Mike Fedyk wrote:
Matt Riddell wrote:
I would instead recommend the SuperMicro 1U servers - we have had a
really
great run with these.
Do you use Opteron or Intel?
I would not suggest that Supermicro are in Intel's pocket, so they must
have had their fingers in their ears going,
hi,
i have an issue that when making a call from a SIP phone going as follows:
phone -- asterisk -- cisco(192.168.0.1) -- terminating voip
platform(10.0.0.1)
i get the cisco sending up an invite to the voip platform followed
directly with a CANCEL message, as follows:
Via: SIP/2.0/UDP
Hi all,
Anyone using their services ?
I'm thinking of setting up my servers with their service.
But before starting to mess with my extensions.conf I thought let's check
the community for their experience.
Thanks,
Michiel van Baak.
___
--Bandwidth
On 06/01/06, Michiel van Baak [EMAIL PROTECTED] wrote:
Anyone using their services ?
I'm thinking of setting up my servers with their service.
But before starting to mess with my extensions.conf I thought let's check
the community for their experience.
I use them - the service works exactly
I just signed up for an account with them yesterday. I need to
configure my asterisk box for my needs and will test them out. I will
post to this thread as well as the wiki after a week or two of testing.
Thanks,
Steve
Hi all,
Anyone using their services ?
I'm thinking of setting up my
Not really, their suggested retail price is USD 300 for the analog unit,
probably because of the intelligent stuff in the box (which we do not
need when using *).
At USD 300 you can find SIP capable devices, for an analog unit the SIPCE
is 3x more expensive than the unit we were discussing.
Michiel van Baak wrote:
Hi all,
Anyone using their services ?
I'm thinking of setting up my servers with their service.
But before starting to mess with my extensions.conf I thought let's check
the community for their experience.
I don't use them from asterisk, but I do use their SMS service
bchan=1-5,7-15,17-31
dchan=16
Why are you excluding channel 6?
jvb
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Remco Barende wrote:
Not really, their suggested retail price is USD 300 for the analog
unit, probably because of the intelligent stuff in the box (which we
do not need when using *).
At USD 300 you can find SIP capable devices, for an analog unit the
SIPCE is 3x more expensive than the unit
HI , I installed asterisk in fedora core 3 machine perfectly. and i have 10 units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do any configurations of any files .What are the
lukeuse the wiki.
(always wanted to do that)
http://www.voip-info.org/wiki/view/Asterisk+phone+grandstream+budgetone
hope this helps,
yair
On 1/6/06, luke devon [EMAIL PROTECTED] wrote:
HI ,
I installed asterisk in fedora core 3 machine perfectly. and i have 10 units
of GrandStream IP
Hi!
The PBX I'm getting ready to replace has a really nifty feature -- one
that I'm not even sure Asterisk -can- do -- though I'm hoping to be proven
wrong. When a call goes to voicemail, the end-user can listen to the VM
as it's being recorded, and can interrupt and answer the call if it's
hi all, I am newbie to asterisk. I have installed asterisk based VoIP gateway in my LAB. Now i want to how many simultaneous calls (internal and external) can this gateway can handle? hereby i m sending my system details: 1) asterisk gateway is running on P-IV 2.6GHz machine. 2) i have
Hi,
I'm just in the process of replacing a crappy Siemens PBX with a new and
shiny Asterisk system. To connect Legacy equipment I hooked up a small
ISDN PBX (DeTeWe OpenCom 36) to one port on a Junghanns.net quadBRI
card. That port is configured for NT Point to Multipoint
(Mehrgeraeteanschluss)
OK all. I need some help. Looking to deploy asterisk servers and want
to get a recommendation on what server to buy. I love Dell's, but from
what I see on the list they seem to have some issues. I would like to
stay with one brand and need systems for small offices (20 users),
medium (50
I don't get it. What is the advantage of using a GSM gateway? VOIP calls are
pretty inexpensive as they are now. Is the use of a gateway intended as a
backup incase a wired network connection goes down? I have being looking
around the net for information on this. Anyone out there using it and if
On Fri, 2006-01-06 at 07:35 -0500, JCC wrote:
I don't get it. What is the advantage of using a GSM gateway? VOIP calls are
pretty inexpensive as they are now. Is the use of a gateway intended as a
backup incase a wired network connection goes down? I have being looking
around the net for
VOIP - GSM calls may be cheap if you call to China.
When you call a cell in The Netherlands it will cost you USD 0.25 per
minute. I am located in NL therefore a lot of calls go to NL mobiles.
