Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?

[Jonathan Rose]

We worked on this bug today and are expecting to release packages with the fix 
soon, possibly tomorrow (Aug 2).  The issue arose from a change in features 
reload which was back-ported to 1.6.2 and was committed without enough testing 
to observe the intermittent crash behavior.

Thanks for your patience,
Jonathan R. Rose

- Original Message -
From: "Vahan Yerkanian" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Sunday, July 31, 2011 3:03:07 AM
Subject: Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame 
to fix such bugs?

On 7/30/11 7:39 AM, Bruce B wrote:
> I think this should be a quick fix since it's rendering the latest
> stable version useless and making the impression that it was released
> just to break things and force people onto 1.8x. Just a thought...no
> blame game. But really something like this should be tackled quickly. No
> point to break things so badly on the last stable version.
>
> Regards,
>

Confirming this issue on 10+ 1.6.2.19 Asterisk servers. This problem 
makes it difficult to do edits on sip.conf on production systems, as 
there is ~25% chance that you'll crash the server and cut the 
established calls. The problem does not exist in 1.6.2.18...

I think this problem should be fixed or the 1.6.2.19 should be removed 
from the digium repo.

Regards,
Vahan

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Re: [asterisk-users] T38 Fax

[Larry Moore]

On 2/08/2011 4:13 AM, Robert Huddleston wrote:


Thanks -- and did you find a provider with T.38 DIDs? I don't see many 
pay as you go providers with T.38





I am not looking for VoIP providers for such functionality and is 
subjective to geographic location, however I am of the opinion that one 
should have a PSTN line connected directly to a fax device at CPE for 
receiving said communications and one could use T.38 for Outbound faxing 
providing the transmissions are of high enough quality


Larry..


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Re: [asterisk-users] Problems with AMI connections (Asterisk 1.8.3.2)

[Kevin P. Fleming]

On 08/01/2011 03:35 PM, Paul Belanger wrote:

On 11-08-01 04:24 PM, Daniel - Asterisk wrote:




You are closing the socket before reading the result of 'Logoff' and
Asterisk is complaining.


Well, he's sending DBPut before reading the result of Login as well.

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Re: [asterisk-users] Problems with AMI connections (Asterisk 1.8.3.2)

[Paul Belanger]

On 11-08-01 04:24 PM, Daniel - Asterisk wrote:



You are closing the socket before reading the result of 'Logoff' and 
Asterisk is complaining.


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[asterisk-users] Problems with AMI connections (Asterisk 1.8.3.2)

[Daniel - Asterisk]

Hi guys, I hope you could help me.

I am trying to get connected through AMI but something is not working. Both
php code and manager.conf were working well in asterisk 1.4

 1. Sometimes it gets connected and sometimes it doesn't:

 == Connect attempt from '192.168.25.241' unable to authenticate
 == Connect attempt from '192.168.25.241' unable to authenticate
 == Manager 'mark' logged on from 192.168.25.241


2. When connected there's a message that appears just before the connection
crashes:

== Manager 'mark' logged on from 192.168.25.241
[Jul 26 16:29:14] ERROR[1579]: utils.c:1178 ast_careful_fwrite: fwrite()
returned error: Broken pipe
[Jul 26 16:29:14] ERROR[1579]: utils.c:1178 ast_careful_fwrite: fwrite()
returned error: Broken pipe
 == Manager 'mark' logged off from 192.168.25.241


*This is my manager.conf:*

[general]

enabled = yes

port = 5038

bindaddr = 0.0.0.0

webenabled = no


[mark]

secret = mysecret
permit=0.0.0.0/255.255.255.0

read = system,call,log,verbose,command,agent,user,originate

write = system,call,log,verbose,command,agent,user,originate


*This is my code in PHP*:




*These are the software I am running:*


Sistema operativo
àDebian 6.0  squeeze   2.6.32-5-686

Version de asterisk
à   Asterisk 1.8.3.2

Version de Php
àphp 5.3.3-7+squeeze1



Thanks ins advance.


