Il giorno 12/mag/10, alle ore 02:59, David Backeberg ha scritto:
I thought setting CallerID like that was for setting callerID on
OUTBOUND calls.
Why on earth would you want to override what's happening on an
inbound call?
What happens if you hairpin it to a local channel, using
Hello list,
when I sent an incoming call first to a queue and after the timeout to a
dial-command, while the correspondent's phone rings there is no ringtone
for the caller...
So it goes like this :
1. dial(SIP/account1,20)
2. queue(myqueue20)
3. dial(SIP/account2)
In step 1 there is
On 12/05/10 09:08, Jonas Kellens wrote:
Hello list,
when I sent an incoming call first to a queue and after the timeout to
a dial-command, while the correspondent's phone rings there is no
ringtone for the caller...
So it goes like this :
1. dial(SIP/account1,20)
2. queue(myqueue20)
hi, all
i want to use PHP agi to do as a soap client. does php agi support
this function?
Thanks!
--
Thanks for your supporting,
have a nice day.
Sucan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Dear all,
using asterisk-1.4.23.1, I encountered a problem of asterisk that cannot
release the channel.* *
We have several of asterisk server(client ,Guest). Now channels remaining
problem occurs only in the server where the number of user agent is more
than 660 and where many simultaneous
I think he need use r option in Dial command, while how I understand in
Queue he need musiconhold.
Dial(SIP/account2,,r)
Vardan
Ishfaq Malik wrote:
On 12/05/10 09:08, Jonas Kellens wrote:
Hello list,
when I sent an incoming call first to a queue and after the timeout to
a dial-command,
Hi all,
I have a demo machine I'm running up on Lenny - it has the packaged
Asterisk version installed (1.4.21.2+stuff).
I'm trying to add an extension to leave a voicemail message, just with
Voicemail(1234), which I've done before (on 1.2 at least), but it's
saying no application 'Voicemail' .
You sure it's not using the URL OPEN parameter for the very queue?
l.
2010/5/11 Carlo Dimaggio jaasmail...@gmail.com
Hi all,
In order to use the open url function of zoiper (it opens an url
based on the asterisk $callerid(dnid)), I need rewriting of the dnid.
In my dialplan I have:
exten
Hi,
I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN (no
NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP. I must be missing something
obvious.
I followed the simple example at
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/.
so
In the queue I need musiconhold indeed, so the 'r'-option is not an
option here...
I did not know there was an 'r'-option for the Dial-command.
However, even with this 'r'-option in the Dial-command, there is no
ringtone for the caller... It just stays silent.
Any other ideas ?
Jonas.
Try so:
1. dial(SIP/account1,20)
2. queue(myqueue,,,20)
3. Ringing
4. dial(SIP/account2,,r)
20 in queue is timeout?
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
Vardan
Jonas Kellens wrote:
In the queue I need musiconhold indeed, so the 'r'-option is not an
option here...
I did not
Hello all,
I have a problem where problem with one way audio, and I think it's
related to a=sendonly and a re-invite. Can anyone please assist?
The scenario is as follows
- We send an INVITE to a peer, and it replies with a 100 Trying, and
then a 183 Session Progress message containing
Hi, I wonder if anyone can help me with a macro issue I have. I need to
set a variable which tells me whether a call has been authenticated
properly. However this authentication is taking place inside of a macro
and I don't want to use a global variable if it will apply to other
channels. I've
Hi,
I still think we've either got a bug in Asterisk or a bug in the
Asterisk::AGI module.
In a separate part of the dialplan we have a call to a (much simpler)
script that begins with the below code.
In the last 1000 calls, I've had a couple of extension not returned by
AGI errors from the
Hi!
I'm trying to setup a SIP trunk between 2 Asterisk servers on the same LAN
(no NAT, no firewalls).
With IAX2 all's fine but I'm unable to setup SIP. I must be missing
something obvious.
Either
a) set a secret and use that on both sides, or
b) look at allowguest= and the default
On Tue, May 11, 2010 at 04:48:30PM +0200, Harel Cohen wrote:
Hi all,
Could someone please tell me what is the relative cost in using conf files
oppose to the astdb? Basically I need to match a name to a phone number in
order to have all users registered by name and not by number (which I
Hello,
I need to store some additional CDR data from the dialplan, like in
example here (down of the page):
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+cdr
However, neither CSV, nor MySQL CDRs have any of these values as the result.
Can you please highlight where can I find the
On 05/12/2010 08:46 AM, David Backeberg wrote:
On Tue, May 11, 2010 at 3:30 PM, William Stillwell (Lists)
william.stillwell-li...@ablebody.net wrote:
Anybody know a reliable fax solution for 1.4.30 branch?
