Guys,
I want to replace Cisco 3620 which currently connected to Siemens HiCom
with
Asterisk PBX. As requirement i need to know how to translate the Cisco
configs
to zaptel configs. Only this information i got from Cisco.
controller E1 0/0
framing NO-CRC4
clock source internal
ds0-group 1
Hi,
I am getting this problem when trying to register with Voipfone.co.uk.
It used to work, and I havent changed anything that I know of.
Jan 13 10:22:37 WARNING[21645]:
acl.c:213 ast_get_ip_or_srv: Unable to lookup
'voipfone.co.uk.voipfone.co.uk'
Why does the domain name appear
Nathan C. Smith [EMAIL PROTECTED] wrote:
If I start Asterisk from the command line
(usually, asterisk -c or asterisk -vvvcp ) I can
receive faxes and mailtofax sends them to me OK. If I start
the asterisk service (service asterisk start) that uses
safe_asterisk, faxes appear
# make CC=gcc
In file included from /usr/include/linux/kernelcapi.h:13,
from /usr/include/linux/capi.h:18,
from chan_capi.c:35:
clean compile with kernel 2.6.10 and gcc 3.3.4
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hello there,
any one who used ASTCC in a real enviroment, or has successfully
handled above 1k simultanous calls. need some evalution of ASTCC. if
any one has such an experience please share it with the rest
To handle 1k concurrent calls, you might perhaps need something !=
asterisk. see
I do no get any audio when using voicemail or the playback application.
However, phone to phone works just fine.
I am on a gentoo 2.4.x kernel and have compiled in the latest cvs. The
CLI indicates everything is fine and playing. Nothing in the logs to
indicate any errors either. Simply no
hi
a short test:
- ssh into a box from an ssh window
- asterisk -gvcp
- close window
asterisk box now hangs and doesn't even take a Alt+SysRq+B
is this pseudo realtime? I thought the pseudo realtime threads
allowed others to run as well
Is it possible to have asterisk start with high
On Thu, 13 Jan 2005, Roy Sigurd Karlsbakk wrote:
is this pseudo realtime? I thought the pseudo realtime threads
allowed others to run as well
Is it possible to have asterisk start with high priority, also for I/O
etc without using -p?
The pseudo in pseudo-relatime means that no
The pseudo in pseudo-relatime means that no guarantees are made. A
better
word would be strict priorities. This means that if such a thread
loops
and is busy no processes with lower (or normal) priorities will be
allowed
to run.
You can have a shell running with higher RT priority, that should
Hello Niksa,
are you using OctoBRI ? If yes does it work ok ?
I just switched our company's asterisk box from quadBRI to octoBRI and
I am experiencng problems with our DDI ISDN Lines (works fine with
non-DDI lines)...
Sometimes it just stops working , I get CHANUNAVAIL ...
I am using
Paul Fielding wrote:
- Original Message - From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 12, 2005 11:13 PM
Subject: [Asterisk-Users] Grandstream Bugetone 101 mwi
On Thu, 13 Jan 2005 10:22:52 +0200, David Norton [EMAIL PROTECTED] wrote:
I am getting this problem when trying to register with Voipfone.co.uk. It
used to work, and I haven't changed anything that I know of.
Jan 13 10:22:37 WARNING[21645]: acl.c:213 ast_get_ip_or_srv: Unable to
Debian woody stable, nothing special most of the trouble are paths
On 13/01/2005, at 1:26 AM, Sam Njenga wrote:
Hi
Am setting up * with R2/MfC support but am 90% done. I seem to be
missing
something in my setup. Can you tell me what Linux distribution and the
packages you have used to complete
Hi!
I have two, not too related questions:
- the probably simpler one: if anyone can help me out using a Cisco
7905G with chan_sccp? I did already managed to get it working with a SIP
image, I'd just like to see it work with this one as well. It's probably
something I screw up with the
Hi there
asterisk goes to 90% cpu usage when trying to authenticate a sip friend using
realtime mysql, no other message does appear at cli and asterisk hungs;
here some info:
*CLI realtime load sipfriends name 104
Jan 13 11:52:21 DEBUG[8928]: res_config_mysql.c:109 realtime_mysql: MySQL
Same thing here,
I am using bristuff0.20-RC2b with an octoBRI card.
It only happens with DDI lines. With normal ISDN lines I don't have a
problem.
