Displaying 20 results from an estimated 7000 matches similar to: "CallerID Auth"
2006 Dec 04
2
ASterisk and SER
HI,
My Asterisk is registed with my SER. My client are connected to asterisk
when they dial any no like 62222 asterisk passes this is ser and then again
ser passes this no 2222 (strip 1) back to my asterisk. but insted of ringing
this exten it says loop detected. can some one tell me what is wrong.
thanks
arun
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2007 Jun 04
2
G729 License
HI
I bought 20 license from Digium and install in my server and b'coz of some
problem I've to change my server is it possible that I can use those lice
and register again in my new server ?
Is it possible that I'll be able to use those lice in my old box also ?
thanks
arun
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2007 May 13
2
TC400B load problem
Hi
Im trying to install my TC400B trans coder card when I do:
modprobe wctc4xxp
tail -f /var/log/messages says:
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Encoder' with
92 transcoders (srcs=0000000c, dsts=00000101)
May 13 14:56:36 pbx2 kernel: Registered codec translator 'DTE Decoder' with
92 transcoders (srcs=00000101, dsts=0000000c)
May 13 14:56:36 pbx2
2007 Apr 24
2
Call Connection Problem
Hi,
I'm running a php script to generate calls using Asterisk Manager and its
working fine. this script call a specified land line number if the phone is
answered then It will connect to an extension and play an IVR. But I see in
Asterisk CLI its placing the call and it shows channel answered but I don't
receive call on my land line and it starts playing the IVR. Please guide me
how to
2007 Apr 19
1
Asterisk Queue Call Transfer
Hi
I've configured the queue on my asterisk box and everything is working fine.
In my queue I've 3 agents logged in the queue. When call comes they are able
to receive the calls without any problem. But some time they are on break
and there extension rings and no one is there to answer the call (we don't
want them to log off from the queue) but we have one normal user in the same
2007 Apr 08
1
Adding Noise or background noise
Hi,
In my dial plan I've configured two trunks to make outbound calls (trunk1
and trunk2) to same service provider but I want when any of my exten starts
with _2. should goto trunk2 and there should be some kind of disturbance
(like some noise or some background noise) when my calls goes to trunk2 to
make the call quality bad. Mainly I want to achieve bad call quality on
trunk2 by adding
2007 Apr 17
2
No of Calls
Hi
sorry for asking the same question again:
here is my details:
I've 50 exten in my sip and I'm using snom300 to my asterisk box this
asterisk box is connected to another asterisk box using IAX trunk over 1MB
full duplex line. I'm using g729 as the preffered codec. Can you please tell
me how many calls can go at the same time without causing the any type of
problem.
thanks
arun
2009 May 20
3
Asterisk CCM, CME Integration
Hi All,
I'm just posting this questions to both forums as its related to both. In
hope to get some help on below issue:
Asterisk 1.4.x
CCM = 4.x
CME = 4.x
codec = g711ulaw
Here is my setup:
600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME
-----> 461X Phones
461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X
Phones
so in
2007 Jun 06
2
iax trunking on OpenBSD
Hi,
do I have a chance to use iax trunking on OpenBSD where there is no zaptel
driver or ztdummy available? Do I can use sth. else as timing source?
kind regards
Sebastian
2007 Jun 27
1
Help with IAX Trunk
Hi
I've two servers :
1. UK
2. Pakistan
Pakistan * server has ISDN30.
Pakistan(ISDN30) <====> UK ===> User
Im planning to setup an IAX2 trunk between these two server ?
so , how much bandwidth I need for 30 simul. calls ?
Im planning to use G729 on both my server ?
to support 30 calls over IAX2 do I've to change some setting during compile
time or not ?
pls suggest.
2007 Jul 17
5
Asterisk PRI Busy Problem
Hi,
I've an PRI coming to my asterisk ,calls are coming fine and my agents are
able to answer no prob. but I've an agreement with my telco with some
incoming no if the no of calls on these no are more then 3 then send to
another no. they use busy signal to divert call on another number so I'm
sending the call to Congestion() if no of calls in this group are more then
3. But my
2007 Apr 19
3
Outgoing CallerID
I am not sure of the best way to do this, so I thought I would query the list.
