Hello,
I am using a Swiss VoIP provider called sipcall. They have what they
call a SIP trunk, and it is less expensive than individual accounts. From
Asterisk's point of view, this is just a regular SIP account, which
can however receive and send calls from multiple numbers. I just migrated
from
appreciated. When this does happen, and we need to
intervene, we try to poke around for a few seconds and test different
things, but again this is a production system, so the quicker its back
up, the better. =)
Thanks,
Marc
Could you please give me a feedback regarding this issue, I'm not sure of the
answer I got browsing the web
Thanks and Best Regards
Le mercredi 19 janvier 2011 09:14:55, Marc Leurent a écrit :
Good morning,
I have a simple question,
Is this problem would affect also an Asterisk 1.4.38
Thank you for the confirmation
Best Regards,
Le vendredi 21 janvier 2011 14:17:20, Kevin P. Fleming a écrit :
On 01/21/2011 05:59 AM, Marc Leurent wrote:
Could you please give me a feedback regarding this issue, I'm not sure of
the answer I got browsing the web
Thanks and Best Regards
Good morning,
I have a simple question,
Is this problem would affect also an Asterisk 1.4.38 if Pedantic SIP
support: No in the Global Signalling Settings
For what I understood, no..
Or is it a simple way to postpone upgrade until next planned upgrade.
Best Regards
Le mardi 18 janvier 2011
phone to an entry in the directory, and displaying that
contact/extension's full name (eg, when I type in 20467 on my phone
and hit dial, my phone would then display Marc Smith while it was
ringing Marc Smith's phone).
I've Google'd quite a bit and haven't really found any solutions; I
found one
Take a look at http://dev.leurent.eu/voip/MOS/
I'v done this a long time ago, hope it will help!
++
Le 08.03.2010 11:10, mosbah.abdelkader a écrit :
Hello All,
MOS and R factor are the two QoS parameters used to estimate VoIP call
quality.
I have found that they are calculated from other
Can I use:
exten = 33,n,Set(ACCOUNT=waitexten()) ???
No. Something like
exten = 33,n,Read(ACCOUNT,,10)
See http://www.voip-info.org/wiki/view/Asterisk+cmd+Read
hth and happy new year to everyone
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with the sip server.
Before sip reload:
sip-out/abc 172.16.200.1 N 5060 UNREACHABLE
After reload:
sip-out/abc 192.168.1.1 N 5060 OK (29 ms)
Regards,
Marc Ketel.
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Am Friday 06 November 2009 00:17:36 schrieb Marc Lindner:
Dear list,
I have problems with DISA on an specific server with Asterisk
1.4.26.2.
After starting DISA I can only press one key and DISA is jumping
direct into the context without waiting for further digits.
The reason and solution
READ I do not have problems to enter digits by DTMF so I
assume its related to DISA.
best regards
Marc
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I have the same result with Asterisk 1.4.21 on a Debian Lenny server
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Marc LEURENT
lf...@leurent.eu
Le mercredi, 28 octobre 2009 12.27:59, Marc Leurent a écrit :
Hello, when I remove a peer from my sip.conf and just do a reload, the peer
is still ping with SIP OPTIONS until I restart
=default
;dtmfmode=info
;insecure=port,invite
;nat=never
;sendrpid=yes
;disallow=all
;allow=alaw
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lf...@leurent.eu
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Thank you Klaus and Martin for your answers!
It's very helpful!
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lf...@leurent.eu
Le vendredi, 23 octobre 2009 20.51:54, Martin a écrit :
You can call application Progress() from within dialplan and it will
cause the Asterisk to send a SIP reply 183
on the call that came
Supported: replaces
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lf...@leurent.eu
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I need to make a conference between 2 numbers, one of them is external and
it has an extension. So, I need to dial the number and later enter the
extension, how can I do that?
something like this :
exten = 5145551212,Dial(Zap/g0/5145556000,20,D(7287))
see
is needed.
Any help is greatly appreciated!
Thanks,
Marc
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Thank you Shaun for your answer!
Indeed, I have made some basic tests to convert a file to g729 using the
software codec and it works!
Have a nice day!
