Sure if you can get up stream carriers to cooperate. Just follow the CDRs.
But short of a subpoena... or enlightened self interest, like the calls take
down a tandem.. (not likely).
We could loop the calls back to get AT&T's attention, but they would just
complain about the loop, not trace them
Hi,
Are you sure that Verizon amswers the call? They should play that
message as 'early media' without answering, after which they cam clear
the call with an appropriate cause code.
That would work for you and still give callers the audible ,essage they want.
Steve
On 3/20/09, drew einhorn wr
But from what I understand of SS7 it can be traced back with the
cooperation of the carriers. I know we have traced a call setup (AMPS)
across 3 carriers before.
Cary Fitch wrote:
> And getting a connection
> to a voice mailbox requires a 3 -way handshake so we can record the ip
> number the cal
And getting a connection
to a voice mailbox requires a 3 -way handshake so we can record the ip
number the call originated from. So it should be possible to generate
a dns black list similar to the ones used for email spam.
[Cary Fitch]
The problem is these are coming in from the PSTN and the S
On Thu, 19 Mar 2009 13:08:55 -0500, Cary Fitch wrote:
>Does anyone know of a phone product that:
>
>1. Would plug into a DHCP IP port and get an address. (i.e. Cable modem)
>
>2. Has a second Ethernet port and would bridge that address (perhaps pseudo
>DHCP so that following computer would be una
On Thu, Mar 19, 2009 at 10:27 PM, Jon Pounder wrote:
> Cary Fitch wrote:
>
> two weeks ago when I said don't ever permit them to have phone service
> again I was labeled a radical.
>
> at&t and the other telcos are just dropping the ball here as I said
> before - with ip address spoofing we all ha
Given the technology that lets folks who have equipment that listens to
the radio and automatically identifies the recordings being played to
generate Top 40 lists, allocate royalty payments to copyright owners,
etc.
It seems that it should be possible to automatically scan voicemail
recordings an
I'm having a problem with Verizon Wireless.
I would be extremely surprised if I was the only one having this problem.
It seems to me that Verizon Wireless might be able to use one of the
Special Information Tones to allow us to solve the problem.
But I really do not whether my suggestion is comp
Cary Fitch wrote:
two weeks ago when I said don't ever permit them to have phone service
again I was labeled a radical.
at&t and the other telcos are just dropping the ball here as I said
before - with ip address spoofing we all have rules to prevent packets
from entering our network which sho
Three or four area codes, all spoofed ANI.
We absorb their war dialing for about 2,000,000 unassigned cell numbers with
two Asterisk server which do nothing else.
Since they are war dialing cell phone numbers, they obviously don't care
about any rules. Trying to get any info from the people who
My mail client is smarter than I am.
CF
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757 area code right? They have been hitting my cell twice a day. I
always press one and go through the process telling them I have a 1959
Edsel dump truck that needs alot of work and how perfect this is going
to work for me... Either that or something else to waste their time.
Their ANI info i
oops, ignore my last post, this looks more appropriate:
http://www.voip-info.org/wiki/view/Asterisk+func+if
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021-295-1923www.knossos.net.nz
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On Thursday 19 March 2009 19:20:42 Cary Fitch wrote:
> The only conditional command I k
On Thu, 19 Mar 2009 19:20:42 -0500
"Cary Fitch" wrote:
> The only conditional command I know of in Asterisk is "GotoIF".
There is also GosubIf and ExecIf..
full list here:
http://www.voip-info.org/wiki-Asterisk+-+documentation+of+application+commands#Alphabeticallist
> Is there a simple "IF" t
On Thu, Mar 19, 2009 at 5:39 PM, Phil Reynolds
wrote:
> Quoting "drew einhorn" :
>
>> This sort of message is usually preceded by some magic tones that
>> allow direct marketing application to immediately drop a call to a
>> dead phone number.
>>
>> What is the proper terminology for the tones?
>
The only conditional command I know of in Asterisk is "GotoIF".
Is there a simple "IF" that doesn't have to goto anywhere?
I simply want to set a variable when a condition is met, for a specific set
of numbers, like:
exten _713NXX,IF $A = $B $C= "Houston Call"
exten _512NXX,IF $A = $B $
On Mar 19, 2009, at 4:34 PM, drew einhorn wrote:
> This sort of message is usually preceded by some magic tones that
> allow direct marketing application to immediately drop a call to a
> dead phone number.
