hello,
i am trying to set up a asterisk server (version 1.2.26 by now) with
realtime configuration but the user shouldnt register directly to the
server, instead i have set up a ser registration proxy. Everything works
fine so far, but i can´t use the hint feature. Its possible to subscribe
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hello,
you have to use following format in den extension key of the snom:
sip:[EMAIL PROTECTED];user=phone|*7
the |*7 is the extension to dial if you want to pickup the ringing
(blinking) line.
maybe you should try sip:[EMAIL PROTECTED]|*7 where 100 is your hint
extension
and *7100 is a
hello,
why do you build a realtime configuration loading the sip users and
extensions from a database?
if you want to use the username as extension it would be quit simple
looking like this:
exten = _X.,1,Select user from DB which has ${EXTEN}
exten = _X.n,GotoIf(Uservar = ?nouser)
exten =
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[EMAIL PROTECTED] schrieb:
I am new to asterisk and am looking to setup a small office with 4-6 IP
phones and 4 analog lines from the local telco (primary line with HUNT
to the other lines). I am considering purchase of a Digium AEX800.
One of the features that will be important
Shaun schrieb:
Hi All,
This is puzzling me greatly.
The setup: PAP2T over ADSL registers to Asterisk 1.4?using SIP. Attached to
Asterisk are SIP clients. Codec throughout G729 (only have 1 license on
Asterisk server loaded though). When calling the SIP clients from PAP2T I
can't
Adrian Marsh schrieb:
Hi All,
I'm trying to figure out why in the below code, the PSTN_NUM variable is
always amended
exten = s,n,NoOp(${PSTN_NUM})
exten = s,n,ExecIf( $[ ${PSTN_NUM:0:1} != 0 ] $[
${LEN(${PSTN_NUM})} = 10 ]|Set|PSTN_NUM=001${PSTN_NUM})
exten = s,n,NoOp(${PSTN_NUM})
code sends the configured user for this button,
after the code i.E. **201
best regards
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Steve Davies schrieb:
Thanks for that excellent information - Now does anybody know the XML
to provision that field? Normally you take the text on the screen
Call Pickup Code and replace space with underscore
Call_Pickup_Code ua=na *8# /Call_Pickup_Code
Unfortunately Call Pickup Code
which does exactly what you need. sorry but i dont remember the name of it.
best regards
Stefan Schmidt
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hello,
i want to build a pickup extension in a multiuser system, which means
there are several different numbers which same extensions and so on. the
phonenumbers are identified only with their ID in the database like
+123- where after the - is the extension.
the normal pickup function works
Klaverstyn, David C schrieb:
Hi All,
I can not install the asterisk-addons as it thinks there is no
mysqlclient installed. I have installed mysql, mysql-server and
mysql-devel and I am still unable to install the addons. I am running
CentOS 5.2 i386.
Please somebody help.
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Nhadie schrieb:
Hi Sir,
basically I need the public IP, of the user who registered to my
asterisk and made call. users IP address might changed anytime, but need
to record the IP during each call so i can be able to trace it.
Regards
Nhadie
Hello,
we use the following:
exten =
list
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to this mailing list.
so dont know rules and regulation, just trying to post my problem of
voicemail.conf
snipped
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Brent Vrieze schrieb:
Here is what happens:
1. Asterisk verifies connection to the server and we get this. (CLI
output)
-- ast_get_srv: SRV lookup for '_sip._udp.galvatron.vtnoc.net'
mapped to host galvatron.vtnoc.net, port 5060
-- ast_get_srv: SRV lookup for
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- Original Message -
*From:* Sriram mailto:d_r_sri...@hotmail.com
*To:* asterisk-users@lists.digium.com
mailto:asterisk-users@lists.digium.com
*Sent:* Friday, October 10, 2008 6:52 PM
*Subject:* [asterisk-users] Block Caller ID
Hi
Is there
an Spa9xx with Firmware greater thatn 5.x.
which is the same for pap2.
I´ve attached you an complete XML file of an pap2 i´ve found.
best regards
Steve Smith
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Robert Augustyn schrieb:
Hi,
Is that reliable? Any known issues? or recommended setups?
I am planning on adding the spa2002 devices and attaching the terminal
to it.
Will this work well?
