I'd frankly hoped that people here would be - on average - a bit smarter
than hey! it's got a catchy name! and ajax buttons too!.
l.
In data Tue, 26 Jun 2007 03:00:48 +0200, Mark Phillips [EMAIL PROTECTED]
ha scritto:
Great! Another one. With such a catchy name too!
On Tue, 2007-06-26 at
by the way selling does not depend on the amount of lines you have and
we are very productive trust me
True, very true. There are lots of very productive sales people that don't
need a phone at all. From the paper boy to car dealers, lots of sales don't
require many phone lines. Of course at
On 6/26/07, Jared Smith [EMAIL PROTECTED] wrote:
I'm making a wild guess here, but I'd say that if you're using
trunking, then you're probably getting close to exceeding the MTU size
or possibly the MAX_TRUNKDATA size as defined in chan_iax2.c. If it's
happening without IAX2 trunking turned
On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
We're looking at a large wifi phone deployment, and we're looking for wifi
phones that:
HAve a look at the Linksys WIP 300 (or something)
Can be charged from the USB port
--
Hendrik Visage
___
Replace with below. Actually Asterisk should only generate ringback when the
SIP phone is ringing.
On 6/25/07, satish patel [EMAIL PROTECTED] wrote:
exten = 222,1,Dial(SIP/222,r)
exten = 333,1,Dial(SIP/333,r)
exten = 555,1,Dial(SIP/555,r)
exten = 100,1,Dial(SIP/100,r)
exten =
Any recommendations on an economical layer 3 switch?
I've been quite happy with the Netgear FS728TP ProSafe switches. These are
24 port 10/100 switches with 4 GigE ports and PoE. They also support 1000-SX
GBIC through an optional module. The total PoE budget for all 24 port is 195
watts. We
Any idea why I can't build HPEC for zaptel 1.2.18?
It builds fine with 1.4.3...
PaulH
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To UNSUBSCRIBE or update options visit:
Hi all,
I have a isue with a Siemens Hicom conected to my asterisk, here is the
scheme:
Telco Asterisk --- Legacy PBX --- Legacy phones
The asterisk box has a TE210 (one PRI conected to Telco another PRI
conected to Siemens)
Everything works ok, but when I make an international
It can be fixed with the patch from
http://lists.digium.com/pipermail/asterisk-dev/2007-June/028093.html
Cheers, Troy
On 26/06/07, Jaswinder Singh [EMAIL PROTECTED] wrote:
This is due to changes in cdr in asterisk 1.4.5 so in all outgoing
dial from macro it shows 's' in cdr . Could this be a
Hi
now, I am using, realtime connection(mysql) for
dialplan,
but the following line must be added ,manualy to
extensions.conf, before reloading.for each new
context.
[NEW_CONTEXT]
switch = Realtime/@extensions
is there any idea, to add this line to dbase too?
thanks in advance
Best
MAni
On Tue, 26 Jun 2007, Hendrik Visage wrote:
On 6/25/07, Michelle Dupuis [EMAIL PROTECTED] wrote:
We're looking at a large wifi phone deployment, and we're looking for wifi
phones that:
HAve a look at the Linksys WIP 300 (or something)
Can be charged from the USB port
I have a
Greetings!
Due to high workload, I am currently checking and responding to e-mail twice
daily at 12:00 PM EST and 9:00PM EST.
If you require urgent assistance (please ensure it is urgent) that cannot wait
until either 12:00 PM or 9:00 PM, please contact me via phone at: 305-338-3867.
Thank
Hello,
as from now on the snom 370 a special firmware exist to build secure
VoIP-Infrastructures via OpenVPN http://openvpn.net-Technology.
For further information go to http://snom.com/wiki/index.php/Networking/VPN
Note: *That is a pre-release, probably the software is still unstable*
It looks like you haven't install some mysql packages BEFORE make clean, make,
make install.
just install:
libmysqlclient15-dev
mysql-client
mysql-server
D
- Original Message -
From: Khaled Chehab
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Cc: [EMAIL
Hi,
I'm a novice trying out an experiment with Asterisk
and was unsure of the hardware needs for it.
I'm wondering if its possible to receive a call from
an external number (PSTN) say A. Then make a call to
another external number (PSTN) say B - and then bridge
the two calls so that A is talking
Hi,
I have put Digium TE120P card in PCI slot. So, lspci command gives the
information in followimg format.
