similar to: OT; What happen with voipuser.org ?

Hi I have been lurking for a while, but now have a small problem or 3. 1) I have my inbound line via sip from VOIPUSER.ORG and have a simple extension selection menu on my * box. Internally the DTMF tones are present, (for xlite and * on same LAN), however calling in via the sip line from a pstn doesn't register any tones in asterisk. I have tried all the different DTMFMODE settings in the

JID Pri S Owner Number Pages Dials TTS Status 58 123 S root 008675533661 0:2 4:12 02:12 No carrier detected Here is the asterisk output: [Mar 28 01:54:00] NOTICE[16753]: chan_iax2.c:6025 update_registry: Restricting registration for peer 'iaxmodem' to 60 seconds (requested 50) -- ast_get_srv: SRV lookup for '_sip._udp.voipuser.org' mapped to host

Hi, I am simulating the sending of fax using sendfax through voip to reach an Asteria server via ZAP/1 ( PSTN phone line ) which then route call to a fax machine at ZAP/2. It seems like I am not able to establish any handshake with the physical fax machine using the sendfax program. Does anyone know why that happens and how to fix it? The scenario will be deployed in remote location in the

Hello, We have a provider which is using Kamailio as front end. Our asterisk 13/chan_sip server has no problem to register and pass/receive calls form this provider. Now we want to move to asterisk 16/pjsip and face problem. Registration is OK but when we pass a call our INVITE never receive answer from the provider. We opened a ticket to their support but in the mean time we want to know

Le 19/01/2020 à 00:31, Joshua C. Colp a écrit : > On Sat, Jan 18, 2020 at 1:14 PM Administrator <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > > Le 17/01/2020 à 11:54, Administrator a écrit : > > > > Le 15/01/2020 à 19:24, Administrator a écrit : > >> Hi all, > >> > >> we face a strange

Hi list, since few days my asterisk says I'm connected to iax2.fwdnet.net but I can't call (even 612 or 613), all my calls finishing with "nobody pickup in 30s" and calls I receive finish the same way (no answer from called party which means my *) I debug iax and saw that call is going through iax2.fwdnet.net My iax show channels also tell me channel is up. If I log inmy

Hi all, we are trying to move our servers from chan_sip to chan_pjsip. At this time no problems with phones, they all register fine and can place calls. But for a trunk we face problem and can't place calls despite the fact that registration is OK. What we get is: [2018-04-16 16:08:33] WARNING[18665]: res_pjsip_outbound_authenticator_digest.c:178

I have a whack of Snom 360's. Occasionally, *some* of them, prompt the user, on the screen, for the registration password. You enter it, everything's OK. Next day, same thing. This is like on 5 or 6 phones out of a lot of 120. It's *always* the same phones. I haven't drilled down to things like firmware rev yet, since I ordered them all as one lot, but I'm wondering if anyone

Hi all, we are running some Asteriks (version 13 on Debian Stretch) as VM/kvm in datacenter and have trunks with other Asterisk (v1.8 11 or 13) instances behind FW. Problem we face is that when we reboot the DC Asterisks, the trunks (SIP or IAX) become alive from DC Asterisks to clients ones but UNAVAILABLE the other way. In clients logs we see Registration for 'XXX at

Here's something that should be easy for RUBY pro's. Here is a script: 1.times do r = $agi.exec('DIAL', SIP/voipuser&Zap/32&Zap/33&Zap/34&Zap/35) r = $agi.get_variable('DIALSTATUS') # $agi.set_variable(' WHOANSWERED

Jerry and Michael, many many thanks for your posts. Erick. On 5/24/06, The VoIP Connection <asterisk-biz@thevoipconnection.com> wrote: > Here are the step by step instructions for setting up a brand new Audiocodes > FXS gateway for use with an Asterisk server: > > Connect the gateway to a network switch and connect a computer to the same > switch. Then configure the IP

Le 17/01/2020 à 11:54, Administrator a écrit : > > Le 15/01/2020 à 19:24, Administrator a écrit : >> Hi all, >> >> we face a strange behavior while connecting an Asterisk16 instance >> with PJSIP to 2 providers: we receive error 401 Unauthorized, both of >> them having Kamailio as front-end. With other providers -we don't >> know if they run

Hi, we had an attack on a server and we don't understand how it was possible, Asterisk 1.4.28/Debian Lenny 5.1 Attacker came from PALTEL, network 188.161.128.0/18 Hacked account had following setup: [111] type=friend username=111 context=from-111 host=11.22.33.44 dtmfmode=auto qualify=yes nat=yes canreinvite=no defaultip=11.22.33.44 port=35060 disallow=all allow=ulaw,alaw call-limit=2

HI; Thanks for your reply. The reason for why I am going through asterisk in such case is just "using asterisk voicemail service" I mean: ATA1 calls ATA2, suppose ATA2 is unreachable or he is not at the office, then the call reroute (my GK is able to reroute calls if the first route is not valid) to atersik for voicemail service. Do you think I can handle it with asterisk native

Le 30/09/2019 à 11:45, Joshua C. Colp a écrit : > On Fri, Sep 27, 2019, at 11:31 AM, Administrator TOOTAI wrote: >> Hi list, >> >> I would like to now what is the sense of such type of entry in security.log >> >> [2019-09-27 15:12:24] SECURITY[26964] res_security_log.c: >>

Hi! Please is it possible for Cisco 5XX phones to use asterisk/FreePBX phone directories, and if so, how? Thanks in advance! On Feb 17, 2013 6:40 PM, <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

dear all, apologizes for this off topic. I would like to inform you that registration and paper submission for the 28th International Workshop on Statistical Modelling (IWSM) to be held in Palermo (Italy) 8-12 July 2013 is now open at http://iwsm2013.unipa.it Register at http://iwsm2013.unipa.it/?cmd=registration and then submit your abstract. Deadlines for Abstract submission is February 4,

Hi;? When connecting via VoipBuster or VoipStunt, "I can hear them but they can't hear me"?. This happens with VoipBuster or Voipstunt. Registration is done correctly. I thought it could be something related to NAT, but I don't have this problem when using VoipUser or Asterisk2PSTN, for example. ? I tried with different codecs: gsm, alaw and ulaw but no change. ? So, now?I

Hi, I face a problem which look like the same as David with his thread "Asterisk not receiving call from VPN address". I had an Asterisk (Elastix) working well in a VM (Debian Wheezy - KVM) having IP 192.168.111.14, my phone network is in the range 192.168.10.x I updated lately to 11.7.0 version and now no one of my phones can register anymore to the asterisk. Ngrep as well as