Sincerely,
Robert Augustyn
hello,
my expierince with data connections like Modem over voip
.
best regards
steve smith
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Olivier schrieb:
Hi,
I've read in this mailinglist archives some notes related to Linksys
SPA3102 provisioning but I couldn't find there the answer I'm looking for.
Is it possible with this box (mine is unlocked) to store its config
file(s) in a TFTP server, and have this(these) file(s)
qualify=yes
secret=
username=xxx
callerid=bla bla
accountcode=xxx
disallow=all
allow=alaw
allow=ulaw
allow=gsm
host=dynamic
best regards
steve smith
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Ankit Agarwal schrieb:
May 18 14:57:15] WARNING[8314]: rtp.c:2433 rtp_socket: Unable to allocate
RTP socket: Too many open files
[May 18 14:57:15] WARNING[8314]: chan_sip.c:6710 sip_alloc: Unable to create
RTP audio session: Too many open files
[May 18 14:57:15] ERROR[8314]: acl.c:481
hello,
i have a problem with stucked or hanging calls in asterisk 1.4.25
we had this problem before and so we upgradet from 1.2.32 to 1.4.25 but
it still exists and as i could see, happens even more.
on this server there are 1500 clients registered all with qualify on
and we had 2 routing
hello
David Backeberg schrieb:
On Wed, May 27, 2009 at 7:32 AM, Stefan Schmidt s...@sil.at wrote:
i have a problem with stucked or hanging calls in asterisk 1.4.25
only appears on this server and not on the routing server. Even if i
I'm confused. So the server where the calls get stuck has
David Backeberg schrieb:
On Wed, May 27, 2009 at 9:26 AM, Stefan Schmidt s...@sil.at wrote:
Server A call it PBX there are the sip clients connected
A call comes from server B or C to server A and then to a client, gets
stucked on Server A when PSTN side hangs up. On server B or C the call
David Backeberg schrieb:
Now that I better understand your problem, I'm out of ideas.
thats the point where i stand ;)
You are correct that if a BYE sip packet gets lost,
a) it won't get retransmitted if it's UDP
b) the side that's waiting for the hangup will think the call is still active
David Backeberg schrieb:
On Wed, May 27, 2009 at 1:49 PM, Stefan Schmidt s...@sil.at wrote:
all server are in one rack in our datacenter and are connected to an HP
Procurve 2650 switch, which has been setup around 3 months ago, cause of
the old switch died silent in the night.
all server
Alex Samad schrieb:
Hi
Hi Alex,
I am new to asterisk so my suggestions might be a bit silly.
Why not setup a iax2 connection bettween the asterisk servers, because
its a lower overhear and more efficient.
We had changed from iax connections to sip connections cause we had
timing
Deepak schrieb:
Thanks. You are right in assumng that we query the database. I was not
aware that there is a limit to the number of DB connections to mysql.
We open/close db connections as needed. I will check if there is such a
limit and if yes, post the result.
Would you happen to know
Benny Amorsen schrieb:
A regular Dial(somepeer) to a SIP peer which doesn't reply at all seems
to not time out, or at least have a very long time out.
We have a set up where we can dial two different peers, a primary and a
backup peer. If the first one dies completely, so that no error
Danny Nicholas schrieb:
There is a timeout function in the Dial command. The folks who wrote the
command obviously felt that setting a programmatic limit on this would cause
somebody a problem. If you expect a reply from your SIP peer in 30 seconds,
just do Dial(SIP/peer,30) and the line
Benny Amorsen schrieb:
Stefan Schmidt s...@sil.at writes:
What kind of client cant handle one packet per minute without getting a
higher load?
It isn't a client. It handles thousands of connected devices, so it'll
be handling perhaps 50 OPTIONS packets every second if I go the qualify
asterisk xload schrieb:
I have intalled Asterisk 1.4.20.1, it saves the voicemails wavs into a NFS
mounted directory and for an unknow reason all messages over 10 seconds was
recorded incorrectly, but if i save to a local directory works fine.
somebody can help me?
Thanks.
Ernesto
But I wonder why there is a problem with writing recordings to an
NFS mount directly. NFS should easily handle that.
hello philipp,
i dont know why this is a problem with nfs, but i had the same issue
with two servers behind one switch. So i know what helps.
I think that NFS had a problem
Adrien Lemoine schrieb:
Hi all,
To remember, Asterisk runs in version 1.2.7.1 on RedHat AS 4.
