search for: exted

--- server/extents.c | 49 ++++++++++++++++++------------------------------ 1 file changed, 18 insertions(+), 31 deletions(-) diff --git a/server/extents.c b/server/extents.c index 2d609652..4ab5946c 100644 --- a/server/extents.c +++ b/server/extents.c @@ -42,6 +42,7 @@ #include <assert.h> #include "minmax.h" +#include "vector.h" #include "internal.h"

Hi, We've implemented a 'page-all' function for some of our customers, and we've noticed that on occasion the page-all will cause asterisk to crash (safe_asterisk then restarts it again). The particular customer has about 20 phones, and also has 5 Linksys 932 to monitor the state of these extensions. I'm not sure whether it is the page-all that causes the crash, or the

I was trying to install ferret 0.9.2 on solaris (SunOS 5.8) which does not have a sys/dir.h nix_io.c:5:21: sys/dir.h: No such file or directory make: *** [nix_io.o] Error 1 I couldn''t find an obvious way around this... any suggestions? Thanks, Rich Marisa Cornell Information Technologies Cornell University

Hi all, using latest asterisk-svn I want to reflect an video call incoming via an PRI EuroISDN channel to another outgoing PRI channel, and I want the the outgoing channel to have the exact same bearer capability < Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Unrestricted digital information (8) < Ext: 1 Trans mode/rate:

On 11/09/2019 21:59, Vincent Sherwood wrote: > Here is the full output > > [user at DCSAMBA4A ~]# ?samba-tool domain join MYDOMAIN ?DC > -U"administrator at mydomain.ext" ?--server=DC2016A.mydomain.ext > Password for [administrator at mydomain.ext]: > INFO 2019-09-11 21:55:09,790 pid:20279 > /usr/local/samba/lib64/python3.6/site-packages/samba/join.py #1563: >

hello everyone, I have a trixbox server with an E1 card(not digium).It connects to an AVAYA pbx use E1. It works fine.But when i change the E1 card to digium TE220B,there is a problem. When sip extension A(on trixbox) call PSTN extension B(on avaya),A must wait longtime before B start to ring.From the log I find there are two times call. I don't know why the first request be rejected

Is there a setting somewhere in * to define whether I am receiving callerID or true ANI? Global Crossing claims they are sending ANI but I dont think so. My understanding of ANI is that it is always sent, regardless if callerID is blocked. If I dial *67 and my DID, I get "Presentation: Presentation prohibited of network provided number" and no number. Before I call GC on Monday

Hi all, I have to asterisk-1.0.9BRIstuffed-0.2.0n boxes each equipped with a TE410P. Box A is connected with pri1 to the PSTN. Box B is connected with pri1 (cpe) to the Box A at pri2 (net). Now I want Box B to dial out to the PSTN tunneled thru Box A and have it get all ISDN indications in case of call failure, eg. unallocated destination number etc. But currently Box B always gets only

I just downloaded the cvs version CVS-07/28/03-14:45:19 and now I cannot recieve the the calls from the zaptel interface which is a E100P with pri signaling. That is something with asterisk becouse rolling back to version from 06/23/03 using the new libpri and zaptel fixes the problem. Here is an exept from the config: [macro-stdexten]; ; ; Standard extension macro: ; ${ARG1} - Extension

Hello all, the scenario: Carrier ----S2M------ * -----S2M------Siemens | | SIP Clients and many other features With much help from the list, the PRI links are without alarms and inbound calls are working fine (from both: Carrier and Siemens). But I am not able to dial wether outbound nor to the Siemens PBX. I allways get the message: == Everyone is busy

Hi All, I am trying to install Dovecot connecting to MySQL and all seems to be working until I try to send/receive. I'm getting a "Unknown setting" error for anything that is inside dovecot-sql.conf.ext dovecot-sql.conf.ext is inside /etc/dovecot and is referenced from /etc/dovecot/conf.d/auth-sql.conf.ext The error I get is: auth: Fatal: sql

Hi, When have a incoming call from E1 to a extension FXS, and this extension is busy, the incoming call recive ring tone, and it is wrong. What can I do? Thanks in advance Pedro Here is the trace: asterisk-1*CLI> < Protocol Discriminator: Q.931 (8) len=41 < Call Ref: len= 2 (reference 66/0x42) (Originator) < Message type: SETUP (5) < Sending Complete (len= 4) < Bearer

Hey all, I am running a Asterisk 1.2.9.1 with Sangoma A101 card, newest firmware, configured with a help from Sangoma Tech Support, running fine. It is connected to a PRI circuit split from Cisco MC 3810, which in turn is connected to a Converged T from CTC Communications. While Asterisk works fine and I can call in/out on my BV account, I am only able to dial in through CTC. I have spent

Hello! Is it possible to connect Siemens HiPath 4000 to Asterisk? What equipment required on Siemens side? I mean IP not E1. Sorry for asking here. Siemens-related websites use "salesperson language". There is no technical information.

Yes and No. It's confusing to me which is why I ask. Per my initial email my password source is PAM. It's the userdb I'm concerned about ... which dovecot is using /etc/passwd. So dovecot is getting user information from passwd file; password information from PAM. I need to add extra fields for qouta but can't add them to /etc/passwd so I have to create a passwd with the extra

Hi, I have this setup: E1 PRI PSTN -- Asterisk -- Alcatel PBX - analog phones Can someone tell me what's wrong with this call initiating from an analog phone connected to Alcatel PBX? It dies with NOANSWER but all works if I call other destination numbers. Dialplan is a simple Dial(zap/g1/0984465691) statement. At the end you'll find also zapata.conf.

Hi, I am trying to add a samba DC to an existing Domain that has 2 Win2016 Domain Controllers but is still running at Domain Functional and Forest levels 2008R2. When I run the join command it goes most of the way through before eventually erring out, and backing out everything it had done. The command I used is samba-tool domain join MYDOMAIN DC -U"administrator at mydomain.ext" -d

Hi, I've been watching my * Console and seems to be one call not well terminated or something: For 5 minutes at least my console is reporting this: ectory|general|ext-local|be: -- Playing 'letters/c' (language 'en') directory|general|ext-local|be: -- Playing 'letters/o' (language 'en') directory|general|ext-local|be: -- Playing 'dir-instr'

Hello, Configuration: Asterisk CVS HEAD 20050730 on RH EL3+ DIGIUM TE110P PRI card + euroISDN E1 I am trying to sort out the problem: 1. Provider's switch sends "SETUP"; 2. Asterisk receives "SETUP", rings allocated extension but does not send "Setup acknowledge" (or any other messages) to switch; 3. After 4 seconds of waiting of *'s response switch sends

Hi All After lots of try I was successfull in connecting to PSTN to make and recevice calls , I used AMP for this purpose , now I wanted to try out this Asterisk server answers the call , ask for the extensions and then after the extension entered the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my