Hi,
I would like to see the DNID in my MySQL CDR logs.
I have read one big thread in the Asterisk Developer List, but I could
not figure out how to implement it ?
Is there a simple step-by-step.
Thx in advance.
Vai
--
_
-- Ba
Use the userfield.
On 03/15/2010 04:25 AM, RSCL Mumbai wrote:
> Hi,
>
> I would like to see the DNID in my MySQL CDR logs.
>
> I have read one big thread in the Asterisk Developer List, but I could
> not figure out how to implement it ?
> Is there a simple step-by-step.
>
> Thx in advance.
>
> Va
Hello Guys,
I have been trying to do this since 2 days but couldn't make itneed your
help..
The scenario is as under:
PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
I am trying to use SIP on Cisco Gateways and Asterisk box for the connection.
The configuration is as under:
Ishfaq Malik wrote:
> Hi
>
> I'm trying to get ExtenSpy to work but it wont, I'm dialling a number
> from my mobile which comes into our server and answering the number on a
> particular SIP extension which all works fine. I'm then dialling an
> exten from my own SIP extension which executes the
Am 12.03.2010 13:17, schrieb Steve Davies:
> Hi,
>
> I am just moving from Asterisk 1.2+bristuff up to 1.6.2, a huge leap
> :) I was wondering if someone could point me at 3 things that I appear
> to have "lost"?
>
> 1) ZapEC(off) - Is there an equivalent dialplan command to request no
> EC on a
Isn't the use of DNID separate to the userfield? I'd like to have this
working also.
Lee
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
Balashov
Sent: 15 March 2010 08:34
To: Asterisk Users Mailing List
I have read 2 solutions
(a) Changing the Dial plan and capturing DNID and inserting it into
one of the existing column in CDR table.
(b) Copy new CDR related .c & .h files which have added the
functionality of recording DNID into MySQL.
For this, CDR table structure needs to be changed and a
Hi!
I just updated from 1.4 to 1.6.2.6 and Asterisk complains about my AEL
dialplan:
application call to Gosub affects flow of control, and needs to
be re-written using AEL if, while, goto, etc. keywords instead
What is the suggested replacement for an explicit Gosub() call? I use it
lik
Klaus Darilion wrote:
> Hi!
>
> I just updated from 1.4 to 1.6.2.6 and Asterisk complains about my AEL
> dialplan:
>
>application call to Gosub affects flow of control, and needs to
>be re-written using AEL if, while, goto, etc. keywords instead
>
> What is the suggested replacement for
FWIW, just received an android-based phone and after installing
"sipdroid" found that it works very well with asterisk ;).
It automatically dials numbers through asterisk if available and
otherwise through the gsm network.
Contacts integrate well too.
No ties to any telco or to google, just a ha
Conrad Wood wrote:
> FWIW, just received an android-based phone and after installing
> "sipdroid" found that it works very well with asterisk ;).
>
> It automatically dials numbers through asterisk if available and
> otherwise through the gsm network.
>
> Contacts integrate well too.
>
> No ties to
What do you mean chief? What am looking at is ability for asterisk to
receive a call and recording until it tier down without bridging it to the
physical device
Sam
> Would you like the advice in all caps?
>
> On 03/15/2010 01:20 AM, RESEARCH wrote:
>
>> Hi there
>>
>> I remember to ask this quest
What do you mean chief? What am looking at is ability for asterisk to
receive a call and recording until it tier down without bridging it to the
physical device
Sam
> Would you like the advice in all caps?
>
> On 03/15/2010 01:20 AM, RESEARCH wrote:
>
>> Hi there
>>
>> I remember to ask this quest
On Mon, 15 Mar 2010, Ishfaq Malik wrote:
> Conrad Wood wrote:
>> FWIW, just received an android-based phone and after installing
>> "sipdroid" found that it works very well with asterisk ;).
>>
>> It automatically dials numbers through asterisk if available and
>> otherwise through the gsm netwo
Jeff LaCoursiere wrote:
> On Mon, 15 Mar 2010, Ishfaq Malik wrote:
>
>
>> Conrad Wood wrote:
>>
>>> FWIW, just received an android-based phone and after installing
>>> "sipdroid" found that it works very well with asterisk ;).
>>>
>>> It automatically dials numbers through asterisk if avail
Muro, Sam escribió:
> What do you mean chief? What am looking at is ability for asterisk to
> receive a call and recording until it tier down without bridging it to the
> physical device
>
> Sam
>
>> Would you like the advice in all caps?
