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 w..
Hey, all.Im 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 provision..
On Wed, Nov 10, 2010 at 3:01 PM, Jose P. Espinalwrote: > Is that documented somewhere? (I could not find details in the ATFOT book) > Im not sure if it is.Ive already add some notes to a wiki page  I am working on.  https://wiki.asterisk.org/wiki/display/~pabelanger/Compiling+Asterisk+for+Develop..
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. Espinalwrote: >> Is it possible to select ODBC_STORAGE without entering to menusele..
to all. I have a problem in the AMI. Sometimes the AMI dont generate the event NewState when the exten of destiny is Ringing and sometimes dont show me the callerid in this events. The event NewState what i refer: Event: Newstate Privilege: call,..
All, Ive 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 ..
, This question is related to an old zaptel version (1.2.14) that Im 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 versi..
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 softph..
, 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. b..
Id love to, but I have bigger trees to chop in Asterisk that messing with DB CDRs, so I just use the flat file /var/log/asterisk/cdr-csv/Master.csv. Ill defer this answer to a poster who has actually climbed that mountain. Apologies for terse ans..