On Tue, 27 Jul 2010, Kyle Kienapfel wrote:
On Tue, Jul 27, 2010 at 12:50 PM, Roderick A. Anderson
raand...@cyber-office.net wrote:
Anyone tried installing Asterisk in a AWS server?
It probably works as well as it does virtualized other ways. I've seen
peoples opinions on how virtualizing
Travis Langhals tra...@netitek.com writes:
[2010-07-27 10:34:42] DTMF[9744] channel.c: DTMF begin '1' received on
SIP/5211-0078
Is SIP/5211 a Linksys or a Grandstream or something else?
Do you have relaxdtmf=no?
Also, your Asterisk version numbers are incorrect. Do you mean 1.6.2.10?
Randy R randulo2...@gmail.com writes:
I'd think twice about trying this, taking into account the recent
spate of attacks to so many of us coming from Amazon EC2 and
particularly their answer to complaints, which was something like
Deal with it.
Indeed, my personal threshold for dealing with
Update on this - breaktrough! :-)
Finally, I was able to do it. Yes you were right again as you said, I saw
that you mentioned using macros but for some reason I thought that macros
were not available in asterisk 1.2...
So what I managed to do is o start a macro and get info about the
Hi,
My problem is as follows.
I registered an xlite client and dialed 1500 extension. In the
extensions.conf i set as follows.
exten=1500,1,AGI(localhost//
hello.agi.
This hello.agi when connected plays a greeting message. Once this is
connected from the script i want to transfer the call to
Just to mention... I also tried:
$my.query(UPDATE call_log SET local='#{$CHANNEL}', endtime = NOW() WHERE id
= #{call_log_id})
But then the local is empty - meaning $CHANNEL is empty in ruby.
The question is how do i pass that macro dumpchan data to ruby?
Zrko
From:
Hi
I've suddenly started encountering a strange issue. Sometimes, when a
call is made into our system, an extension answered the phone but I can
see no mention of it being bridged in the console. Also, the server does
not seem to think that it is answered and then goes to voicemail. We are
If you run a sip debug at the same time you will get some more usefull
logs.
What sip client are you using?
Ishfaq Malik wrote:
Hi
I've suddenly started encountering a strange issue. Sometimes, when a
call is made into our system, an extension answered the phone but I can
see no mention
On receiving a call, try using the 'Answer()' command before anything else.
I once had some issues, though not similar, which were solved by this
command, as it sends back a SIP acknowledgement to the calling party which
is otherwise not sent.
Zeeshan A Zakaria
--
www.ilovetovoip.com
On
Hi
The problem is that this is a production server with usually about 10
concurrent calls going on and also that if I just run a sip debug on the
customers peer, I still don't know when it's either this issue or if it
genuinely went to voicemail. That's why I'm trying to consistently
Hi
Unfortunately this isn't an option as we allow customers to forward
incoming calls back out to POTS or mobile. If we use an explicit
Answer() all forwarded calls show as answered even if they weren't by
the POTS or mobile end point.
Ish
On 28/07/10 11:48, Zeeshan Zakaria wrote:
On
On 10-07-27 10:02 PM, Michelle Dupuis wrote:
From: asterisk-users-boun...@lists.digium.com
[asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
[leif.mad...@asteriskdocs.org]
Sent: Tuesday, July 27, 2010 9:22 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] Grab
Hi,
I had the following odd behaviour in Asterisk 1.2 - We are migrating
to 1.6, and I will re-test ASAP, though it is quite hard to replicate,
but I am curious to know whether it is a known IAX issue in 1.2.
We had 2 users in iax.conf:
[user1]
username=user1
secret=secret1
context=context1
Thanks to help from Jim Dickenson I managed to start a macro and get info
about the channel that picked up the call from my ruby script.
The only thing that I cant do so far, is capturing the ${CHANNEL} variable
in the ruby script that started the macro.
Is that variable accessible from
Hi!
- upgrade to a current 1.4 version, 1.4.17 is very old (you probably run
this because of the zaptel -- dahdi change, but still)
- do you have a SIP proxy or any SIP-aware hardware in your network
that might play tricks on you, e.g. a SIP ALG (application layer gateway)
on your Internet
Works like charm Danny,
Tested and works fine.
I sent this to Jim, but this is something you could know:
I managed to start a macro and get info about the channel that picked up the
call.