>
> What is the proper terminology for the tones?
>
> Where can I find information about h
I don't think the telemarketers care about them. Right now we get thousands
of "Car warranty" phone calls everyday, now for months, and given that they
are illegally war dialing cell phone numbers, I don't think they listen for
the Special Information Tones.
Cary Fitch
-Original Message-
Quoting "drew einhorn" :
> This sort of message is usually preceded by some magic tones that
> allow direct marketing application to immediately drop a call to a
> dead phone number.
>
> What is the proper terminology for the tones?
Special Information Tone.
> Where can I find information about
This sort of message is usually preceded by some magic tones that
allow direct marketing application to immediately drop a call to a
dead phone number.
What is the proper terminology for the tones?
Where can I find information about how this is implemented?
--
Drew Einhorn
On Thu, Mar 19, 2009 at 6:20 PM, Christian Victor
wrote:
>> > grandstream gxp-2000 works fine for that.
>> > depending on firmware rev its two ports are either a switch or router.
>>
>> Grandstream removed this functionality in recent softwware upgrades - I
>> guess they needed the code space for
>
> > grandstream gxp-2000 works fine for that.
> > depending on firmware rev its two ports are either a switch or router.
>
> Grandstream removed this functionality in recent softwware upgrades - I
> guess they needed the code space for other things.
Why would you want a router in the phone and
You can also do a set variable in the call file. I don't really know how to
do that, but you can probably find the command and syntax on voip-info.org.
The reason it works on certain numbers has to do with switch timing. If *
can complete the call within a certain time frame, all is well. If no
Robert Broyles wrote:
> Is there a way to override the queue wrapup time on the fly?
>
> I would like to allow a longer wrapup time for my agents, but if they
> are already done with closing up the call ticket, I would like them to
> be able to dial an extension or something to override the wrap
I dont want to change it within my extensions.conf, because I have many
dids, and change them on the fly according to the call i am making. I have
a web interface where I fill a form that gets the number I am calling, the
caller id and context to go etc...
I dont want to keep editing extensions.co
GLOBAL_OUTBOUNDCID = XX in extensions.conf [globals] should do the trick
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Thursday, March 19, 2009 3:45 PM
To: Asterisk Users Mailing List - No
Is there a way to override the queue wrapup time on the fly?
I would like to allow a longer wrapup time for my agents, but if they
are already done with closing up the call ticket, I would like them to
be able to dial an extension or something to override the wrapup.
Is there a way to do that?
I did mean multiple chips, not multiple cores.
Thanks
Mike
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Spiro Harvey
> Sent: Thursday, March 19, 2009 16:36
> To: asterisk-users@lists.digium.com
> Su
Pascal Bruno wrote:
> Also very strange, when in my call file I change the callerid line to
> SIP/whatever like Danny said, the call go through, but I dont want
> that, because when I do so, it is displaying the main number on my T1
> account as caller id and I dont want that, I want to display
> >> I'm shooting from the hip here, but I don't think dual CPU gives
> >> you
> > redundancy. If one chip fries I am pretty sure the machine will
> > crash.
> >
> > This was sort of a question disguised as a statement. Can a CPUs
> > function when it's neighbour is fried?
Dualcore means two co
This is one approach. I'm sure there are better answers available. This
just seemed to be a simple one.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
Aloi
Sent: Thursday, March 19, 2009 3:19 PM
To: Asterisk U
If you were using 1.6 then you could do it in one queue with the new queue
rules, at least as I read the docs.
--
Jim Dickenson
mailto:dicken...@cfmc.com
CfMC
http://www.cfmc.com/
From: Christopher Aloi
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Date: Thu, 19 Mar 2009
Also very strange, when in my call file I change the callerid line to
SIP/whatever like Danny said, the call go through, but I dont want that,
because when I do so, it is displaying the main number on my T1 account as
caller id and I dont want that, I want to display one of my other DID as
callerid
Here is what I get from the console with the call file:
-- Attempting call on DAHDI/g1/1201XXX for s...@fortest:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- PROGRESS with cause code 127 received
[Mar 19 16:12:47] DEBUG[2892]: pbx_spool.c:413 scan_service: Delaying ret
Ahh - so use three queues and not one queue with three penalties?
