The cellphone can be presented to Asterisk as SIP device using OpenBTS (GSM
to SIP conversion).
On Tue, Feb 15, 2011 at 10:40 PM, Faisal Hanif fai...@vopium.com wrote:
Hi,
Your question is not clear but below are possible answers to your question,
If you want to attach you cell-phone to
On Tue, 15 Feb 2011, Hans Witvliet wrote:
On Tue, 2011-02-15 at 09:01 +, Steve Howes wrote:
On 15 Feb 2011, at 03:39, Jeff LaCoursiere wrote:
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
Hi every one,
When i run asterisk i am getting error as
ERROR[2109]: chan_sip.c:3963 __sip_reliable_xmit: Serious Network Trouble;
__sip_xmit returns error for pkt data
i am unable to understand what it is and how come.. can any one help
me regarding
On Mon, Feb 14, 2011 at 10:39:20PM -0500, Jeff LaCoursiere wrote:
Now this is what I call uptime...
minipbx*CLI show uptime
System uptime: 41 years, 7 weeks, 6 days, 3 hours, 26 minutes, 46 seconds
Last reload: 8 hours, 3 minutes, 51 seconds
Bizarre bug?
Hi. I see that 41 years, 7
Hi group,
I have a simple call center scenario set up on Asterisk. Customer calls the
DID and gets placed in Queue waiting for their turn to talk to the available
agent.
Sometimes Customer hangs up in between and in this case I want to get the
last position of customer in Queue.
I know there is a
I'm running Asterisk 1.6 and was wondering if anybody have a workig barge
in solution running.
I was thinking of using chanspy, but i would like that the original call
would be dropped, and the new call would be the only one there.
--
On 16 February 2011 00:22, Ernie Dunbar maill...@lightspeed.ca wrote:
At 12:12 PM 2/15/2011, you wrote:
I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
v1.6.2.1. They can call each other's extensions (and make and receive
calls otherwise), but they cannot transfer calls,
On 16 February 2011 10:13, Peter den Hartog peterdenhar...@gmail.com wrote:
I'm running Asterisk 1.6 and was wondering if anybody have a workig barge
in solution running.
I was thinking of using chanspy, but i would like that the original call
would be dropped, and the new call would be the
Hi,
Anyone who has deployed Cisco 7945G phone with asterisk, kindly share your
experience.
/ag
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar
Okay, so let me try to make it more clear to be sure everybody gets it :-),
i can be a bit unclear from time to time ;-).
100 is in a call with 101.
102 has a higher priority and calls 100. The call between 100 101
disconnects, and 102 100 are connected.
Peter
On Wed, Feb 16, 2011 at 11:29
Dear Mr,Ms;
I am planing for a custom IVR, for example to act as a simple installer!
I mean there is some choice via 0-9 and # as *Next* and * as *Back* button.
is there any way for me to detect if the caller pressed # vs * on Dialplan ?
--
Regards,
Ali R. Taleghani
0936 322 4069
--
Try it with your own AGI-Script - this is more flexible.
http://www.voip-info.org/wiki/view/Asterisk+AGI
Am 16.02.2011 11:45, schrieb shayne.al...@gmail.com:
Dear Mr,Ms;
I am planing for a custom IVR, for example to act as a
If you use Asterisk 1.8.x you can have this in channel vars and can collect
and add to DB or file on h extension.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Man
Sent: Wednesday, February 16, 2011 3:06 PM
To:
On Wed, Feb 16, 2011 at 7:32 AM, ast guy ast...@gmail.com wrote:
Hi,
Anyone who has deployed Cisco 7945G phone with asterisk, kindly share your
experience.
/ag
I did some 7941's a few months ago with SIP. They work pretty well.
Make a console cable for the AUX port and you can see them
Hello,
can anyboby tell me, how can I disable the echo cancellation for sip?
thx a lot...
best regards,
Felix
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
You might use a SIP-to-cellular gateway as I on
did.
http://www.mgraves.org/2008/11/how-to-add-a-cellular-trunk-to-your-voip-
system-part-1/
Michael
--Original Message Text---
From: logan
Date: Tue, 15 Feb 2011 21:49:26 -0800
Hello,
Are there any gateways which allow me to hook a cellphone
Hi guys,
the function Echo() did work on CAPI, but doesn't work for SIP connection.
