Displaying 20 results from an estimated 1000 matches similar to: "MWI Assistance"
2015 Apr 08
0
Asterisk Inbound calls, multiple SIP accounts, calledID
Solved it, kinda. It's not cute. I'm sure this is the way NOT to do it but
it does work. For prosperity, the SIP service is through Internode.
Here is my "extensions.conf" file:
exten => s,1,Set(thedid="${SIP_HEADER(TO)}"); ignore this one
exten => s,2,Set(pseudodid=${SIP_HEADER(To)})
exten => s,3,Set(pseudodid=${CUT(pseudodid,@,1)})
exten =>
2008 Jul 25
2
Strange checkpassword issue
I'm helping a friend setup a small mailserver using dovecot, and I'm
finding a strange problem with checkpasswd that I haven't had on my
servers.
How is the following debug output even possible?
Jul 25 12:12:20 company2 dovecot: auth(default): master out: USER 5 joe home=/var/mail/joe.com/joe/Maildir/ uid=1005 gid=1005
Jul 25 12:12:20 company2 dovecot:
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
2015 Apr 08
0
Asterisk Inbound calls, multiple SIP accounts, calledID
Hi, Andrew.
You are trying to solve two tasks: definition through what line the call
came and a beautiful display of this information.
1. definition through what line the call came. If the username and
password for inbound and outbound registration the same, then try the
following:
a) delete "register" lines.
b) add option "callbackextension=Company1" to Company1 friend
2010 Nov 09
0
Asterisk Voicemail Realtime and 'VirtualBoxing'
Hello
I'm about to set up a voicemail system for multiple wholesale customers.
So I use a realtime mysql config for the mailboxes.
All single mailboxes have their information about the number, emailaddress,
password in the database. This works fine.
Now the notification emails of course should be customized per wholesale
customer.
I added a 'mandate' table to the database and
2015 Apr 07
5
Asterisk Inbound calls, multiple SIP accounts, calledID
Hi Dmitriy and others and thanks for your help so far.
The option "match_auth_username=yes" seems to have had no effect. From my
reading, this option will try to match the username of the incoming SIP
account to a section heading. If that is how it must work then i can see a
big problem. I'm trying to present the receptionist with a nice display of
which line the call came in on.
2004 Nov 28
0
Fwd: Re: very newbie question
> On Sat, 27 Nov 2004 19:37:54 +0000, Corvin <corvin.dun@wp.pl> wrote:
> > I have very simple question, how to limit SIP phone user making
> > calls to for example longdistant calls?
>
> This is how I do it -
Thank you very much to all of you.
I have one more question which troubles me.
We have scenario:
(only SIP is considered now)
Subscriber A registered in Asterisk
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]:
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:
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 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
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
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
2007 Aug 09
1
The quest for making "hint" more flexible continues - using Realtime now
Ok, now that I've learned I cannot use any variables when using the `hint`
priority (for BLF), I figured I'd try to use the next best thing: hardcoded
values using realtime. This way I avoid variables such as ${ACCOUNTCODE}
but I can at least change the DB more easily than text files. This is the
appropriate line in the DB:
2006 Mar 21
2
Voice mail not working with Asteriks 1.2.5
Hi,
I have upgraded my PBX to Asterisk 1.2.5 , previously I was using
Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not
working. The error I am receiving in log files is like following,
WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12'
I have searched for solution a lot can Any one of you let me know how can I
solve this issue
2011 Nov 21
1
video calls not working
Hi list,*
*I am not able to make video calls between two sip accounts. below is the
information. please help me where I am missing the configuration.*
Extensions.conf*
exten => 111,1,Answer()
same => n,Dial(SIP/2206,60,r)
same => n,Hangup()
*SIP.conf*
[2218]
type=friend
secret=*******
callerid="Virendra" <9172341457>
host=dynamic ;
2006 Dec 17
5
BLF on GXP2000
I am trying to set up the BLF on a GXP2000.
Currently what I have is
extensions.conf:
[globals]
polycom430=SIP/101
[internal]
exten => 101,1,Macro(voicemail,${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}
2004 Apr 03
7
Few question on HTB
Dear All,
Sorry to trouble again..... After go through www.lartc.org I have implemented the HTB instead of CBQ
for the same scenario.
Now following files are under /etc/sysconfig/htb directory.
eth0 DEFAULT=30 R2Q=10
eth0-2.root RATE=256kbps BURST=25k
eth0-2:10.comp1 RATE=120kbps BURST=12k PRIO=0 LEAF=sfq RULE=192.168.200.0/24
eth0-2:20.comp2