Leo Ann Boon wrote:
Kyle Gordon wrote:
Hi all,
I'm currently on an NTL PSTN line, using Asterisk 1.2 and a X101P
cheapo card.
The problem lies with detecting when the far end has hung up. It fails
to detect it, and will only cleardown when the silence timeout has
been reached. Now, I've
I had this problem and in the end it appeared to be slot timing on the mobo. I
had to put the TE110P in the 1st slot - which happened to be a PCI-X slot.
That was using a Supermicro motherboard too.
Regards
Lee
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
Hello List
I am having a rather big problem with a sangoma A104 card, I just installed to
replace a Digium TE410 card, that was acting up.
But now we have a problem with the sangoma card. It runs great after being
started, and calls proceed as normal, but after about 1 hour, it stops
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
E.g: because you have a valid PID file of the controlling process. If
you actually want to kill it, you can.
And you don't need physical access to the system to get to the one and
only real console. OTOH, if you do have physical access,
On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
Hello List
I am having a rather big problem with a sangoma A104 card, I just installed
to replace a Digium TE410 card, that was acting up.
But now we have a problem with the sangoma card. It runs great after being
started, and calls proceed
hello,
i want to make a dialplan where i can pickup calls to an extension when
there are internal and external calls.
i want to use only one prefix for pickup both situations so there is a
plan how to check if the incoming call is an internal call or an
extern???
regards rene
I am running the newest version, from the sangoma wiki.
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: 26. januar 2007 10:56
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sangoma card
Ok this is a simple question...
What has been your experience with the WellTech 38xx series (I'm looking
specifically at the 3802)
VoIP gateway? I'm looking for a good (and hopefully not too expensive)
VoIP/T.38 gateway for my office.
Asterisk intergration is not a major factor at this time
Hello everybody i am Ashish here.
i am new to this mailing list.
so dont know rules and regulation, just trying to post my problem of
voicemail.conf
Actuallt right now i am using Asterisk 1.2 on my LAN environment.
i am able to call all my extension very nicely.
Right now i am trying to
Why there is no asterisk.conf.sample file?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
winmail.dat___
--Bandwidth and
Which asterisk versions etc etc?
On 1/26/07, Jon Schøpzinsky [EMAIL PROTECTED] wrote:
I am running the newest version, from the sangoma wiki.
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: 26. januar 2007 10:56
To: Asterisk
it is in doc/ directory
asterisk-conf.txt
Tomislav Parčina wrote:
Why there is no asterisk.conf.sample file?
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
imho, ci$co doesn't support anything other than callmanager as signaling
server :-(
Peter Mitchell wrote:
79X1 phones now come bundled with licences - and I can't find a separate SIP
licence like the old 79x0 models.
Whats the non callmanager - SIP licence number for 79X1 ?
Could it be related to this? http://bugs.digium.com/view.php?id=8507
Did you try to contact Sangoma Support? Their replies are prompt.
Regards, Alex
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: Friday, January 26, 2007 12:03 PM
To:
hello,
maybe you should try adding this to your voicemail configuration.
mailcmd=/usr/sbin/sendmail -t
or whereever your sendmail is located.
then your mails should be send to the wanted adress.
Best regards.
Stefan
Ashish Barot schrieb:
Hello everybody i am Ashish here.
i am new to this
Hi,
I have Asterisk 1.2 + Adit600 Channel bank(which gives analog output and
also takes PSTN lines into the Asterisk system).
Conversations recorded by the ASTERISK comes in two separate Files:
xx.0-in (GSM Audio) for the Asterisk Extension Side of the
conversation;
xxx.0-out (GSM
Hi all,
I dont have any problem, my asterisk is working fine. but on the cli,
asterisk keeps saying Got SIP response 603 Declined (no dialog) back
from 192.168.0.100. trixbox running on another machine is registered to our
server from address 192.168.0.100. whats the reason of this msg?
--
Thanks for your answers :)
I use another server to test digium board and my config, and it works
well, so... I think the problem is between chipset, Intel 5000P and
digium card. I will try to put the digium board in other PCI-X slots,
and change some timing PCI parameters in the BIOS.