Also, as said in the thread, i was able to start moh beside the macro and it
all works like
Hi!
Three notes:
* as others have already mentioned: personally I would not Dial() from
within AGI using EXEC, but rather set extension and context and then let
the dialplan handle the Dial, and therefore complete that AGI before the
Dial; then possibly run another AGI after the call in the h
Leif Madsen wrote:
I have a client using QueueMetrics and they seem to be fairly pleased
with it. Their response times on issues has been pretty good from
what I can tell (I had the client communicate with them directly
where necessary).
Unless you build it yourself, I'm not sure there
I resolved this isue using odbc.
On Mon, Jul 26, 2010 at 11:27 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Mon, Jul 26, 2010 at 10:05:27AM +0200, Andraž wrote:
Hi,
I have Ubuntu server 10.04 64bit, and Asterisk 1.4.34, compiled from
sources. I installed
Hi,
I try to record a conversation trough AMI (Monitor). In the documentation is
written, that if you would like recordings to be in one file, just use Mix:
1. I use this parameter, but still I have 2 files.
Filename-in.wav and Filename-out.wav.
Regards Andraž
--
Hi,
Since f2b is one of the topics du jour here, I was wondering if
someone would mind telling me what these pf stats mean:
Evaluations: 964303 Packets: 12176 Bytes: 648408 States: 0
Looks like pf examined nearly a million cases from fail2ban in 24h?
thanks,
/r
--
On Tue, Jul 27, 2010 at 6:08 PM, bruce bruce bruceb...@gmail.com wrote:
:-) I knew someone would bring up FreePBX. I have FreePBX installed and it's
not good for Queues at all. It's using the reporting tool from Areski and
One of the several things you asked for was GUI for cdr database logs.
Hi!
I'm working for Zoiper, you can contact us directly on supp...@zoiper.com
Zoa
I will do a test call from a soft phone to my mobile. I can speak into
my headset and the audio is heard instantly. But if I speak into my
mobile there is a 1-2 second delay in the Audio. I am using SIP.
Hi List,
Asterisk 1.4.22 built by root @ carl on a i686
Purely SIP
Linksys SPA962 with 932 sidecar and also Cisco SPA508 / 525G with Sidecars
Have an issue with this happening with a number of my customers.
Customer hits the ringing BLF on the sidecar to pickup the call incoming on
another
On Wed, Jul 28, 2010 at 6:38 AM, Randy R randulo2...@gmail.com wrote:
Hi,
Since f2b is one of the topics du jour here, I was wondering if
someone would mind telling me what these pf stats mean:
Evaluations: 964303 Packets: 12176 Bytes: 648408 States: 0
Looks like pf examined nearly a
On Wed, Jul 28, 2010 at 02:59:23PM +0530, Janu Mukherjee wrote:
Hi,
My problem is as follows.
I registered an xlite client and dialed 1500 extension. In the
extensions.conf i set as follows.
exten=1500,1,AGI(localhost//
hello.agi.
I wonder why you use the odd name 'localhost' here.
hi there,
i have posted earlier on the list but got no satisfying answer. the problem
is not big.
I have asterisk server directly connected with internet (79.80.x.x) and
clients are behind router. clients/users are registered with asterisk and
are using sipura and xlite softphone.
Now problem
hello
i was wondering what is the use of rinstance in SIP Headers. I noticed
that this parameter is visible only when there is NAT invloved.
I am experiencing one way audio when dialing a registered user by his
IP:port. I this case rinstance parameter is missing.
when i dial SIP/username audio
On Wednesday 28 July 2010 06:49:01 Steve Davies wrote:
Hi,
I had the following odd behaviour in Asterisk 1.2 - We are migrating
to 1.6, and I will re-test ASAP, though it is quite hard to replicate,
but I am curious to know whether it is a known IAX issue in 1.2.
We had 2 users in iax.conf:
Do you have your softphone setup to use a stun server so it can send it's
public IP address in the SIP packets? I see in the SIP debug output a 192.168
address for the RTP packets to go to which of course will not work.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
On
On 28 July 2010 17:32, Tilghman Lesher tles...@digium.com wrote:
On Wednesday 28 July 2010 06:49:01 Steve Davies wrote:
[snip] to avoid repetition below
I don't see a 'type' argument to either of the above, so neither of these
would at all be used. That said, you're assuming that the deny
On 07/28/2010 11:32 AM, Tilghman Lesher wrote:
They permit what packets will even reach user2
It should also be pointed out that the config option is permit, and not
allow.