I am using PikaFax on another server and works very well (about 3000 faxes
a
--- On Wed, 5/12/10, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
Either
a) set a secret and use that on both sides, or
b) look at allowguest= and the default context and maybe
the domain=
settings, or
c) use insecure=invite
Thanks Philipp.
I'm trying option
Are there any online training courses , similar to the Asterisk fast start
course, available.
I would really like to take something like the fast start course , but
travelling at this point is out of the question.
Any help or advice appreciated.
Joe
jj...@yahoo.com--
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten = _X.,3,Hangup
Server2:
sip.conf
I will give this a shot and see how well it will work.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, May 11, 2010 3:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Dual PRI using Sangoma DAHDI
Anywhere from 1 to 10 faxes a minute, averaging 2000+ a week..
Zero outbound, all inbound faxing., using about 50 did numbers.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny
Vardan wrote:
Hello
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[callfromserver2]
exten = _X.,1,Noop(Call from server2)
exten = _X.,2,Dial(SIP/${EXTEN})
exten =
And also please show your settings and logs (without debug)
Vardan
Vieri wrote:
--- On Wed, 5/12/10, Philipp von
Klitzingklitz...@pool.informatik.rwth-aachen.de wrote:
Either
a) set a secret and use that on both sides, or
b) look at allowguest= and the default context and maybe
the
Hi All,
I seem to have stumbled on a bit of a problem. When trying to send a fax
with Fax For Asterisk on 1.6.2.x (have tried 1.6.2.5, 1.6.2.7 and the
current svn version, with FFA 1.2 I get a core dump each time.
Here is an extract form the console:
[May 12 22:47:09] DEBUG[22584]:
I have forget to write for outcall in extension
server1:
[calltoserver2]
exten = _X.,1,Noop(Call to server2)
exten = _X.,2,Dial(SIP/interboxserver2/${EXTEN})
exten = _X.,3,Hangup
server2:
[calltoserver1]
exten = _X.,1,Noop(Call to server1)
exten =
I would doubt that anything you could do online (other than working with one
of the on-line asterisk providers), could match the experience of going to
the site and working with equipment. Jared Smith could provide a much
better answer, since this is what he does.
_
From:
Hi!
I'm trying option c) which is the simplest.
used insecure=invite but failed with the same SIP messages.
Tried also insecure=yes but the same messages show up:
SIP/2.0 407 Proxy Authentication Required
Then you have another entry in sip.conf that uses the same IP address.
Delete that,
On Wed, May 12, 2010 at 7:45 AM, Steve Underwood ste...@coppice.org wrote:
On 05/12/2010 08:46 AM, David Backeberg wrote:
So buy an asterisk appliance that supports fax, and then you can pay
somebody else to do the upgrade.
Does that appliance actually support FAX? The web pages don't mention
On 05/12/2010 08:12 AM, Ben Dinnerville wrote:
[May 12 22:47:15] ERROR[22725]: res_fax_digium.c:2114 dgm_fax_start: FAX
handle 0: failed to queue document '/var/spool/asterisk/fax/campaign_70.tif'
[May 12 22:47:15] ERROR[22725]: res_fax.c:834 generic_fax_exec: channel
On 05/12/2010 08:22 AM, David Backeberg wrote:
On Wed, May 12, 2010 at 7:45 AM, Steve Underwood ste...@coppice.org wrote:
On 05/12/2010 08:46 AM, David Backeberg wrote:
So buy an asterisk appliance that supports fax, and then you can pay
somebody else to do the upgrade.
Does that appliance
Hi again!
--- SIP read from 192.168.250.111:5060 ---
SIP/2.0 407 Proxy Authentication Required
You need to run the SIP debug on 192.168.250.111 to learn more about WHY
the 407 is issued. Have a close look and you are likely to understand it
right away.
Also: Do not forget the reload after
El 12/05/10 08:22, David Backeberg escribió:
On Wed, May 12, 2010 at 7:45 AM, Steve Underwoodste...@coppice.org wrote:
On 05/12/2010 08:46 AM, David Backeberg wrote:
So buy an asterisk appliance that supports fax, and then you can pay
somebody else to do the upgrade.