On 25 Jan 2007, at 06:57, Brad Templeton wrote:
On Mon, Jan 22, 2007 at 09:59:06AM +, Tim Panton wrote:
In the meanwhile, use IAX, which understands about NAT pretty well.
If you have multiple SIP phones on a LAN behind a NATing router, just
put a small asterisk box on the LAN. It can
On 1/25/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Thursday 25 January 2007 6:30 pm, David Gomillion wrote:
I mean that I would like to have a system in place so that Asterisk, as
a
privileged service, can gain access to Courier's IMAP storage. Having to
keep track of all of our users'
I have had no trouble but I record in .wav format and it automatically
mixes it together if I use the *1 or the mixmon app.
on Friday 01/26/2007 [EMAIL PROTECTED]([EMAIL PROTECTED]) wrote
Hi,
I have Asterisk 1.2 + Adit600 Channel bank(which gives analog output and
also takes PSTN lines
On Friday 26 January 2007 7:42 am, David Gomillion wrote:
I'm not talking about setting the voicemail password. I'm talking about
not having to put my users' email passwords in the voicemail.conf file.
Asterisk, if I understand correctly, needs each user's email password to
deliver the
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2 is one of
the easiest configs to put together. Works extremely well and requires
opening a single port on each NAT.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Thursday,
Hi Asteriskers,
I have the following :
exten = 1,1,Playback(sample)
exten = 1,2,Read(response,,1)
exten = 1,3,GotoIf($[${response} != *]?300:100)
exten = 1,100,Playback(hello)
exten = 1,101, [[[ do stuff ]]]
exten = 1,300,Playback(reject)
exten = 1,301,Hangup
Which plays a confirmation sample,
Anyone experience ring oddities with extensions.conf rollovers? Let me
summarize...
One of my extensions.conf file is built to ring during the day, ring/go
to voicemail after a certain time:
[main-aa]
exten = s,1,GotoIfTime(17:00-8:30|mon-fri|*|*|*?main-night-aa,s,1)
exten =
Does anyone know of a wireless 802.11 sip phone with an auto answer mode?
THanks,
Jerry
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Background *does* do what I need, so problem solved.
Thanks to fenlander on #asterisk-uk for the help :)
--
Tony
_
MSN Hotmail is evolving check out the new Windows Live Mail
http://ideas.live.com
Asterisk is version 1.2.14, zaptel 1.2.12, libpri is whatever version was with
zaptel 1.2.12 :)
Jon
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Davies
Sent: 26. januar 2007 12:03
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hi,
I would like to know what is Asterisk Dynamic Loader. Let me
explain what I'm about to ask.
I have three Asterisk servers running my in-house built app_xyz.so
application. Now what I do to save time is compile application on one
server and scp app_xyz.so on rest of servers. All servers have
Has anyone successfully installed spandsp and rxfax and txfax applications on
1.4.0 release of Asterisk?
I tried the latest snapshot of spandsp, as well as couple other previous
versions. I compiled it fine, downloaded the asterisk.patch, manually patched
the asterisk files, run .configure,
On 26 Jan 2007, at 10:43, Ashish Barot wrote:
Upto this moment the voicemail is generating, but it is not e-mail
to any email id. But it comes on [EMAIL PROTECTED]
[...]
[worldbiz]
exten = _111X,1,Dial(SIP/${EXTEN},4)
exten = s-BUSY,2,Goto(s,1)
exten = _111X,2,VoiceMail([EMAIL PROTECTED])
AT the risk of being rude with a follow up of the same information and a
top post, change the AMDSTATUS of AMD_PERSON to HUMAN. The example does
not work, if you look at the source for AMD you will see that the status
returned is:
This application sets the following channel variable upon
On Thu, 25 Jan 2007, Yuan LIU wrote:
Thanks for this information. Does this mean two IAX boxes can talk behind
their respective NAT's (without any server sitting in voice path)? I'm
imagining this:
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2
Using IAX, yes. It's quite
Olle E Johansson wrote:
24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling:
James Fromm wrote:
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
interface from another SIP device and immediately hanging up will
Problem solved :) :)
I change the OS, I install a Debian Etch x86_32bits and it works
perfectly with the following software versions:
asterisk 1.2.13
zaptel 1.2.12
so... I don't understand where was the problem. TE110P driver
version
The previus OS was a Ubuntu 6.10 (codename
cb wrote:
On Jan 25, 2007, at 5:38 PM, Leif Neland wrote:
A Grandstream GXW-4108 with 8FXO cost $400, almost the same as a
TDM404B fully populated 4FXO card.
