Yes,
transcoding is not going to work for that density.
asterisk doesn't do g723, and even if it would your system would not be
able to handle more than 150 simultaneous g711 to g729/g723 transcodings.
If you would go for plain g711, you could do 500, but i don't recommend
it, especially if
Rehan Ahmed ha scritto:
I dont see the ip in the Master.csv but you can view the IP when the
call comes in on the CLI Window.
I am guessing there must be a command or a way to record this ip in
your CDR using AGI, we are using agi to make our own CDR but i would
apreciate if some one
What about plain g729?
My main concern is the Hardware, anyone that can tell me if this
Supermicro 6014H-32 is stable and sutible for asterisk?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Zoa
Sent: den 8 december 2005 09:00
To: Asterisk Users Mailing
How would one go about to implement such a
cluster?
How do the different Asterisk boxes know of the
extensions on all the other boxes?
Is each client bound to it's box or can it connect
to any box in the cluster, ie if one fails can the
other take over and share the load of the failed
on between
Krystian Filiks wrote:
What about plain g729?
My main concern is the Hardware, anyone that can tell me if this
Supermicro 6014H-32 is stable and sutible for asterisk?
Supermicro Superservers are traditionally extremely stable and reliable.
--
Roman Volf
Keystreams Internet Solutions
[EMAIL
can u paste your sip.conf general section,,?
there have another possible cause... the both side use different codecm and asterisk can not translaste it ...
-- Jeffery
On 12/8/05, chawki hammoud [EMAIL PROTECTED] wrote:
Hi:i added these two lines to my general context ,butnothing happened the
I am having problems with echo, first let me
explain my setup:
I have a Gateway box, which basically is an
Asterisk with a PRI card. It's only job is to
interface with 2 incoming ISDN PRI connections.
Then there are two other asterisk boxes to which
my users are registered.
Dialing from a phone
Take a look at the function VoiceMailMain()
You will need to put it somewhere in your extensions.conf.
On 12/8/05, Tejas Shah [EMAIL PROTECTED] wrote:
hi all,
I have made two voice mail boxes for 2 users on asterisk server
(/var/spool/asterisk/voicemail/testmail/inside this 2 boxes for 2
Hello Patrick,
I have an Eicon Diva PRI-30M card and use the Eicon Linux drivers with
chan_capi_cm.
I am able to do ISDN to SIP calls with this.
Have you tried using the Eicon drivers instead, rather than zaptel and zib pri.
Instruction for doing this can be found here:
Hey,
I´m just wondering if its possible to login an ISDN phone connected to my
asterisk box via an S0 Trunk line.
My setup is:
Siemens Hipath 4000 S0 Asterisk --- SIP --- x SIP Phones
I would like to setup several queues on my asterisk and allow both SIP
Users as well as the external
Hi!
How many register lines does Asterisk support i sip.conf? I may need a setup where Asterisk should act as a many sipclient (over 10.000) (seen from the telco side) and then forward the calls to the real sipclients. Is that possible?
Is it possible to use SER to do this instead of
hello,
I always update
trough CVS from the cvs tree but i only see this revision 7230 in the asterisk
all the days but the changelog say there are already newer
versions.
Did i updated wrong
or is the revison wrong?
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I have setup a2billing to be used as more
of a billing software.
I have made a card for each of my
customers and the call through and get authenticated with callerID and then the
call is placed using dnid.
This is working great.
The problem that I just ralized is that
when I get a call
Hi all,
I have problem to get MYSQL cmd work with Asterisk Realtime (Asterisk
1.2.1 and Asterisk-addon 1.2.1). It happened like what being described
in this post
http://lists.digium.com/pipermail/asterisk-users/2005-May/107956.html.
The ${resultid}reference on the fetch line using asterisk
Hi, I got this message in the asterisk console while sending the dtmf from a phone.Dec 8 14:55:50 WARNING[29098]: codec_ilbc.c:163 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? Please help me to solve this.Thanks Jibu
Yahoo! India
Hello,
I'm using Swissvoice IP10S phones. When I dial from number 11 to number
22, on the phone with number 22 there is displayed a name set in
sip.conf for user 11 and a number 22 (not 11). When I try to call back,
it calls to the 22 (self). How to correct this?