Elder
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Re: [asterisk-users] T38 Fax

[Robert Huddleston]

Thanks - and did you find a provider with T.38 DIDs? I don't see many pay as
you go providers with T.38

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Larry Moore
Sent: Monday, August 01, 2011 4:08 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] T38 Fax

 

On 2/08/2011 1:02 AM, Robert Huddleston wrote: 

Anyone have any testing experience with T38 and HT-502 Grandstream?

 

I just want to confirm that t.38 is working on this device.

 

Thanks

 

 


Yes, it works.

I currently have latest firmware installed and it still works in T.38. I am
using UDP transport for this device as I seem to encounter problems with TCP
or TLS.

I am currently running Asterisk 1.8.5.0.


Product Model: 

  HT-502 V1.1C 


Software Version: 

  Program-- 1.0.5.5Bootloader-- 1.0.0.9Core-- 1.0.5.2Base--
1.0.5.2


Some settings I have set and you may wish to check for the FXS port are;


Force INVITE: 

  (X) No ( ) Yes (Always refresh with INVITE instead of UPDATE)


Send Re-INVITE After Fax: 

  ( ) No (X) Yes 

 


VAD: 

  ( ) No   (X) Yes 


Symmetric RTP: 

  (X) No   ( ) Yes 


Fax mode: 

  (X) T.38 (Auto Detect)   ( ) Pass-Through 


Fax tone detection mode: 

  ( ) Caller   (X) Callee   ( ) Caller or Callee


Jitter buffer type: 

  (X) Fixed   ( ) Adaptive 


Jitter buffer length: 

  (X) Low ( ) Medium   ( ) High 



You will need to ensure you are using redundancy mode instead of FEC.

I am able to send a fax via my voice provider seemingly without errors even
though ECM is not enabled, this is because redundancy mode is working as
expected on the outbound communication.

Unfortunately my voice provider only sends one data item in the incoming
UDPTL hence the occasional missed line.

Here is an extract from my sip.conf

[general]
.
.
t38pt_udptl=yes,redundancy,maxdatagram=400
.
.
[906]
; Grandstream HT502 FXS Port
; Analogue FAX Modem attached
type=friend
defaultuser=906
md5secret=c5bca943c9b0cc303c496fbf9d48a48e
call-limit=1
disallow=g722
transport=udp
qualify=yes
directmedia=no
host=dynamic
context=FAX-T38
faxdetect=no
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/255.0.0.0
permit=172.16.0.0/255.240.0.0
permit=192.168.0.0/255.255.0.0

Larry.

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Re: [asterisk-users] T38 Fax

[Larry Moore]

On 2/08/2011 1:02 AM, Robert Huddleston wrote:


Anyone have any testing experience with T38 and HT-502 Grandstream?

I just want to confirm that t.38 is working on this device.

Thanks




Yes, it works.

I currently have latest firmware installed and it still works in T.38. I 
am using UDP transport for this device as I seem to encounter problems 
with TCP or TLS.


I am currently running Asterisk 1.8.5.0.

*Product Model: * HT-502 V1.1C
*Software Version: * 	  Program-- 1.0.5.5Bootloader-- 1.0.0.9 
   Core-- 1.0.5.2Base-- 1.0.5.2



Some settings I have set and you may wish to check for the FXS port are;

/Force INVITE: /No Yes (Always refresh with INVITE instead of UPDATE)
/Send Re-INVITE After Fax: /No Yes


/VAD: / No Yes
/Symmetric RTP: /   No Yes
/Fax mode: /T.38 (Auto Detect) Pass-Through
/Fax tone detection mode: / Caller Callee Caller or Callee
/Jitter buffer type: /  Fixed Adaptive
/Jitter buffer length: /Low Medium High



You will need to ensure you are using redundancy mode instead of FEC.

I am able to send a fax via my voice provider seemingly without errors 
even though ECM is not enabled, this is because redundancy mode is 
working as expected on the outbound communication.


Unfortunately my voice provider only sends one data item in the incoming 
UDPTL hence the occasional missed line.


