Displaying 20 results from an estimated 4000 matches similar to: "Dialing SIP server user extension... Dial string issue..."
2003 Oct 08
1
BudgeTone 102 flakey sound
I have experienced lots of apparently dropped packets (in other words,
lots of short interruptions of what the other party tries to tell me)
with a GS102 and chan_capi. The GS102 is connected through a lightly-loaded
switch directly connected to the * server, so bandwidth/latency
shouldn't pose a problem. Funny thing is that the switch indicates
10mbit on the GS102 port - is that correct?
2015 Apr 13
1
dial out with channel variable; sub-string usage
On 15-04-09 12:06 PM, Chad Wallace wrote:
>> but don't know where to put those lines. I have BABY defined as
>> >channel variable:
>> >
>> >BABY = SIP/babytel_out
>> >
>> >but that seems circular, somehow.
> You put them in the context for your clients... From what you show
> below, I'd say they go in the "local_200"
2003 Dec 04
4
Channelbank Recomendation and GS102 question
Hi All.
I'm working on an * configuration. We require 8 inbound POTS lines, and
CT1 or PRI seems like it will be
quite expensive at that level. I've read that a T1 Channelbank plus
the T100P would be a (the?) way to go
for this situation. What is the recommended channelbank for use in this
scenario? From searching the archives
I see a lot of suggestions to get "a
2015 Feb 20
2
sipsak 200 for a user, but 404 for a different user...why?
What's the difference between user "123" and "devries"? Based on the
output here, they seem the same..?
tleilax*CLI>
tleilax*CLI> sip show users
Username Secret Accountcode
Def.Context ACL Forcerport
201 password 201
default No Yes
123
2015 Feb 16
3
LAN sip-to-sip
I'm reading the O'Reilly "Asterisk the definitive guide", 4th ed, with a
starfish on it. In some ways, astonishing that it's not really that
definitive, it's more general -- and it only clocks in at one ream of
paper!
In any event, I'm having some port problems on my home network:
http://security.stackexchange.com/questions/81752/
I need to open ports for
2005 Mar 04
1
Zap channels intermittently bridging with SNOM190
Hi guys/girls,
We are running a TDM04B card with Asterisk in a Linux box that has 15 GS102 extensions and 1 SNOM190 phone which we are using as an operator console. The FXO ports in the TDM04B are plugged directly into our telecoms provider's analogue lines.
Something I've picked up with the SNOM is that sometimes when there are two active incoming calls via Zap channels and the first
2015 Apr 08
2
dial out with channel variable; sub-string usage
I want to do something like:
exten => _NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN})
exten => _Nxxxxxx,1,Dial(${BABY}/${EXTEN})
exten => _1NXXNxxxxxx,1,Dial(${BABY}/${EXTEN})
exten => _011.,1,Dial(Dial({TOLL}/${EXTEN})
exten => _9NXXXNxxxxxx,1,Dial(${BABY}/${EXTEN})
exten => _9Nxxxxxx,1,Dial(${BABY}/${EXTEN})
exten => _91NXXNxxxxxx,1,Dial(${BABY}/${EXTEN})
exten =>
2015 Feb 19
0
sipsak: 404 error
Hi,
I **think** that I have user of thufir101, because I get a 200 response
below, but I also get a 404. It seems to depend on how I send the ip
address/fqdn?
tleilax*CLI>
tleilax*CLI> sip show users
Username Secret Accountcode
Def.Context ACL Forcerport
201 password 201
default No Yes
2006 Apr 12
1
SIP call hangup from asterisk CLI
Hi,
We are using Vicidial and sometime even when agent disconnects, outgoing
call originated by dialer is still active. Since call was initiated by
dialer and then bought into meetme conference of agent and we can't corelate
this call to any agent channel.
When agents are dialing, channels doesn't show calls
vicidial2*CLI> show channels
Channel Location
2015 Feb 20
0
sipsak 200 for a user, but 404 for a different user...why?
On 2/20/15 6:15 AM, thufir wrote:
> What's the difference between user "123" and "devries"? Based on the
> output here, they seem the same..?