I'm currently testing a GXW-4108... my verdict is still out. I've had
some problems, some minor, some major.
In the minor
Gordon Henderson wrote:
On Thu, 25 Jan 2007, Yuan LIU wrote:
Thanks for this information. Does this mean two IAX boxes can talk
behind their respective NAT's (without any server sitting in voice
path)? I'm imagining this:
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2
Using
Moises,
I 've stated testing by raising all timers a bit. Everything went
worse, now there are more failed calls. Can you give me a hint of
which timers to modify and, if you know, a more extensive explanation
of each one? I know it's documented into the file but I cannot catch
the concept.
Hello everyone!
I 've looking for carriers which can terminate my international calls.
They must accept payments from Argentina and give me interconection to
my Asterisk. I'd appreciate your help or recomendations.
Regards.
--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088
Share your
Just for giggles can you set an absolute timeout in the dialplan for
all calls in and out of that span?
On 1/25/07, kjcsb [EMAIL PROTECTED] wrote:
I have a situation where the two Zap channels on a TDM400 are staying
offhook after a random period of time; it is not (I believe) related to the
Hi:
I am working in a VoIP Carrier Company, I could provider you the service
for your internationals calls.
Please visit
www.fonetglobal.com and call me, my phone number is +52 442 167 08 00
x214 Rafael Canchola.
Thanks.
At 09:54 a.m. 26/01/2007, Facundo Ameal wrote:
Hello everyone!
I 've
On 1/26/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: C F [EMAIL PROTECTED]
Cory, it's called dialplan magic it realy depends what PBX it is, not
all of them allow dial plan magic. But it is possible on most pbxes.
CF: What exactly is diaplan magic? I googled but found little info.
Did you
Anybody else having trouble with hangup detection on the TDM2400 FXO
modules? It works most of the time, but sometimes I get hung lines in
Voicemail. And only with the TDM24xx not with the TDM400 or Adit 600
CB. Anybody else have this?
___
--Bandwidth
On 26 Jan 2007, at 06:19, Yuan LIU wrote:
From: Brad Templeton [EMAIL PROTECTED]
I have a really dumb question. It appears that Yahoo, MSN, AIM,
you name
them, they don't have a NAT problem, and some use SIP. I don't
think they
all stay in voice path, either. What takes?
When you
On 25 Jan 2007, at 01:40, Bastian Schern wrote:
Hello Asterisk fans,
I try to connect an analog modem to Asterisk. The modems are connected
e.g. to alarm systems or a cash terminals (POS). As PSTN-Interface I'm
using a Wildcard TE110P (E1).
Is it possible to connect the modems to an ATA?
check register expiration on polycom , probably is higher than 3600 sec
(default on asterisk) , so after this 3600 , imagine polycom as an expire of
6000sec, there's a gap of 2400sec that polycom isn't registred!
On 12/10/06, C F [EMAIL PROTECTED] wrote:
While what you say might/should help,
try to use sipura SPA2102. it has T38 and works well with WellGate 5250 adn
cisco 53xx on other end.
Hope this helps...
Vlad
- Original Message -
From: Mark Coccimiglio [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, January 26, 2007 3:27 AM
Subject:
26 jan 2007 kl. 16.31 skrev James Fromm:
Olle E Johansson wrote:
24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling:
James Fromm wrote:
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
interface from another
On Jan 26, 2007, at 10:39 AM, Drew Gibson wrote:
You can get the option numbers and values from the source html of
the web page. (I am assuming the GXW-4108 works the same as other
Grandstream products)
I'll try that out, thanks!
I did see a thread on another forum mentioning the HTML
Hi all,
I would like to make a script/program that would be able to show lots of
status information from my analog FXO lines (and FXS lines in the near
future).
Example of interesting status information:
- Hook status: is there a call being made with that zap?
- Voltage status: cable
Hi list
I conncte my Panasonic KX-TCD705TW cordless phone to TDM400P FXS module.
