search for: caller's

...guments.as | 165 ++++++++++++++++++----------------- 14 files changed, 351 insertions(+), 340 deletions(-) New commits: commit 15402b24b998b110dad41d2b1559267d474ba892 Author: Pekka Lampila <pekka.lampila at iki.fi> Date: Thu Nov 15 20:48:40 2007 +0200 Improve arguments caller/callee test somewhat diff --git a/test/trace/arguments-5.swf b/test/trace/arguments-5.swf index aece37e..50d9be0 100644 Binary files a/test/trace/arguments-5.swf and b/test/trace/arguments-5.swf differ diff --git a/test/trace/arguments-5.swf.trace b/test/trace/arguments-5.swf.trace index 02fbeb4.....

...t/test_bins.sh +++ b/test/test_bins.sh @@ -52,8 +52,8 @@ flac --help 1>/dev/null 2>/dev/null || die "ERROR can't find flac executable" run_flac () { if [ x"$FLAC__TEST_WITH_VALGRIND" = xyes ] ; then - echo "valgrind --leak-check=yes --show-reachable=yes --num-callers=100 flac $*" >>test_bins.valgrind.log - valgrind --leak-check=yes --show-reachable=yes --num-callers=100 --log-fd=4 flac $* 4>>test_bins.valgrind.log + echo "valgrind --leak-check=yes --show-reachable=yes --num-callers=50 flac $*" >>test_bins.valgrind.log + valg...

...d capability: G.729{hw} <3> 2:21.278 ThreadID=0x49399b30 RFC2833 Handler created 2:21.278 ThreadID=0x49399b30 H323 Added capability: G.729{hw} <1> 2:21.278 ThreadID=0x49399b30 H323 Created new connection: ip$localhost/23866 2:21.279 H225 Caller:80f1490 H225 Started call thread 2:21.328 H225 Caller:80f1490 H323TCP Started connection: host=192.168.1.80:1720, if=192.168.1.50:10001, handl$ 2:21.328 H225 Caller:80f1490 H225 Sending Setup PDU 2:21.329 H225 Caller:80f1490 H225 Check for Fast start by loc...

Hi list, I am in need of a "special" asterisk conference room with the following constraints: - there is one admin / moderator and several "normal" callers. - the callers must not hear any other caller, only the moderator - the moderator must be able to mute and unmute any caller at any time - the moderator must be able to talk to all callers or to a specific caller. - the modetator must be able to kick off any caller at any time... Any hints on how...

...uot;asm goto with outputs", a long requested feature. We want to continue building our relationship with the Linux community. KSPP is a project to improve security in the Linux kernel, through both kernel changes and compiler features. One compiler feature they want is the ability to zero out caller-saved registers on function return as a defense against stale register contents being used as a side-channel or speculation path. The option will be "opt-in" for each target. Targets that don't support the flag should probably emit a warning or error. Our proposal for the feature is...

Hi. I'm using /var/spool/asterisk/outgoing files to place automatic calls, but I'm having trouble setting the Caller ID for the second half of the call. In other words, when we call the first number, we want the Caller ID set to our number, but then when we connect them to the second number, we want _their_ number to be the Caller ID. I've tried the following (and various approximations): Channel: Local/16...

Hi all, I have the following setup PSTN -> ASTERISK -> IVR (using dialogic card) 1) Caller id information is presented to asterisk during the first and second ring. 2) Hence, Asterisk waits for 2 rings before pickup the call and forwarding to the appropriate FXS port. 3) The IVR application also waits for 2 rings before picking up the call to get the caller id. 4) Hence any caller cal...

...All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN Line" tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes "Display Name", "User ID" and any "PSTN CID Number Prefix" I have configured. I have adjusted "PSTN Ring Thru Delay" to 10 as I realise caller ID is not presented until the second ring in oz. I have also verified that caller ID is e...

# ./testcall testcall.conf Chan 1, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025860' to '013331339767' Chan 2, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025861' to '013331339768' Chan 3, class 'mfcr2', variant 'ar,10,4', end 1, caller 0, from '30025862' to '013331339769...

