similar to: Again Sip Registration Fail

I'm using asterisk with XLite everything is working. But in the asterisk console I always receive some notice of Registration failed . What is the reason for this? How Can be fixed? message : Jul 1 16:18:29 NOTICE[65541]: chan_sip.c:6731 handle_request: Registration from 'damian <sip:damian@10.1.1.11>' failed for '10.1.1.11' Asterisk and Sip phones are all in one

I have * box on machine with external ip address and internal one I'm tring to register to it from to machines - one from innternet ( everything is ok - in sip.conf nat=yes)\ and the other one is in the internal network (in sip.conf - nat=no ) and it say 403 Forbidden? Any Ideas ? here are the logs and configs From the external SIP Client whic registers.

I'm trying to call from XLite phone to PSTN (I've tried this from internet and from local network the same) The Xlite doesn't write that it is connected but receives excelent audio. At the other end comes only noise. Some times only for a second you can here the caller voice , but this was only one time :) I saw with ethereal that UDP packets are coming and going to the asterisk

Is it possible to call a macro, which is defined in extensions.conf from a realtime extension configured in Mysql. Beacuse when i try i receive an error - no such context. -- Executing Macro("SIP/1007-2165", "dialnumber_wvm,1004,SIP/1004") Dec 11 12:51:04 WARNING[22551]: app_macro.c:100 macro_exec: No such context 'macro-dialnumber_wvm,1004,SIP/1004' for macro

I've just installed asterisk on a Debian Linux (apt-get it) And i have placed two sip phones in sip.conf and i'm testing parking with them I have phone1-SIP/1000 and phone2-SIP/1007 The following happens if i park from calling party and everything is OK 1. Pickup Phone2 and call to Phone1 2. Talk 3. Phone2 dials #700 and parks the call (it is placed in 701) 4. Phone2 is hangup 5. Pickup

11-10-2004 there was a subject: Re: Where did USE_SIP_MYSQL_FRIENDS go?: on asterisk.user list. >All db specific code has been removed from the code in favor of the >currently-in-development "RealTime" method of configuration from >database. >You are most likely not using the 1.0 stable branch. >You need to use the new RealTime configuration method. And currently,

No Caller ID comes from the FXO line ( The caller id is on and is working with a standard phone) in zapata.conf everything looks fine usecallerid=yes hidecallerid=no When the call comes in there are some warnings in Asterisk Console -- Starting simple switch on 'Zap/4-1' Jun 22 11:20:24 NOTICE[213006]: callerid.c:281 callerid_feed: Unknown IE 17 Jun 22 11:20:24 NOTICE[213006]:

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten,8006,Sip/8006) [macro-stdexten] ; ; Standard extension macro: ; ${ARG1} -

When I have an ip hardphone username setup in my sip.conf : [109] type=friend username=ipphone9 secret=bla-la host=dynamic dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info defaultip=172.20.0.139 mailbox=109 ; Mailbox for message waiting indicator callerid=ipphone9 <109> callgroup=1 pickupgroup=1 and this user has a wrong password then calls are denied, but

How can i control the codec for the calls. For example I have 3 SIP phones registered to asterisk The firs two are in the local area network (behind nat)- I want to use g711 between them and to connect directly (canreinvite=yes) and the third is in internet - want all calls to it and from it to use g729 and media to go through asterisk. So if Phone 1 calls Phone 2 the codec to be g711, but when

While on conversation with another party, I dial the atxfer key sequence. Asterisk says "Transfer" then gives you a dial tone, while put the other party on hold music. I dial the transferee number and talk with the transferee, then I hang up and the other party must be connected with the transferee. But this doesn't work the transferee hears a beep. -- Playing 'beep'

Hi folks, I want to fix hole in my asterisk set up. I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN, Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go 'other' places. This senario works fine. Now the issue is someone else running a vocal or another SIP proxy can redirect his calls to my * as well. Those calls two will come through general

I'm tring to compile fax applications on Debian system. the spandsp library compiles ok, and when i try to patch the make file in apps directory as is said in the instructions it returns errors. I'm using cvs version of asterisk . -------------------------- voipgw:/usr/src/asterisk/apps# patch < Makefile.patch patching file Makefile Hunk #1 FAILED at 35. Hunk #2 FAILED at 68. 2 out of

Hello, Does anyone know of a "slick" way to get R to convert a string which is all upper case to a string where the first letter in each word is upper case and all others are lower case? I suspect the solution is to begin by parsing the string, convert the appropriate letters to upper and lower case using "toupper" and "tolower", and then to paste the pieces back

We've just set up an internal domain, and had a single XP Pro machine join without any difficulties. But we need this machine (and the other handful with it) to use Roaming Profiles, which is posing something of a speed issue. I have logged on, killed the 'You do not own the profile' tidbit, fixed up permissions, and finally gotten XP to read the profile from the SAMBA server

On 20 Jul 2020, at 1:12, Damian wrote: > I solved the issue by adding my own m3u files to > /usr/share/icecast/web but I am not sure if this is icecast > best-practice. This also raises a few other issues as well… http > stream links don’t appear on the icecast stream directory and stream > titles and track metadata does not display when opening m3u files > created in this

Thanks for the response. I can confirm that the recommended settings worked and now IPV6 is working as it should. If anyone is interested, these are the <listen-socket> settings I am using. <!-- You may have multiple <listen-socket> elements?> <listen-socket> <port>8000</port> <bind-address>0.0.0.0</bind-address> </listen-socket>

On 17 Jan 2023, at 23:16, Damian wrote: > Thanks for this. I?m on a Debian server so it?s looking good. > Yes, then there should be no issues whatsoever using IPv6 with Icecast 2.4 >> On 18 Jan 2023, at 07:59, Marvin Scholz <epirat07 at gmail.com> wrote: >> >> Hi, >> >> On 17 Jan 2023, at 22:51, Damian wrote: >> >>> Hi, >>>

Good morning, On Sun, 2020-10-11 at 10:24 +1000, Damian wrote: > Can I also ask, which is the recommended course of action? Sure, just keep in mind that we had weekend, so people were off. ;) > Should I rebuild icecast with OpenSSL or is there now a fixed version > via backports that I can use? Based on... On Sat, 2020-10-10 at 23:06 +1000, Damian wrote: > Edit (apologies for the

Thanks for this. I?m on a Debian server so it?s looking good. > On 18 Jan 2023, at 07:59, Marvin Scholz <epirat07 at gmail.com> wrote: > > Hi, > > On 17 Jan 2023, at 22:51, Damian wrote: > >> Hi, >> >> I would like to know whether Icecast version 2.4.4 supports IPV6 listener connections? I have some mobile phone users that connect to my web server