Has anyone had success patching chan_sip.c so that Asterisk will recognize
the tel: header for RDNIS information?
exten = get_in_brackets(tmp);
if (!strncasecmp(exten, sip:, 4)) {
exten += 4;
} else if (!strncasecmp(exten, sips:, 5)) {
exten += 5;
We're experiencing an issue where calls disconnect after 15 minutes. It
seems to happen just after Asterisk sends an update mesage.
RTP is being set up directly. Asterisk is only in the SIP dialog.
Has anyone experienced this issue?
4 PRIs inbound, 4 PRIs outbound, asterisk provides
Greetings,
I'm hoping that an extra pair of eyes might help me to solve a challenge...
Anyone have any idea why the following would not work? I'm trying to test
for a callerid value that is 4 digits in length..
exten = s,1,NoOp(CLID is ${CALLERID(all)})
exten =
supp...@drdos.info wrote:
Positively Optimistic wrote:
Anyone have any idea why the following would not work? I'm trying to
test for a callerid value that is 4 digits in length..
Differences between yours and mine:
Yours:
Gotoif($[${LEN(${CALLERID(number)})} != 4]?true)
Mine:
Gotoif
the ISDN message via SIP, it all
comes down to how the gateway handles the desired functions.
Sincerely,
Brian LaVallee
On 6/26/14, 11:24 PM, Positively Optimistic wrote:
We're using a Earthlink PRI converted to SIP via a MediaGateway. I
assume
the mediagateway will convert the headers
We would like to present a toll free CallerID when making outbound toll
calls. In the past, when our PRIs were directly connected to a Nortel
CS1000 we could do this, without issue. Now that the PRIs are front ended
by a mediagateway facing asterisk, we can no longer do this.
Is it possible to
On Thu, Jun 26, 2014 at 8:10 PM, Positively Optimistic
positivelyoptimis...@gmail.com wrote:
We would like to present a toll free CallerID when making outbound toll
calls. In the past, when our PRIs were directly connected to a Nortel
CS1000 we could do this, without issue. Now that the PRIs
on your account.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Positively
Optimistic
*Sent:* Thursday, June 26, 2014 10:11 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] CLID
We are using vpn routers to connect home users back to our office network.
Basically, shipping a mikrotik router that 'calls home' and establishes a
vpn connection for the pc and phone that are connected to the mikrotik...
user plugs router in, plugs phone and computer into router, and that
In a multi-tenant environment, we are sending various CallerIDs outbound
from asterisk based on who the user is. We have an insurance agency who
would like to present a toll free callerid. This works.. unless they're
calling a toll free number. In that case, occasionally, the call fails.
Check out transnexus We use their product... Seems to work well
On Oct 18, 2013 2:09 AM, binary dreamer binary.vor...@gmail.com wrote:
hello everyone. i am concerned about security to the PBX and i would like
to discuss different fraud detection methods.
Apart from making everything to
We, along with a lot of other people, have a phone number that is pretty
important to us. Yesterday, our VoIP provider went down... won't call
any names VI, but it was pretty bad...
Our goal is to create a script within asterisk, that will place a call out
one SIP trunk provider (not the one
Define it as a variable, use the variable to define the filename
Ex.
exten =
529,n,Set(monfile=num-${CALLERID(num)}_name-${CALLERID(name)}_${EXTEN}_UID-${UNIQUEID})
exten = 529,n,MixMonitor(/var/spool/disa/${monfile}.wav,,)
hello list,
i have asterisk 1.4 installed i use MixMonitor to
We have a scenario where we wish to present a toll-free caller id, yet have
our calls rated based on our billing-telephone-number. Is it possible to
present a number in the sip header for billing and another number in the
header for jurisdicional call rating?
Whereas today, all of our calls are
Have you looked at rtp debug? Is it possible reinvites are enabled?
On Mar 9, 2012 9:20 PM, sean darcy seandar...@gmail.com wrote:
On 03/09/2012 07:20 PM, Arstan Jusupov wrote:
It may sound silly but did you configure/open firewall ports on amazon
ec2? The instance itself as we as from the
-info.org/wiki/view/Asterisk+auto-dial+out*
Regards
Dhaval
On Wed, Jun 15, 2011 at 12:15 PM, Positively Optimistic
positivelyoptimis...@gmail.com wrote:
Greetings!!
We're getting some strange results using call files.. no matter the
technology, DAHDI, SIP, etc., we get a Call failed
Has anyone successfully made this scenario work in 1.4. I found info at
http://www.voip-info.org/wiki/view/Asterisk+presence indicating that this
does not work with 1.4 implementations.
--
_
-- Bandwidth and Colocation Provided
Is it possible to send sip messages (debug) to a file or syslog server
without having them present in the console? If so, does anyone know what
kind of performance hit this would create.
Instance has approx 800 sip peers.
--
_
Has anyone found a way to detect a loop condition in the dialplan.?? We
had a condition where this filled up 47 PRI channels in an NFAS
group connected to our media gateway... and endless loop if you will..
Thanks
--
_
--
Ladies, Gentlemen
We are experiencing an unusual problem in our asterisk 1.4.34.. We are
attempting to determine if channels are in use before paging to them.
This works correctly, as in it pages the phone.. however, we see the error
message below on the console... after googling, we
| Asterisk Freezes ? 1.4
Positively Optimistic wrote:
We have what I consider to be a large dialplan (-= 1501 extensions (2559
priorities) in 99 contexts. =-)
If we have more than 10 or so channels up (all SIP, no TDM) and issue
the extensions reload command.. quite often, asterisk
Good day..
We have what I consider to be a large dialplan (-= 1501 extensions (2559
priorities) in 99 contexts. =-)
If we have more than 10 or so channels up (all SIP, no TDM) and issue the
extensions reload command.. quite often, asterisk will completely freeze
up... requiring us to either
In followme , is it be possible to have a third option
Whereas,
takecall=1
declinecall=2
proposed option
transfercall=3 or, transferring the call directly from followme
isn't really neccessary, if the callee could answer the call, and transfer
it someplace, that would work as
...@lists.digium.com] On Behalf Of Positively Optimistic
Sent: Thursday, January 14, 2010 8:16 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Followme Options
In followme , is it be possible to have a third option
Whereas,
takecall=1
declinecall=2
proposed option
transfercall
We have used a lot of the GXP400x series.. In my option, they have a high
failure rate...we've been testing the SPA8000s in our lab... my
opinion is that the architecture, everything from the software to the metal
chassis is superb to the grandstream. The SPA8000 has a fan built in for
This works for us
exten = h,1,Set(CDR(userfield)=${RTPAUDIOQOS})
exten = h,2,Hangup()
results in
Set(SIP/rpx2399a-b61fc5e0,
CDR(userfield)=ssrc=213416392;themssrc=0;lp=0;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=0;rlp=0;rtt=0.00)
*From:* [EMAIL PROTECTED]
In theory ParkAndAnnounce has a lot of usefulness, however, that we've had
very little success with application...Our application is similiar to
the local Walgreens pharmacy.. Dr. Calls in, selects the Im a doctor with
a prescription option...call is parked, and announcement overhead is
We've searched but thus far have not successfully found a solution for this…
We're looking for a way to set a variable using get digits for a DISA
application. Sometimes we're away from the office and get a voicemail that
I need to respond to quickly and would prefer for the caller to be
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