Hello All.
I'm using sangoma card A-101. tested successful with E10 (ACATEL) Exchange,connection with E1, CAS, (using unicall-0.0.3pre4).
my systemrun success,incoming call and call out are good.
when iswitch to EWSD (SIEMENS) R-15. my asterisk faill, cannot connect with EWSD.
(E10 and EWSD
Paul Belanger wrote:
#root service asterisk start
Starting asterisk: [ OK ]
# ps aux
does asterisk show up as a process?
nope. But it does if I manually type safe_asterisk or asterisk
Julian
PB
___
HI,
Has someone sucessfully connected this channelbank to an asterisk through a digium card?
I would rather use this channel bank, since is the only one I have!
thanks to all
Alejandro
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Nguyen Trung Tin wrote:
Hello All.
I'm using sangoma card A-101. tested successful with E10 (ACATEL)
Exchange, connection with E1, CAS, (using unicall-0.0.3pre4).
my system run success, incoming call and call out are good.
when i switch to EWSD (SIEMENS) R-15 . my asterisk faill, cannot
--- Waldo Rubinstein [EMAIL PROTECTED] wrote:
I tried the same experiment with all the queueing
strategies and the
behavior was the same. The only exception was with
ringall. The
problem with ringall is that it shows the same
caller-ID to all
agents. Once the first agent picks up
Hi,
we want to build a Asterisk server for a branch office in Australia.
At the moment they use 5 analog lines. We will need at least 8 lines.
What hardware would you recommend for the 8 analog PSTN lines?
Thanks
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I've copied over all the asterisk configuration settings. There was nothing
decent to see in the logs, so I#12288;didn't copy those.
http://zanshin.tsumelabs.com/
The system was working a couple days ago until I had the server rebooted.
there are 2 zap cards, both are working fine, all lines
From the command prompt type: help-aah
This will give you a list of commands to
change passwords. For example:
Commands Descriptions
---
config set the local time
zone and keyboard type
netconfig configure
Does anybody know if the Digium TE series cards will work with NI-1 (SBC
California) ISDN BRI? If not can anyone make recommendations as to reliable
cards to use? My end goal is to use the BRI lines for incoming fax
(spandsp) only.
Thanks in advance,
Aaron Picht
On Thu, 1 Sep 2005, Aaron Picht wrote:
Does anybody know if the Digium TE series cards will work with NI-1 (SBC
California) ISDN BRI? If not can anyone make recommendations as to reliable
cards to use? My end goal is to use the BRI lines for incoming fax
(spandsp) only.
I cannot tell you
Try doing an strace on it and seeing what the last section shows you.
i.e. strace asterisk -vvvc
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Julian
Lyndon-Smith
Sent: 31 August 2005 22:39
To: Asterisk Users Mailing List - Non-Commercial
[apologies if this comes through twice - the original
doesn't seem to have shown up even after 16 hours]
In the handling of agents, when using AgentCallbackLogin, a call placed to
an agent needs to be accepted by the agent pressing the '#' key.
I'm trying to replicate that kind of operation in a
Hi Steve :)
The problem is not with the asterisk command, nor with safe_asterisk but
with the /etc/init.d/asterisk script
if I manually run
/etc/init.d/asterisk start
all's ok
if I manually run
service asterisk start
it says that it has started, but hasn't :)
Julian
Steve Hanselman
Hello Paul,
Thursday, September 1, 2005, 4:38:42 AM, you wrote:
PH I am setting up a snom 360, and the lights come on OK when the mapped
PH user makes an outgoing call, but when the user takes an incoming call
PH the light does not come on.
PH I do not want to install the bristuff patch if
Hi
I have setup a queue with 2 agents in it...
one on an extension
the other an outgoing call - Cell phone
If I have to callers in the queue, and pickup the the first caller
with my cell phone
the other caller gets a all circuits are busy, please try again later
Why is that.
I have
Hello,
We recently bought 2 TDM400P Rev I boards with in total 8 FXS ports to be
used with Asterisk. I use Asterisk CVS Head (version around 15 aug 2005).
I have an ISDN Quad boards towards the national Telco. The TDM400P has
his own Interrupt line.
I encounter 2 major problems:
1) Transmitting
I want to speed-up dialing on X101P clone (Ambient modem). I probably
must change wcfxo.c, but what line to change?
I found what to change: digits.h line 23
from
#define DEFAULT_DTMF_LENGTH 100 * 8
to
#define DEFAULT_DTMF_LENGTH 50 * 8
and my dialling is now much faster.