Here is an extract from my sip.conf

[general]
.
.
t38pt_udptl=yes,redundancy,maxdatagram=400
.
.
[906]
; Grandstream HT502 FXS Port
; Analogue FAX Modem attached
type=friend
defaultuser=906
md5secret=c5bca943c9b0cc303c496fbf9d48a48e
call-limit=1
disallow=g722
transport=udp
qualify=yes
directmedia=no
host=dynamic
context=FAX-T38
faxdetect=no
deny=0.0.0.0/0.0.0.0
permit=10.0.0.0/255.0.0.0
permit=172.16.0.0/255.240.0.0
permit=192.168.0.0/255.255.0.0

Larry.
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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[Ishwar Sridharan]

Richard,

Thanks for the explanation. You were right about the lack of signalling to
indicate that the call has been rejected, One particular service provider,
instead of signalling rightaway that the call has been rejected, gives a
voice message saying 'The user is busy. Please call later.', and doesn't
send any more signals till the default timeout.

Total face-palm moment :)

--
Regards,
Ishwar.


On Mon, Aug 1, 2011 at 7:36 PM, Richard Mudgett  wrote:

> > There is no event for Asterisk to recognize. The PROGRESS message just
> > says that there is an audio message available for the caller to listen
> > to. Asterisk just passes the indication to the peer channel and opens
> > the audio path. It is the caller who must recognize any audio message
> > that their call has been dropped.
> >
> > Thanks for the explanation. Any suggestion on how to recognise that
> > the call has been dropped?
> >
> >
> > As far as ISDN is concerned, the
> > call has not been answered yet so Asterisk must keep waiting.
> >
> As far as the ISDN signaling is concerned, the call is still going.
> There is no signaling to indicate the call is not going to proceed
> any further.
>
> Richard
>
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Re: [asterisk-users] T38 Fax with Grandstream HT-502

[Robert Huddleston]

My apologies - yes.. Grandstream HT-502...

Apparently finding a t.38 provider is also another struggle...

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P.
Fleming
Sent: Monday, August 01, 2011 1:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] T38 Fax with Grandstream HT-502

On 08/01/2011 12:02 PM, Robert Huddleston wrote:
> Anyone have any testing experience with T38 and HT-502 Grandstream?
>
> I just want to confirm that t.38 is working on this device.

You'd be more likely to get relevant responses if you had included the 
information about the HT-502 in your message subject :-)

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] T38 Fax with Grandstream HT-502

[Kevin P. Fleming]

On 08/01/2011 12:02 PM, Robert Huddleston wrote:

Anyone have any testing experience with T38 and HT-502 Grandstream?

I just want to confirm that t.38 is working on this device.


You'd be more likely to get relevant responses if you had included the 
information about the HT-502 in your message subject :-)


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[asterisk-users] T38 Fax

[Robert Huddleston]

Anyone have any testing experience with T38 and HT-502 Grandstream?

 

I just want to confirm that t.38 is working on this device.

 

Thanks

 

 

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Re: [asterisk-users] ISAC and Asterisk

[Steve Underwood]

On 08/01/2011 07:43 PM, Kevin P. Fleming wrote:

On 08/01/2011 04:12 AM, CB wrote:

On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote:

Are there any plans to include the ISAC codec in Asterisk? Is it

possible or

even desirable? Is ISAC open source (nothing indicates it is from the

WebRTC

website http://www.webrtc.org)?


What do you need it for?


The possibility of having a web-based softphone without requiring any
plug-in is interesting. The adaptive nature of the ISAC codec could also
prove useful. I see lots of possibilities in the mobile device space.

I guess the lack of responses gives me the answer anyway!


The IETF Opus codec is nearing completion, and it is very likely that 
it will be incorporated into the WebRTC stack soon after that. Given 
that, there's not much reason to spend time working on ISAC.


A counter argument to that might be that Opus is fresh and new and 
nobody knows what patent issues might come crawling into view. iSAC has 
been around for a while. The source wasn't open until recently, but 
licenced users have had it for a long time. There has been much more 
opportunity for patent issues to show up.


