similar to: Can't compile Asterisk on Ubuntu 16

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:

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

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

Hello I am experiencing a freeze of the Asterisk proces when issuing a 'sip reload'. I have this issue every time on asterisk versions : 13.11.2, 13.11.1, 13.10.0 and certified-13.8-cert3. I do not have this on versions certified-13.8-cert2, certified-13.8-cert1 and asterisk 1.8.32.3. The only solution is a cold restart of Asterisk. I can execute any command on CLI except 'sip

Hi I'm trying to use Asterisk running as non-root user and selinux enabled. Asterisk is running ok, but astdb not works. When i try to put in astdb, console shows this message: WARNING[1853]: db.c:350 ast_db_put: Couldn't execute statment: SQL logic error or missing database CentOS 7.5.1804 Asterisk certified/13.21-cert3 [root at sv03 asterisk]# ls -lahZ /var/lib/asterisk/astdb.sqlite3

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:

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

All; I'm running Asterisk 1.8.15-cert3 with the newest version of spandsp. I've even tried unloading that and using Digium's FFA module but I receive the same error on an outbound transmission: [2014-02-06 14:35:14] ERROR[19066]: udptl.c:294 encode_open_type: UDPTL (SIP/XXXXXXXXXXX_outbound-00000000): Buffer overflow detected (59 + 127 > 175) I only get this with one

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

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, Currently I'm experiencing crashes on Asterisk more recently, see messages below (crashed reason: segfault signal 6). abrt-hook-ccpp[19864]: Process 7082 (asterisk) of user 0 killed by SIGABRT - dumping core asterisk: ERROR[15373][C-0004e304]: astobj2.c:131 in INTERNAL_OBJ: FRACK!, Failed assertion bad magic number 0x0 for object 0x7fbd2c 00d170 (0) After running the backtrace for the

I'd like multiple people to be able to dial in and listen to various live radio streams. I was told that the correct resource-friendly way would be to setup a MoH class, and then select that from the dialplan. This works well, but how do I switch between streams? Someone correct me if I'm wrong, but from previous similar questions a few years ago it seems like once you've entered a

Hi guys, I'm having a really strange problem, which I'm pretty sure has only appeared since my last upgrade (1.2.12.1) . It's about NAT and Qualify. I'm using Asterisk to register with some external SIP providers. However, they're always marked as UNREACHABLE, when they weren't before! A typical debug looks like this: hera*CLI> sip reload Reloading

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

Asterisk Project Security Advisory - AST-2017-001 Product Asterisk Summary Buffer overflow in CDR's set user Nature of Advisory Buffer Overflow Susceptibility Remote Authenticated Sessions Severity

Hi all, I'm new to VoIP, now we have a project that needs a PBX with client APPs. In our team we have argument for choosing PBX. By so far, we have following candidates: A: Open source 1) Asterisk PBX (http://www.asterisk.org) (with longest history that almost every one knows it, now the last version using the PJSIP stack) 2) FreeSwitch (http://www.freeswitch.org) (A lot people

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?

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert3. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert3. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert3 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following