similar to: Wrap-up

Hi. I was wondering if there's a parameter to limit the number of concurrent sessions in IAX, globally or on a per-user basis. That could be needed for security purposes (to prevent dos attacks), to limit bandwidth / cpu usage, or to not allow more than N guest connections, for example. Any other VoIP channel support that? (like SIP, MGCP) Matteo. -- Brancaleoni Matteo

To all who need more queue functionality, We are contracting Digium to enhance the queue app for our call center needs. Please read the following email conversation and give your ideas. Unless a glaring omission is found in my specification we will have them start tomorrow (Wednesday). I may not have thought of something important. It will be released to all Asterisk users by Digium. Thanks for

Just cvs'ed about 40 minutes ago (10:15 CST 8-8-03). Segfaults when I use a queue app in many different scenarios. When calling phone is only member of queue I get a segfault. When 1st called extension is outside line I get a segfault. Many other scenarios as well. Unsure how to go about troubleshooting. Any ideas? Jim Friedeck

Your log file almost looks like a bug in Asterisk doesn't it? Why call poll() with a zero timeout while passing only one FD? and then why do the read when there is no data? Read the man pages for all the system calls Take a look at the source chan_sip.c /* Wait for sched or io */ res = ast_sched_wait(sched); if ((res < 0) || (res > 1000))

Thanks to Digium for the fine work they are doing on queue app! Does anyone know the progress on the monitor app recording to one file or synchronizing the two files' start and end times? Thanks. Jim Friedeck

I am having trouble with the AgenCallBackLogin app. I can't seem to define a context for the queue. Here is the relevant configs: queues.conf: [general] [default] [q_lo_1] music = default strategy = ringall context = c_in_1 timeout = 15 retry = 2 maxlen = 0 member => Agent/@3 agents.conf: [agents] autologoff=10 wrapuptime=15000 group=1 agent => 1001,1234,Agent1 agent =>

I seem to be encountering a problem with the db.h file when compiling gastman. Incorrect definitions for functions and so on. Here are the errors: If I don't copy the db.h from asterisk source to the include directory of gastman I get: gui.c:31:16: db.h: No such file or directory If I do then I get: gui.c:743: too many arguments to function This is for the line: if (!(res =

How do I stop asterisk when it is in a bad mood? It keeps dialing extensions and won't listen! I tried kill <PID>. No go. I don't want to have to reboot again. Thanks. Jim Friedeck P.S. I love it when my boss looks over my shoulder and I don't have an answer when he says: 'So, what are you doing?'

Our CSR people need to be informed when a call is ringing in when they are on the phone. Is there a mechanism for informing an off-hook target channel of an incoming call? We have a guy who should get first shot at all incoming calls on our local lines and our customer service line. If he is on the phone, he should get beeped and then be able to place the current call on hold to answer the

Anyone know how to specify a call group (as specified in the example zapata.conf) when using Dial? No other reference I can find. Jim Friedeck

Hello. I have this issue, when I pickup a call that is ringing in a SIP Phone, it keeps ringing. There is bug #116 that mention something about these, but it does not seem to be resolved , at least, not yet. Anybody else has seen it behavior ? Thank's.

Nobody in this long thread appears to have yet explained why it's a bad idea to allow atomic fragments of code/data (whatever you want to call them: atoms, sections, who cares) to have more than one global symbol attached to them in LLD's internal representation. That seems like it'd provide the flexibility needed for ELF without hurting MachO. If that change'd allow you to avoid

Hi there, Besides the problem of Voicetronix dialing too early before the carrier gives a dial tone, there also appears to be issues with the audio quality when more than 1 channel is utilized.

I've a big problem with SIP forwarding back into 'ringing groups' creating what can only be described as call storms :-( I have a 'ringing groups' of SIP phones with an effective dialplan (much simplified) like so: ; Purchase ledger [ptsn_inbound] exten => _846061,1,Dial(Local/6061 at groups) .... [groups] exten =>

Folks, please, take a look at this asterisk log message: [Oct 2 08:55:13] WARNING[10290] app_queue.c: Announcement file 'atcert' is unavailable, continuing anyway... [Oct 2 08:55:13] WARNING[10290] app_queue.c: Agent on Agent/1001002 hungup on the customer. but: -bash-3.1$ whoami asterisk -bash-3.1$ ls -ls $HOME/sounds/atcert* 12 -rw-r--r-- 1 asterisk asterisk 10956 Oct 2 07:00

My colleague left our company, then I have to manage all our phones lines and asterisk: please, apologize me because I'm 'absolute beginner' about voip/asterisk!! Well... all seems work fine; we have some queues and some agents; the "music on hold" works fine when the agent press the hold button on the phone (thomson); the agents have the 'autoanwser' flag

I am looking for a way to quickly and easily test for on-hook channels from within a C-language AGI app. CHANNEL STATUS works but is a bit clumsy. I don't want to rely on strcmp'ing the returned '200 result=-1' as the meaning of this might change in the future. I am trying to create an ACD using MySQL and want to test each channel before I Dial it. Any ideas? Jim Friedeck

On Mon, May 11, 2015 at 11:13 AM, Rui Ueyama <ruiu at google.com> wrote: > If you attach two ore more symbols along with offsets to a chunk of data, > it would be a pretty similar to a section. That means that if you want to > do something on the atom model, now you have to treat the atoms like > sections. > What do you lose/pay by having to treat the atoms like sections?

Ok, a little patch that adds a little functionality to call parking. With that, you can pickup the older parked call, if many are in the parking lot. The default exten to do that is 750, but can be changed by setting "parkpick => exten" on parking.conf , like [general] parkext => 800 ; What ext. to dial to park parkpos => 801-820 ; What extensions to

Hi, I have a Asterisk that connects to the PSTN via a PRI. After Asterisk sends the setup message it immediately sends a 183 Session Progress. Is there a way I can change it so that it sends a 100 Trying instead? Because I am having some issues with a equipment where it does not play a busy tone as a result of sending a 183 Session Progress then the 486 Busy. Thanks Remi