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…
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…
On Wed, Nov 10, 2010 at 3:01 PM, Jose P. Espinal
> 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 
I am working on.  https://wiki.asterisk.org/wiki/display/~pabelanger/Compiling+Asterisk+for+Developers
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
>> Is it possible to select ODBC_STORAGE without entering to 'menuselect'?
> $ ./configure
> $ make menuselect.makeopts
> $ menuselect/menuselect --enable ODBC_STORAGE menuselect.makeopts
> $ make
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
Uniqueid: 1289414204.29705 This guy is ok. But sometimes the event come like this: Event: Newstate
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
Todd1 (SIP/14153436587@jnctn) (dynamic) (Not in use) has taken
no calls yet
No Callers Here is the…
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…
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 22.214.171.124
7. 1 x IAX2 incoming trunk from phone provider for 1 phone number (2
8. 1 x IAX2 outgoing trunk (theoretically unlimited channels) to phone
how can I set up an peer, so that behind one IP (NAT) multiple devices
can access to this single peer to make outbound calls.
Some of these multiple devices will be SIP phones and these SIP phones
are trying to make registrations to this peer. best regards
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:firstname.lastname@example.org">email@example.com
[mailto:firstname.lastname@example.org] 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…