02:0a.0 Ethernet controller: Unknown device d161:0120 (rev
11)
Following modules are running when seen through lspci
wcte11xp
Sorry to clutter up the mailiing list, but I've been unable to post to this
list for the past 2 WEEKS!
My ISP's blocking SMPT from other than his own servers.
I think I've worked around it. - But if I see this message in the digest
then I know I'm okay.
Again. - Sorry for any inconvenience.
Gary
Setting subject to a more descriptive one.
On Tue, Jun 26, 2007 at 03:35:12PM +0530, [EMAIL PROTECTED] wrote:
Hi,
I have put Digium TE120P card in PCI slot. So, lspci command gives the
information in followimg format.
02:0a.0 Ethernet controller: Unknown device
Dear ALL
I have audiocode MP -124 with configure in asterisk Endpoint
configuration means every analog phone register in asterisk now thing is that i
have one more SIP trunk with mediant 2000
[auodiocode-mp-124]-[ * ]--[mediant 2000]-E1
When i call from audiocode MP
Don Kelly wrote:
I've tried timing faxes two ways:
From a fax machine on a station port of an AltiGen PC/PBX served by an MCI
PRI calling back into the same PRI and reaching a RightFax server on a
station port behind the AltiGen.
From the same fax machine on the same station port of the AltiGen
Carsten Bock schrieb:
José Luis Ledesma schrieb:
In my asterisk 1.4.5 chan_features.so has been installed properly...
check in your asterisk-source if /channels/chan_features.so is present
regards,
Jack escribió:
Is chan_features.so deprecated for asterisk 1.4.5 or why is
Give the new site a break. I think it's a good idea. Sure there are lots of
news sites for VoIP, but many of them are poorly designed, and I can't
recall any that are very good at letting the users provide the news content.
I agree that the name could be better, but after having just tried it
I haven't been too impressed with the WIP330 - but my experience with
Linksys tech support has been disastrous!
I spent approximately 50 minutes on hold, I was transferred between 4
different people (all of whom had a poor grasp of the English language),
none of them understood the features of
I'm running Trixbox 1.2.3 with 2 TDM400s (FXOs).
With Trixbox out of the mix and a regular phone connected I get the CID
fine yet Trixbox shows 'unknown':
dialparties.agi: Caller ID name is 'unknown' number is 'unknown'
dialparties.agi: Methodology of ring is 'ringall'
Here is my Zapata.conf
Jack wrote:
Thanks for your answer and sorry for my late response.
So what does this exactly mean to me? Can I keep chan_features.so from
1.4.4? What consequences does it have when chan_features.so is disabled
und why has this been done? Is chan_features.so related to features.conf?
Whats wrong w/ voip-info.org?
Jon Weisman | Sales Engineer
International Bell Communications
www.ibell.net
- Original Message -
From: Alex Robar
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Tuesday, June 26, 2007 8:38 AM
Subject: Re: [asterisk-users]
John Faubion wrote:
by the way selling does not depend on the amount of lines you have and
we are very productive trust me
True, very true. There are lots of very productive sales people that don't
need a phone at all. From the paper boy to car dealers, lots of sales don't
I don't think anything is _wrong_ with VoIP-Info at all, I just think the
sites serve different purposes. This is all just personal preference, but to
me VoIP-Info does not work that well as a social news site, as all
stories/headlines, good or bad, have equal weight. With the Pligg system,
the
I am looking for a gateway that has several FXS ports and uses IAX. I have
a need for 16 ports, but will accept 6 or 8 port gateways as well.
-
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com http://www.ics-il.com
You do not need an L3 switch for this, just any managed switch which
does vlans
Unless there is something else?
On Jun 26, 2007, at 12:07 AM, Marty Mastera wrote:
Any recommendations on an economical layer 3 switch? Preferably
something that you have hands on experience with connecting to
Can you post your zaptel.conf so we can verify your timing settings?
---
Matthew Fredrickson
Software Engineer
Digium, Inc.
On Jun 25, 2007, at 11:10 PM, Don Kelly wrote:
I've tried timing faxes two ways:
From a fax machine on a station port of an AltiGen PC/PBX served by
an MCI
PRI
I posted this bug yesterday:
http://bugs.digium.com/view.php?id=10058
but really was hoping that one of you would be willing to try something
simple for me and reply back with your results.
Basically - I have run into a problem where Asterisk RFC2833 DTMF does not
seem to be compatible with
Jason,
I am at least having similar issues with rfc2833 DTMF:
http://bugs.digium.com/view.php?id=10058
On 6/20/07, Jason Ma [EMAIL PROTECTED] wrote:
Hi buddies,
I encountered DTMF issue when I tried to place call from x-lite to a
sip conference serice,here is the diagram.