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Marc LEURENT
lf...@leurent.eu
Le mercredi, 9 septembre 2009 19.32:30, Shaun Ruffell a écrit :
On 09/09/2009 09:33 AM, Marc Leurent
service_notactivated.g729: empty
service_notactivated.gsm: data
I was able to create the gsm file with the command, but the g729 one is empty.
Have you got any idea how I can solve this?
Thanks
PS: I'm able to place call in g729 without problem and the TC400B works well
--
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Marc LEURENT
lf...@leurent.eu
What I have is 1 front desk phone only with 6 lines
Front Desk Phone line 1 - incoming extension 1
Front Desk Phone line 2 - incoming extension 2
Front Desk Phone line 3 - incoming extension 3
Front Desk Phone line 4 - incoming extension 4
Front Desk Phone line 5 - incoming extension 5
I mean when the subs dials the digits with some delay between entering the
digits sequentially . At our current case , the Asterisk will wait about 2
seconds to see if another digit will be dialed or not and then he will route
the dialed digits according to the pre-defined routing table or he
Hello everybody,
I was wondering what is postponing the 1.4.26 release? I thought it was
scheculed for last week.
Is there something we can do to help to release this version?
There is no more issue reported on https://issues.asterisk.org/ for the time
being.
Best Regards,
-- --
Marc LEURENT
lf
Hi,
I want to install an Sangoma A200 together with an BRI card.
I would like to use Asterisk 1.4
Are there any howto or tips?
First compile bristuff and after compile wanpipe?
thanks...
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Hi,
Just wondering what the popular open source call statistics / metrics
packages are for Asterisk? Preferably an all-in-one package that
supports queues and calls from the CDR information generated by
Asterisk.
Whats everyone using? Favorites?
Thanks,
Marc
On Mon, Jun 8, 2009 at 9:18 AM, Christopher
Stamperchristopherstam...@gmail.com wrote:
I'm considering implementing an Asterisk PBX for conferencing. Before I get
started, I wanted to make sure that it supports the features that I need.
I plan to use Asterisk as a conference bridge only. I
On Fri, Apr 10, 2009 at 7:46 PM, John Rogers j...@wizworks.net wrote:
Thank you for the links! Of course if anyone else knows of other IAX ATA
offerings, please *DO* share. Really looking for a good solution. I will
buy one of each of these offerings to test and I'll share my findings with
://www.itu.int/ITU-T/studygroups/com12/emodelv1/tut.htm
Have a nice day!
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
Rue Eugène-Marziano 15 - 1227 Les Acacias
http://www.vtx.ch - marc.leur...@vtx-telecom.ch
:\2sip:
$(hdr(X-number-to-dial))@\3/ig');
}
Have a nice day!
-- --
Marc LEURENT
Le Monday 23 March 2009 13.41:59 Marc Leurent, vous avez écrit :
I have spoken to quickly,
Usually Asterisk on an incoming call sends an INVITE
Reg.Contact
Number@Reg Contact IP to the Peer IP
Hello all, I have put my MOS.ods file into
http://dev.leurent.eu/voip/MOS/
My problem is to add the jitter value into the formula
Have you got any idea how to do it?
-- --
Marc LEURENT
Le Thursday 02 April 2009 11.20:06 Mindaugas Kezys, vous avez écrit :
Could you share with us your
On Thu, Apr 2, 2009 at 11:37 AM, Cary Fitch ca...@usawide.net wrote:
Is there a way to program an FXO device to totally ignore incoming calls?
put the port in that context :
[incoming-noanswer]
exten = s,1,Hangup()
hth
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(rtpqos|audio|all)})
exten = s,n,ResetCDR(vw)
exten = s,n,NoCDR()
So I retrieve these values in my MySQL CDR table in order to calculate a MOS
value:
ssrc=592614191;themssrc=0;lp=1;rxjitter=0.00;rxcount=0;txjitter=0.00;txcount=20734;rlp=0;rtt=0.094000
codec used: g711a
--
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Marc
from the [Open]SER family.
lftsy wrote:
Hye everybody, anyone has any idea how to help me?
To resume, I just want to know how to change the IP in the URI sent by
Asterisk (first line of SIP packets)
Thanks for your time!
