I am currently testing a 57i unit. No problems configuring the phone's
config via phone/web UI.
We are trying to avoid using the web UI, the reason is it will take a
long time typing the softkey xml applications URIs on each phone, so
we chose TFTP.
Tried configuring the phone via a TFTP config
I've got a Grandstream 487 in a home-office. The phone-side is working
fine, but the user is complaining that his internet connection keeps
disappearing. The Grandstream is set up as NAT router, and there's
just one PC hanging off the LAN.
Has anyone experienced anything similar?
/Per
How is the billsec field calculated in CDRs?
I have a situation where billsec is being reported as 0 despite the call
being answered and a conversation occurring. An example record follows:
'2007-11-06 21:36:50', '6495566778', '6495566778', '0116495566778',
'1100012_1', 'Local/[EMAIL
I just had a SIP call drop out with the following in the logs and would be
interested to know where to start tracking it down if it recurs.
I'm still running svn trunk due to the experiments I was carrying out earlier
this week to find out whether that port issue had been resolved (which it
In article [EMAIL PROTECTED],
Carles Pina i Estany [EMAIL PROTECTED] wrote:
On Nov/06/2007, Tony Mountifield wrote:
I have a number of systems that have a single Pentium 4 @ 2.8GHz (with HT),
1GB RAM and a 4xE1 PRI card (TE410P), and they regularly have conferences
with up to 90
6 nov 2007 kl. 23.52 skrev CSB:
Asterisk 1.4.12
I wish to extract some custom headers from a SIP REFER message but
am unable to do so. However I can extract them from an INVITE. The
code is:
exten = _.,n,Set(custom-id=${SIP_HEADER(custom-id)}) ;
exten =
I've asterisk stop (presumably segfaulting) a couple of times, and I was
just beginning to look at how to keep it running - what have others
done?
I was thinking of wrapping a script around asterisk like this:
while 1
do
asterisk -f
done
/Per Jessen, Zürich
--
http://www.spamchek.com/ -
Hello,
We're about to deploy an Asterisk system.
So far we have the following (below) configuration
but before we start anything we would like to hear some
suggestions on it, specification or recommendation too...
Thank you!
miniITX board
dual core CPU
2 LAN
1x IDE (or more)
2x SATA (or more)
Sorry guys, I should have already sent such details...so
1. Yes, device is behind NAT (for ram)
2. Bellow is sip.configuration file
[general]
bindport=5060
bindaddr=0.0.0.0
context=invalid-context
musicclass=default
externip=56.236.64.79
allowguest=no
useragent=PBX
maxexpirey=7200
On Wed, Nov 07, 2007 at 11:33:02AM +0100, Per Jessen wrote:
I've asterisk stop (presumably segfaulting) a couple of times, and I was
just beginning to look at how to keep it running - what have others
done?
I was thinking of wrapping a script around asterisk like this:
while 1
-bash: 1:
On 11/6/07, Kim Joung-il [EMAIL PROTECTED] wrote:
Hello!
We are using several Linksys SPA-941 in our office. After IP change occur
devices seems not to be reachable, actually unavailable! Devices is
connected, e.g. we can place a call using SPA-941 but can not receive any
calls...
Why is
Ciao Per,
I've asterisk stop (presumably segfaulting) a couple of times, and I was
just beginning to look at how to keep it running - what have others
done?
I was thinking of wrapping a script around asterisk like this:
while 1
do
asterisk -f
done
Doing so you won't be able to
In article [EMAIL PROTECTED],
Per Jessen [EMAIL PROTECTED] wrote:
I've asterisk stop (presumably segfaulting) a couple of times, and I wasjust
beginning to
look at how to keep it running - what have othersdone?
I was thinking of wrapping a script around asterisk like this:
while 1do
Tzafrir Cohen wrote:
while true
do
asterisk -f
done
And if Asterisk decides to die? If you have a wrong module in
/var/lib/asterisk/module ?
Well, if asterisk decides to die, I want to restart it. A bad module
would be spotted prior to going into production.