But, I have new
Hi all,
I have a strange problem with a quadbri
card and my asterisk box with installed verson 1.0.7 of asterisk Bristuffed.
I have connected to the card 3 isdn
in ptp mode configured in selection passing (I don't know if is exact the
english traduction but I have 3 isdn with 99 numbers and
I'm trying to find a way to generate an 'internal
extensions' tonelist but I can't seem to find anything on how to do this.
My idea was to start a Playtones(intercom) tonelist and not indicate ringing to
the line (dead air). But then, somehow StopPlayTones needs to be run once
the ringing
Hi everybody,
Ive proxy server IP, user ID and password. Now I need
to connect to a remote Asterisk server as a SIP using my Asterisk @ Home box. That
Asterisk server will make PSTN calls for me. I think I am making mistake while
setting up the Trunk because when trying to make calls,
Issue #3644 has recently been committed to CVS-HEAD which allows for full device state notification via subscriptions for Snom 360 and other supporting phones w/o the need for additional patches.
On 9/1/05, Alessio Focardi [EMAIL PROTECTED] wrote:
Hello Paul,Thursday, September 1, 2005, 4:38:42
On Thursday 01 September 2005 06:35, Alex Ongena wrote:
I encounter 2 major problems:
1) Transmitting and receiving van Fax is very unreliable (on the CLI I see
a Native bridging (seems to be 911.ulaw, 64 kbit high quality).
Sometimes the Fax is Ok, sometimes I miss a few lines, sometimes
On Tue, Aug 30, 2005 at 07:45:50PM +0200, [EMAIL PROTECTED] wrote:
I have a strange problem with a quadbri card and my asterisk box with
installed verson 1.0.7 of asterisk Bristuffed.
I have connected to the card 3 isdn in ptp mode configured in selection
passing (I don't know if is exact
Hi,
Ive Sipura 1001 phone adapter. In the settings it has
separate Line 1 and Line 2 tabs, which apparently means it can control two
separate phone lines. Ive [EMAIL PROTECTED] server and want to setup two
different extensions for two phones, i.e. 201 and 202. After doing all this, I
Hi,
In asterisk at home, in Outbound Routing menu, under the trunk
sequence (e.g. IAX2/FWD), what does little red cross mean beside the selected
trunk.
Thanks
Zeeshan
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Erick,
After reviewing your original message a little closer it occurs to me
that you may be able to trunk Asterisk---Meridian with 2 Digium TDM400
cards. These are Quad FXS or FXO cards that could receive the lines from
your 8 analog line card.
You'll still need an E1 card (Digium or Sangoma)
PH I am setting up a snom 360, and the lights come on OK when the mapped
PH user makes an outgoing call, but when the user takes an incoming call
PH the light does not come on.
PH I do not want to install the bristuff patch if possible.
PH (although I can see that with the devstate command I can
2) I have 2 TDM400Ps installed in a system. I need the audio on the
analog phone (FXS modules) to be amplified somewhere between 10 and
15db so I set rxgain and txgain to 15 for the FXO interfaces and FXS
interfaces. When a call comes in on the FXO at this setting, the call
sometimes has
Hi there.
The title basicly explains it. When we get
incomming calls from cellular phones, the callers tend to echo ALOT. They hear
their own voice at very high volums.
This is a problem only for mobilphone
users that calls in to us.
Im using wifi IP-phones.
Asteirsk:
Is your Asterisk server listening on port 5061? If not, just change the
entry to 5060.
On Wed, 2005-08-31 at 16:36 -0600, Andres Paglayan wrote:
Hi,
I am having problems on getting the second line to work on a Polycom 301,
this is the phone.cfg file,
the * box is 192.168.1.8 and the phone
IS there a way to make the phone reboot each day at a time?
On Thu, 2005-09-01 at 07:24 -0500, Jeff Brownlee wrote:
PH I am setting up a snom 360, and the lights come on OK when the mapped
PH user makes an outgoing call, but when the user takes an incoming call
PH the light does not come on.
I have upgraded the GXP-2000 to the newest firmware 1.0.1.12 and the
phone is much more usable However, I still have two slight sound
quality issues:
1) There is static on the line at all times. It is not that
noticable to me, but when I make calls out the PSTN the person on the
other end hears
+ sign to add another trunk to the route
Zeeshan Zakaria wrote:
Hi,
In asterisk at home, in Outbound Routing menu, under the trunk sequence
(e.g. IAX2/FWD), what does little red cross mean beside the selected trunk.
