: [Asterisk-Users] GSM Gateway / Terminal for sale
Hi,
I am interested. Could you please send me over brand and model? I will
check it out and confirm order back by email.
Jose Limeres
M.: +34 690-351498
SPAIN
www.boratelecom.com
On 17/11/05, Sam Tam [EMAIL PROTECTED] wrote:
Single port GSM Gateway
The 1900 is the tri band one right?
Can you drop me an email and I will work
out a price for the tri band one for you
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rusty Dekema
Sent: 17 November 2005 15:35
To: Asterisk
Users Mailing List - Non-Commercial
Well try to setup some QoS service on both router to let VoIP calls take
priority over any others data.
Also try to do some pinging test for 1 day or so and see if you are
suffering from any packet loss.
Packet loss can do a lot of harm to VoIP calls..
-Original Message-
From: [EMAIL
I am also interested in finding out if there is anywhere that I can grub
hold of a opensource SIP softphone with sourcecode that I can play around
with..
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Time Bandit
Sent: 20 November 2005 13:11
To:
I think I have heard in the past that someone mentioned to me there is a
codec that does not getting affected much because of packet loss.
Is there such thing?
Sam
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to get rid of so there will
be plenty left
Sam
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GSM 900MHz / 1800MHz GSM Gateway
Including antenna and power supply.
Limited Stock, please email gsm AT cyber-telecom.com for more info
Or visit www.cyber-telecom.net to purchase right away.
Sam
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the areskicc support but no response.
If anyone has this problem or got a bit of time please feel free to help me in
any way ..
Cheers
Sam
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Asterisk
Just wondering whether
there is any DID or relatively cheap E1 / T1 provider in HK that provide
incoming / outgoing calls minutes within HK.
Drop me an email if you know
any
Sam
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any recognition that such a problem exists let
alone that it has been fixed.
Regards,
- Sam
--
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Sam Tilders
[EMAIL PROTECTED]
(Move to Jupiter)
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.
- Sam
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[EMAIL PROTECTED]
(Move to Jupiter)
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Has anyone found a solution for asterisk and r2
signaling ? Steve Underwood had given some information saying he had a working
asterisk working. I need it to work with Argentina R2 signaling
Sam
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Hi
I have reason to believe that I have errors in my
configuration because when I make a call I can see the H323 call executed ok but
not being processed by Zap. I am using R2 signaling (which I know is
incomplete but should I not see it when I debug Zap channel?). I think there is
a
'
-- Executing Dial("H323/ip$80.247.147.146:3852/19860", "Zap/g1/203755343") in
new stack -- Couldn't call
g1/203755343 -- Hungup 'Zap/1-1'
h323 -- Asterisk --- h323 works ok
Regards
Sam
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[E
Hi
I have reason to believe that I have errors in my
configuration because when I make a call I can see the H323 call executed ok but
not being processed by Zap. I am using R2 signaling (which I know is
incomplete but should I not see it when I debug Zap channel?). I think there is
a
).
Perhaps it's a bug that ringing Agent channels have no pickupgroup/callgroup
set?
- Sam
On Tue, Sep 14, 2004 at 01:00:04PM +1000, Sam Tilders wrote:
Hi folks,
Recently we assigned our users agent id's and switched to
having them use agentcallbacklogin instead of just ringing the phones
UK, London Based DID £1 per month
All number begin with 0208 0xx
If you are interested please email ukdid AT cyber-telecom.net
SIP based and support standard ulaw or alaw.
Unlimited incoming minutes.
For multi channels please email for pricing.
Sam
Antenna connection: SMA antenna tie-in, N type port(optional).TNC
port(optional)
For more info please email gsm AT cyber-telecom.net for more info or visit
www.cyber-telecom.net to purchase right away.
Sam
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Anyone know how to do it?
Sam
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The long waited Ultimate GSM Gateway is finally out. This time we have managed
to source a new patch of brand NEW GSM Gateway at prices that is only 50% of
what the market rate. And with the SMS Function and many more...
For purchase please email gsm AT cyper-telecom.net. We accept paypal and
be a pain in the ass..
/b
On Thu, 2005-12-08 at 23:41 +0800, Sam Tam wrote:
The long waited Ultimate GSM Gateway is finally out. This time we
have
managed to source a new patch of brand NEW GSM Gateway at prices that
is
only 50% of what the market rate. And with the SMS Function and many
It supports US power or EU power system.
