Dear,
the sip phones that registered, in to the asterisk 1.4.x have the echo in their
callings to pstn.
how this echo can be canceled?
Best
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Pezhman Lali wrote:
Dear,
the sip phones that registered, in to the asterisk 1.4.x have the echo
in their callings to pstn.
how this echo can be canceled?
H - you don't give much to go on...
What is the connection to the PSTN (i.e. what kind of card, interface
etc...)
The echo is
On Thu, Nov 20, 2008 at 08:25:54AM +0100, Olivier wrote:
2008/11/17 Philipp Kempgen [EMAIL PROTECTED]
Tilghman Lesher schrieb:
On Thursday 13 November 2008 08:16:42 Klaus Darilion wrote:
Is there somewhere a statement from Digium how long they will support
Asterisk 1.4?
There
On Wed, Nov 19, 2008 at 09:40:49PM -0500, Tom Browning wrote:
FWD (Free World Dialup) allows any SIP call to US toll free numbers via *
[EMAIL PROTECTED] This works WITHOUT the need to be registered at
FWD so in my dialplan I have something like:
exten =
thanks for your writing
the network is like behind:
Sip phone(caller)---SERAsterisk world telephone
carrier---pstn--callee
the carriers and their routers are not accessible for us , the callees have not
any echo in their callings but the caller(sip phones) has the echo.
there isn't
hi,
I am working on a project to perform the voip call quality.
i want to get some statistics about the call quality with asterisk.
I used the following command: iax2 show netstats
and the result changes depending on the configuration of iax.conf.
When i enable jitterbuffer=yes and
Hakan C wrote:
I've never used BRI but you can take a look to wiki.sangoma.com
http://wiki.sangoma.com
As you probably could have imagined after reading my post, I did this
already.
But while their support for analog and PRI cards seems to be really good
(from what you hear around), I
Dear Sir,
I need to configure my Voice Mail on asterisk...I made the following
configuration:
*
extensions.conf:*
exten = _999.,1,VoiceMail(${EXTEN})
exten = _999.,2,HangUp()
If the customer dial 9991234 then a prompt message should ask him to enter
his voice message and this what is not
michel freiha wrote:
Dear Sir,
I need to configure my Voice Mail on asterisk...I made the following
configuration:
_
extensions.conf:_
exten = _999.,1,VoiceMail(${EXTEN})
exten = _999.,2,HangUp()
exten = _999.,1,VoiceMail([EMAIL PROTECTED]|u)
You need to include the context a2billing
Is this a competition about how many levels of quotes the list
can handle or something? SCNR. ;-)
Steve Totaro schrieb:
On Wed, Nov 19, 2008 at 8:58 PM, Jeff LaCoursiere [EMAIL PROTECTED] wrote:
On Wed, 19 Nov 2008, Steve Totaro wrote:
On Wed, Nov 19, 2008 at 8:29 PM, Jeff LaCoursiere [EMAIL
Hello All,
I want to collect the Digits input by the Callee after
the Call is connected, i use the Dial Application to connect the Caller
and Calllee, please
can someone tell me how to do this ? ( Asterisk is in the media path (
Asterisk Version 1.4) ) .
I have looked at
Dear All,
Kindly let me know please where I can fix the payload of DTMF to 101...I'm
using RFC2833
Regards
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Simith Nambiar wrote:
Hello All,
I want to collect the Digits input by the Callee after
the Call is connected, i use the Dial Application to connect the Caller
You'll want to look at the read application.
Doug
--
Ben Franklin quote:
Those who would give up
Tom Browning wrote:
FWD (Free World Dialup) allows any SIP call to US toll free numbers
via [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
This works WITHOUT the need to be registered at FWD so in my dialplan
I have something like:
exten = _8.,1,Dial(SIP/fwd.pulver.com/*${EXTEN:1},60,r
Yeah what Doug said ;), for more info check out:
http://www.voip-info.org/wiki-Asterisk+cmd+Read
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Thursday, November 20, 2008 8:49 AM
To: Asterisk Users Mailing List - Non-Commercial
Hello Darrin / Doug,
Thank you for your response, i find
that the Read Aplication blocks for input and returns when a DTMF is
dialled, which is fine.
My problem is that when i use the Dial Application , it is blocking too,
so wheee do i put the Read call in my
Hello!
We're looking for a solution to reliably load balance our
Asterisk boxes. So far we've been using a hodge-podge of
directing different services to different boxes/IPs, but
eventually I'd like to consolidate things so we can present
a single IP address to the outside world.