When the system boots or reboots, the LED on the backlit of TDM400P OFTEN
gets off and dmesg shows problem with FXS module, as follows
Zapata Telephony Interface Registered on major 196
Zaptel Version: 1.4.0 Echo
I am getting the following compile error on h323.
Its an old redhat 7.3 system with asterisk 1.2.14, zaptel 1.2.12
pwlib 1.5.2 and openh323 1.12.2
I have pwlib compiled and installed.
I have openh323 compiled and installed.
I went in the channels/h323 directory and did make opt
What shall I
Andrew Joakimsen ha scritto:
I know of the call pickup issues but what asterisk issue and what BLF
issue?
On 1/25/07, Alberto Pastore [EMAIL PROTECTED] wrote:
Andrew Joakimsen ha scritto:
Actually I noticed just three days ago there is a new release, and the
releae notes seem to address
I am trying to set callerid from a PHP script, using one of two functions as
shown below (setid1 and setid2). The first function works great with
regular names and numbers, BUT, if I call the function with
(Test,UnknownNumber), the cid number gets set to asterisk. Why is my
passed number
Hi All,
Asterisk on IBM NEBS compliant Blade Server sounds great.
There is some information at
http://www.voip-info.org/wiki/view/Asterisk+hardware#IBMNEBScompliantBla
deServerforTelcoappli
I couldn't find further details on this, Have some one used this ? or
have any details on
From:"Ken Williams" [EMAIL PROTECTED]Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2 is one ofthe easiest configs to put together.Works extremely well and requiresopening a single port on each NAT.
Now I realize that I took the wrong assumption that all NAT traversal is blind traversal.
I downloaded version 1.4.0 compiled and installed it. This is my
extensions.conf:
[outboundmsg1]
exten = s,1,NoCDR
exten = s,n,AMD
exten = s,n,NoOp(${AMDSTATUS})
exten = s,n,NoOp(${AMDREASON})
exten = s,n,GotoIf($[${AMDSTATUS}=HUMAN]?humn:mach)
exten = s,n(mach),WaitForSilence(2500)
exten =
Hi there, im looking for another place that provides manuals and
firmware updates for the ATCOM AT 468 and their configuration with
asterisk.
the site www.atcom.com.cn has non functional download links.
I have several of these units but it came only with one CD, I
misplaced it and I cant
On Fri, Jan 26, 2007 at 06:17:03PM +0100, François Delawarde wrote:
Hi all,
I would like to make a script/program that would be able to show lots of
status information from my analog FXO lines (and FXS lines in the near
future).
Example of interesting status information:
- Hook status:
Of note, I tried the same call using IAX2 instead of SIP, and it was fine.
This may either be 1) a configuration problem or 2) a SIP provider problem.
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Hi all,
I have a problem with Asterisk dropping audio.
While in call, audio gets dropped for a while (from 5 to 60 secs, and
obviously users often hangup, this means that I'm not sure the audio is
always coming back after 60 secs), in the meantime the call remains up
and no SIP signalation is
Dear all,
How may I configure my extensions.conf so that only the boss's secretary
can call the boss through his extension, all others when dial his
extension only makes the boss's secretary phone ring, not his. If she
wants, she can transfer the incoming call to the boss dialling his
Why don't you just give the secretary the boss' REAL extension and give a
different extension to the world that just rings the secretary?
-jonathan
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Ricardo Carvalho
Sent: Friday,
From what I know this log show everything working Perfect.
Then it goes to the Welcome screen then after a long time of processing,
it errors out with a 0x1 error
Any Ideas?
1005195711|so |4|00|-- Initial log entry --
1005195711|so |4|00|+++ Note that bootrom log
This is typically an error in one of your config files, either
0004f2023ecc.cfg or sip.cfg. What does your 0004f2023ecc.cfg file look
like?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Walker
Sent: Friday, January 26, 2007 12:10
To: Asterisk
Maybe you could use something like:
exten = boss_ext,1,GotoIf($[${CALLERID(number)}=secretary_ext]?boss:secretary)
exten = boss_ext,n(boss),Dial(SIP/boss_ext)
exten = boss_ext,n(secretary),Dial(SIP/secretary_ext)
## nini @ www.modulo.ro ##
Jonathan k. Creasy wrote:
Why don't you just give
Looks like the network time server isn't provisioned.