Which card are you using ?
Remco Barende wrote:
I installed bristuff0.20-RC3 (which includes * 1.0.3 stable)
It works fine until I disconnect the
Hi,
Il giorno mer, 12-01-2005 alle 20:29 +0330, Paradise Dove ha scritto:
just to make sure:
when i have zaptel devices on my box and i also use meetme and iax2,
do i need to have USB device enabled and it's modules loaded?
no, no, no
just usbcore+uhci loaded + ztdummy
Matteo.
--
Matteo
Good day all
We have one Bt-100 that logs on to the server,works for a few min and
then just starts loosing registration
Jan 13 13:10:05 NOTICE[-1101505616]: chan_sip.c:7503 handle_request:
Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.0.145'
Jan 13 13:10:05 NOTICE[-1101505616]:
Two Asterisk machines, different CVS, both say no
application MeetMe, show application does not show MeetMe, when I browse
to /asterisk/apps/ I notice that it is the only app that has not installed?
Do I need to install ZAPRTC first then try to install the
MeetMe application?
I do
inline
On Jan 10, 2005, at 10:12 PM, Christian Peter wrote:
- If I call outside (with Nikotel to German Telekom) there is a remote
hangup after 2 minutes. I've seen other people posting this but nothing
helped. I luckily managed to get around this issue with the following
workaround: The provider
Followed instructions from these old post, CVS updated my asterisk too,
edites makefile... but
--
Get oh323 from
http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/
openh323-Janus_patch4-src-tar.gz
Get pwlib from
I suspect that app_radius has not been updated in a while. That's why you
are getting those errors since Asterisk has been updated a lot since 2004-04-
19.
I am having trouble building appradius from
http://appradius.minitelecom.org/
I configure, make, make install cpprad-1.0, but when I
Hi,
Could someone please give me some advice on how to get * to perform the
following :-
Answer incoming call (from IAX Trunk)
Play a prompt to the caller like Thanks for calling please hold while you
call is transferred to first available operator
The caller hears MusicOnHold
Asterisk begins
Check the used codecs in sip.conf. For playback voicemail-audio files,
gsm-codec is used.
disallow = all
allow = alaw
allow = ulaw
allow = gsm
Hope, this helps.
Guido Hecken
-Ursprüngliche Nachricht-
Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Gesendet: Donnerstag, 13. Januar
Can you tell, which CVS-Version of Asterisk and which firmware of
snom-phones you used? I wasn't able to do this on snom 190 phones. Did you
apply any patches to asterisk, to get this to work?
Thanks for any informations
Guido Hecken
-Ursprüngliche Nachricht-
Von: Altus Snyman
On Thu, Jan 13, 2005 at 01:09:54PM +0200, Kelemen Zoltan arranged a set of bits
into the following:
Hi!
I have two, not too related questions:
- the probably simpler one: if anyone can help me out using a Cisco
7905G with chan_sccp? I did already managed to get it working with a SIP
I'm tring to use the function named sipgetheader in asterisk, but I
downloaded the asterisk version 1.0.3 in which this function doesn't appear.
What the simplier solution to my problem? May I download something
else?
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I have provisioned with iaxy.conf:
snip
What do you see when you run the provisioning prog?
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Hello,
Does anyone know ifzaptel withlibpri(Euro-ISDN implementation for PRI) ofasterisk serveruse HDLCunder LAPD(Q921)?
Regards,
Nauman
Do you Yahoo!?
The all-new My Yahoo! Get yours free!
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We have one Bt-100 that logs on to the server,works for a few min and
then just starts loosing registration
There is one recent firmware version that has a registration fallout
problem. I'm afraid I don't know which version. I have my own problem
I am discussing with GS.
Any firmware after
I can't build it, errors:
chan_zap.c:61: #error You need newer libpri
chan_zap.c: In function `zt_call':
chan_zap.c:1806: warning: implicit declaration of function
`pri_sr_set_redirecting'
chan_zap.c: In function `pri_dchannel':
chan_zap.c:7776: structure has no member named `redirectingreason'
Hello,
I am a Kirk IP600 user too, and I had partial success in getting it to
work with chan_sccp. I changed the line 133 in chan_sccp.c to the
following:
if ( (!s-device) (mid != RegisterMessage mid != AlarmMessage
mid != KeepAliveMessage mid != IpPortMessage)) {
And then I was able to
Cool idea.