Thanks
Zeeshan
Sorry, went of at a tangent (that's what you get for half reading emails
I guess!)
Ok, guess the easiest thing to do is to check in the contrib directory,
diff your one against the redhat (are you running redhat?)
Mine is the same and it works fine, maybe you're running an outdated
init script.
On Thu, 2005-09-01 at 08:12 -0400, Andrew Kohlsmith wrote:
On Thursday 01 September 2005 06:35, Alex Ongena wrote:
I encounter 2 major problems:
1) Transmitting and receiving van Fax is very unreliable (on the CLI I see
a Native bridging (seems to be 911.ulaw, 64 kbit high quality).
Yes. You could send a sip notify via asterisk and set that up via cron to happen once per day.
eg.
asterisk -rx sip notify reboot-snom sip peername the snom is at
Make sure that the reboot-snom clause is setup in sip_notify.conf before attempting this.
On 9/1/05, altus [EMAIL PROTECTED] wrote:
If you do not want to buy a timer based power supply ;) you can send a NOTIFY
with 'Event: reboot' to the phone.
Nils
On Thursday 01 September 2005 14:41, altus wrote:
IS there a way to make the phone reboot each day at a time?
On Thu, 2005-09-01 at 07:24 -0500, Jeff Brownlee wrote:
PH I
I tried to modify the extension but
* soon receive the extension and the other hand get busy
i.e
First:
The caller digit 123456 (at the fourth
digit connect get the ring back)
exten = _1234,1,dial,sip/25
exten = _123456,dial.sip/30
in this way the other hand get the first
estension and does not
Where in Australia?
We are based in Sydney. One of our clients have 4 analog lines using 2 dual
FXO Cards and seems to work fine. Most of our clients have an E1.
I Have not tried the 4 port cards however I would suggest 2 of them would be
sufficient.
OCTTEL have a 4 port router which is
I have just signed up for 2 landline numbers in China. They have
offered to sell me 2 h323 compatible handsets which I have declined as
I want these numbers to ring into my * box.
They have given me the following info (modified for security)..
Protocol = H323
Gatekeeper = 210.21.118.xxx
Hi,
Has anyone out there managed to do a hookflash transfer with a Micronet
5050s gateway ?
We have just tried out this gateway and it seems to do everything we need
except this
particular feature. Also if you have succeeded where is the hookflash time
specified in the
gateway. There does not
Hi Jason,
Is there a specific version of DIAX that I should use? I grabbed the
latest
release...Looking at the DIAX site, 910g has the URL feature fixed.
Is it
broken again in 915a?
URL feature works in 0.9.15a.
Take care that it is implemented JUST for the Dial command.
Best regards,
Dan
IS there a way to make the phone reboot each day at a time?
You could do it via a cron job by wget'ting the reboot uri (on the advanced page again),
but there really shouldn't be any need to do so. The only time subscriptions should
disappear is when you do a reload or restart on asterisk.
I have a Digital
Toshiba Strata PBX with an IP Card (Model BIPU-M2A) that supports 16 connections
and I want to Integrate Asterisk via this card but I found out it supports
MEGACO+(I believe share a lot with MGCP). I am not sure how to go forward with
this. (Note: Theproprietary IP Phones
Not sure this applies, but there does seem to be a problem in the
zaptel area that also impacts service... command. If I kill asterisk
and try 'service zaptel stop', it now fails consistently as its
unable to stop 'ztdynamic' (leaving zaptel still running since it
can't unload). However, 'service
Is your Asterisk server listening on port 5061? If not, just change
the
entry to 5060.
Also, I'm not sure how your sip.conf is set up for asterisk, but if
you've set it up like:
[203]
type=friend
username=blah
secret=blah
etc...
Your Polycom config file will generally look like this.
Do i have to change the adsl routers? or just do QoS with the Layer 3 switches?
Will my ADSL router respect the QoS setting when sending the packet to
the Internet?
On 9/1/05, asterisk groups [EMAIL PROTECTED] wrote:
Erick,
After reviewing your original message a little closer it occurs to
Erick- Can't say if they will or not. In theory they should respect all
outgoing traffic unless being filtered by another device such as your
PIX. You might want to check with the ADSL router manufacturer just to
be safe.
On Thu, 2005-09-01 at 09:25 -0500, Erick Perez wrote:
Do i have to change
Hey all,
I know you all saw the topic and let out a groan. However, I understand
how to get an overhead paging system to work with respect *, however I am
now looking for a small(?) paging amp, that I could hook 3 or 4 horns to.