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote:
I have forget to write for outcall in
extension
server1:
[calltoserver2]
exten = _X.,1,Noop(Call to server2)
exten =
_X.,2,Dial(SIP/interboxserver2/${EXTEN})
exten = _X.,3,Hangup
server2:
[calltoserver1]
exten =
Kevin P. Fleming wrote:
Like I said, it's a known problem, and the fix should be out within a
day or two. It was reported to us about a week ago, so if you had
contacted the support department, it's likely they would have been able
to shortcut your hair-pulling experience :-)
Hi Kevin,
--- On Wed, 5/12/10, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
--- SIP read from 192.168.250.111:5060 ---
SIP/2.0 407 Proxy Authentication Required
You need to run the SIP debug on 192.168.250.111 to learn
more about WHY
the 407 is issued. Have a close look
All:
Getting ready to put the system in production.
Any suggestions on stress testing the system? I'd like to initiate
say 10 sip phone calls to make sure the provider has the bandwidth. Can
you do that in CLI? I've called 4 numbers simultaneously with the hard
phones I currently have and
--- On Wed, 5/12/10, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
SIP/2.0 407 Proxy Authentication Required
Then you have another entry in sip.conf that uses the same
IP address.
Delete that, or change the port on one of them, and adjust
insecure=
accordingly.
Please look in any conf file that have any relations with sip.conf.
I think you have some records.
And one also, you take this message when calling in both direction?
(server1 call server2 and server2 call server1)
Vardan
Vieri wrote:
--- On Wed, 5/12/10, Vardanhvarda...@gmail.com wrote:
please show sip show users and sip show peers
vardan
Vieri wrote:
--- On Wed, 5/12/10, Philipp von
Klitzingklitz...@pool.informatik.rwth-aachen.de wrote:
--- SIP read from 192.168.250.111:5060 ---
SIP/2.0 407 Proxy Authentication Required
You need to run the SIP debug on
Hi Vardan
I did same as you told and deleted the SIP information in Astdb and
restarted asterisk. but the result was same.
as you said there might be mistake in sip.conf so i am pasting both servers
configuration here..
1- nasir.server.com
[abc]
username=abc
type=friend
secret=mysecret
nat=yes
Here's one way - set up calls to the sip provider using local channels
instead of actual phones.
In extensions.conf
[monkeys]
Exten = s,1,playback(tt-monkeys)
Exten = s,n,hangup
Create Call file (monkey1.call)
Channel: sip/5551212
CallerID: Local/8
MaxRetries: 1
WaitTime: 60
retryTime: 5
And sip show registry
Vardan
Vieri wrote:
--- On Wed, 5/12/10, Philipp von
Klitzingklitz...@pool.informatik.rwth-aachen.de wrote:
--- SIP read from 192.168.250.111:5060 ---
SIP/2.0 407 Proxy Authentication Required
You need to run the SIP debug on 192.168.250.111 to learn
more about
here i am attaching debug trace of sip in case of sccessfull call when using
register string...
*CLI [May 12 19:21:14]
--- SIP read from 192.168.0.254:5060 ---
INVITE sip:17185594...@nasir.server.com
sip%3a17185594...@nasir.server.comSIP/2.0
Via: SIP/2.0/UDP
If you can call yourself via the provider just setup a dialplan which
spirals the call,e.g. from softphone call via provider one of your
numbers. Then incoming call route to your next DID, and so on, and after
some spiraling just connect the call to the Milliwatt() application.
Milliwatt is
Look, you do again with registration.
remove any registration information.
Look this config, I think it can help you
Server1:
sip.conf
[interboxserver2]
type=friend
host=192.168.250.112
context=callfromserver2
disallow=all
allow=ulaw
allow=alaw
allow=g729
extensions.conf
[calltoserver2]
Regarding functions and applications options, the only authoritative
source is the console:
core show application ...
core show function ...
regards
Klaus
Am 07.05.2010 18:37, schrieb Tim Densmore:
Hi Folks,
Is there a generally accepted Asterisk bible for current versions? I
poked around
Hi again,
below is debug trace of * cli when i remove register string from sip.conf
*CLI [May 12 19:33:06]
--- SIP read from 192.168.0.254:5060 ---
INVITE sip:17185594...@nasir.server.com
sip%3a17185594...@nasir.server.comSIP/2.0
Via: SIP/2.0/UDP 192.168.0.254:5060;branch=z9hG4bK56e3b44a;rport
This code is really ugly und hard to verify.
Please file a bug report at https://issues.asterisk.org/
thanks
klaus
Am 06.05.2010 23:54, schrieb Richard Kenner:
I can confirm that the following fixes my problem:
--- chan_sip.c (revision 261450)
+++ chan_sip.c (working copy)
@@ -10357,12
hi all.
