The best format would be in whatever format asterisk is sending the
final audio out in. Even if you store it in the highest quality asterisk
may have to transcode it on the fly so its best to store it in an
already transcoded format to reduce the cpu load.
For dahdi you would want to use the
Klaus Darilion wrote:
Hi!
Since some time the download of the newest Asterisk does not contains
the version number anymore, but is just called asterisk-1.4-current.tar.gz
This gives me a tarball where I do not know the version without looking
into the tarball.
Thus, IMO it would be
bayardo.sanc...@gmail.com wrote:
This week I was experiencing attacks sip log into my accounts were more than
1,000 requests for records Sip accounts in less than an hour THROUGH deny
the ip of my router access list in cisco and my asterisk server to go through
the iptables drop ip
One of the main benefits of qualify=yes is to detect network problems
with peers.
We send a lot of calls via a service provider using SIP but we have
qualify-yes set so that if it becomes unreachable the dial fails
immediatly without having to wait for a timeout which enables us to
seamlessly
Shariq Khan wrote:
Is there a way skip / ignore the member whose status is busy in the Queue.
I have two channel member in queue and i have set the peer limit 2 for
these members.
I want to skip those member who are currently on the call (answered to
calls) and now their status is busy,
PM, Gareth Blades
list-aster...@skycomuk.com mailto:list-aster...@skycomuk.com wrote:
Shariq Khan wrote:
Is there a way skip / ignore the member whose status is busy in
the Queue.
I have two channel member in queue and i have set the peer limit
2
I cant help you with fixing the actual cause but have you considered
moving the mysql and as much of the associated logic to an AGI running
something like a perl or php script. From previous posts that generally
seems to me the more reliable way of making mysql queries.
Jonas Kellens wrote:
Rough area. Consider yourself lucky you haven't been ripped apart :P
Pete wrote:
I hope someone has helped poor Rob, I would as I am just over the bridge
in Bristol, UK but some evil internet scammer has stolen all my money! ;)
Cheers!
On 15/09/10 12:14, Rob Fugina wrote:
It is with
...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Gareth Blades
Sent: Wednesday, September 15, 2010 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial
Dan Journo wrote:
Hi,
Im looking at using MixMonitor to record calls and I know that I need to
set the filename first.
However, with the number of calls coming in, hard coding the filename
isnt an option.
So I need to do something like this:-
Jonas Kellens wrote:
Hello list,
getting warning : *syntax error, unexpected 'token'*
dialplan :
exten = pbx,n,Macro(CheckNetworkProblems,${custID})
exten = pbx,n,NoOp(status = ${STATUS})
exten = pbx,n,GoToIf($[${STATUS}=congestion]?backup:nocongestion)
CLI :
[Sep 9
Jonas Kellens wrote:
Hello,
in asterisk 1.4.30 all realtime configurations go well.
In asterisk 1.6.2.11 the following appears on CLI :
[Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
MySQL RealTime: Invalid database specified: MyDBase (check res_mysql.conf)
Jonas Kellens wrote:
On 09/08/2010 04:50 PM, Gareth Blades wrote:
Jonas Kellens wrote:
Hello,
in asterisk 1.4.30 all realtime configurations go well.
In asterisk 1.6.2.11 the following appears on CLI :
[Sep 8 16:43:43] WARNING[1843]: res_config_mysql.c:325 realtime_mysql:
MySQL
1) The file is written in real time. Personally I would add a dialplan
entry into the 'h' extension to move the recording into a different
directory when the call ends. That will make your syncronisation much
easier.
Dan Journo wrote:
Hi,
1) I want to create add *1 call recording
The DTMF mode can cause problems. The main rule is to make sure
everything is using the same method. I normally use SIP-Info as the
method as it allows to rtp stream to be switch directly between the two
end points but asterisk still sees all the dtmf digits.