Don’t worry about it.
It has been tested before and it should be
able to be used world wide without a problem
Sam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rusty Dekema
Sent: Friday, December 09, 2005
11:01 PM
What hardware/?
Sorry I have missed part of the message?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Evert Meulie
Sent: Monday, December 19, 2005 4:03 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Re: Does hardware like this
Drop me an email and I will send you a list
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dakota
Sent: Tuesday, December 20, 2005 9:14 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Affordable IP Phones for
Antenna connection: SMA antenna tie-in, N type port(optional).TNC
port(optional)
For more info please email gsm AT cyber-telecom.net for more info or visit
www.cyber-telecom.net to purchase right away.
Sam
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We have ran out of stock in our office in UK. All GSM Gateway are now being
send from HK therefore the shipping will be more expensive than usual.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bails
Sent: Friday, January 06, 2006 12:18 AM
To: Asterisk
Have a look at cyber-telecom.net. CT-GSM-1000
seems to be one of the cheapest GSM Gateway that you can buy right now.
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Tuesday, June 27, 2006 11:41
PM
To: 'Asterisk
Users Mailing List -
Get an GSM Gateway from cyber-telecom.net
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 4:57
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [asterisk-users] call
forwarding to mobile phone
regards,
Sam Liang
Support Network Engineer
ISPhone Australasia Pty Ltd
Tel: +61 (0)2 8213 8686
Fax: +61 (0)2 8213 8685
Mob: 0413509414
email: [EMAIL PROTECTED]
web:www.isphone.com.au
- /* Begin receive call from 203.166.103.233 with caller ID
'0413509414
a gsm network.
On 7/18/06, Sam Tam
[EMAIL PROTECTED] wrote:
Get an GSM Gateway from cyber-telecom.net
From: [EMAIL PROTECTED]
[mailto:
[EMAIL PROTECTED] ] On
Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 4:57
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Yes Get an X100P
Sam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lito Lampitoc
Sent: Tuesday, July 18, 2006 5:16
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [asterisk-users] call
forwarding to mobile phone
is there a way I
WE have found this type of phone work better than E61
http://cyber-telecom.net/shop/product_info.php/cPath/21/products_id/31
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Fredrik Emil
Jensen
Sent: Tuesday, July 18, 2006 4:04 AM
To: Asterisk Users
PortaSIP and
the Asterisk-client is, we want to be benefit from the IAX trunk of packet
saving and etc.
But we just can not pass the DNID(B-party number to the client Asterisk)
Hope that you can have some idea to help us.
We did not get the DID from didx.
Best regards,
Sam
-Original Message
PortaSIP and
the Asterisk-client is, we want to be benefit from the IAX trunk of packet
saving and etc.
But we just can not pass the DNID(B-party number to the client Asterisk)
Hope that you can have some idea to help us.
We did not get the DID from didx.
Best regards,
Sam
-Original Message
for help and idea therefore I could give my friend a
bit of help
Sam
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Any one know how can I use asterisk to record phone call i.e situration like
this
T1 - channels bank - 24 lines - PBX - Recording using asterisk -
24 phones.
Sam
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Is your server located locally ?
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of support
Sent: Friday, August 18, 2006
11:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] call
barge
Hi
i have running a call center and I
have 5 agents ,
port with a max of 16 ports in 1
chassis.
Sam
-Original Message-
From: Matteo Brancaleoni [mailto:[EMAIL PROTECTED]
Sent: Monday, October 30, 2006 12:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] VoIP GSM Gateways
Hi,
On Sun, 2006-10-29
Happy New Year …..
Sam
_
From: Dovid B [mailto:[EMAIL PROTECTED]
Sent: Monday, January 01, 2007 12:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Happy 2007!!!
Its the new year. Cant we all be semi nice for atleast a lil bit
I am in the process of implementing a new ISDN pri and have a couple of
questions. This is a full 24 channels (23 B and 1 D) delivered over a T1
interface. The interface looks good and is not showing any errors. Any help
that you can provide would be greatly appreciated.