...1:21:34.937 ThreadID=0x06f2bbb0 rfc2833.cxx(81) RFC2833 Handler created 1:21:34.971 ThreadID=0x06f2bbb0 h323ep.cxx(1393) H323 Created new connection: ip$localhost/8176 1:21:34.971 ThreadID=0x06f2bbb0 tlibthrd.cxx(688) PWLib Created thread 0x834f9d0 H225 Caller:%0x -- Called 192.168.1.107 H225 Caller:834f9d0 tlibthrd.cxx(1199) PWLib Started thread 0x834f9d0 H225 Caller:834f9d0 1:21:34.978 H225 Caller:834f9d0 h323ep.cxx(712) H225 Started call thread 1:21:34.984 H225 Caller:834f9d0 transports.cxx(1578) H323TCP Conne...

Nicol?s Gudi?o <asternic@gmail.com> wrote: >>> Paul, 1.0.5 stable suffers from caller id issues as well, at least for >>> SIP channels. What fixed things for me was swapping in app_dial.c from >>> 1.0.2 stable (didn't try others). You could also just diff app_dial.c >>> between versions to find the problem but I took the lazy way out the >>> f...

I am trying to implement a change to our Dialplan that will thwart tele-spammers that are calling us with blanked out caller ID. The caller IDs seem to vary between originating callers when they block caller ID. I've seen the following: "anonymous" "" So I'm checking for these. However recently one company seems to be bypassing this, so what I wanted to do was implement some logic that...

Hello, I am trying to use * to handle anonymous ISDN callers. Something like exten => 5150/0,1,Congestion should work but doesn't. Apparently because the ISDN CAPI doesn't use 0 for callers who don't send their number. Is there a way to make * identify ISDN callers who use CLIR? -Walter -- Walter Doerr =*= wd@infodn.rmi.de =*...

Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password of caller. One possible solution to this is for the agent...

Hi, I would like to use the from caller id, to allow calls yes or no. 101, and 111 should be allowed to use the Trunk, the rest of the phones are not. Is this even possible? So if the from caller id is 101 or 111, then allow the call, otherwise hangup. Thanks, Peter -------------- next part -------------- An HTML attachment was scrubbe...

...>> This is exactly what I do - “press 1 for a human” >> Works great > I do this as well, but I also do a database lookup to see if the number > is on our speeddial list and if so, pass the call directly on without > the IVR prompts. I do something similar for calls without caller ID, but I was still getting robocalls with spoofed caller ID. I have now changed the dialplan slightly so that the first time people call they are asked to dial 1. After the first call, they are added to a known caller list and get straight through, and any robocalls at that point are blacklisted m...

I need some advice on "forwarding" arguments to a callee. Suppose I have a function F that is called at the beginning of all other functions in the module. From F I need to access (read) the arguments passed to its immediate caller. Right now I do something like boxing all arguments in the caller inside a struct and passing a pointer to the struct to F, alongside an identifier telling which caller F is being called from. F has then a giant switch that branches to the appropriate unboxing code. This is obviously suboptimal...

...======== Init TPM BACK ================ Thread "tpmback-listener": pointer: 0x20010043f0, stack: 0xf0000 ============= Init TPM TIS Driver ============== IOMEM Machine Base Address: FED40000 Enabled Localities: 0 Map 1 (fed40, ...) at 0x1006000 failed: -1. Do_exit called! base is 0x10fcb8 caller is 0x1f24d base is 0x10fcd8 caller is 0x27658 base is 0x10fd88 caller is 0x2772b base is 0x10fde8 caller is 0x26bf6 base is 0x10fe28 caller is 0x26c1e base is 0x10fe38 caller is 0x1ba94 base is 0x10fe78 caller is 0x6f84 base is 0x10ff38 caller is 0x353c base is 0x10ff68 caller is 0x1fa80 base is 0x...

Hello All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

I am an Asterisk newby, and I cannot seem to get Caller ID information from our T1 line. When calls appear at the phones, they say the call came from "asterisk" and unknown number. I know how Caller ID information is passed on an analog phone line (between the rings) but with a T1 line, I don't know technically how it is done. I do...