hello, all of users:
sorry, resend it again for clarifying the message. I have implemented cha_ss7
in china. initially, the
chan_ss7 can not support the call group. i modify the code.
now the problem is that, both sides can hear the ring, but i
can not hear the voice from each other.
i think
On Thu, 19 Mar 2009, Christian Victor wrote:
grandstream gxp-2000 works fine for that.
depending on firmware rev its two ports are either a switch or router.
Grandstream removed this functionality in recent softwware upgrades - I
guess they needed the code space for other things.
Why would
On Fri, 20 Mar 2009, Stephen Davies wrote:
Hi,
Are you sure that Verizon amswers the call? They should play that
message as 'early media' without answering, after which they cam clear
the call with an appropriate cause code.
Similar issue in the UK and yes, the carriers do answer the call
On Thu, 19 Mar 2009 16:38:02 -0300, David fire ddf...@gmail.com
wrote:
dive in the mailing list archive in February a very nice user sent an email
about how to do load balancing using opensip.
I don't suppose you know the Subject line, do you David?
I can't find it!
2009/3/20 D Tucny d...@tucny.com
2009/3/19 Oguzhan Kayhan oguzh...@bilkent.edu.tr
Hi, i know i am asking a lot of questions lately in this forum..sorry
about that first of all. :)
Ok, here is the deal..
I am trying to make a hybrid system with an ericsson MD110 and
asterisk.
Hi all,
I mentioned in asterisk.conf there is a property maxcalls...I know that
this is the max number of concurrent calls but i need to know please if this
entry is commented out, what is the default number of MAX concurrent calls
supported by asterisk?
Regards
Hi to all the ML. I'm new here.
I start to use asterisk with realtime configuration, with pgsql
backend connected via odbc.
The connection between asterisk and pgsql works fine.
I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501.
Those are the records:
asterisk= SELECT
Dear All,
I'm trying to send FAX to an endpoint Behind NAT...The scenario i the
following:
PSTN_GW--Asterisk--asterisk--OpenSIPS--Endpoint behind NAT..
The FAX is failed and I got the following error log on asterisk:
Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite:
On Fri, 13 Mar 2009 09:22:12 +0100, Lenz Emilitri wrote:
I'm only half joking: what about parsing the full log looking for
command inviocations and channel IDs? this would be completely
transparent, albeit insane :)
The full log is insane on a busy server. You get no decent call tracing,
you
On Tue, 17 Mar 2009 13:34:34 +, Geraint Lee wrote:
what about relogging the information using:
Set(CDR(customfield)=${CDR(originalfield)})
That won't work because there's no way (that I know of) to collect the
DTMF numbers which the Dial() command consumes in order to complete
the
Steve Underwood schrieb:
Hi Olivier,
Olivier wrote:
T.38 says that if the call starts in audio mode it is the called end
which should initiate a re-invite to change from audio to T.38. This
makes sense, as that is the end which has the best chance of figuring
out if a FAX
Configure emaildateformat in voicemail.conf.
I worked around the english weekdays by using numeric weekdays (see man
strftime)
emaildateformat=%d. %m. %Y um %H:%M Uhr
If you need the weekday in French you have to set the Linux Locale to
french. But this affects all parts of Asterisk where
Quoting Oguzhan Kayhan oguzh...@bilkent.edu.tr:
Yes seems pretty simple,
so can i use a wildcard instead of giving all numbers one by one?
Such
instead of exten = 1500,1,Dial(SIP/1500)
exten = _15XX,1,Dial(SIP/_15XX) ?
assuming all my local numbers are on 15 prefix??
Almost.
exten
SS7 doesnt send any voice. It sends call info, and tells the switches
which trunk to use for the voice. Trunks are two-way as far as audio
content, though they maybe designated is inbound or outbound trunks.
An audio problem is possibly a NAT or other issue.
Since you are modifying the SS7
On Fri, Mar 20, 2009 at 1:53 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Fri, 20 Mar 2009, Stephen Davies wrote:
Hi,
Are you sure that Verizon amswers the call? They should play that
message as 'early media' without answering, after which they cam clear
the call with an
On Fri, Mar 20, 2009 at 5:36 AM, michel freiha mich...@gmail.com wrote:
Can you please help me in order to find the real issue?
Try taking out three or four pieces of your architecture, and then try again.
How about PSTN - Asterisk?
___
-- Bandwidth
From a cell user level perspective...
The cell companies are doing it like they think makes sense.
If they know your cell is off/out of range they route instantly to VM.
They could give 4-10 rings of fake effort, but why. With follow me
roaming and such, they want to process the call as fast as
2009/3/20 Klaus Darilion klaus.mailingli...@pernau.at
Steve Underwood schrieb:
Hi Olivier,
Olivier wrote:
T.38 says that if the call starts in audio mode it is the called end
which should initiate a re-invite to change from audio to T.38. This
makes sense, as that is
Hello All,
I have a little complicated question about the Dial command.
I use OpenSIPs to loadbalance Asterisk Servers, and Users are registered on
Asterisk servers.