Dan Journo wrote:
1) I want to
Ken D'Ambrosio wrote:
I'd *really* like to be able to have a call ring three different cell
phones; then, if someone answers, they have to somehow acknowledge the
call for it to be directed to them. That way, if one of the phones is
off, or out of range, it doesn't go straight to that phone's
bruce bruce wrote:
Hi Everyone,
I can connect the Jabra GN2124 + GN2100 (smart cord) to the Aastra 53i
receiver port and I get a tone. But when I connect it to the headset
port there is no tone. I am running firmware 2.4 and I can't seem to
find that DHSG, EHS or whatever the setting
Zeeshan Zakaria wrote:
Hi list,
I am planning a migration to virtual machines, and was considering with
it to move from 1.4 to one of the later versions. My and my clients' 1.4
setups have been rock solid and I don't want to put myself into any
unnecessary trouble. Those of you with
Asterisk can convert from wav but it still needs to be in the correct
format. See
http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk
Jonas Kellens wrote:
Hello list,
when putting the class 'default' in comment, then this happens :
[Aug 13 12:36:34]
/asterisk/testing/01Long.wav
2. This does not explain why I can't use class default AND class
whatever.
Jonas.
On 08/13/2010 12:47 PM, Gareth Blades wrote:
Asterisk can convert from wav but it still needs to be in the correct
format. See
http://www.voip-info.org/tiki-index.php?page
We use the Sangoma PRI/E1 cards and they work perfectly. If I wanted a
USB based analogue solution I would go straight for the U100.
Eric Merkel (Mail Lists) wrote:
I am looking to build a small PBX for an office that has 3 incoming
analog lines and less than 10 extensions.
For
Tino wrote:
Hello,
Is it possible to install Asterisk on Vmware(centos) from source. Is
there any difference or disadvantage for this compared to asterisk
running on physical machine.
What version of vmware?
Generally it works but it could be a problem if you require access to
dahdi
'latest version' doesnt really help. There are multiple products.
Tino wrote:
Thanks Gareth for your quick reply.
It is the lateset version and i think i need access to Dahdi interface.
Is there any disadvantages other than this.
On Wed, Aug 11, 2010 at 2:11 PM, Gareth Blades
list-aster
Ron wrote:
Hi,
Is there a way to check on AMI if a user is currently engage on the
phone? i would like to display on my portal whether a user is calling or
not.
thank you
regards
Ron
You could get it to run a command and do 'core show hints' and parse the
result. You will need to
Danny Nicholas wrote:
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Subject: Re: [asterisk-users] AMI Command
Actually, what you probably want is the CoreShowChannels command.
Tilghman Lesher
To second this;
Alejandro Cabrera Obed wrote:
Dear all, I'll install Asterisk 1.4 in an IBM xSeries 226 server with
four HD's available, using CentOS as the OS.
What's the best RAID type recommendation ??? RAID 1 or RAID 5 ???
Regards
Alejandro
Either RAID1 with a couple of spare drives or RAID5
bruce bruce wrote:
Hi Everyone,
I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The
phones occasionally go into No Service mode. The POE switch doesn't
seem to be the problem as it's tested fine. I think the router sometimes
gives up and comes back quickly. Or something
Harel Cohen wrote:
Hi all,
Can the Asterisk do “things” not during a call? For example I would like
to change my dial plan during certain hours\dates or I would like to
check some information in the astdb (e.g. counters of al sort) and
handle it as required and so on. All of this is not
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
Have a look at fail2ban
mosbah abdelkader wrote:
An attacker is scanning my Asterisk Switch to gain illegitimate access
to VoIP call functionality.
Using a sip scanning tool, *it* sends REGISTERs with random identities.
And when it discovers one identity subscribed in my switch, it
Zhang Shukun wrote:
hi,list
Could Asterisk-addson-1.6.2.0 install in 64bit Cent-OS ? after
i make and make install. i cant find the .so file.
is this mean it can't install on 64bit Cent-OS. ps: it works fine on
the 32 bit Cent-OS
Thanks very much!
I have a live system running
Paul Belanger wrote:
On Thu, Jul 22, 2010 at 6:53 AM, Chandrakant Solanki
solanki.chandrak...@gmail.com wrote:
Is anything missing in above configuration or something goes wrong.?
kamailio != asterisk
Wrong list.
Looks like opensips from the code that was pasted.
--
Christian wrote:
Hi all,
Does anyone know any good SIP based provider that offers free calling
within europe for some monthly fee?
Many thanks!
It will always depend on call volume and a fixed monthly fee may not
always be the best value. A lot of ITSPs have a monthly charge almost
that
If you add qualify=yes to the setting in sip.conf it will send a sip
message to the peer every 60 seconds to check if it is alive.