You can buy sim cards that offer minutes for USD 0.02 per minute, if you
can recommend a carrier
Hi all
I need to configure call forwarding for particular
extension is busy.how to configure this my extension
configuration is like following.
exten = 2006,1,Dial(SIP/sipura2)
regards
ramakrishnan.n
__
Yahoo! DSL Something to
We use Dell PE 1650 upto 2850 servers for all of our Asterisk and SER
applications and they work just fine. Not sure what others are experiencing
but our systems have been rock solid.
-Steve
B. Keith Murphy wrote:
OK all. I need some help. Looking to deploy asterisk servers and
want to get
exten = 2006,2,goto(s-${DIALSTATUS},1)exten = s-BUSY,1,DIAL(SIP/sipura3)exten = s-NOANSWER,1,exten = s-www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
Cheers,Giovanni Miano2006/1/6, nr k [EMAIL PROTECTED]
:Hi allI need to configure call forwarding for particularextension is
busy.how
I don't get it. What is the advantage of using a GSM gateway?
VOIP calls are pretty inexpensive as they are now.
It largely depends on the country you're calling. Here in the UK, calls to
mobiles are maintained at an artificially high rate because the terminating
network (the mobile networks)
call deflection does not work with bristuffuse CAPI2006/1/6, Pisac [EMAIL PROTECTED]:
Do bristuff/zaphfc support CD (Call Deflection)?How to deflect call (transfer before answering) with bristuff?
___--Bandwidth and Colocation provided by Easynews.com
I can't say for sure that it's 10.. but it's somewhere between 8 and
13 as I hit * to cycle.. when I get up in that range... it will stop
spying.. and asterisk will stop taking calls until I do a restart.
On 1/5/06, Tom Vile [EMAIL PROTECTED] wrote:
I have not had that issue. Are you saying 10
Hello,
It didn't work...
I used "Data: SIP/dov.bigio-9949" which was the
channel being used, and the call I received just had beeps... no
conversation.
According to the documentation on (http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy),
ChanSpy doesn't take a channel as parameter, does
Hi,
Recently the Dutch ISP Xs4all started a SIP based VoIP service with free
087 numbers to their subscribers. Has anyone been able to get this
service to work with Asterisk? So far I had no luck. It seems they use
MD5 authentication with a realm of sip.xs4all.nl. And for those
interested: they
Hello,
Giovanni Miano schrieb:
call deflection does not work with bristuff
this is no longer true - at least not when using a recent bristuff version and a
point-to-multipoint trunk.
exten = 37,1,Wait(0.5)
exten = 37,2,ZapCD(destination-number)
exten = 37,3,Progress()
exten = 37,4,4,Hangup
Lol, so Dell must be doing the same thing.
Did you ever consider that Supermicro are an enterprise setup to make
money, and that possibly their financial interests are served by
sticking with Intel?
You would have to figure that Dell is doing something right to get to
the size they currently
On Friday 06 Jan 2006 00:46, A_ Navone wrote:
make[2]: *** [obj_linux_x86_r/simph323] Error 1
make[2]: Leaving directory `/usr/src/openh323/samples/simple'
make[1]: *** [opt] Error 2
make[1]: Leaving directory `/usr/src/openh323'
make: *** [optshared] Error 2
any idea ?
None unless
On Fri, Jan 06, 2006 at 01:23:26PM -, Chris Bagnall wrote:
I don't get it. What is the advantage of using a GSM gateway?
VOIP calls are pretty inexpensive as they are now.
It largely depends on the country you're calling. Here in the UK, calls to
mobiles are maintained at an
On Friday 06 Jan 2006 08:11, Richard Scobie wrote:
Mike Fedyk wrote:
Matt Riddell wrote:
I would instead recommend the SuperMicro 1U servers - we have had a
really
great run with these.
Do you use Opteron or Intel?
I would not suggest that Supermicro are in Intel's pocket, so they
Here is the issue:
[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# lsmod | grep zaptel
zaptel206724 0
crc_ccitt 2113 1 zaptel
[EMAIL PROTECTED] ~]#
[EMAIL PROTECTED] ~]#
[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is
Hi,
Im a new user of Aterisk, and I have to
configure a VoIP Gateway.
I have an Alcatel PBX with an E1 card, connected, for
the moment, to a local carrier.
I would like work with a french VoIP provider, but,
for this, I need to use a VoIP Gateway for connect my E1.
Thus, I want to
I work for a call center and we are looking at using asterisk to have
our operators take calls. Our message taking software records all the
calls on the operators computers. Right now we use these recording
controls from radio shack that plug in between the wall jack and the
phone and plug in
Tom wrote:
Here is the issue:
[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# lsmod | grep zaptel
zaptel206724 0
crc_ccitt 2113 1 zaptel
[EMAIL PROTECTED] ~]#
[EMAIL PROTECTED] ~]#
[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice:
Is anyone aware of the details of this in Australia?
I'd love to be able to let tech's have calls route straight to their
mobiles when 'in-house'
Steve Kennedy wrote:
On Fri, Jan 06, 2006 at 01:23:26PM -, Chris Bagnall wrote:
I don't get it. What is the advantage of using a GSM
With our current pbx system, a call comes in from the PSTN to the
receptionist. She then hits flash, which puts the caller on hold, calls
my extension, says so and so is on the phone for you, I say ok put
him through, she hangs up and I am connected to the caller.
With [EMAIL PROTECTED] I can
However there are some disadvantages, the main being you cant set CLI of
the outgoing call as it will always be tied to the SIM of the mobile
terminal.
That's true. You can however choose to mask the caller ID.
Another is that you can NOT run a GSM gateway (as they're known) for 3rd
In article [EMAIL PROTECTED],
Tom [EMAIL PROTECTED] wrote:
Here is the issue:
[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# lsmod | grep zaptel
zaptel206724 0
crc_ccitt 2113 1 zaptel
[EMAIL PROTECTED] ~]#
[EMAIL PROTECTED] ~]#
[EMAIL
Apparently we've been having calls sporadically drop. We're using an
IAX outbound trunk and SIP adapters on the inside.
Below is a log excerpt detailing one of the calls which dropped, and it
looks largely normal to me except for this:
Jan 5 13:31:07 DEBUG[3776] channel.c: Didn't get a
On Fri, January 6, 2006 15:37, Michael Sampson said:
I work for a call center and we are looking at using asterisk to have
our operators take calls. Our message taking software records all the
calls on the operators computers. Right now we use these recording
controls from radio shack that
Look for the option of attended transfer.
On 1/6/06, Michael Sampson [EMAIL PROTECTED] wrote:
With our current pbx system, a call comes in from the PSTN to the
receptionist. She then hits flash, which puts the caller on hold, calls
my extension, says so and so is on the phone for you, I say ok
On Fri, January 6, 2006 15:46, Michael Sampson said:
With our current pbx system, a call comes in from the PSTN to the
receptionist. She then hits flash, which puts the caller on hold, calls
my extension, says so and so is on the phone for you, I say ok put
him through, she hangs up and I am
This can be accomplished in the DP with ChanSpy, and this bug:
http://bugs.digium.com/view.php?id=5841
On 1/6/06, Philipp von Klitzing [EMAIL PROTECTED] wrote:
Hi!
The PBX I'm getting ready to replace has a really nifty feature -- one
that I'm not even sure Asterisk -can- do -- though I'm
With our current pbx system, a call comes in from the PSTN to the
receptionist. She then hits flash, which puts the caller on hold,
calls my extension, says so and so is on the phone for you, I say
ok put him through, she hangs up and I am connected to the caller.
With [EMAIL PROTECTED] I
Grandstream has been very well detailed on the
wiki.
www.voip-info.org
- Original Message -
From:
luke
devon
To: Astericks
Sent: Friday, January 06, 2006 6:10
AM
Subject: [Asterisk-Users] Budge Tone-100
as a Ext in the LAN
HI ,
I installed asterisk
Asterisk has call recording capabilities built in. it will offer you far
more functionality than what you currently are using (better control,
archiving and ability to export to third party analysis).
I suggest you do some research on this area of asterisk capability and
then suggest to the call
On Fri, Jan 06, 2006 at 06:48:27PM +0400, Jean-Michel Hiver wrote:
However there are some disadvantages, the main being you cant set CLI of
the outgoing call as it will always be tied to the SIM of the mobile
terminal.
That's true. You can however choose to mask the caller ID.
Yup, for
Are GSM gateways allowed in Canada?
And can we resell it?
Robert
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Steve Kennedy
Sent: Friday, January 06, 2006 9:17 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] GSM Gateway /
Hi,
Yes InternalExtension is the context and 2093 the extension.