You're reimplementing
Andrea Spadaccini wrote:
IMHO it's better to build a FSM (Finite State Machine) that handles
the Asterisk process and other collateral processes (like the MAPI
proxy) and let it monitor the process.
Moreover, you should make this FSM sensible to UNIX signals in order
to start, stop,
Tony Mountifield wrote:
Have a look at the safe_asterisk script, which should automatically be
in /usr/sbin/safe_asterisk. It does this automatically, including
emailing a notification (if you set the NOTIFY variable).
Thanks, I didn't know that script (well, until Tzafrir mentioned it :-)
Per Jessen wrote:
Andrea Spadaccini wrote:
IMHO it's better to build a FSM (Finite State Machine) that handles
the Asterisk process and other collateral processes (like the MAPI
proxy) and let it monitor the process.
Moreover, you should make this FSM sensible to UNIX signals in order
to
Have you looked at your TFTP server logs?
On 11/7/07, Roi Stork [EMAIL PROTECTED] wrote:
I am currently testing a 57i unit. No problems configuring the phone's
config via phone/web UI.
We are trying to avoid using the web UI, the reason is it will take a
long time typing the softkey xml
Per Jessen wrote:
I don't know why it's stopping, but I'm pretty certain it's a segfault.
Next time it happens, I should be getting the core dump.
I'm running 1.4.13, no AGI scripts.
Per,
You should be able to determine if it was a segfault by looking at your
system log. For example, on one
On Wed, 7 Nov 2007 11:47:50 -0500, [EMAIL PROTECTED] wrote:
I am very happy with the Linksys WRT54GS v4 routers and the WRT54GL
(which are both supposed to be the same hardware). Also the Buffalo
WHR-G54S, WHR-G125 WHR-HP-G54S models all running the DD-WRT
firmware of Sebastian Gottschall.
At 02:47 11/7/2007, CSB wrote:
Content-Type: multipart/alternative;
boundary==_NextPart_000_0007_01C82187.BC96F350
Content-Language: en-nz
How is the billsec field calculated in CDRs?
I have a situation where billsec is being reported as 0 despite the
call being answered and a
On Wed, 2007-11-07 at 20:29 +1100, Jason White wrote:
[Nov 7 20:11:01] WARNING[3486] chan_sip.c: Maximum retries exceeded
on transmission [EMAIL PROTECTED] for seqno
212385315 (Critical Response)
This is usually an indication that Asterisk has tried several times to
send an important message
Have you tried a second unit? I don't trust the Grandstream ATA at
all. We only bought 3 but none worked!
On 11/7/07, Per Jessen [EMAIL PROTECTED] wrote:
I've got a Grandstream 487 in a home-office. The phone-side is working
fine, but the user is complaining that his internet connection keeps
Add qualify=5000 in the relevant section of your sip.conf (under the
[6464]) and also make sure the phone is configured NAT Keep Alive
Enable = YES.
On 11/7/07, Kim Joung-il [EMAIL PROTECTED] wrote:
Sorry guys, I should have already sent such details...so
1. Yes, device is behind NAT (for
Hello,
I have some extensions that are using variables loaded by an AGI program.
Everything works fine and I am able to use NoOp to see the value of my
variables when using IAX, but the same variables don't work when using SIP.
I can provide further details, but right off of the bat does is
On Wed, 2007-11-07 at 21:47 +1300, CSB wrote:
How is the billsec field calculated in CDRs?
I have a situation where billsec is being reported as 0 despite the
call being answered and a conversation occurring. An example record
follows:
'2007-11-06 21:36:50', '6495566778',
C F wrote:
Why is it stooping on you? What version are you running? Are you
running any AGI scripts?
I don't know why it's stopping, but I'm pretty certain it's a segfault.
Next time it happens, I should be getting the core dump.
I'm running 1.4.13, no AGI scripts.
/Per Jessen, Zürich
Well, if asterisk decides to die, I want to restart it. A bad module
would be spotted prior to going into production.