Hello,
i am not sure which bug this may be, but i am sure that it has been
fixed since the last 6 years since 1.2.7.1 was up2date.
update to 1.2.31 or newer and you wount have the bug again.
know the bug
reference ?
I'm interested to find the bug report but I don't know how to formulate my
search.
Regards,
Adrien
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Stefan
Schmidt
Envoyé
apply a fix or the only issue consiste in updating Asterisk ?
Regards,
Adrien
-Message d'origine-
De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Stefan
Schmidt
Envoyé : lundi 22 juin 2009 19:19
À : Asterisk Users Mailing
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jonas kellens schrieb:
Is it possible to have several clients behind NAT to register to an
Asterisk-server with a public IP-address ?
When Asterisk receives an incoming call, how will it know @ which
private IP-address the client is reachable ?
I guess it is impossible for Asterisk to
Hello,
This is just a warning, that a snom phone tries to subscribe an
extension which has no hint entry. You should try to find the snom which
has set up the subscription which wasn´t found. You should search in the
snom webif in the function keys for the function nebenstelle or
extension in
with the
${ACCOUNTCODE} which is empty.
I hope someone could see an error i havent found.
Best Regards
Steve Smith
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Hello,
i´ve a question about the Meetme Options. How could i play a enter and
leave sound but without recording the user name first. I just want
something like User joined conference and a User leaved.
With the i or I Option i have to record the name first.
Is there any way of doing this? As i
Oguzhan Kayhan schrieb:
Hi,
I am using asterisk 1.6.0.10
For debugging i set verbosity to 10 with asterisk -vvr..
now i am trying to set it lower but..
when i type asterisk -r it starts with Connected to Asterisk 1.6.0.10
currently running on asterisk1 (pid = 2408)
Verbosity is at
Hello,
iam searching for an Firefox plugin which can make an sip Invite and
Redirect after 200 OK, so i dont have to use a softphone, just to
initialise a call by clicking on a number
i've found some plugins which only works with a softphone installed on
the system but nothing which works good
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Matt Riddell schrieb:
On 5/11/09 9:14 PM, Stefan Schmidt wrote:
Hello,
i use sendjabber notifications when a call is answered to send the
answering user information about the caller also with links to our CRM
or ticket system.
My problem is that i dont know how i can make a link like CRM
Hello,
i have a problem with a Sip trunk to a SAP-BCM PBX.
In and Outbound Calls works fine but when the SAP tries to transfer an
inbound call to an outbound call there is no-way-audio. Two outbound
calls could be transfered without any Problem.
In the sip trace i see that the SAP BCM make
Hello mike,
this feature is only available with an higher firmware for the spa941 (
5.x.x)
You can set this up on the Phone itself or over the Web IF (the USER part).
in the SPA941 its called Call Waiting Service. This would also do what
you want.
Best regards
Steve Smith
Mike A. Leonetti
Hello,
maybe you could find a core dump file mostly in /tmp where you can use
gdb to find which thread has killed your asterisk.
have a look at
http://www.voip-info.org/tiki-index.php?page=Asterisk%20debugging
Backtracing a core dump file in /tmp
best regards
steve
Nitesh Divecha schrieb:
Hello,
i´ve got this when i asterisk has died / killed and was restarted but i
dont have seen that it will collapse then.
i also got this after restarting asterisk from the CLI with restart now.
so dont worry ;)
best regards
steve smith
Danny Nicholas schrieb:
You’d think that this is/was
Hi,
sounds for me like when i use an headset and the microfone handle
scratches on my beard while i talk ;)
maybe you have a network cable whitout screening. I had bad problems on
different phones which sounds like that you have cause of electric or
magnetic inteferences but when i changed
debugging and watch for resend
sip messages.
best regards
steve
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Alejandro Recarey schrieb:
Stefan
How do you dial the users? direct with the peername or something like
ex...@ipofpeer ?
i know this problem when dialing a patton ISDN ata without an extension.
The call is established but when the T1 sip timeout fires the call gets
disconnected. Maybe you
hello,
mike mosier schrieb:
Howdy all
1. does anyone know a good voip / sip / qos monitoring tool?
you could try smokeping or iperf but real monitoring of the dsl quality
isnt easy.
2. Has anyone had luck running asterisk phone systems over DSL?
we dont run asterisk itself over dsl, but
oder
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hello,
which phone do you have behind the pap2 cause the hook flash time
sometimes could be set in the phone and then it will work with the pap2
also.
you should have a look at spaconfig.de (its a german website) but the
default parameters in sip and regional conf, may help you.
best regards
Julien Claassen schrieb:
Hello everyone!