Joshua Colp schrieb:
Jack wrote:
Thanks for your answer and sorry for my late response.
So what does this exactly mean to me? Can I keep chan_features.so from
1.4.4? What consequences does it have when chan_features.so is disabled
und why has this been done? Is chan_features.so related
Polycom Phones
1. New call
2. Press 9 access outside line
3. Dial Cell Number
4. Transfer the call that way.
Once you initiate a new call you will tie up the second line. Your asterisk
box will now be bridging the two lines. The lines will stay tied up until
the salesman drops the call.
One
Sorry, I forgot to mention that I want to route between VLANs without an
external router and do some simple ACLs to allow PCs on the data VLANs to
access the web interface of the Trixbox on the voice VLAN.
thanks
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On
This is a follow up to an earlier post.
Looking for a means to individualize incoming FAX, so as to distribute them
to the intended recipient.
While the PBX is based on Asterisk, it is not possible for me to enter the
box to modify things, to any great degree. I thank those who mentioned
Niothing, it just serves a different purpouse. A site like voip-info,
astrecipes, asterisk book wiki is for adding content; a site like digg is
more for pointing out things you find on the various sites and to share
them with other people.
l.
In data Tue, 26 Jun 2007 15:12:38 +0200, Jon
Doug Zingel wrote on Tuesday, June 26, 2007 4:39 AM
I'm wondering if its possible to receive a call from
an external number (PSTN) say A. Then make a call to
another external number (PSTN) say B - and then bridge
the two calls so that A is talking to B?
Yes, look at blind transfer or
On 6/26/07, Marty Mastera [EMAIL PROTECTED] wrote:
Sorry, I forgot to mention that I want to route between VLANs without an
external router and do some simple ACLs to allow PCs on the data VLANs to
access the web interface of the Trixbox on the voice VLAN.
thanks
The only reason to route
Does doing it this way give you the correct DST?
[macro-dialout]
exten = s,1,Goto(${MACRO_EXTEN},1)
exten = _X.,1,Set(TOUCH_MONITOR=${TIMESTAMP}_${CALLERID(num)}-${ARG1})
exten = _X.,n,NoOp(CID_NAME : ${CID_NAME})
exten = _X.,n,NoOp(CID_NUMBER: ${CID_NUMBER})
exten = _X.,n,NoOp(CID_CLIR :
This is usually a Cisco issue.
You need to set the Cisco to use RFC2833 DTMF. Check the Cisco docs.
tracinet wrote:
Jason,
I am at least having similar issues with rfc2833 DTMF:
http://bugs.digium.com/view.php?id=10058
On 6/20/07, Jason Ma [EMAIL PROTECTED] wrote:
Hi buddies,
I
Does anyone know if Asterisk can natively support the SVP protocol from
SpectraLink?
Thanks,
MD
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Jared:
As you see i have the s option. That works fine on Version 1.2. Let me
see config the call limit con sip channels it works.
Thanks.
On 6/25/07, Jared Smith [EMAIL PROTECTED] wrote:
On 6/25/07, Alvaro Parres [EMAIL PROTECTED] wrote:
I'm having the next problem, it appear that
To configure the Cisco for RFC 2833 add the following line to the desired
dial-peer
dtmf-relay rtp-nte
Hope this helps.
Ed Nuñez
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric
ManxPower Wieling
Sent: Tuesday, June 26, 2007 11:41 AM
To: Asterisk
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote:
This is a follow up to an earlier post.
Looking for a means to individualize incoming FAX, so as to distribute
them to the intended recipient.
While the PBX is based on Asterisk, it is not possible for me to enter
the box to modify things, to
Marty Mastera wrote:
Any recommendations on an economical layer 3 switch? Preferably
something that you have hands on experience with connecting to IP
phones with attached PCs? Specifically I need the ability to set the
VLAN in the phone to tag voice packets and to set a native VLAN on a
Can anybody at least point me in a direction??
On 6/25/07, Alex Mcdowell [EMAIL PROTECTED] wrote:
I don't think my cards are bad, but maybe there is a problem with the
one. It has been two weeks since I put my ticket in with Digium...and
still no word. I am starting to get frustrated.
On
If you really need layer3 support, I would steer clear of the Netgear.
I've had a lot of problems
with them, and the support was disappointing.
What model did you use? I've been very happy with the FS728TP as I mentioned
earlier. I haven't had any problems so far. Granted I haven't had to call
. . .