In sip.conf I have:
Russ Price wrote:
So, are there any IP faxes?
Sort of.
But I'm talking about hardware IP faxes.
--
Best regards,
Bartosz Piec
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Dear Gulzar,
Thank you for your reply, I am using same configs. I have tried both 0
1 in timing but no luck. I will try again with 'timing' parameter = 1 in
zapata.conf
best Regards,
--
Atif Rasheed
Gulzar Hussain wrote:
I am using EWSD's PRIs and I am not having this
problem my configs
Hi all,
I'm trying to change the CDR userfield in a macro which is executed upon
call pickup (option M in Dial command). The goal is to log the answer
time (in the default CDR it is not correct as the call is picked up to
play music on hold to the caller before Dialing the called extension). I
Title: Message
Hi
all,
Did
any one succeeded to configure MOH (Asterisk 1.2.0) to play audio from the
streaming source?
Sample
musiconhold.conf has entry like this:
[stream]
mode=custom
application=/usr/sbin/streamplayer192.168.100.132
8088 format=ulaw
I used
JMStudio (java JMF
hi,
is it possible to use GROUP_COUNT function in AGIs.
i could not make it work. :-(
thanks
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Hey everybody,
Is here anyway to change the name asterisk on the caller id inbound to the
client/sip app?
Thanks,
Oliver
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Oliver Vermeulen wrote:
Is here anyway to change the name asterisk on the caller id inbound to the
client/sip app?
In sip.conf type in the section of desired user:
callerid=Caller Name 11
Where 11 is your caller number.
--
Best regards,
Bartosz Piec
Hi,
Is there any existing tools for IAX stress testing?
I need to know with how mach IAX - H.323 and IAX - IAX simultaneous
calls can my server works. Now it is few clients, but in the future it
will be mach more. Can I emulate many dummy IAX clients on single computer?
--
Thanks,
Eugene
Hi all!
I have an [EMAIL PROTECTED] 2.1 setup here which is working 99% the way it
should. The only thing that does not work is Meetme/Conferences...
In the log-file I see:
Dec 8 11:51:28 WARNING[3288] pbx.c: No application 'MeetMe' for extension
(from-internal, 8125, 6)
This is when I
Hi all,
I've noticed that Polycom and Snom each offer
attendant console expansions. As far as I understand, the point in using all
thebuttons they provide is to program them to register as extensions in
order to be able to monitor the status of each extension at any given time and,
also,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
on voip-info there's two howtos to connect asterisk with CM and/or CME
Cisco, but always with sip trunk.
What about h323 instead of sip? there's someone that has tested
something like that? MWI will work too?
Your feedbacks will be
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi folks,
my topology is like that:
ISP --[sip]-- Asterisk --[sip]-- CME Cisco -- ip phones
ISP services are g711 and g729 enabled.
My Asterisk is registered on ISP with two sip UA.
Then I've forwarded calls from ISP to ip phones registered on CME
yusuf wrote:
Hi all,
can anyone tell me when (or how) * starts generating ring tone when a
call is made. The reason I ask this is I have an E1 coming from a PBX
into my * box (CVS 19/07/2005). I have some intermittent problems.
1. Sometimes no ringtone is generated, so I dial a number, and
Thanks, I did that with upper and lower case, using 1.3. I have another
issue then because it is still not loading, it appears the phone is
loading but when I check the configs aren't there.
Lists wrote:
According to the wiki page
Bartosz Piec wrote:
Russ Price wrote:
So, are there any IP faxes?
Sort of.
But I'm talking about hardware IP faxes.
There are a number of IP capable FAX machines. It seems most don't obey
the standards (T.37 and T.38), though.
Regards,
Steve
Hi
In my
extensions.conf shown below when the external
number 123 is dialed it goes to phone ext1. I can forward to another phone
using exampleline below but I would only like to forward after 5pm and
before 9am. How can this be done?
Thanks for your help.
exten =
James Steven wrote:
Hi
In my extensions.conf shown below when the external number 123 is
dialed it goes to phone ext1. I can forward to another phone using
example line below but I would only like to forward after 5pm and
before 9am. How can this be done?
