Displaying 20 results from an estimated 1000 matches similar to: "controlling SIP subscriptions from SNOM phones"
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 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 Mar 28
1
Redirect problem/bug/feature
I have a major problem with SIP redirects, or maybe just do not understand
how they are supposed to work. We are using Cisco 7940s and 7960s with SIP
version 6.3. Asterisk 1.2.5.
A call come in to extension 944 over the IAX trunk. Extension 944 has
forward all to extension 904 set on the phone. According to the dialplan.
extension 904 should ring for 90 seconds, then ring another extension, and
2006 Apr 25
1
CHANUNAVAIL, busy and congestion
Greetings to all,
I ma having a problem with channel variables on a couple of our Asterisk
boxes.
Here is the setup. Asterisk on customer's site (1.2.5), using IAX to our
external GW (1.2.5), IAX to PSTN GW (1.0.10), E1/PRI to PSTN.
On the External GW, we also have an IAX trunk to a VOIP provider if for some
reason the E1 is down. If the DIALSTATUS is CHANUNAVAIL, which should be
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
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
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
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 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
2004 Sep 03
7
Dropping incompatible voice frame
Hi: i have a problem.
Mi extensions.conf:
exten => _N.,1,Setvar(VOICEMAILREQ=${EXTEN})
exten => _N.,2,SetAccount(${customer})
exten => _N.,3,SetCDRUserField(${VOICEMAILREQ:1})
exten => _N.,4,ResponseTimeout(5)
exten => _N.,5,Background(ifyou)
exten => _N.,6,Background(silence/1)
exten => _N.,7,Background(ifyou)
exten => _N.,8,Background(silence/5)
exten
2006 Mar 15
1
dropping voice frame ulaw - slin?
Mar 15 12:54:01 NOTICE[24269] channel.c: Dropping incompatible voice
frame on Local/[removed number]@context-5c3e,2 of format ulaw since our
native format has changed to slin
Can anyone provide an English translation of what this means?
The extension is a Polycom IP 501
The only allowed formats are g.711u
MOH is MP3 files (obvious)
All prompts have been re-recorded in .ul uLaw
2006 Mar 13
1
Scrolling messages
Several times a day I get this meesage scrolling on one of our asterisk
boxes:
Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping
incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our
native format has changed to alaw
Mar 13 14:43:47 NOTICE[11264]: channel.c:1911 ast_read: Dropping
incompatible voice frame on Local/928@sanset-d88e,2 of format slin since our
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
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:
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 ;
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