++
On Fri, 20 Mar 2009 15:09:55 +0100, Marc Leurent
that only background [general]
sip.conf settings will then apply:
Dial(SIP/1...@ip.of.peer.not.in.sip.conf)
Marc Leurent wrote:
Hello,
it is not an OpenSIPs problem I have, it's an Asterisk one,
I would like to change the URI in message generated by Asterisk.
Thanks
Le
:55 Marc Leurent, vous avez écrit :
Thank you, this is exactly what I needed!!
In order to Dial any number to a registered peer, I just have to enter
Dial(SIP/anynum...@sippeername)
Best Regards!
Le Monday 23 March 2009 11.31:31 Alex Balashov, vous avez écrit :
The Request URI generated
. Contact of the main number) but it
doesn't work
Have you got any idea how to rewrite the IP of the URI sent?
Thanks!
--
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Marc LEURENT
lf...@leurent.eu
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asterisk-users
an RPM from for libxml2 from
ftp://xmlsoft.org/libxml2/, but the dependencies for it are causing me
headaches. Any suggestions would be helpful. Thanks.
--
-- --
Marc LEURENT
Ingénieur VoIP
DECKPOINT SA
Une société du groupe VTX Telecom
-Andrei Iancu, vous avez écrit :
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
I was looking at the aastra 9133i, however I was informed that this phone is
no longer supported. What are good phones around the $100 - $125 price
point? (Need POE)
I like the Polycom IP-330. 2 lines, nice speakerphone, dual ethernet,
support PoE and works with 2.5mm headset.
$110 at
Asterisk doesn't use PING to check the STATUS, it uses a SIP OPTION message.
Regards,
Marc
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: mardi 17 février 2009 14:06
To: Asterisk Users Mailing List - Non-Commercial
The Attrafax software that was mentioned at the beginning of the thread does
support Gateway mode.
Regards,
Marc
-Original Message-
Fabio Mosti wrote:
2009/2/16 Steve Underwood ste...@coppice.org:
You don't indicate the kind of setup you are using.
I use asterisk
the devicestate for a channel?
Marc Hudson
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Philipp Kempgen wrote:
Jared Smith schrieb:
On Thu, 2008-10-23 at 09:36 -0700, Marc Hudson wrote:
I've looked at 'core show hints' and it is in fact reporting INUSE when
it's not, and NOT_INUSE
when it is.
That definitely sounds like a bug to me. Could you please report
'/var/lib/asterisk/agi-bin/cid-to-acct.php': No such file or directory 2
== cid-to-acct.php: Failed to execute
It is not complaining about the lack of /usr/bin/php, but about the
fact that the file /var/lib/asterisk/agi-bin/cid-to-acct.php is
nowhere to be found.
Probably asking the obvious
on these boxes and can't just upgrade to any old
version to see if it fixes it. I need to figure out what the bug is. I
did some research, but couldn't find it.
Peder
Do the rt* options in sip.conf have any effect? Maybe one of those might help?
--Marc
On Fri, Jul 11, 2008 at 11:46 PM, Marc Smith [EMAIL PROTECTED] wrote:
Hi,
I'm having a problem with IMAP storage and asterisk. Here is the error
message I get (in this instance its checking messages):
[Jul 11 23:14:12] WARNING[9888]: app_voicemail.c:8738 mm_log: IMAP
Warning: SECURITY
if GMail/Asterisk-IMAP integration is going to
be right for our institution, even if I get this working correctly,
there is still the problem of how to turn on IMAP access for all GApps
accounts and make it stay on. I don't believe the GApps API supports
changing user's options that way.
--Marc
2008/6
Anyone else ran across something like this? Ideas?
Thanks,
Marc
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Hi,
Anyone using Asterisk IMAP voicemail storage with Google Apps / GMail
IMAP? If so, does their IMAP implementation support any kind of
master user (Dovecot) abililty? Good? Bad?
--Marc
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On Wed, Jun 4, 2008 at 9:04 AM, Steven Howes [EMAIL PROTECTED] wrote:
On 4 Jun 2008, at 11:43, Joao Ferreira gmail wrote:
can I connect 2 FXS plugs to the same analog phone ?
No. Fire and death.
Unless you use a 2-lines analog phone :)
___
--
I have a wav file recording that i want to use on my voicemail, how
can i set this up?