Hi Evert,Do you have the zaptel/ztdummy modules installed ?KunalOn 12/8/05, Evert Meulie [EMAIL PROTECTED]
wrote:Hi all!I have an [EMAIL PROTECTED] 2.1 setup here which is working 99% the way it should. The only thing that does not work is Meetme/Conferences...
In the log-file I see:Dec 8
Read before you reply... ;-)
To be 100% clear on zaptel/ztdummy, here's the output of my lsmod:
[EMAIL PROTECTED] ~]# lsmod
Module Size Used by
md5 8001 1
ipv6 240097 16
autofs422085 0
i2c_dev14273 0
Hi Dionisis,
please search the Wiki/ Google for hint in connection with asterisk
and you will find.
Philipp
I've noticed that Polycom and Snom each offer attendant console
expansions. As far as I understand, the point in using all thebuttons
they provide is to program them to register as
I'm currently starting development of an add-on to a program designed to
be used in a call-centre type environment that will interface very
closely with Asterisk - quite possibly to the point that the add-on
itself will be a softphone as well.
In order to test this application properly, I
Is there a way to optionally keep asterisk in the media path in order
to record calls using the Monitor command? For example, if I have a
SIP peer that is defined with canreinvite=yes, this means that if
possible, Asterisk will not be in the media path. However, what
happens if the user
Use asterisk itself to build a box which generates the calls. Maybe what
some people misses (call simulators are quite a recurrent query on the
list) is that you can move a text file with the equivalent of a manager
API action Originate in the spool/asterisk/outgoing/ directory and the
call
Is there a way to optionally keep asterisk in the media path in order
to record calls using the Monitor command? For example, if I have a
SIP peer that is defined with canreinvite=yes, this means that if
possible, Asterisk will not be in the media path. However, what
happens if the user
If that's the case, is it possible to override the canreinvite
attribute of a SIP peer in extensions.conf before a call is made or
answered by that peer?
- Waldo
On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote:
Is there a way to optionally keep asterisk in the media path in order
to
On Thu, 2005-12-08 at 07:00 -0500, [EMAIL PROTECTED] wrote:
Thanks, I did that with upper and lower case, using 1.3. I have another
issue then because it is still not loading, it appears the phone is
loading but when I check the configs aren't there.
How are you checking, with the web
Hi all, just got an iaxy box for a customer and its great, but!
I really dont want to host and bill this customer myself and i cannot
find a voip to pstn breakout that will let him have a dynamic IP.
Gradwell require a static ip
Voiptalk wont support it
Any Ideas where else to try?
Thanks
Hi Bails,
We'll help. Drop me a mail off-list.
Simon
http://www.esms.comOn 12/8/05, bails [EMAIL PROTECTED] wrote:
Hi all, just got an iaxy box for a customer and its great, but!I really dont want to host and bill this customer myself and i cannot
find a voip to pstn breakout that will let him
I am having problems with echo, first let me
explain my setup:
I have a Gateway box, which basically is an
Asterisk with a PRI card. It's only job is to
interface with 2 incoming ISDN PRI connections.
Then there are two other asterisk boxes to which
my users are registered.
Dialing from
Hi All
I have a call center working on 8 FXO Channels,
everything working fine except one little problem, I
am using asterisk queues with
monitor-format = wav49
and
monitot-join = yes
asterisk is recording all conversations but the
problem is that the volume of Zap Channel is too low
hi all,
I have made two voice mail boxes for 2 users on asterisk
server (/var/spool/asterisk/voicemail/testmail/inside this 2 boxes for
2
users). i have made following settings in voicemail.conf :
[testmail]
vipul=,vipul patel, [EMAIL PROTECTED]
tejas=,tejas
Note:
I upgraded Zaptel to the 1.2 stable and changed digits.h line to #define
DEFAULT_DTMF_LENGTH 250 * 8.
I was told that there is still a problem.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - -
Alvaro Parres wrote:
Hi any one can recommend me a company in the USA that can sell me a Toll
Free Number
and send me the call via IP.
Thanks.
___
--Bandwidth and
Hi
all,
I'd like to set up a
box with asterisk and the following cards in it :
- one E1 card (from
digium)
- one Junghanns
OctoBRI
My question is
:
1) Is it possible
such a configuration ?