We tend to find that on the rare occasions asterisk does decide to die, it very
often doesn't die completely. The asterisk process is still running, but
running asterisk -r
Why is it stooping on you? What version are you running? Are you
running any AGI scripts? I have asterisk running for months unless I
stop it manually it just doesn't die, specific system is with a quad
PRI cards using 3 of them, but just one for a PRI the rest to channel
banks. The other busy
Kim Joung-il wrote:
IP is changing because it is simply an public dynamic IP address, and
our provider change the IP every 8 hours
1) is the phone set up as being behind a NAT router?
2) have you got a STUN server?
I have a couple of SPA-921s in just such a setup with no problems.
/Per
Hi,
I am using OpenSer + Asterisk. I am using a Audiocode MP112 over a
satellite link. The ping time to the server is about 700ms. When
connecting to another carrier there is no delay what so ever. When I
connect it to my test server there is a 3 second delay. From what I
heard my test carrier is
Quick question for those who know the innards of chan_sip:
Does chan_sip use the To: header of an incoming INVITE request,
for anything other than setting SIP_HEADER(TO) ?
As far as I can tell so far, the target extension is taken from the
request URI, i.e. sip:[EMAIL PROTECTED], and the target
On Wed, 2007-11-07 at 02:35 -0800, Kim Joung-il wrote:
externip=56.236.64.79
$ whois 56.236.64.79
OrgName:United States Postal Service.
register = pbx1:[EMAIL PROTECTED]
$ whois 196.222.62.196
descr: --
descr: Temporary
Tzafrir Cohen wrote:
That's even worse. I can imagine what happeens if you actually decided
you wanted to stop Asterisk.
The question was how to start Asterisk from dying not, how do I stop
Asterisk from dying but make sure it doesn't restart when I stop it. In
all seriousness, safe_asterisk
On Wed, Nov 07, 2007 at 07:16:48AM -0500, J. Oquendo wrote:
Per Jessen wrote:
Andrea Spadaccini wrote:
IMHO it's better to build a FSM (Finite State Machine) that handles
the Asterisk process and other collateral processes (like the MAPI
proxy) and let it monitor the process.
Ciao Per,
IMHO it's better to build a FSM (Finite State Machine) that handles
the Asterisk process and other collateral processes (like the MAPI
proxy) and let it monitor the process.
Moreover, you should make this FSM sensible to UNIX signals in order
to start, stop, restart
It should be possible to get the video call over PRI or ISDN and
depending on the codec in theory it could just be throwing packets
into SIP.
On 11/1/07, voip Server asterisk [EMAIL PROTECTED] wrote:
Hi..
Iam new with asterisk PBX, and i have read about asterisk video call.: my
question:
1.
On Nov 3, 2007 9:03 PM, Bert Haverkamp [EMAIL PROTECTED] wrote:
This is generally not possible. The 3G phones (GPRS will be a strech
wrt bandwidth) that do video telephony, do not support any SIP. So the
(...)
Not true - Nokia N95, 3G phone with video telephony, SIP support included.
Makes
I managed to use Cisco IP phones 7911g with asterisk with Sccp and
chan_skinny without any configuration files in tftp. Only settings in
dhcpd to indicate the tftp address and skinny.conf settings. the problem
that I have is that from 8 phones two of them after working a while now are
Paradise Dove wrote:
hi
is there any way to find out that an fxo module is connected to telco
line or not?
not in Asterisk.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update
Thomas Stein wrote:
Hi.
Asterisk 1.4.12.1.
I get a lot of message like this. Someone knows what this message mean? Do i
have to worry about it?
[Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on
channel 'Local/[EMAIL PROTECTED],1' not posted
[Nov 7 15:24:25]
Hi.
Asterisk 1.4.12.1.
I get a lot of message like this. Someone knows what this message mean? Do i
have to worry about it?
[Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free: CDR on
channel 'Local/[EMAIL PROTECTED],1' not posted
[Nov 7 15:24:25] NOTICE[31145]: cdr.c:434 ast_cdr_free:
Hello,
On Nov/06/2007, [EMAIL PROTECTED] wrote:
Just remember if you don't have any Zaptel cards you are going to have
to use ztdummy to run app_meetme. Ztdummy essentially requires Linux
2.6, which you should be using anyways.
yes, and this is the reason that we have setted up a new server
Patrick,
Thank you for your effort, you make me laugh however, IP address, name,
password, are not true details :) so do not pay attention to that part.