I have about 100 internal extensions ranging from 2000 - 2100. Each
internal extension has a external DID number. For example: 2001 =
5552871620. As you can see the internal externsion and DID don't
match in any way. What would be the best way to set the DID for when
a extension dials out on the PRI? In
2010 Jan 05
5
CallerID on Indian PSTN is not working.
Hi,
I am using asterisknow 1.5.0 and Wildcard TDM410P card. Everything is
working fine except the caller ID of incoming call from PSTN line. The phone
display is showing "Unknown" when there is an incoming call. I think the
same problem listed here: https://issues.asterisk.org/view.php?id=6683
There is one patch on this link but i don't know how to apply patch on
asterisknow.
2001 Apr 23
1
several bugs (PR#918)
# Your mailer is set to "none" (default on Windows),
# hence we cannot send the bug report directly from R.
# Please copy the bug report (after finishing it) to
# your favorite email program and send it to
#
# r-bugs@r-project.org
#
######################################################
1. as.numeric behaves differently in R than in S and I think this
shows a bug in how S3
2009 Nov 05
1
Asterisk 1.4 DISA is jumoing after one digit in the DISA context
Dear list,
I have problems with DISA on an specific server with Asterisk 1.4.26.2.
After starting DISA I can only press one key and DISA is jumping direct
into the context without waiting for further digits.
In dtmf.log I found this:
[Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin '7' received on
SIP/214-00d92db0
[Nov 6 00:09:28] DTMF[2413] channel.c: DTMF begin passthrough
2007 Aug 16
1
Set CALLERID(num) to a specific number only if ${CALLERID(num)} is not an NANP number
Im trying to figure out the base way to check the callerID being sent
to my Asterisk box and use it if it is a valid NANP number, but
replace it with a static NANP number if it is not. (Why? I have a
few carriers that require this, and a few international users - if it
happens to take one of the carriers that require it, I want it to set
a static number that is valid).
I'm playing
2013 Aug 29
4
Add new calculated column to data frame
Hi,
I have a following data set:
id event time (in sec)
1 add 1373502892
2 add 1373502972
3 delete 1373502995
4 view 1373503896
5 add 1373503996
...
I'd like to add new column "time on task" which is time elapsed between two
events (id2 - id1...). What would be the best approach to do that?
Thanks,
Srecko
[[alternative HTML
2006 May 11
3
Please help me to combine two datasets.
Dear r-users,
Suppose I have two data sets
data set-1
Date height
------------------------
1/11/2005 10
2/11/2005 23
3/11/2005 54
4/11/2005 21
5/11/2005 22
data set-2
weight
--------
32
45
11
Now I want to combine this two data sets. i.e. i want to see:
Date height weight
-------------------------------------------
3/11/2005 54
2005 Aug 08
4
[PATCH] Fix TypeError when starting VMX domains
The attached patch fixes this traceback:
Traceback (most recent call last):
File "/usr/lib/python/xen/xend/XendDomainInfo.py", line 510, in construct
self.construct_image()
File "/usr/lib/python/xen/xend/XendDomainInfo.py", line 574, in
construct_image
self.store_channel)
File "/usr/lib/python/xen/xend/xenstore/xsobj.py", line 310, in
2013 Feb 17
3
Select components of a list
Hi Gustav,
Try this:
lapply(1:length(models),function(i) lapply(models[[i]],function(x) summary(x)$coef[2,]))[[1]] #1st list component
[[1]]
#??? Estimate?? Std. Error????? z value???? Pr(>|z|) # pm10
#5.999185e-04 1.486195e-04 4.036606e+00 5.423004e-05
#[[2]]
#??? Estimate?? Std. Error????? z value???? Pr(>|z|) #ozone
#0.0010117294 0.0003792739 2.6675428048 0.0076408155
#[[3]]
#???