If you try to make a call while the peer is not alive it will fail
immediatly rather than the caller hearing silence while your box waits
for a reply timeout.
Andy
or will that be a waste of time?
Cheers,
Andy
On 20/07/2010 05:42 PM, Gareth Blades wrote:
If you add qualify=yes to the setting in sip.conf it will send a sip
message to the peer every 60 seconds to check if it is alive.
If you try to make a call while the peer is not alive it will fail
amit mehta wrote:
Hello,
I am looking for Voip providers for voip minutes to Mali(South Africa)
Kindly provide the ratesheet for the same.
Regards,
Amit Mehta
AQL - 0.1816 GBP/min
Magrathea high call volume rate - 0.126 GBP/min
They are a couple of UK providers. If it is only that
,
Amit
On Mon, Jul 19, 2010 at 3:42 PM, Gareth Blades
list-aster...@skycomuk.com mailto:list-aster...@skycomuk.com wrote:
amit mehta wrote:
Hello,
I am looking for Voip providers for voip minutes to Mali(South
Africa)
Kindly provide the ratesheet
We are a telco so when we receive calls via ISDN and the number is
witheld we see the correct presentation value but also still see the
actual callers number in the callerid(num) variable.
I am trying to forward some of these calls over to one of our other
boxes via SIP but I have found that
Have you restarted zaptel since making any changes?
You are receiving FCS errors but you dont appear to have crc4 specified
in your span lines.
If you have removed the option but not restarted zaptel yet then do that
to see if it fixes the problem.
Chetan Meshram wrote:
Hi All,
I
Ken D'Ambrosio wrote:
Hey, all. I'm trying to do some fun with auto-provisioning of Polycom
phones, and one thing that would make life easier for me would be if I
could have a per-phone sip.conf file. If not, no biggie -- but if there's
a way to do an include (as per extensions.conf) or
exchange.
Chetan Meshram wrote:
I did restart the zaptel after making changes.. but just to reconfirm I
restarted it again..
but the problem still persists.
On Mon, 19 Jul 2010 16:43:57 +0100, Gareth Blades
list-aster...@skycomuk.com wrote:
Have you restarted zaptel since making any
leonimar cape wrote:
Hi Group,
Is there anyway to force asterisk to use the ip address instead of the
hostname
in the sip via header.
Our client's gateway is using a not FQDN as the hostname of their gateway.
And I
am suspecting that the asterisk is dropping the call because it
Thermal Wetland wrote:
I have a virtual server with godaddy but can not compile DAHDI as it
complains that I do not have the correct kernel source.
The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686:
Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and
latest
bruce bruce wrote:
Hi Guys,
Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i,
and 6730i, but none of them indicate the voic-email. Where should I look
for trouble to find the root issue for MWI?
Thanks,
For each extension in sip.conf I have :-
Jonas Kellens wrote:
Hello list,
this is the setup :
analogue phone -- Grandstream GXW4008 -- Linksys WAG160N --
Asterisk-server (public)
and
Zoiper softphone -- Linksys WAG160N -- Asterisk-server (public)
When calling with an analogue phone + Grandstream GXW and also when
.
On 06/30/2010 11:06 AM, Gareth Blades wrote:
Echo cannot be caused by a router.
The zoipher softphone is probably being used with a headset and I
suspect the microphone is picking up the sounds from the earphones
resulting in echo. Try turning down the earphone volume to see
, Gareth Blades wrote:
Routers wont cause echo. In order for them to do so they would have to
store the outbound voice traffic, delay it and then mix it into the
inbound voice.
Telephones inherently cause echo. For domestic calls the audio path is
normally so short that any echo arrives back so
Kenny Watson wrote:
Hi,
I had a breif telco outage with one of my sip providers.
Is there a way to add failed calls to the cdr aswell as the connected ones?
I was also thinking about having an automated process that monitored
congested calls vs Succesful ones on a carrier and
on this box. I've been
trying to keep things as light as possible.
If I can get congestion into a cdr and have it sending cdr off to a SQL db it
would be ideal.
Thanks
Kenny
Support contact details: supp...@geniusgroupltd.com
- Original Message -
From: Gareth Blades
Jonas Kellens wrote:
On 06/30/2010 12:20 PM, Gareth Blades wrote:
So you get echo when calling from the softphone to the analogue phone?
From softphone to analogue phone is echo.