With all due respect, this project should be handed over to whomever has
authorization to administer the Asterisk box. We can tell you how to do it
in Asterisk, but if you can't take our advice, our ability to help you will
be severely limited.
Thanks. Point taken. I'm, unfortunately,
All,
I'm using a Digium TE410P w/ Asterisk 1.2.18. Trying to connect it to our NACT
STX switch via PRI, d channel is up, T1 shows normal, but I'm getting crazy
errors. I rewired this thing three times, then I connected the same cable from
the STX to a Cisco AS5300 (same pri settings as
I have seen this on cards waiting for the callerID and there being a problem
with the callerid signal. Is callerid working on theses lines?
On 6/26/07, Alex Mcdowell [EMAIL PROTECTED] wrote:
Can anybody at least point me in a direction??
On 6/25/07, Alex Mcdowell [EMAIL PROTECTED] wrote:
I
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote:
Thoughts vary to second T1, with channel bank, breaking out some DS0's into
a channel bank, or finding a T1/fax board (do they exist?), to go directly
into the FAX server (PC/linux based)
It looks to me like you have two choices. The first
First of all, Alex, sorry for not seeing your reply. Nearly two weeks
ago now :(
Honestly, with canreinvite=yes, I'm not sure what is meant by the
signalling still travels through asterisk... I would ASSUME that
includes out-of-band dtmf as well. Sorry!
Moj
Alex Crow wrote:
Moj,
Does
Daniel already pointed you in the right direction.
I have seen this error many times, but it never causes a problem.
Alex Mcdowell wrote:
Can anybody at least point me in a direction??
On 6/25/07, Alex Mcdowell [EMAIL PROTECTED] wrote:
I don't think my cards are bad, but maybe there is a
Buddies,
Thanks for you response.
I have resolved the issue,it was not the DTMF mismatch between
Asterisk and Cisco proxy.
In fact,there is a Convedia media box behind Cisco proxy as conference
bridge,after checked the whole trace through the patch,I found that my
asterisk send video codec
The only reason to route the voice VLAN is if you need the phones to access the
Internet and/or vice-versa. If you only need to worry about the computers on
the data VLAN accessing Trixbox's web interface, I would suggest using the
Ethernet VLAN capabilities of Linux. You can create eth0.vlan1
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote:
This install uses a Sangoma card.
Could you expand on redirect to a channel bank? Could you illuminate
the connectivity for me?
A single T1 connects to??? Is the Digium card smart as in, can it
break out DS0 line(s) on a second port (to go
. . .
It looks to me like you have two choices. The first you probably
can't do. That is, get a two port board in the Asterisk system with
the second T1 going into an Eicon Board in a Hylafax system. Then,
you can assign DIDs with whatever web interface you have on this
Asterisk system to
On 6/26/07, Marty Mastera [EMAIL PROTECTED] wrote:
The only reason to route the voice VLAN is if you need the phones to
access the Internet and/or vice-versa. If you only need to worry about the
computers on the data VLAN accessing Trixbox's web interface, I would
suggest using the Ethernet
I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server.
My PBX has experienced several core dumps the last couple of days and I am not
sure if this is what's causing it, but it always seems to happen when a
particular extension on a grandstream phone uses ChanSpy SIP group.
I
Has anyone successfully run * 1.4 with the following configuration (or
something very similar)?
HP DL380 G5 (3Ghz Xeon)
CentOS 4.5 (kernel 2.6.9-55)
Asterisk 1.4.5 (or 1.4.4)
Zaptel 1.4.3 (or 1.4.2.1)
TE412P
TDM400B (2x FXO and 2x FXS modules)
I've had this rig running * 1.2.18 with Zaptel
I would also like to know if Asterisk can be setup to automatically re
start if there is a core dump.
Sure! You should already have the required script. Just run it from
safe_asterisk. Here is a link with more info:
http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs
use the safe_asterisk script
it will restart asterisk if it crashes and it enables core dumps (your core
size limit is probably set to 0 when you start asterisk).
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ed Nuñez
Sent: Tuesday, June 26, 2007 2:22 PM
To: Asterisk
Well seems like I am already doing first method minus the extension. We
do have full features on our lines so both lines are free once the
transfer is complete. We also have toll calls on our lines so it would
not be a problem, so I do not have to worry about ATT dropping the
call. I tried to
On 6/26/07, Ed Nuñez [EMAIL PROTECTED] wrote:
I have not been able to locate where the core dump file is being saved. I
can't find it in my TMP directory.
