Displaying 20 results from an estimated 113 matches for "restrictcid".
2006 Jun 08
6
revisit to legacy PBX and CID over PRI
...ut asterisk thinks that it is still ringing.
I have added "hidecallerid=yes" to zapata.conf and no longer have the problem.
But, now I have Legacy PBX users complaining about having no caller ID.
I tried this, but it still would not complete the call.
hidecallerid=no ;fix for no answer
restrictcid=yes
usecallingpres=no
I have also tried to make the callerid name null, but asterisk still tries to send the data.
I have also tried to dumb it down from NI2 to NI1, but asterisk still tries to send the callerID name in the PRI debug.
Is there a way to send callerid number and not the name?
ref....
2005 Feb 11
2
Question about DID
...would be great!!
Here is my configs
cat zaptel.conf
#PRI to Telco
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
# PRI to Fax server
span=2,0,0,esf,b8zs
bchan=25-47
dchan=48
zapata.conf
[channels]
context=from-analog
signalling=pri_cpe
switchtype=dms100
group=1
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
useincomingcalleridonzaptransfer=yes
callerid=asreceived
faxdetect=no
musiconhold=default
channel => 1-23
context=from-sip-internal
switchtype=dms100
signalling=pri_net
group=2
overlapdial=yes
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
useincomingcall...
2006 Dec 12
3
outgoing call on ISDN PRI
HEllo list !
When user A calls user B via Asterisk (Users A and B are registered on
the same Asterisk server ) and an ISDN PRI, user B phone
always shows Asterisk server telephone number. How to hide it and how
to forward user A number ?
We tried usecallerid, callerid, hidecallerid, restrictcid,
usecallingpres in zapata.conf but we always see Asterisk server
telephone number !
Thanks you!
2005 Jan 26
1
mySQL-sipfriend dials to another SIP-endpoint - How to set the from-user
Hi,
I have some mySQL-sipfriends and connectivity to PSTN.
When a call from PSTN comes, it shows a callerid,
and that callerid is displayed at the called sip phone.
When the call comes from another sip user (defined as
mySQL-sipfriend), no callerid is displayed at the called
sip phone.
I turned on sip debug and discovered, that in the last case
in the SIP-header to the called phone:
From:
2004 Jul 07
0
IP Dialog Hangup problem
...t; s,103,Hangup()
IpDialog SipTone II
[600]
type=friend
secret=blah
username=600
context=default
canreinvite=no
host=dynamic
dtmfmode=inband
;Choices are inband, rfc2833, or info
defaultip=XXX.XXX.XXX.XXX
callerid = "Blah" <600>
mailbox=600 ; Mailbox for message waiting indicator
restrictcid=no ; To have the callerid restriced -> sent as ANI
insecure=yes ; To match a peer based by IP address
only and not peer
insecure=very
Mediatrix 1102
[603]
type=friend
secret=Blah
username=603
context=default
host=dynamic
canreinvite= no
qualify=200
dtmfmode=inband
defaultip=XXX.XXX.XXX.XXX...
2011 Jan 24
2
Outgoing FXO calls have no audio with callprogress=no
...eduration=40
;usedistinctiveringdetection=yes
answeronpolarityswitch=yes
usecallerid=yes
cidsignalling=bell
cidstart=ring
;hidecallerid=yes
;hidecalleridname=yes
;waitfordialtone=yes
;mwimonitor=no
;mwilevel=512
;mwimonitornotify=/usr/local/bin/dahdinotify.sh
;mwisendtype=rpas,lrev
callwaiting=yes
;restrictcid=no
usecallingpres=yes
sendcalleridafter = 1
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
;echotraining=no
group=1
callgroup=1
pickupgroup=1
;immediate=yes
immediate=no
callerid = asreceived
useincomin...
2004 Dec 14
3
sip_buddies mysql table
...; canreinvite
this looks to be 3 chars not 1 (Null, no, yes)
>>> context
See name above
>>> incominglimit/outgoinglimit
these have been depreciated and probably should be removed unless there is
come Realtime coding that requires these fields to be present.
>>> restrictcid
currently 1 char long, should be 3 chars for values (Null, no, yes)
>>> pickupgroup
Since callgroup was set to 30 I just set this value to 30 chars since
the example shows the same number of characters.
CREATE TABLE `sip_buddies` (
`uniqueid` int(11) NOT NULL auto_increment,
`...
2007 Jan 26
2
Hello Everybody, my problem with voicemail.conf
....conf , extensions.conf and voicemail.conf
Pl,do the needful.
Thanks.
With Warm Regards,
Ashish Barot.