Unfortunately I don't know anyone in Mexico City to call
Miguel Cavazos ([EMAIL PROTECTED]) wrote:
Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if
i can fill this 30 channels with REAL traffic for 2 or 3 days I can
find new bugs on chan_unicall or I can
I read somewhere that it works but you must
recompile the kernel...
there is some
info in:http://www.voip-info.org/tiki-index.php?page=Asterisk%20Data%20Configuration
look for a thread with this subject (it has
plenty of info)
[digium.com #12961] T100P as bandwidth
hope this
I have an asterisk system down here in Oaxaca. I don't know anyone there to
call but I can call some hotels
in the area for possible reservations and perhaps ticket information for the
theater.
- Original Message -
From: Miguel Cavazos [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List -
Hello all, I'm trying to figure out if there is
some way I can log agents in and out by just having them call an extension.
Ideally I'd like to have it set up where each agent just dials an extension to
log in and the same one or possibly another one to dial out. I got the logging
in
I read the data sheet on the MAX TNT and didn't see
anything indicating it supports VoIP/SIP have you used it to interface to *
using VoIP??
-- Mike
- Original Message -
From:
TC
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday,
Hi!
Both of your answers helped to some extent thank you. The rest remains up to me
to play a bit with the configurations.
thanks,
Zoltan
Quoting Niksa Baldun [EMAIL PROTECTED]:
Hello,
I am a Kirk IP600 user too, and I had partial success in getting it to
work with chan_sccp. I changed the
I
beleive both are locked into a VOIP carrier (Vontage?)
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]On Behalf Of James H.
ThompsonSent: Wednesday, January 12, 2005 11:54 PMTo:
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
In that example you could make outgoing calls only correct? (since incoming
likely needs port forwards)
I guess the questions becomes how often are you going to do that to justify
the extra $100 or so you going to pay for a wifi sip phone?
Paul Fielding ([EMAIL PROTECTED]) wrote:
I think
Hi,
I still have some trouble with the AGI interface:
- I can use EXEC now, but it never gives me the error returned by the executed
application, if an error occurs
- I can use ANSWER, but I have to put something else behind ANSWER. If I say
ANSWER, I get 510 Invalid or unknown command. If I
Hi,
-Original Message-
Hello all, I'm trying to figure out if there is some way I
can log agents in and out by just having them call an
extension. Ideally I'd like to have it set up where each
agent just dials an extension to log in and the same one or
possibly another one to
Steve Totaro wrote:
Paul Fielding wrote:
- Original Message - From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 12, 2005 11:13 PM
Subject: [Asterisk-Users] Grandstream
Wilson Pickett wrote:
I have provisioned with iaxy.conf:
snip
What do you see when you run the provisioning prog?
___
# iaxyprov 192.168.250.112 iaxy.conf
01:
00 00 00 04
0f:
3d dc 79 12
10:
11 d9
06:
49 41 58 79 .
07:
any feedback would be awsome, the idea is to fill in the 30 channels of
the E1 all at the same time and see how stable it can be
On 13/01/2005, at 8:28 AM, Don Dawson wrote:
I have an asterisk system down here in Oaxaca. I don't know anyone
there to
call but I can call some hotels
in the area
(i skipped over this email at first because the subject is msql not
mysql.)
what version MySQL did you compile against? what version of res_config_mysql
are you running? there was an update to it a few days ago.
-matthew
- Original Message -
From: Maurizio Marini [EMAIL PROTECTED]
To:
Hello,
I read that page, I guess I have to jump into the asterisk database a
little bit more to see what I can accomplish. Thanks for the input.
-Matt
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Craig Waddington wrote:
Two Asterisk machines, different CVS, both say no application MeetMe,
show application does not show MeetMe, when I browse to /asterisk/apps/
I notice that it is the only app that has not installed?
Do I need to install ZAPRTC first then try to install the MeetMe
Has anyone been able to get the Dial Macro Patch applied to the
current CVS stable?
http://search.ebay.com/x100p_W0QQfkrZ1QQfromZR8
I know that this is in the CVS-HEAD, but I need the CVS-stable so that
I can utilize app_suppervaletparking
Thanks in advance,
Brian
I tried to call the mexico city airport and got the following
-- Executing Dial(SIP/9104044010-541d, IAX2/[EMAIL PROTECTED]/57644910
@guest|90.Tf) in new stack
-- Called [EMAIL PROTECTED]/57644910 @guest
Jan 13 10:20:59 WARNING[1142135600]: chan_iax2.c:5339 socket_read: Call
rejected
by
Ronald, it's the context listed in voicemail.conf (I got caught on this
as well)
I really wish Asterisk was better documented; it's bullshit the way it
stands at the moment.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronald
Wiplinger
How do I set asterisk not to answer incoming PSTN POTS calls? I want to
be able to use the line for outgoing calls only.