On Thu, Mar 19, 2009 at 4:04 PM, Danny Nicholas wrote:
> Wouldn’t this work?
>
> Exten => s,1,Queue(level1,20)
>
> Exten => s,n,Queue(level2,20)
>
> Exten => s,n,Queue(level3,20)
>
> Exten => s,n,voicemail ; nobody answered
>
>
On Thu, 19 Mar 2009, Mike wrote:
> Hi,
>
> I`m looking for reliable and redundant hardware for Asterisk. I`ve been
> leaning towards buying one of these (HP 360 G5 with everything as redundant
> as possible), which I know will be good enough for a few months before
> needing to upgrade:
>
> http:
Wouldn't this work?
Exten => s,1,Queue(level1,20)
Exten => s,n,Queue(level2,20)
Exten => s,n,Queue(level3,20)
Exten => s,n,voicemail ; nobody answered
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
Aloi
Sent
On Thu, 19 Mar 2009, Mike wrote:
>> You can reliably run asterisk on just about any x86 hardware. You don't
>> mention what kind of stresses you are going to put on it, so your sizing
>> questions are impossible to answer. How many extensions? How many
>> simultaneous calls? Will you be tran
Mike wrote:
You can reliably run asterisk on just about any x86 hardware. You don't
mention what kind of stresses you are going to put on it, so your sizing
questions are impossible to answer. How many extensions? How many
simultaneous calls? Will you be transcoding? Routing to/from the PSTN
Hey All -
I've got an interesting problem, here is what I'm trying to accomplish:
Six agents, two queues, three skill levels
Queue A (queue B is the same)
- Level 1
-- Agent 1
-- Agent 2
- Level 2
-- Agent 3
-- Agent 4
- Level 3
-- Agent 5
-- Agent 6
I'd like a call to come in to Queue A
Don't know enough to properly term the problem I'm seeing... sorry if subject
appears vague. And I have other questions too, but "Newbie Help Wanted" isn't
exactly more specific... ;-)
My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an
X100p clone. Regular phone line pr
> You can reliably run asterisk on just about any x86 hardware. You don't
> mention what kind of stresses you are going to put on it, so your sizing
> questions are impossible to answer. How many extensions? How many
> simultaneous calls? Will you be transcoding? Routing to/from the PSTN?
> Wh
i am very far away to be an expert
in my experience i prefer to use a cluster of normal computers instead of an
expensive one.
if one go down you can trhow it and buy a new one any where very fast.
using opensip and *Heartbeat* you you can have an failsafe system.
dive in the mailing list archive i
On Thu, 19 Mar 2009, Mike wrote:
> Hi,
>
>
>
> I`m looking for reliable and redundant hardware for Asterisk. I`ve been
> leaning towards buying one of these (HP 360 G5 with everything as redundant
> as possible), which I know will be good enough for a few months before
> needing to upgrade:
>
>
Anyone hear anything? Very down for me right now.
Fred Posner
f...@teamforrest.com
Main: +1 (212) 937-7844
Direct: +1 (503) 914-0999
www.teamforrest.com
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asteri
Ricardo Melendez wrote:
> Hi to All, I dont know much about PCI express slots in newer Servers, my
> doubt is if the Digium and Sangoma PCI express cards, are compatible
> with the x8 PCI express slots that come in the HP Proliant ML150 G5 server.
Yes. All PCI Express x1 cards will work in x4, x8
Hi,
I`m looking for reliable and redundant hardware for Asterisk. I`ve been
leaning towards buying one of these (HP 360 G5 with everything as redundant
as possible), which I know will be good enough for a few months before
needing to upgrade:
http://h10010.www1.hp.com/wwpc/us/en/en/WF05a/1535
On Thu, 19 Mar 2009, Jon Pounder wrote:
> Cary Fitch wrote:
>
> grandstream gxp-2000 works fine for that.
> depending on firmware rev its two ports are either a switch or router.
Grandstream removed this functionality in recent softwware upgrades - I
guess they needed the code space for other th
Cary Fitch wrote:
> Great, how do you relate firmware rev to this feature, and I wonder if they
> do it with a Budge Tone 200?
>
I think the newer revs its just a switch not a router, never really paid
much attention since we don't use them for that.
> Cary
>
>
> -Original Message-
> Fr
Great, how do you relate firmware rev to this feature, and I wonder if they
do it with a Budge Tone 200?