>
> tleilax*CLI>
> tleilax*CLI> sip show users
> Username Secret Accountcode
> Def.Context ACL Forcerport
> 201 password 201
> default
2009 Jan 28
5
Inbound Call Disconnect in 3 seconds
My Inbound calls lands , but line get disconnect in exactly 3 secs.
Here goes my extension.conf setting :
[from-ipkall]
exten => 901835,1,Ringing ; call ringing
exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI
exten => 901835,3,Answer ; Answer the line
exten =>
2015 Feb 16
0
LAN sip-to-sip
It looks as if that is more of a question/issue with your router, rather than Asterisk.
I have SIP devices working on my LAN, all hardwired, and have no need to open any ports or have the router address SIP in any way
My switch is not managed, and the router ports on the LAN side are all unmanaged, just a huge Ethernet "wirenut"
You SHOULD be able to communicate between devices on the
2020 Apr 16
4
Crash after Update to 4.12.1 with vfs full_audit
Hello alAl,
after update of our test server to 4.12.1 from 4.11 it crashes. If the
vfs module is removed from the config everthing works as before. Logs
from the crash see here:
.0.31:445]
Apr 16 13:36:47 lx-sv-03 smbd_audit[6263]: [2020/04/16 13:36:47.546559,
0] ../../source3/lib/util.c:830(smb_panic_s3)
Apr 16 13:36:47 lx-sv-03 smbd_audit[6263]: PANIC (pid 6263):
vfs_full_audit.c: name table
2010 Sep 14
4
If then else with command for
Hey listers,
I am trying to do something simple... Check the program below...
I would like to create a variable named COLOR according to the conditions
that I stablished... But the problem is that it seems that my variable COLOR
is checking just on sample, may be last in the loop... Certainly, I am
missing something...
Thanks in advance,
Marcio
2015 Feb 16
1
SIP show peers: UNREACHABLE
I'm trying to configure SIP trunking. Now, I'm referencing "Asterisk
the definitive guide", 4th ed. While I don't have the page handy, I was
reading the suggestion to try SIP to SIP before proceeding to outside
connectivity. I'm aware that SIP trunking is a construct, but am,
obviously, learning the system.
What I'd like to do is from the CLI "ping"
2015 Mar 23
0
trying to connect to asterisk with softphone (logs, etc)
In the Asterisk log I see:
---
[Mar 23 19:25:29] VERBOSE[4067] chan_sip.c: [Mar 23 19:25:29]
<--- SIP read from UDP:198.38.7.34:5065 --->
SIP/2.0 200 OK
To: <sip:16046289850 at sip.babytel.ca>;tag=sd3D4swKRc
From: <sip:16046289850 at sip.babytel.ca>;tag=as07c833c5
Via: SIP/2.0/UDP 96.48.217.39:5060;branch=z9hG4bK13c68eb7;rport
Call-ID:
2009 Apr 15
4
[Release Planning 3.4] 3.4.0pre1 will be delayed
Hey folks,
the release of Samba 3.4.0pre1 will be delayed until April 30, 2009 due to
the samr access check bugs and bug #6263 (Domain login problems in Windows
XP without SP3).
@Developers: There is still some space left to place your changes in the
release notes.
Karolin
--
Samba http://www.samba.org
SerNet http://www.sernet.de
sambaXP http://www.sambaxp.org
-------------- next part
2003 Sep 26
3
dialing out with the outgoing queue problem.
Hi,
I have cvs updated all my modules (zapata, libpri, zaptel and asterisk).
I have also read in the archives & seems that no-one has run into this
problem.
What I'm trying to do is simple. Just make and outbound call using the
/var/spool/asterisk/outgoing directory.
I copied /usr/src/asterisk/sample.call and only changed the context &
extension.
I configured my Zap1 to the same
2008 Feb 10
0
CentOS-announce Digest, Vol 36, Issue 4
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When
2009 Apr 17
1
[Announce] Samba 3.2.11 Maintenance Release Available
================================================================
"You can''t have everything.
Where would you put it?
Steven Wright
================================================================
Release Announcements
=====================
This is a maintenance release of the Samba 3.2 series.
Major enhancements in 3.2.11 include:
o Fix domain logins for WinXP