My question is
Hi
I do have asterisk running in real time I do want to add voicemail to real
time. I did follow :
http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail
However when I do try to make a voicemail I do get :
[Nov 20 12:17:04] NOTICE[16501]: chan_sip.c:5506 process_sdp: No compatible
codecs,
2008/11/20 Nitzan Kon [EMAIL PROTECTED]
Hello!
We're looking for a solution to reliably load balance our
Asterisk boxes. So far we've been using a hodge-podge of
directing different services to different boxes/IPs, but
eventually I'd like to consolidate things so we can present
a single IP
--- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote:
2 openser servers with 3 ip adresses (1 virtual) +
heartbeat to ensure the
failover + watchdog to ensure if opensips/kamalio/openser
crashes a nice
failover reboot, it is working stable here
(dispatching to 10 servers +
What do you mean by hardware options? There are no ASIC-assisted SIP
load balancers out there. :-) The embedded hardware-based options
are load balancers built just like PCs - often on top of a UNIX kernel -
that run a software application-aware load balancing suite.
Your best bet is a
Ali Jawad schrieb:
I do have asterisk running in real time I do want to add voicemail to real
time. I did follow :
http://www.voip-info.org/wiki-Asterisk+RealTime+Voicemail
However when I do try to make a voicemail I do get :
[Nov 20 12:17:04] NOTICE[16501]: chan_sip.c:5506
Simith Nambiar wrote:
Hello Darrin / Doug,
Thank you for your response, i find
that the Read Aplication blocks for input and returns when a DTMF is
dialled, which is fine.
My problem is that when i use the Dial Application , it is blocking too,
so wheee
Hardware solutions are of course simply packaged software solutions.
Personally I would go with something that has this wonderful support base
and quick solutions versus dealing with a vendor. You did mention that
price was a consideration, right?
j
On Thu, 20 Nov 2008, Nitzan Kon wrote:
Always a self appoited list Nazi. If it bothers you, then don't
bother reading.
On Thu, Nov 20, 2008 at 7:23 AM, Philipp Kempgen
[EMAIL PROTECTED] wrote:
Is this a competition about how many levels of quotes the list
can handle or something? SCNR. ;-)
Steve Totaro schrieb:
On Wed,
Alex,
I realize and agree that hardware load balancers are actually
software based. I'm less concerned about that and more about the
general specs:
Foundry ServerIron XL: rated for 1,000,000 concurrent connections
Linux box where OpenSIPS is sitting: rated for ...???
Not to mention a simple
Nitzan Kon wrote:
Foundry ServerIron XL: rated for 1,000,000 concurrent connections
Linux box where OpenSIPS is sitting: rated for ...???
Because OpenSER's load balancer is hash-based and not stateful, it is
rated for far, far more than that.
--
Alex Balashov
Evariste Systems
Web:
Unless the LB is SIP-aware, and can maintain a SIP session, I don't see
how it would work. As the SIP command stream sends discrete commands,
without some sort of basic level of session awareness, there's no
guarantee over a reasonable-length call that the INVITE and BYE would
even get sent to the
The solution to make this work and still work statelessly is to hash
various unique identifying bits of the SIP headers without maintaining
transactional, session or dialog information as such.
SIP wrote:
Unless the LB is SIP-aware, and can maintain a SIP session, I don't see
how it would
This baby talks about being able to do hardware SIP load balancing.
http://www.f5.com/news-press-events/press/2007/20070212.html
I've never used an f5 product so I can't provide any comments from
experience. I did look at an f5 load balancer product once and the
price was over 6 figures that was
N,
SIP-aware LBs do exist - but way way out of my price range.
Alex,
Remember we are an Asterisk-based provider. I'm not going
to drop enough money on a load balancer to go bankrupt. ;) That's
exactly why I'm wondering if it's possible to do this with a
DUMB load balancer. i.e. one that would
I was about to say, I'm sure F5 can do it... but...
price was over 6 figures
Why??!
It's spending money on these types of things when they are unnecessary
that is the undoing of every struggling VoIP provider I watch, in the
misguided belief that only will half a million dollars get you
Jared Smith had written:
To answer the second portion of your question (which I forgot to do in
my earlier email)... yes, Asterisk can be a registration server as well.
--
Jared Smith
Training Manager
Digium, Inc.