What if they call a regular telephone number?
Calling to a cellphone number or a fixed number
Jonas Kellens wrote:
Internet Telephony Service Provider = SIP provider. The company that
connects the Asterisk-server via a SIP trunk with the other networks
like GSM, analogue carriers...
Jonas.
By ITSP do you mean a SIP provider?
Thats where I believe the problem lies. You are
your advise and try with a SIP-phone (snom 320).
What do I do if :
1. I also have echo with a SIP-phone ?
2. I do not have echo with a SIP-phone ?
Jonas.
On 06/30/2010 03:52 PM, Gareth Blades wrote:
Thats where I believe the problem lies. You are sending audio to them
of the other calls ??
jbenable = no
I must say I'm not really into these jitter-settings in asterisk. I made
jbenable=yes as it can do no harm...
Jonas.
On 06/30/2010 04:24 PM, Gareth Blades wrote:
Try the SIP phone. If it is better then you might try looking to see
Zhang Shukun wrote:
hi, list
i want to know what is the best OS for install Asterisk 1.6.2.9,
which should work properly on working system.
i want to use CentOS5.2 or CentOS 5.4. Which is better and stable?
Thanks for your help.
Whatever system you go for it should have a long
Zhang Shukun wrote:
hi, all
after a long time development, i need to deploy a production system.
i want to install latest Asterisk 1.6.2.9 on Centos 5.4 . one thing confused
me.
my computer hardware support 64 bit OS.
my question is : should i use Centos 5.4 64bit or Centos
Remco Bressers wrote:
Hi,
The subject says it all. Is it possible to put the IP address of the
peer in the CDR records? Using AGI maybe?
Yes you can either put the information in the userfield if you are using
a plain text file.
If you are storing to a mysql table for example then you
bruce bruce wrote:
Hi Everyone,
I am accustomed to PUTTY and it's very nice as in it allows many many
SSH profiles to be saved and allows tunneling etcbut it's not very
good when it comes to scrolling up and down, colors, text size, and
specially it doesn't give a title to the opened
Remco Bressers wrote:
Hi,
Sorry, but i forgot to notice that i am already using the 'userfield'
column so that's not a possibility. Is there any way i can add the IP
address to a custom MySQL field in CDR? With AGI possibly? The problem
is, that the CDR entry is written in MySQL when the
Rodrigo Lang wrote:
Hello list.
I'm trying to find a way to block any ip that tries to login more than
three times with the wrong password and try to log in three different
extensions. For I have suffered some brute force attacks on my asterisk
in the morning period.
The idea would
Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for
mobile phone calls coming from a GSM Gateway.
All the components are set up in DTMFMODE = RFC2238, and so when the
caller from mobile touches the IP phone LAN extension, the call is
in the GSM
Gateway or everywhere.
Thanks again.
2010/6/28 Gareth Blades list-aster...@skycomuk.com:
Alejandro Cabrera Obed wrote:
Dear all, I have Asterisk 1.4.23 implemented with an IVR wainting for
mobile phone calls coming from a GSM Gateway.
All the components are set up in DTMFMODE
Randy R wrote:
Hi,
I know some of you are very experienced as to the working of
networks. I wondered whether there is some accepted way of determining
bandwidth needs based on the network traffic over time. For example,
looking at the figures for the network traffic through the server
Hugo Serrano wrote:
On 25/06/10 10:44, Gilles wrote:
On Fri, 25 Jun 2010 09:53:34 +0200, Randy Rrandulo2...@gmail.com
wrote:
IMO, if it's a business phone, you'd do well to just reboot it at 3AM
once a week or once a month or some interval that you're comfortable
with. We used to do
Scott Stingel wrote:
Hello-
After configuring DAHDI and starting asterisk, I get the following
message continuously on the Asterisk console:
WARNING[2057]: chan_dahdi.c:4158 pri_find_dchan: No D-channels
available! Using Primary channel 16 as D-channel anyway!
My card is a D410P
Scott Stingel wrote:
On 6/17/2010 2:12 AM, Gareth Blades wrote:
Scott Stingel wrote:
Hello-
After configuring DAHDI and starting asterisk, I get the following
message continuously on the Asterisk console:
WARNING[2057]: chan_dahdi.c:4158 pri_find_dchan: No D-channels
available! Using
.