Can anybody help?
thanks a lot.
best regards,
Felix
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
--Original Message Text---
From: Michael Graves
Date: Wed, 16 Feb 2011 05:46:30 -0600
You might use a SIP-to-cellular gateway as I on
did.
http://www.mgraves.org/2008/11/how-to-add-a-cellular-trunk-to-your-voip-
system-part-1/
Here's a shortened URL for convenience:
http://j.mp/gQMjNR
Michael
Hello,
if you want use a Huawei 3G usb modem, take a look at chan_datacard module:
http://wiki.e1550.mobi/doku.php
Regards
- Andrea--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
On Wed, Feb 16, 2011 at 2:49 AM, logan logan...@gmail.com wrote:
Hello,
Are there any gateways which allow me to hook a cellphone to Asterisk and
use that line for routing my calls? Basically, I'm looking to play around a
bit and if I can get to connect a cellphone with Asterisk then that
On Wed, Feb 16, 2011 at 12:01:20AM +0100, Hans Witvliet wrote:
kept on reading the thread...
Wouldn't it be better, for asterisk at least, to get rid of all this
identification / authentication stuff?
Keeping config files holding pain passwords or simple md5 isn't the way
to solve this...
It is in client but not in asterisk sip channel
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Wednesday, February 16, 2011 4:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to diable echo
Did you executed Answer() before it?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Wednesday, February 16, 2011 4:48 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] function Echo() doesn't work
On Tue, Feb 15, 2011 at 11:51:26PM +0100, Hans Witvliet wrote:
On Tue, 2011-02-15 at 07:18 -0500, Richard Kenner wrote:
Anyway, the answer is: No, it's mathematically impossible to do
that. Even if the passwords were stored encrypted, Asterisk itself
has to be able to get the plaintext
Yes, I did it. The function Echo() works both on CAPI and Datacard (UMTS
Stick). Just only no echo on SIP. Any suggestion?
2011/2/16 Faisal Hanif fai...@vopium.com
Did you executed Answer() before it?
*From:* asterisk-users-boun...@lists.digium.com [mailto:
Check if you have incoming SIP call in supported codec or send CLI log for
further troubleshooting.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Wednesday, February 16, 2011 5:14 PM
To: Asterisk Users Mailing
* == Using SIP RTP CoS mark 5*
*-- Executing [1174614@von-voip-provider:1]
Answer(SIP/sipgate-account-, ) in new stack*
*-- Executing [1174614@von-voip-provider:2]
Echo(SIP/sipgate-account-, ) in new stack*
* == Spawn extension (von-voip-provider, 1174614, 2) exited
Hi Hanif,
I indeed use 1.8 .0 but couldn't find the channel variable for caller's
last position in queue anywhere in documentation.
Would you please let me know the channel variable name?
Thanking you.
On Wed, Feb 16, 2011 at 4:40 PM, Faisal Hanif fai...@vopium.com wrote:
If you use Asterisk
On 02/15/2011 06:08 PM, Jian Gao wrote:
How about encrypt the whole hard drive?
If I built a server and give to other people, there is no easy way to
stop them reset the root password or just mount my drive to read
everything on it. But if build an encrypt OS then it will be secure. My
question
I faced same issue for sipgate but got it resolved by allowing all codec in
sipgate peer config.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Wednesday, February 16, 2011 5:33 PM
To: Asterisk Users Mailing List -
In which conf-Data should I allow all codec? Thank u for explaining.
2011/2/16 Faisal Hanif fai...@vopium.com
I faced same issue for sipgate but got it resolved by allowing all codec in
sipgate peer config.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
You have enable following in queue configuration,
setinterfacevar=yes
setqueueentryvar=yes
setqueuevar=yes
and you will find your data in following variables,
${QEORIGINALPOS} will have position when caller enter the queue.
${QUEUEPOSITION} will have position when caller left the
I tried to set allow=all in sip.conf. But it still doesn't work.
2011/2/16 Faisal Hanif fai...@vopium.com
I faced same issue for sipgate but got it resolved by allowing all codec in
sipgate peer config.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
ken...@gnat.com (Richard Kenner) writes:
Here's a possible design:
- There's optionally a file in the config
directory called master_key. It contains just a string.