Valentin Bud wrote:
Hello Mr. Smith,
snip
If you know any kind of books
i want to get some statistics about the call quality with asterisk.
I used the following command: iax2 show netstats
and the result changes depending on the configuration of iax.conf.
When i enable jitterbuffer=yes and forcejitterbuffer=yes, i get the
following result:
voip*CLI iax2 show
Nitzan Kon wrote:
My concerns with OpenSIPS:
1. It's a software based solution, which means higher chance
of software-related failure, and higher chance of failure due
to problems with the Linux box hosting it.
A little bit of proper engineering will overcome that reasonably.
2. Overkill
Alex Balashov wrote:
I was about to say, I'm sure F5 can do it... but...
price was over 6 figures
Why??!
It's spending money on these types of things when they are unnecessary
that is the undoing of every struggling VoIP provider I watch, in the
misguided belief that only will half a
2. Overkill to install and maintain (if we can get a simpler
solution)
I am not agreed on point 2:
If I understood how to install opensips + heartbeat WITHOUT knowing any
program (opensips ? heartbear ?) or programming language(hell yes!) in a
week ( just knew what's invite and bye ;) a more
On Thu, Nov 20, 2008 at 5:35 PM, Bill Andersen [EMAIL PROTECTED] wrote:
Jared Smith had written:
To answer the second portion of your question (which I forgot to do in
my earlier email)... yes, Asterisk can be a registration server as well.
--
Jared Smith
Training Manager
Digium, Inc.
--- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote:
I am not agreed on point 2:
If I understood how to install opensips + heartbeat WITHOUT
knowing any
program (opensips ? heartbear ?) or programming
language(hell yes!) in a
week ( just knew what's invite and bye ;) a more
I create my sip users using a common macro in 1.4:
[internal]
exten = 200,1,Macro(phones|200|SIP/200)
[macro-phones]
exten = s,1,Dial(${ARG2}|45|Tt)
etc...
But now in 1.6 this fails:
-- Executing [EMAIL PROTECTED]:1] Macro(SIP/201-0942b530,
phones|200|SIP/200) in new stack
[Nov 20 08:55:55]
Nitzan Kon wrote:
--- On Thu, 11/20/08, Grygoriy Dobrovolskyy [EMAIL PROTECTED] wrote:
I am not agreed on point 2:
If I understood how to install opensips + heartbeat WITHOUT
knowing any
program (opensips ? heartbear ?) or programming
language(hell yes!) in a
week ( just knew what's
Hi,
For a long time, I was wondering if I should use MAC address instead of
Extension number to identify SIP endpoints (as I'm mostly not using
softphones).
Before diving into this, I wondered how people using MAC address are using
CLI as it seems more natural and simple to type
sip show peer
On Thu, Nov 20, 2008 at 2:50 PM, SIP [EMAIL PROTECTED] wrote:
Tom Browning wrote:
FWD (Free World Dialup) allows any SIP call to US toll free numbers
via [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
This works WITHOUT the need to be registered at FWD so in my dialplan
I have something like:
3. Incoming calls - I admit complete ignorance. I don't know
how OpenSIPS handles incoming calls, but for those to arrive
at the user reliably they must arrive from the same IP address
the user is registered to. Otherwise their broadband router's
NAT firewall will just block the connection.
On Thu, Nov 20, 2008 at 5:57 PM, Joseph L. Casale
[EMAIL PROTECTED] wrote:
I create my sip users using a common macro in 1.4:
[internal]
exten = 200,1,Macro(phones|200|SIP/200)
[macro-phones]
exten = s,1,Dial(${ARG2}|45|Tt)
etc...
But now in 1.6 this fails:
-- Executing [EMAIL
It will auto-complete if you hit tab, just like the shell. But I
would recommend against it. I can't really think of a good reason to
do it. 'sip show peer 268' I can remember to see that status of
extension 268 when somebody calls and says I can't dial 268.
Whereas 'sip show peer
On 20 Nov 2008, at 16:14, Daniel Hazelbaker wrote:
Any reason you want to use the MAC address?
Bet he used to use Cisco ;)
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Olivier schrieb:
For a long time, I was wondering if I should use MAC address instead of
Extension number to identify SIP endpoints (as I'm mostly not using
softphones).
Before diving into this, I wondered how people using MAC address are using
CLI as it seems more natural and simple to
2008/11/20 Philipp Kempgen [EMAIL PROTECTED]
Olivier schrieb:
For a long time, I was wondering if I should use MAC address instead of
Extension number to identify SIP endpoints (as I'm mostly not using
softphones).