You could play that file before sending the person to your voicemail
and pass the s option to it
Type show application voicemail on asterisk CLI to see the options.
hth
can anyone help me. I'm finding the softphone which can trigger web
browser and use callerid to go web page
You don't say on what OS you need it to run.
Mine is for Windows and support receiving URL (ex.:
Dial(IAX2/7003|20|trw|http://asterisk.org)
You can get it here :
(alternative title - what did I do wrong? or suggestions to make this
work)
Thought I'd try 1.6 beta5 (and 1.4.18 didn't want to compile vpb
/usr/lib/gcc/i386-redhat-linux/4.1.2/../../../../include/c++/4.1.2/i386-redhat-linux/bits/gthr-default.h:48:
error: â does not name a type )
1.6 did
with nat=yes and nat=no and almost any possible combination of the
SPA942 options so I don't paste the configs ;)
Thanks,
Marc
--
http://www.marcfargas.com -- will be finished some day.
signature.asc
Description: Esta parte del mensaje está firmada digitalmente
On Feb 19, 2008 10:44 AM, Anton Krall [EMAIL PROTECTED] wrote:
Guys, Im looking for a good text file editor for asterisk config files
that can be embedded on a web page for online editing (on an interface),
any recommendations?
You mean, something like this :
I found that there will be a memory leak if asterisk running day by
day without restart. Is it good to restart asterisk service daily?
What is the better way to restart it daily like apache?
Probably depends on the version of Asterisk, but I don't restart daily
From one in production used
Are other clients I should know about?
http://www.zoiper.com/
http://www.counterpath.com/
Add to that list
- Mozphone (http://mozphone.mozdev.org/) that can be installed in Firefox
-Kiax : http://sourceforge.net/projects/kiax
- shameless plugMy MediaX softphone :
Marc, does your client play nicely with Vista? We've been having some
problems with softphones that work fine in XP, but choke in Vista.
I don't know, never tried it since I couldn't find a machine with
enough power to run Vista decently ;)
Try it and let me know how it goes.
If it doesn't
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good morning,
Is it possible with asterisk to allow to share the same account on 2 different
devices, for example I want both my fix phone and my wifi phone to ring
in the same time.
I want to do it without making ringroups...
Any idea how to do it?
Any one advise a good strong softphone that can work with IAX fine?
samelessplugTry my softphone :
http://www.marccharbonneau.com/asterisk/mediaxphone.php/samelessplug
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Diax is probably the smallest Windows softphone.
Add to that list Mozphone (http://mozphone.mozdev.org/) that can be
installed in Firefox
Kiax : http://sourceforge.net/projects/kiax
shameless plugMy MediaX softphone :
http://www.marccharbonneau.com/asterisk/mediaxphone.php/shameless
plug
iaxcomm
add the following line: file:///Users/YOURUSERID/ports
portindex
sudo port install asterisk
Please report success or issues directly to me.
Thanks, Marc.
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-04 at 17:20 +0100, Marc LEURENT wrote:
It's just that I received SIP notify message saying that there is
nothing in the voicemail even when there is a message...
Do you have a mailbox defined for the SIP device in sip.conf? If you
don't, Asterisk has no way of matching up a mailbox
, Marc LEURENT wrote:
Good evening, I have something strange,
I have unread message in my voicemail box but the SIP NOTIFY that are
received by my telephone are like:
whereas there is voice messages inside!
Any idea how to solve that? Thanks
PS: I'm using asterisk 1.4.13 + Freepbx
#
U
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good evening, I have something strange,
I have unread message in my voicemail box but the SIP NOTIFY that are
received by my telephone are like:
whereas there is voice messages inside!
Any idea how to solve that? Thanks
PS: I'm using asterisk 1.4.13
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Just download the g729 module that fits your hardware at
http://downloads.digium.com/pub/telephony/codec_g729/ and follow the
README: http://downloads.digium.com/pub/telephony/codec_g729/README
PS: do a 'cat /proc/cpuinfo' to know what it your
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good morning,
I would like to find a simple PCI express card with only one FXS module,
do you know where I can find such a card?
Thanks
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (Darwin)
Comment: Using GnuPG with Mozilla -
/component/option,com_remository/Itemid,40/func,
fileinfo/id,25/
And yes - it's FREE as name suggests.