Regards,
Faisal Hanif
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Tuesday, June 15, 2010 9:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
Deepika Nijhawan wrote:
It just gives no matching peer error and doesn’t pick their sip
configuration, so do not go to any context in extentions.conf.
VERBOSE[3252] chan_sip.c: No matching peer for 'calling number' from
IP:4604'
So the question is why didnt it match anything.
If
Basically it means that one of the messages it received on the PRI D
channel failed the checksum.
I take it that in your span command you have 'crc4' or similar specified
as an option for all of your spans?
If thats the case its probably a faulty port on the card, cable, or a
card in the
Gilles wrote:
Hello
I just read this article and would like some feedback from
experienced Asterisk users:
===
Failed open source VoIP deployment leads to hosted VoIP strategy By
Jessica Scarpati
When budgets are crimped, open source voice over IP (VoIP) solutions
dle...@lstelcom.com wrote:
Hi fellow asterisk users,
I am running Asterisk 1.4.29 with an Digium TE121 card (wcte12xp kernel
module) an approx. 100 snom320. The whole installation is running
without issues since 5 months.
Without having changed anything on the asterisk server for at least
Does anyone know if ANI is supported in the standard version of libpri?
We are currently running the latest asterisk 1.4 but with an older
version of zaptel.
ANI seems to be supported in asterisk 1.4 in that least it is one of the
variables within callerid but would I need to upgrade to the
--[ UxBoD ]-- wrote:
Looking for some help from the UK please. I backed up all my Asterisk
configuration before re-installing the server from 32 - 64 bit.
Unfortunately I did not transfer the backup to another machine!
I now have a TDM400P that is not picking up the line. Can you
Olivier wrote:
2010/5/21 Leif Madsen leif.mad...@asteriskdocs.org
mailto:leif.mad...@asteriskdocs.org
Olivier wrote:
As 1.6.2 seems pretty close to 1.6.1 but I gave 1.6.2 a try but I
met an
issue with BLF-pickup which kept me from going further.
Which bug
Olivier wrote:
2010/5/21 Gareth Blades list-aster...@skycomuk.com
mailto:list-aster...@skycomuk.com
Olivier wrote:
2010/5/21 Leif Madsen leif.mad...@asteriskdocs.org
mailto:leif.mad...@asteriskdocs.org
mailto:leif.mad...@asteriskdocs.org
Vieri wrote:
--- On Fri, 5/14/10, Steve Edwards asterisk@sedwards.com wrote:
I'm supposing my system is using the DAHDI-driven
Digium cards on my
motherboard. I don't know how hardware timers work and
if Digium
hardware rely on the motherboard (my system clock is
going too fast
Klaus Darilion wrote:
Am 17.05.2010 10:46, schrieb Zhang Shukun:
Hello,
you know , when a call setup, either caller hangup first or callee
hangup first , the hangupcause will set to 16(means Call Clearing
Causes)
My question is how could i identify whether the caller or callee
hangup
Show the details on the active channels when using both methods and
check what codecs are being used.
Vieri wrote:
Hi,
I have an audio quality problem regarding IAX2. I have 2 Asterisk servers
interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).
One trunk is SIP and the other
There should be no noticeable difference between slin, ulaw and alaw so
what you have is fine. The problem must be elsewhere.
Vieri wrote:
--- On Thu, 5/13/10, Gareth Blades list-aster...@skycomuk.com wrote:
Show the details on the active
channels when using both methods and
check what
I am running asterisk 1.6.2.6 and have configured hints for our
extensions and have a couple of Aastra 6755i test phones. The phones
register fine but 'core show hints' shows the lines as idle even if they
are in use.
I read the wiki and see mention about needing to set call-limit in
asterisk
Richard Kenner wrote:
I read the wiki and see mention about needing to set call-limit in
asterisk 1.4 but that has been depreciated in 1.6 so what is the way it
should be done in 1.6?
I set
callcounter=yes
in sip.conf.
Thanks that works perfectly.
--
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running on watermelon (pid = 10365)
watermelon*CLI sip show registry
Host
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me credentials to register to.
Connected to Asterisk 1.6.2.5 currently running
Mike A. Leonetti wrote:
On 05/07/10 12:14, Gareth Blades wrote:
Mike A. Leonetti wrote:
On 05/07/10 11:52, Gareth Blades wrote:
Mike A. Leonetti wrote:
In an attempt to connect our Asterisk 1.6 phone system with another
phone system called Broadsmart, they gave me
In my previous company we bought about 30 Grandstream GXP2000 phones.