2) Because of the
Junghanns card, I will have to use the bristuff package, but I'd like to
Forward UDP Ports 1-2 to your asterisk box.
On 12/8/05, Jeffery Chen [EMAIL PROTECTED] wrote:
can u paste your sip.conf general section,,?
there have another possible cause... the both side use different codecm and asterisk can not translaste it ...
-- Jeffery
On 12/8/05, chawki hammoud
He said that he is using a crossover but for some reason I think the
crossover may be the problem. Try making a new one. Cross pin one with
four and two with five. Also try a straight through cable. Your
configs look fine on the asterisk side although I am not real cluefull
on the Meridian.
Hi,
If you were to lead someone (with a UI) through the process of
configuring a a Digium T1/E1 card with asterisk and a T1/E1 trunk from
a
provider, would the following questions cover most scenarios? i.e.
given
the following questions and assumptions, would the configurations
below
If the digium card is good then he has the proper cable config,
although his send may not be getting to the nortel. This is layer one
which must work before layer two, ie d channel.
What does the nortel say regarding the T1?
If this is good then your issue is with configuration not
i'm using 1.2.
get the right patch from http://bugs.digium.com/view.php?id=5281
patch fie is: Patch-5281-v2.txt
On 12/6/05, Alvaro Parres [EMAIL PROTECTED] wrote:
which version of Asterisk do you have ?, Becouse when i change the function
to your code, every time that one phone with call-limit
Hi,
is it possible to run an octoBri card together with a TE405P card in one system
with bristuff?
If yes, how should the zaptel.conf look like?
Thanks and regards,
BK
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You can't use a ethernet crossover cable, make sure you are using a T1 crossover cable. (you will definately need to use a T1 crossover cable).
I'm running a Nortel Option 11 and Asterisk connected in this manner.
On 12/8/05, Steve Totaro [EMAIL PROTECTED] wrote:
He said that he is using a
Cabling is always the first thing to check in these types of issues.
Sure, he may need a T1 crossover but maybe the cable was made
incorrectly or the connections are not crimped well. Just because
someone says they have a crossover cable does not mean that it is OK.
Again, check the cable
Greetings All,
I intend to buy a Polycom IP 500CS with part number 2201.11500.001 but
I'm not sure if it will work with Asterisk. Is this a SIP capable
phone? Moreover, what does CS stand for? Done some research at
http://www.voip-info.org/wiki/view/Polycom+Phones but the original info
on part
Well, then set canreinvite=no
If that's the case, is it possible to override the canreinvite
attribute of a SIP peer in extensions.conf before a call is made or
answered by that peer?
- Waldo
On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote:
Is there a way to optionally keep
Anyone know how to do it?
Sam
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Hello Rob,
Our OrderlyQ system is designed to pass (real) calls to call centre
agents and queues at a constant rate (or at least can easily be
configured to do this). I can think of several ways the system could be
'rigged' to produce the calls automatically too...
We've also built
Steve Totaro wrote:
Hi,
If you were to lead someone (with a UI) through the process of
configuring a a Digium T1/E1 card with asterisk and a T1/E1 trunk from
a
provider, would the following questions cover most scenarios? i.e.
given
the following questions and
Yes, that is why I said cross pins one with four and two with five (a T1
crossover cable configuration)
You can't use a ethernet crossover cable, make sure you are using a T1
crossover cable. (you will definately need to use a T1 crossover
cable).
I'm running a Nortel Option 11 and
I understand. But because the majority of calls are not to be
recorded, I don't have a need to keep Asterisk in the media path all
the time. That's why I'm wondering if you could dynamically keep it
in the media path or not.
- Waldo
On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote:
Well,
[EMAIL PROTECTED] wrote:
Thanks, I did that with upper and lower case, using 1.3. I have another
issue then because it is still not loading, it appears the phone is
loading but when I check the configs aren't there.
I looked at this last night. You need to have an aastra.cfg file in
your
The long waited Ultimate GSM Gateway is finally out. This time we have managed
to source a new patch of brand NEW GSM Gateway at prices that is only 50% of
what the market rate. And with the SMS Function and many more...