Regards/Pagarbiai,
Mindaugas Kezys
Advanced Billing for Asterisk PBX
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marc LEURENT
Sent: Friday
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
Is the asterisk B2BUA patches useful anymore??
I'm trying to set a prepaid SIP network and the only way seems to get
through a patched asterisk with B2BUA functions..
The patches failed, Hunk + problems: I have repaired them, but is it
very
Good evening,
Have you got any idea which prepaid application will be the best to do
simple prepaid calls with a MySQL storage...?
PS: I have a compiled by hand Asterisk 1.4.13 on a Debian Etch
Thanks
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On Nov 13, 2007 6:14 PM, Marc LEURENT [EMAIL PROTECTED] wrote:
Good evening!
I was wondering one thing,
I'm using freepbx to configure my asterisk server and I have a problem
with some inbound calls.
When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an
inbound
Good evening!
I was wondering one thing,
I'm using freepbx to configure my asterisk server and I have a problem
with some inbound calls.
When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an
inbound route! It matches a DID number.
How can I route an INVITE sip:[EMAIL PROTECTED]
Of course, use the codec for the pentium 4!!
bilal ghayyad a écrit :
Dear Marc;
Thanks a lot for your kindly help.
My output of the command cat /proc/cpuinfo is:
[EMAIL PROTECTED] /]# cat /proc/cpuinfo
processor : 0
vendor_id : GenuineIntel
cpu family : 15
model
:
Dear Marc;
I readed your email about the codec G729a and I am now
also need to install the codec on my Asterisk.
I typed from Asterisk CLI:
core show version and I got the following:
Asterisk SVN-branch-1.4-r72556 built by root @
localhost.localdomain on a i686 running Linux on
2007
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good evening,
I have something strange, when I add an ALERT_INFO variable to a ring group,
the invite generated contains 2 lines with Alert-Info and my phones return a
400 Bad Request...
I've checked in my config files, there is only one line with
/40kbps) data 0
format_g723.so G.723.1 Simple Timestamp File Format 0
The codec_g729a.so doesn't appear..
Any idea how to solve the problem.
Thanks
Best Regards,
Marc LEURENT
-BEGIN PGP SIGNATURE-
Version: GnuPG v1.4.7 (GNU/Linux)
Comment: Using
appear..
Any idea how to solve the problem.
Thanks
Best Regards,
Marc LEURENT
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have license for g729a audio codecs and I would like user to use them and
when the limit of 10 is reached, I would like the others to use ulaw...
Do youu know how to do it...
I have put:
allow=g729,ulaw
disallow=all
But ulaw is always chosen
Thx Steve!
Steve Totaro wrote:
Qualify=yes?
Thanks,
Steve
Marc Patino Gómez wrote:
Hi Adam,
thanks for your quick answer, I try your tip but the problem persist,
so... It seems not to be a dns problem
Asterisk executes the Dial command and it tries to reach the VoIP
provider
of servers), I try to use dnsmgr without solving the isue
Thanks in advance,
Marc
PD: I have used more sophisticate configs using DIALSTATUS variable, but
the problem persists
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command don't wait until timeout when the
provider host is unrechable?
Cheers,
Marc
Adam KOSA wrote:
Marc Patino Gómez wrote:
in most cases it works well but, if my internet connection is down
Asterisk tries to Dial voipprovider, but it can't resolve the dns name,
so it waits 60
support.
Thanks to all,
Marc
Matthew Fredrickson wrote:
Arthur Miller wrote:
The Digium cards are known to steal IRQ's.
The Sangoma cards do not
Not to appear defensive, but that is a technically inaccurate and also
technically ambiguous statement. To correct
card I
will post my feelings and results to the list.
Thanks to all,
Marc
PD: I hope after solve my issue, I will wear my Asterisk t-shirt (with
Digium logo on its back) as same as proud that I wear my Debian t-shirt ;)
Russell Bryant wrote:
Marc Patino Gómez wrote:
I have
Yes.. OUR rollouts work fine, because we use a version of asterisk
that we are comfortable with. However, I'm talking about when we do
consulting for someone who has installed their own asterisk and then
they have some issues with it...
This is the problem to use the last release of
, so now I'm running the same Asterisk config
in other server with the same Digium card and there is no noise in PRI.
Any advice to solve the problem with Dell box?
