similar to: RemoveQueueMember and realtime queues

Hello list, I try to pick up a ringing extension but nothing works. To be clear, I'm trying to pick up extension 10. [Jun 14 17:37:34] -- Executing [**10 at from-TESTCORP:4] Pickup("SIP/testcorp3-00000041", "10 at 123456") in new stack [Jun 14 17:37:34] NOTICE[16555]: app_directed_pickup.c:159 pickup_exec: No target channel found for 10. [Jun 14 17:37:34] --

Hello, this is a test to add a channel to multiple GROUPs. this is my dialplan : exten => s,n,NoOp(groepcount = ${GROUP_COUNT(40)}) exten => s,n,Set(GROUP(40)=40) exten => s,n,NoOp(This channel is member of : ${GROUP_LIST()}) exten => s,n,NoOp(groepcount = ${GROUP_COUNT(40)}) exten => s,n,NoOp(groepcount = ${GROUP_COUNT(40L)}) exten => s,n,Set(GROUP(40)=40L) exten =>

Hello list, I have the following problem with MySQL-queries : it seems that the resultid and connid are not cleared ! [macro-GetMailboxFromSIPuserID] exten => s,1,MYSQL(Connect connid localhost xxx xxx xxx) exten => s,n,MYSQL(Query resultid ${connid} SELECT\ extensie FROM\ tbl_SIPaccounts\ WHERE\ ID="${ARG1}") exten => s,n,MYSQL(Fetch fetchid ${resultid} extensie) exten

Hello list, why is it that GoToIfTime thinks a date of **|*|29-*|jun *is not valid ?? [Jun 29 14:06:34] -- Executing [s at macro-vac:10] *GotoIfTime*("SIP/testcorp-00000036", "**|*|29-*|jun*?onvac") in new stack [Jun 29 14:06:34] WARNING[3076]: pbx.c:4127 get_range: Invalid end day '*', assuming none [Jun 29 14:06:34] -- Executing [s at macro-vac:11]

Can asterisk act as a SIP registrar or location server? I would like to be able for a user agent(client) to register with whatever client they are using as "username@domain-name.com". Rather than the entry/username/password that is setup in the sip.conf file. That way a user could log into any SIP enable client and their calls would follow them around. I have read the sip.conf man pages

Here's a revised patch, plus patches showing how fpmath metadata could be turned on in clang and dragonegg (it seemed safest for the moment to condition on -ffast-math rather than on one of the flags implied by -ffast-math). Major changes: - The FPMathOperator class can no longer be used to change math settings, only to read them. Currently it can be queried for accuracy info. I split the

Hi, I'm running Asterisk 1.8.11.1 @office. The Broadvoice service work fine with all 1.6 version and early 1.8 behind a NAT but about 2 months ago stop working. No made changes in the firewall NAT rules. Right now I'm @home via my Xlite softphone working fine without problems Any suggestions or thoughts? Alex Celi This is the info central*CLI> sip show peers Name/username

I have about 20+ phones on a server, all set for registry expiry 1 min. But only this one, with 2 accounts, keeps re-registerting itself. All the time this is what I see on asterisk CLI and it is kind of annoying. What only this phone does this and no other. Its on a remote location. All phones are Grandstream GXP-2000. -- Registered SIP '502' at 64.101.221.250 port 18639 expires 60

Hello everyone, I've suffering cut offs after 6 or 7 seconds a call is answered, incoming calls are working fine, but outgoing ones show the gollowing messages when are being dropped: [2013-05-15 12:55:14] WARNING[3569]: chan_sip.c:3641 retrans_pkt: Retransmission timeout reached on transmission ZjJkZjlkZWMyZTE4ZmY2NWZlZTExNDM1MDRhMTY4MTc. for seqno 2 (Critical Response) -- See

http://bugs.freedesktop.org/show_bug.cgi?id=17359 Summary: NV50 XV crash often Product: xorg Version: unspecified Platform: x86-64 (AMD64) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau AssignedTo: nouveau at lists.freedesktop.org ReportedBy:

Just upgraded to 1.8, we use the multi lot parking feature by dialling *4. We are not getting the parking slot announcement being played to the person who parks the call, so it's impossible to tell which slot they've gone into. Could someone check our config? On Debian Squeeze using packages from http://packages.asterisk.org/debsqueeze main (Asterisk 1.8.11.1-1digium1~squeeze)

Asterisk Project Security Advisory - AST-2012-006 Product Asterisk Summary Remote Crash Vulnerability in SIP Channel Driver Nature of Advisory Remote Crash Susceptibility Remote Authenticated Sessions Severity

Hi folks, Following an upgrade to Debian wheezy, I'm now running Asterisk 1.8.13.1. As opposed to Asterisk 1.6.2.9 that I ran with squeeze, this version can support IPv6. However, it seems that I can't get it to support both IPv4 and IPv6 at the same time. For example, if in sip.conf I set the bindaddr variable to '::' it will only listen on IPv6 and none of my IPv4-only

In testing Asterisk 11, I've found that Asterisk doesn't seem to be sending BLF updates to SIP peers that have subscribed to a hint looking at a Custom device if that Custom device state is RINGING or RING_INUSE. All other states seem to be working correctly. The hint section of the dialplan is: [hints] exten => _3XX,hint,Custom:${EXTEN} Console shows the following for core show

Dear Alexander Morozov, From the output of the command "ps -ef | grep wine", we can observe that the native Windows USB driver ICDUSB2.sys for the Sony IC Recorder gets loaded with winedevice.exe. enming 18814 18781 0 16:15 ? 00:00:00 C:\windows\system32\winedevice.exe ICDUSB2 However, Digital Voice Editor 3 (under Wine) simply would not detect the Sony IC Recorder. The

The attached patch is a first attempt at representing "-ffast-math" at the IR level, in fact on individual floating point instructions (fadd, fsub etc). It is done using metadata. We already have a "fpmath" metadata type which can be used to signal that reduced precision is OK for a floating point operation, eg %z = fmul float %x, %y, !fpmath !0 ... !0 = metadata