- A CLI command core encrypt string is added to Asterisk. It takes the
provided string, encrypts it using the string in
Did you make any peer for sipgate if yes then do for that peers. Please also
note that disallow line should be before allow line.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Felix Dong
Sent: Wednesday, February 16, 2011 6:22 PM
- The config file reader looks for strings of the form {enc:string}:
and replaces them, before otherwise parsing the line, with the decrypted
version of the string using the key in the master_key file.
This sounds pretty reasonable, except perhaps that you might only want
to convert
Hi,
I am not sure if it is doable:
1. We originate one call from Asterisk
2. Asterisk plays one audio file to the called part when the called part picks
up the phone.
3. Asterisk establish one real connection between the caller part and the
called part.
Thanks,
Songtao Yu
Yes, I did it exactly as what you said. It still doesn't work. :-(
2011/2/16 Faisal Hanif fai...@vopium.com
Did you make any peer for sipgate if yes then do for that peers. Please
also note that disallow line should be before allow line.
*From:* asterisk-users-boun...@lists.digium.com
You can do it using callback files or AMI.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Songtao Yu
Sent: Wednesday, February 16, 2011 6:41 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Play one audio file to the
It is somehow back to normal. Nothing change. May be the sip provider
problem. However, it lasts for quite a while.
Thanks
On Wed, Feb 16, 2011 at 12:04 PM, Faisal Hanif fai...@vopium.com wrote:
You can also append add dtmf logging to cosole and see if dtmf is coming
from carrier.
*From:*
In your sip.conf, in trunk parameters use:
dtmfmode = INFO
Date: Wed, 16 Feb 2011 23:07:16 +0800
From: aster...@ck-lee.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] DTMF not detected, time out
It is somehow back to normal. Nothing change. May be the sip provider
Hi Felix,
Am Mittwoch, den 16.02.2011, 12:47 +0100 schrieb Felix Dong:
Hi guys,
the function Echo() did work on CAPI, but doesn't work for SIP
connection. Can anybody help?
thanks a lot.
are You trying to echo between local phones or is it a external call via
some VoIP Provider?
In
Hello all,
Thank you for the responses. I really appreciate it.
Since I'm just trying it out for fun, I will begin by using mobile-chan and
see how that goes.
Thanks a lot,
Hitesh
On Wed, Feb 16, 2011 at 4:05 AM, Andrew Latham lath...@gmail.com wrote:
On Wed, Feb 16, 2011 at 2:49 AM, logan
How should I configure my asterisk server so that I can receive calls from
an unregistered peer from whom I also receive registrations of sip phones?
I'm asking you this, because with my actual configuration, when I register a
contact from that peer's IP, no more inbound calls are accepted from
Well a quick n easy fix for you is you can configure you call sending peers
to use username secret in INVITE. As far as I know it possible in almost
all CISCO, Avaya and all other standard Gateway and SBCs which follows full
SIP RFCs.
If you can't do it then you need to use curl as realtime
I was messing with something in conf dir. I reinstalled asterisk and
removed extensions.conf and lua extensions is working now.
I think lua in dialplan is a killer feature. It enables complex apps
to be done in a much easier way now.
2011/2/16 Faisal Hanif fai...@vopium.com:
You may need to
Isn't this a limitation that can be surpassed with some configuration that
I'm lacking in my sip.conf or extensions.conf of my asterisk?
Ricardo.
On Wed, Feb 16, 2011 at 4:54 PM, Faisal Hanif fai...@vopium.com wrote:
Well a quick n easy fix for you is you can configure you call sending
Hi,
I'd like to know if there's an easy way of doing the following:
SIP phone dials a custom feature code in Asterisk,
call gets answered within a custom context (Answer()),
anything that the caller says should be redirected/piped to an external
application.
Something like monitor except audio
On 16 February 2011 00:22, Ernie Dunbar maill...@lightspeed.ca wrote:
At 12:12 PM 2/15/2011, you wrote:
I have two Aastra phones, a 6730 and a 6757, both connected to Asterisk
v1.6.2.1. They can call each other's extensions (and make and receive
calls otherwise), but they cannot transfer calls,
Hi,
I'm looking to get an ISO of FreePBX or AsteriskNOW installed on a USB that
I can boot from and also be able to save my changes. Is this possible?
