Re: [asterisk-users] SIP trought Firewall
Hi suffered that issue since I started that´s the course oif all of us newbies, noone is willing to help/and even answer, I don't even know if my messages are being read on this list cause not evena "google for it" i've received. I'm now acroos the rive with that problem you're being the victim of an unconfigured sip_nat.conf, in thre you have to specify you public static ip or dynamic domain name, your internal lan like 192.168.1.0/255.255.255.0 and natt=yes restart asterisk and you're problem will be solved, here's my config, fell free to copy/paste it but remember to change the lines to suit you're setup. nat=yes ; key line externip=my.dynamic.hot.tld; This is your addres externrefresh=30 ; some refresh time, still don't know what it does :-P localnet=192.168.1.0/255.255.255.0 ; this is the LAN setup, these are the adress range that you're DHCP NAT device is giving you. qualify=yes ; I hope you're at least sneaked-peaked Asterisk TFOT and know what qualify mean. Try this out and you'll be very happy that people outside your lan will hear you and you will hear them too. Thanks for using Asterisk and though supporting OSS the good people that developed it. On 10/1/07, Emiliano Vazquez <[EMAIL PROTECTED]> wrote: > > Hi to everyone! > > I have succerfully instaled my new Asterisk 1.4 on my debian etch. > > I have my users in sip.conf like this: > > [200] > type=peer > host=dynamic > context=home > secret=200 > callerid= 200 > dtmfmode=rfc2833 > nat=yes > [EMAIL PROTECTED] > disallow=all > allow=ulaw > > I can make calls in my LAN but i can´t ear comunications with another > client > trought Internet. > My Asterisk is in my LAN and i not have a DMZ. I search in the list and > find > something about "rtp" ==> rtp.conf. I found rtpstart and rtpend and > forward > those Ports on my firewall, but this don´t work for me. > > What´s wrong??? > > If you need some info please tell me. > > Thanks in advance! > > Emiliano Vazquez. > > > ___ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- DAVID GONZALEZ H. GNU/Linux Debian+SuSE+RedHat+LFS TECNICO EN REDES NETWORK ADMIN http://www.computrabajo.com.co/cvs/dgonzalezh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk+Sipura 3102+PSTN line
Hello Gurus I've installed my Asterisk server for testing on the company I work the setup or the approach let's call it is: 1 Asterisk Server fully configured and with some SIP extensions setup on two cities A and B. 2. One local PSTN line connected thru a x01p card to call local phone numbers numbres on city A. 3. A Sipura 3102 Gateway on city B connected to a city's B PSTN line. I wnat to be able to call from city A to city B PSTN phone numbers from city A using Internet and vice-versa. What is the proper config on Asterisk and the SPA-3102 so that I can call SIP extension on that device plus PSTN phone lines. Thanks for the tips or the pages/guides I can be referred to. Thanks! :) -- DAVID GONZALEZ H. GNU/Linux Debian+SuSE+RedHat+LFS TECNICO EN REDES NETWORK ADMIN http://www.computrabajo.com.co/cvs/dgonzalezh ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: Routing issue
Hi list
I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk
solutions and appliances.
I installed TrixBox on a litle PC @ home and a x100p card which is
recognized as a Zaptel card, I made some in/outbound routes and they seem to
work but I have a problem with SIP softphones. I created 2 estensions 1000
and 1001 they're both in different cities, when I 1000 (on the same network
as TrixBox) dial 1001 (the other city) they answer and can hear me, but I
don't hear them, and when they call *43 for echo test it plays the "You're
entering echo test..." but when it tries to start echo it just hangs up. and
the log says
-- Executing [EMAIL PROTECTED]:1] Answer("SIP/1000-08939150", "") in
new stack
-- Executing [EMAIL PROTECTED]:2] Wait("SIP/1000-08939150", "1") in new
stack
-- Executing [EMAIL PROTECTED]:3] Playback("SIP/1000-08939150",
"demo-echotest") in new stack
-- Playing 'demo-echotest' (language 'en')
== Spawn extension (from-internal, *43, 3) exited non-zero on
'SIP/1000-08939150'
-- Executing [EMAIL PROTECTED]:1] Macro("SIP/1000-08939150",
"hangupcall") in new stack
-- Executing [EMAIL PROTECTED]:1] ResetCDR("SIP/1000-08939150", "w")
in new stack
-- Executing [EMAIL PROTECTED]:2] NoCDR("SIP/1000-08939150", "") in
new stack
-- Executing [EMAIL PROTECTED]:3] GotoIf("SIP/1000-08939150",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [EMAIL PROTECTED]:6] GotoIf("SIP/1000-08939150",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [EMAIL PROTECTED] :9] GotoIf("SIP/1000-08939150",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [EMAIL PROTECTED]:11] Hangup("SIP/1000-08939150", "") in
new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/1000-08939150' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/1000-08939150'
I'd appreciate your help a lot, I'm not if this is a forewall issue or
something wrong with my asterisk config.
Thanks a lot.
--
DAVID GONZALEZ H.
GNU/Linux Debian+SuSE+RedHat+LFS
TECNICO EN REDES
NETWORK ADMIN
http://www.computrabajo.com.co/cvs/dgonzalezh
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Routing issue
Hi list
I'm kinda new to asterisk and I'm woriking for a company that sells Asterisk
solutions and appliances.
I installed TrixBox on a litle PC @ home and a x100p card which is
recognized as a Zaptel card, I made some in/outbound routes and they seem to
work but I have a problem with SIP softphones. I created 2 estensions 1000
and 1001 they're both in different cities, when I 1000 (on the same network
as TrixBox) dial 1001 (the other city) they answer and can hear me, but I
don't hear them, and when they call *43 for echo test it plays the "You're
entering echo test..." but when it tries to start echo it just hangs up. and
the log says
-- Executing [EMAIL PROTECTED]:1] Answer("SIP/1000-08939150", "") in
new stack
-- Executing [EMAIL PROTECTED]:2] Wait("SIP/1000-08939150", "1") in new
stack
-- Executing [EMAIL PROTECTED]:3] Playback("SIP/1000-08939150",
"demo-echotest") in new stack
-- Playing 'demo-echotest' (language 'en')
== Spawn extension (from-internal, *43, 3) exited non-zero on
'SIP/1000-08939150'
-- Executing [EMAIL PROTECTED]:1] Macro("SIP/1000-08939150",
"hangupcall") in new stack
-- Executing [EMAIL PROTECTED]:1] ResetCDR("SIP/1000-08939150", "w")
in new stack
-- Executing [EMAIL PROTECTED]:2] NoCDR("SIP/1000-08939150", "") in
new stack
-- Executing [EMAIL PROTECTED]:3] GotoIf("SIP/1000-08939150",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [EMAIL PROTECTED]:6] GotoIf("SIP/1000-08939150",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [EMAIL PROTECTED]:9] GotoIf("SIP/1000-08939150",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [EMAIL PROTECTED]:11] Hangup("SIP/1000-08939150", "") in
new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/1000-08939150' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/1000-08939150'
I'd appreciate your help a lot, I'm not if this is a forewall issue or
something wrong with my asterisk config.
Thanks a lot.
--
DAVID GONZALEZ H.
GNU/Linux Debian+SuSE+RedHat+LFS
TECNICO EN REDES
NETWORK ADMIN
http://www.computrabajo.com.co/cvs/dgonzalezh
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
