similar to: Asterisk V10, SIP MESSAGE fails for unknown reason, missing DNS-lookup?

Easy questions for you guys probably, I'd like to serve 10 parallell incoming calls at the same time, so I bought a lot of Zap-channel cards for analog phone lines. But I want all users to be able to use the same phone number to dial in, but I want the number to be switched to an avaiable zap-channel. Do I need some kind of switch for this? It sounds reasonable, but I'm not sure. :) Am

Hi, I'm calling asterisk with a swedish PSTN-phone line with caller presentation (DTMF) activated. I'm using asterisk 1.4.20.1 and cannot upgrade unfortunately, so I have to stay with this release. I use a TDM800P 8 channel PSTN card working as answering phones (I connect a phoneline with carrier signal to my TDM-card). Using zaptel-1.4.12.1. I verified that the DTMF tones of the number

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd

I'm trying to play a wav-file on a channel. This is what I see in the asterisk debug console AGI Rx << STREAM FILE "test.wav" "12345" [Aug 20 16:10:19] WARNING[25219]: file.c:602 ast_openstream_full: File test.wav does not exist in any format So it doesn't find the file, or it's in a wrong format? I can listen to it with windows media player... it's a

Hi there! Trying to originate and dial a number using Zap-8, used to work, but now it just fails. I enabled all debug I found in the source-code and this is the output from asterisk. Can someone understand something from the debug-output what is wrong and direct me to what the problem might be? The setup is correct, trust me, it worked some hours ago, haven't changed anything. Just dialing

Hi We are working on a speaker phone system using PJSIP and Speex Speech processing API on an ARM platform. Currently we have spent about a month on getting the AEC to work properly and we have worked through the most common causes of problems (such as clock drift, synchronization problems and non-linearity's in echo path). Now we achieve ERLE of about 17 dB which tells me that the AEC is

hi list, i recently recognized, that winbind on my fileserver (needed for allocating SID->UIDs when setting ACL's from windows box) adds ldap attributes although the SID already exists !!??!?! example i have a user "install" # install, users, eva.mpg.de dn: uid=install,ou=users,dc=eva,dc=mpg,dc=de objectClass: posixAccount objectClass: person objectClass: sambaSamAccount cn:

On Mon, Sep 21, 2015 at 9:45 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Mon, 21 Sep 2015 06:48:52 +0000 > Emil Ohlsson <emo at svep.se> wrote: >> [sip-im] >> exten _X!, 1, NoOp(Got message) >> exten _X!, n, Answer() >> exten _X!, n, Agi(agi://localhost/messagehandler.agi?...) >> exten _X!, n, SendText(Message received) > > I am not

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

Hi, I am trying to minimize a rather complex function of 5 parameters with gafit and nlm. Besides some problems with both optimization algorithms (with respect to consistantly generating similar results), I tried to run this optimization about a hundred times for yet two other parameters. Unfortunately, as the log below shows, during that batch process R starts to eat up all my RAM,

Hello List! My Asterisk stopped making SIP-calls today, I could call from external, and saw Call coming in over PRI, but calling the SIP/Device wont work. I saw 5 open channels - all chan_spy. Only a restart helped. In the messages-file i found from yesterday: [Mar 4 17:28:01] NOTICE[25769] app_chanspy.c: Attaching SIP/209-0000170fto SIP/210-0000170e [Mar 4 17:29:38] NOTICE[25790]

http://bugs.freedesktop.org/show_bug.cgi?id=25824 Summary: KMS, Dualhead: Strange bigger console Product: xorg Version: unspecified Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org

Sorry for the delay here. For some reason the mail from Joshua Colp failed to deliver to my mailbox. So, anyway, I've set up a local scenario on my computer a PJSIP client and Asterisk 11.17.1 (On a fedora linux workstation) with the settings listed below. In this scenario I've used UDP, but I want a configuration that can be used with any transport protocol. I can see that the context

Hi, I'm having trouble configuring Asterisk to respond to an incoming out of call SIP MESSAGE. The transport protocol is TLS and the Asterisk version is 10 (it's old, but I'm kind of stuck with it at the moment). Currently I have roughly the following configuration and handling: sip.conf: [general] accet_outofcall_messages=yes outofcall_message_context=sip-im and extensions.conf

Those are just late DNS replies--port 53 is DNS, and the IP you gave points to a DNS server (ns1.gci.net). "dig -x" is your friend :) The connection tracking table used by iptables to masquerade your internal network will only "hold open" a UDP connection for a certain amount of time; if no traffic flows in either direction, the entry in the connection tracking table will be

Hi I have a question about the AEC and preprocessor. I have seen that when near end is talking for a longer time (about 10 s) without being interrupted by the far end the residual echo (or rather leak_estimate) increases making the preprocessor suppress the near end talk when it shouldn't. Is there a way to make the leakage estimate only update when far-end is present or similar in order to

On 15-09-28 10:19 AM, Emil Ohlsson wrote: > (Still no not receiving the mail, revisited the settings.) > > OK, so SendText doesn't work with this scenario. But can MessageSend > handle this, and respond even when the transport protocol is TLS? Or > do I need to modify Asterisk to add this support? MessageSend has no concept of TLS, it gets passed to chan_sip which then sends

Hi all, I'm preparing a package which uses .Fortran to interface a Fortran 95 function. This F95 function simply receives the name of a file from R, opens this file and forwards its content to a F95 module, which, in turn, makes the real computation. The F95 module is a pre-existing one and I'm trying to use it in its actual state. Thus, data transfer between R and this F95 module is

The Wine development release 1.7.17 is now available. What's new in this release (see below for details): - More implementations for the Task Scheduler. - C runtime made more compatible by sharing source files. - Fixes in the Mac OS X joystick support. - Various bug fixes. The source is available from the following locations:

hello all... i am trying to install r v2.0.1 on my suse 9.2 pro box... when i run configure, i get the following error: checking how to get verbose linking output from g77... configure: WARNING: compilation failed checking for Fortran libraries of g77... checking for dummy main to link with Fortran libraries... none checking for Fortran name-mangling scheme... configure: error: