similar to: The quest for making "hint" more flexible continues - using Realtime now

Displaying 20 results from an estimated 6000 matches similar to: "The quest for making "hint" more flexible continues - using Realtime now"

2010 Jul 16
1
BLF - Realtime & Asterisk
Hello Asterisk-Community, I'm having an error with my BLF configuration on my asterisk...i've configured the sip peer like this: [8250] type=friend callerid=Extensi?n 8250 <8250> canreinvite=no context=pbx9 dtmfmode=rfc2833 host=dynamic insecure=no language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100
2005 Dec 20
4
Got SUBSCRIBE for extensions without hint
Hi there, I'm getting a bunch of these errors from Polycom phones in 1.2.1: ERROR[24301]: chan_sip.c:10790 handle_request_subscribe: Got SUBSCRIBE for extensions without hint. Please add hint to 4003 in context internal I've searched the Wiki and archives to no avail - what do these errors mean? Regards, -- Anthony Rodgers Business Systems Analyst District of North Vancouver Web:
2012 Jul 26
1
Asterisk Realtime issue after registering with x-lite
Hi All, I have an small issue, which is not creating any problem on working syatem but not sure about the problem that is why eager to know about it. I had installed Asterisk realtime with Asterisk 1.4.41. Every thing is working good but getting warning at Asterisk CLI. [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a valid host [Jul 26 21:17:36] WARNING[17811]:
2006 Dec 17
5
BLF on GXP2000
I am trying to set up the BLF on a GXP2000. Currently what I have is extensions.conf: [globals] polycom430=SIP/101 [internal] exten => 101,1,Macro(voicemail,${polycom430}) [macro-voicemail] exten => s,1,Dial(${ARG1},10,tT) exten => s,2,VoiceMail(u${MACRO_EXTEN}@default ) exten => s,102,VoiceMail(b${MACRO_EXTEN}@default) [ext-local-custom] exten => 101,hint,${polycom430}
2005 Aug 02
1
Polycom Soundpoint 500
I have a Polycom Soundpoint IP 500 that I have been using with Asterisk for a few weeks. It has been working OK, no major problems other than a freeze up every now and then, until today. The power apparently went out last night and for some reason the phone appears to be working but I keep getting the following errors repeating over and over in my Asterisk log file (IP's X'ed out): Aug
2006 Jan 06
2
controlling SIP subscriptions from SNOM phones
We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one phone setup as the receptionist phone, using hints to show busy office lines. This all works as expected. This is a new installation, and people are just starting to setup their phones. For those of you not familiar with SNOM phones, there is a row of keys on the right side of the phone which SNOM calls function keys. In
2007 Jan 20
1
error message
Recently, I got the following error messages in CLI periodically. Jan 20 17:43:18 ERROR[8641]: chan_sip.c:11002 handle_request_subscribe: Got SUBSCRIBE for extension XXXXXXXXXXX@from-int from 192.168.0.123, but there is no hint for that extension I have no idea what the error message tell me. I am sure I haven't that account XXXXXXXXXXX in my database and there is no hint extensions in dial
2010 Jul 14
2
BLF with Realtime
Hello Asterisk community, I'm trying to use BLF with Asterisk Realtime, i've been searching for some info but nothing seems to be clear, can anyone help me eith some ideas to make this work ok? I'va my dialplan with Realtime Thanks in advance -- Saludos Danny Dias SkypeID: danny.dias1
2006 Dec 23
1
CLI Errors and warnings
Hi all, I am getting the following popping up in my asterisk CLI. Everything seems to working ok, but I'm curious as to what exactly these messages mean: >>>> Dec 23 12:28:44 NOTICE[9858]: chan_sip.c:11123 handle_request_subscribe: Got SUBSCRIBE for extension 95555555555@Management from 192.168.1.104, but there is no hint for that extension <<<< Thanks for
2007 Mar 24
2
Can be called on FreeWorldDialup/IAX channel, but can't make calls
Hi, I have an FWD account and it's configured in asterisk. I can be called by people using FWD, but I cannot make FWD calls myself. Every number dialed with a 8 prefix goes to FWD, if for example I call the echo servie I get this: Connected to Asterisk 1.2.13 currently running on asterisk (pid = 2865) Verbosity is at least 35 -- Executing SetCallerID("SIP/timothy-08224f08",
2008 Feb 22
1
Message waiting light on polycom 301 using asterisk 1.4.14
All, I am setting up asterisk on a nslu2 (Linksys) using unslug. Everything is working great except that I have a polycom 301 and I cannot get the message indicator to work. I have created the users and mailbox in users.conf and I can manually dial the mailbox (*986000). Last thing is I am not using config files for the polycom just web browser. Can anyone point me in the right direction I
2015 Sep 13
4
Fail2ban
Hello I'm using the Fail2ban. I configuration below. I want to try to prevent the continuous password. Fail2ban password that does not prevent this form. (Asterisk 1.8 / Elastix interface) What could be the problem ? Asterisk log; "Registration from '<sip:3060 at sip.x.eu;transport=UDP>' failed for 'x.x.x.x:32956' - Wrong password" Fail2ban asterisk
2011 Nov 21
1
video calls not working
Hi list,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [2218] type=friend secret=******* callerid="Virendra" <9172341457> host=dynamic ;
2009 Dec 05
2
How to use SIP hints and BLF for realtime extensions on Aastra phones?
Hi, I need to make use of BLF feature on Aastra 6757i phones but its an Asterisk 1.4 using realtime architecture. Extensions are defined in realtime database and dial plan is in AEL. I am able to correctly setup hints in the dialplan, but they don't work. Did some research and found out that hints don't work work with realtime extensions. Is there any work around? On voip-info I read
2006 Nov 02
3
mpg123 new version
Hi there, It seems mpg123 is now maintained again and there are a few new version released. The latest is 0.61. Does asterisk work with the newer versions? What audio options shall we use when compile mpg123 v0.61? Thanks! Frank
2018 Aug 01
0
ecryptfs (from centosplus) and libvirt lxc quest on it - quest upon start shoots host in the head
hi guys I have a lxc guest on Dell's R815(AMD Opterons) which I just moved to an ecryptfs mounted folder(using centosplus' kernel C7.5) Copying data to that folder seems to work just fine, but I as soon as I start the quest, the host gets shot in the head almost immediately, hard reboot, no dump. I also have another, virtually identical guest but on Intel platform and that one does
2018 Jul 31
0
ecryptfs (from centosplus) and libvirt lxc quest on it - quest upon start shoots host in the head
hi guys I have a lxc guest on Dell's R815(AMD Opterons) which I just moved to an ecryptfs mounted folder(using centosplus' kernel) Copying data to that folder seems to work just fine, but I as soon as I start the quest the host gets shot in the head, hard reboot, no dump. I also have another, virtually identical guest but on Intel platform and that one does not do anything but runs
2004 Dec 10
2
include and hint in extensions.conf with new realtime feature - how?
hi, i'm a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten =>
2010 Jul 15
0
WARNING[15867]: chan_sip.c:15766
Hello Asterisk-Community, I'm having an error with my BLF configuration on my asterisk...i've configured the sip peer like this: [8250] type=friend callerid=Extensi?n 8250 <8250> canreinvite=no context=pbx9 dtmfmode=rfc2833 host=dynamic insecure=no language=es nat=yes pickupgroup= callgroup= qualify=2000 secret=cyx2mo type=friend username=8250 subscribecontext=pbx9 call-limit=100
2015 Jan 08
4
SEMI OFF-TOPIC - Fail2ban
Hi list , someone on the list has seen this type of connection attempts in asterisk, fail2ban does not stop 2015-01-08 14:59:47] SECURITY[21515] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="1420750787-386840",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:100 at