I have a problem with my voicemail. When someone leaves a message - using
googletalk at least - the message file starts silnet, stays that way for a
few
seconds and then is cut short at the end.
The last test we did ended up more than 10
hello,
sounds like a T1 timeout hangupt. The T1 timeout has the default value
of 30 seconds and hangs up a call when for example the 200 OK to the
client doesnt get the ACK back.
you should look at the sip debug of client 3000 maybe you could see that
packets are resend to the client.
maybe
James Lamanna schrieb:
It appears as though the 489 Bad Event response to the NAT keep alive
event responds to the local address, instead of responding to the
NATted address.
This causes Linksys phones to go amber (no registration) after a short
amount of time after placing calls.
Turning
James Lamanna schrieb:
If you've used Linksys phones against recent Asterisk 1.4.x you may
have noticed
that they may drop registration for a quick bit and then go back to being ok
if your phone is behind NAT.
If you turn Asterisk's sip debug information on, you'll probably find
errors like
oder
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Alexander Aksarin schrieb:
Hello, All. I have a problem with receiving fax through T.30. I'm
calling 543 number from fax machine, then start sending fax and fax
machine send document without problem. But asterisk don't receive fax.
I can't find good documentation for app_fax and I'am googled
Rodrigo Lang schrieb:
Good afternoon list.
I'm experiencing a problem with my SIP channel's. When I have an
external connection for one of my SIP carrier's, I can listen to the
client and the client listens to me normally. The problem is when I
will transfer this connection, the call is
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dotnetdub schrieb:
Hi List,
snip
core show channels
Channel Location State Application(Data)
SIP/102--08e1 *...@from-inside Down(None)
SIP/102--08d6 *...@from-inside Ring(None)
SIP/102--08d7
Steve Davies schrieb:
I need suggestions please on how to determine where it is locking, and why.
Many thanks,
Steve
hello,
have you allready tried strace ?
you could just easily start asterisk with this command:
strace asterisk -
or whatever options you want.
maybe you could
Steve Davies schrieb:
On 23 August 2010 18:32, Stefan Schmidt s...@sil.at wrote:
hello,
have you allready tried strace ?
you could just easily start asterisk with this command:
strace asterisk -
Yes, I tried this. Output just stops along with everything else
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Danny Nicholas schrieb:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dario
Quiroz
*Subject:* [asterisk-users] MOH in the middle of the call
Hi, I have a very strange problem. In the middle of the call the MOH
starts for
Am 06.09.2010 00:20, schrieb Gautam Desai:
Can I generate SIP registration and call from Asterisk without a SIP client?
I
need to initiate a call from asterisk and play a recorded message.
Gautam
hello,
have a look at the sip.conf.sample file how to register asterisk as
Hello,
Am 13.09.10 11:56, schrieb Steve Davies:
Hi,
We have a user who is putting large call volumes through Asterisk, and
wants to BLF monitor up to 90 extensions. We are struggling to find a
handset that can keep up with Asterisk :)
1) Is there a handset that will do this?
we only use
Am 07.10.10 10:52, schrieb Steve Davies:
Hi,
snipped
Hello,
i just want to say something about point 4 which comes to my mind about
security.
4) I am not sure whether it is worth dropping through and testing auth
against other peers if there is no username match. Can auth ever
succeed
Am 09.10.2010 20:34, schrieb bruce bruce:
And that is exactly what is done on the device: Nat=yes but Asterisk still
sees the SIP packet coming in to register with a local IP an so it responds
to a local IP which doesn't even exist on the Asterisk network. This is what
frustrates me that it's
Am 10.10.10 15:46, schrieb dotnetdub:
Hi List,
I need to modify the callerID name of the call coming back when a parked
call returns to the extension that parked it when it times out.
Looking at app_parkandannounce.c
/* Now place the call to the extention */
snprintf(buf,
if you dont know someone in china, it would be a good idea to block
this AND set allowguest=no to prevent this in future.
best regards
stefan schmidt
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on sip debug you will see several retransmits for the 200 ok
message which comes at the real beginning of a call (when you answer the
phone) cause the ACK package to this 200 ok could not be received.
same to Bye at the end of a call.