Cary
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jon Pounder
Sent: Thursday, March 19, 2009 1:16 PM
Cary Fitch wrote:
grandstream gxp-2000 works fine for that.
depending on firmware rev its two ports are either a switch or router.
seemed to work fine once configured just plug in wherever and it
registers and works.
only issues have been bad nat router which are not friendly to the udp
streams
Does anyone know of a phone product that:
1. Would plug into a DHCP IP port and get an address. (i.e. Cable modem)
2. Has a second Ethernet port and would bridge that address (perhaps pseudo
DHCP so that following computer would be unaware of subterfuge.)
3. Would be a SIP phone doing the "usua
Hi to All, I dont know much about PCI express slots in newer Servers, my
doubt is if the Digium and Sangoma PCI express cards, are compatible with
the x8 PCI express slots that come in the HP Proliant ML150 G5 server.
Thanks in advance.
Ricardo
_
Matt Riddell schrieb:
> On 17/03/2009 9:10 a.m., Doug wrote:
>> >
>> >So to make extension 201 in pickup group 1 just do:
>> >
>> >asterisk -rx 'database put pickupgroup 201 1'
>>
>> So this is a command line argument. Can this
>> be automated? Whenever we do a reload, can
>> this be st
Dear all, I want to know if anybody has implented an Asterisk server
(1.4 or 1.6) with SRTP and SIP with TLS, in order to encrypt both
signaling and voice packets.
Is it possible ??
And in the affirmative case, does encryption increase the delay and so
the voice quality becomes wrong ???
Thanks
2009/3/20 D Tucny
> 2009/3/19 Oguzhan Kayhan
>
>> Hi, i know i am asking a lot of questions lately in this forum..sorry
>> about that first of all. :)
>>
>>
>>
>> Ok, here is the deal..
>> I am trying to make a hybrid system with an ericsson MD110 and asterisk.
>> Internally we have 4 digit phon
2009/3/19 Oguzhan Kayhan
> Hi, i know i am asking a lot of questions lately in this forum..sorry
> about that first of all. :)
>
>
>
> Ok, here is the deal..
> I am trying to make a hybrid system with an ericsson MD110 and asterisk.
> Internally we have 4 digit phone extensions on ericsson.. and
Locked channel does not react to 'soft hangup' command.
That's why it is called - LOCKED.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.co
On Thu, Mar 19, 2009 at 06:37:26PM +0200, Mindaugas Kezys wrote:
> Any guidelines how to solve locked channels problems?
>
> E.g. to find out which part of the code has problems and causes locks.
Build Asterisk with locking debugging?
(sure, this hurts performance, but one day with decreased per
Monitoring my own line when making a call, it goes red once I press send.
But if I use a button for an unrelated line (not in the same context) then
the light doesn't show busy.
So if calling from xxx-yyy-zzz1, the blf for xxx-yyy-zzz1 shows red.
But if calling from xxx-yyy=zz23, the blf for xxx-
Any guidelines how to solve locked channels problems?
E.g. to find out which part of the code has problems and causes locks.
Upgrade to newer versions are not an option.
Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions
-Original Message-
From: asteri
On Thu, 19 Mar 2009, Miguel Molina wrote:
> Mindaugas Kezys escribi?:
> >
> > As Asterisk has inner problems and channels very often locks we have
> > such script to restart Asterisk each midnight.
> >
>
>
Why restart Asterisk, free up the channel...
>From cron, you can clear up any calls ove
> Probably same thing I did.
>
> In the GXP2000 BLF setup, set the "account" field to the Line that relates
> to the extension you are trying to monitor. You can't monitor "just any
> old
> hint" it has to be related to a number that is in the same context as a
> number on the phone and the BLF en
Hi,
Currently, in both 1.4.23.1 and 1.6.0.5, "sip show peers" displays lines of
data.
In each data line, the first field is "Name/username".
Let's say the value of this field is "Foo/0123456789".
If I type "sip show peer Foo", I've got a long value list.
Would it be easy to add "sip show usernam
use ami
http://www.voip-info.org/wiki/view/Asterisk+manager+Example%3A+Java
or
Ajam
http://www.voip-info.org/wiki/view/Aynchronous+Javascript+Asterisk+Manager+(AJAM)
2009/3/19
> I have to develop a VoIP application. I need to know how to use Java APIs
> to communicate to my client applicati
Try this call file - replace XXX with your number and YYY with a valid SIP
exten on your system
Channel: DAHDI/g1/1XX
Callerid: SIP/YYY
MaxRetries: 1
RetryTime: 5
WaitTime: 60
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.co
Mindaugas Kezys escribió:
>
> As Asterisk has inner problems and channels very often locks we have
> such script to restart Asterisk each midnight.
