Maybe this happens if you have a short delay to your dns servers. This
could increase the load very fast and after some seconds it might be
over again.
I have installed a dns recurser with own caching on all of my asterisk
servers and now everything runs much more smoothly.
best regards
Am 21.01.13 08:22, schrieb Ding Peng:
Hi, all,
I need remove some header from sip message, such as removing the
privacy:id from receiving INVITE and sending out?
Is there any method to do that?
Thanks in advance.
Ding Peng
Hello,
core show application SipRemoveHeader should do what
Am 31.05.2012 21:37, schrieb equis software:
Anybody knows a good SBC to download?
Thanks
Hello,
it depends on what you want to do with this SBC but asterisk itself can
do some SBC features like simple nat traversal, nat keep alives ...
i also have read about opensbc but i dont know how far
Am 24.05.12 23:46, schrieb bilal ghayyad:
Thanks for all for the help and kindly reply.
One last point that will help me alot:
I am thinking to have 4 Servers running Asterisk and 2 Servers to be for
database. The load to be distributed on the 4 Asterisk Servers with ability
to be
Am 26.04.12 13:23, schrieb Arjan Kroon | Mobillion:
I'm using Bria,but X-Lite from counter path
I have good result with these programs under Lion
I had very good results using jitsi for video calls. maybe its also
worth a look
best regards
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Am 24.04.12 09:27, schrieb Administrator TOOTAI:
Hello,
I opened bug #19763 on jira last friday (20/04) and didn't get any
feedback till now. Is this a normal delay?
Regards
Hi,
i didnt want to say that it is a normal delay but most bug marshals
and devs work on asterisk bugs in their
Am 20.03.12 10:15, schrieb Olivier:
Hi,
I would like to test the following COLP use case :
Alice and Bob are both using a SIP phone registered on a Asterisk 10 server.
Alice dials Bob's extension.
While Bob's phone is ringing, Asterisk updates Alice phone screen with
Bob's name, so that
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Am 13.03.2012 21:13, schrieb Steve Edwards:
On Tue, 13 Mar 2012, Amit Patkar | Avhan Technologies Pvt Ltd wrote:
[Amit Patkar] I completely agree with you on distributing the load. At
the same time, I am looking at juicing hardware as well. Can you share
the number instead of saying couple
Am 23.02.12 07:18, schrieb Goyal, Amit:
Hi,
Can someone share how can I configure asterisk to get P-Associated-Uri header
in 200 Ok to the REGISTER.
Thanks,
Amit
Hello amit,
AFAIK P-Associated-Uri is not supported by asterisk in any version so
you cannot configure it. I even dont think
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Am 07.02.12 12:38, schrieb virendra bhati:
Hi List,
Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What
technology FreeSwitch is used and asterisk don't. I don't know it's the
right or wrong but this question come to my mind...
I had done some load tests with asterisk
some 1000+ CCs on asterisk
server on a regular dev machine with choppy audio on an actual call while
still under stress.
Kindly please confirm your stats.
Regards,
Sammy
On Thu, Feb 9, 2012 at 4:49 PM, Stefan Schmidt s...@sil.at wrote:
Am 07.02.12 12:38, schrieb virendra bhati:
Hi
Am 09.02.12 14:19, schrieb Bryant Zimmerman:
Stefan
This is on target with my configuration I am working on. What kind of
dialplan were you using when running the tests.
Were you doing database lookups or just answering the calls and playing
hold music. Any example would be
Am 09.02.12 16:45, schrieb Patrick Lists:
Iirc a long time ago there was a discussion about load testing by
playing MoH was not a realistic test. Something about all MoH music
getting streamed synchronized so basically Asterisk only has to stream
one file and sorta multiplex that single output
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Am 19.12.11 14:26, schrieb Zoel Hairi:
Hello All,
I have a problem with Fax For Asterisk, the Successful Rate when sending Fax
are very Low especially when we send the Fax just once. Now I’m trying to
modify the dialplan so it will keep trying to send the fax for maximum 5
times at
Am 18.12.11 20:19, schrieb Carlos Rojas:
Hello everybody
I'm setting, heartbeat and asterisk, with rsync, anyone, work them fine?
I've been find any information and saw heatbeat + cysnc2 and heartbeat +
rdbd, any one has worked any these aplications fine?
Best regards
Hello,
I dont
Am 23.11.11 11:39, schrieb Ishfaq Malik:
Hi
How much impact on performance do DONT_OPTIMISE and BETTER_BACKTRACES
have on a busy (13000+ entries in cdr for yesterday) server?
I'm trying to decide whether to have them on in case of crashes or not.
Hi,
IMHO a very big impact. for my system
Am 14.11.11 06:54, schrieb Linux:
I tried to understand the rfc4235 which states the following:
However, using this package to model state for non-
session dialog usages is out of the scope of this specification.
Does this actually mean that the device state of being offline is
Am 27.10.2011 19:29, schrieb Vinod Dharashive:
Hi Richard,
The link is up on 16 th channel. My objective is to have 16 E1 to be
configure on single machine with two 8 port sangoma card. Which is problem I
am facing. Please let me know if you have any solution.
Thanks
Vinod dharashive
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Am 04.10.11 20:40, schrieb Esteban Cacavelos:
someone have been installed Asterisk (Trixbox) on VirtualBox which is
installed on a Linux host (Ubuntu server 10.04 specifically).
I want to know if it is convenient or not, and the reaseons if i should on
shouldn't do it.
Thanks in
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
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 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
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
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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
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 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 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 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 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
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
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
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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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,
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 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
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 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
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
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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
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
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
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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
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
oder
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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
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
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
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,
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
oder
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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
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
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
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
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 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,
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
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
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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
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,
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
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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
apply a fix or the only issue consiste in updating Asterisk ?
Regards,
Adrien
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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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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
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De : asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] De la part de Stefan
Schmidt
Envoyé
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
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
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
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
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
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
qualify=yes
secret=
username=xxx
callerid=bla bla
accountcode=xxx
disallow=all
allow=alaw
allow=ulaw
allow=gsm
host=dynamic
best regards
steve smith
--
Für weitere Fragen stehen wir gerne unter v...@sil.at oder
059944 - 2440 zur Verfügung.
Mit freundlichen Grüssen
--
Stefan Schmidt
Sysadmin
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)
.
best regards
steve smith
--
Für weitere Fragen stehen wir gerne unter v...@sil.at oder
059944 - 2440 zur Verfügung.
Mit freundlichen Grüssen
--
Stefan Schmidt
Sysadmin/VOIP // s...@sil.at // Tel 059944-2440//
-
SILVER SERVER GmbH // Lorenz-Mandl
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
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
--
Für weitere Fragen stehen wir gerne unter v...@sil.at oder
059944 - 2440 zur Verfügung.
Mit freundlichen Grüssen
--
Stefan Schmidt
unter v...@sil.at oder
059944 - 2440 zur Verfügung.
Mit freundlichen Grüssen
--
Stefan Schmidt
Sysadmin/VOIP // s...@sil.at // Tel 059944-2440//
-
SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 //
A-1160 Wien // Fax 059944-9000 // www.sil.at
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