On Thu, Jan 26, 2006 at 03:10:09PM -0600, Mike Hammett wrote:
I'm running a VPS and I need to pass the device drivers from the
host OS to the VPS. What files do I need to pass through for
ztdummy to work? I'm assuming they're in /dev/zap, but I'm not
sure which ones are needed.
ztdummy
There is no such thing as a hard limit in asterisk. (Except for zap
channels, those are limited to 256 iirc).
With iax you can go higher, but the limit might be lower than 256 if you
are doing a lot of transcoding.
The limit depends on what exactly the server has to do with your call,
and
Do anyone know how to setup asterisk to authenticate the user through IP
rather than username and password?
I know most carriers will do that but smaller end user providers will not
do.
Sam
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Hi! For reasons that I won't bore people with, I'd like to disable
echo cancellation on-the-fly, depending on which DID a call came in
on. I've seen things like spandsp disable EC for faxes, so I know
it's possible. Any idea where to start looking? (I assume I'll have
to make a helper
Andrew Nowrot a écrit :
Hi,
Does anyone know what is the amount of max concurrent calls that can
be made in one Asterisk box?
I heard that it is 256 and it doesn't depend on how good your machine
is. It is the program constraint.
I wasn't aware of such limit and I seriously doubt it. Where
Hi
Some petty notes notes regarding the perl:
On Thu, Jan 26, 2006 at 11:23:27PM -0800, Jeremy wrote:
1. you didn't use strict and -w. Debugging will be a whole lot tougher
2. Consider using the nagging -T (taint mode), to explicitly know when
you trust the input.
3. Consider the latency this
/etc/init.d/asterisk stop
Stopping Asterisk PBX: .
censys:/usr/src/asterisk-8632# cd ..
censys:/usr/src# asterisk -vc
== Parsing '/etc/asterisk/asterisk.conf': Found
== Parsing '/etc/asterisk/extconfig.conf': Found
Asterisk SVN-trunk-r8620, Copyright (C) 1999 - 2006 Digium, Inc. and
Hi,
Has anyone implemented astpp? I'm configuring one right now and I have a
problem on the pricelist.
I followed the steps here
http://www.astpp.org/index.php?n=ASTPP.Installation and created tables
using http://www.astpp.org/index.php?n=ASTPP.Structure, but i didn't see
there a query on
Hi,Yeah, I think it was all about thew zap channelsBut what opportunities I have when I need to connect two or more Asterisk boxes. IAX, SIP or what?What is most efficient.CheersAndrew
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Hello list,
I've got a problem provisioning my snom 360's in the office (about 20 of them).
I'm using DHCP option 66/67 to set the provisioning URL but the phone
won't connect to it to retrieve it's configuration.
We are using a Cisco Catalyst Epress 500 to power the phones (poe), however if
i
I would like to add that I did have at one point
problems figuring out 4.0 and there were no problems
downgrading. Also I made a special email account
@mydomain for SNOM liscence's. This helps if at a
later dat you need to re-enter it again.
Regards,
Dovid
--- Christian Stredicke [EMAIL
Hello Sam,
use host=IP_ADDRESS when defining user in sip.conf
regards,
Umair Bari
On 1/26/06, Sam Tam [EMAIL PROTECTED] wrote:
Do anyone know how to setup asterisk to authenticate the user through IPrather than username and password?
I know most carriers will do that but smaller end user
Hi, Where are you pulling this number from? (other than the obvious traditional 2^8)?
That is not my imagination ;).Actually I talked with a guy who was one of the designers of Asterisk. He told me about this limitation but I don't know if he was talking about Zap channels only or in general. I
Hi,
We have a task to reduce voice call bandwidth. IP+UDP+RTP are using 40 bytes per
packet and for voice GSM FR 33 bytes. We are trying to reduce this bandwidth
accommodating multiple GSM frames in one packet. If we want to use per packet
10 GSM frames how to do this using asterisk? Assume the
Andrew Nowrot a écrit :
Hi,
Yeah, I think it was all about thew zap channels
But what opportunities I have when I need to connect two or more
Asterisk boxes. IAX, SIP or what?
What is most efficient.
Your question doesn't make any sense.
Tell us what you are trying to do and you might
Andrew Nowrot a écrit :
Hi,
Where are you pulling this number from? (other than the obvious
traditional 2^8)?
That is not my imagination ;).
