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Can someone recommend the best way to view current calls in progress on the Asterisk console? Neither the 'show channels' or 'sip show channels' commands are easy to read. hestia*CLI> show channels Channel Location State Application(Data) SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2) SIP/2944079-e7f2 <mailto:2944093@one_start:2> 2944093@one_start:2 Up Dial(SIP/2944093|36|tr)...

I'm having some issues with Background() and Read() commands. See the example below. This is when I wait for Background to finish playing the sound file, before entering '12345#'. All works fine. hestia*CLI> -- Executing Answer("SIP/2944093-3366", "") in new stack -- Executing Wait("SIP/2944093-3366", "1") in new stack -- Executing BackGround("SIP/2944093-3366", "if-u-know-ext-dial") in new stack -- Playing 'if-u-k...

...ample.com_>doing parameter log file = /var/log/samba/%m.logdoing parameter kerberos method = secrets and keytabdoing parameter client signing = yesdoing parameter client use spnego = yespm_process() returned Yeslp_servicenumber: couldn't find homesNetbios name list:-my_netbios_names[0]="HESTIA"added interface eth0 ip=<client_ipv6> bcast= netmask=ffff:ffff:ffff::added interface eth0 ip=10.0.0.8 bcast=10.0.1.255 netmask=255.255.254.0libnet_Join: libnet_JoinCtx: struct libnet_JoinCtx in: struct libnet_JoinCtx dc_name : NULL machine...

...rver #61-Ubuntu SMP Fri Apr 8 19:44:42 UTC 2011 x86_64 GNU/Linux We have on running virtual root at jupiter:~# virsh list --all Id Name State ---------------------------------- 1 kvmtik.4safety.cz running I run script for create virtual system wrote root at jupiter:~# make-hestia.4safety.cz.sh 2011-04-24 21:13:25,613 ERROR : ubuntu-kvm already exists Traceback (most recent call last): File "/usr/bin/vmbuilder", line 24, in <module> cli.main() File "/usr/lib/python2.6/dist-packages/VMBuilder/contrib/cli.py", line 81, in main raise VMBu...

...g Asterisk in a production environment and we're using > 1.2, not 1.4. I don't have the resources to work with 1.4. > Last time I filed a bug against 1.2 I got told off. > > Here's an example of that cruddy output. > > hestia*CLI> dundi show peer 00:0e:0c:a1:92:4d > Peer: 00:0e:0c:a1:92:4d > Model: Symmetric > Host: xxx.187.142.203 > Dynamic: no > KeyPend: no > Reg: No > In Key: dundikey > Out Key: dundikey >...

...var/log/samba/%m.log > doing parameter kerberos method = secrets and keytab > doing parameter client signing = yes > doing parameter client use spnego = yes > pm_process() returned Yes > lp_servicenumber: couldn't find homes > Netbios name list:- > my_netbios_names[0]="HESTIA" > added interface eth0 ip=<client_ipv6> bcast= netmask=ffff:ffff:ffff:: > added interface eth0 ip=10.0.0.8 bcast=10.0.1.255 netmask=255.255.254.0 > libnet_Join: > libnet_JoinCtx: struct libnet_JoinCtx > in: struct libnet_JoinCtx > dc_name...

I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in extensions.conf changed recently? exten => 5555,1,NoOp(${CALLERID}) hestia*CLI> -- Executing NoOp("SIP/2944093-d24d", ""Cletus the Slaw Jawed Yokel" <2944093>") in new stack == Auto fallthrough, channel 'SIP/2944093-d24d' status is 'UNKNOWN' This does not match... exten => 2944093/5555,1,NoOp(${CALLERID})...

...h number 3254103. I have configured the phone to forward to a new number, 18059999999. Here's my dialplan: exten => 3254103,1,Dial(SIP/3254103,10,tr) exten => 18059999999,1,Dial(SIP/11101553818059999999@proxy2,40,tr) When Asterisk dials 3254103, here's what comes up on the console: hestia*CLI> -- Executing Dial("SIP/2944093-6935", "SIP/3254103|10|tr") in new stack -- Called 3254103 -- Got SIP response 302 "Moved Temporarily" back from xxx.187.128.19 -- Now forwarding SIP/2944093-6935 to 'Local/18059999999@betty_start' (thanks...

