similar to: Asterisk does not function without a DNS server

> We have our Asterisk server running smoothly with a SIP BRI gateway > for inbound calls. However if the Internet connection goes down and a > DNS server becomes unreachable Asterisk basically does not function. > By this I mean it does not answer call coming in from the gateway > (which is on the local LAN) and you can't even reload it - just hangs > there. If I change the

Hi, I'm getting some strange behaviour on Asterisk 1.4 running on Debian Stable (Lenny). I suspect it's something to do with my setup, rather than a bug, but I'm struggling to see it, and would appreciate any input. Setup: PC with two ethernet cards: eth0 goes to local network, including two SIP phones (Aastra 9112i, wired, and Nokia E75, over WIFI); eth1 goes to router and

Hallo Group! My Name is Guenther Sohler and I registred to this group, because I think asterisk could be interesting for me. I have got a small server at home running linux. It does NAT and a Firewall. There is an intranet with my home PC and a hardware SIP phone. This SIP phone registers at mujtelefon.cz Now I got another account at sipgate.at My idea is following: I want to be reachable at

Hello, I am using a system with an AVM ISDN PCI card (fcpci) and asterisk with chan_capi-cm-0.6. The hardware is connected to a Siemens Hipath 3550 PBX. As a BRI connection has 2 channels and allows 2 simultaneous calls, numbers/MSNs 6391 and 6392 were for provisioned for each channel. The system is working (partly, read on), the trick is the correct cable wiring and setup the PBX's port

Hi , I had some problems in the past with sip trunks, asterisk-users Digest, Vol 68, Issue 4, message 6, and had a reply (message 9) saying that It could be a dns issue. Well today I had a problem again with sip module and it really seams a dns issue. I have an asterisk, version 1.4.26.1, that has 4 bri access and two sip trunks. I'm having internet access problems and when this happens

Hello all, I've a little clue with zaphfc used to connect to a BRI linethat probably can be a configuration issue.... (really I hope so) Here, telcos (expecially mobile operators) use to substitute the dialtone with some vocal indication without answer the line. (Indications like "The customer is not reachable" or "wait because the customer is on the phone" ecc..) For

>Just to check, have you set up >srvlookup=yes > >under the general context in your sip.conf? > >Alyed No, but I put it now but the result is the same. And googleing further https://issues.asterisk.org/view.php?id=3723, it seems that is an old issue... Don't know for witch version is, 1.2?... But is what is happening to me. I'm putting bind in the asterisk server and

Just for the sake of this thread I'll paste part of the last post regarding this issue in the asterisk bug tracker. kpfleming on 2005-03-10 post: "Essentially, what we are saying is that if you are going to use DNS to resolve critical information in your Asterisk configuration, you need to do everything possible to ensure that the DNS lookups will not block for long periods of time.

Hi all We use pci/pci-e BRI cards in our installations. Due to echo problems (that was before Oslec and others), we quickly switched to cards with hardware-based EC. So we use exclusively Digium B410p cards that provide 64ms tail EC. For several reasons we'd like to switch to external BRI gateways like the Patton smartnodes (the price is getting really close to a B410p). I'm

Does any body know if it is possible to group SIP channels just like it is possible with Zap channels? I have a group of FXO gateways (Sipura 3000's) and I would like to treat them as a group the same as I would Zap channels. Does anyone know if this is this possible?

All, I have an issue with IAX that I can't comprehend. Approximately every eight minutes my servers go unreachable. They stay unreachable for exactly 10ms. I have two servers running IAX and it happens on both servers simultaneously. I have searched the archives and see similar issues, but not the exact same one. I am on the current CVS stable version of *. Also, during IAX calls,

I have added in my zone file; _sip._udp.elmit.com. IN SRV 20 0 5060 vpbx.elmit.com. As I understand it should mean that any sip connection to <anyname>@elmit.com should go to the udp port 5060 at the host vpb.elmit.com. In Asterisk's extensions.conf I have in the context [default] exten => ronald,1,Dial(${PHONE_615},60,tr) exten => ronald,2,Voicemail,u615@office exten =>

Hello, I am getting lots of messages as the ones attached below. Is this a problem anybody can explain. (My internet connection is NOT slow or instable... thus I don't get it.) Maybe does this result from incorrect registration? Cheers, Arik ----- sip.conf ------ [general] qualify=no srvlookup=yes canreinvite=yes register => xxxxx:xxxx@sipgate.de/xxxx [sipgate] type=friend

Trying to read and understand bits and pieces of chan_sip.c I've found these I would like someone to clarify: * srvlookup=yes|no * pedantic * canreinvite=update|yes --"update" seems new Being curious, especially for "srvlookup" functionality... /O

Hi all, For a while now I've been using Asterisk together with HFC-PCI cards (Cologne chipset) for Euro-ISDN BRI support. However, I do not consider this to be the most reliable solution and believe that the most stubborn problems have always been software related. If my clients are willing to spend a bit more money on different hardware, what do you think the best solution would be?

Hi! I want my PBX to be reachable at my ekiga.net account. It seems I am registered: vajna2*CLI> sip show registry Host Username Refresh State Reg.Time ekiga.net:5060 magwas 585 Registered Sat, 13 Nov 2010 13:48:22 However when others try to call magwas at ekiga.net, they find me unavailable. My asterisk

Hi all, We have several sipura 3000's working well for outbound calls, however the issue we have is that when calls are sent to the Sipura with Dial(SIP/${EXTEN:0}@sipura1) the Sipura does a SIP answer immediately and then proceeds with the call "in band" therefore sending dialing sounds back to the caller. Other SIP gateways we have notably the Vegastream and others do not do a SIP

Hello list, I have nat=no and qualify=no in my sip peer definition and still my CLI is flooded with : [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (30ms / 2000ms) [Mar 12 10:17:26] NOTICE[20278]: chan_sip.c:12985 handle_response_peerpoke: Peer 'mysippeer' is now Reachable. (24ms / 2000ms) [Mar 12 10:17:26]

On Tue, Oct 22, 2019 at 5:07 PM Rowland penny via samba < samba at lists.samba.org> wrote: > On 22/10/2019 15:52, Thomas Schweikle wrote: > > > > On Mon, Oct 21, 2019 at 5:03 PM Rowland penny via samba > > <samba at lists.samba.org <mailto:samba at lists.samba.org>> wrote: > > > > On 21/10/2019 15:47, Thomas Schweikle via samba wrote: >

Dear All, Can help to advise if there are any name planing for dns? e.g: I've a domain amtb-m.org should my samba4 server be ad.amtb-m.org? OR should I create another non-reachable internal domain e.g: ad.amtb-m.lan For them? What is the benefit on this or that? Any documentation about that? Thank you.