>
That is the things we must help to solve for not having to do to
something like this on asterisk servers. Fortunately I use 1.4.22
version which
I have this "weirdness" as well - depending on the phone (all gxp2000's)
I either get steady green regardless of sip registered or not OR no
green ever, and red always works as expected. Never yet seen green
follow the sip registration of that device like I would expect.
not sure if my message
On 19 Mar 2009, at 15:08, ameu...@yahoo.fr wrote:
> I have to develop a VoIP application. I need to know how to use Java
> APIs to communicate to my client application with asterisk.
Ok.
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Oguzhan Kayhan wrote:
actually further to my last email another related question :
when programming the side buttons what are people using - ring groups or
individual extensions, other ?
I have setup sip devices with one series of extension numbers (ie every
line button on the phone, softclien
- ameu...@yahoo.fr wrote:
>
> I have to develop a VoIP application. I need to know how to use Java APIs to
> communicate to my client application with asterisk.
I tried looking for some answers based upon your subject but nothing came up.
This may be what you're looking for: http://lmgtfy
Oguzhan Kayhan wrote:
> Hi,
> Previously i was using asterisk 1.4 with freepbx installation.
> To try the 1.6 version i installd anc configured everything..
> Just one thing didnt work so far..
> I am using grandstream 2000 and it has a line busy indicator for chef
> secretary phones.
> But now, th
Probably same thing I did.
In the GXP2000 BLF setup, set the "account" field to the Line that relates
to the extension you are trying to monitor. You can't monitor "just any old
hint" it has to be related to a number that is in the same context as a
number on the phone and the BLF entry pointed t
Hi,
Previously i was using asterisk 1.4 with freepbx installation.
To try the 1.6 version i installd anc configured everything..
Just one thing didnt work so far..
I am using grandstream 2000 and it has a line busy indicator for chef
secretary phones.
But now, this feature does not work.
I can see
>>> On Mon, Mar 16, 2009 at 6:42 PM, Pascal Bruno wrote:
I have a weird problem with call using my T1 card. I can make calls
fine using my analog and IP phones, but when I try to initiate a call
using a .call file, I get the following error
-- Attempting call on DAHDI
I have to develop a VoIP application. I need to know how to use Java APIs to
communicate to my client application with asterisk.
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As Asterisk has inner problems and channels very often locks we have such
script to restart Asterisk each midnight.
We (our clients) mostly use v1.4.18.1. We can't upgrade to newer versions
because there are too much changes which would brake our system
(realtime/sip/iax2/cdr/etc/etc).
Scri
Just to add
P[ 1] Transmitting 128 samples 2 misdn
P[ 1] writing 128 bytes 2 asterisk
P[ 1] Sending :160 bytes 2 MISDN
P[ 0] misdn_jb_fill: written:160 | Buffer status:256 p:861fee0
P[ 0] misdn_jb_empty: read:128 | Buffer status:128 p:861fee0
P[ 1] Transmitting 128 samples 2 misdn
P[ 1] writin
Here is what my extensions.conf file has:
exten => _NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
exten => _NXXNXX,n,Hangup()
exten => _1NXXNXX,1,Dial(DAHDI/g1/${EXTEN})
exten => _1NXXNXX,n,Hangup()
Using the phone, I can dial any numbers succesfully.
And here is my call file:
Channel: DAHD
- "Oguzhan Kayhan" wrote:
> Hi, as far as i know 's' is wildcard for "all calls" because as i see on
> asteriskgui it is written as 's' (CatchAll) which means redirect all
> calls to that extension.
That is not correct. The 's' extension only matches analog calls (because they
have no dia
On Mar 19, 2009, at 9:05 AM, Ken D'Ambrosio wrote:
> Hey, all. I'm all over MWI, but I gotta say that I think the
> Polycoms go
> a bit over the top. The blinking LED is enough for me; how do I
> disable
> the stuttered dialtone and the audible warble? I've looked through
> the
> config
On my EuroISDN PRI link, a pri debug on the same type of call yields the
messages below. What could I try to do to see why the QSIG pri link doesn't
work (times out)?