Check the directory in which you're starting Asterisk. It doesn't
sound like you're using the Red Hat initscript to start Asterisk, so
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote:
One idea is to utilize DID, and have Asterisk forward the calls to the
current FAX lines, preserving the DID as Caller ID. I am fairly sure
Asterisk itself can do this. (The call would appear to be from this
assigned ID). If so, I could,
On 6/26/2007 at 3:04 PM, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote:
One idea is to utilize DID, and have Asterisk forward the calls to the
current FAX lines, preserving the DID as Caller ID. I am fairly sure
Asterisk itself can do
Ed,
I am having a problem with Asterisk frequently crashing on me as
well. I just run it under supervise:
http://cr.yp.to/daemontools/supervise.html
This way it will be restarted if svc determines it isn't running.
Eric
On Tue, 2007-06-26 at 13:22 -0500, Ed Nuñez wrote:
I am running
Hi,
Has anyone met any success, installing localized (ie non-english) menus
within SIP firmware enabled Cisco 7941 ?
Those phones seem to be trying to download localized menus from Cisco Call
Manager but as they are managed by an Asterisk server, I'm looking for a
workaround.
Any advice ?
I don't have caller ID at all, not on the verizon side and
usecallerid=no in zapata.conf. I do, however have the DSL on this
line. I have a splitter and then I have a filter on the asterisk side.
I am guessing this is the root of the problem. Thanks for any
insight.-Alex
On 6/26/07, Eric
On Tue, Jun 26, 2007 at 09:45:48PM +0200, Olivier wrote:
Has anyone met any success, installing localized (ie non-english) menus
within SIP firmware enabled Cisco 7941 ?
Those phones seem to be trying to download localized menus from Cisco
Call Manager but as they are managed by
Matt, thanks for pointing me to zaptel.conf; I'm new to Asterisk and can use
all the help I get!
Here are the non-comment lines from zaptel.conf (not set up by me):
span=1,1,0,esf,b8zs
span=2,1,0,esf,b8zs
bchan=1-23
dchan=24
bchan=25-47
dchan=48
loadzone = us
defaultzone=us
The first span is
On 6/26/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
On 6/26/07, Joe acquisto [EMAIL PROTECTED] wrote:
One idea is to utilize DID, and have Asterisk forward the calls to the
current FAX lines, preserving the DID as Caller ID. I am fairly sure
Asterisk itself can do this. (The call
2007/6/26, Steve Kennedy [EMAIL PROTECTED]:
On Tue, Jun 26, 2007 at 09:45:48PM +0200, Olivier wrote:
Has anyone met any success, installing localized (ie non-english)
menus
within SIP firmware enabled Cisco 7941 ?
Those phones seem to be trying to download localized menus from
Vadim Berezniker wrote:
use the safe_asterisk script
it will restart asterisk if it crashes and it enables core dumps (your
core size limit is probably set to 0 when you start asterisk).
*From:* [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Ed Nuñez
*Sent:* Tuesday, June 26,
Paul Hales wrote:
Any idea why I can't build HPEC for zaptel 1.2.18?
It builds fine with 1.4.3...
I don't know why this change just happened, but it has been fixed in
revision 2668 of SVN branch-1.2 of Zaptel.
You can fix it on your system by adding the following line:
Is there a way to let chan_sip skip host lookup?
Problem is I have to have a peer host config for every sip message outgoing.
For example, I cann't have this
in extension.conf
exten = 500,n,Dial(SIP/[EMAIL PROTECTED])
It'll return,
chan_sip.c:2738 create_addr: No such host: 192.168.1.79
when
On 6/26/07, Doug Zingel [EMAIL PROTECTED] wrote:
I'm wondering if its possible to receive a call from
an external number (PSTN) say A. Then make a call to
another external number (PSTN) say B - and then bridge
the two calls so that A is talking to B? What hardware
will I need to be able to do
I figured it out.
srvlookup=no
On 6/26/07, Lucian Romi [EMAIL PROTECTED] wrote:
Is there a way to let chan_sip skip host lookup?
Problem is I have to have a peer host config for every sip message
outgoing. For example, I cann't have this
in extension.conf
exten = 500,n,Dial(SIP/[EMAIL
http://www.abptech.com/support/qa/index.php?target=linksy_remote_p
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
Dupuis
Sent: Sunday, June 24, 2007 10:53 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users]
I am having a problem with Asterisk frequently crashing on me as
well. I just run it under supervise:
But that's just a band-aid. If it crashes, it takes all calls with it.