Just to explain something odd thats happening (and Im very stumped
with this)
.I think my contexts are definately the reason that I
cant interrupt the menu for incoming pstn calls to choose a submenu:
My users register with my
On Fri, Jan 06, 2006 at 02:17:47PM +, Bob Goddard said:
On Friday 06 Jan 2006 08:11, Richard Scobie wrote:
Supermicro do not do Opteron (or Athlon64) systems.
Supermicro DO do Opteron.
Model numbers please? Searching through SuperMicro's web site shows ZERO
AMD based models. ONLY Intel.
Is there a protocol I'm supposed to use here? It seems that people are asking
100 questions a day and SOMEONE is helping them, and I've posted this three
times and not even an I Don't Know.
My third repost:
Ok, I've been trying to figure out why my [EMAIL PROTECTED] won't answer the
lines
On Fri, 2006-01-06 at 08:47 -0700, [EMAIL PROTECTED] wrote:
Is there a protocol I'm supposed to use here? It seems that people are
asking 100 questions a day and SOMEONE is helping them, and I've posted this
three times and not even an I Don't Know.
My third repost:
Ok, I've been
[EMAIL PROTECTED] a écrit :
Is there a protocol I'm supposed to use here? It seems that people are asking 100
questions a day and SOMEONE is helping them, and I've posted this three times and not
even an I Don't Know.
You know, if thoushands of people had to answer I don't know, it would
- Original Message -
Sent: Friday, January 06, 2006 10:44 AM
Subject: Re: [Asterisk-Users] Asterisk on Dell blade servers
On Fri, Jan 06, 2006 at 02:17:47PM +, Bob Goddard said:
On Friday 06 Jan 2006 08:11, Richard Scobie wrote:
Supermicro do not do Opteron (or Athlon64)
People don't usually respond with I don't know. They just don't respond
unless they can help. This helps reduce the clutter on the list. And for the
record I do not have an answer to this issue.
[EMAIL PROTECTED] wrote:
Is there a protocol I'm supposed to use here? It seems that people are
I had a similar problem , and then used GoTo instead of include
Iqbal
Aisling O'Driscoll wrote:
Hi,
Yes InternalExtension is the context and 2093 the extension.
Just to explain something odd that’s happening (and I’m very stumped
with this)….I think my contexts are definately the reason
[EMAIL PROTECTED] wrote:
Is there a protocol I'm supposed to use here? It seems that people are asking 100
questions a day and SOMEONE is helping them, and I've posted this three times and not
even an I Don't Know.
My third repost:
Ok, I've been trying to figure out why my [EMAIL PROTECTED]
On Friday 06 Jan 2006 15:44, Walt Reed wrote:
On Fri, Jan 06, 2006 at 02:17:47PM +, Bob Goddard said:
On Friday 06 Jan 2006 08:11, Richard Scobie wrote:
Supermicro do not do Opteron (or Athlon64) systems.
Supermicro DO do Opteron.
Model numbers please? Searching through
Hi All
I am trying to simplify a dialplan for a few thousand users.
Would what I have below work?
If someone dials exten 710001 would it go through answer and then to the macro
to try dialing the SIP phone thats registered on 710001 and then onto voicemail
if no answer or not signed on?
Hi, I haven't found anything about the message below on the mailing list, Does anyones knows why this notice is being appearing?
-- Executing Dial(Local/[EMAIL PROTECTED],2, IAX2/CallOut/12365533643|30|otT) in new stack -- Called CallOut/12365533643
-- Call accepted by 12.11.11.11 (format ulaw)
If you have to ask this question, please get professional help to
install this, otherwise you might end up with a few thousand users
picketing at your door.
On 1/6/06, scott [EMAIL PROTECTED] wrote:
Hi All
I am trying to simplify a dialplan for a few thousand users.
Would what I have below
Chris,
I've done several customized versions of iaxComm (including two for call
centers)
Contact me off-list if you're interested.
On Thu, 5 Jan 2006 05:37:59 -, Chris Bagnall [EMAIL PROTECTED] wrote:
I've been working my way through the softphones listed on voip-info over the
last few
Don't forget, patterns (for matching) must begin with an underscore (_)
I find it nicer to just use ${MACRO_EXTEN} rather than declaring $
{ARG1} for the sake of it.
-
exten = _71,1,Answer()
exten = _71,2,Macro(71macro)
exten = _71,3,Hangup()
[macro-71macro]
exten =
Change the RTP Packet Size: 0.010
to
RTP Packet Size: 0.020
Asterisk only work with 2 frames.
I can't send any fax with other values.