-Thanks
Tim
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Asterisk-Users@lists.digium.com
On 13/01/2005, at 9:35 AM, Miguel Cavazos wrote:
Really weird calls are still getting in and i just called the same
number as you did. I will investigate.
here is the context on extensions.conf
[guest]
exten = _,1,Dial(Unicall/g1/${EXTEN},90,Tt)
On 13/01/2005, at 9:22 AM, Gary Carr
I would like to apply the app_dial macro patch referenced in:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=2905
To my stable version of Asterisk:
Asterisk CVS-v1-0-12/21/04-14:14:46 built by [EMAIL PROTECTED] on a i686
running Linux.
Mantis has 5 attached patch files. It looks
Depending on what you have the context of this setup for in your
zapata.conf, don't have the following statement in that section of your
extensions.conf.
exten = s,1,Answer
or
whatever the CID of this FXO port is as in
exten = 4792739992,1,Answer
---
Kelly D Griffin
Network Engineer
Tantella
I changed to line to :
exten = _,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN},90,Tt
and it works fine.
- Original Message -
From: Miguel Cavazos [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, January 13,
On January 13, 2005 10:31 am, Tim Lewis wrote:
How do I set asterisk not to answer incoming PSTN POTS calls? I want to
be able to use the line for outgoing calls only.
Put it in a context that lacks an Answer().
-A.
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Asterisk-Users mailing list
I don't know if anyone noticed my post a few months ago on the
asterisk-user mailing list, but I've been writing a queue log parser. I
was wondering if anyone had any queue_logs (the bigger the better) that
I would use as sample data? I would of course be willing to post the
stats up for the
Wow, I hate bad cut and pastes. This should have been:
http://bugs.digium.com/bug_view_page.php?bug_id=0002905
(I guess you all know what I was looking at before :)
On Thu, 13 Jan 2005 at 10:18 Brian S. Adelson ([EMAIL PROTECTED]) wrote:
Has anyone been able to get the Dial Macro Patch
I'd dearly love to be able to give an Asterisk demo by just toting my
notebook, a PC/PCMCIA card, and a couple SIP phones. Is there any way
to do this? Or should I look for a small-profile box with PCI slots,
instead?
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Asterisk-Users mailing list
Kelly Griffin [EMAIL PROTECTED] wrote:
Depending on what you have the context of this setup for in your
zapata.conf, don't have the following statement in that
section of your
extensions.conf.
exten = s,1,Answer
or
whatever the CID of this FXO port is as in
exten =
I am just now investigating Asterisk. Can Asterisk provide 6-10 party
teleconferencing when configured properly?
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To UNSUBSCRIBE or
Kelly, when I tried this it didn't work for me. What ever I tried *
picked up. I know in theory this works, but have you tried it?
On Thu, 13 Jan 2005 09:56:31 -0600, Kelly Griffin
[EMAIL PROTECTED] wrote:
Depending on what you have the context of this setup for in your
zapata.conf, don't have
Hello,
Heres what Id like to do: call my
Asterisk box from a phone, hangup after a few rings, then Asterisk calls me
back and presents a dialtone, than I can dial any valid number in the context
the call originated.
Ive done it with CAPI (thanks to the script on
http://lists.digium.com/pipermail/asterisk-users/2004-December/080417.html
http://www.voip-info.org/wiki-Asterisk+cmd+Dial
Both of the above links outline the same technique.
On Thu, 13 Jan 2005 23:31:07 +1100, RockWater ! [EMAIL PROTECTED] wrote:
Hi,
Could someone please give me some advice
Well that didn't workI now get this error
Jan 12 16:56:21 NOTICE[4989]: app_dial.c:746 dial_exec: Unable to
create
channel of type 'SIP'
== Everyone is busy/congested at this time
-- Executing VoiceMail(IAX2/[EMAIL PROTECTED]:4569/5, b) in
new
stackJan 12 16:56:21 WARNING[4989]:
On January 13, 2005 11:08 am, Patrick Lidstone (Personal e-mail) wrote:
I don't think Kelly's response is correct, at least for TDM FXO boards.