Before diving into this, I wondered how people using MAC address are
AFAIR it was mentioned in UPGRADE.txt that argument separator was
changed from pipe to comma. Unless you read it, you might also
experience lot of other problems.
Whoops, missed that! I did see the suggestion on GoSub's but as it
stated Macros would still be supported I neglected to attempt to
2008/11/20 Daniel Hazelbaker [EMAIL PROTECTED]
It will auto-complete if you hit tab, just like the shell. But I
would recommend against it. I can't really think of a good reason to
do it. 'sip show peer 268' I can remember to see that status of
extension 268 when somebody calls and says I
A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
having a conversation. Call quality is reported as good except for an
echo with a 3 second delay.
Most of my searches are saying echo happens only on the PSTN piece, but
there isn't one here.
Can someone point me in the right
Mike wrote:
I tried using this iptables sample, and did not see duplicate packets
on '--to-ports' port
Has some verified this is working for them?
I listened on both ports with tcpdump command.
Mike,
I can confirm that it's working. Admittedly, I never looked at the
packets with tcpdump
Nobody responded, but I was able to resolve this issue the way I wanted.
In my extensions.conf I put the following:
[callback-dialtone-auth]
exten = s,1,answer()
exten = s,n,authenticate(5678)
exten = s,n,Read(fwd_callback_to)
exten = s,n,NoOP(${fwd_callback_to})
exten = s,n,Dial(SIP/[EMAIL
I'm not sure about the 3 second delay, but I've seen plenty of echo issues on
Polycom phones when the gain has been changed on the handset. Check the
voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're
not too high.
You also may want to make sure there aren't any
SIP wrote:
As for the current F5 SIP load balancer, we tried it a few years back
and it was a dismal failure. It wanted to do cookie-based SIP load
balancing and only worked with certain SIP proxies.
I assume that is because there is no way RFC-supported way to insert a
cookie into a SIP
On Nov 20, 2008, at 9:02 AM, Olivier wrote:
2008/11/20 Daniel Hazelbaker [EMAIL PROTECTED]
Any reason you want to use the MAC address? If it is just for easy
provisioning, I just put a MAC address field in the realtime SIP table
and use a php script to take the phone's MAC address and feed it
Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
IAX2 is not all it is cracked up to be.
Also, do a ping to see latency, 200ms is pretty much my standard.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
On Thu, Nov 20, 2008 at 12:16
I tweaked the voip-info page a bit to reflect your example correctly (my
example stripped the first digit as I am using 8 as the dial prefix to toll
free via free SIP providers )
On Thu, Nov 20, 2008 at 11:02 AM, Atis Lezdins [EMAIL PROTECTED] wrote:
Wow, that's helpful.
I googled a bit,
Managed to get the portech 370 up and running with asterisk (even got
the callerid working!), but was wondering how (if) it is possible to
send / receive sms messages through the device . All I could find
googling was people asking how ;(
Does anyone have sms working with this device ?
Julian
Philipp Kempgen wrote:
Olivier schrieb:
For a long time, I was wondering if I should use MAC address instead of
Extension number to identify SIP endpoints (as I'm mostly not using
softphones).
Before diving into this, I wondered how people using MAC address are using
CLI as it seems
Hi All,
I have a ticket open with Digium, but based on their previous lack of support
for the Asterisk Appliance, I'm not really holding my breath - and, honestly,
I'm not 100% convinced it's a Digium issue in the first place (but I don't know
where else to point fingers).
We have an AEX-804E
Background:
WAN1 - Fixed IP low latency, low jitter
WAN2 - Fixed IP medium latency, higher jitter than I like for good VoIP
Firewall/Router not SIP aware
NATed LAN
Asterisk on server located on LAN.
Most, but not all ATA/IP phones on LAN
In the past I was running a v1.2 Asterisk which acted as a
2008/11/20 Eric ManxPower Wieling [EMAIL PROTECTED]
Personally I use the MAC-x wherex=the line appearance number. MAC-a for
first line appearance, MAC-b for 2nd, etc.
Is it easy to use (CLI, logs...) ?
Would you step back to an extension-based identification scheme ?
Greetings,
We recently moved our public subversion mirror to a new server. It is
currently down for maintenance while we resolve some unforeseen
problems. It should be back up by the end of the day.
