* You are viewing the archive for November 10th, 2010

1.4.36 – Warning Dropping incompatible voice frame on Local/ on multiple atxfer a->b->c…->d…

Hi

Does anyone have the same problem, or know the solution?

Multiple warning messages on Asterisk 1.4.36: Dropping incompatible
voice frame on Local/….

when receiving calls with codec A and doing multiple attended
transfers to codec B

Reproduced with the following channel combinations

SIP -> SIP -> SIP…
IAX -> SIP -> SIP…
DAHDI -> SIP -> SIP..

Tested in different systems that I’ve upgraded from 1.4.22 to 1.4.36,
tested with different codecs.

g729 -> alaw
ilbc -> alaw
alaw -> ilbc
alaw -> g729

Tried to set transcode_via_sln=no and restarting *, it has no effect.

moh files in wav format, also tried moh files in multiple formats to
avoid transcoding..

===============
Test Multiple call transfer
===============

A with codec X calls B with codec Y

B answers the call

B attended Transfer to C -> Ok

C attended Transfer to D -> Multiple warning messages 139…. 150
Dropping incompatible voice frame on Local/

D attended Transfer to E -> Multiple warning messages 139…. 150
Dropping incompatible voice frame on Local/

[Nov 10 16:12:38] NOTICE[4463] channel.c: Dropping incompatible voice
frame on Local/413@ramais-administrativo-81ea,2 of format slin since
our native format has changed to 0x8 (alaw)

[Nov 10 16:12:38] NOTICE[4463] channel.c: Dropping incompatible voice
frame on Local/413@ramais-administrativo-81ea,2 of format slin since
our native format has changed to 0x8 (alaw)

Can it cause a problem to asterisk?

I have about 7K to 10K of this warning messages per day in all my
systems with 1.4.36 version.

Reboot any(?) SIP Polycom — provisioned or no.

Hey, all. I’m working on making a script to auto-provision my Polycoms.
I wanted one that:

– Gets the MAC by itself
– Fills in the provisioning info you supplied on a web page
– Creates appropriate files
– Reboots the phone (which then gets provisioned)

The last part was the sticking one, though. I found plenty of ways to
make them reboot — but most required an already-provisioned phone, kind
of defeating my purpose.

This will work with these two limitations:
* The phone has default username/password
* You don’t care about your NAT keepalive time; I imagine most don’t.
(See inline comments for more info.)

If that’s your type of thing, enjoy!

-Ken

Selecting ‘ODBC_STORAGE’ from outside of ‘menuselect’

On Wed, Nov 10, 2010 at 3:01 PM, Jose P. Espinal wrote:
> Is that documented somewhere? (I could not find details in the ATFOT book)
>
I’m not sure if it is. I’ve already add some notes to a wiki page [1]
I am working on.

[1] https://wiki.asterisk.org/wiki/display/~pabelanger/Compiling+Asterisk+for+Developers

Selecting ‘ODBC_STORAGE’ from outside of ‘menuselect’

Thank you very much Paul.

Is that documented somewhere? (I could not find details in the ATFOT book)

Paul Belanger wrote:
> On Wed, Nov 10, 2010 at 2:11 PM, Jose P. Espinal wrote:
>> Is it possible to select ODBC_STORAGE without entering to ‘menuselect’?
>>
> $ ./configure
> $ make menuselect.makeopts
> $ menuselect/menuselect –enable ODBC_STORAGE menuselect.makeopts
> $ make
> …
>

Problem with AMI

Hi to all.

I have a problem in the AMI. Sometimes the AMI don’t generate the event
NewState when the exten of destiny is Ringing and sometimes don’t show me
the callerid in this events.

The event NewState what i refer:

Event: Newstate
Privilege: call,all
Channel: SIP/17-00006fd6
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum: 4191920902
CallerIDName: 4191920902
Uniqueid: 1289414204.29705

This guy is ok. But sometimes the event come like this:

Event: Newstate
Privilege: call,all
Channel: SIP/17-00006fd6
ChannelState: 5
ChannelStateDesc: Ringing
CallerIDNum:
CallerIDName:
Uniqueid: 1289414204.29705

And sometimes these events don’t come. I think this is a bug, correctly?

My Asterisk version is the 1.6.0.28. I use ATA Linksys in the extensions and
this problem happen in every hardware’s (khomp, xorcom) and in the SIP
protocol to.

Note: Every teste made is from the same number (my cell) and to same
extension of destiny.

Thanks a lot to all.

Asterisk 1.8 — queue not recognizing that agent is busy

Hi All,

I’ve got a realtime queue in place (strategy is “wrandom”), and have
added a member dynamically via “queue add member “. My agent shows in
the queue, but when he gets the call is not recognized as “In Use”.
Here is the output from “queue show” prior to the call:

*CLI> queue show
QUEUE_3 has 0 calls (max unlimited) in ‘wrandom’ strategy (0s
holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
Todd1 (SIP/14153436587@jnctn) (dynamic) (Not in use) has taken
no calls yet
No Callers

Here is the output when actually connected to the inbound caller:

*CLI> queue show
QUEUE_3 has 0 calls (max unlimited) in ‘wrandom’ strategy (1s
holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
Members:
Todd1 (SIP/14153436587@jnctn) (dynamic) (Not in use) has taken
no calls yet
No Callers

If I make a second inbound call while the agent is still connected to
the first, the second call is also routed to the agent. Queue doesn’t
appear know the agent is already busy.

My question: doesn’t the Queue application keep track of the agent’s
interface status?

Todd