Asterisk use the Reg. Contact entry to reach the UAC via the OpenSIPs server.
Everything works except for trunk numbers:
For each
On Fri, 20 Mar 2009, Stephen Davies wrote:
Hi,
Are you sure that Verizon amswers the call? They should play that
message as 'early media' without answering, after which they cam clear
the call with an appropriate cause code.
Similar issue in the UK and yes, the carriers do
Hello,
I've been looking into OpenSIPS to see if it's a worthwhile addition to our
setup. We're currently running a cluster, using Heartbeat, between two servers.
It works well but I'm interested in seeing if we can improve it. My manager
heavily uses RPM's for installations rather than
I still find it weird as even if it is a switch timing problem. Because
when is it calling my phone *all the time *and that other area code it *never
*calls it. Does that mean asterisk always complete my number in a certain
time frame, and the other number no? Also I get the progress code 127
GrandCentral/Google Voice does just this, although I have no idea what
they use for a back end to make it happen. When someone calls your
GC/GV number, it forwards out to a list of numbers you have given the
service. You can choose to answer the call, send it on to voicemail, or
a couple of
Hello,
I want to ask that if thee are some ATA decives that i can use to connect
mutliple analog phone lines to my VOIP system..
I mean for example an ATA device with 24 ports with 24 independent SIP
accounts.
For example for some dormitories in my area, i want to put an ATA device
and move
Don't really know the answer, but these are givens:
1. your phone is (most likely) in the same area code as the asterisk
installation
2. NY is most likely not in the same area code.
3. Even though the T1 is a dedicated digital service, the code that
handles all of this is/was
Grandstream makes an 8-port unit which we've had success with, you could
use three of them.
Hello,
I want to ask that if thee are some ATA decives that i can use to connect
mutliple analog phone lines to my VOIP system..
I mean for example an ATA device with 24 ports with 24 independent SIP
Hello,
I want to ask that if thee are some ATA decives that i can use to
connect
mutliple analog phone lines to my VOIP system..
I mean for example an ATA device with 24 ports with 24 independent
SIP
accounts.
For example for some dormitories in my area, i want to put an ATA
Hello Darrin,
Maybe you should ask this question on OpenSIPs mailing list.
I have build a rpm for CentOS 5.2 using and updated opensips.spec from svn
1) retrieve opensips.init and opensips.spec-4.4 from
https://opensips.svn.sourceforge.net/svnroot/opensips/branches/1.4/packaging/rpm
2) retrieve
On Friday 20 March 2009 00:38:45 Cary Fitch wrote:
Sure if you can get up stream carriers to cooperate. Just follow the CDRs.
But short of a subpoena... or enlightened self interest, like the calls
take down a tandem.. (not likely).
We could loop the calls back to get ATT's attention, but
I am trying to connect asterisk 1.4.23 to a customer that has eon from
eoncc.com
Has anyone done this before?
They dont have a sip trunk so we are using sip a SIP extension.
All I get is Registration timed out, trying again
my register line is something like:
register = 5...@ipaddress
[VOIP]
The message I play to people (or machines) answering is something like DEX
Yellow Pages call for Jones Bail Bonds, press [1]
This information, along with the caller's phone number, etc., is logged for
follow-up.
The key, as Cary points out, is to look for DTMF to confirm that the call is
not
This information appears to be relevant, but useless?
--Don
Don Kelly
PCF Corp
People Come First
651 842-1000
888 Don Kell(y)
651 842-1001 fax
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Well it will get me off my rant in this forum. Isn't that worth something?
Seriously, as users some of us have one 2 line system and others are
running multiple systems, absorbing hundreds of thousands of calls a day.
Where the %#! warranty calls are coming from or not coming from is useful
Cary Fitch wrote:
The problem has two prongs - first we are in control of our own
landlines and can use asterisk to screen whatever crap we wish before
disturbing a real user or allowing a vm to get stored (but it would be
nice not to have to).
The other issue is we are not for the most part
Cary Fitch wrote:
SS7 doesn’t send any voice. It sends call info, and tells the switches
which trunk to use for the voice. Trunks are two-way as far as audio
content, though they maybe designated is inbound or outbound trunks.
An audio problem is possibly a NAT or other issue.
Since you
Hi Darrin, Hi Marc,
Darrin, with an OpenSIPS frontend you can do more things actually:
1) move the HA in OpenSIPS - it will be able to re-route if one of the
Asterisk boxs is down
2) do LB - you can use in parallel multiple Asterisk boxes and to
balance the traffic between
3) you can terminate
hello,
please any asterisk Management application that use the WXWidget
Graphicalll User Interface (GUI) ?
the FreePBX is not fully accessible to my screen reader.
thanks.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Thanks for your help
Don’t really know the answer, but these are “givens”:
1. your phone is (most likely) in the same area code as the asterisk
installation
My phone has a different area code than the asterisk installation. The
asterisk box is in FL and I can call a number in MN but
Verizon wireless filed a lawsuit against the perpetrators of the car
warranty scam. I hope to hell they win.
http://www.foxnews.com/story/0,2933,501404,00.html
Cary Fitch wrote:
The problem has two prongs - first we are in control of our own
landlines and can use asterisk to screen
I am trying to get a queue to do more than just play music and hold calls.