Regards,
Marc
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Hi Steve,
All my cards are Digium, I tried diferent Digium cards and I had the
same problem.
Regards,
Marc
Steve Totaro wrote:
Marc Patino Gómez wrote:
Hi list,
I have a terrible noise issue with Dell SC1430 + Digium TE110P. The
digium card is not sharing interrupts with any other
Hi Steve,
Thanks for your advice, I will order a Sangoma card and test the box. A
part from this, you know any other point to recomend Sangoma cards
versus Digium cards?
Many thanks,
Marc
Steve Totaro wrote:
That is why I suggested Sangoma. Ask them if you can return it if it
does
Hi John,
thanks for this usefull info
Marc
John Novack wrote:
Marc Patino Gómez wrote:
Hi Steve,
Thanks for your advice, I will order a Sangoma card and test the box. A
part from this, you know any other point to recomend Sangoma cards
versus Digium cards?
Many thanks,
Marc
This 2 line code is doing what I wanted.
exten = 200,1,voicemail(200)
exten = 200,2,Hangup
What I've been told is that they want the 20 year old phone system to
light up the message bulb. (yea, a filament bulb, not an LED) To do
this you pick up on the line that goes into Asterisk and do a:
So I've got a Voicetronix card and it looks like the kernel driver works.
Other than the 0's for ID info.
vpb: Driver Version = 4.0
vpb: major = 251
vpb: tmp [0xfc8fec00] dev-res3 [0xfc8fec00]
vpb: tmp [0xfc8c] dev-res2 [0xfc8c]
vpb: 1WS Write cycle
vpb: Manufactured 00/00/
vpb: Card
,NoOP
exten = _X.,n,Dial(Zap/g1/${EXTEN})
exten = _X.,n,Hangup
The other calls works great, incoming calls and outgoing calls. Any help
will be very apreciated, I'm a newbie doing this kind of asterisk
config, so any advice will be helpful.
Best regards,
Marc
Hello,
In Asterisk 1.4.4, the SIP channel names (SIP//peer/-/id/) do not seem
to be unique. That means that the id associated to a peer is not random.
Is that normal ? Because other asterisk versions give random id for each
generated SIP channels of a peer.
Regards,
- marc
projects:
http://www.voip-info.org/wiki-MediaProxy
Joss.
On 5/11/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote:
Hi all,
I have been using asterisk to do such kind of thing,
But I must admitt, this is not 100 % conveniant (Mainly because Asterisk
isn't a SIP Proxy).
I just wanted to know
Hi all,
I have been using asterisk to do such kind of thing,
But I must admitt, this is not 100 % conveniant (Mainly because Asterisk
isn't a SIP Proxy).
I just wanted to know if you knew/used some kind of SBC or packages which
would deal both with SIP AND RTP !
SER/OpenSER woulc be a good SIP
replacing this one ?
Thanks,
Jean-Marc
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*Jean-Marc Salsa [EMAIL PROTECTED]*
Sent by: [EMAIL PROTECTED]
05/02/2007 07:03 PM
Please respond to
Asterisk Users Mailing List - Non-Commercial Discussion
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To
Asterisk Users Mailing List - Non-Commercial Discussion
asterisk
Hi,
I would like to return different values/cause to another SIP Server with
Hangup cmd.
I tried to put different values in Hangup(xx) ...
but it always returns the same value !
How can I send back different error cause ?
Thanks,
Jean-Marc
registered (or I may be misinterpretting this command.
What can I do to investigate this registration message ? Is there an
special debug command ?
thanks :)
From: Jean Marc Le Fevre [EMAIL PROTECTED]
Date: Wed, 18 Apr 2007 18:14:41 +0200
Hello all,
I'm having a quite simple configuration
Hello and thanks for answering,
As I just answer to Yuan LIU, what I don't understand, is that I can
place an outbound call from asterisk to a gsm at the same time I
can't get asterisk thought a inbound call. But I'll try what you
advice me.
I'll tell you the result of it
Jean-Marc LE
Hello all,
I'm having a quite simple configuration like:
SIP provider = asterisk SIP = lan
Everythings works fine but sometime I can't get incoming call.
here are some of the logs from set debug 25 set verbosity 25 sip show
debug and sip.conf and a part of extension.conf
thanks in advance
-Marc
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