Best regards
Stefan Schmidt
You are missing the point completely. Maybe I did not explain myself
clearly. The problem is that when you connect to the server from
outside the network (Internet), Asterisk does not see the IP address of
the device, it thinks the device is connecting from the IP address of
the
Am 14.10.2010 21:06, schrieb Tim Nelson:
The TCP header is exactly what the NAT changes, no?
--Tim
to the outside yes but not inside.
for example thats how a typical nat table looks like. (its from a zyxel
adsl router with nat)
Nat session
Am 21.10.2010 19:30, schrieb Ricardo Melendez:
Hi to all, I am in the process of setup a new asterisk server, I think in
the HP Proliant ML350 G6 Server (aprox. 100 SIP Users), and Sangoma A102DE
Card.
The specs of the Proliant (HP PART 487932-001) about PCI are the next.
1
Am 21.10.2010 20:03, schrieb sean darcy:
I have a 100MB internal lan. aastra's are wired. asterisk box is wired
next to the switch. But look:
sip show peers
142/14210.10.10.42 D A 5060 OK (137 ms)
144/14410.10.10.44 D
Am 03.11.10 15:14, schrieb satish patel:
Hello Everyone,
We are running asterisk 1.2.x version in production environment since last 5
year and we have no issue at all, But now time to upgrade. and i heard about
1.8 which has introduce many features. I am wondering should I use asterisk
Am 04.11.10 13:14, schrieb Glenn O Larsen:
What often happens, is that most of the peers is getting UNREACHABLE
or Lagged When I try to call during this time, I get a timeout...
Any ideas on where to start debugging?
I'm running on Asterisk 1.4, with realtime users, with cache and
Am 04.11.2010 18:16, schrieb Glenn O Larsen:
Hi Stefan,
Yes, the 1.4-svn works a lot better... Do you have the bug # ? I tried
to find it, but I couldn't locate it.
I'm still able to make the Asterisk not respond (timeout for phones
trying to call) when all clients are subscribing at
Am 17.11.2010 18:06, schrieb Andrew Latham:
John Todd should have a good answer for this. I would start my
estimate at 200,000+ if you are including all of the versions and
types. Software like BigBlueButton includes Asterisk so it can get
confusing real fast.
~
Andrew lathama Latham
Am 24.11.2010 13:48, schrieb Bayardo Sanchez:
My Asterisk is Asterisk 1.2.30.2 currently running on viciserver the problem
is :
Nov 24 06:45:01 WARNING[3335]: channel.c:780 channel_find_locked: Avoided
deadlock for '0x861f6d8', 9 retries!
Nov 24 06:45:01 WARNING[3335]: channel.c:780
Am 01.12.10 05:10, schrieb Duane Larson:
For me OpenSIPS will do most of the work. Asterisk will only handle Hunt
Groups/Queues, IVRs, and Voicemail when OpenSIPS forwards that traffic to
Asterisk. And since I already have MySQL Cluster working in a redundant
fashion I am not sure I want to
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Stefan Schmidt
Sysadmin/VOIP // v...@sil.at // Tel 059944-2440
://www.asterisk.org/hello
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Für weitere Fragen stehen wir gerne unter v...@sil.at oder
059944 - 2440 zur Verfügung.
Mit freundlichen Grüssen
--
Stefan Schmidt
Sysadmin/VOIP
Hello,
what i want to do is to find a way how i can solve the following problem.
we want to offer our customers in the country side also isdn over voip
but we have to use internet connections from another company for this.
This company offers a QoS on this connections but only with 192kbit
Am 05.04.11 20:35, schrieb satish patel:
If i want to watch every phone status Idel or Inuse the how should i use hint
in my dialplan. I meant should i need to specify each and every extension ?
or is there any catch-all extensions ?
-Satish
Hello,
You can use a hint wildcard like
Am 18.07.11 16:15, schrieb Alex Vishnev:
I am wondering if anyone hit this case yet. I am using 1.6.2.19 and doing an
attended transfer. The transfer is going to an outbound number (normally AA
on another IP PBX). the audio on the first transfer is fine. But if the user
requests a transfer
Am 15.09.2011 21:18, schrieb ERIC HERRON:
Asterisk 1.4.26 keeps randomly crashing then restarting itself on my
live production.
I cannot run valgrind and I do not have the right flags set in menuselect.
I can however at the dead of the night run stress tests.
I want
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