I apologize for the inconvenience,
--
Russell Bryant
Senior Software Engineer
Open Source
Tim Nelson wrote:
I'm not sure about the 3 second delay, but I've seen plenty of echo issues on
Polycom phones when the gain has been changed on the handset. Check the
voice.gain.tx and voice.gain.rx settings in your sip.cfg to make sure they're
not too high.
You also may want to make
Steve Totaro wrote:
Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
IAX2 is not all it is cracked up to be.
Also, do a ping to see latency, 200ms is pretty much my standard.
Coming from outside the network, setting up for a couple rounds of
NATting isn't going to work
On Thu, Nov 20, 2008 at 1:13 PM, c james [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
IAX2 is not all it is cracked up to be.
Also, do a ping to see latency, 200ms is pretty much my standard.
Coming from outside the
On Thu, Nov 20, 2008 at 3:38 AM, Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Thu, Nov 20, 2008 at 08:25:54AM +0100, Olivier wrote:
2008/11/17 Philipp Kempgen [EMAIL PROTECTED]
Tilghman Lesher schrieb:
On Thursday 13 November 2008 08:16:42 Klaus Darilion wrote:
Is there somewhere a
There are also settings which will turn on local echo cancellation for
the handset, headset and/or speaker phone. I don't recall their names at
the moment. They are off by default on the handset and headset unless
you're using a very recent (3.0+) SIP app.
Tim Nelson wrote:
I'm not sure about
Simple tests. Change from the highly touted IAX2 to SIP, but before
that, download X-Lite and see if you have the same delay. If you
don't then look at your Polycoms, if you do, then switch to SIP.
--
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
On
Olivier wrote:
2008/11/20 Eric ManxPower Wieling [EMAIL PROTECTED]
Personally I use the MAC-x wherex=the line appearance number. MAC-a for
first line appearance, MAC-b for 2nd, etc.
Is it easy to use (CLI, logs...) ?
Would you step back to an extension-based identification scheme ?
I
Hello Everyone,
I’ve sold an asterisk system to a client that has a custom written CRM package
written in VB6 with an MS-SQL backend. They want to “unplug” the application
from their old phone system and “plug” it into the asterisk system. The program
has pop-up screens based on incoming
- Mensaje reenviado
De: sas sas [EMAIL PROTECTED]
Para: asterisk-users@lists.digium.com
Enviado: miércoles, 5 de noviembre, 2008 9:20:34
Asunto: sas
I would like to know how can I send and receive sms using asterisk 1.4.22.
If anyone know, please tell me soon.
Thanks.
c james wrote:
A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
having a conversation. Call quality is reported as good except for an
echo with a 3 second delay.
Most of my searches are saying echo happens only on the PSTN piece, but
there isn't one here.
Which end
Eric ManxPower Wieling schrieb:
Philipp Kempgen wrote:
Olivier schrieb:
For a long time, I was wondering if I should use MAC address instead of
Extension number to identify SIP endpoints (as I'm mostly not using
softphones).
Before diving into this, I wondered how people using MAC address
Hello there,
I am wondering if some one could tell me on the average in the U.S. What does a
person with DCap certification make on a standard Asterisk installation and
configuration process as well as a Sip Master. I am looking to go to the
Asterisk course and I am blind and have a state
I would say little if not none. I always give my clients a
satisfaction guaranteel and if they are any good at google, they find
me or get referrals. I actually do not do any marketing and stay
busy. I cannot comment on Sip Master.
I think it is more of a status thing in the community at this
I am calling the console dsp and speaking just fine.
however, I am hearing crackling and all kinds of static.
I go into alsamixer and mute the MIC channel, bring the levels to zero
and nothing affects it.
Why might that be? I expected to not hear anything - especially after
muting the channel
This Wednesday, November 26th, the Toronto Asterisk Users Group invites
all in the area to join us for a telephony workshop and talk sponsored by
Sangoma Inc.[1]
Jose Landivar, co-founder of PaloSanto Solutions[2], creators of Elastix,
will be running a getting started workshop on Elastix,
Ok, I'll bite, what possible IAX bugs/shortcomings/features can cause
echo ?
Tim.
On 20 Nov 2008, at 18:47, Steve Totaro wrote:
Simple tests. Change from the highly touted IAX2 to SIP, but before
that, download X-Lite and see if you have the same delay. If you
don't then look at your
Hi all,
Kindly note that I got the below message when sending DTMF in RFC2833
through asterisk PBX...The DTMF is not going through
RTCP Read too short
I'm using G729 codec and asteriks Asterisk 1.4.21.2
Regards
___
-- Bandwidth and Colocation
c james [EMAIL PROTECTED] writes:
A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
having a conversation. Call quality is reported as good except for an
echo with a 3 second delay.
