similar to: # -key not to be 'transfer'

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Hello, I read on the wiki : Asterisk 1.8 will allow to read SIP response codes in the dialplan via *${HASH(SIP_CAUSE,<channel-name>)}*. Additionally make sure you're using the destination channel, not the source channel. But when I use this in my dialplan, this 'variable' is empty. Dialplan : exten => h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})}) exten =>

Hello, I have this in my dialplan : exten => s,n,Set(vgLabel=vg(${number}+1)) exten => s,n,GoTo(${vgLabel}) But in stead of vgLabel becoming the SUM of 2 numbers, it is just a string : [Nov 3 16:17:27] -- Executing [s at macro-f:43] Set("SIP/test-00000002", "vgLabel=vg(1+1)") in new stack [Nov 3 16:17:27] -- Executing [s at macro-f:44]

Hello list, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw that a lot of SIP accounts where UNREACHABLE and needed to register again (what they slowly did). These are realtime SIP peers that reside on

Hello list, my CLI is spammed with : [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:38] > doing dnsmgr_lookup for 'ssw6.itsp.tld' [Sep 13 08:31:47] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:48] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep 13 08:31:49] > doing dnsmgr_lookup for 'ssw4.itsp.tld' [Sep

Hello. I'm using asterisk 1.4.30. I've found this patch for app_queue.c : https://issues.asterisk.org/view.php?id=11700 Can I easily implement this by issuing : */wget 'https://issues.asterisk.org/file_download.php?file_id=17192&type=bug' -O - | patch -p0/* ?? Does this mean I have a "patched" asterisk ? (I ask this because some applications require a

Hello, what does this message mean ? [Oct 12 14:03:32] DEBUG[9064] chan_sip.c: Trying to put 'SIP/2.0 401' onto UDP socket destined for public_ip:2049 I find this in my debug log file when "core set debug 25". Is something failing, or is this just informative ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, can I use include-statements in the calendar.conf configuration file ? Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131113/8aaffcd8/attachment.html>

Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH" Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",

Hello, when a call comes in and is answered by colleague A, this colleague A sees the CallerID of the external calling number. When colleague A transfers the call to colleague B, attended or unattended, then colleague B sees the number of colleague A on his screen while talking to the external calling number. I expect here that colleague B would see the external calling number on the screen

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

Hello, I would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Since it is Saturday evening (7PM EST) I am asking this on the list in case someone who knows sees it and can answer. Astricon is in my back yard for the first time, and I could hit you with a rock. I would always like to attend, and spoke at the 2007 Astricon in Phoenix but don't have the idle cycles. Question: Can I just go to Astricon and take the dCAP exam only? In and out? Cost? I

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted

How can I let asterisk immediately dials a trunk when off hook?

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body text="#000000" bgcolor="#ffffff"> <font size="+1">Does anyone have links to the most recent audiocodes

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show <queue_name> I get the following numbers: <queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s So from that data we look at 17s holdtime And assume that is the

Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing "sip set debug peer PROVIDER": Sending to 123.123.123.123 : 5060 (no NAT) ^^^^ That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm