Displaying 20 results from an estimated 6000 matches similar to: "CLI Errors and warnings"
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:
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
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]:
2007 Jan 16
2
Polycom IP601 - some hints working, not others?
Are all of the sip phones in the same context?
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Robert Jenkins
> Sent: Tuesday, January 16, 2007 1:44 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: [asterisk-users] Polycom IP601 - some hints working,
2006 Dec 31
8
(OT) Where to post free source for AGI?
Hey all,
After figuring out a problem with AGI and freepascal, I have finished
writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a
small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt.
I'd like to give it to the community (source/binary) and was wondering
where to post it?
The wiki?
Also, anyone have suggestion on licensing? LGPL? FreeBSD?
Thanks
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}
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:
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
2005 May 16
3
CLI and DNIS presented to Analog extension
My company is based in Australia and we have a need to be able to
present CLI (ANI) and DNIS to an analog extension. Currently our PABX
vendor is saying they can only deliver CLI between the first and second
cadence.
The system will collect calls via an aggregate of ISDN PRI services from
the PSTN and then direct them to the analog extensions in a hunt group
configuration. It is important that
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 ;
2012 Nov 18
1
How to MessageSend to a SIP from AMI Or CLI?
Hello all,
I am running Asterisk 10.10.0 and I can send Message between SIP's no
problem. However, I would like to be able to send send Message to a
SIP from AMI Or CLI. I check the ListCommands On the AMI and it
don't have MessageSend. Therefore, I try the SendText.
AMI:
Action: SendText"
Channel: SIP/600"
Body: This is a test.
Message: This is a test.
Extension: 600";
2006 Nov 21
1
Hairping calls and Originating CLI
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2007 May 21
3
Aastra MWI
I need to setup MWI on a few Aastra 9112's. I've tried doing so in the
web interface by setting "Explicit MWI Subscription" to true, but no
lights, no stutter tone.
Firmware: 1.4.0.1048
Thanks!
--
Warm Regards,
Lee
2007 Jan 16
2
force ulaw passthrough if call from modem extension?
I have Teliax trunk set to ulaw and g729 and I have a modem/fax extension
from a sipura forced to ulaw. When the call goes out through Teliax IAX
trunk, asterisk transcodes to g729. Is there a way to tell asterisk not to
transcode calls from/to a specific extension?
I'm running asterisk 1.2.4 and that extension is for my home alarm/dish
network and fax calls.
Thanks
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2006 Dec 30
4
WIFI SIP- The Best phone
Hello Everyone,
I can see that a few people are interested in SIP WIFI phones. I have
tested several Linksys 300,and it is OK. More of a toy then a business
tool. It a poor built in ear speaker, which makes all calls sound tinny,
and the unit is known to hang. I have two Linksys 300's that are fun to
play with however, I wont hand them out to users.
HOWEVER- The Zultys WIP 2 is an