>>
>>
He means that you put the subject in all caps.
Oh.. I didnt know that.
Thanks
Sam
> Muro, Sam escribió:
> What do you mean chief? What am looking at is ability for asterisk to
> receive a call and recording until it tier down without bridging it to the
> physical device
>
> Sam
>
>> Would you like the advice in all caps?
>>
>>
> He means that
On Mon, Mar 15, 2010 at 4:42 AM, Mohit Saxena wrote:
> I have been trying to do this since 2 days but couldn't make itneed your
> help..
Well, you could certainly ask Cisco for help.
You did pay Cisco money, right?
> PSTN-Cisco AS5350---Asterisk BoxVoIP Providers
> I am abl
On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik wrote:
> Hi
>
> I'm trying to get ExtenSpy to work but it wont, I'm dialling a number
> from my mobile which comes into our server and answering the number on a
> particular SIP extension which all works fine. I'm then dialling an
> exten from my own
On Mon, 15 Mar 2010, David Backeberg wrote:
>
>> and also to do LCR and Quality based routing of International calls?
>
> I don't know what that means.
>
Least Cost Routing. Asterisk doesn't have anything built in for this. We
do it with an in-house AGI. Others have done similar things that
The problem is not with cisco as the SIP header on debug doesn't contain the
called number. It only says To:sip:ip add of cisco gw. It should say number:ip
addr of cisco gw.
Br,
Mohit C. Saxena I Data/ISP Department
Starcomms Plc.
1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-7
We are a mobile operator so has to work with the PSTN side E1s from the Mobile
switch. This is the reason for using Cisco Media gateways.
Kindly help
Br,
Mohit C. Saxena I Data/ISP Department
Starcomms Plc.
1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709
email:moh..
Continuing with the top posting parade...
Can you post your {sanitized} sip.conf and your extensions.conf for inspection?
--Tim
- "Mohit Saxena" wrote:
> The problem is not with cisco as the SIP header on debug doesn't
> contain the called number. It only says To:sip:ip add of cisco gw. It
On Mon, 15 Mar 2010, Mohit Saxena wrote:
> We are a mobile operator so has to work with the PSTN side E1s from the
> Mobile switch. This is the reason for using Cisco Media gateways.
I know you may be stuck with them, but you could just as easily plug in a
Digium/Sangoma/Rhino T1/E1 card (or
David Backeberg wrote:
> On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik wrote:
>
>> Hi
>>
>> I'm trying to get ExtenSpy to work but it wont, I'm dialling a number
>> from my mobile which comes into our server and answering the number on a
>> particular SIP extension which all works fine. I'm th
Bumping a thread without adding anything useful is annoying. If you do
it again, I won't be helping.
Although I have gotten quite a chuckle from your posts, it's really going to
hurt when you fall from that high horse.
--
_
-
> and also to do LCR and Quality based routing of International calls?
I don't know what that means.
LCR = "Least Cost Routing"
Routing a call based on the quality or cost of a route (PSTN term vs SIP to
PSTN term vs SIP to SIP) is actually quite common.
--
Sip.comf
[PCCW-KPN]
type=peer
host=41.205.190.15
allow=ulaw
qualify=100
nat=no
canreinvite=no
user=07028000709
extension.conf
exten=07028XX,1,Dial(SIP/PCCW-KPN)
Cisco Gateway:
dial-peer voice 110 voip
description Voip peer to test the server
destination-pattern 1234
session protocol sipv2
s
Yes, I mean the same Least Cost routing.
Br,
Mohit C. Saxena I Data/ISP Department
Starcomms Plc.
1261c Bishop Kale Close, Victoria Island, Lagos, Nigeria, +234-702-8000-709
email:moh...@starcomms.com
www.starcomms.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[
I did a clean install to freepbx 2.6.1 and now when i do *76 i get a 1 second
flash on the screen then the phone hangs up. the FOP says it is on DND
but some ext are still getting calls. once i do a *76 FOP still says I
am on dnd. I am running asterisk 1.6.0.21.
before i was getting a message li
exten=07028XX,1,Dial(SIP/${ext...@pccw-kpn)
You aren't sending an outbound DID with just SIP/PCCW-KPN.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mohit Saxena
Sent: Monday, March 15, 2010 12:42 PM
To:
- "Mohit Saxena" wrote:
> extension.conf
> exten=07028XX,1,Dial(SIP/PCCW-KPN)
Here is your issue. Shouldn't you be sending the number you'd like to dial with
the call? Try this:
exten => 07028XX,1,Dial(SIP/PCCW-KPN/${EXTEN})
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)72
On Mon, Mar 15, 2010 at 1:19 PM, Ishfaq Malik wrote:
> David Backeberg wrote:
>> On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik wrote:
>> You didn't mention version. Could be relevant.
