similar to: SIP with Qualify and NAT

I have an odd problem. I have just installed asterisk on an ubuntu box, and migrated the previous configuration of asterisk (on another ubuntu box) to this new server (scp -pr xxx at oldserver:/etc/asterisk/* /etc/asterisk/) Asterisk worked fine on the old server, but on this server my SIP trunk peer does not login after initial server startup. "sip show peers" shows my phones

Dear all I've got call queuing working when calls originate from my local site. After testing I migrated it to calls originating from our voip provider- it should ring an extension, then queue . All works well apart from if the caller hangs up when queued: the call hangs around in the queue as a phantom until one of the extensions answers it and it is destroyed Am I doing something wrong?

Hi, I am getting this problem when trying to register with Voipfone.co.uk. It used to work, and I haven't changed anything that I know of. Jan 13 10:22:37 WARNING[21645]: acl.c:213 ast_get_ip_or_srv: Unable to lookup 'voipfone.co.uk.voipfone.co.uk' Why does the domain name appear twice? I don't know when it stopped working. In SIP.CONF [sip_proxy-out] type=peer

Hi all, I have recently installed Asterisk@home and outbound calling is working great. However I am strugglings with inbound calls. I have setup a trunk for my provider, voipfone and in the inbound area on AMP I have the following :- user context name = 3011XXXX context=from-pstn dtmfmode=rfc2283 fromdomain=voipfone.co.uk host=voipfone.co.uk insecure=very secret=XXXXXX type=user user=3011XXXX

We're trying to forward an incoming SIP call from voipfone (UK ITSP) that originated from a UK landline back up a SIP trunk to the same ITSP and on to another UK landline number. UK Landline->voipfone->asterisk 1.4->voipfone->UK landline About 1 in 3 times the call at the final landline is silent and we see "RTP Read too short" scrolling on the console log. Where do we

Hey; Thank you very much. I was able to install asterisk from your link. One other question. How are you starting asterisk? Do you use an init script or systemd? Do you think that you could share the script you use? Thanks Again; John V. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H Sent:

Hi If I have an incoming call coming down a SIP trunk to a particular internal SIP extension- I can answer the extension fine, all works well However, if I change extension.conf from dialling the internal extension to forward the call to an external cell phone (up the same trunk as the incoming leg of the call) I cannot get any audio and get the following error message on the console: [Jan 30

Hi there; I didn't see any "G" option in the example above, and the usage for the option parameters is entirely undocumented at https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial The G options are as below G - If the call is answered, transfer the calling party to the specified priority and the called party to the specified priority plus one. context exten

Hi List, I'm having a bit of an odd problem with asterisk and outgoing zap calls. Tzafrir has been kind enough to help me get the logging sorted out so I have some idea of what's going wrong, but I'm a little flummoxed. Essentially the symptoms are as follows; Make a SIP call from Cisco 7960 or 7940 to asterisk, where it is routed out on a ZAP (x100p) line. After

My firewall and asterisk pjsip config only has "permit" options for my ITSP's (SIP trunk) IPs. Here's the script that sets it up. -------------------------------------------------- #!/bin/bash EXIF="eth0" /sbin/iptables --flush /sbin/iptables --policy INPUT DROP /sbin/iptables --policy OUTPUT ACCEPT /sbin/iptables -A INPUT -i lo -j ACCEPT /sbin/iptables -A INPUT -m

On Wed, 8 Feb 2017 15:59:16 -0600 Paul Klapperich via samba <samba at lists.samba.org> wrote: > I have a FreeNAS 9.3 server running Samba Version 4.3.6 and a bunch of > Windows and Linux clients. Everything's been running fine for a while > and nothing changed on the server. > > Recently (Jan 27th) some of the Archlinux clients updated from a 4.8.x > kernel to a 4.9.x

here is the debug from the CLI. I think I know where the problem is I just can figure out how to fix it. The IP in the From and To i think is where the problem is. When I make an outbound call. i get the message "the call cannot be completed as dialed". if i call another ext it works. I posted the debug for both calls. ==============outbound call=========================== <---

I have a FreeNAS 9.3 server running Samba Version 4.3.6 and a bunch of Windows and Linux clients. Everything's been running fine for a while and nothing changed on the server. Recently (Jan 27th) some of the Archlinux clients updated from a 4.8.x kernel to a 4.9.x kernel. Again, things ran fine. Then on Jan 30th around 2am the Archlinux clients using 4.9.x kernels and utilizing mount.cifs to

Very well. Here is the affected smb.conf. ------ [global] server min protocol = NT1 server max protocol = SMB3 interfaces = 127.0.0.1 10.0.0.8 bind interfaces only = yes encrypt passwords = yes dns proxy = no strict locking = no oplocks = yes deadtime = 15 max log size = 51200 max open files = 2830016 logging = file load printers = no

All; I am trying to build and install certified Asterisk 13.13 cert3 on a Ubuntu 16.04.2 LTS host without much success. I am getting the following errors when I try to compile. [CC] res_pjsip/config_transport.c -> res_pjsip/config_transport.o res_pjsip/config_transport.c: In function 'transport_apply': res_pjsip/config_transport.c:572:6: error: 'pjsip_tcp_transport_cfg

Hello, I have a linux bridge setup and am using HTB to bandwidth manage traffic passing through the bridge. I want to manage "some" of the traffic passing through the bridge. I figured I can do this if I dont define a "default" class on the qdisc. I will only config classes for traffic I want to manage. Assume that i Have traffic for network 10.0.0.0/24 passing through the

Hi I''m very very new to tc iproute etc and have read the LARTC howto. What I want to do is create some "master" classes of bandwidth limit and below that per ip address which "inherits" from this master class. Example: one queue for 128Kbps other queue for 256Kbps What I want now is that for example in "class" 128Kbps the ip 10.0.0.5, 10.0.0.8 etc. goes

Hi, I am using Trxibox 2.6 latest ISO install. Following is the output of : "sip show channels" [trixbox ~]# /usr/sbin/asterisk -rx "sip show channels" Peer User/ANR Call ID Seq (Tx/Rx) Format Hold Last Message 212.53.40.40 0218245 6cfb845d050 09011/00000 0x0 (nothing) No 192.168.1.116 (None) YTc4ZmM3NjV 00101/00006 0x0

Try to ping from client to server with its hostname. Sounds like dns problem. ping server Then try to ping its ip address. Then try to add server address to host file. Ex 192.168.8.30 server.example.com[1] server Best M On May 26, 2016 12:02, "Nico Speelman" <nico at speelmanrobben.nl[2]> wrote: Hello, I've been trying to add a new server to my Samba 4 Active directory, but

So if someone has their own hardware and infrastructure but wants a software (not FreePBX but perhaps similar) what options do we have? Would like to virtualize it and not stuck with any one virtualization technology. Discuss... :) Travis -------------- next part -------------- An HTML attachment was scrubbed... URL: