similar to: Re :Re: Sip module and dns (Alyed)

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 , 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

Hi, I need some help debugging a sip situation. I started to have problems with sip trunks, using more than one trunk (and sometimes using only one) the sip module seems to freeze. My local extensions lost registration and also the trunks. The only way that I can restart the sip is removing the trunks. If I make sip reload or restart asterisk the sip module takes many many time before

=20 Andrew Fenn wrote: > You don't need their program to use justvoip, voipdiscount, etc=2E You > can use any sip client to connect to Betamax servers=2E Try Twinkle=2E >=20 > On Mon, Jul 27, 2009 at 11:24 PM, miroa84<wineforum-user at winehq=2Eorg> wrote: >=20 > > I tried to install justvoip several times and I cannot install it=2E Can somebody tell me how to

I have been encountering a rather hard to debug problem for the last couple of months: * Calls are setup fine. * After a couple of minutes, two way audio becomes one-way and the remote or local party drops out of the call. Setup: * Nokia E71i sip on NAT'd network (multihomed linux box) * Remote asterisk 1.4.21 on Ubuntu on public network * using a Finera/Betamax provider to route calls to

I tried to install justvoip several times and I cannot install it. Can somebody tell me how to install it on ubuntu? Meybe next version of WINE will support it?

Hi, I'm having a problem trying to register sip trunks. I using asterisk 1.4.39.1, freepbx 2.5.2 in centos 5.5 and I'm trying to configure several sip accounts from my provider. The accounts are individually configured as sip trunks. With only one account everything is ok, it registers and I can make and receive calls. My problem is when I try to put more accounts, it seems to start

Lista dos aprovados em concurso Lagoa da Canoa: V?rzea Grande: ANA MARA DE SOUSA PEREIRA, LUANA FERNANDA FERNANDES ANDRADE, FRANCISCO ROGER GARCIA DE ALMEIDA, POLLYANNA DE O, JO?O CARLOS MOREIRA DE CARVALHO, DANIELLA FERNANDES DA SILVA, MARIA JOS? DA SILVA FERREIRA, J?SSICA VENTURA FREIRE. SOLANGE PAULINO CORREIA, BRUNA KETHEY DA SILVA PEIXOTO, LUIS HENRIQUE FREITAS GOMES, HELIO SOARES DE ARAUJO,

externhost is monitoring for ip changes with an interval of externrefresh, so far so good. Wouldnt it be handy if asterisk would do an sip reregister if it detects an ip change? My SIP provider has sometimes very high intervals of 1 hour and if ip changes, the registration doesnt work until it expires or asterisk is restarted or sip reload. Or just everyone uses fixed ip addresses? For now,

Hi all, we do have a callcenter system running with 1.4.21.1 - the agents are connected used sip phones. SIP accounts are configured using realtime (sip buddies) - and are configured with call-limit=1. It is operating just fine - but from time to time it does happen that an agent with an active call (inbound or outbound) does start to get a second call offered. I have taken a look at the

Hi, I've been having a strange problem recently where real-time asterisk will unregister a IAX friend at random times when the registration should not have expired. I have a Zoiper soft phone client (on windows) connecting to asterisk over a LAN (no firewalls). The default reregister time of 60 seconds is used, but the asterisk server unregisters the client (sets regseconds to 0 in the

Hi, im trying to setup ASTCC but I'm getting it difficult. I've correctly set up the mysql database astcc and added a brand, trunk, route and a card as follows: brands +------+----------+------+--------------+------+--------+------+------+ | name | language | inc | publishednum | did | markup | days | fee | +------+----------+------+--------------+------+--------+------+------+ | FWD

Adding back the nouveau list... On Sun, Feb 9, 2014 at 8:28 AM, Attila T?th <tothsoft at gmail.com> wrote: > Linux mint 16 (64bit) > > Repo: Ubuntu X-SWAT <ubuntu-x at lists.ubuntu.com><ubuntu-x at lists.ubuntu.com> > Mesa: 10.2.0~git20140205.44338cd8-0ubuntu0sarvatt~saucy > libdrm: 2.4.52+git20140121.46d451c9-0ubuntu0sarvatt~saucy > libg3dvl-mesa:

El dom, 10-02-2013 a las 17:47 -0300, Jos? Luis Artuch escribi?: > El dom, 10-02-2013 a las 10:07 +0200, R?cker Thomas escribi?: > > > Hi, > > > > On 10/02/13 06:53, Jos? Luis Artuch wrote: > > > > > > > I'm understanding why Icecast2 is an audio server, wow !!! > > > http://www.icecast.org/docs/icecast-2.0.1/icecast2_config_file.html

Hello, I wonder what the "open error -1" / "release open_disk error" messages in sanlock.log actually mean. I saw these messages in the log on a KVM host that rebooted, and after running "/usr/sbin/virt-sanlock-cleanup" on that host. The resources where disks from 2 guests running on another KVM host. So in fact the disks are still in use, bot got cleaned up by

Rela??o de aprovados Mar Vermelho: ?gua Clara: ANA PAULA RODRIGUES DA SILVA, LUCAS ARAUJO GOMES FROTA, GABRIEL VICTOR BARROS FORTE DA SILVA, QUIT?RIA DA SILVA G?IS, JO?O CARLOS MOREIRA DE CARVALHO, DAYANA MARIA DE SOUSA TAVARES, MARIA JULIENE CORDEIRO, JO?O PAULO DA SILVA. TALITA FERNANDES GONCALVES, BRUNO RAMOS FERNANDES, LUIZ HENRIQUE ALVES DAMASCENO, IAGO DA SILVA NOBRE, RITA ANGELA DA SILVA.

Lista dos aprovados em vestibular Junqueiro: Vale de S?o Domingos: ANA L?CIA MENDES DOS SANTOS, LUANA FELIX BIE, FRANCISCO PAULO DE OLIVEIRA MESQUITA, POLLIANA BRASILIANA DE SIQUEIRA, JO?O CARLOS MOREIRA DE CARVALHO, DANIELE SILVA OLIVEIRA, MARIA JOS? BATISTA LEITE, J?SSICA MAYARA P. PAULINO. SOLANGE FERREIRA ANDRADE, BRENA RODRIGUES MACIEL, LUIS FABRICIO DE FREITAS SOUZA, H?LIO BARROS FERREIRA,

Oops! My fault, sorry!! :P El 10/02/2013 03:01, "Jos? Luis Artuch" <artuch at speedy.com.ar> escribi?: > Ah ... thanks Dave, will give light to the subject: > <relay> > <server>127.0.0.1</server> > <port>8001</port> > <mount>/example.ogg</mount> >

Hi, On 10/02/13 06:53, Jos? Luis Artuch wrote: > I'm understanding why Icecast2 is an audio server, wow !!! > http://www.icecast.org/docs/icecast-2.0.1/icecast2_config_file.html > We're not an 'audio server', that's pulse-audio, esd, etc. territory. We're a multimedia streaming server. Slight difference. :) > Testing the relay option: > Two mountpoints

No problem Xabier, so we go, thank you !! El dom, 10-02-2013 a las 11:21 +0100, Xabier Oneca -- xOneca escribi?: > Oops! My fault, sorry!! :P > > > El 10/02/2013 03:01, "Jos? Luis Artuch" <artuch at speedy.com.ar> > escribi?: > > Ah ... thanks Dave, will give light to the subject: > <relay> >