The build and design quality of those phones were terrible (not to
mention firmware bugs).
Speakerphone and headset ports were unusable.
The external powersupply would only last a year or two before it failed.
The screen was
I have got call recording working on our 1.4.30 asterisk box together
with a recording pause ability and being able to play different audio to
each party at the start and end of the pause. This all works perfectly
but one wish is to have the audio files have a beep or something in them
so when
Ignore me I figured it out. The dangers of copy and paste.
After looking through the code line by line I noticed the 'b' parameter
to monitor(). Fine to use before the dial command but shouldnt be used
when a call is in progress.
Gareth Blades wrote:
I have got call recording working on our
Try this.
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2},,rtT)
Peter Gelencser wrote:
Hi,
I need a feature from asterisk with dahdi channels, if there is an
incoming call, it should ring on several dahdi channels.
My channels look like:
typo ...
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}${OFFICE2},,rtT)
Gareth Blades wrote:
Try this.
OFFICE1=DAHDI/13
OFFICE2=DAHDI/14
exten = 12345678,1,Dial(${OFFICE1}{OFFICE2},,rtT)
Peter Gelencser wrote:
Hi,
I need a feature from asterisk with dahdi
Can you post a sip debug
Tarek Sawah wrote:
Greetings List.
I'm facing a strange issue with one of my providers.. after sending an INVITE
request my server places the call on hold.. until the call is answered..
this is happening only with this provide although i have 3 other providers i
I have upgraded Asterisk from 1.4.22 to 1.4.30 and I have noticed I am
getting a lot of errors like this on the console :-
ERROR[23912]: utils.c:968 ast_carefulwrite: write() returned error:
Broken pipe
I have tracked it down to a perl AGI script which performs our own CDR
recording. It is
Danny Nicholas wrote:
Check out this snippet from Tilghman Lesher (one of the true Asterisk
Guru's)
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg220482.html
Thanks but that appears related to AMI not AGI.
--
Philipp von Klitzing wrote:
Hi!
Why is asterisk so slow in sending the call info via STDIn in these cases?
Is there any way this can be fixed?
Your AGI script is faulty: In at least one place you have missed to READ
the output right after you have issued a command. So go check your
Danny Nicholas wrote:
Can you post the script?
Yes private stuff is in a separate file. $mode=start works fine but
answered and completed cause the problem.
I dont know if it is a problem with teh AGI script or just the newer
asterisk reporting it as an error. It doesnt effect functionality
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Wednesday, April 28, 2010 9:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.4.30 is slow sending STDIN to AGI
script
Danny Nicholas
Steve Edwards wrote:
On Wed, 28 Apr 2010, Gareth Blades wrote:
The script does not issue any commands. The same script is called at all
3 stages but with different parameters on the command line to indicate
the call status. Works fine before the call is answered but during and
at the end
Danny Nicholas wrote:
Darn, that should have worked. The improvement from 1.4.22 to 1.4.23+
basically requires that every print STDOUT line be followed by a STDIN
to make util.c not choke when doing commands/setting variables. I wonder
how this rewrite would work?
sub set_variable
{
Steve Edwards wrote:
On Wed, 28 Apr 2010, Gareth Blades wrote:
The script does not issue any commands. The same script is called at
all 3 stages but with different parameters on the command line to
indicate the call status. Works fine before the call is answered but
during and at the end
Steve Edwards wrote:
Steve Edwards wrote:
How do you reconcile your assumption that the Perl module is reading
STDIN and your statement that your AGI quits before asterisk has
finished sending the information about the current call via STDIN.
On Wed, 28 Apr 2010, Gareth Blades wrote
You can use the hangupcause variable which us the pri cause code
supplied when a call is ended over a PRI line. For example this is the
maco we use to dial a number over PRI.
[macro-pridial]
exten = s,1,GotoIf($[${ARG1:0:2} != 00]?noint)
exten =
We currently have the Grandstream GXP-2000 phones which generally work
very well except that we cannot get find a headset which works reliably
with them. Either the sound quality is poor or the other party has
difficulty in hearing us.
We therefore want to get a couple of different phones and
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