search for: ast_test_flag

...s causing us a problem. I would prefer to solve the problem by changing a configuration option somewhere but I'm running out of ideas. I've had a look in chan_sip.c and have seen this: case 180: /* 180 Ringing */ case 182: /* 182 Queued */ if (!ast_test_flag(req, SIP_PKT_IGNORE) && (p->invitestate != INV_CANCELLED) && sip_cancel_destroy(p)) ast_log(LOG_WARNING, "Unable to cancel SIP destruction. Expect bad things.\n"); if (!ast_test_flag(req, SIP_PKT_IGNORE) && p->owner) {...

...#39;ve grep'd the entire src folder for \(5\) as well as qxd trying to find all instances of this, and the only ones are listed in the app_page.c file. Any suggestions on where to get this rogue (5) out of here? snprintf(meetmeopts, sizeof(meetmeopts), "%ud|%sqxdw", confid, ast_test_flag(&flags, PAGE_DUPLEX) ? "" : "m"); and if (!res) { snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%sqxd", confid, $ pbx_exec(chan, app, meetmeopts, 1); } are the only sections of the app_page.c that have the meet...

...are different, although the call-id is the same. We have used pedantic checking. Could it be considered as a bug? Looking at the code of chan_sip.c (version 1.4.23.1), we have observed that in function 'find_call' line 4667, asterisk is considering the call as FOUND because of this test: !ast_test_flag(&p->flags[1], SIP_PAGE2_DIALOG_ESTABLISHED). Commenting out this comparison, the call proceeds correctly. Sure, there is some reason for this checking and we would like to know which is and in what does it affect. How could we fix it? The following is the asterisk console output when the ca...

Hello, My configuration is : Card 0 - kernel dummy sound card Card 1 - my soundcard I have a jackd running in background. My jackd launch command is : jackd --port-max 16 --realtime --no-mlock -d alsa --playback hw:1,0 --capture hw:1,0 --rate 8000 --period 1024 --shorts --inchannels 2 --outchannels 2 --dither triangular & 1 ) I open asterisk with chan_alsa.so connected (with asoundrc) to

List, Does anyone know how to limit the amount of registrations that a sip user can have? For example, I have 2 softphones that I use on my laptop & desktop, both use the same username & password. If I have both softphones up at the same time, I can make simultaneous calls with each of them. I know you can have call-limit=1 but in this case, I want to allow them to have 3 way calling

...re we are) where [AgentQ] is called by the queue command to a member added by addqueuemember(Local/99@AgentQ) why don't I get to the NoOp if the agent hangs up during the announcement message (to the agent) ? I see in the app_dial.c program that the "g" flag is tested thus: if ((ast_test_flag(peerflags, OPT_GO_ON)) && (!chan->_softhangup) && (res != AST_PBX_KEEPALIVE)) res = 0; So this would indicate that if all three of these conditions are met then res would be set to 0, and things would behave how I want. In chan_agent.c, the following line is where the agent...

Hi List, I am currently using the H parameter in the dial command. The issue that I am having is that if the user is calling an ivr that requires him to press the * key then the call gets hung up on. How would I go about changing it so that the user will have to press say ** for the H parameter to come in to effect ? Thanks a lot. Dovid -------------- next part -------------- An HTML attachment

I have read about that on asterisk 1.6, there will be a parameter "o" (Only listen to audio coming from this channel), I have tried, but I still get inbound and outbound audio from the spied channel. Has anyone used this feature? Is it working? Is there any work-around? I will like to only spy the outbound audio from a channel, I dont want to hear the incomming audio of that channel. I

Hi everyone, I am trying to figure out the behavior of trustrpid Basically its not behaving the way I expected it to or maybe I am missing a configuration option or something else. When a call from a phone is sent to the * box it has the following sip headers: From: "From Phone" <sip:1001 at 10.0.0.29>;tag=4bf4bb4e11e92476. Remote-Party-ID: "Cloutier"

Hi folks, we use meetme application with pin so when a customer joins he's prompted for his name. Then the voice say:"press one to accept the recording..." My question is, is it possible to cut off that request to"press one"? Thanks to all -- .:FaberK:.

I am still trying to figure out a reasonable way to exit the chanspy application in a dialplan. For the most part I understand how things are working and there is one change I would like to propose. The way the 1.4.23.1 code seems to work is that if there are no channels that match the chanprefix argument the chanspy code stays in a loop waiting for a new channel to come into being that matches

...rport parameter out of the via headers. Here's my scenario: UA -> Snom NATf -> Snom 4S Proxy -> Asterisk Echo Test Function NATf, the proxy, and Asterisk are all on public IPs. So my question is: In chan_sip.c, copy_via_headers function, I see an if statement testing for "(ast_test_flag(p, SIP_NAT) == SIP_NAT_ALWAYS)" What in sip.conf do I do to toggle/change SIP_NAT to try to match this if statement? Following is my sip.conf for the proxy. Note I've tried nat=yes, nat=no, nat=always but the darn thing always takes the "else" instead of matching the if....

Hi asterisk-users, We are using Asterisk 1.2.12.1, and are trying to use the Page application. It seems to work but after approx 4-5 seconds the call is hung up. The dialplan code look like this: exten => _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2}) exten => _*2XX,n,GotoIf($[ "${PAGING_DEVICES}" = "invalid" ]?i,1) exten => _*2XX,n,SIPAddHeader(Call-Info:

Can someone help me with upgrading to the lastest version. I am using the same sip.conf file, but the headers have changed and registration fails. Has something change in the conf file that would cause this? Notice 1.2.5 has no Authoization at all... Regards, Jason Version 1.0.9 --------------------------- REGISTER sip:voip-ca35323.ocn.ne.jp SIP/2.0 Via: SIP/2.0/UDP

Hello list, I'm observing what I believe to be inconsistent behaviour regarding "Newstate" AMI events for the "Ringing" state. As such I come to you asking for experience or advice: am I wrong or should I file a bug ? I present you a short introduction which I feel is relevant; however, if you want to go straight to my technical question, please scroll