--
_
-- Bandwidth and Colocation Provided by
On Wed, Jul 28, 2010 at 9:03 AM, Kyle Kienapfel doctor.w...@gmail.com wrote:
On Wed, Jul 28, 2010 at 6:38 AM, Randy R randulo2...@gmail.com wrote:
Hi,
Since f2b is one of the topics du jour here, I was wondering if
someone would mind telling me what these pf stats mean:
Evaluations: 964303
Hi
We need to evaluate some open source project that supports 3G-324M on top of
Asterisk.
What do you recommend ? What has been your experience ?
Thanks.
regards,
Anita Hall,
Simmortel.
--
_
-- Bandwidth and Colocation
On Wednesday 28 July 2010 12:18:04 Steve Davies wrote:
When a call arrives from IP address 10.2.3.1 with a username of
user2, then [user2] is used for authentication, but the call
proceeds using [user1] and a channel name of IAX/user1-xxx after
authentication is complete. In the example above
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
We are running asteriskNow 1.4.18 and after a few days it becomes unresponsive
and inbound INVITEs timeout.
We just reboot the box to resolve it. But it seems to be occurring more
regularly now.
I am hesitant to move to latest version, but will do if needed.
Any guidance or troubleshooting
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval
Karihaloo
Subject: [asterisk-users] Asterisk unresponsive
We are running asteriskNow 1.4.18 and after a few days it becomes
unresponsive and inbound INVITEs timeout.
We just
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ujjval
Karihaloo
Subject: [asterisk-users] Asterisk unresponsive
We are running asteriskNow 1.4.18 and after a few days it becomes
unresponsive and inbound INVITEs timeout.
We just
Hi Guys,
I am getting a complain that call on analogue lines (Sangoam A400D) drops
all of a sudden. Here is what I see in logs:
[Jul 28 15:49:08] DEBUG[21438] dsp.c: ast_dsp_busydetect detected busy,
avgtone: 75, avgsilence 135
[Jul 28 15:49:08] VERBOSE[21438] logger.c: -- Executing
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of bruce bruce
Subject: [asterisk-users] Why do Zaptel calls drop all of a sudden?
Couldbusy detect be the problem?
I am getting a complain that call on analogue lines (Sangoam A400D) drops
From: Ujjval Karihaloo
We are running asteriskNow 1.4.18 and after a few days it becomes
unresponsive and inbound INVITEs timeout.
We just reboot the box to resolve it. But it seems to be occurring more
regularly now.
On Wed, 28 Jul 2010, Danny Nicholas wrote:
Assuming you aren’t “around
On 28 July 2010 21:42, Stefan Schmidt s...@sil.at wrote:
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)
Hmmwhat about call waiting?
You mean, when a call comes in on that specific line, it generate two beep
tones and hence the system hangs up thinking it's end of the call?
Interesting!!!
If it is call-waiting do I have to set all of the following off for it to
not give me problem again:
We are running asteriskNow 1.4.18 and after a few days it becomes
unresponsive and inbound INVITEs timeout.
Search this list for DNS.
Philipp
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New
Furthermore, these are lines in Hunt, so, I am not sure if Call-Waiting is
turned ON on these lines at all. But it's definitely an interesting idea.
On Wed, Jul 28, 2010 at 5:54 PM, bruce bruce bruceb...@gmail.com wrote:
Hmmwhat about call waiting?
You mean, when a call comes in on that
Hello.
I'm trying to set TTS with Cepstral and Swift but can't get it to work. I get
this error when testing it:
-- SIP/101- Playing 'welcome.gsm' (language 'es')
-- Executing [...@local-calls:3] Swift(SIP/101-, Hello this is
ceptral) in new stack
[Jul 28 18:29:16]
Good call! I was just reading this thread and was preparing to write a
reply mentioning DNS and SIP channel lockups...
Basically, OP, Asterisk's SIP channels don't like not being able to do
timely DNS queries, so you end up with a very unresponsive Asterisk
server if you don't have local DNS
Manmohan wrote:
I can see the path does exists but i cant see any recordings
happening inn there. There are no files in it
Following is the output:
/var/lib/asterisk/sounds
drwxrwxrwx 2 asterisk apache 4096 Jun 27 20:54 conf-recordings
I hope m understandly this correctly but m sure m
SIP/5211 is a Grandstream device.