Actually I talked with a guy who was one of the designers of Asterisk.
He told me about this limitation but I don't know if he was talking
Hello,
I'm trying to catch channel hangup in DeadAgi script. For example, A
calls to DeadAgi script which connects (Dial) to B. After Dial command
exits I need to identify where hangup came from: A or B. CHANNEL
STATUS returns 6 (Line is Up) regardless of who hungup.
In CLI show channels states
Hi all,
I'm running Asterisk SVN-trunk-r8643M and face following problem:
I'm trying to get incoming call from a provider and calls ended with a
404 error. On the INVITE I get Found no matching peer or user for IP
address:5060 and then Looking for UserName in SIP default context
(domain
This is not a problem of the ISDN line (or chan_capi), Asterisk is just
not doing anything after
-- Executing GotoIfTime(CAPI/ISDNL1/5912211-0,20:01-7:59|mon-sun|*|*?9)
in new stack
and without further commands (like Ringing(), Answer(), ...) the ISDN line
timed out and disconnects.
So
I've installed the last released asterisk 1.2.2 on my own HLFS system
with a 2.6.14.3 kernel. I've also a 2 FXO/ 1 FXS digium card on it.
Every thing is working correctly.
For ODBC, I'm using UnixODBC with pgsql. The voice messages are
correctly written to the database and also their number is
Ronald Ramos wrote:
Hi,
Has anyone implemented astpp? I'm configuring one right now and I have
a problem on the pricelist.
I followed the steps here
http://www.astpp.org/index.php?n=ASTPP.Installation and created tables
using http://www.astpp.org/index.php?n=ASTPP.Structure, but i didn't
Hi, It does sound like a typical case of urban legend, where Zap is limited
to 256 channels becomes Asterisk is limited to 256 channels. Asterisk!= Zap.I've never said that Asterisk is limited to 256 channels. I only asked a question. That is the main reason of this list isn't it?
But leave the
I need to connect two (or more) asterisk boxes. They will exchange a
lot calls. What is the best approach? Which protocol should I use IAX
or SIP or what? I never did that so first I want to ask people who
have some experience.
If you're connecting Asterisk boxes between each other, it
HiI had same problems yesterday but ts fine now,DanOn 27/01/06, Mark Adams [EMAIL PROTECTED]
wrote:
I would expect a reply in about 4-5 days …
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
Angelito Manansala
Sent: Thursday, January 26, 2006
8:44 PM
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen
Sent: 27 January 2006 08:21
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] paging agi
Hi
Some petty notes notes regarding the perl:
SNIPPED
Hi,
I'm using asterisk 1.2.1.
Is there anybody out there who knows what this warning means?
*WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer:
image 5004 udptl t38*
Google does not help at all.
TIA
Giorgio Incantalupo
___
Hi,
I'm currently in the process of building
Asterisk for our new office and have hit a snag. We need two internal
Analog lines for a modem and fax machine. Am I right in thinking
I can use two ATA's, one on each piece of equipment which will then talk
to Asterisk and route via our ISDN30?
If
HiIn my environment I have to connect 6 * boxes with each other so IAX is probably the best solutionThanksCheers
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Hi all,
When I do outgoing calls via my FXO card (TDM400, analog line), they get
always marked ANSWERED in my CDR. I guess it is not that easy for fxo
to determine if there is actually a call or just ringing.
But anyway, is there a way to get this working right?
Thanks in advance,
Henry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Jean-Michel Hiver
Sent: Friday, January 27, 2006 10:49 AM
If you're connecting Asterisk boxes between each other, it
would make sense to use IAX as it's Asterisk's 'native' protocol.
...
So I
Hello all,
I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal
analogue lines. The same number is assigned to these lines. These lines are
connected to 2 spa3k registered to my asterisk box.
When calls arrive, TR1 try to pass call to the first spa. If spa not takes
the call
Since I passed from version 1.0 to the 1.2.3. I have Pb with the
callerid. If somebody call with presentation of the number all is well.
If somebody make call in masked number, i couldn't send a callerid to
the phone.
It is in a call center and i use the callerid to present the name of the
Hi Phil,
if you want to use ATAs take a look at grandstream site...they are
better than digium but you could use a card, TDM400 is excellent for
analog lines and devices.
Giorgio Incantalupo
[EMAIL PROTECTED] wrote:
Hi,
I'm currently in the process of building Asterisk for our new
hello all,
i have a * 1.2.1, in a lab, only for test,
with 4fxo clone - md3200 - intel537, connect to pstn.