Please, can you help me, anything useful - thank you!
Patrick [EMAIL PROTECTED] wrote: On Wed, 2007-11-07 at 02:35 -0800, Kim
Joung-il
That's odd because in my world I *NEVER* have a CDR show ANSWERD and
anything besides 1 billing seconds. Also -- Dave shows up with the
stuff and isn't confused about his name.
CSB -- I'd say the reason you are having this problem is you are
dialing a local channel. Have you tried otherwise?
[EMAIL PROTECTED] wrote:
Have you tried a second unit? I don't trust the Grandstream ATA at
all. We only bought 3 but none worked!
Nope, just the one. It's really a temp solution, so I don't want
to stock up on them. Also, it has worked fine previously, albeit in
a different location.
/Per
On 11/7/07, Tzafrir Cohen [EMAIL PROTECTED] wrote:
I'm not really sure if Callweaver has this limitation or not. But they
did aim at using high-resolution timers from the Linux kernel.
Callweaver does. Asterisk does not. I'm awaiting their next release
its supposed to have proper faxing
IP is changing because it is simply an public dynamic IP address, and our
provider change the IP every 8 hours
Steve Davies [EMAIL PROTECTED] wrote:
Why is the device IP address changing? If the IP address changes, then
the device will need to re-register with Asterisk so that asterisk
knows
On Nov 7, 2007 9:16 AM, Luis Antonio Prata Barbosa wrote:
Hi,
About APs I think DD-WRT firmware is a very good option, and it
could be used in some versions of WRT-54G...
That's all.
as far as QoS for * goes, Tomato offers a very easy to use tool
that works really well, I am able
Per Jessen [EMAIL PROTECTED] wrote:
I've asterisk stop (presumably segfaulting) a couple of times, and I was
just beginning to look at how to keep it running - what have others
done?
Monit http://www.tildeslash.com/monit/ does the job really well. It
certainly isn't a replacement for
Sajith T S wrote:
It certainly isn't a replacement for fixing the root causes of
whatever that makes asterisk die, though.
Completely agree. I intend to have a look at that when it happens next
time.
/Per Jessen, Zürich
--
http://www.spamchek.com/ - your spam is our business.
On Tue, Nov 06, 2007 at 05:04:50PM -0500, Lutgring, Sam wrote:
When I execute a pickup on a ringing phone I get CALL FAILED REASON CODE
603. I am dialing **212 with the following config. Anyone have a
suggestion?
I am not sure, but in the context where your extensions are, have you done:
I have a situation where billsec is being reported as 0 despite the
call being answered and a conversation occurring. An example record
follows:
'2007-11-06 21:36:50', '6495566778', '6495566778', '0116495566778',
'1100012_1', 'Local/[EMAIL PROTECTED],2',
'SIP/64.192.001.001-08893238',
On Wed, 2007-11-07 at 00:08 -0800, Roi Stork wrote:
1) No DHCP, so I manually set the network settings via phone UI.
2) The files aastra.cfg and mac address.cfg are in the TFTP root folder.
3) Restarted the phone.
Here's what I'd do to troubleshoot the problem:
1) First make sure that your
Alvin Austin wrote:
Anyone successfully using the Polycom SoundStation VTX 1000
with Asterisk?
These are analog phones so if an analog phone works, this will also. We
are using them in a couple of our conference rooms and they are working
great! We connect them to asterisk through a channel
Hi,
About APs I think DD-WRT firmware is a very good option, and it could be
used in some versions of WRT-54G...
That's all.
Luis A P Barbosa
2007/11/7, Michael Graves [EMAIL PROTECTED]:
I'd like to survey those on-list who actually use wifi SIP handsets.
What type of wifi access point do
Thanks for the suggestion. Everything was there except for the context
in the Pickup()cmd and that did not fix it.