--
Bill
1005195752|app1 |4|00|Could not load time from 0.0.0.0(0.0.0.0).
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I have an X100P which I have set up as per the guidelines:
http://www.x100p.com/support/doc/quick_start_fxo.php
The card is recognized by the system:
Zaptel Version: 1.4.0
Echo Canceller: MG2
Configuration
==
Channel map:
Channel 01: FXS Kewlstart
Fixed that issue but it does not change the error
0126204105|cfg |3|00|Image sip.ld has not changed
0126204105|copy |3|00|Download of 'sip.ld' succeeded on attempt 1 (addr
1 of 1)
0126204105|cfg |3|00|Downloaded application image is identical to
current version
0126204105|cfg |3|00|Phone
How may I configure my extensions.conf so that only the boss's secretary
can call the boss through his extension, all others when dial his
extension only makes the boss's secretary phone ring, not his. If she
wants, she can transfer the incoming call to the boss dialling his
extension.
the
Be sure that your mac.cfg file is pointing to a valid configuration
file, I believe the 0x1 error is a missing file error.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Walker
Sent: Friday, January 26, 2007 12:49
To: [EMAIL PROTECTED];
?xml version=1.0 standalone=yes?
!-- Default Master SIP Configuration File--
!-- Edit and rename this file to Ethernet-address.cfg for each phone.--
!-- $Revision: 1.14 $ $Date: 2005/07/27 18:43:30 $ --
APPLICATION APP_FILE_PATH=sip.ld CONFIG_FILES=jason.cfg, sip.cfg
MISC_FILES=
I noticed the same problem as well. I will see if I the old sound files
corrects the problem, or if it's actually a timing problem.
I have to say, I like the old sounds better, they sounded softer.
-Ry
On 12/17/06, Gil Kloepfer [EMAIL PROTECTED] wrote:
Is anyone else finding in the new audio
Hi all,
Running Asterisk 1.2.12 (a bit out dated, but it was fully operational until
a few days ago), I'm seeing the following message in my logs, repeated
literally millions of times:
channel.c: Nobody there, continuing…
We've started to see some odd behavior (incoming callers can hear us, we
Our operator has asked if it is possible that when a call times out in
the call parking and comes back to her, if there is someway to show that
call has come back from parking. I have looked all over the
documentation and have come up with nothing so far.
All I see when a call times out is:
Hello,
I just buyed a SPA3102 from Linksys. Can anyone help me or guide me to
configure voice part on it. I cannot get it if I can use like peer for my
asterisk. Please help me with some tips.
Thank you guys.
Regards,
Jonson.
___
--Bandwidth and
Hello,
Maybe using app_backticks will solve your problem.
I use it to call a script and saved the result into an Asterisk variable.
http://www.pbxfreeware.org/app_backticks.c
http://www.voip-info.org/wiki/view/Asterisk+cmd+Backticks
Regards,
## nini @ www.modulo.ro ##
Pavel Jezek wrote:
any
Looks alright there. The next config to check is where it loads your
'jason.cfg', any errors will be in your app logfile (as opposed to the
boot one you pasted).
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jason
Walker
Sent: Friday, January 26, 2007
Hello,
We have an asterisk system with about 40 cisco 7940/7960 phones and a few
linksys SPA941. I recently analyzed our network and discovered that the rtp
packet size from the cisco phones is 10ms. We want to change the rtp packet
size of the Cisco phones from 10ms to 20ms. I know how to do
Asterisk User List wrote:
Our operator has asked if it is possible that when a call times out in
the call parking and comes back to her, if there is someway to show that
call has come back from parking. I have looked all over the
documentation and have come up with nothing so far.
All I see
Kyle Gordon wrote:
fxsks=1 #X100P
Is your line truly a kwelstart line? try fxsls
SNIP
busydetect=yes
You may need to add these 2 values to help the busydetect
busycount=3
busypattern=375,375
busypattern tells asterisk how your busy tone sounds like, in UK it
should be 400Hz 0.375s ON and
On Fri, Jan 26, 2007 at 08:42:36PM -, Charlie Grosvenor wrote:
I have an X100P which I have set up as per the guidelines:
http://www.x100p.com/support/doc/quick_start_fxo.php
The card is recognized by the system:
Zaptel Version: 1.4.0
Echo Canceller: MG2
Configuration
On Sat, Jan 27, 2007 at 07:40:31AM +0800, Leo Ann Boon wrote:
Kyle Gordon wrote:
fxsks=1 #X100P
Is your line truly a kwelstart line? try fxsls
And if the line is ls, indeed, what harm is there in setting it up as
ks?
