similar to: differentiate incoming calls on SIP clients

Displaying 20 results from an estimated 40000 matches similar to: "differentiate incoming calls on SIP clients"

2004 Jan 09
2
asterisk sip with voicemail
Hello all, I have setup my sip.conf so users can register etc in the following format, [person] type=friend username=nick secret=******** host=dynamic mailbox=101 in my voicemail.conf I have an entry like 101 => 1234,Nick Knight,nick@omniis.com Leaving a voicemail works fine after I have my dial command time out but on sip clients which display whether voicemail is
2003 Oct 31
2
asterisk and pingtel
Hello All, I have pingtel and asterisk working really well. I have a really annoying little problem - mainly with pingtel. When a call comes in pingtel displays the caller ID on the phone. If I miss it then I click on the number for redial - this doesn't include a 9 to dial an outside line. The second problem is with the dialer from outlook again it bypasses the outlook dialing rules so
2009 Sep 06
0
[LLVMdev] How to differentiate between external and internal calls in llc?
I have a MachineFunctionPass plugged into llc during LLVMTargetMachine::addPreRegAlloc. In this Pass I need to extend calls (i. e. CALL32m, CALL32r) iff they call function within the program. CALL32m has, I think, ten different possibilities for the four operands giving the target address. At the moment I have excluded calls that give the displacement as GlobalAddress or JumpTableIndex
2013 Oct 27
2
Blocking certain hostnames/clients
Hello, As a result of learning of the new 'Intro' App introduced by LinkedIn, and discussing how to block SMTP access to my postfix server from these clients, I'm now interested in doing the same for dovecot. Bottom line desire is to avoid scraping/hijacking email stored on my dovecot server by any client other than a users client. This includes Intro (so, LinkedIn), Blackberry,
2016 Jan 22
4
[Bug 2530] New: Client does not differentiate between more keys on Smart card, signs always with first one
https://bugzilla.mindrot.org/show_bug.cgi?id=2530 Bug ID: 2530 Summary: Client does not differentiate between more keys on Smart card, signs always with first one Product: Portable OpenSSH Version: 7.1p1 Hardware: Other OS: Linux Status: NEW Severity: enhancement Priority: P5
2014 Jan 13
3
[LLVMdev] How to differentiate standard libc calls from intrinsics
Hi, My pass scans call instructions for standard C library calls. For some libc functions, however, LLVM uses intrinsics instead. For example, I see that my memcpy calls are replaced by the llvm.memcpy.* intrinsics. This is not a problem because I can simply look for llvm.memcpy calls when scanning for memcpy calls. The problem arises when LLVM implicitly inserts llvm.memcpy intrinsics into my
2019 Apr 22
2
Incoming SIP call, outgoing SIP registration. PJSIP.
Hi, Thank for your answer. 22.04.2019 23:47, Joshua C. Colp пишет: > On Mon, Apr 22, 2019, at 1:43 PM, Pavel wrote: >> Hi, >> >> Got problems with incoming SIP calls. >> >> Scenario: >> >> Server1: 3cx or any other server >> >> Server2: Asterisk 16.2.1 . PJPROJECT 2.8 >> >> Server2 registers on Server1 with SIP ID 1121.
2003 Jul 24
2
Voicemail() problems - Long pause after incoming message recording ended.
I'm having the following problem: I call into my Asterisk box (RedHat Linux 9.0, 1 Digium X100P card) to access voicemail. After dialing the appropriate extension I get voicemail, am presented with the user's unavailable message, and can leave a message normally. The problem comes when I press "#" to end the recording, at which point I am told "Your message has been
2009 Jun 10
0
sip calls not going through
Hello, i've recently configured my asterisk for internal sip calls. while testing, i noticed that 1 out of 10 calls works.. at first i thought my router dropping packets around the way as it were a bottle neck.. so i've added a switch. once i tested again same prob occurs... im using xlite as a softphone on clients pc and centos server on a dedicated machine. at times the phone call
2019 Apr 22
2
Incoming SIP call, outgoing SIP registration. PJSIP.
Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple
2006 May 29
2
Asterisk Internal sip calls I can´t send/recive
When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call
2007 May 15
1
differentiate groups on barplot
To differentiate between groups on the barplot, I guessed that col = colr[test$group] would have worked. How can I do this? Many Thanks Murray test <- structure(list(patient = 1:20, score = c(100, 95, 80, 75, 64, 43, 42, 40, 37, 35, 30, 29, 27, 26, 23, 22, 19, 18, 17, 16), group = c(1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 1, 1, 0, 1, 0, 1, 0, 1, 0)), .Names = c("patient",
2008 Jun 10
1
Delaying SIP disconnect after incoming call hangs up?
I'm looking for a way to delay the disconnection of a call to a SIP extension (or pad it with silence) for a few seconds, after an incoming call to that extension hangs up. Rationale: I have an Asterisk PBX (current 1.4.20 codebase), with a Leadtek BVP8051S ATA hooked to an analog phone which has a built-in answering machine. Incoming SIP connections to the appropriate extension are dialed
2006 Oct 30
0
Is there a way to differentiate PV-on-HVM devices from qemu-dm devices?
Hi folk, I''m recently doing some work on VNIF driver, however, when I enabled both qemu-dm devices and pv devices, the pv driver complains. It seems two kinds of devices have very similar entries in xenstore, which misled the driver. I remembered in the past, there is a `type:ioemu'' entry in xenstore to represent dm devices, but it was removed. There used to be a
2008 Mar 03
0
Fwd: R: Studdy Missing Data, differentiate between a percent with in the valid answers and with in the different missing answers
Begin forwarded message: > Subject: R: Studdy Missing Data, differentiate between a percent > with in the valid answers and with in the different missing answers > take a look at the Website from Dirk Enzmann: > http://www2.jura.uni-hamburg.de/instkrim/kriminologie/Mitarbeiter/ > Enzmann/Software/Enzmann_Software.html > He provides a function called "freq.r" - maybe
2020 Feb 17
3
Differentiate array access at IR level
Hi LLVM community, I am trying to differentiate access to different array elements, for example: for (int i = 1; i < 10; i++) { a[i] = a[i] + 10; b[i] = a[i - 1] * 2; } If it is possible to tell it loads/stores 3 different array elements: a[i], b[i] and a[i - 1] at IR level? Thanks for your time in advance! Best, Michael -------------- next part -------------- An HTML
2008 Apr 25
2
Differentiate alphanumeric vs numeric strings
I have a bunch of tables in a Microsoft Access database. An updated database is sent to me every week containing a new table. I know that is inefficient and weird but welcome to my life. I want to read the tables whose names are something such as "040207" but not the ones that have alphanumeric names such as "everyone". Using RODBC I am easily able to create a character vector
2020 Aug 13
0
Voicemail: don't play vm-intro if custom intro is recorded.
Hi Gang We migrated our voicemail system from asterisk 13 to 16 a couple of months ago. Right after the migration, we got the complaint that vm-intro is being played when the customer had recorded a own announcement. So I assumed we had replaced that file by a zero lenght one on the previous installation and did the same to suppress that surplus intro. Now I got the opposite complaint: If the
2005 Jul 10
0
How to properly handle incoming SIP and IAX calls, so user can call back and how to properly make outgoing sip/iax calls through Asterisk ?
Hi, I'm aware that incoming and outgoing calls are going fine when isdn channels are involved - caller id properly identifies calling party, so user can call back.... But how to properly handle this for iax, sip calls.... I have few questions : - BTW, what to type for instance in remote firefly to make standalone calls to Asterisk default context or particular extension ? - If I receive
1999 Sep 29
1
getenv() can't differentiate "defined but empty" and "undefined"
getenv(<varname>) currently returns "" if the <varname> is undefined. However, if <varname> is defined but empty, getenv(<varname>) still only returns "". I think this is quite unfortunate but consistent with the prototype. --- I'd propose to change the current behavior. Something which should be pretty back compatible would for the first