I could not find a way of preventing the FXO board grabbing the line
when it rang, and subsequent enquiries on this list at the time
suggested that it
C F wrote:
Kelly, when I tried this it didn't work for me. What ever I tried *
picked up. I know in theory this works, but have you tried it?
Asterisk will NOT answer the line unless it's told to by using something
like immediate=yes, Answer, Playback, Background, etc. I suspect you
have
On Jan 13, 2005, at 17:12, Matt Burleigh wrote:
I am just now investigating Asterisk. Can Asterisk provide 6-10 party
teleconferencing when configured properly?
Yes Matt, it can ;)
P.S.: Ask Andrew, it's running at ZC
---
Jens Vagelpohl [EMAIL PROTECTED]
Software
Hi, im using asterisk for a voip (sip) solution, so i dont have any
zap/e1/t1 card and works great, but..., i have customers complaining that
the cdr begins the accounting of the call just when the phone is ringing and
not after the user have picked up the phone, i have verified this, is this
You might need to go for [EMAIL PROTECTED] Avery simple and easy to install
version of Asterisk. Just burn the ISO image to a CD and boot with it and it
will automatically install everything for you. However, it will wipe out all
your HD and install CentOS then Asterisk.
For SIP, you can start
Why is SER considered a better SIPserver than asterisk , why is it that SER
can handle more clients than asterisk can. And if this is just cause of say
poor SIP handling code in asterisk then is there anything being done to fix
it. Just wanted to know why SER claims to be better than asterisk as a
On Thu, 2005-01-13 at 16:08 +, Patrick Lidstone (Personal e-mail)
wrote:
I don't think Kelly's response is correct, at least for TDM FXO boards.
I could not find a way of preventing the FXO board grabbing the line
when it rang, and subsequent enquiries on this list at the time
suggested
I do not see any value set in the asterisk database when agents are logged
in our out how does asterisk keep track of agenst that are logged in or
out?
- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Because SER does not process the RTP stream, it just directs it around.
Greg
Vikram Rangnekar wrote:
Why is SER considered a better SIPserver than asterisk , why is it that SER
can handle more clients than asterisk can. And if this is just cause of say
poor SIP handling code in asterisk then is
Hi all,
I have setup my Cisco 79XX phone. Did the tftp, put the config files in the
right location with the right names. Booted my phone, it does the tftp
things,
the screen shows my extensions everything seems fine. However, when I
come offhook and try to dial 11 which is just a playback of
Hello all
Has anyone had any success with the Manager API ?
I am trying to check an extension status without too much luck I have
the following
?php
$fp = fsockopen(127.0.0.1, 5038, $errno, $errstr, 30);
if (!$fp) {
echo $errstr ($errno)br /\n;
} else {
-Original Message-
From: Vikram Rangnekar [mailto:[EMAIL PROTECTED]
Sent: 13 January 2005 16:51
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] SER vs Asterisk for SIP
Why is SER considered a better SIPserver than asterisk , why
is it that SER can handle more
From my (fairly limited) understanding, I think the fundamental
difference is that Asterisk is a pbx (offering all the features
associated with a pbx, voicemail, call transfer, call detail
recording etc) whereas SER is just a sip proxy (albeit a good one).
Therefore Asterisk deals in terms of
Works for me, too. But I found that the Benito Juarez
International airport was reachable by 9-011-52-5-571-3600.
To get this from my PBX-like setup, I have the following in
extensions.conf:
exten = _901152.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:6},90,Tt)
and the following in
I am currently working on a bounty to have queue_logs written directly to
database. It will become open source once finished.
-Matthew
- Original Message -
From: Ben Merrills [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Hi,
-Original Message-
I do not see any value set in the asterisk database when
agents are logged
in our out how does asterisk keep track of agenst that
are logged in or
out?
Yes, that is what I said in my previous mail. Asterisk holds that
information in memory. You have to
Hello,
Can anybody help me with this issue?
-- Called 999302
-- Got SIP response 488 Not
Acceptable Here back from 202.125.154.12
== No one is available to answer at this time
Why am I getting error 488. Im using Sipura SPA-2000
Thanks
David
What can I do next?