1) What
(${EXTEN:2})
exten = _**XXX,n,Hangup()
[BLF]
; Defines a BLF Hint for phones
exten = 212,hint,SIP/sam
-snip-
SIP.CONF
-snip-
[sam]
type=friend
username=sam
fromuser=sam
callerid=sam
host=dynamic
dtmfmode=RFC2833
disallow=all
allow=ulaw
call-limit=20
subscribecontext=BLF
Thanks in advance for any
not working
On Tue, Nov 06, 2007 at 05:04:50PM -0500, Lutgring, Sam wrote:
When I execute a pickup on a ringing phone I get CALL FAILED REASON
CODE 603. I am dialing **212 with the following config. Anyone have
a suggestion?
I am not sure, but in the context where your extensions are, have you
done
it is not doing the
pickup.
Thanks for the help though.
-Original Message-
From: Baji Panchumarti [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 07, 2007 4:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Lutgring, Sam
Subject: Re: [asterisk-users] Pickup Command
Why not simply store voicemail local so there are no issues if the VPN
goes down. Then set up your dial plan at each site to allow the PSTN
access to your remote (other site) extensions. You can use IAX to trunk
a PSTN call just like you can a local caller, just give them access to
the context.
is what it looks like:
SIP.CONF
[sam-X-1433]; This is my X-lite phone
type=friend
username=sam-X-1433
-SNIP-
[sam-G-1433]; This is my desk phone
type=friend
username=sam-G-1433
-SNIP-
EXTENSIONS.CONF
exten = 1433,1,Dial(SIP/sam-G-1433SIP/sam-X-1433,22,Tt)
Hope you find this to be useful
Is there a way to allow a user to dial an extension after listening to
your voicemail instead of leaving a message? Example would be the big
boss is on vacation and changes his out message to say you can reach my
assistant at by dialing 1234 now or leave me a message.
Thanks in advance.
I have a strange issue with CLID that I would appreciate if someone
could point me in the right direction. When a call comes in (either
from another SIP user on the same Asterisk box or from the ISDN PRI) the
Caller ID Name is displayed correctly, but the Caller ID Number seems to
be empty. My
Has anyone experienced the situation where you receive a
PRI_EVENT_PROGRESS message from a PRI that is then sent to a SIP channel
where the SIP client (tried 2 different phones/manufactures) never
acknowledges, Asterisk resends the message two more time and then begins
hanging the call up?
I have mine set up to ring a group of designated phones. Each one of
those phones has a dedicated line button that subscribes to their
particular account in the group. This way when the phone rings the user
KNOWS that it is the main building number that is ringing.
-Original Message-
I am having an issue with the CallerID Number not being passed to my
phone in the SIP packet. The CallerID Name is passed just fine and
displayed on the phone with no issue. I have done a NoOp() in my
extension.conf and successfully seen both the CallerID name and number
correctly. So that leads
this code at checkout and enjoy 15% off on the total order
excluding delivery charges.
Kind Regards
Sam Tam
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I think what you need is a GSM Gateway
You can find a cheap one at cyber-telecom.net
The model you should be looking for is CT-G1000 or 2000
Plug the SIM in there and it will give you a RJ11 telephone port where you
plug into something like X100P then you are ok to go..
Sam
-Original
Try cyber-telecom.net
May be get a X100P with a CT-G1000 or G2000
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gordon
Henderson
Sent: Tuesday, January 15, 2008 3:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Try cyber-telecom.net
May be get a X100P with a CT-G1000 or G2000
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Sunday, January 20, 2008 11:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk
It is more economical to get a hardware GSM Gateway from places like
cyber-telecom.net and then plug it in a X100P
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bilal ghayyad
Sent: Monday, January 14, 2008 8:43 PM
To: asterisk-users
may be I should keep my month shut..
But hey I though a mailing list is trying to get other users helping each
other.
There is no way I cannot see my post being non constructive ..
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Kennedy
Sent
in. Then they are setup just like a normal phone line
Try to look at the voip wiki for x100p config
2. get a 8 ports fxo voip gateway and 8 ports gsm gateway then put those
together. They end up working the same way.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
Try the CT-G1000 from cyber-telecom.net it is 39.99 GBP atm.
Sam
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos Rojas
Sent: Wednesday, January 30, 2008 10:35 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk gateway
Hello everybody
Well I think you need a GSM Gateway
You can find some info on cyber-telecom.net
For a cheap option you can try a CT-G1000 or CT-G2000 and then plug it in a
X100P or something similar then it would be very economical.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Well alternatively you can look up straight forward gsm voip gateway.
Like CT-375
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves
Sent: Thursday, February 07, 2008 8:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I am using about 70 Grandstream GXP2000 phones on 1.1.5.15 code with *
1.4.16.2 and have not experienced any of these issues. The one thing
that I would suggest is make sure that you are using RFC2833 for you
DTMF Mode. I was originally using INFO and ran into some strange issues
with dropped
Try switching your DTMF mode on both asterisk and the phone to RFC2833. I have
not seen the issue that you are describing, but I had some very strange issues
like call hang-ups when I was using INFO. After switching the issues were gone
and I have had no further troubles.
Hope this helps
I know this is a bit off the thread
But I am trying to see if anyone in here know how to config a AS5300 with 2
T1.
Please contact me off list if you can give me a bit of help
Sam Tam
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gateway, and have the
ability to make more than 1 call at once through it.
Thanks
-Kev
Sam Tam wrote:
Try cyber-telecom.net
May be get a X100P with a CT-G1000 or G2000
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Graves
Sent: Sunday
I have an IAX trunk configured between 2 Asterisk servers. Everything is
working great except if the caller presses # during the call. If they press #
the local PBX comes on and says transferring and tries to transfer to a blank
extension. Does anyone know how to turn this off? There is no
/Asterisk-Partner+ACS+for+Voicemail
The problem i have is how to use this info to replace nice/witness
recording server
Sam
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If anybody who want to earn a quick $50 via paypal and can help me on
setting up a polycom ip7000 to work with asterisk please email sam __ tam AT
hotmail DOT com
Do not email me through my gmail account.
Sam
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Hi Shahnawaz
Have you considered how you are going to address location issue for Mobile
users calling 911. You should think of SS7 MAP/TCAP to atleast know their
Cell ID
Regards
Sam
Thanks very much everybody who contributed their thoughts. I would try
to get some DID's so that each physical
Hi Shahnawaz
Have you considered how you are going to address location issue for Mobile
users calling 911. You should think of SS7 MAP/TCAP to atleast know their
Cell ID
Regards
Sam
Thanks very much everybody who contributed their thoughts. I would try
to get some DID's so that each physical
:37.83 events/1 28 root
10 -5 000 S 0.0 0.0 0:15.67 events/2 29 root 10
-5 000 S 0.0 0.0 0:40.36 events/3 30 root 10 -5
000 S 0.0 0.0 0:16.45 events/4
*
Thanks
Sam
Hi Steve
Even though you shouldn't have to, have your rebooted? 200 days of
uptime and this just started?
It seems this problem is common as i have three boxes of the same capacity
with exactly the same problem. So reboot should only solve the problem for
a while
Have you recently updated
Hi Team
Can someone advice me on how i can lower the load average on my asterisk
server?
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.10.1
asterisk-1.4.25.1
2 X TE412P Digium cards on ISDN PRI
Im using the system as an IVR without any transcoding or bridging
**
top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55,
55.75
Hi Sam!
Hello Steve!
Are there any side-effects from the high load average? The system
doesn't seem to be CPU or disk bound from the look of the CPU stats.
System %age is
high by way
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 question
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 question
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 you put
with alternative solution?
Muro, Sam wrote:
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
(currently is done via Nice) on asterisk - This's the problem
Sam
--
Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
Hi there
Has anyone know how to configure asterisk to be able to query Corba
interface directly from the dialplan
Sam
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Hi there
Does anyone know how to configure asterisk to be able to query Corba
interface directly from the dialplan
Sam
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0117473789
NAME: Franklin John
STATUS: Active
Can someone advice on how i can catch this values from AGI or directly on
dialplan.
Thanks
Sam
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Kyle Kienapfel wrote:
On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam resea...@businesstz.com
wrote:
I am having a problem understanding the way to retrieve some parameters
to
asterisk via AGI or what ever method that fits. I have an executable
program that accept one parameter (CALLERID
Steve Edwards wrote:
On Sun, 25 Jul 2010, Muro, Sam wrote:
I am having a problem understanding the way to retrieve some parameters
to asterisk via AGI or what ever method that fits. I have an executable
program that accept one parameter (CALLERID) and return customer status
from
Kyle Kienapfel wrote:
On Sun, Jul 25, 2010 at 9:04 AM, Muro, Sam resea...@businesstz.com
wrote:
Kyle Kienapfel wrote:
On Sun, Jul 25, 2010 at 8:18 AM, Muro, Sam resea...@businesstz.com
wrote:
I am having a problem understanding the way to retrieve some
parameters
to
asterisk via AGI
Steve Edwards wrote:
On Sun, 25 Jul 2010, Muro, Sam wrote:
I am having a problem understanding the way to retrieve some
parameters to asterisk via AGI or what ever method that fits. I have
an executable program that accept one parameter (CALLERID) and return
customer status from the database
Use the *SIPAddHeader(Header:Content)* application in dialplan. I don't
think Method specific SIP headers can be done via asterisk.
On Fri, Aug 26, 2011 at 3:05 PM, Jaime Lozano jaimelozan...@gmail.comwrote:
Hello everybody,
I want Asterisk Server to send packets (SIP packets) to some 3Com
Alternative work around to this could be:
1- Make two different dialplan extensions. One to dial DAHDI numbers with
setting for DAHDI and other extension for SIP dialing. Both extensions
setting different CallerID presentation
2- Create a queue with Local extensions as static members
Though this might have been resolved/accomplished already but I've couple of
questions for Virendra Bhati.
1- If you are doing this to make new accounts for new users, why couldn't
you use Asterisk realtime(DB) based configurations of
Voicemail/MoH/SIP/dialplan etc wouldn't it be much easier than
AM, Sam Govind govoi...@gmail.com wrote:
Though this might have been resolved/accomplished already but I've couple
of questions for Virendra Bhati.
1- If you are doing this to make new accounts for new users, why couldn't
you use Asterisk realtime(DB) based configurations of
Voicemail/MoH/SIP
and last things which hurt me.
On Mon, Sep 5, 2011 at 12:48 PM, Sam Govind govoi...@gmail.com wrote:
1- Per my experience I've used DB with configuration files and I was
amazed that Asterisk was taking a union of DB + conf file configurations and
accepting both.So if you just make a simple
There could be as easy solutions as using teamviewer or use tools like
Hamachi used in combination with dyn-dns etc. IP-tunneling I guess needs
static public IPs for the sake of completing the route.
On Wed, Sep 7, 2011 at 5:30 AM, A Dunor alsta...@gmail.com wrote:
Thanks for the speedy pointer
See this link:
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
You'll find similar pages where you can setup to store queue logs/events(as
Alex mentioned) in MySQL DB and further do your triggers or functions on
them.
On Wed, Sep 7, 2011 at 10:46 AM, Michael
See absolute timeout. I think yours' a complex thing to achieve I guess
absolute timeout may be the thing that can help. In older versions
absoluteTimeoute(n) could take you to exten T when time n elapsed. now I
guess funtion Timeout() is used as replacement.
here's an excerpt from somewhere:
;
and queue_log
doesn't exist either (as a file or as a db table). We're using AsteriskNOW,
so maybe these files/tables were not created.
How should we add them?
Thanks.
On Wed, Sep 7, 2011 at 8:54 AM, Sam Govind govoi...@gmail.com wrote:
See this link:
http://www.voip-info.org/wiki/view/Asterisk
Thats goood ! :) thanks for updating.
On Fri, Sep 9, 2011 at 2:16 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
Can you please provide an excerpt of your logs when this happens?
Regards
Ish
On Fri, 2011-09-09 at 09:05 +, Vinod Dharashive wrote:
Hi sam,
Have solved the problem
Hey,
I think I remember the same post before. previously I heard someone telling
to use vicidial or some other thing like that.But I don't think that those
are totally AMI based call-generators.
What I'd recently done is make a php page which connects to Asterisk's AMI
port. I send page request
1- *-bash: obd-demo.ulaw: No such file or directory* // try use absolute
file path i.e /usr/src/mymp3.mp3 . I guess that's why its saying no such
file or directory.
2- http://lists.digium.com/pipermail/asterisk-users/2006-March/144689.html Go
through this thread.
3- When everything fails from
Hey krishnan,
Everything happens for a reason. The most intuitive cause of this issue
seems to be network change. Can you confirm that no change in networking
happened! because your server is sending register requests but not getting
responses. Meanwhile the same server replying to scenarios2 can
it .call file start playback at it's own channels but I am not able to
hear anything into conf.
As i know localdial is not joining into the conf. but how I will do it so
that I will be able to hear any played file into conference ?
On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com
to
hear anything into conf.
As i know localdial is not joining into the conf. but how I will do it so
that I will be able to hear any played file into conference ?
On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com wrote:
Good to know,
I think it'll be a feedback score or a poll
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