All work well, but, 1 once day 2 of this cards,
stop make call, and receiv call thought.
i kill the asterisk, remove modules, wcfxo and zaptel,
mount the modules again,
I didn't find that exact message in the RFC's, but I did find something
similar in RFC 3407 (http://www.rfc-archive.org/getrfc.php?rfc=3407),
a=cdsc: 4 image udptl t38
Which means that the sender is capable of sending T.38 fax over UDP.
I wouldn't worry about it unless you were trying to
Hi asterisk and ser users,
Is there a solution to monitor asterisk and ser with
snmp ?
Regards
Harry
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What caller id method is used in spain? Is it before or after the ring.
If you can set the ISDN termination box for UK caller id then the ID is sent
before the first ring.
on the sipura thats ETSI FSK with PR(UK)
Chris
- Original Message -
From: Manuel Dominguez [EMAIL PROTECTED]
Could someone please outline the differences between:
allowed_not_screened: Presentation Allowed, Not Screened
allowed_passed_screen : Presentation Allowed, Passed Screen
allowed_failed_screen : Presentation Allowed, Failed Screen
allowed : Presentation Allowed, Network
Sean Cook wrote:
Ok... I am having a serious brain fart this evening. IIRC, the next sip
draft addresses shared lines and I thought I remembered something on the
list about support for it in the near future.
'the next sip draft'? There are probably 150+ IETF drafts circulating
regarding SIP
Hi.
Use massdeployment for putting the licenses on to your phones.
There is a setting called license_url you can use like the firmware update
URL, the macro {mac} will be replaced by the MAC address of the phone. So if
you provide the setting like this:
license_url:
Hi I'm looking for a pinout for the above. Note this has
what i'd call
RJ45 sockets (or someone smart can correct me). I need to
plug into BT (rj13?).
Are you sure the TDM400 has RJ45 sockets? The pair I've got here have RJ12
sockets.
I assume with the mention of BT,
it means that your sender is capable of sending t38, but asterisk (without at
a minimum the t38-patches for passthrough) is not capable of handling this.
if you have reinvite for this channel allowed and your sender can send the
fax over g711 asterisk will send a reinvite and the fax has a
[EMAIL PROTECTED] wrote:
Hi,
I'm currently in the process of building Asterisk for our new office and
have hit a snag. We need two internal Analog lines for a modem and fax
machine. Am I right in thinking I can use two ATA's, one on each piece
of equipment which will then talk to Asterisk
Phil
I have very good experience
with the vegasteam ATAs devices.(you might also want to look @ sipura
ATAs, since vegastream is doing an oem on there boxes)
They support modem until
v.90 speeds and faxes for g3.
They are expensive, and
again, work great and configure very easy
Ronald Wiplinger wrote:
does still not do the trick!
Show your Dial command from extensions.conf file.
--
Best regards,
Bartosz Piec
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There is no doubt that given a particular scenario, anything won't work
properly. This is not necessarily a problem with the SPA3000 or the TDM
cards, this is much more of a phone line issue. Granted, those devices don't
handle line issues as well as some other devices (such as the long
Henry Margies wrote:
Hi all,
When I do outgoing calls via my FXO card (TDM400, analog line), they get
always marked ANSWERED in my CDR. I guess it is not that easy for fxo
to determine if there is actually a call or just ringing.
But anyway, is there a way to get this working right?
If
[EMAIL PROTECTED] wrote:
Hi,
We have a task to reduce voice call bandwidth. IP+UDP+RTP are using 40 bytes
per
packet and for voice GSM FR 33 bytes. We are trying to reduce this bandwidth
accommodating multiple GSM frames in one packet. If we want to use per packet
10 GSM frames how to do
Have any providers started to offer T.38 yet? I am anxious to find a
solution for faxing.
--
Chris Mason
--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.
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Whatever happened to Google? why don't people use that?
Tha actual limit according to Google/wiki is/was 255 for zap channels:
http://voip-info.org/tiki-index.php?page=Asterisk+dimensioning
However, in that same post someone corrected it that it is no longer limited.