-------------------------------------
*sip.conf*
[general]
bindport=5060
context=worldbiz
mailbox=_111X@worldbiz
nat=yes
allow=all
[1111]
type=friend
context=worldbiz
secret=1234
host=dynamic
restrictcid=no
canreinvite=no
mailbox=1111@worldbiz
[1112]
type=friend
context=worldbiz
secret=1234
host=dynamic
restrictcid=no
canreinvite=no
mailbox=1112@worldbiz
[1113]
type=friend
context=worldbiz
secret=1234
host=dynamic
restrictcid=no
canreinvite=no
mailbox=1113@worldbiz
-------------------------
*ext...
2005 Aug 19
4
Overriding Caller ID
...pets of some of our configs:
extensions.conf
;
; Local calls
;
exten => _NXXNXXXXXX,1,CallingPres(32)
exten => _NXXNXXXXXX,2,SetCallerID(2125551234)
exten => _NXXNXXXXXX,3,Dial(${TRUNK_LO}/${EXTEN})
zapata.conf
[channels]
usecallerid=yes
cidsignalling=bell
cidstart=ring
hidecallerid=no
restrictcid=no
usecallingpres=yes
callerid=asreceived
switchtype = dms100
signalling = em_w
group = 1
context=inbound
callerid=asreceived
channel => 1-24
Does anyone have any suggestions?
Thanks,
Waldo
2004 Dec 14
3
Problems with app_realtime
...deny | varchar(95) | YES | | NULL | |
| pickupgroup | varchar(10) | YES | | NULL | |
| port | varchar(5) | | | | |
| qualify | varchar(4) | YES | | NULL | |
| restrictcid | char(1) | YES | | NULL | |
| rtptimeout | char(3) | YES | | NULL | |
| rtpholdtimeout | char(3) | YES | | NULL | |
| secret | varchar(30) | YES | | NULL | |
| type...
2007 Aug 28
1
E911 mf camma Trunks
...e even using the
e911, and if so how do you have it set up? I'll post my relevant configs
below:
ZAPTEL.conf
=======================
#Sangoma A102 port 2 [slot:11 bus:1 span: 2]
span=2,0,0,esf,b8zs
e&m=25-48
=======================
ZAPATA.conf
=======================
;911 group
group = 2
restrictcid=yes
signalling = e911
channel => 25-26
=======================
EXTENSIONS.conf
=======================
[globals]
DIGG1=Zap/g1
DIGG2=Zap/g2
; 911
exten => 911,1,NoOp(911 ANI CALLER ID INFO IS: ${CALLERID(ani)} REGULAR
CALLER ID: ${CALLERID(all)})
;exten => 911,2,Dial(${DIGG2}/${EXTEN})
ex...
2003 Nov 07
0
Possible fix for grandstream outgoing
...02:17:47 2003
+++ chan_sip.c Fri Nov 7 02:16:23 2003
@@ -3928,8 +3928,8 @@ static int check_user(struct sip_pvt *p,
p->callgroup = user->callgroup;
p->pickupgroup = user->pickupgroup;
p->restrictcid = user->restrictcid;
- p->capability = user->capability;
- p->jointcapability = user->capability;
+ /*p->capability = user->capability;
+ p->jointcapabilit...
2006 May 13
0
Spam? Re: Cisco 7970 problems
I don't see that anywhere. Here is my zapata.conf This is only happing
on my 7970 all other phone are working without trouble.
[channels]
context=pri
signalling=pri_cpe
switchtype=dms100
group=1
usecallerid=yes
hidecallerid=no
restrictcid=no
usecallingpres=no
useincomingcalleridonzaptransfer=yes
callerid=asreceived
faxdetect=incoming
musiconhold=default
echocancel=yes
echocancelwhenbridged=yes
channel => 1-23
context=Fax
switchtype=national
signalling=pri_net
group=2
overlapdial=yes
usecallerid=yes
hidecallerid=no
restrictcid=no...
2004 Jun 17
1
Zap dropping calls
.../24/04-17:37:48 on kernel
2.4.25-gentoo-r3. I have a Digium TDM-400P card with 4 FXO ports. Here
are the pertinent files:
zaptel.conf:
fxsks=1-4
loadzone = us
defaultzone=us
zapata.conf:
[channels]
context=north_in_pots_vip
group=1
signalling=fxs_ks
usecallerid=no
hidecallerid=no
callwaiting=no
restrictcid=no
threewaycalling=no
echocancel=1
echocancelwhenbridged=no
echotraining=1
rxgain=10.0
txgain=2.0
immediate=no
musiconhold=default
jitterbuffers=4
relaxdtmf=yes
busydetect=yes
callprogress=yes
callerid => 1234567
channel => 1-2
extensions.conf:
[north_in_pots_vip]
exten => s,1,Answer
exte...