Consider, e.g.
Tzafrir Cohen wrote:
On Sat, Jan 27, 2007 at 07:40:31AM +0800, Leo Ann Boon wrote:
Kyle Gordon wrote:
fxsks=1 #X100P
Is your line truly a kwelstart line? try fxsls
And if the line is ls, indeed, what harm is there in setting it up as
ks?
I understand ks is ls with a
Yes the line is connected, a standard phone works fine when connected to
the line.
Thanks
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir
Cohen
Sent: 26 January 2007 23:45
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] X100P -
Charlie Grosvenor wrote:
Yes the line is connected, a standard phone works fine when connected to
the line.
There're 2 ports on the card. Which port are you using? One of the ports
is for connecting another phone in parallel to the card.
Leo
___
Olle E Johansson wrote:
26 jan 2007 kl. 16.31 skrev James Fromm:
Olle E Johansson wrote:
24 jan 2007 kl. 18.10 skrev Eric ManxPower Wieling:
James Fromm wrote:
The behavior we see is that the SIP interface in the queue will
sometimes not release from the in-use state. Connecting to the
- Original Message -
From: kjcsb [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 24, 2007 8:24 AM
Subject: Re: [asterisk-users] realtime sipusers and rtcachefriends...
bigheadache!!
hi folks,
Hi!
I don't understand what you mean by : „configure voice part on it, but I
can give general guidelines:
First you setup SPA3000 web UI:
1) Line1 Tab:
Sip settings:
SIP port : 5060
Proxy and Registration:
Proxy: Asterisk IP
Subscriber Information:
Display Name:
Hi there,
We traced this issue to transfers (see http://bugs.digium.com/
view.php?id=8848). On Monday, I will be attaching the various debugs
to the case as requested.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
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Dial(SIP/[EMAIL PROTECTED]) will ring forever even if no application is
listening. How can Asterisk tell if the user is not answering or simply not
having SIP?
Yuan Liu
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asterisk-users
The SM56 MODEM manual says it does. But when used with zaptel 1.2.12,
nothing shows up.
Yuan Liu
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From: Yuan LIU [EMAIL PROTECTED]
The SM56 MODEM manual says it does. But when used with zaptel 1.2.12,
nothing shows up.
Debug level 6 (Asterisk 1.4.0) only shows:
[Jan 26 22:56:46] ERROR[25603]: callerid.c:564 callerid_feed: fsk_serie made
mylen 0 (-14)
[Jan 26 22:56:46] WARNING[25603]:
From: Tim Panton [EMAIL PROTECTED]
Thanks for this information. Does this mean two IAX boxes can talk
behind their respective NAT's (without any server sitting in voice path)?
I'm imagining this:
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2
If Asterisk1 can talk to Asterisk2
On Thu, Jan 25, 2007 at 10:19:06PM -0800, Yuan LIU wrote:
Asterisk1 -- NAT1 --- { Internet } --- NAT2 -- Asterisk2
If Asterisk1 can talk to Asterisk2 at trunk level, I'll be happy.
While I'm not sure of what tricks * plays at all levels, you
can certainly make this work if you have control of
On Fri, Jan 26, 2007 at 12:34:30PM +, Tim Panton wrote:
For a remote phone, not on the same network as the Asterisk
box (in which event the NAT worries are different) you definitely
want to use the same protocol for the phone as for your
term/orig provider. Otherwise you will be forced
From: Yuan LIU [EMAIL PROTECTED]
Debug level 6 (Asterisk 1.4.0) only shows:
[Jan 26 22:56:46] ERROR[25603]: callerid.c:564 callerid_feed: fsk_serie
made mylen 0 (-14)
[Jan 26 22:56:46] WARNING[25603]: chan_zap.c:6389 ss_thread: CallerID feed
failed: Success
[Jan 26 22:56:46] WARNING[25603]:
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