Thanks
< Call Ref: len= 2 (reference 1065/0x429) (Originator)
< Message type: SETUP (5)
< [04 03 80 90 a3]
< Bearer Capability
Hey, all. I'm all over MWI, but I gotta say that I think the Polycoms go
a bit over the top. The blinking LED is enough for me; how do I disable
the stuttered dialtone and the audible warble? I've looked through the
config files, but there are a HELL of a lot of options, and I haven't been
able
Please paste the call file content (with the number 'ed of course) and
the Dial section from extensions.conf.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pascal Bruno
Sent: Wednesday, March 18, 2009 6:24 PM
To: Aster
Hi, i know i am asking a lot of questions lately in this forum..sorry
about that first of all. :)
Ok, here is the deal..
I am trying to make a hybrid system with an ericsson MD110 and asterisk.
Internally we have 4 digit phone extensions on ericsson.. and so in asterisk.
So, what i want to do is
Paulo Santos > I did some tests on it, not many. Without going higher
than 2.0 load average I managed to do 10 calls per second, lasting 5
seconds each. During those 5 seconds, 2 sound files were played
(sln). MySQL CDR was enabled, so that's also 10 DB writes/second. I
don't know exactly what
Hello,
I work on voicemail.conf and I need that ${VM_DATE} is in french!
How can I do it?
Thank you
Cordialement,
BERGANZ François
P Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
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> 2009/3/19 Oguzhan Kayhan
>
>> Hi, i managed to connect to Ericsson MD110 with PRI at last.
>> And made a successful call thru asterisk to ericsson.
>>
>> But when i try to call from ericsson to asterisk i got an error on
>> asterisk side.
>> And i couldnt figure out why.
>>
>> Here's my extensio
Wolfgang Pichler wrote:
> Or can anyone here tell me where to get good (and not to expensive)
> 2.5mm plug connection binaural headsets ?
>
>
Ebay might be a source for these:
http://shop.ebay.com/items/?_nkw=2.5mm+to+3.5mm+adapter+headphone
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-- B
Hi,
I have just discovered (a year after it was implemented) a possibly
undocumented incompatability between IAX in Asterisk 1.4 and any
version of Asterisk pre-March 2008.
It seems an AST_CONTROL_SRCUPDATE frame type was added (in March '08),
but no mechanism to negotiate whether it can be sent
Hi to all the ML. I'm new here.
I start to use asterisk with realtime configuration, with pgsql
backend connected via odbc.
The connection between asterisk and pgsql works fine.
I create a table sip_conf with 2 user (for testing purpose), 1401 and 1501.
Those are the records:
asterisk=> SELECT nam
2009/3/19 Oguzhan Kayhan
> Hi, i managed to connect to Ericsson MD110 with PRI at last.
> And made a successful call thru asterisk to ericsson.
>
> But when i try to call from ericsson to asterisk i got an error on
> asterisk side.
> And i couldnt figure out why.
>
> Here's my extensions.conf abo
I have a PRI E1 link between Asterisk 1.4.24 and Alcatel-Lucent OmniPCX
Enterprise R9.0.
As EuroISDN it works fine.
However, I need to move to QSIG because of a firmware upgrade on the Alcatel
PBX which doesn't support EuroISDN (please don't ask why).
Besides, I've read somewhere that 2 B Cha
Hi, i managed to connect to Ericsson MD110 with PRI at last.
And made a successful call thru asterisk to ericsson.
But when i try to call from ericsson to asterisk i got an error on
asterisk side.
And i couldnt figure out why.
Here's my extensions.conf about incoming calls.
[DID_span_1]
include
Hi All,
I'm trying to configure a Digium T100P to talk to a legacy voicemail
system. I have the signaling specs verbatim from the original manufacturer
documentation as follows:
[T1 Signaling]
Service Type: T1,D4 format, AMI(Super Fram)
Signaling: Four wire, terminated, E&M (Robbed bit)
Start Pro
Laurent CARON wrote:
> I'm experiencing a quite strange behavior while trying to receive faxes
> through Asterisk (either directly through app_rxfax or with spandsp +
> hylafax).
Hi,
I should have mentionned (known?) that the telco is using G729 compression.
Obviously FAXes will never get thro
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