Hardly a good thing, unless you only have 1 call at a time -- then
it's probably no the end of the world.
I still don't
Hi,
I have been looking for an example of accomplishing this, but
I've been unable to locate something similar to what I'm trying
to do.
Here's the scenario:
Users caller ID is set to their internal extension (200-250).
This is set in sip.conf for each user. Each user has a local
When you see [ERROR] in the Message Log, either the MP firmware is buggy
or the far end is sending something out of spec in the SIP Message.
You'll need to upgrade to the latest MP firmware then report this to
whomever you bought it from. Or fix the far end to send the message in spec
or form
Hi All,
I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router
with a PRI card in it, handing off to a PBX and vise verse. Calls in
and out are working fine except for DTMF from Asterisk to the 2600.
DTMF from the 2600 to Asterisk is fine.
Here are the Asterisk console warnings I
Great examples Matthew, really appreciate it. This is exactly what I've been
searching for!
On 6/26/07, Matthew Brothers [EMAIL PROTECTED] wrote:
Hi,
I have been looking for an example of accomplishing this, but
I've been unable to locate something similar to what I'm trying
to do.
just an idea, but maybe qmail, samba, and bind have a smaller memory
footprint than an in-use asterisk? can you take the hardware offline
long enough for a memtest?
Moj
Luki wrote:
It's no unusual seeing uptime for say
qmail, samba or bind of 200+ days.
Do any one any clue.
This is what I need.
I have a Polycom 501 phone, which support multiple lines ie on the LCD you can
see the extensions asssigned to a user as.
555 --- Line 1 -- Extensions which registers with SIP(Asterisk) -- User A
8555 --- Line 2 -- Extensions which registers with
From what you provided, I'm not sure that 'firewall is disabled' will
help you. Your firewall probably needs to be configured to forward some
ports to the asterisk box's internal IP address. I usually do the
following:
For SSH connections to the box to manage it:
forward a.b.d.c's external
Sounds to me like inband vs rfc2833 issues.
I found that one has to use the same codec throughout in order to make
DTMF function and then use inband. This in turn forces you down the road
of alaw or ulaw codecs.
On Tue, 2007-06-26 at 18:01 -0500, JR Richardson wrote:
Hi All,
I have
It turns out that we were/are having trouble with out uplink
On Tuesday 19 June 2007 05:39:27 am Dave Bour wrote:
Have you tested the actual throughput on the link? What's it max out...
What kind of latency are you seeing as it gets loaded.
Can you do a local call to your own internal
See http://www.dyndns.com/services/dns/dyndns/
You can establish a name like yassir.dyndns.org that will point to the
dynamic ip address for your Asterisk server. You should be able to use this
for the domain for your VoIP service provider.
--Don
Don Kelly
PCF Corp
Real Support for your
I just had a similar problem and solved it with..
[from-internal-intldial]
exten = _+61X,1,Goto(from-internal,0${EXTEN:-9},1)
exten = _X.,1,Goto(from-internal,${EXTEN},1)
And put the E65 into the new contect. The first line stops all +61 (my
default country in australia)
The second
Hi,
I've experiencing this kind of problem.
Actually, my asterisk is running perfectly. I've tested it, and I called some
computer in my LAN. Then I enter the CLI and entered these commands
- show modules
- modules status (or so.. I forget)
- restart now
After I enter the last command, the
OCOSA ListAcc wrote:
Can you give an example of creating an extension which points to a cell
phone. Secondly how can you have if no one answers an extension it dials
the cell number next. That maybe answered in the example. I have the
system setup so it just dials out which ever line is
In Paul's defense, it looked to me like his original post was simply
a joke that was misunderstood. (I thought it was funny, anyway)
I have written a few jokes for this list over the years - it's nice to
know that some people find them funny.
PaulH
just an idea, but maybe qmail, samba, and bind have a smaller memory
footprint than an in-use asterisk?
No, probably not. Asterisk's is about 20-40 MB depending on the number
of extensions, etc. Smbd's is similar, bind's is actually 90 MB (with
about 600 zones).
can you take the hardware
Isn't this what you are looking for?
http://voipspeak.net/index.php?option=com_contenttask=viewid=72Itemid=28
On 6/26/07, Deepak Naidu [EMAIL PROTECTED] wrote:
Do any one any clue.
This is what I need.
I have a Polycom 501 phone, which support multiple lines ie on the LCD you
can see the
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