Watching the cli in debug you can see it dial the **212 and fall
straight through the first step exten = _**XXX,1,Pickup(${EXTEN:2})
and move to the second step exten =
Hi,
...
drivers/isdn/capi/kcapi.c:1014:47: error: macro INIT_WORK passed 3
arguments, but takes just 2
make[2]: *** [drivers/isdn/capi/kcapi.o] Error 1
make[1]: *** [drivers/isdn/capi] Error 2
make: *** [_module_drivers/isdn] Error 2
#+ LOG INFO: pwd:/usr/lib/eicon/divas/src
#!
Anyone successfully using the Polycom SoundStation VTX 1000 with Asterisk?
I can't see any mention of it on the wiki page:
http://www.voip-info.org/wiki-Polycom+Phones
Thanks,
Alvin
___
--Bandwidth and Colocation Provided by
I am very happy with the Linksys WRT54GS v4 routers and the WRT54GL
(which are both supposed to be the same hardware). Also the Buffalo
WHR-G54S, WHR-G125 WHR-HP-G54S models all running the DD-WRT
firmware of Sebastian Gottschall. However the management featureset is
still that of a consumer
To determine what you are going to need, you'll want to start by discussing
what you plan to do with the system, including capacity (e.g. number of
users/extensions, types of calls, etc.). I'd suggest some research,
starting with the following resources which have a lot of good background,
That's odd because in my world I *NEVER* have a CDR show ANSWERD and
anything besides 1 billing seconds. Also -- Dave shows up with the
stuff and isn't confused about his name.
CSB -- I'd say the reason you are having this problem is you are
dialing a local channel. Have you tried otherwise?
We've finaly solved it...
For once, when we plugued the PRI into the first port, all began to work
better.
As for my dial-out into my RAS, I had to put pridialplan=unknown into my
channels group of incoming line, and pridialplan=national in my group for my
pri going to my RAS. That did the
On Nov 7, 2007 11:03 AM, Lutgring, Sam wrote:
Thanks for the suggestion. Everything was there except for the context
in the Pickup()cmd and that did not fix it.
Watching the cli in debug you can see it dial the **212 and fall
straight through the first step exten =
Hello,
Today we setted up a server that needs to use MeetMe but doesn't have
any Zap hardware. So we need to use ztdummy (at least, this was our
idea).
Rarely: zttest is not working at all (100% bad, using zttest -v doesn't
give anything, etc.). Of course, after load ztdummy, there isn't any
Sajith T S [EMAIL PROTECTED] writes:
Per Jessen [EMAIL PROTECTED] wrote:
I've asterisk stop (presumably segfaulting) a couple of times, and I was
just beginning to look at how to keep it running - what have others
done?
Monit http://www.tildeslash.com/monit/ does the job really
I get the same result even if I don't use pattern matching and hard set
the extension like **212,1,Pickup(212). I know that the * is allowed
because I have used it before.
I added the NoOp and saw it execute it from the CLI. I know that it is
executing the extension, I just don't know why it
I have done some thing similar the past with .call files. I had a mysql db
set up with all the calls that needed to go out and the times they needed to
go out. I had several asterisk servers that would poll the data base once a
minute to see if any calls needed to go out at that minute. It
On Wed, 7 Nov 2007, Marek B wrote:
On Nov 3, 2007 9:03 PM, Bert Haverkamp [EMAIL PROTECTED] wrote:
This is generally not possible. The 3G phones (GPRS will be a strech
wrt bandwidth) that do video telephony, do not support any SIP. So the
(...)
Not true - Nokia N95, 3G phone with video
Are all of your settings getting lost, or just some? We've encountered some
interesting bugs in the Aastra's...(tech support said wait for the next
firmware release, for 8 months - and yes there have been firmware releases
in-between).
If your tftp is in fact working, strip you .cfg down to the
Baji Panchumarti wrote:
what happens if you replace the pattern matching expr with _.XXX
Not what you expect, that is for sure! The . pattern MUST be the LAST
character in the pattern. Once Asterisk sees a . in a pattern it
stops looking for any more pattern characters.