When I use conference call, my setting about jitterbuffer on sip.conf
doesn't work.
### sip.conf #
jbenable = yes
jbforce = yes
jbmaxsize = 100
jbresyncthreshold = 1000
jbimpl = fixed
###
And I understood how to be effective jitterbuffer on conference call.
I have to
Un-top-posting...
Am 07.05.2010 18:37, schrieb Tim Densmore:
Is there a generally accepted Asterisk bible for current versions? I
poked around the forums and there didn't seem to be a real consensus,
and there are lots of options out there. I need something that focuses
on Asterisk
This code is really ugly und hard to verify.
Since the computation of the is being done with separate code from the
actual output, the code in that part of the module is indeed ugly. But I
wanted to make the smallest possible change. However, I do suggest that
the full output string be built
Anyone else get this issue - around 200 entries per second of this in
the Asterisk messages file:
astobj2.c:115 INTERNAL_OBJ: bad magic number 0x27b4113a
Seems to happen after several hours of receiving a steady stream of test
calls.
My messages file is 7.5 gigs...
John
--
You are right that PIKA no longer just sells Fax licenses to be used
with 3rd party boards.
However the PIKA Warp appliance is great for Faxing with Asterisk.
http://www.pikatechnologies.com/english/View.asp?x=1009
Rod
==
On Tue, May 11, 2010 at 3:30 PM, William
Yes, 20 in Queue is timeout... works fine.
Also with the Ringing() command, there is no dialtone... It's just
silence... With or without the r-option, always the same.
When there is no Queue in between the 2 dial-commands, then the ringtone
is there as it should be !
So when I change to
But that can't handle the call volume, and doesn't support (2) 23B+D now
does it?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rod Boileau
Sent: Wednesday, May 12, 2010 11:23 AM
To:
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote:
please show sip show users and sip
show peers
SERVER 2:
sip show users (trimmed to just my sip test trunk):
Username Secret Accountcode Def.Context
ACL NAT
interboxsip
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote:
And sip show registry
sip show registry doesn't list anything regarding my interboxsip test trunk
because I'm trying to setup a straightforward link such as this one described
here (without user/password):
Please change the peers name in any server.
for example:
server1:
interboxsip1
server2:
interboxsip2
Vardan
Vieri wrote:
--- On Wed, 5/12/10, Vardanhvarda...@gmail.com wrote:
please show sip show users and sip
show peers
SERVER 2:
sip show users (trimmed to just my sip test trunk):
Hi,
when i include a sip configuration from another file in my sip.conf
using #include /etc/asterisk/sip-sipgate.conf everything seems to be
working.
The peer is listed when i execute sip show peers and Status is OK.
But the peer is not listed using sip show registry.
I need to place the register
Hi,
i need to match a number with like 03012345678*0 or 03012345*9
I tried _X.*X and _X!*X but both are maching 03012345678 too, ignoring
that *X is required at the end.
The interesting part is that like expected _X*X is matching only numbers
like 1*1 and not 11
Regards
Robert Wagner
What are your allowguest= and domain= settings in the global section of
sip.conf?
And which version of Asterisk exactly are you using?
Philipp
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
On 05/12/2010 01:03 PM, Robert Wagner wrote:
Hi,
when i include a sip configuration from another file in my sip.conf
using #include /etc/asterisk/sip-sipgate.conf everything seems to be
working.
The peer is listed when i execute sip show peers and Status is OK.
But the peer is not listed
On 05/12/2010 01:03 PM, Robert Wagner wrote:
when i include a sip configuration from another file in my sip.conf
using #include /etc/asterisk/sip-sipgate.conf everything seems to be
working. The peer is listed when i execute sip show peers and Status
is OK. But the peer is not listed
Many are having this problem.
goto http://issues.asterisk.org and search for 'bad magic number'
Notably, a few reports have come up in recent days.
Alec Davis
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John Rose
Hi Guys,
Anyone might know why this error keeps showing up and inbound/outbound is
not working on a Bell PRI with Sangoma A101D?
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement
No calls can be made inbound/outbound.
Keeps repeating. No alarms ON and no changes been made to
- bruce bruce bruceb...@gmail.com wrote:
Hi Guys,
Anyone might know why this error keeps showing up and inbound/outbound is not
working on a Bell PRI with Sangoma A101D?
-- Got SABME from network peer.
Sending Unnumbered Acknowledgement
No calls can be made inbound/outbound.
What is wrong with IAX2 protocol?
If IAX2 is so much better than SIP so why providers discontinuing support for
IAX2
I was with provider callwithus but they discontinue IAX2
I switched to checkbox.cc but they discontinued it as well.