2010 May 10
1
Dialing a SIP Peer without using register strin
...i m posting here. please help
me out
currently I am working on dialing a sip peer on an asterisk server from 2nd
asterisk server. scenario is like this.
on my system i am using this peer in sip.conf.
[abc]
type=peer
username=abc
secret=mysecret
host=192.168.0.20
context=default
dtmfmode=rfc2833
;restrictcid=no
canreinvite=yes
insecure=invite,port
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=gsm
nat=yes
qualify=yes
and using following register string
register => abc:mysecret at 192.168.0.20 <abc%3Amysecret at 192.168.0.20>
now problem is that when i use register string everything g...
2005 Jul 21
1
account code missing in csv cdr
...on the list and the solution
seems to be setting up a dialplan for incoming calls from a particular
sip peer.. in my opinion this does not scale well at all and I am
looking for a solution to correct this problem.
example sip peer:
[asterisk_gw]
type=friend
accountcode=asteriskgw
host=x.x.x.x
restrictcid=no
fromdomain=voip.livewirenet.com
context=lwn
canreinvite=yes
Incoming calls from the peer lack an accountcode field.. a
NoOp(${ACCOUNTCODE}) is blank for calls coming from the above peer..
forcing an accountcode does indeed work with SetAccount() however, as I
stated, this does not scale.
I s...
2004 Jul 06
2
Mediatrix 1102 Problems
...le times and have done multiple cvs updates. Any help with this
would be most appreciated.
[602]
type=friend
secret=blah
username=602
context=default
host=dynamic
canreinvite= no
qualify=200
dtmfmode=inband
defaultip=XXX.XXX.XXX.XXX
callerid="MediaTrix Port 1" <602>
mailbox=602
;restrictcid=yes
;insecure=yes
;insecure=very
[603]
type=friend
secret=blah
username=603
context=default
host=dynamic
canreinvite= no
qualify=200
dtmfmode=inband
defaultip=XXX.XXX.XXX.XXX
callerid="MediaTrix Port 2" <603>
mailbox=603
;restrictcid=yes
;insecure=yes
;insecure=very
JP McInnis, D...
2008 Jan 17
1
Device state of SIP doesn't change
...1168>
canreinvite: no
context: default-sip
defaultip: NULL
dtmfmode: rfc2833
fromuser: NULL
fromdomain: NULL
fullcontact: NULL
host: dynamic
insecure: NULL
language: NULL
mailbox: 21168 at device
md5secret: NULL
nat: yes
deny: NULL
permit: NULL
mask: NULL
pickupgroup: NULL
port: 5061
qualify: no
restrictcid: NULL
rtptimeout: NULL
rtpholdtimeout: NULL
secret: xxx
type: friend
username: 21168
disallow:
allow: all
musiconhold: NULL
regseconds: 1200593168
ipaddr: xxx.xxx.xxx.xxx
regexten:
cancallforward: yes
setvar:
Any help would be appreciated.
Regards,
Atis
--
Atis Lezdins
VoIP Developer,
IQ Lab...
2005 Feb 03
1
403 Forbidden when registering sip user database on backend
...-------+-----------+
| uniqueid | name | accountcode | amaflags | callgroup
| callerid | canreinvite | context | defaultip |
dtmfmode | fromuser | fromdomain | host |
incominglimit | outgoinglimit | insecure | language |
mailbox | md5secret | nat | permit | deny |
pickupgroup | port | qualify | restrictcid |
rtptimeout | rtpholdtimeout | secret | type |
username | allow | disallow | regseconds | ipaddr |
auth |
+----------+------+-------------+----------+-----------+----------+-------------+---------+-----------+----------+----------+------------+---------+---------------+---------------+-------...
2004 May 23
5
PRI problem???
...zs
# span 2 is for NuVox PRI (1-23B, 24D)
span=2,1,2,esf,b8zs
# span 3 is unused
span=3,0,0,esf,b8zs
# span 4 is unused
span=4,0,0,esf,b8zs
#
#
zapata.conf
[channels]
;
; Default language
;
language=en
;
; Default context
;
context=dialnational
;
signalling=fxo_ks
use_callerid=yes
callwaiting=no
restrictcid=no
usecallingpres=yes
callwaitingcallerid=no
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=no
echotraining=yes
relaxdtmf=no
rxgain=0.0
txgain=0.0
callgroup=1
pickupgroup=1
immediate=no
amaflags=billing
<channels 1-24 omitted>
;
; NuV...