I would like to just have the * extention be routed to a soundcard
Hello,
I have a customer who said that their Snom 360 is joining calls by accident.
The situation is that they had one call on the line and another call came in.
She pressed the hold button on the phone and the two calls were joined
together.
I do have Call join on Xfer set to yes, but I
I have two call queues in which the agents are added through
AgentCallbackLogin. Whenever they answer the call then TRANSFER, the call
is transferred but the agent status continues to show them talking on the
zap channel. Eventually, timeout with no RTP traffic the agent appears to
be
hello
I am looking for Israel and Russia DiDs.
Please email me to [EMAIL PROTECTED]
REgards
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Hello
Personaly i prefer oh323, i am using for one year whitout problems.
and is more easier to configure.
regards
On 9/1/05, Steve Ducat [EMAIL PROTECTED] wrote:
I have just signed up for 2 landline numbers in China. They haveoffered to sell me 2 h323 compatible handsets which I have declined
has anyone an idea how to display incoming national/international
isdn-pstn-calls correctly to internal isdn AND sccp/sip-phones ?
without nationalprefix=0 and internationalprefix=00 I get incoming phone
numbers correctly on isdn-phones
but the leading zero's are stripped of for non-isdn phones
service zaptel start|stop works fine for me.
service asterisk start does not
[EMAIL PROTECTED] res]# service asterisk start
Starting asterisk: [ OK ]
[EMAIL PROTECTED] res]# service asterisk status
asterisk dead but pid file exists
[EMAIL PROTECTED]
Chris Deserva wrote:
I have written it in C++, because I used an OCI
interface library (ORAPP). I want to post it
opensource so that I could get help in its development
and testing, and be a part of Asterisk modules.
You cannot make this open source. The Oracle client libraries are not
Aaron Picht wrote:
Does anybody know if the Digium TE series cards will work with NI-1 (SBC
California) ISDN BRI? If not can anyone make recommendations as to reliable
cards to use? My end goal is to use the BRI lines for incoming fax
(spandsp) only.
No Digium boards work with BRI circuits
The two most common companies to make paging equipment are Viking and Bogen.
Bogen even resells ATAs for paging now. http://www.bogen.com or
http://www.vikingelectronics.com
Chris Coulthurst
[EMAIL PROTECTED]
- Original Message -
From: [EMAIL PROTECTED]
To:
I'm using oh323 too without any issues,
but in Steve specific configuration, depends on how his provider expect to be
register as (Terminal or Gw) afaik, oh323 just could be binded as gateway, so
better ask the provider.
LTenorio
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
Hi,
The PBX recives alarms from the TE110p card and are mainly pointing at frame
errors and Loss of signal.
Asterisk is configured as
Zapata.conf
signalling=pri_cpe
switchtype=national
rxwink=250
channel = 1-15,17-31
Zaptel.conf This is what I need to know - the SPAN is currently set to
Hi all,
I would like to enable SIP incoming calls from any origin (not configured as
peer in sip.conf). Is this possible?
A workaround could be to put a SER in front of the Asterisk, and configure
this as a peer in sip.conf, but I would like to find other simpler way if
possible.
Thank you
Hi,
Did anyone successfully installed and setup Astaro
Security Linux V6 SIP Proxy with Asterisk behind the
Astaro and clients bedinde another NAT?
Regards
Start your day with Yahoo! - make it your home page
Michiel van Baak wrote:
On 17:22, Wed 31 Aug 05, Eric Skippy Hope wrote:
We're using 1.0.9 and the powers that be are wary of moving beyond
stable. If I'm reading the wiki correctly, incominglimit is to limit
the calls coming _from_ the extension and coming into the server, and
I am still getting an error compiling the 1.2-Beta version. The
tarball works fine, but I have never been able to compile the 1.2beta
from CVS. I have been compiling CVS-HEAD on the machine for quite
some time.
It goes into this loop:
if cmp -s include/asterisk/version.h.tmp
I want to be able to authenticate each user that dial out
(PSTN or IP), ideally using their mailbox and voicemail password.
Should I use VMAuthenticate or
Billing?
Any pointers on setting up either?