On 1/5/06, Joash Herbrink [EMAIL PROTECTED] wrote:
You could use a cisco ata 186.There aren't very cheap, but I have made them work on several of
I'm running * 1.2.1 on Slackware.
I have several queues configured to record incoming calls once
answered (without joining the in and out files). Yesterday, I showed
my agents how to transfer a call received from a queue to another agent.
What I realized today is that when listening to
Since not all of our operators are going through asterisk I can't switch
over to using asterisk. I agree that it is a much better system to
record the calls at the server, but thats just not an option. The call
recording software we use now is too integrated into our message taking
system not
show channels concise
it spits out colon : delimited fields with lots of information
That was one of the more frustrating changes from 1.0 to 1.2 but in
the end it provides much more data. Just beware if you use SIP or IAX
trunks that have colons in them, it will throw off the order of the
On Thu, 5 Jan 2006 17:57:47 +0100, Erwin de Raad wrote:
You should be able to run SIP through m0n0wall quite happily - we have a
number of client sites with SIP phones offsite which connect to the *
server
behind a m0n0wall box. You'll need to allow 5060 (UDP) for SIP, then an
appropriate
Jerry Geis wrote:
I am getting TRUNCATED call information
the IAX2/muncie_to_ge is truncated. How do I
get the need call information to transfer the call.
'show channels' is used for human-readable output on a console screen.
If you need the information in a complete form for some
Jerry Geis wrote:
/ I am getting TRUNCATED call information
// the IAX2/muncie_to_ge is truncated. How do I
// get the need call information to transfer the call.
/
'show channels' is used for human-readable output on a console screen.
If you need the information in a complete form for
If this or any other example is available, I would be most thankful to
have it.
I got the go ahead on this project to day so now I have to start seeing
how to do this.
Thanks,
Wiley
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Wiebe
Sent:
try this
?php
$socket = fsockopen(localhost,5038, $errno, $errstr, $timeout);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: 1212\r\n);
fputs($socket, Secret: 1212\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: reload\r\n\r\n);
* fputs($socket, Action:
I need to match an incoming call based on a prefixed string, and this
solution was suggested to me some time back.
exten = _conf.,1,Answer
exten = _conf.,2,MeetMe(${EXTEN:4}|d)
exten = _conf.,3,Hangup
However incoming calls never match this pattern, and I cannot
find any evidence in the wiki or
The match doesn't work because n in conf will never match to the
letter n (it's a pattern for a digit).
try _co[n]f. instead.
On Fri, 2006-01-06 at 10:33 -0800, Dan Austin wrote:
I need to match an incoming call based on a prefixed string, and this
solution was suggested to me some time back.
A really neat thing about this, you could make it interactive, and also
post the response back from each user on if they accepted it or not. and
then call them back in 5 min again :) LOL
But someone could be seeing what the system is doing realtime...
./Ben
Hello All,
I am having
Just to make it easy, I will be reading the caller list from a another
server via a web page, parsing it and dialing.
After each pass, I just post back to the server web page and it updates
the other system.
Our tech just needs to review the log once daily.
W
-Original Message-
From:
Jerry Geis wrote:
Is there another wayin the manager API I'm not aware of to get this
information?
No, I was mistaken. Matt Florell's response about using 'show channels
concise' is probably the best way to go, since it produces output
designed for automated interpretation.
I'm sure there must be a setting I'm missing somewhere, so I thought I
might was well ask here.
Conversations are punctuated by sudden replacement of a given syllable
or so of conversation with a DTMF tone.
I would hope perhaps there's some kind of setting that has to do with
the way it
Dean Collins wrote:
Lol, so Dell must be doing the same thing.
Did you ever consider that Supermicro are an enterprise setup to make
money, and that possibly their financial interests are served by
sticking with Intel?
Absolutely. However, it looks as though their lack of AMD product is
Hi I have two asterisk servers. I just want to connect two asterisk server using IAX2.But the Asterisk Servers are not able to register each other. If some body have done thisthen Please send me the configuration they have done in
iax.conf and extensions.conf.I simply want to connect and call
I have a question on this. It isn't readily obvious to me, upon issueing a 'sip
show channels' command which call legs are related to which call.