Feedback from speaker to microphone. The problem is always at the end
which doesn't hear it.
Drew Gibson wrote:
c james wrote:
A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are
having a conversation. Call quality is reported as good except for an
echo with a 3 second delay.
Most of my searches are saying echo happens only on the PSTN piece, but
there isn't one
Steve Totaro wrote:
On Thu, Nov 20, 2008 at 1:13 PM, c james [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
Just for sh1t$ and giggles, try sip to sip and drop the IAX piece.
IAX2 is not all it is cracked up to be.
Also, do a ping to see latency, 200ms is pretty much my standard.
Coming
Hi -
I found the maxusers defined in meetme.c, but I'm not sure how this
value is set. Does anybody know if one can limit the number of users
permitted in a meetme conference? I know there's MeetmeCount(), but
I'd rather avoid the dialplan logic and just set maxusers instead.
Thanks,
Noah
In my meetme.c, users is defined as an int on line 328. This gives a
possibility of 35768 people in a conference. If you cbanged that to a
signed char, you would limit it to 127.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Noah Miller
Sent:
I've got a user with Linksys ATA's for their analog phones. At random times
during calls, the other party hears DTMF tones during the call.
Is there a way to solve this?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com
Is there a way to inject sound from a sound file into an established call
using AMI?
I have an established call from which I can record either or both legs. I
can additionally spy on the call. Is there any way I can play a sound file
into the call and not loose the ability for the people to
Be careful there, the size of a int is not statically defined in c. The
rules in the standard that applies to an int is as follows:
An |int| must not be larger than a |long int|
A |long int| must be at least 32 bits long int
An |int| must be at least 16 bits long.
So an int is at least 16
Noah wrote:
I found the maxusers defined in meetme.c, but I'm
not sure how this value is set. Does anybody know
if one can limit the number of users permitted in a
meetme conference? I know there's MeetmeCount(), but
I'd rather avoid the dialplan logic and just set
maxusers instead.
That
Just set up a new spy in the dialplan that performs a Background on the
sound file, then hangs up.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Dickenson
Sent: Thursday, November 20, 2008 4:34 PM
To: Asterisk User MailList
Subject: [asterisk-users]
Hi All
Just been looking at stats for one of my sites, and I'm conserned about
the number of error cause codes being returned from the telco
for example
12000 calls processed
131 are cause code 31* normal. unspecified.*
139 are cause code 28 * invalid number format (address incomplete).*
I have IMAP voicemail working with Exchange 2003 using a single username
and password for multiple mailboxes.
Right now, I am setting up asterisk to use voicemail with my Cisco Call
Manager (Which I detest BTW...) and I have everything working, EXCEPT:
I cannot get my externnotify script
What is the source of the numbers you are calling? Are they
previously-verified numbers from your database? Are some of them
fumble-fingered manually-dialed calls? I'm pretty sure that I goof on more
than 3% of calls that I manually call. Have you researched some of the
failures (examining the
Coming from outside the network, setting up for a couple rounds of
NATting isn't going to work well. They are not seeing it between
phones. Others, using the polycom phones have reported echo between two
SIP on a 4ms ping trip.
Could this be due to a purely acoustic echo within the Polycom
Does any know what happens with svn repository on svn.digium.com ?
--
-
Luis Morales
Consultor de Tecnologia
Cel: +(58)416-4242091
-
Jeffrey Phelps schrieb:
I have IMAP voicemail working with Exchange 2003 using a single username
and password for multiple mailboxes.
Right now, I am setting up asterisk to use voicemail with my Cisco Call
Manager (Which I detest BTW...) and I have everything working, EXCEPT:
I cannot get
Luis Morales schrieb:
Does any know what happens with svn repository on svn.digium.com ?
http://lists.digium.com/pipermail/asterisk-users/2008-November/222147.html
Philipp Kempgen
--
http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com
Amooma GmbH - Bachstr. 126 - 56566
I am not sure how one would do that. If I have a dialplan that does a
chanspy the dialplan hangs on that step so how could you do a background
step?
I am clearly missing something in your suggestion.
I am using version 1.6.0.1 if that makes a difference.
Here is more about my setup.
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