On 1/27/06, Andrew Nowrot
Hi I basically allow=all and NAT=no for all the phones. but still can't see why I can't receive calls (i.e. in-bound)but I can make outbound calls. also there is no debuging on pbx for sip (unless it's outbound call). do you have anymore advice?thanks AmaMd Sani Johari [EMAIL PROTECTED]
Ronald Wiplinger wrote:
exten = 600,1,Dial(${PHONE_LOCAL},60,tr)
Type this:
exten = 600,1,Dial(${PHONE_LOCAL},60,tTwWr)
dial at 600 and see if this helps. If so, change all commands in that
way (tT is for transfer, wW is for recording).
You must also have sox installed for calls
On Friday 27 January 2006 13:29, Manuel Dominguez wrote:
Hello all,
I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal
analogue lines. The same number is assigned to these lines. These lines are
connected to 2 spa3k registered to my asterisk box.
When calls arrive, TR1
Hi Guys,
We are using Grandstream BT-102 phones internally to talk directly to
our SIP provider's SIP server. Each of our phones is configured with a
CLI provided by our SIP provider.
I have a couple of spare phones and about 5 spare CLI's, so I decided to
set up AsteriskAtHome to see what
On 1/27/06, Chris Mason (Lists) [EMAIL PROTECTED] wrote:
Have any providers started to offer T.38 yet? I am anxious to find a
solution for faxing.
commpartners does offer it. I haven't personally used it yet, but I
know they offer the service.
--
Bird's The Word Technologies, Inc.
Hi all
I have installed AAH 2.2 in my P4 PC
following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp
and made as per the guide says
and downloaded SJ Phone, and registered user
and when i try to dial the 19197543700
i get message that, all circuits are busy now, please try
there is no error message coming up on the pbx for in-bound calls (there is only debugging messages for outbound calls).thanks in advance for any hint or suggestion. AmaI just post my configuration file here for sip phone:
Bartosz Piec wrote:
Ronald Wiplinger wrote:
exten = 600,1,Dial(${PHONE_LOCAL},60,tr)
Type this:
exten = 600,1,Dial(${PHONE_LOCAL},60,tTwWr)
dial at 600 and see if this helps. If so, change all commands in that
way (tT is for transfer, wW is for recording).
You must also have sox
It does use the same kernel for everything. It's a specially modified
kernel for the VPS support. I guess the only way to see if ztdummy works in
the VPS is to try it.
Mike Hammett
Intelligent Computing Solutions
http://www.ics-il.com
- Original Message -
From: [EMAIL
Can anyone shed some light on rules that might make the task of
parsing the country code and city codes from a dialed number in the
CDRs?
I know that there is almost never a case where a concatenated country
and city code could overlap with another country code, but what about
city codes and
Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension 1055.
Outbound calls to 1055 work perfectly.
Inbound calls
Aha, I see it's 4.1, cool. So I just have to do a straight upgrade to 5.0
and I have this new toy to play with, correct?
-Original Message-
From: Sven Fischer (support) [mailto:[EMAIL PROTECTED]
Sent: Friday, January 27, 2006 5:27 AM
To: Asterisk Users Mailing List - Non-Commercial
On Fri, January 27, 2006 15:13, ram said:
Hi all
I have installed AAH 2.2 in my P4 PC
following AAH handbook PDF and http://mundy.org/blog/index.php?p=62#amp
and made as per the guide says
and downloaded SJ Phone, and registered user
and when i try to dial the 19197543700
i get
On Fri, January 27, 2006 16:09, Ian Cowley said:
Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg 10.10.10.10
It is NAT'd 1:1 via Checkpoint firewall to external public IP eg
50.50.50.50
The remote IAX2 phone (ATCOM320) is configured to call 50.50.50.50 on
extension
Hi,
I've ordered a few IP501s from PC Connection, basically since we have an
account with them. I like the phones for what they do, and now would like
establish a relationship with a reseller that can give us maintenance and
access to the most current firmware.
What are some good resellers out
Ram,
On my AAH the stock dial plan requires a 9
first. For kicks, try dialing 919197543700 and see what you get.
-MC
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ram
Sent: Friday, January 27, 2006
6:14 AM
To: Asterisk
Users Mailing List - Non-Commercial
We use APIC on all servers, so interrupt sharing is not an issue :)
On Jan 26, 2006, at 3:02 PM, Damon Estep wrote:
And in some (many) cases it will do so while sharing an interrupt
with a
NIC and disk controller!
We run sangoma a104 cards in Dell SC1425 1U servers with great success
under
hello,
can someone help me with ser redirect to asterisk.
any help appreciated.