Steve


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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[Ishwar Sridharan]

Richard,

I tried calling the same number outside of Asterisk, by making direct calls
from a landline telephone and a mobile phones. When the user rejected the
call, the call was immediately cancelled.

This implies that for whatever reason, the call reject signal is not
available for asterisk to process, even though the network processes the
event.

We use a digium TE420 card(
http://www.digium.com/en/products/digital/te420.php) to connect to the PRI.
To debug the scenario, how do I do the following:
a. Is there an ISDN signal for call reject/denied?
   I'm sure there is, for reasons explained above.
b. Is my service provider passing along the call reject/denied signal to the
PRI?
c. If the signal is passed along to the PRI, why is the card not recognising
the signal?

Call Reject is a pretty common feature and is in common use everywhere.
There must be a simple way to fix this.

--
Thanks,
Ishwar.

On Mon, Aug 1, 2011 at 7:36 PM, Richard Mudgett  wrote:

> > There is no event for Asterisk to recognize. The PROGRESS message just
> > says that there is an audio message available for the caller to listen
> > to. Asterisk just passes the indication to the peer channel and opens
> > the audio path. It is the caller who must recognize any audio message
> > that their call has been dropped.
> >
> > Thanks for the explanation. Any suggestion on how to recognise that
> > the call has been dropped?
> >
> >
> > As far as ISDN is concerned, the
> > call has not been answered yet so Asterisk must keep waiting.
> >
> As far as the ISDN signaling is concerned, the call is still going.
> There is no signaling to indicate the call is not going to proceed
> any further.
>
> Richard
>
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Re: [asterisk-users] Serious bug in 1.6.2.19 - what is the time frame to fix such bugs?

[Bruce B]

Thanks for feedback. Yeah, tell me about it. Your description is very
accurate of the situation. I can't believe it's in the repo without any
tests done; even the simplest reload. I don't mean to be a whiner but
honestly the repo is a joke with such an obvious flaw for so long
now

On Sun, Jul 31, 2011 at 4:03 AM, Vahan Yerkanian  wrote:

> On 7/30/11 7:39 AM, Bruce B wrote:
>
>> I think this should be a quick fix since it's rendering the latest
>> stable version useless and making the impression that it was released
>> just to break things and force people onto 1.8x. Just a thought...no
>> blame game. But really something like this should be tackled quickly. No
>> point to break things so badly on the last stable version.
>>
>> Regards,
>>
>>
> Confirming this issue on 10+ 1.6.2.19 Asterisk servers. This problem makes
> it difficult to do edits on sip.conf on production systems, as there is ~25%
> chance that you'll crash the server and cut the established calls. The
> problem does not exist in 1.6.2.18...
>
> I think this problem should be fixed or the 1.6.2.19 should be removed from
> the digium repo.
>
> Regards,
> Vahan
>
>
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Re: [asterisk-users] Capturing call Reject/Decline events on a PRI line

[Richard Mudgett]

> There is no event for Asterisk to recognize. The PROGRESS message just
> says that there is an audio message available for the caller to listen
> to. Asterisk just passes the indication to the peer channel and opens
> the audio path. It is the caller who must recognize any audio message
> that their call has been dropped.
> 
> Thanks for the explanation. Any suggestion on how to recognise that
> the call has been dropped?
> 
> 
> As far as ISDN is concerned, the
> call has not been answered yet so Asterisk must keep waiting.
> 
As far as the ISDN signaling is concerned, the call is still going.
There is no signaling to indicate the call is not going to proceed
any further.

Richard

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Re: [asterisk-users] sip attacks

[Faiyaz Ahmed]

Dear Robert

Are you at live IP ???