> Apologies for not adding the version, it's 1.4.17
Yeah, that's relevant.
> I will try ChanSpy to see what h
Am 15.03.2010 13:48, schrieb Kevin P. Fleming:
> Klaus Darilion wrote:
>> Hi!
>>
>> I just updated from 1.4 to 1.6.2.6 and Asterisk complains about my AEL
>> dialplan:
>>
>> application call to Gosub affects flow of control, and needs to
>> be re-written using AEL if, while, goto, etc. ke
David Backeberg wrote:
> On Mon, Mar 15, 2010 at 1:19 PM, Ishfaq Malik wrote:
>
>> David Backeberg wrote:
>>
>>> On Fri, Mar 12, 2010 at 11:31 AM, Ishfaq Malik wrote:
>>> You didn't mention version. Could be relevant.
>>>
>
>
>> Apologies for not adding the version, it's 1.4.17
David Gibbons wrote:
>
> Bumping a thread without adding anything useful is annoying. If you do
> it again, I won't be helping.
>
>
> Although I have gotten quite a chuckle from your posts, it's really going to
> hurt when you fall from that high horse.
>
>
I thought that was a little harsh m
Hi folks,
I am struggling to install cdr_pgsql in asterisk 1.6.0.26. When I do the
./configure, it complains about the function PQescapeStringConn not
existing in -lpq, so when I do a make menuconfig, I can't select the
cdr_pgsql module.
I am using CentOS 5.4 with the yum PGDG repository for 8
Hi,
I have Asterisk 1.6.0.20 running on Red Hat Enterprise Linux Server
release 5.4 (Tikanga).
I am trying build an app_swift module which uses the Cepstral software.
I am compiling it with the following command line
gcc -I/opt/swift/include -g -Wall -D_REENTRANT -D_GNU_SOURCE -fPIC -c
-o app_
Hello all,
I would like to share with you an article [1] we have issued last week
(sorry, currently only in Romanian language - we plan to provide an
English version soon).
This article is describing a method to be used for obtaining the
maximum number of SIP simultaneous calls an Asterisk server
> I have read 2 solutions
> (a) Changing the Dial plan and capturing DNID and inserting it into
> one of the existing column in CDR table.
> (b) Copy new CDR related .c & .h files which have added the
> functionality of recording DNID into MySQL.
> For this, CDR table structure needs to be
Ioan-
Sounds like this would give a useful measurement regardless of server type,
network config, and other variable issues.
That should be a great tool.
Do you have any plans to test with Asterisk in 'native bridging' mode? I.e.
with RTP streams not touched in any way
by Asterisk? I assume
Vikram-
> http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly
>
> The link above indicates that it is possible to setup RTP streams to
> directly flow between endpoints and completely bypass Asterisk. I would
> like to know if this configuration would work when,
>
> a)
Are there DUNDi CLI commands for Asterisk 1.4? I have searched google
and I only see the dundi commands in Asterisk 1.6, although I see
reference to them in older version's of Asterisk such as Asterisk 1.4
here: http://www.asteriskguru.com/tutorials/cli_cmd_14.html. When I
view the CLI commands thr
Hi all,
This question has been asked for days, I think that would be more
comprehensible if i post it in a new thread.
What i want to do is something like karaoke. when users call to
asterisk, a music song is played while caller sings. Their voice
will be recorded and mixed with the music. To do
On Tue, 16 Mar 2010, Pham Quy wrote:
> Hi all,
>
> This question has been asked for days, I think that would be more
> comprehensible if i post it in a new thread.
>
> What i want to do is something like karaoke. when users call to
> asterisk, a music song is played while caller sings. Their voic
> On Tue, 16 Mar 2010, Pham Quy wrote:
>
>> How can I count down 60s? MixMonitor app doesnt have any time out
>> argument.
On Tue, 16 Mar 2010, Jeff LaCoursiere wrote:
>
> I think you would be more successful and have more control if you wrote
> it as an AGI. Then you could set a timer that wou
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