Thanks,
AA
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-Original Message-
From: Ian Cowley
Sent: 27 January 2006 15:10
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] External IAX2 phone defined as internal
behaving as from PSTN
Have [EMAIL PROTECTED] 1.2.1
The server is on an internal network eg
I have installed a Digium card TE210P and unicall for use MFC/R2. I think
that it´s all right but I can´t make and receive calls. I´m using asterisk 2.1
with the patch made by José P. Leitão andthe follow libs:
libsupertone-0.0.2libunicall-0.0.3libmfcr2-0.0.3zaptel
2.1
My number is
On 1/27/06, Kevin P. Fleming [EMAIL PROTECTED] wrote:
Sean Cook wrote:
Is there an implementation for shared line support in asterisk? I know
that with hint I can watch line status... I just want to be able to
pick up on an extension when ringing or jumping in on a call by punching
the
Iax.conf
[general]
;bindport = 4569 ; Port to bind to (IAX is 4569)
bindport = 5036 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=g729 ; 4 simultaneous allowed
allow ilbc ; prefered for iax2
allow=gsm
Hi,
I like to forward an incoming call on an ISDN line to my mobile phone.
Since ISDN offers two channels, I thought that this should work, but Asterisk
tells me, that there is no channel available.
There is no one else using this line, so guess I made a mistake in the
configuration or it might
I have an analogue trunk to an ATT Definity.
It has a DISA context defined.
From a Definity handset call the analogue port extension 1008 and wait
for dial tone from asterisk. It takes between 34 rings.
Likewise from Asterisk SIP handset PBX Access NoPBX Extn takes
nearly 10 secs to ring.
Is
I've been running 1.6.4.0064 for the last few weeks..
I've had no problems with it, I haven't done a whole lot of speaker
phone with it yet though.. Once my IP4000 reboots It'll be running it as
well so that will be a good test.
Chad
-Original Message-
From: [EMAIL PROTECTED]
To which context of the dial-plan does asterisk tries to
match incoming calls when acting as a sip client?
To be more specific:
In extensions.conf Under which context should I place
exten = 648064,1,Dial(TECH/peer)
for an entry like this register = 648064:[EMAIL PROTECTED]/648064 ?
On Fri, January 27, 2006 17:23, Ian Cowley said:
Iax.conf
[general]
;bindport = 4569 ; Port to bind to (IAX is 4569)
bindport = 5036 ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=g729 ; 4
I've been running 1.6.4.0064 for the last few weeks..
I've had no problems with it, I haven't done a whole lot of speaker
phone with it yet though.. Once my IP4000 reboots It'll be running it as
well so that will be a good test.
Which bootrom version are you using?
-Ron
The card is telling:
CAPI INFO 0x34a2: No circuit / channel available
so the other channel must be in use by something else.
Maybe another device on the ISDN line?
Armin
On Sat, 28 Jan 2006, Ralf Mueller wrote:
Hi,
I like to forward an incoming call on an ISDN line to my mobile phone.
From The CLI with iax debug (IP address faked)
asterisk1*CLI
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
NEW
Timestamp: 0ms SCall: 22774 DCall: 0 [99.99.99.212:1720]
CALLING NUMBER : 1055
CALLING NAME: 1055
FORMAT : 256
CAPABILITY
Hi,
I'm trying to configure some Quality Of Service among an Asterisk server
with RedHat3 and some IP phones on my LAN.
I read about 802.1p (level 2) QoS, using 3 bits of VLAN tag.
Two questions:
- do I need to use tagged links (trunks) end-to-end? In other words, do all
ports on all switches
Hi Gavin -
I've ordered a few IP501s from PC Connection, basically since we have an
account with them. I like the phones for what they do, and now would like
establish a relationship with a reseller that can give us maintenance and
access to the most current firmware.
What are some good
Hi
all of them thanks for the quick reply
i was tried adding 9 as well as 00
but i get number invalid if i put any of the digits
what kind of config files need to post here to resolve the problem
please assists
ram
On 1/27/06, Michael Collins [EMAIL PROTECTED] wrote:
Ram,
On my AAH the
Start with extensions.conf and also the
debug lines from the Asterisk console.
-MC
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ram
Sent: Friday, January 27, 2006
9:54 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: Re:
Hi -
I'm running 1.6.2.0041 according to my phone.
