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

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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…

Asterisk Users 4.9 years ago 0 Answers

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

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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…

Asterisk Users 4.9 years ago 0 Answers

Selecting 'ODBC_STORAGE' from outside of 'menuselect'

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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
> ...
>

Asterisk Users 4.9 years ago 0 Answers

Problem with AMI

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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…

Asterisk Users 4.9 years ago 0 Answers

Asterisk 1.8 -- queue not recognizing that agent is busy

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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…

Asterisk Users 4.9 years ago 3 Answers

/dev/zap/channel missing

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Dear All, This question is related to an old zaptel version (1.2.14) that I'm using for specific reasons on one of the machines in combination with Sangoma cards. The version of zaptel/asterisk/libpri cannot be changed :-). Even though this is an old version, I still hope somebody will be kind enough to help me a bit. The following problem came out of the blue: whenever I stop Sangoma drivers (command "wanpipe stop") the /dev/zap/channel /dev/zap/pseudo /dev/zap/timer and /dev/zap/transcode devices disappear. Executing command "ls /dev/zap" returns "ctl" only. Starting the drivers and executing "ztcfg -vvv" doesn't help. Asterisk will not start…

Asterisk Users 4.9 years ago 2 Answers

Random call drops on IAX2

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Hello list, I have an Asterisk setup with the following details: 1. 3 x internal extensions / sip hardphones - Grandstream 2000
2. 2 x internal extensions / dahdi cordless phone
3. 1 x 2 FSX ports OpenVOX pci card
4. 1 x internal sip extension / sip softphone (linphone)
5. 1 x 800Mhz Asterisk + Linux server
6. Asterisk version is 1.6.2.13
7. 1 x IAX2 incoming trunk from phone provider for 1 phone number (2
channels).
8. 1 x IAX2 outgoing trunk (theoretically unlimited channels) to phone
provider.

Asterisk Users 4.9 years ago 1 Answer

Store CDR (call detail record) toOracledatabase

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I'd love to, but I have bigger trees to chop in Asterisk that messing with
DB CDR's, so I just use the flat file /var/log/asterisk/cdr-csv/Master.csv.
I'll defer this answer to a poster who has actually climbed that mountain.
Apologies for terse answer and top-post. _____ From: href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Phuong Hoang
Sent: Wednesday, November 10, 2010 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Store CDR (call detail record)
toOracledatabase Thanks Danny.
Can you talk clearer about using ODBC to store cdr…

Asterisk Users 4.9 years ago 1 Answer