Did not add relaxdtmf=no, but sip show settings verifies it's already set to
no.
Fat fingered the version, it should have said 1.6.2.6 through 1.6.2.10
Travis
On Wed, Jul 28, 2010 at 3:12 AM, Benny Amorsen
benny+use...@amorsen.dkbenny%2buse...@amorsen.dk
Sorry, I came into this late...what codec is the device using, and is
the audio being trascoded?
Back at Voxitas, we had a couple of customers complain about random
DTMF tones coming across their line, and Asterisk WAS actually
hearing DTMF tones...want to know what it was?.
In that
On 7/28/2010 6:22 PM, Landy Landy wrote:
[Jul 28 18:29:16] NOTICE[5191]: app_swift.c:304 engine: Text to Speak : Hello
this is ceptral
[Jul 28 18:29:16] ERROR[5191]: app_swift.c:338 engine: Failed to set voice.
Do you have cepstral installed and have the voice(s) registered ?
try: swift
Do you have cepstral installed and have the voice(s)
registered ?
try: swift --voices
asterisk:~# swift --voices
Swift command-line synthesis program
Version 5.1.0 of July 2008
Copyright (c) 2000-2006, Cepstral LLC.
Voice | Version | Lic? | Gender | Age | Language | Sample Rate
Hi Everyone,
This is probably more related to Linux than to Asterisk. Analogue channels
on a system were un-responsive on Monday morning. Apparently something
happened over the weekend and the router went off or it lost it's DSL
connection.
[Jul 23 22:50:01] VERBOSE[12437] logger.c: --
On 7/28/2010 8:33 PM, Landy Landy wrote:
asterisk:/home/landysaccount# grep ^[a-z] /etc/asterisk/swift.conf
buffer_size=65535
goto_exten=no
voice=Marta-8kHz|David-8kHz
afaik, the voice parameter is simply the default voice when not
specified via the swift binary or the Swift asterisk
Jeremy,
Thanks a lot that helped and solved the problem. I had it as: voice=Marta-8kHz
before and that didn't work and now changed it to voice=Marta.
Thanks. I apreciate it.
--- On Wed, 7/28/10, Jeremy Kister asterisk...@jeremykister.com wrote:
From: Jeremy Kister
Just a note, the asterisk mailing list server continually gets
blacklisted over at
http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
mail to spamtraps. Perhaps something needs to be looked into...
Regards,
Sam
--
My guess is on spammers signing up the spamtraps for mailing lists ;)
On Wed, Jul 28, 2010 at 6:45 PM, Sam aster...@net153.net wrote:
Just a note, the asterisk mailing list server continually gets
blacklisted over at
http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
SIP wrote:
what can you do ? simple discard spam don't bounce it.
On 7/28/10 9:45 PM, Sam wrote:
Just a note, the asterisk mailing list server continually gets
blacklisted over at
http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
mail to spamtraps. Perhaps
On 7/28/10 9:45 PM, Sam wrote:
Just a note, the asterisk mailing list server continually gets
blacklisted over at
http://www.uceprotect.net/rblcheck.php?ipr=216.207.245.17 for delivering
mail to spamtraps. Perhaps something needs to be looked into...
Regards,
Sam
Spammers sign up to the
On Wed, Jul 28, 2010 at 6:06 PM, bruce bruce bruceb...@gmail.com wrote:
Hi Everyone,
This is probably more related to Linux than to Asterisk. Analogue channels on
a system were un-responsive on Monday morning. Apparently something happened
over the weekend and the router went off or it lost
Hi Dan,
Following is the output for core set verbose 5, also i am really not able to
get on the admin pin thing? Do you mean, that with admin pin configured we
cant use recording?
LinuxTest*CLI core set verbose 5
Verbosity was 3 and is now 5
== Using SIP RTP CoS mark 5
-- Executing
Also following is what i am putting in lib/define.php
define (RECORDING_PATH, /var/lib/asterisk/sounds/conf-recordings/);
On Thu, Jul 29, 2010 at 9:20 AM, Manmohan Singh Jandu
manmoha...@gmail.comwrote:
Hi Dan,
Following is the output for core set verbose 5, also i am really not able
to get
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