Cheers
Graham___
--Bandwidth and Colocation
Greetings
We have all those problems and then some... after a while,
the phone starts degrading: The ringing becomes lower and lower and
there is a lot of stuttering in the conversation. Also, if I stop/start
asterisk, half of the phones reconnect while the rest don't. I was
using the same
Viking makes everything you might need for paging and door control.
www.vikingtelecomsolutions.com
William Boehlke
Signate
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Thursday, September 01, 2005 7:34 AM
To:
On Wed, Aug 31, 2005 at 09:46:37PM -0400, Doug Lytle wrote:
When there is a call on zap 1, from a sip phone on the remote office
side and typing 'zap show channel 1' shows echo cancel is on, doing the
same thing from the Definity to a SIP phone shows echo cancel off.
Shouldn't it be on
And to add to what Kevin said, we don't want any closed source stuff, be
it a database module or a device driver, to be a part of Asterisk as a
standard module, for obvious reasons.
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kevin P.
Matt Fredrickson wrote:
On Wed, Aug 31, 2005 at 09:46:37PM -0400, Doug Lytle wrote:
When there is a call on zap 1, from a sip phone on the remote office
I have not seen it myself, but I have heard that some people have ahd trouble
with
overlapdial and echo cancellation. I have not
On Wed, Aug 31, 2005 at 08:45:34PM -0500, chris gamble wrote:
the echo isnt horrible most of the time, and seems extremely random in
that i can call a number once without echo, then dial the same number a
second time and get echo.
things i am currently considering (and would like to know if
Hi all,
Im installing two HFC pci card (both in TE
mode), I dont have problem when load module, but whrn I give ztcfg
vv, I see 6 the six channels that I configured, then my computer
hang and I have to reboot it. (Im using a VIA Epia-M 1000 with Via C3
processor)
[EMAIL PROTECTED]:~#
Hi
When a a context , and you dial a extension, will the current one be
looked into, or the default incoming one.
My call scenario is to bring in all users into a default context and
then GoTo others based upon some parameter. Now when a user dials a
extension, the match should occur within
On Thu, Sep 01, 2005 at 01:24:02AM -0500, Kristian Kielhofner wrote:
Nguyen Trung Tin wrote:
I'm using sangoma card A-101. tested successful with E10 (ACATEL)
Exchange, connection with E1, CAS, (using unicall-0.0.3pre4).
my system run success, incoming call and call out are good.
when
Has anyone gotten *66 (busy callback) to work with asterisk and
devices like the sipura SPA-2002? When I dial *66 and hangup.. the
sipura seems to immediately try again (which is probably normal) and
then ring me (Even if the line is busy) any idea why the sipura is
not detecting the line as
Thanks to all that have replied thus-far.
I have talked to Viking and they recommended the PA-2A paging amp.
(http://www.vikingelectronics.com/products/view_product.php?pid=317). It
requires a 600ohm input. Before I go beating my head against the wall
with this, has anyone else installed
I was wondering if anyone could shed some light on what options I
have for mapping incoming/outgoing SMS messages to/from a telephone
number that I am given by a VoIP provider who does not currently
offer SMSC services?
In other words, Voicepulse, my VoIP provider, provides me with a PSTN
Does anybody have experience with this all-in-one router? (triple play,
ADSL, VoIP, Wifi).
I needed to upgrade the firmware, or otherwise the ADSL Internet connection
would drop during a call.
This has been fixed now. Also the device registers fine with my asterisk.
Incoming calls through the FXO
Hi,
I put on sip.conf the following line
#include sip.d/*.conf
inside I have files like that
provider1.conf
provider2.conf
But asterisk does not want to load it
This is the error
Sep 1 13:18:35 VERBOSE[8756]: == Parsing
'/etc/asterisk/sip.d/*.conf': Sep 1 13:18:35
VERBOSE[8756]: == Parsing
Other than DIDx what is another DID provider that I can buy DIDs from?
Joshua
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Hi,
you can get the following by typing 'help-aah' from the
CLI
Commands
Descriptions---config
set the local time zone and keyboard
typenetconfig
configure ethernet
interfacegenzaptelconf
autoconfig Zaptel
Thanks..I am begining to agree with you about these phones. Which
poylcoms do you have? I have been looking at the polycom
soundpoint IP501. It seems like a good phone for just under
200USD.
Thanks again,
Aaron On 9/1/05, Jesus Mogollon [EMAIL PROTECTED] wrote:
Greetings
We have all those
Good Day all,
I have a box connected to a netgear switch which allows me to
set priority based upon DSCP Values. This switch has listings from
value 1 - 63. And can be set to normal, high, etc. Does anyone know
what or how to translate TOS= line in the sip.conf file to in order to
have
great nevermind. I was relying on my friend's configurations which are
broken at agent-add. He was logging in the phones by just going through
typing in each login manually on each phone. :| I'm curious then why the
heck he placed a login key on each phone.
