similar to: trunk peer not registering after migrating installation

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 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 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

I have recently moved our asterisk server from our LAN to a Debian Lenny server (Asterisk 1.4.21.2~dfsg-3 ) with a public IP outside our network. Our phones are behind a natted firewall. An ITSP provides a PSTN to SIP termination for incoming calls Public ITSP -->Asterisk server-->Natted firewall-->extension (192.168.1.x) Everything works fine (incoming/outgoing audio etc.) except

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

> I have a problem and I hope someone can help me... > I configured an Asterisk on a VM to serve more accounts and act as a proxy to > other SIP-providers. > > The first account running on my phone works without any problem. > A second account, running on the phone of my wife, is always UNREACHABLE. > I can just see in the log: > > [May 28 21:48:46] NOTICE[3646]:

I'd start by turning on sip debugging in asterisk >sip set debug ip [your_phone_ip] and use tcpdump or wireshark to see what the OS sees tcpdump host [your_phone_ip] and udp port 5060 On 15-05-28 03:58 PM, Luca Bertoncello wrote: > Hi list! > > I have a problem and I hope someone can help me... > I configured an Asterisk on a VM to serve more accounts and act as a proxy

Hi list! I have a problem and I hope someone can help me... I configured an Asterisk on a VM to serve more accounts and act as a proxy to other SIP-providers. The first account running on my phone works without any problem. A second account, running on the phone of my wife, is always UNREACHABLE. I can just see in the log: [May 28 21:48:46] NOTICE[3646]: chan_sip.c:22933 sip_poke_noanswer: Peer

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

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

Specifics samba-3.0.0beta1-1.i386.rpm on RedHat 9 and smbpasswd authentication, the machine is the PDC and security is set for user. The machine account was setup on the fly and it appears in passwd, shadow, and smbpasswd files as it should. However it seemed to take a very long time to join the domain, about 1-1/2 minutes. I can browse the Samba PDC machine and access the shares, etc. After

I have two sites running Asterisk PBX. Normally the inbound calls go through a 3rd (colocated) server and are routed via IAX to the site (the site registers with the main server) I created a macro that tries to ring one location and then another. Each site explicitly Answer() the call even though it will only ring all the sip phones at the relevant location. When fall back is in effect it goes to

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

Hello sip show peers does not mark hosts as NAT even though sip.conf and sip_peers table has nat=yes. spitfire*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status voipuser.org/gdsm 216.127.66.119 N 255.255.255.255 5060 Unmonitored 5560/5560 192.168.4.5 D N A 255.255.255.255 5060

Hello. A Mac client of mine has a problem deleting an empty folder from the root of a Samba 4.12 server share, reporting a permission issue; however, the more I look at it, the more I am convinced it should be able to delete it. smb.conf: > [global] > workgroup=XXXXXXXX > realm=XXXXXXXX.local > interfaces=em0 > hosts allow=192.168.XXX. 10.0.XXX.2

Hello everybody, I am having a problem with realtime SIP peers. On Asterisk 1.8, I had SIP peers for external SIP providers configured in database and additional register lines in sip.conf so they would register. Now I upgraded to Asterisk 11.3.0, partly because of the promised callbackextension feature for realtime peers (https://reviewboard.asterisk.org/r/1717/). Removed the 'register'

When doing a "sip show peers" I might see something like: Name/username Host Dyn Nat ACL Port Status devcentos5x64_to_mmfirepa 192.168.1.177 5060 Unmonitored devcentos5x64_to_bt610tMM 192.168.1.159 5060 Unmonitored devcentos5x64_to_am2mm/de 192.168.1.178 5060

> On 27 December 2018 at 19:13 Subscription <leo1subscr at zudiewiener.com> wrote: > > > I'm trying to move from my exising server to a new site. In preparation > for this I've set up the new server as per the first attachment. > > I've added additional (temporary) setting to the new site as per these > instructions > >

> On 27 Dec 2018, at 19.13, Subscription <leo1subscr at zudiewiener.com> wrote: > > but when I try to do a backup with the following command from the old to the new site > > sudo doveadm -D -o imapc_user=user1 at oldserver -o imapc_password=pw-oldserver backup -R -u user1 at newserver imapc: > Since both of your servers are running dovecot it would be probably better

Hello Everyone - Well, I think I'm getting closer with the asterisk connection. This is my setup and I keep getting this error below in ,my /var/log/asterisk/messages file. I have opened 5060 port on the firewall box. I would this is Warning which I can ignore! But I see the connetcion coming but NO DIAL TONE on mt ata186 box sitting in my 192.168.200.x site! I'm using ATA186(cisco