Which firmware worked for you?
It was the old firmware from when we first got the phones actually.
1.4.x I think. Then I read that they fixed the CID issue and decided
we needed an upgrade. I tried it out on my phone, but didn't really
if you are using AAH, please post extensions.conf,
extensions_additional.conf - also send us more info on your phones.
thanks
rajeev
ram wrote:
Hi
all of them thanks for the quick reply
i was tried adding 9 as well as 00
but i get number invalid if i put any of the digits
what kind of
Hi, I've got the exact same problem here with Asterisk 1.2.1, at 2
locations.
The first one have 8 sipura 3000 with 5 pstn lines and 8 standard phones.
There is also 4 Mitel 5215 phones
The secon one have 8 sipura 3000 with 5 pstn lines and 8 standard phones.
There is also 8 Mitel 5215 phones
Hi rajeev
i have posted the extension.conf before
now iam posting extension_additional.conf
[EMAIL PROTECTED] asterisk]# more extensions_additional.conf[globals]#include globals_custom.confVM_PREFIX = *RINGTIMER = 15REGTIME = 7:55-17:05REGDAYS = mon-friRECORDEXTEN =
PARKNOTIFY = SIP/200OUT_2 =
Hmm.. I definitely have type=friend in the sip.conf and I added
qualify=yes but, I think that's default anyways.. When I call from the
outside and enter his extension it goes through to him fine but, when I
go extension to extension it automatically goes to voicemail.. Here are
the messages from
Hi, we have the same problem here at 2 location that we just installed
Asterisk 1.2.1
P4 3.0Ghz
Motherboard ASUS P4S800-VM
2 SATA disk in software Raid-1
We use 2 nic, one (onboard) to talk to the network (1Gbps link that we use à
100Mbps) and the other realtek 8139 from Startek that talk to
Eric Bishop wrote:
Do you have step by step instructions on how you created these RPMs. I
would like to create a few of my own but compiled for my own custom
kernel and patchea and am not very familiar with RPM packaging
A good starting point is to download and install the source RPMs in:
I have had no problems running the Sip.cfg from 1.5.2 with 1.6.4 so far,
but I am looking to update in the next while.
Chad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah
Miller
Sent: January 27, 2006 1:07 PM
To: Asterisk Users Mailing List -
Editing subject line to reflect current status.
On 1/26/06, Nilesh Londhe [EMAIL PROTECTED] wrote:
Since T616 is not answering (and incoming calls are going to Cingular
voicemail after 30 sec,) I suspect the problem focus area is...
-- Executing Answer(BLT/T616, ) in new stack
Is
Is anyone using Asterisk (and Festival) to make calls to appropriate
persons (techs, etc. ) when Nagios generates a particular type of alert?
If so, I would love to hear how people are doing it.
Thanks,
--
Darrell S. Long
BestWeb Corporation
I've got an SPA-841 SIP hardphone connecting to my asterisk server across
the internet through a couple of NAT routers. Everything works great (I can
initiate calls, receive calls, hear audio both ways, etc.) except for one
thing. When I hang up the phone, the connection in asterisk doesn't
amen! http://www.tritechcoa.com/ is a great supplier :)
Noah Miller wrote:
Hi Gavin -
I've ordered a few IP501s from PC Connection, basically since we have an
account with them. I like the phones for what they do, and now would like
establish a relationship with a reseller that can give us
I have used both, just not together. I have a possible idea though.
If they're running on separate servers, you can have nagios send an
email that the asterisk server receives. Have different email aliases
for different alerts, or have a script parse the email to see what
kind of alert it is.
Accounts 3-4 are disabled, Account 1 is the only account.
Thats it. Nothing special. If you have problems, try doing *70.
-Kerry
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
tracinetSent: Wednesday, January 25, 2006 8:57 PMTo:
Asterisk Users Mailing List
I tried through voipdiscount as well.
Even my older account through voipbuster started to behave this way and it
used to be ok on IAX.
I would expect at least some reply.
Rudolf
- Original Message -
From: Aryanto Rachmad [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
Boy oh boy. This blows. I upgraded to 1.2.2 from 1.0.9, and of course had
the timebomb bug. Immediately after upgrading to 1.2.3 we were ok, for 24
hours or so.
Since upgrading to 1.2.3, though, the whole system has locked up twice. Once
on Thursday, and then about a half hour ago. The server
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