Hello, I'm still looking for any ideas on this problem: I've got 3 sip clients behind the router, and they all register with Asterisk using the same IP address. Now, wenn all are registered, all the calls get routed to the client that registered most recently, but not to the correct client. Also, there seems to be some problems registering all 3 clients simultaneously. I have

I apologize for the double post. I am curious as to what the usefullness is of the multiple line appearance feature on Polycom phones. I setup our phones to register one line per extension but I hear the IP501's can do three line appearances. Why and how could this feature be applied? Thanks again all. Kenny ______________________________________________________ Click here to

How do I set each extension to play it's own voicemail prompts? I have vm working in that it plays the standard "person at extension 1234 is not available....." and takes the message. I've recorded seperate .gsm files for each user but can not figure out how to use them. - Gary Edison Information Technologies www.EdisonInfo.com P.O. Box 554

Greetings, I am trying to implement distinctive ringing on a Cisco 7960. I have tried setting alert_info to chirp1 or chirp2 before dialing the phone, but it has no affect. If you have successfully implemented distinctive ringing on a 7960, I would really appreciate seeing the snipit of code that works. Thanks in advance Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775)

Are there any standards for setting up pbx dialplans for businesses/offices? What I mean is that, which numbers are reserved for a specific use ex. 0 for operator ? Putting Zero for operator in the dialplan seems to be the common practice of businesses. If there is such a standard, * and # are used for what ? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all! I would like to solve some problems: I have a sip provider that lets me make pstn calls after listening some stuff and entering a pin number: 1) How can I make Asterisk enter the pin number? Then wait 1 second and enter the phone number? I have in extensions.conf: exten => 6*,1,Dial,SIP/2002@myprovider,60,tr I have tried with w (like with ZAP channels) but it does not work, nor

Hi, I'm working with this issue for a while, Now I already solve the dialplan issues, but I still have a question about the Callgroups, I read at www.voip-info.org that , there is a 63 limit of callgroups. And I'm wondering why?? and if the 1.2.0beta version supported more than 63 Groups?? (I did'nt find any Changelog for 1.2) or If not There is any unoficial patch for that ? Thanks

Following on from my below email, things have taken another bizarre twist.. I have continued getting the error when 2092 tries to listen to messages by dialing 9999. --Executing VoiceMailMain ("SIP/2092-6918", "2092") in new stack --Playing 'vm-password' (language 'en') WARNING: app_voicemail.c:4922 vm_authentication: Unable to read password. Then I

Hi! I'm trying to use dongle in my Asterisk But appear for me all time this error [Feb 12 13:49:10] ERROR[13347]: chan_dongle.c:442 do_monitor_phone: [dongle0] timedout while waiting 'OK' in response to 'AT' -- [dongle0] Error initializing Dongle -- [dongle0] Dongle has disconnected -- [dongle0] Trying to connect on /dev/ttyUSB1... -- [dongle0] Dongle has

Yes, I used IMEI. But in CLI appearing nothing and it not register. 2016-02-12 14:27 GMT-02:00, Shabbir abbasi <shabbirabbasi92 at gmail.com>: > have you tried imei discovery > imei=123456789012345 > > > write imei number instaed of 12345... > > On Fri, Feb 12, 2016 at 8:51 PM, Vitor Mazuco <vitor.mazuco at gmail.com> > wrote: > >> Hi! >>

Yes I used. The problem can be the version of Asterisk? I use Asterisk 13 instead of 11. 2016-02-12 14:31 GMT-02:00, Shabbir abbasi <shabbirabbasi92 at gmail.com>: > have changed this > [dongle0] > audio=/dev/ttyUSB1 > data=/dev/ttyUSB2 > > To > > [dongle0] > imei=123456789012345 > > and imei exact same as on your device ? > > On Fri, Feb 12, 2016

From: Sylvain Gault <sylvain.gault at gmail.com> The command-line parsing used to return a pointer to a local array. The code used to work by chance, but now, gcc 5 is able to detect it and return a NULL pointer instead. The buffer is now marked static. This shouldn't be a problem as only one command line can be read at a time. Signed-off-by: Sylvain Gault <sylvain.gault at

Dear Srs, I'm getting this error when trying to connect to a Infortrend A16E- G2130-4 storage v?a iSCSI. > Apr 29 10:24:40 vz-10 kernel: scsi1 : iSCSI Initiator over TCP/IP > Apr 29 10:24:41 vz-10 kernel: Vendor: IFT Model: A16E- > G2130-4 Rev: 361F > Apr 29 10:24:41 vz-10 kernel: Type: Direct- > Access ANSI SCSI revision: 04 > Apr

Hi. I'm running the 9.1-RC1 release. At the end of the boot sequence, I see: mps0: mpssas_scsiio_timeout checking sc 0xffffff8000759000 cm 0xffffff8000784380 (probe19:mps0:0:19:0): TEST UNIT READY. CDB: 0 0 0 0 0 0 length 0 SMID 240 command timeout cm 0xffffff8000784380 ccb 0xfffffe0006918000 mps0: mpssas_alloc_tm freezing simq mps0: timedout cm 0xffffff8000784380 allocated tm

Within a function I'd often like to obtain a text string equal to the name of the function. One use for this: To generate a filename for use in pdf(). This enables me to keep track of which function generated a particular graphic came. match.call() puts parentheses at the end of the name. I don't want parentheses in a filename. The following kludgey function gives the desired result.

Hi, I'm having trouble installing the LDAP support for Samba passwords. I confess I am no LDAP expert... but.... With the latest version of samba 3.0beta I install the ldap.conf entries on on restart get this... Shutting down ldap-server done Starting ldap-server/etc/openldap/schema/samba.schema: line 318:

I'm getting this error when compiling:- make[1]: Entering directory `/usr/src/asterisk.svn/sounds' --09:22:12-- http://ftp.digium.com/pub/telephony/sounds/releases/asterisk-core-sounds-en-wav-1.4.0.tar.gz => `asterisk-core-sounds-en-wav-1.4.0.tar.gz' Resolving ftp.digium.com... 216.27.40.102, 69.16.138.164 Connecting to ftp.digium.com|216.27.40.102|:80... connected. HTTP

Am 27.09.18 um 22:52 schrieb Bjorn Helgaas: > [+cc LKML] > > On Tue, Sep 18, 2018 at 04:32:44PM -0500, Bjorn Helgaas wrote: >> On Thu, Sep 13, 2018 at 11:37:45AM +0800, Daniel Drake wrote: >>> On 38+ Intel-based Asus products, the nvidia GPU becomes unusable >>> after S3 suspend/resume. The affected products include multiple >>> generations of nvidia GPUs