similar to: SIP credentials in the dialplan

On Tue, Jul 9, 2019 at 6:05 AM Joshua C. Colp <jcolp at digium.com> wrote: > On Tue, Jul 9, 2019, at 7:00 AM, Dovid Bender wrote: > > Hi, > > > > Looking at http://the-asterisk-book.com/1.6/applikationen-dial.html you > > should be able to dial with SIP credentials in the DP. Is this still > > possible in recent versions of Asterisk either with chan_sip or

Hello, I would like to have "whisper" channel spy (not private whisper) in Asterisk 1.2. I see here: http://www.the-asterisk-book.com/unstable/applikationen-chanspy.html That is only available for Asterisk 1.4. I wonder if there is any way to emulate it in Asterisk 1.2. For example, to "join" two calls: one to use a private whisper and other one to use a normal chanspy.

Hi I am getting the following error when i tried to install rails in windows using command prompt.... Attempting remote installation of ''rails'' Updating Gem source index for: http://gems.rubyforge.org ERROR: While executing gem ... (Gem::RemoteSourceException) Error fetching remote gem cache: getaddrinfo: no address associated with hostname. There are proxy settings in

Darryl Moore <darryl at moores.ca> schrieb: > Ahh. Seen that before! That suggests to me that you don't have your > sip.conf records setup right. > > What's your sip.conf look like? Well, here what I wrote in my sip.conf: register => 00493511111111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register =>

Hi All My first post to the list, so "hello world"! Having searched the list archives and the wiki for an answer to this, I don't think it is possible. However, let me ask nonetheless... Is it possible for a Dovecot proxy's login process (IMAP and POP3) to include the "destuser", i.e. the uid used to authenticate to the backend IMAP/POP3 server, in its logging?

I have configured yum to use a proxy via the yum.conf file by adding: proxy=http://myproxy.com:8080/ What I noticed when running yum check-up date is that some requests are going through the proxy while the system seems to be trying to resolve the domains of other hosts in the repo and trying to establish direct connections to them instead of going through the proxy. Can anyone explain this

I would like to use the getBioC.R script from http://www.bioconductor.org/getBioC.R to install the biobase packages. Unfortunately, url() and download.file() die when trying to talk to my proxy: > getBioC("exprs") connect: Cannot assign requested address unable to connect to 'gproxy1.pfizer.com'. connect: Cannot assign requested address unable to connect to

I would like to use the getBioC.R script from http://www.bioconductor.org/getBioC.R to install the biobase packages. Unfortunately, url() and download.file() die when trying to talk to my proxy: > getBioC("exprs") connect: Cannot assign requested address unable to connect to 'gproxy1.pfizer.com'. connect: Cannot assign requested address unable to connect to

Kevin Larsen <kevin.larsen at pioneerballoon.com> schrieb: > The phone you gave your wife is really old. Are you sure it supports SIP > OPTIONS? Can you make a call in or out to it? If you can, it is more > likely that it just doesn't support that and you can't use a qualify > statement. No, I'm not sure. And no, I can't make any call, right now... At least,

Hello, I'm using 1.6.1-rc4. When sending a userevent, (with "UserEvent(MyEvent);" in extensions.ael), I've got : Event: UserEvent Privilege: user,all UserEvent: MyEvent I can't see any Uniqueid field as mentioned http://www.voip-info.org/wiki/view/Asterisk+cmd+UserEvent or http://www.the-asterisk-book.com/unstable/applikationen-userevent.html Is this Uniqueid mandatory ?

Hello, I've got a bunch outgoing-only SIP trunks "connected" to an asterisk 11 setup. I've read the following doc [1] stating you can pass username/password in a dial string. My goal is to dial from asterisk through one SIP trunk or another without touching my sip.conf file. In other words, I'm planning to use: Dial(technology/user:password at

Why will gems only install rails 1.0 when using a proxy? gem install rails -p http:\\myproxy will only install rails 1.0!!! why is this? it''s very irritating Chris -- Posted via http://www.ruby-forum.com/.

Hi again! About my previous E-Mail... I though about it and I think, that maybe I'm just very stupid... Since I called an INTERNAL number, Asterisk tried to call it. I tried right now to call an EXTERNAL number (using my context [myproxy]) and the behavior is NOT the same... Not 100% correct, but it tries the right way... Now my problem is to check in my dialplan if the peer, that

Hi list! So, new day, new problem... I tried right now to call from my cellphone a peer in my Asterisk. The cellphone has correct credentials, but it's NOT registered on my Asterisk, now. I just tried to call a peer in my network, from a peer not yet registered. And it works... :( The very curious thing is, that I can't find how the call will be accepted... Every section in my dialplan

Hi All, I have been trying to solve the issue of Xen-based Virtual Machines (as I described earlier); I have the problem of ethernet ports locked to a single Mac address in my lab--hence VM''s can''t communicate using bridging other than their own dom0. To further test/confirm, yesterday I took my laptop at home and connected to my Home-Hub (which issues DHCP based IPs); I

How can I get wine from git through a proxy server? I tried the usual clone command but substituting http:// for git://, set $http_proxy (per git man page) to the name of my proxy server and got: $ git clone http://source.winehq.org/git/wine.git wine Initialized empty Git repository in /cygdrive/d/wine/.git/ Cannot get remote repository information. Perhaps git-update-server-info needs to be run

I have a group of users configured as extensions in *.These users are registered with a SIP Proxy Server and can receive calls very well. The problem happens when any user tries to make an outbound call. The proxy replies with a "401 Unauthorized" and * don't try another INVITE including credentials. Here is part of the content of sip.conf. [general] port = 5061 bindaddr = *.IP

Hi all, Any one know how to increase the Linux limit? I am hiting a wall on 200 calls playing files at the same time. From Asterisk console, I am getting messages like Sip_request_call: Unable to build sip pvt data for "asterisk1/700" Too many open files Is this a limit of my Linux box? I only have 512MB of ram. Will increase it to 2G help or I have to change some configuration in

I think your phone may be trying to register with the username '1234', while your sip configuration is expecting 'luca'. Can you try changing your phone registration credentials to use 'luca'? Can you give us a sip transcript when you try to place a call from it? On 15-05-28 05:09 PM, Luca Bertoncello wrote: > Darryl Moore <darryl at moores.ca> schrieb: >

Semi Off Topic My searching hasn't found what I consider superior info, and we are wondering from others experience on this list... In the logwatch results we all see the info below on almost a daily basis I have taken the liberty of combining logwatch results from centos 4 and 5 machines for extra info and future searchability ----- Centos 4 ----- --------------------- httpd Begin