similar to: Asterisk not relaying SIP messgaes

Strange things are happening at my asterisk box :) I've got asterisk setup to dail out with sip to my SIP provider. I've got NO fromuser/fromdomain stuff setup in my sip.conf When I place a call with my Siemens Optipoint 400 SIP phone everything is allright, the From: header is stating the username of the Siemens phone. When I place a call with X-Lite the From: header is altered and reads

Hello, I've got the following queue.conf: [testQ] music=jr_80 ;Bore the caller with some 80's music announce=queue-testQ ;Announcement to play to the Agent answering strategy=ringall ;Let all hell break lose timeout=60 ;We should answer within 60s retry=5 ; announce-frequenty=15 ;Tell them where the are every 15 seconds announce-holdtime=yes ; Give them

List, I've got 2 queues with 10 agents in both queues. One of the agents is mainly responsible for queue_1, and the others mainly for queue_2 so i've defined the following in my queues.conf [queue_1] strategy=ringall member=>Agent/1,2 member=>Agent/2,1 member=>Agent/3,1 member=>Agent/4,1 [queue_2] strategy=ringall member=>Agent/1,1 member=>Agent/2,2

I am curious as to why I quite often receive responses to questions on the help before receiving the actual question. For example Mr Graves has responded to Mr Lumley's response to Mr Wegelin yet I ave not received Mr. Lumley's post while I have received the other two. Cheers, Jim [[alternative HTML version deleted]]

Ah, ok Satz, I did not know that's right :( Now I am trying: <server>194.232.200.156</server> <port>8000</port> <mount>/listen.pls</mount> <local-mount>/oe3.mp3</local-mount> ... but I do not get the mountpoint "oe3.mp3" in http://186.56.96.133:8200 May be something related with the extension "pls" ?. El mi?, 20-02-2013 a

Hi all! I've setup an icecast2 relay server, which connect to an icecast1 server mountpoint. the stats.log of icecast1 lists an/my icecast2 relay connect - so far so fine :) BUT: if i try to listen to the relayed stream, my winamp couldn'd connect: 404 file not found :(( the access.log lines are like: 213.23.134.235 - - [13/Apr/2004:08:05:29 +0200] "GET /rc2 HTTP/1.0" 404

Hi; Has anyone successfully relayed an Icecast stream using Shoutcast? I realize that this is backwards! However, several organizations that want to use our streams use Shoutcast, I really don't want to setup shoutcast and Icecast, but unless I can relay an Icecast stream with Shoutcast, I'm going to have to. Thanks, Fred -------------- next part -------------- An HTML attachment was

Hiya, 1> Master/slave relay should work for you.. something i found in the version i am using you have to define the mount name in the master config file for it to be relayed to the slaves relays. The current relesed version supose to resolve this issue, I still trying to compile it on my linux systems to test it out. 2> For non static IP address you can use a dynamic hosting service

Hello, I would like the dialing party to know what happened to the call, since asterisk doesn't relay a sip error back to the originating sip channel (would be nice, a if (org_channel = sip && dst_channel = sip, relay error to sip client) I want to set up audio feedback on the call status. I've changed the county setting to NL in indications.conf and created this test

HI all, I have the following setup running: EP<--->Calling Asterisk<--->Relaying Asterisk<--->Softswitch<---> PSTN The Endpoint EP is registered with the Calling Asterisk. Calls are forwarded from this machine to Relaying Asterisk which in turn forwards it to the Softswitch. In addition, this machine also relays back responses from the Softswitch to the Calling

I've been thinking more about this. It would be really nice if Icecast could add a port to a mountpoint and emulate whatever information Shoutcast requests when it relays a shoutcast stream. This way Shoutcast could relay Icecast streams. Fred _____ From: Fred Black [mailto:fred@batanga.com] Sent: Monday, July 18, 2005 2:36 PM To: 'icecast' Subject: RE: [Icecast]

Hi all, I've been trying to hit on the right configuration combo to allow relaying from specific users and/or domains to an internal box running Sendmail. Reading the docs at http://www.sendmail.org/m4/anti_spam.html#relay and http://www.sendmail.org/m4/anti_spam.html#access_db_fine I would appear that I should be able to all per-address relaying in /etc/mail/access by enabling

Well, I've hit another problem: Icecast crashes if I define more than 10 listen-socket ports. I have 44 streams that I'm trying to relay via Icecast. It works just fine until I try to add the listen-socket port definitions. If I put the 11th one in, it will crash on startup without entering anything in the log files (log level on 4). I've use sysinternal tcpView to verify that the

I think my sendmail is not correctly set. It says the default is to disallow relaying. I am running centos 4.8 and I think sendmail is relaying for anything. Based on this: http://www.sendmail.org/~ca/email/chk-89f.html#RELAYING I added|: FEATURE(relay_hosts_only) to my sendmail.mc file and restarted sendmail. I am still seeing connects that get relayed and are not in my /etc/mail/access

The relaying-configuration seems to be straight-forward, unfortunately it does not work. That's my configuration: <relays-on-demand>1</relays-on-demand> <relay> <server>http://mp3stream7.apasf.apa.at</server> <port>8000</port> <mount>/</mount> <local-mount>/oe3.mp3</local-mount> <on-demand>1</on-demand>

Hello, I've got a setup with NUT, where we've got two UPSes, ups1 and ups2. UPS1 supports our servers, while UPS2 supports our desktop boxes. We monitor both of them via USB on a box now called "upsmon". The interesting point comes here. We monitor both UPSes on the upsmon box, but if UPS2 fails, which supports our desktop boxes, NUT shouldn't initiate a shutdown in

At 10:06 AM 9/4/02 +0200, you wrote: >Hi, > >Can someone enlighten me, how does icecast2 stream relaying work? Is it >same (pull-style) setup like in icecast 1.x? in the config file, I see >elements like: > > <master-server>127.0.0.1</master-server> > <master-server-port>8001</master-server-port> >

Hello, I'm trying to implement a function available on the PSTN net here, if you dial a number which is busy and you press 5, you will be called back when the busy party hangs up. Figuring out if a SIP user is busy isn't to hard, ${DIALSTATUS} produces a BUSY message, however, how can I implement the call back? IE, I dial to extension 712, but that extension is busy, I dial 5 and

Ok, Found the issue, I needed to specify another listen-socket for the port I specified in the alias section. The alias works now, AND, I successfully relayed it to Shoutcast! <listen-socket> <port>8000</port> </listen-socket> <listen-socket> <port>8010</port> </listen-socket> <paths> .. ..

Hello, If one would like to build a redundant Asterisk setup, would it be possible to exchange the locationdb for the SIP users between then? IE, the following setup: SIP Phones -------------- Asterisk ------------------------ SIP carrier | | ------- Asterisk (standby) ------ Asterisk is used as a PABX in this setup, so the