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

>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

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

=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 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

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:

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

Hi all my asterisk server, 2 sip client softphones are the same LAN asterisk ip address : 192.168.1.5 sip client 1 : 192.168.1.4 sip client 2 : 192.168.1.2 asterisk starts ok with sip setup the sip.conf [test] type=friend username=test secret=1000 host=dynamic context=cucku directmedia=yes directrtpsetup=yes [1000] type=friend username=1000 secret=1000 host=dynamic context=cucku

Hi! I'm trying to communicate a Cisco CCM 4.0 with Asterisk 1.2.11, I 've followed the info in http://www.voip-info.org/tiki-index.php?page=Asterisk+Cisco+CallManager+Integration but still not able to make Asterisk communicate with Cisco. I keep on receiving --- SIP/2.0 400 Bad Request - 'Malformed/Missing URL' --- and --- SIP/2.0 404 Not Found --- messages

Hello, I have a problem with movistar in Mexico with a sip calls. Movistar send to me T38 and G729 in the INVITE and they say that I have to ignore T38 and use G729 in the voice call. When a fax call is made Movistar send only T38 in the INVITE. Invite example: v=0 o=GDL-BMSW-12D 19913379 19899826 IN IP4 192.168.1.2 s=sip call c=IN IP4 192.168.1.2 t=0 0 m=audio 6370 RTP/AVP 18 101 a=fmtp:18

hi,all when i send a call to other server use SIP trunk, i got this below, <--- SIP read from 222.46.18.52:5060 ---> SIP/2.0 403 Forbidden what's rong with is? > Asterisk 1.4.21.2, Copyright (C) 1999 - 2008 Digium, Inc. and others. > Created by Mark Spencer <markster at digium.com> > Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for

i have a prablom here, i want to send a call from A to B use sip trunk , the call can sended B,but not work to PSTN. the message from B server. help pls,what's rong? > > <--- SIP read from 192.168.0.176:5060 ---> > INVITE sip:15921256331 at 192.168.0.151 <sip%3A15921256331 at 192.168.0.151>SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.176:5060;branch=z9hG4bK51a51b96;rport

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

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,

Hello everyone, I'm fairly new to asterisk and this list. Currently I'm working on IAX trunks to send/receive calls between 2 asterisk boxes with asterisk 1.6.1.1+asterisk gui 2.0. After some work in the gui, two boxes can send/receive calls to/from the other just fine, the only problem I have is the caller id. Here is my setup: 1. on both boxes, I added an IAX user in the gui, say the

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, 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

Dear sir, We have been using asterisk for 4 years. Now we have got problems which occurs during the attended transfer. But we are not always getting this problem. Sometimes it happens. But now we cannot understand why this is happening? problem is:"Failed to play transfer sound! " The log of asterisk is as like as followings: [Mar 25 17:58:40] DEBUG[23168] chan_sip.c: Invalid SIP

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.