What is wrong with IAX2?
--
Joseph
--
Hi,
I'm going abroad shortly and want to be able to dial into asterisk and get it
to call me back so that I can make an outgoing call through my voip provider,
rather than paying crazy international rates.
Can anyone point me in the right direction with regards to the dialplan?
Im using
OK thanks.
Yes I see this is reported in 1.6.0.27 which is where I started seeing
it.
John
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alec Davis
Sent: Wednesday, May 12, 2010 1:52 PM
To: 'Asterisk Users Mailing List -
I should have added, that if you havn't already, please report your senario
with example dialplan etc to one of the open bug reports related to you
problem, otherwise feel free to open a new one.
Also 'many' was a bit strong, should have said 'others'.
Alec Davis
_
From:
Hello,
i'm using asterisk 1.4.9 in fedora 7, i was compiled its with this package:
libpri-1.4.2
asterisk-1.4.9
spandsp-0.0.4
unicall-0.0.5pre1
libmfcr2-0.0.3
libsupertone-0.0.2
libunicall-0.0.3
zaptel-1.4.4
i'm using a E1 pci card with R2 but they not work, when I start the asterisk
What is wrong with IAX2 protocol?
If IAX2 is so much better than SIP so why providers discontinuing support for
IAX2
I was with provider callwithus but they discontinue IAX2
I switched to checkbox.cc but they discontinued it as well.
What is wrong with IAX2?
The same thing that's wrong
--- On Wed, 5/12/10, Philipp von Klitzing
klitz...@pool.informatik.rwth-aachen.de wrote:
What are your allowguest= and domain=
settings in the global section of
sip.conf?
And which version of Asterisk exactly are you using?
I have no such settings defined yet. Still haven't tried to
--- On Wed, 5/12/10, Vardan hvarda...@gmail.com wrote:
Please change the peers name in any
server.
for example:
server1:
interboxsip1
server2:
interboxsip2
If I understand correctly, the peer names can be identical on both servers.
What counts is the host entry, I guess. But then
SIP is just more supported so its easier for the providers to deal with it.
I personally also believe that IAX is not supported by big providers because
if they do so, it'll just make asterisk more famous than they want it to be.
Secondly, as IAX name suggests, it was primarily designed for
On 05/12/10 16:04, Joe Greco wrote:
What is wrong with IAX2 protocol?
If IAX2 is so much better than SIP so why providers discontinuing support
for IAX2
I was with provider callwithus but they discontinue IAX2
I switched to checkbox.cc but they discontinued it as well.
What is wrong with
At 8:04 PM on 12 May 2010, Robert Wagner wrote:
i need to match a number with like 03012345678*0 or 03012345*9
I tried _X.*X and _X!*X but both are maching 03012345678 too, ignoring
that *X is required at the end.
The interesting part is that like expected _X*X is matching only
numbers like
Quoting Alfredo Peña arp...@gmail.com:
Try using this line in the [general] section of sip.conf in your
simulated SIP provider machine:
realm=sip.provider.com
No, that didn't seem to make any difference. However, this did:
insecure=invite
This prevents the Failed to authenticate on
Well, I have managed to get my hands on a copy of 1.2.1 rc1 FFA which
seems to have fixed the core dumping issue but does not appear to have
fixed the issue that was causing the core dump.
We are still getting an issue with a particular file which I have tried
multiple different ways to create
On Wed, May 12, 2010 at 9:53 PM, Ben Dinnerville b...@voicelogic.com.au wrote:
We are still getting an issue with a particular file which I have tried
multiple different ways to create to no avail. The tiff file is created
with ghostscript from a pdf as per the guidlines but every time we try
I already replied to you in the asterisk-r2 mailing list. Your lines are
blocked, the log is telling you that:
[May 12 08:58:43] WARNING[2689]: chan_unicall.c:1034 unicall_call: Make call
failed - Blocked
The only way you get that is if the line is blocked ( rx ABCD bits are 1101
or equivalent
Hi all,
I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1.
It started ok with out cdr_addon_mysql.o. But when I put
cdr_addon_mysql.o in to modules folder, it fail at start and the
following out has been thrown:
--
[r...@localhost modules]# /usr/sbin/safe_asterisk: line 145:
On Thu, 13 May 2010, Pham Quy wrote:
Hi all,
I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1.
It started ok with out cdr_addon_mysql.o. But when I put
cdr_addon_mysql.o in to modules folder, it fail at start and the
following out has been thrown:
--
[r...@localhost
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