For example:
*CLI sip show channels
Peer User/ANRCall ID Seq (Tx/Rx) Form Hold Last
Message
192.168.10.121 a00090601
Here, this may be of use:
http://mundy.org/blog/index.php?p=95
On 1/6/06, Wiley Siler [EMAIL PROTECTED] wrote:
If this or any other example is available, I would be most thankful tohave it.I got the go ahead on this project to day so now I have to start seeing
how to do
On 1/6/2006, Michael Sampson [EMAIL PROTECTED] wrote:
Since not all of our operators are going through asterisk I can't switch
over to using asterisk. I agree that it is a much better system to
record the calls at the server, but thats just not an option. The call
recording software we use now
We have an Asterisk server with a single Digium E1. Everzthign works as it
should except for one minor issue.
When we place a call to a number that is busy, Asterisk does not seem to
properly send the busy signal back to the SIP phones. There is no indication
on the phone of anything at all, just
On Demand-monitoring? If your referring to monitoring specific agents calls,
I'm still trying to work out how to do that. You can either monitor all calls
for a queue, or all calls for all agents, but not all calls for a specific
agent. I tried to use the Monitor() command on it's own to start
pwlib ver 1.5.2
/usr/bin/ld:
./obj_linux_x86_d/asn_grammar.o(.gnu.linkonce.r._ZTV5PListI7PStringE[vtable
for PListPString]+0x1c): unresolvable relocation against symbol
`PAbstractList::Compare(PObject const) const'
/usr/bin/ld: final link failed: Nonrepresentable section on output
collect2:
Looks like you got a configuration issue, you should test for the
${DIALSTATUS} variable and set the signalling to the phones based on
that.
You can do:
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Goto,s-${DIALSTATUS},1)
exten = s-CANCEL,1,Playtones(congestion)
exten = s-CANCEL,2,Congestion
We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one
phone setup as the receptionist phone, using hints to show busy office
lines. This all works as expected.
This is a new installation, and people are just starting to setup their
phones. For those of you not familiar with
Hi,
Does anyone know if it is possible to setup an SJPhone with an external
ringer of some sort. One of the operators may not always be at her desk
and when she is not wearing a headset she cannot hear the phone ring -
is there some way to fix this?
Thanks
On 1/6/06, Chandan Mishra [EMAIL PROTECTED] wrote:
Hi
I have two asterisk servers. I just want to connect two asterisk server
using IAX2.
But the Asterisk Servers are not able to register each other. If some body
have done this
then Please send me the configuration they have done in
It may be better to change it to NOTICE instead of ERROR. You would want
to know it you are trying to subscribe to an unknown 'hint', misconfigs
or lack of should cause NOTICE instead of ERROR.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
We have done some work on this since my last post.We added some code
to update new fields in the realtime SIP database. Status, Qualify, and
Host Server. We then place the call directly to the phone the SIP
full contact (i.e. dial(sip/[EMAIL PROTECTED]:5060) Via a AGI
script. Our AGI
On 1/6/06, Chandan Mishra [EMAIL PROTECTED] wrote:
Hi
I have two asterisk servers. I just want to connect two asterisk server
using IAX2.
But the Asterisk Servers are not able to register each other. If some
body
have done this
then Please send me the configuration they have done in
On Fri, January 6, 2006 20:20, Chandan Mishra said:
Hi
I have two asterisk servers. I just want to connect two asterisk server
using IAX2.
But the Asterisk Servers are not able to register each other. If some
body
have done this
then Please send me the configuration they have done in
Hi,
What do I have to do to get local\number to work in a context?
It works from my [from-internal]... however from subcontexts it does not work:
Jan 6 15:55:32 VERBOSE[20237] logger.c: -- AGI Script Executing
Application: (Dial) Options: (Local/570323)
Jan 6 15:55:32 NOTICE[20237]
Here is what I used to do it:
http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers
Worked for me J
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
Sent: Friday, January 06, 2006
2:20 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Not Able
Is it a female talking on the other end from which hears the DTMF? If
so, that's life. It happens on the PSTN network too from time to
time.
On 1/6/06, Brian Capouch [EMAIL PROTECTED] wrote:
I'm sure there must be a setting I'm missing somewhere, so I thought I
might was well ask here.
On Fri, 2006-01-06 at 10:27 +, scott wrote:
[macro-71macro]
exten = s,1,Dial(SIP/${ARG1},30,tr)
exten = s,2,VoiceMail(${ARG1})
exten = s,3,PlayBack(vm-goodbye)
You appear to be missing some potential error codes. Here is what I use
which is just an extended version of what you have.
Can someone explain how to use groups? I can't seem to wrap myself around
this, though I know it is something simple.
I have 3 zap lines, and when placing an outgoing call, would like to 1) use
a zap line if and only if 1 or fewer zap lines are being used at the time,
and 2) if more than 1 zap
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