similar to: My dialplan just stopped working one day

In my test configuration, I have a Budgetone, an Iaxy and two computers running X-Lite. My server has one X100P in it (no line hooked up yet). Currently, I can call from any phone to any phone except on one, when the caller calls me, I can't hear the caller (using an X-Lite) but the caller can hear me. If I call him, everything works fine. If I pick up another phone while two phones are

I'm trying to setup a couple of Cisco 7960's in asterisk. I have asterisk working fine for sip clients, and can call the 7960's just fine, but I can't seem to dial out on them. As soon as I enter the first digit, the phone attempts to dial it without waiting for the rest. I've changed timeout settings, etc but can't seem to get it to work. Any ideas? Asterisk

Guys, Ok - nowhere near as complex as most of the discussions on here ( ex telco engr for 18 years here).. But thought I'd ask for some assistance. Have just set up my first * Pbx - having a play with it and a couple of Cisco 7960 (configured as SIP) phones. The phones are tftp'ing into the server ok, and picking up the configs all ok. Everything _seems_ to be working, but I

Hi list, i've problems with my * server and the 4 phones which are linked to it. i've 2 grandstream bt100 with the firmware upgraded to 101, a wi-fi phone (i don't know its brand) and another ip phone i don't know its brand. with this sip.conf : [general] port = 5060 bindaddr = 192.168.100.229 context = default ;x changed from default to sip localnet = 192.168.100.0/24

Hi all, -------- I have installed a TDM400 with one active FXS port (TDM10B) an connected it to a Siemens Euroset 2015 analogue phone. I have installed some smom IP phones to the network as well and configured them as usual (sip.conf). For configuring the TDM10B I have used FXO signalling in /etc/zaptel.conf and in /etc/asterisk/zapata.conf. I definded the TDM channel and the Snom phones to the

Hi, I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular phone. Asterisk configuration is listed below. When I attempt to place a H.323 call, I receive the following errors: - Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20") in new stack Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path exists

I have successfully configured two servers and I am now trying to connect via iax. When I attempt to call from one ext, 2006(server viop1) to extension 3006 (server voip2) I receive a timeout or "call failed 403 forbidden. The information I am receiving from the console is below. Apr 26 10:53:32 WARNING[311313]: channel.c:1745 ast_request: No channel type registered for 'IAX'

Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI> Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)??? are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering... ________________________________ From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa Sent: Fri 7/1/2005 6:43 PM

An awful response: change tombstoneLifetime : ) When deleting objects they go to recycle bin then to deleted objects then are deleted. This if you have some recycle bin working which is not the case by default I reckon. tombstoneLifetime is the number of days AD has to keep deleted objects before the real deletion. If you use the recycle bin this very same tombstoneLifetime is also used to

Hy Samba users, I've got a problem with an samba/ldap setup. As I set an ACL to a domain group in an windows client, a group mapping entry will be created in the Idmap ou at the ldap server. I discoverd the OpenLDAP logfiles. There, the server sends a search request for the domain group sid to the ldap backend will retreive an entry back: Jan 15 20:19:24 225 slapd[4518]: conn=190 op=24

The system described by https://lists.samba.org/archive/samba/2016-May/199829.html (Invalid data for index DN=@INDEX:OBJECTCLASS:DNSNODE) now appears to perform DNS updates correctly, all systems are 4.2.10-Debian, and we've been able to add a user and a new DC. (Thanks for the help!) Synchronisation between v-ward (the new local DC), and empire isn't entirely working, though. >

I have the following two mapping data frames (r) and (h). I want to fill teh value of r$seid with the value of r$seid where r$cid==h$cid. I can do it with sapply as such: > r$seid = sapply(r$cid, function(cid) h[h$cid==cid,]$seid) Is ther a better (faster) way to do this? > r <- data.frame(seid=NA, cid= c(2181,2221,2222)) > r seid cid 1 NA 2181 2 NA 2221 3 NA

https://bugzilla.mindrot.org/show_bug.cgi?id=2181 Bug ID: 2181 Summary: error message changed for sftp(get command) Product: Portable OpenSSH Version: 6.0p1 Hardware: Other OS: Other Status: NEW Severity: enhancement Priority: P5 Component: sftp Assignee: unassigned-bugs at

Hello all :) While trying to bind to 0.0.0.0 having lo0, eth0 and ippp0 up no matter whether on- or offline, i get this: sshd[4518]: error: getnameinfo failed sshd[4518]: fatal: Cannot bind any address. It seems sshd tries to get reverse-resolved the ippp0's address. Is there a configuraiotn fault on my behalf? Did i misunderstand the way sshd works? Shouldn't sshd rather not rely on

CentOS Errata and Security Advisory 2018:2181 Important Upstream details at : https://access.redhat.com/errata/RHSA-2018:2181 The following updated files have been uploaded and are currently syncing to the mirrors: ( sha256sum Filename ) x86_64: 4d58715b4fb8e1a09e0f18ec69e93edc6e4b0558639d37421e8a7c5bd48ad344 gnupg2-2.0.22-5.el7_5.x86_64.rpm

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Evnin'... Anybody got an idea where those CLI messages come from? [Mar 24 20:30:05] WARNING[4518]: chan_sip.c:12296 handle_response: Remote host can't match request NOTIFY to call '0354c42214142e5d6cb8e05568c59837@10.0.2.2'. Giving up. Interestingly all are caused by local IP used by asterisk-1.4.1 cheers rick

Jan 3 08:05:03 NOTICE[66269]: app_dial.c:1056 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) I end up getting this when I call from 2000 to 2001. 2000, 2002, and 2001 all exist in sip.conf and I connect using them. I have all three setup to use the from-sip context. Any suggestions on what is happening? [from-sip] exten =>