For purchase please email gsm AT cyper-telecom.net. We accept paypal and
Is there a way to have control go back to the dialplan after a call gets tovoicemail?
I'm looking to implement findme and campon, but I wantthe options to be hidden, so if someone calling got a voicemail they could key in *1 (or whatever) and it would go back to the dialplan so I can implement
Well up until I saw the 100-220V and 50HZ I was sold.. But if you dont
support 60HZ it will never work in North America. Well it could but it
would be a pain in the ass..
/b
On Thu, 2005-12-08 at 23:41 +0800, Sam Tam wrote:
The long waited Ultimate GSM Gateway is finally out. This time we
On 12/8/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I understand. But because the majority of calls are not to be
recorded, I don't have a need to keep Asterisk in the media path all
the time. That's why I'm wondering if you could dynamically keep it
in the media path or not.
Some options of
There may be a better way but off the top of my head this idea jumped
out. It assumes that you know prior to making the call that you need to
record it and that you have phones capable of multiple lines.
Setup a second line with a different entry in sip.conf with
canreinvite=no and use that
I have an extremely simple AGI script like this:
#!/bin/bash
ISODATE=`date -iso-8601=seconds`
echo SET VARIABLE ISODATE \$ISODATE\
The script's intent is to return the current date and time back to Asterisk
in an ISO format as a variable. It works fine. Calling the script from
Asterisk returns
Hi,
I am using ooh323.
I cannot setup a call towards a cisco gateway.
The cisco rejects the call right away with :
Cause value: Mandatory information
element is missing (96)
This
is in the q931 part.
Cisco explanation
Indicates
that the equipment that is sending this code
I have a big dilemma.
I have a client who is looking for a big installation.
I am looking at the digium product and have the following
Questions.
Difference between Asterisk and Asterisk Business Edition.
My Client has 300 personal split between two office and
wants to
What will work in the US and also have SMS with multiple ports?
Well up until I saw the 100-220V and 50HZ I was sold.. But if you
dont
support 60HZ it will never work in North America. Well it could but
it
would be a pain in the ass..
/b
On Thu, 2005-12-08 at 23:41 +0800, Sam Tam
Hello,
In the process of upgrading a couple of voicemail servers from CVS (end of
august 2005) to 1.2.1
This is a purely voicemail system using mysql configurations.
All my mailboxes are in the default context and it worked fine under the
CVS version. But with 1.2.1 the voicemailmain fails to
This and Time Bandit's comment makes sense. I didn't realize that
these options in the Dial string will force Asterisk to stay in the
media path even if canreinvite=yes.
I'll give it a try.
Thanks,
Waldo
On Dec 8, 2005, at 11:18 AM, Steve Totaro wrote:
There may be a better way but off
Voicemail in itself does not hangup, * will bring you back to the DP
(to exten a). So if a user exits VM (I think they can exit by pressing
# after recording) then you can drop them in a context that does what
you want, you can do the same at exten a.
On 12/8/05, Joe Pukepail [EMAIL PROTECTED]
I have a new * 1.2 server running on a dual-processor
machine, 1GB of RAM, Gentoo with Linux 2.6 and a Digium TDM400 (four fxo
boards) installed. Everything has been working great until we tried our
first Meetme conference call yesterday.
I have a total of 12 extensions. 9 of them are
If you have to ask this question, then the answer is, use Asterisk
Business Edition.
On 12/8/05, Goran Donev [EMAIL PROTECTED] wrote:
I have a big dilemma.
I have a client who is looking for a big installation.
I am looking at the digium product and have the following Questions.
We finally got the T1 connection working between the Nortel and the Asterisk
box, but only with Robbed bit signalling. For some reason, the D channels
would not come up when using PRI ISDN. No clue why, but I'm just happy to
have it up and running.
Anish Basu
Field Systems Engineer
Softel, Inc.
On Wed, 2005-12-07 at 21:45 -0500, Lists wrote:
According to the wiki page
http://www.voip-info.org/tiki-index.php?page=Aastra+480i+Configuration it
shows lowercase file name and then there is a comment at the bottom that it
needs to be capitalized.
I have tried it both ways with no luck.
Is there a document/wiki/web site that maps the various SIP.conf settings to
the structure of the actual IP packet?
If so please advise.
