similar to: fallthrough extension.

hi all, is there any way to force all sip peers to re-authenticate themselves? thanks, Paradise Dove

I upgraded from 1.0.9 to 1.2.1 My IVR which worked perfectly on 1.0.9, now hangup with no reason (at least I could not find a cause) When this hangup happen, I can read: == Auto fallthrough, channel 'IAX/user-20' status is 'BUSY' This happening also with ZAP channels I'm really disappointed with 1.2.1, what is benefit from upgrade if I must spend couple days to get my system

I was posed this question: A T1 set up for voice carries 24 conversations on a circuit that is 1.544 megabits/second. Right? Well, if you set that T1 up to carry data and run a link between two IP networks over it, how many SIP conversations could it be expected to carry? How about IAX? How would one extend this calculation to varying bandwidth circuits and various VOIP protocols (MGCP,

Hi. I upgraded my main build host and the clang -Werror builds started failing. This is because clang 10's -Wimplicit-fallthrough doesn't understand /* FALLTHROUGH */ but rather requires __attribute__((fallthrough)): clang -Wall -O2 [...] -Wimplicit-fallthrough [...] -D_XOPEN_SOURCE=600 -D_BSD_SOURCE -D_DEFAULT_SOURCE -DHAVE_CONFIG_H -c /openbsd-compat/base64.c

Great! Thanks everyone for the input, I'm going to start splitting up the patch. I'll send out the non-mechanical parts separately. On Wed, Oct 31, 2018 at 10:03 PM Chris Lattner <sabre at nondot.org> wrote: > On Oct 31, 2018, at 2:24 PM, Reid Kleckner <rnk at google.com> wrote: > > Alex Kornienko proposed enabling this warning back in 2012 here: >

Hello, > ready> ERROR: Program used external function 'putchard' which could not be > resolved! > Any idea of what could be wrong? Please make sure you're using C linkage for such functions, due to mangling the name of function being emitted is not "putchard". Something like this: extern "C" void putchard(char c) { ... } Or, just provide a mapping

Replace the existing /* fall through */ comments and its variants with the new pseudo-keyword macro fallthrough[1]. Also, remove unnecessary fall-through markings when it is the case. [1] https://www.kernel.org/doc/html/latest/process/deprecated.html?highlight=fallthrough#implicit-switch-case-fall-through Signed-off-by: Gustavo A. R. Silva <gustavoars at kernel.org> ---

Anton Korobeynikov wrote: > > Hello, > >> ready> ERROR: Program used external function 'putchard' which could not >> be >> resolved! >> Any idea of what could be wrong? > Please make sure you're using C linkage for such functions, due to > mangling the name of > function being emitted is not "putchard". Something like this: >

The persistentmembers=yes is suppose to keep agents in a queue over a restart. It might do this, but it doesn't do much good as even if they all remain in the queue, they are all logged out on a restart. Is there any way to keep the agents that are logged in, logged in across a restart? Thanks, Jon.

Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten => 1637,1,Dial(SIP/100) Exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _NXXXXXXXXX,2,Congestion Exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _1NXXXXXXXXX,2,Congestion These

Hi, All: I have a testcase which produced incorrect result, it's caused by the combination of nsw flag and ExtLdPromotion, I am leaning to say Clang set nsw flag incorrectly, but please let me know if I was wrong. Here is the reduced testcase: long long foo(int *a) { long long c; c = *a * 1405; return c; } Clang emitted the following IR (It is done by EmitMUL() in

hello how do I set up asterisk to play music on hold to callers while it rings my phones? I am using the amp portal to configure the asterisk pbx just to let you all know. thanks hank email: hanksmith4@earthlink.net gmail: hanksmith5@gmail.com msn messenger: hanksmith4@earthlink.net aim: hanksmith5 skype: hanksmith5 -------------- next part -------------- An HTML attachment was scrubbed... URL:

fallthrough to a separate case/default label break; isn't very readable. Convert pseudo-keyword fallthrough; statements to a simple break; when the next label is case or default and the only statement in the next label block is break; Found using: $ grep-2.5.4 -rP --include=*.[ch] -n "fallthrough;(\s*(case\s+\w+|default)\s*:\s*){1,7}break;" * Miscellanea: o Move or coalesce a

Greetings, Running CVS HEAD about 3 weeks old, I have been beating my head trying to get this to work properly.. Or at least figure out what's going on. Maybe I have done things wrong... I have created a 'catch all' extension at the end of our last context where all phones & voicemail extension exist. This catch all is included in all and works quite nicely except when voicemail

Hi all, I have a connect between a siemens hipath & Asterisk system over PRI The connection works perfectly I can call from the Hipath to an Asterisk Extension. I want to allow the hipath extensions to dial out over a SIP trunk on asterisk but I keep getting "The number you have dialed is not in service" In this CLI I'm dialing from hipath extension 9 (prefix for sip trunk)

Hi people, question one i see that asterisk is now in 1.x release. having tried it in the past i want to know if i can use a voice modem as an outgoing line. i know in the past that was not possible/supported so im just asking in case the option is now available. question two im planing to use asterisk as a pure voip solution with sip phones and h323 phones no need for digium/dialogic hardware

Hi All, I'm at the end of my tether here and would really appreciate some help. I'm trying to implement DTMF based pause/resume of call recording. I'm using Asterisk 1.4.22.1. Here's the scenario: The caller (SIP or ISDN, doesn't matter) dials into the asterisk which executes the following code: exten => _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb)

Hi, all. I've got app_rxfax going and nicely receiving a fax, which I then throw to a script, and have it convert it to a PDF and mail it. Works great... a lot of the time. But a fair bit of the time, rxfax throws errors, and extensions.conf seems never to invoke my script. Here are the pertinent lines: exten => _6403,n,rxfax(${FAXFILE}) exten =>

R users, I got a problem to analyze with probit model. What package contains the algorithm to do probit model. Lawrence N.M Kazembe Mathematical Sciences Department Chancellor College University of Malawi P.O. Box 280 Zomba Malawi Tel: (265) 524 222 ext 284 Fax: (265) 524 046 e-mail: lkazembe at chirunga.sdnp.org.mw url: kazembe.cjb.net kazembe.tsx.org

Hi All, When I tried to write my dial plan as below for my FXO port, which connects one PSTN line: [from-pstn] exten =>s,1,Answer() exten =>s,n,Wait(1) exten =>_X.,1,Dial(DAHDI/1) exten =>_X.,n,Hangup I got the following message: Connected to Asterisk 1.6.2.13 currently running on fax (pid = 8154) Verbosity was 0 and is now 4 -- Starting simple switch on 'DAHDI/1-1'