search for: voipfon

...A typical debug looks like this: hera*CLI> sip reload Reloading SIP == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/sip_notify.conf': Found <registration> Reliably Transmitting (no NAT) to 195.189.173.10:5060: OPTIONS sip:sip.voipfone.co.uk SIP/2.0 Via: SIP/2.0/UDP 87.194.194.249:5060;branch=z9hG4bK07c29ff6;rport From: "asterisk" <sip:asterisk@87.194.194.249>;tag=as38a9e906 To: <sip:sip.voipfone.co.uk> Contact: <sip:asterisk@87.194.194.249> Call-ID: 7dd0587b016684785b7bda1e6f1b2478@87.194.194.24...

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 username=3011XXXX To be honest a lot of this is guesswork so could be wrong. I...

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...

...hould 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? Am using asterisk 1.4.16.2 Relevant part of files: sip.conf [voipfone] type=friend secret=xxxxxxxx username=xxxxxx fromuser=xxxxxx fromdomain=sip.voipfone.co.uk host=sip.voipfone.co.uk insecure=very dtmfmode=rfc2833 context=fromvoipfone [s450] type=friend context=phones host=dynamic [xlite] type=friend context=phones host=dynamic [consult] type=friend context=pho...

...even display: "sip show peers" Name/username Host Dyn Nat ACL Port Status 204/204 192.168.xxx.xxx D 2048 Unmonitored 203/203 192.168.xxx.xxx D 2048 Unmonitored "sip show registry" sip.voipfone.co.uk:5060 xxxxxxxx 45 Registered Thu, 27 Nov 2008 11:01:56:03 "sip reload" or restarting asterisk with /etc/init.d/asterisk restart fixes the problem and I get the following output: Name/username Host Dyn Nat ACL Port Status 204/204...

...es at lists.digium.com] On Behalf Of Jonathan H Sent: Tuesday, April 18, 2017 09:10 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't compile Asterisk on Ubuntu 16 Feel free to take a look at https://github.com/lardconcepts/asterisk-digitalocean-voipfone-config/blob/m aster/Asterisk-14-on-Ubuntu.md Ignore the bit about Voipfone and just skip to the "Install Asterisk" bit. I've used this same script with Asterisk 12,13 and 14 on Ubuntu 15,16 and 17 so this should work! Let me know how you get on. On 18 April 2017 at 13:41, Tech Su...

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" scr...

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

...alling 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 08:38:42] WARNING[27575]: rtp.c:1145 ast_rtp_read: RTP Read too short i.e. change from [voipfone_incoming] exten => s,1,Dial(SIP/203,20,t) to [voipfone_incoming] exten => s,1,Dial(SIP/07123123456 at voipfone,20,t) What's wrong?! John

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:

...e "called party" plays back an example piece of "menu" audio while the MoH is playing, but seems to ignore the keypresses. -- Executing [s at streamdemo:1] Answer("Local/s at root-00000002;2", "") in new stack -- Local/s at root-00000002;1 answered PJSIP/voipfone-201-00000002 -- Channel Local/s at root-00000002;1 joined 'simple_bridge' basic-bridge <3ac0c9be-2817-48e7-bcd8-4318eb1f9c2b> -- Channel PJSIP/voipfone-201-00000002 joined 'simple_bridge' basic-bridge <3ac0c9be-2817-48e7-bcd8-4318eb1f9c2b> -- Executing [s at...

...T /sbin/iptables -A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT /sbin/iptables -A INPUT -p tcp ! --syn -m state --state NEW -j DROP /sbin/iptables -A INPUT -f -j DROP /sbin/iptables -A INPUT -p tcp --tcp-flags ALL ALL -j REJECT /sbin/iptables -A INPUT -p tcp --tcp-flags ALL NONE -j DROP # Voipfone /sbin/iptables -A INPUT -p tcp -i $EXIF -m state --state NEW -s 195.189.173.0/24 -j ACCEPT /sbin/iptables -A INPUT -p udp -i $EXIF -m state --state NEW -s 195.189.173.0/24 -j ACCEPT /sbin/iptables -A INPUT -p tcp -i $EXIF -m state --state NEW -s 46.31.225.0/24 -j ACCEPT /sbin/iptables -A INPUT -p...

...ls on the same zap line, nor with internal SIP-->SIP calls, or external SIP-->SIP or SIP-->IAX calls. With full logging enabled, just before the hang up, this is written to the log: Oct 26 18:04:59 DEBUG[10822] chan_sip.c: = No match Their Call ID: 5907c40463e18 ebd6d9c2136105a444a@voipfone.co.uk Their Tag as6e46885a Our tag: as2b18a370 Oct 26 18:04:59 DEBUG[10822] chan_sip.c: = Found Their Call ID: 003094c4-4ea9000 8-37353896-0b0743f4@10.0.1.6 Their Tag 003094c44ea90008745cabf1-6f4660c7 Our tag : as56514c99 Oct 26 18:04:59 DEBUG[10822] chan_sip.c: **** Received CANCEL (14) - Comm...

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

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 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

Thanks Joshua and everyone, Joshua's solution seems a lot simpler and works well. Only one thing now - The reason I named the classes as I did, was so that I could select the class based on callerID plus extension. Unless I've misread it, I'm limited to 9 switchable classes via the "digit=#" option, is that correct? Or is there a clever hack around this? extensions.conf