--
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You can use app_nv_faxdetect.
Hi:
I want to use the same phone number for the fax and voice conversations.
How do I redirect a call to the iaxmodem extension? Should my VOIP
provider support the slinear codec?
Thanks
Miguel
-Original Message-
From: Miguel Soto [mailto:[EMAIL
I have a big dilemma.
I have a client who is looking for a big installation.
I am looking at the digium product and have the following Questions.
Difference between Asterisk and Asterisk Business Edition.
My Client has 300 personal split between two office
Yeah, makes sense now that I think about it a little more. Guess you
will have to prefix your exten so that the dial string with the H is
used and dial that prefix when you know or think that you may have to
record a call.
This and Time Bandit's comment makes sense. I didn't realize that
List users,
Please provide me with tips on how to replicate a single file to a
separate machine as changes are made to it. I would prefer a method
that reacts to file modifications (ie. FAM/gamin) as opposed to timed
loops/polling (cron + rsync). I'd also like to avoid NFS altogether.
I'd greatly appreciate any help or thoughts!
try: RTP Packet size on SIP tab
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Any ideas what could be going on and how to fix it. I thought it could
be a timing thing. The documentation on the Sipura phones is
non-existent at the moment, so I have no idea what might be able to be
changed.
I’d greatly appreciate any help or thoughts!
How about disabling silence
Are any of the phones setup to use a codec payload of more
than 20ms? Bugid 5697 on the
bug tracker has a patch to deal with very poor MeetMe
performance when any of the participants
are using audio packetization greater than
20ms.
Beta1 and beta2 did not have this problem, and I am not
Upgrade if you can. I remember submitting a report to
the ooH323c developers about this
some months ago and the fixed it right
away.
Dan
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
jacobso1Sent: Thursday, December 08, 2005 8:21 AMTo:
I have a related issue.
I have everything set up correctly so that I CAN use live recording
(Press *1 to start and stop recording.)
When I press *1, the console indicates user pressed *1 to start
recording. I also hear the beep and an audio file is created.
The problem is that the audio
Asterisk can only be configured as a GW AFAIK, what ever flavor of H323
you use with Asterisk it will not work as a GK.
Atif
rommel malana wrote:
Hello,
Right now i'm trying to set-up a gatekeeper and i'm having a
hardtime doing it, what i'm thinking is instead of having a
On Thu, Dec 08, 2005 at 04:21:00PM +0100, Kib Eki wrote:
Hi,
is it possible to run an octoBri card together with a TE405P card in one
system with bristuff?
One question regarding zapbri: the bristuff patch is basically:
1. a major change to libpri
2. some relevant adjustments to chan_zap
3.
Currently Im running asterisk @ home 1.5 and a Lucent
Max TNT. I want to use the Max as a PSTN gateway for @home. To do
this I have a PRI terminated to the Max TNT.
As you can see below I have established a SIP trunk between
@home and the MAX TNT.
asterisk1*CLI sip show peers
gNumber has written an application, UnWired Buyer, based on Asterisk. To
show our thanks, we would like to extend an offer to the community. We
are currently offering an PayPal credit of $10 to everyone that signs up
and uses the service within the first 30 days. However if you use the
Hi!
This and Time Bandit's comment makes sense. I didn't realize that
these options in the Dial string will force Asterisk to stay in the
media path even if canreinvite=yes.
You might even have another option: DTMF via SIP INFO
Quote from asterisk-devel two days ago:
This depends on the type of signalling you use!
ISDN uses only 1 D channel that is chan 16 for EuroISDN. All other
variants of ISDN (Q.Sig, 1TR, DPNSS, PSS1 etc) I know do the same.
SS7/C7 is a different story as they can use up to 30 'links', but the
most common is actually still 1 link at
http://www.voip-info.org/wiki-asterisk
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
On 12/8/05, John Voss [EMAIL PROTECTED] wrote:
Is there a document/wiki/web site that maps the various SIP.conf settings to
the structure of the actual IP packet?
If so please
This sounds like a prime candidate for a database implementation. That way
you can get very near real-time stats without the overhead of frequent
cronjobs or polling. You number crunching computer would then just grab
the data and crunch away. I'm just now getting started on using Asterisk
in the
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