--- On Sun, 7/31/11, Robert-iPhone  wrote:

From: Robert-iPhone 
Subject: Re: [asterisk-users] sip attacks
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Cc: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Date: Sunday, July 31, 2011, 4:26 PM

hard to equate sip attack to ping performance.. Run mtr for a bit.
Also try tcpdump or wireshark or tethereal.
If you are really paranoid recycle all your passwords

Sent from my iPhone

On Jul 31, 2011, at 7:04 PM, "Dave George"  wrote:

> My asterisk server is getting bogged down every 5 minutes.  My ping time is
> going from 60ms to 800 ms and the call quality is bad.
> 
> I have fail2ban running and I am using iptables.  I have two ip connections
> to the box.
> 
> How can I tell if the poor performance is due to sip attacks?   I don't see
> any reg attempts in my asterisk cli.  I use to get frequent attacks but
> fail2ban seems to be taking care of that.
> 
> See how ping time gets worst in a short space of time and server performance
> at the time:
> 
> 
> 64 bytes from 4.2.2.1: icmp_seq=6 ttl=55 time=87.8 ms
> 64 bytes from 4.2.2.1: icmp_seq=7 ttl=55 time=99.8 ms
> 64 bytes from 4.2.2.1: icmp_seq=8 ttl=55 time=107 ms
> 64 bytes from 4.2.2.1: icmp_seq=9 ttl=55 time=115 ms
> 64 bytes from 4.2.2.1: icmp_seq=10 ttl=55 time=120 ms
> 64 bytes from 4.2.2.1: icmp_seq=11 ttl=55 time=122 ms
> 64 bytes from 4.2.2.1: icmp_seq=12 ttl=55 time=123 ms
> 64 bytes from 4.2.2.1: icmp_seq=13 ttl=55 time=126 ms
> 64 bytes from 4.2.2.1: icmp_seq=14 ttl=55 time=122 ms
> 64 bytes from 4.2.2.1: icmp_seq=15 ttl=55 time=142 ms
> 64 bytes from 4.2.2.1: icmp_seq=16 ttl=55 time=142 ms
> 64 bytes from 4.2.2.1: icmp_seq=17 ttl=55 time=137 ms
> 64 bytes from 4.2.2.1: icmp_seq=18 ttl=55 time=186 ms
> 64 bytes from 4.2.2.1: icmp_seq=19 ttl=55 time=255 ms
> 64 bytes from 4.2.2.1: icmp_seq=20 ttl=55 time=310 ms
> 64 bytes from 4.2.2.1: icmp_seq=21 ttl=55 time=387 ms
> 64 bytes from 4.2.2.1: icmp_seq=22 ttl=55 time=445 ms
> 64 bytes from 4.2.2.1: icmp_seq=23 ttl=55 time=514 ms
> 64 bytes from 4.2.2.1: icmp_seq=24 ttl=55 time=583 ms
> 64 bytes from 4.2.2.1: icmp_seq=25 ttl=55 time=650 ms
> 64 bytes from 4.2.2.1: icmp_seq=26 ttl=55 time=715 ms
> 64 bytes from 4.2.2.1: icmp_seq=27 ttl=55 time=783 ms
> 64 bytes from 4.2.2.1: icmp_seq=28 ttl=55 time=821 ms
> 64 bytes from 4.2.2.1: icmp_seq=29 ttl=55 time=810 ms
> 64 bytes from 4.2.2.1: icmp_seq=30 ttl=55 time=832 ms
> 64 bytes from 4.2.2.1: icmp_seq=31 ttl=55 time=812 ms
> 64 bytes from 4.2.2.1: icmp_seq=32 ttl=55 time=821 ms
> 64 bytes from 4.2.2.1: icmp_seq=33 ttl=55 time=826 ms
> 64 bytes from 4.2.2.1: icmp_seq=34 ttl=55 time=815 ms
> 64 bytes from 4.2.2.1: icmp_seq=35 ttl=55 time=821 ms
> 64 bytes from 4.2.2.1: icmp_seq=36 ttl=55 time=824 ms
> 
> top - 19:02:38 up 4 days, 11:26,  4 users,  load average: 0.36, 0.75, 0.82
> Mem:   4051312k total,  1062964k used,  2988348k free,   167004k buffers
> Swap:  6094840k total,        0k used,  6094840k free,   680144k cached
> 
>  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
> 4245 root      15   0  791m  86m  10m S 39.6  2.2   1192:32 asterisk
> 18280 root      15   0  3812  600  516 S  2.0  0.0   0:59.00 pppoe
> 2582 root      15   0  5912  628  504 S  0.3  0.0   2:02.19 syslogd
> 18978 root      15   0 12744 1096  812 R  0.3  0.0   0:00.02 top
>    1 root      15   0 10352  700  588 S  0.0  0.0   0:01.14 init
>    2 root      RT  -5     0    0    0 S  0.0  0.0   0:00.01 migration/0
>    3 root      34  19     0    0    0 S  0.0  0.0   0:31.90 ksoftirqd/0
>    4 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00 watchdog/0
>    5 root      RT  -5     0    0    0 S  0.0  0.0   0:00.01 migration/1
>    6 root      34  19     0    0    0 S  0.0  0.0   0:08.43 ksoftirqd/1
>    7 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00 watchdog/1
>    8 root      RT  -5     0    0    0 S  0.0  0.0   0:00.13 migration/2
>    9 root      34  19     0    0    0 S  0.0  0.0   2:40.56 ksoftirqd/2
>   10 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00 watchdog/2
>   11 root      RT  -5     0    0    0 S  0.0  0.0   0:00.05 migration/3
>   12 root      34  19     0    0    0 S  0.0  0.0   0:44.56 ksoftirqd/3
>   13 root      RT  -5     0    0    0 S  0.0  0.0   0:00.00 watchdog/3
>   14 root      10  -5     0    0    0 S  0.0  0.0   0:00.02 events/0
>   15 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 events/1
>   16 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 events/2
>   17 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 events/3
>   18 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 khelper
>   55 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 kthread
>   62 root      10  -5     0    0    0 S  0.0  0.0   0:00.07 kblockd/0
>   63 root      10  -5     0    0    0 S  0.0  0.0   0:00.01 kblockd/1
>   64 root      10  -5     0    0    0 S  0.0  0.0   0:00.00 kblockd/2
>   65 root   