...
:(
Original Message:
Did the sip jitter buffer make it into 1.2? anyone using it?
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To
sorry to report that I reported exactly the same error, twice. Never got
a response from anybody on the 1.2beta team.
CVS head compiles just fine.
Julian.
Geoff Karl wrote:
I am still getting an error compiling the 1.2-Beta version. The
tarball works fine, but I have never been able to
I'm using Asterisk 1.09 with bristuff 0.2.0-RC8n, one BRI line and a
Sipura SPA-2000. I have a HP LaserJet 3330mfp all-in-one. I can receive
faxes but not send them. The faxes start whistling to each other but the
transmission is stopped with a communication error
To receive a fax I have this
Matt Fredrickson wrote:
On Wed, Aug 31, 2005 at 08:45:34PM -0500, chris gamble wrote:
the echo isnt horrible most of the time, and seems extremely random
in that i can call a number once without echo, then dial the same
number a second time and get echo.
things i am currently considering (and
On Thu, 1 Sep 2005, Jesus Mogollon wrote:
We have all those problems and then some... after a while, the phone starts
degrading: The ringing becomes lower and lower and there is a lot of
stuttering in the conversation. Also, if I stop/start asterisk, half of the
phones reconnect while the
Il Neofita wrote:
Hi,
I put on sip.conf the following line
#include sip.d/*.conf
You neglected to include the most important piece of information: what
version of Asterisk you are using.
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Hi I currently have a 3072kbps line that I'm splitting in half for 5 of
my phones. That's 307.2kbps +/- a couple of kpbs.
What is the minimum kbps for a phone to maintain clarity and volume?
Joshua
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not set -- nothing to do
-- AGI Script recordingcheck completed, returning 0
-- Executing SetVar(Zap/1-1,
CALLFILENAME=g3-20050901-115459-1125593688.105) in new stack
-- Executing Goto(Zap/1-1, s|14) in new stack
-- Goto (macro-record-enable,s,14)
-- Executing GotoIf(Zap/1-1, 0?15
Hello Everyone,
I have one machine (asterisk server) that is DMZ behind my nat firewall
on my client end (at home) i have a linksys wrt54g with 16384-32766
forwarded to my cisco 7960 (which works fine) and 16000 - 16383 forwarded
to my sipura 2100 (which is set to these ports)
For some reason,
Just in case somebody else has this problem, it seems that there is a bug in
the 3.1.3a version of firmware of the Sipura SPA-841. Updating to the 3.1.4a
version of the firmware solved the problem.
On Sun August 28 2005 01:55, Juan Jose Comellas wrote:
I have just bought several Sipura
this is a link where you can understand the relationship between tos and
dscp
http://www.speedguide.net/tcpoptimizer.php
Ros
- Original Message -
From: Ronald Hartmann [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Thursday, September 01, 2005 1:37 PM
Subject:
I am using it with CVS-HEAD but it is currently a patch. So far
the version of the patch I have (which was the first one released)..
seems to be working very well.. and definately makes a noticeable
improvement.
On 9/1/05, Damon Estep [EMAIL PROTECTED] wrote:
Did the sip jitter
Pause betwen incoming rings on my phone line is 4s, so when x101p
clone
(wcfxo driver) do not receive next ring signal after 4.5 sec, call
should be consider as ended.
What should I change to set that time (4.5 sec) for incoming ring end
detection?
I measured, event -- Hungup 'Zap/1-1' is
On Thu, Sep 01, 2005 at 06:47:10PM +0200, Giordano Grandis wrote:
Hi all,
I'm installing two HFC pci card (both in TE mode), I don't have problem
when load module, but whrn I give ztcfg -vv, I see 6 the six channels
that I configured, then my computer hang and I have to reboot it. (I'm
On Thu, Sep 01, 2005 at 10:15:04AM +0100, Julian Lyndon-Smith wrote:
Hi Steve :)
The problem is not with the asterisk command, nor with safe_asterisk but
with the /etc/init.d/asterisk script
if I manually run
/etc/init.d/asterisk start
all's ok
if I manually run
service
On Thu, Sep 01, 2005 at 09:13:36AM -0600, Rich Adamson wrote:
Not sure this applies, but there does seem to be a problem in the
zaptel area that also impacts service... command. If I kill asterisk
Do you manually kill the asterisk process? If so: what happens with the
pid files and lock files?
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