My search on web doesn't seem to find anything useful. For now I don't have
the option of having a spare machine or creating a partition on my
I have played a lot on this issue with asterisk config but in realtime it
doesn't supported static peers with version 1.6.2.14.
From: Ricardo Carvalho [mailto:rjcarvalho.li...@gmail.com]
Sent: Wednesday, February 16, 2011 10:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
EAGI could be your target application.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent: Wednesday, February 16, 2011 11:01 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] pipe audio
You can simply use Portable LinuxLive USB Creator 2.6 or grub4dos. And
make your USB bootable by any Linux Live ISO.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of logan
Sent: Wednesday, February 16, 2011 11:24 PM
To: Asterisk Users
On 16 February 2011 00:22, Ernie Dunbar maill...@lightspeed.ca wrote:
At 12:12 PM 2/15/2011, you wrote:
I have two Aastra phones, a 6730 and a 6757, both connected to
Asterisk
v1.6.2.1. They can call each other's extensions (and make and receive
calls otherwise), but they cannot transfer calls,
I am posting here since you guys are my last hope.
I am trying to configure a Polycom Soundpoint IP 335 with MWI.
Is there any way to eliminate the scrolling messages and Msgs softkey?
I am trying to get it where it's just the light that indicates the new
messages.
I don't know if Asterisk
On Wed, Feb 16, 2011 at 4:51 PM, ERIC HERRON e...@lanline.com wrote:
I am posting here since you guys are my last hope.
I am trying to configure a Polycom Soundpoint IP 335 with MWI.
Is there any way to eliminate the scrolling messages and Msgs softkey?
I am trying to get it where it’s
What I am trying to achieve is not in there.
Thanks though.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham
Sent: Wednesday, February 16, 2011 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
On Wed, Feb 16, 2011 at 2:51 PM, ERIC HERRON e...@lanline.com wrote:
I am posting here since you guys are my last hope.
I am trying to configure a Polycom Soundpoint IP 335 with MWI.
Is there any way to eliminate the scrolling messages and Msgs softkey?
I am trying to get it where it’s
I have it on the 430s.
I think it's a firmware issue but I am having trouble replicating it on the
430
I could have sworn I had it on one phone before I rebooted it but memory
might be influenced by hopes.
From: asterisk-users-boun...@lists.digium.com
On Wed, Feb 16, 2011 at 3:05 PM, ERIC HERRON e...@lanline.com wrote:
I have it on the 430s.
I think it’s a firmware issue but I am having trouble replicating it on the
430
I could have sworn I had it on one phone before I rebooted it but memory
might be influenced by hopes.
What
Hi Everyone,
I have a SIP turnk which works fine with both inbound and outbound calling.
However, the only issue is that there is no Ring Tone if someone calls us.
The phones used are Aastra and Polycom connected to the PBX via VPN (SIP).
I do get an outbound ring tone, so it's not that there is
dear
I have a good exp in setting up 79xx on sccp, with sccp-b library, and tftp
server, which part is the main problem for you?
best
On Wed, Feb 16, 2011 at 3:10 PM, Andrew Latham lath...@gmail.com wrote:
On Wed, Feb 16, 2011 at 7:32 AM, ast guy ast...@gmail.com wrote:
Hi,
Anyone who
On IP430s
cat sip.ver
VVX-1500 3.2.2.0481
All others 3.2.2.0477
2345-11402-001.bootrom.ld sip.ld
Phone1.cfg
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=*97 msg.mwi.2.subscribe=
Sip.cfg
up.mwiVisible=0
some outside sip provider does not accept dtmf,
if you have not this problem in your local, ask your outside carrier
best
On Wed, Feb 16, 2011 at 7:27 AM, asterisk asterisk aster...@ck-lee.comwrote:
In the past it was set as auto and worked. I change to RFC2833 but did not
work.
How can I
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce B
Sent: Wednesday, February 16, 2011 2:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] No ring tone on inbound call - but
Thanks. Indeed ringing instead of MoH which was missing files fixed the
issue. Thanks for the quick great tip. Simple things hide from us sometime.