Re: [asterisk-users] ISAC and Asterisk

[Kevin P. Fleming]

On 08/01/2011 04:12 AM, CB wrote:

On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote:

Are there any plans to include the ISAC codec in Asterisk? Is it

possible or

even desirable? Is ISAC open source (nothing indicates it is from the

WebRTC

website http://www.webrtc.org)?


What do you need it for?


The possibility of having a web-based softphone without requiring any
plug-in is interesting. The adaptive nature of the ISAC codec could also
prove useful. I see lots of possibilities in the mobile device space.

I guess the lack of responses gives me the answer anyway!


The IETF Opus codec is nearing completion, and it is very likely that it 
will be incorporated into the WebRTC stack soon after that. Given that, 
there's not much reason to spend time working on ISAC.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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_
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Re: [asterisk-users] different format in asterisk

[Kevin P. Fleming]

On 08/01/2011 12:53 AM, Nikhil wrote:

Does anyone know about this...

On 06/20/2011 04:34 PM, Nikhil wrote:

Hi
In asterisk channel ,I so number of variable regarding the Codec ,Can
anyone explain what are those variable variable means.Below are the
variables

1. chan->readformat

2. chan->writeformat

3. chan ->rawreadformat

4. chan ->rawwriteformat

5. chan->nativeformats


Code questions should be posted to the asterisk-dev mailing list.

--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

--
_
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Re: [asterisk-users] ISAC and Asterisk

[CB]

> On Thu, Jul 21, 2011 at 06:29:38AM +1200, CB wrote:
> > Are there any plans to include the ISAC codec in Asterisk? Is it
> possible or
> > even desirable? Is ISAC open source (nothing indicates it is from the
> WebRTC
> > website http://www.webrtc.org)?
> 
> What do you need it for?
> 
The possibility of having a web-based softphone without requiring any
plug-in is interesting. The adaptive nature of the ISAC codec could also
prove useful. I see lots of possibilities in the mobile device space.

I guess the lack of responses gives me the answer anyway!


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

asterisk-users mailing list
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   http://lists.digium.com/mailman/listinfo/asterisk-users