Specifically, making some comforting voice announcements would be nice.
Below is the queues.conf file relevant portions.
Member phone number is munged to protect the guilty.
We shouldn't need the announcement source
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Moffett
Sent: Friday, March 20, 2009 12:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Area code 757 Car
On Fri, Mar 20, 2009 at 1:23 PM, Adam Moffett a...@plexicomm.net wrote:
Verizon wireless filed a lawsuit against the perpetrators of the car
warranty scam. I hope to hell they win.
http://www.foxnews.com/story/0,2933,501404,00.html
Thanks for the link, now we got a name to the scam fwitw :P
Hello Bogdan,
I have set a small rpm repository for opensips 4.4 for CentOS (with el5) x86_64
(and later i386 32bits)
Simply visit http://centos.leurent.eu/ and read the README.txt
Maybe we could just do the same on the official OpenSIPs website?
++
Le Friday 20 March 2009 18.12:25
The most popular answer I've seen here is to replace the regular music with
a streamed audio feed which can be anything you have access to. I'd try and
give you details, but they wouldn't be correct. This information is pretty
easy to locate in the digium site, viop-info.org or google.
- Cary Fitch ca...@usawide.net wrote:
I am trying to get a queue to do more than just play music and hold
calls.
Specifically, making some comforting voice announcements would be
nice.
You may want to take the quotes off of the filenames in your queues.conf config
file... they're not
[Mar 20 12:45:33] WARNING[4940]: app_queue.c:3136 try_calling: The device
state of this queue member, SIP/3617001000, is still 'Not in Use' when it
probably should not be! Please check UPGRADE.txt for correct configuration
settings.
[Cary Fitch]
We are running 1.4.22 and this message popped up
On 3/20/09, Cary Fitch ca...@usawide.net wrote:
I am trying to get a queue to do more than just play music and hold calls.
Specifically, making some comforting voice announcements would be nice.
Below is the queues.conf file relevant portions.
Member phone number is munged to protect the
Hi Yehavi,
Please see my inline comments:
Yehavi Bourvine wrote:
Hello,
Sorry for the delay - was out of office. I also cross-posting it to
OpenSIPS list.
I have a small pilot (20-30 phones) which also does some sort of SIP
to PRI transcode for our old PBX. The pilot is base on
The Asterisk.org development team is pleased to announced the release of
Asterisk release candidates 1.6.0.7-rc2, 1.6.1.0-rc3, and beta release
1.6.2.0-beta1. Additionally, new release candidates of Asterisk-Addons
1.6.0.2-rc1 and 1.6.1.0-rc3 have been created. Note that the 1.6.1 series of
Jon Pounder wrote:
Cary Fitch wrote:
The problem has two prongs - first we are in control of our own
landlines and can use asterisk to screen whatever crap we wish before
disturbing a real user or allowing a vm to get stored (but it would be
nice not to have to).
The other issue is we
Hey all;
I am experiencing an issue with music on Hold. I am running asterisk version
1.4.22, and have a default script set up in two places for MoH playback. For
internal devices to my network that are SIP peering with asterisk, they simply
dial 123 and hear the MoH music immediately. I'm
Hi,
I'm starting testing 1.6.2 beta. CentOs 5.2
I found my first crash, first I have
[Mar 20 20:30:41] WARNING[11201]: res_config_mysql.c:611 update_mysql:
Attempted to update column 'useragent' in table 'sip', but column does not
exist!
[Mar 20 20:30:41] ERROR[11201]:
Dear all, I want to know if anybody has implented an Asterisk server
(1.4 or 1.6) with SRTP and SIP with TLS, in order to encrypt both
signaling and voice packets.
Is it possible ??
And in the affirmative case, does encryption increase the delay and so
the voice quality becomes wrong ???
Thanks
Seems to be a crash with func_odbc.
Let me know what info you need to check it.
Thnks
From: Sebastian [mailto:s...@adinet.com.uy]
Sent: viernes, 20 de marzo de 2009 10:37 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: 1.6.2 beta 1 crash
Hi,
I'm
On Mar 20, 2009, at 7:49 AM, Oguzhan Kayhan wrote:
Hello,
I want to ask that if thee are some ATA decives that i can use to
connect
mutliple analog phone lines to my VOIP system..
I mean for example an ATA device with 24 ports with 24 independent SIP
accounts.
For example for some
Hi Cary,
I glanced at the code in main/devicestate.c and it seems the
AST_DEVICE_NOT_INUSE
is set when the agent channel does not exist ... and that's most likely true
when the agent is not logged in
or you had asterisk reloaded and the information about agent has been lost
... (true only for
Do you have zaptel timing working ? (or dahdi) ?
If no timing is available then if there's no incoming audio frames coming
from the external SIP channel then no outgoing audio will be produced (even
if you have
MOH application working)
The trigger to fire the outgoing audio frame comes from the
57 matches
Mail list logo