According to http://www.voip-info.org/wiki/view/Asterisk+cmd+Pickup and in
particular the REMARK section:
Remark: This command pickups up the dialed EXTENSION, not the ringing
channel/device.
So when the dialplan says;
200,1,Dial(SIP/100SIP/101)
And someone calls extension 200
The call cannot
On our tests using asterisk, some calls have been terminated
abruptely with exact 185 seconds. This is happening with all our
incoming calls from a trunk from 1 of my DID providers ( other
providers or trunks are fine) and I could reproduce it by calling a
queue from my Wengophone
The following two wiki pages explain it much better than I could.
http://www.voip-info.org/wiki/view/Asterisk+Dialplan+Patterns
http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching
I'm not a fan of using the Wiki as a reference, but there really isn't
any info like this in
Here's what I did:
1) Reduced the local config to just network settings.
2) mac.cfg contains network + sip settings.
3) Restarted the phone.
The result was only some sip settings like auth name, user name,
password get updated.
Fields such as proxy ip and registrar ip didn't get updated. I
Use the web interface of the phone to retrieve the config file that you
uploaded. Is it only partially there?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Roi Stork
Sent: Wednesday, November 07, 2007 9:27 PM
To: Asterisk Users Mailing List
Thanks! We checked the TFTP server and there seems to be no problem.
It's up and listening, and looking at the tcpdump and the log there
really was traffic between the phone and the server. We also
successfully downloaded files using another TFTP client.
On Nov 7, 2007 5:33 AM, Jared Smith [EMAIL
One more Polycom IP601 question please (sorry for the long intro here
to document) ...
In order to closely approximate the behavior of the previous telephone
system that many of the users are familiar with, I have set up call
parking like this:
- features.conf [general] section contains:
We tend to find that on the rare occasions asterisk does decide to die,
it very often doesn't die completely.
Agreed. Need to also watch out for SIP deadlocks (asterisk is up, you
can connect to the CLI, but it does not respond to any SIP traffic, or
sip reload, or unload chan_sip, or restart).
Hi everybody,
I have six cisco 7911g connected on asterisk over chan_skinny. Four of them
are working OK. two of them even the screen on the phone is indicating that
is registered and has number loose connection to asterisk . On asterisk the
message is Skinny Client was lost, unregistering.
Matthew J. Roth wrote:
Per Jessen wrote:
I don't know why it's stopping, but I'm pretty certain it's a
segfault. Next time it happens, I should be getting the core dump.
I'm running 1.4.13, no AGI scripts.
Per,
You should be able to determine if it was a segfault by looking at
your
Hi Polycom experts,
I'm having a problem getting changes to the Polycom IP 601's
(mac)-directory.xml file to update the button list on the phone. If
the phone is newly provisioned (i.e. if I Format File System on the
phone) then the new list will show up on the buttons, but of course
this is
On Thursday 08 November 2007 00:13:44 Andre Quintaes wrote:
On our tests using asterisk, some calls have been terminated
abruptely with exact 185 seconds. This is happening with all our
incoming calls from a trunk from 1 of my DID providers ( other
providers or trunks are fine) and I could
On Wed, Nov 07, 2007 at 08:35:14AM -0500, Jared Smith wrote:
This is usually an indication that Asterisk has tried several times to
send an important message to the device, but it hasn't responded, so
we're giving up trying to talk to it. If I were you I'd start by doing
a SIP trace and
On Nov 7, 2007 7:55 PM, Eric ManxPower Wieling wrote:
Baji Panchumarti wrote:
what happens if you replace the pattern matching expr with _.XXX
Not what you expect, that is for sure! The . pattern MUST be the LAST
character in the pattern. Once Asterisk sees a . in a pattern it
stops
Where did you get this CDR? CDRs should look more
like:
http://www.asterisk.org/doxygen/1.2/AstCDR.html
clidCaller ID
src Source
dst Destination
dcontextDestination context
channel Channel name
dstchannel Destination channel
lastapp
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