You have obviously cycled the power as stated in instructions. I don't
know what else could be wrong.
Could you consider dumping this?
PS: Spam prevention!
Our system is protected with a spam prevention program.
If you send us an e-mail, our system will send you a
you need to look at billsec and not duration in the cdr.
-Matthew
- Original Message -
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, January 13, 2005 10:28 AM
Subject: [Asterisk-Users] Asterisk CDR
Hi, im using asterisk for a voip (sip) solution, so i
From zapata.conf
context=inbound-analog
signalling=fxs_ks
channel = 1
from extensions.conf
[inbound-analog]
;Let fax answer this line
;Only use for outbound calls
---
Kelly D Griffin
Network Engineer
Tantella Wireless
http://tantella.com
479.273.9992 Voice
479.464.8998 Fax
-Original
I'd dearly love to be able to give an Asterisk demo by just toting my
notebook, a PC/PCMCIA card, and a couple SIP phones. Is there any way
to do this? Or should I look for a small-profile box with PCI slots,
instead?
Sure, and it works just fine. I even have my laptop set up to create
Just wanted to catch everybody up. So many people contributed to the
work list that I'd probably miss several if I tried to notify just
them. Many people contributed funds, too, which I greatly appreciate.
We were able to get the entire list done, which added up to two full
hours of work on
Simon wrote:
Hello all
Has anyone had any success with the Manager API ?
I am trying to check an extension status without too much luck I have
the following
?php
$fp = fsockopen(127.0.0.1, 5038, $errno, $errstr, 30);
if (!$fp) {
echo $errstr ($errno)br /\n;
} else {
any feedback would be awsome, the idea is to fill in the 30 channels of
the E1 all at the same time and see how stable it can be
- Original Message -
From: Miguel Cavazos [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Enable debugging to see the reason:
CLI sip debug
quote
A UAS rejecting an offer contained in an INVITE SHOULD return a 488 (Not
Acceptable Here) response. Such a response SHOULD include a Warning
header field value explaining why the offer was
Sorry I forgot to mention that. It's just a cheap ass HFC-S single BRI
card (manufactured by E-Tech). I googled around and I know it can take
some time to recover for the NT1 but I think this doesn't apply for s0.
Even after waiting for 10 minutes I do not get any connectivity but
unloading
[EMAIL PROTECTED]
It occurs to me, do you have the numbered extension set the same as the
context name for the phone in sip.conf? For example, in my sip.conf, the
context names for each phone are [7001], [7002] etc. However, this doesn't
necessarily need to be true. If it's not true, try:
For those who doesnt have an asterisk setup or cant make it work you
can use any iaxsoftphone and use the user guest with no password using
codec gsm and start dialing as if you are in mexico city. We need to
have alot of calls going! The ip for the server is 200.53.121.233
On 13/01/2005, at
Has anyone had any success with the Manager API ?
Yes, lots of success. I think we found that tracking the status of an
extension is the most reliable way to monitor extension state. We do issue
a PeerStatus request at the beginning but that's pretty time and resource
intensive. Since the
Hi all.
I've installed a TDM card, with 1 FXO port. I've configured the zaptel
driver and everything seems to be ok: Asterisk answers the calls. Now,
the problem is that even when the caller hangs up (the caller is my
self from another PSTN line) Asterisk doesn't detect it, and it goes on
You shouldn't need any port forwarding. I've found any SIP phones I've
worked with have happily moved from site to site behind NATs, etc. So I see
no reason to believe that the WiFi phone would be any different - it's just
connecting wirelessly instead of with a wire.
And to answer your
Hello,
Can anybody help me with this issue?
-- Called 999302
-- Got SIP response 488 Not
Acceptable Here back from 202.125.154.12
== No one is available to answer at this time
Why am I getting error 488. Im using Sipura SPA-2000
Thanks
David
if you dial this to reach the airport (using international long distance):
9-011-52-5-571-3600
in extensions.conf
exten = _90115255.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:8},90,Tt)
or
exten = _901152.,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:8},90,Tt)
you should configure the extension in this
I just started getting this error and its preventing me from having any
incomming calls:
chan_zap.c:7542 pri_dchannel: Ring requested on channel 0/2 already in use
on span 1. Hanging up owner.
PRI has been working fine. I didn't know anything was wrong until someone
came and said their DID
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