Displaying 20 results from an estimated 4000 matches similar to: "calleridname from chan_sip (mysql_sipfriends)"
2006 Jun 13
1
calleridname.agi patch to only overwrite name if it is missing
I edited the calleridname.agi patch to only overwrite the name if it is missing.
The asteridex option still overwrites the name since it is our master list for known numbers.
--
Steven
calleridname.agi.patch:
--- C:\Documents and Settings\steveb\Desktop\calleridname.agi-orig Tue Jun 13 14:37:09 2006
+++ C:\Documents and Settings\steveb\Desktop\calleridname.agi Tue Jun 13 14:37:09 2006
@@ -16,6
2006 Jan 18
2
CALLERIDNAME/CALLERIDNUM Deprecation
Previously, when I wanted to forward to incoming callerid when I
forwarded a call to another number I had to set the callerid on the
outgoing call to be that of the incoming number. So today I do this:
exten => s,n,Set(CALLERID(name)=${CALLERIDNAME})
because I want the outgoing callerid that I forward to not be the normal
callerid of the local extension but I want to forward the incoming
2003 Dec 29
0
FW: Weirdness with CALLERID / CALLERIDNAME from incoming PRI
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-
> admin@lists.digium.com] On Behalf Of Adams, Gavin
Is there any additional information I could provide to start tracking
this down? I was thinking about looking into the various applications
source to see how they access the data elements for callerid. I know
where the values are pulled
2005 Jan 27
2
Q: Can I over-ride the value of ${CALLERIDNAME} ?
Folks,
I'd like to change the value of ${CALLERIDNAME} for incoming PSTN
calls from certain numbers, but haven't found a way that works. The goal is
to provide more informative names on my phones' caller ID displays--e.g., I
would prefer to display "ROB CELL" instead of "CELLULAR CALL" when I call
home from my cell phone.
This is what I tried in the context
2003 Dec 24
2
Weirdness with CALLERID / CALLERIDNAME from incoming PRI
Hey all,
We've upgraded our PRI trunk to support what BellSouth calls "enhanced
caller id name delivery". The weird part is, I'm only capable of seeing
the name portion on incoming calls within voicemail2's email delivery.
For example, on an incoming call, asterisk is reporting this:
Context from extensions.conf (BS delivers 7-digit DIDs):
exten => 9133727,1,Answer
2005 Jan 03
3
oh323 context for peers
I am experimenting with oh323 channels and h.323 gateways and a Cisco
CallManager. I am not using a gatekeeper at this time. Is it possible to
place calls coming into Asterisk from specific peers into specific
contexts?
In iax.conf eaxh peer has a context in which I can specify the context an
inbound call will be placed in. I don't see anything like this in the
oh323.conf file or the oh323
2005 Jan 13
1
asterisk realtime msql
Hi there
asterisk goes to 90% cpu usage when trying to authenticate a sip friend using realtime mysql, no other message does appear at cli and asterisk hungs;
here some info:
*CLI> realtime load sipfriends name 104
Jan 13 11:52:21 DEBUG[8928]: res_config_mysql.c:109 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sipfriends WHERE name = '104'
Jan 13 11:52:21 DEBUG[8928]:
2010 Jan 27
1
Asterisk Database Configuration
Hello
I need to add sip extensions from my UI so without going through sip.conf so
i created table
CREATE TABLE `sipfriends` (
`name` varchar(40) NOT NULL default '',
`username` varchar(40) default '',
`secret` varchar(40) NOT NULL default '',
`context` varchar(40) NOT NULL default '',
`ipaddr` varchar(20) NOT NULL default '',
`port`
2005 Jan 01
1
Problems to use asterisk with mysql /odbc
hi, i.m. newbie in asterisk. asterisk 1.0.3 is my current version.
i like to store usernames and passwords in a sql database.
i like to log failed authentification-passwords, to create a blacklist for
securityreasons.
i thingk a sql-database is a good way to log these actions.
i don.t find debugging-options to output invalid login-passwords.
Ok, i have made the following:
debian is my OS.
2007 Dec 10
0
diferents events between ast1.2 & ast1.4 ??
Hi all,
I'm new in the list, and I have a problem upgrading from asterisk 1.2 to
asterisk 1.4:
There is a diference from asterisk1.2 to asterisk1.4 in AMI events.