...st, NTLMSSP_NEGOTIATE <- Session Setup AndX Response, NTLMSSP_CHALLENGE, Error: STATUS_MORE_PROCESSING_REQUIRED -> Session Setup AndX Request, NTLMSSP_AUTH, User: \ <- Session Setup AndX Response -> Tree Connect AndX Request, Path: \\HESTIA\HESTIAROOT <- Tree Connect AndX Response, Error: STATUS_ACCESS_DENIED .... etc. It's true that User '\' has no access to \\HESTIA\HESTIAROOT. What is obscure is why '\' is involved ! There are a number of questions I have no answer for: 1. the seque...

So I went to System > Administration > Server Settings > Samba and added a folder to share and initially set it to guest access to get things going. I couldn't access this one and only share on this server with guest access from either my Windows laptop (XP Pro SP3) or a Leopard server (10.5.8) so I changed the settings to use user authentication and added my local user details

Ok, this is highly confusing. hestia*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 2944030 2944030 oneeighty_start No RFC3581 2944035 2944035 oneeighty_start No RFC35...

Hello all, I need to write a queue_log parser that is going to implement more or less the functionalities described here http://lists.digium.com/pipermail/asterisk-users/2003-July/014965.html of course not everything from scratch, but this is where I'd like it to go. I am looking for - previous work (maybe it's ready somewhere and I've never heard of it) - suggestions -

...om> doing parameter log file = /var/log/samba/%m.log doing parameter kerberos method = secrets and keytab doing parameter client signing = yes doing parameter client use spnego = yes pm_process() returned Yes lp_servicenumber: couldn't find homes Netbios name list:- my_netbios_names[0]="HESTIA" added interface eth0 ip=<client_ipv6> bcast= netmask=ffff:ffff:ffff:: added interface eth0 ip=10.0.0.8 bcast=10.0.1.255 netmask=255.255.254.0 libnet_Join: libnet_JoinCtx: struct libnet_JoinCtx in: struct libnet_JoinCtx dc_name : NULL...

...iven company go to the SAME box as the phones are registered to. Queues won't work correctly if they're are split between servers. If a phone registered to pbx1 wants to call the queue at 2944000, the call comes into pbx1. If I do a dundi lookup on that number on the console I get this: hestia*CLI> dundi lookup 2944000 @180q <mailto:2944000@180q> 1. 200 IAX2/dundi2:cx8tTjXAoQaPfr9V+C6EWw@xxx.187.142.204/oe_main (EXISTS) from 00:0e:0c:a1:90:82, expires in 0 s 2. 300 IAX2/dundi3:zGFQTD6qixAkSNVpQkviQw@xxx.187.142.232/oe_main (EXISTS) from 00:0e:0c:a1:92:6f, expi...

...k zaptel svn checkout http://svn.digium.com/svn/libpri/trunk libpri Compiled and installed zaptel, libpri, asterisk and finally asterisk-addons. Following errors ocurrs when compiling asterisk-addons. Also tried again after blowing away /usr/lib/asterisk. Is it normal for svn code not to compile? hestia:(pbx1)asterisk-addons # make ./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/local/mysql/include -I/usr/include/mysql `ls *.c` make -C format_mp3 all make[1]: Entering directory `/root/svn/asterisk-addons/format_mp3' gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-de...

...\MAILSLOT\NET\GETDC834 SMB MailSlot Protocol Opcode: Write Mail Slot (1) Priority: 1 Class: Unreliable & Broadcast (2) Size: 63 Mailslot Name: \MAILSLOT\NET\GETDC834 Microsoft Windows Logon Protocol (Old) Command: SAM Response - user unknown (0x15) Server Name: \\HESTIA User Name: Domain Name: RHEA NT Version: 1 LMNT Token: 0xffff (Windows NT Networking) LM20 Token: 0xffff (LanMan 2.0 or higher) -- Chris Hall @ Home +44 (0)7970 277 383 -------------- next part -------------- A non-text attachment was scrubbe...

The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.

I'vce got three Asterisk systems here that I'd like to be able to place calls between with IAX. As usual, I've spent several hours playing with it, really getting nowhere. Asterisk is so mentally draining. Each system, pbx1, pbx2, pbx3, should be able to connect to every other. Do I need separate user/peers or can I combine them into a single user=friend for each system? if I place a