On Wed, Feb 16, 2011 at 3:48 PM, Danny Nicholas da...@debsinc.com wrote:
--
*From:*
On 02/16/2011 09:43 PM, ERIC HERRON wrote:
On IP430s
cat sip.ver
VVX-1500 3.2.2.0481
All others 3.2.2.0477
2345-11402-001.bootrom.ld sip.ld
Phone1.cfg
msg msg.bypassInstantMessage=1
mwi msg.mwi.1.subscribe= msg.mwi.1.callBackMode=contact
msg.mwi.1.callBack=*97 msg.mwi.2.subscribe=
Sip.cfg
On Wed, Feb 16, 2011 at 5:49 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
I share your pain. I have an IP335 and IP670 here. Have not configured the
IP335 yet but using the latest Admin Guide (3.3.1) did configure the IP670
running the latest bootrom (4.3.0) and firmware (3.3.1).
I haven't played with the backlights yet.
One annoyance at a time.
To disabled the mwi chirp can be set to silence.
MESSAGE_WAITING se.pat.misc.1.name=message waiting
se.pat.misc.1.inst.1.type=silence ..
I am trying the different firmwares now to see if it makes any difference.
What module are you using?
I have the cdr_mysql_addon.so module, and I can define alias for the
collums. I have an alias for start and end variable, and they both get
recorded as Unix EPOCH integers values on mysql. In Asterisk1.8 the collum
duration and billsec have milisec durations if you
On 02/17/2011 12:10 AM, Ryan Wagoner wrote:
[snip]
Backlight works fine on a IP550 with 3.3.1 . I have mine set to off
when idle. I like that the 3.3.x series doesn't required the default
sip.cfg and phone1.cfg files. The structure of the XML seems cleaner
and more consistent.
up
Any thoughts?
http://googleblog.blogspot.com/2011/02/simple-way-for-publishers-to-mana
ge.html
10% sounds like a bargain for what amounts to a license server
Am i missing something?
I'm wondering how this can be used on the asterisk platform?
Cheers,
Dean
Posted By Dean
On 02/17/2011 12:17 AM, ERIC HERRON wrote:
I haven’t played with the backlights yet.
One annoyance at a time.
Agreed :)
To disabled the mwi chirp can be set to silence.
MESSAGE_WAITING se.pat.misc.1.name=message waiting
se.pat.misc.1.inst.1.type=silence ….
Thanks for the tip Eric. The
On 02/17/2011 12:17 AM, ERIC HERRON wrote:
[snip]
I am trying the different firmwares now to see if it makes any difference.
In the admin guide I just came across: up.oneTouchVoiceMail default 0
If set to 1, the voice mail summary display is bypassed and voice mail
is dialed directly (if
On Wed, Feb 16, 2011 at 8:38 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 02/17/2011 12:10 AM, Ryan Wagoner wrote:
up
up.backlight up.backlight.idleIntensity=0
up.backlight.onIntensity=3
/up.backlight
/up
Here's what I have:
up
up.idleTimeout=10
If set to 0, when you press Msgs, it goes to the message center.
If set to 1, it dials the voicemail box directly bypassing the message
center.
I been all over..lol.
Thanks though!
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
On Tue, Feb 15, 2011 at 10:31 AM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F
Sent: Tuesday, February 15, 2011 9:29 AM
To: Asterisk Users Mailing List -
On 02/17/2011 03:20 AM, Ryan Wagoner wrote:
[snip]
whichsection it is under. My 3.2.x config file worked except for
alert info, ringer, and feature settings, which was outlined in
Simplified_Configuration_Improvements_in_UC_Software3_3_0_TB60519.pdf
Just went through that doc. Interesting
On 02/17/2011 12:17 AM, ERIC HERRON wrote:
To disabled the mwi chirp can be set to silence.
MESSAGE_WAITING se.pat.misc.1.name=message waiting
se.pat.misc.1.inst.1.type=silence ….
This did not work but looking at the example files in the 3.3.1 firmware
the snippet below did work (mind the
Hi
I wanted to use asterisk as SIP client in my centOS box.I should
able to make calls and receive calls.and should able to talk and listen
from the headset that I connected to my CentOS box.
I need a direction to start on this.
Thanks
Nikhil
--
Install asterisknow and begin from there.
http://www.asterisk.org/asterisknow/
and don’t miss to read the documentation
https://wiki.asterisk.org/wiki/display/AST/Home
Regards
Khaled Chehab
NGN Eng.
Operations Office - Lebanon
Office : +961 1 868686 ext 115
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