When I do a call to a queue (with the same extensions.conf dial plan)
with ast1.2 and ast1.4, in ast1.2 apper 3 newcallerid event in ast1.4
apper only 2.
It is normal? anyone knows it? what is the reason?
I
2004 Dec 16
0
Making "sip show channels" show sane results with sipfriends from mysql?
hi
using sipfriends from mysql from asterisk 1.0 branch, how can I make
asterisk show the true channel's current codec with SIP SHOW CHANNELS?
This does not seem to work, and bkw_ said sipfriends from mysql didn't
have that info at all. For what it may seem, asterisk uses G.726 as
told, giving me a
-- Format for call is g726
at the start of the call, but in SIP SHOW CHANNELS all these
2010 Jan 28
0
Database Configration
Hello
I need to add sip extensions from my UI so without going through sip.conf so
i created table
CREATE TABLE `sipfriends` (
`name` varchar(40) NOT NULL default '',
`username` varchar(40) default '',
`secret` varchar(40) NOT NULL default '',
`context` varchar(40) NOT NULL default '',
`ipaddr` varchar(20) NOT NULL default '',
`port`
2007 Mar 30
1
call file vs. originate
I'm having trouble getting the manager interface to behave properly;
specifically the Originate event.
If I create an originate event as below, the calling phone will
auto-answer (as it's supposed to) but the receiving phone never rings.
It will timeout at 20 seconds.
Action: Originate
Channel: Local/201@from-sip2
Context: from-sip
Extension: 154
Priority: 1
CallerID: John Doe
2005 Jan 14
2
Realtime / sip.conf
I am currently in the process of testing out realtime support for
sip.conf. I have followed all of the directions that are listed in
the Wiki, but for some reason this does not work.
When utilizing a flat file, I am able to register endpoints without
any problems, and calls can proceed. One interesting side effect that
I have noticed is that when I am using realtime for sip, I am unable
to see
2009 Feb 04
0
Problems with 9133i config
I am unable to get my 9133i to register with my asterisk server. I am
including config files below, this a simple test network so there's nothing
secret in the config files. I have upgraded the phone to the latest software
version (1.4.3) I'm not sure what the problem is. I can call the phone from
a softphone, but the 9133i says "no service" on the screen and I can't dial
2007 Mar 30
2
Replacing slot of S4 class in method of S4 class?
Dear all,
Assume that I have an S4 class "MyClass" with a slot "myname", which
is initialized to: myname="" in method("initialize"):
myclass <- new("MyClass", myname="")
Assume that class "MyClass" has a method "mymethod":
"mymethod.MyClass" <-
function(object, myname=character(0), ...) {
2007 Jan 08
0
SIP rt load from db
Anyone know the command that tells * to load a sipfriend
from the realtime db rather than saying no such host? I've tried various
combinations of the rt commands:
rtcachefriends=yes;
;rtcache=yes
;rtAutoClear=yes
;rtautoreg=yes
;rtIgnoreRegExpire=yes
;rtupdate=yes
rtfromcontact=yes
Basically I have a group of 4 * servers all routing calls, but only two
are hearing the phones
2005 Feb 16
2
Sip Notify PAP2-NA?
I am using mysql sipfriends and can't seem to get the MWI to work. From what
I've read it seems this is not supported with that dynamic system, and
probably never will be.
I was thinking of just setting a cron job or something to check every minute
for voicemail and set our sip NOTIFY messages as needed.
Also, the PAP2-NA has the ability to reboot via a sip notify and I would
like to be
2004 Apr 21
2
help with smbmount and permissions
ok, on my windows machine, i see:
myname on 'computer01\home' (H:)
so i went to my linux box and did:
smbclient -L //computer01 -U myname
and i see Home listed as a sharename (why does windows show it as "home" but its really "Home" as reported by smbclient?).
well everything works fine if i mount it like this:
smbmount //computer01/Home /mnt/computer01/Home -o
2003 Dec 11
0
getting Samba 3.0.1 to use NIS UID's/GID's instead of its own
There are two NT domains, the resource domain (pretend it's RESDOM) and the user domain (MASDOM) where RESDOM trusts MASDOM. My Samber server (COOL) is registered in the RESDOM domain. I can do smbclient -L COOL -U MASDOM/myname and it asks for a password and displays the shares correctly, even picking up the NIS logon share.
The problems start, however, when I want to connect to the myname