similar to: SIP, NAT, and bindaddr

hI, Trying to understand how to setup two PRIs in sip.conf. Using Asterisk 1.4.23. I have a provider giving me two PRI (different rate centers) through SIP. Both PRI comes in from the same IP on the provider side, but go to two different IPs (both on the same box) on my side. How can I setup two different SIP peer, one for each of the PRIs I get, if all I can use to differenciate them

Hi to everybody, My IAX is not working, When I type reload IAX it returns me: AsteriskSlave*CLI> iax2 reload == Parsing '/etc/asterisk/iax.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13546 set_config: Ignoring bindport on reload [Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13610 set_config: Ignoring bindaddr on

Hi, I have already tried to change for bindaddr=0.0.0.0 but it didn't worked. 2016-08-26 11:44 GMT-03:00, Frank Vanoni <mailinglist at linuxista.com>: > On Fri, 2016-08-26 at 10:12 -0300, Vitor Mazuco wrote: > >> bindaddr = all > > Try: > > bindaddr=0.0.0.0 > > > > > -- > _____________________________________________________________________

I'm running CVS HEAD. The last time I updated was January 7th, at which time everything was fine. Having updated again today, January 17th, I'm having problems with IAX2. I use the "bindaddr" directive for both SIP and IAX2, and while SIP respects it, IAX2 doesn't. It listens on every interface, and uses every one of them for outgoing source addresses. This breaks IAX2

Hi all, is it possible to bind SIP protokol not to all but to more that one interfaces. I did try use bindaddr, but i don't know right syntax. Could anyone help me. Thanks, Bob.

Hi, I'm trying to configure a dual homed asterisk server with iax accepting connection on all address. from iax.conf.example bindaddr = 0.0.0.0 Address to bind to ( all addresses on machine ) but if i register a client using the second ip address i will receive the response from the first ip address and obviously the client discard this. let me explain more: Client : 192.168.0.4

Hello all, I am running asterisk on VMs with standby heartbeat configuration, Heartbeat assigns a virtual IP 172.20.255.40 on machine afterwards asterisk is started. In the sip.conf, I have explicitly define bindaddr=172.20.255.40 but sometimes I see packets coming from physical IP 172.20.255.41 I have both tcp and udp transport enabled Here is the lsof -ni :5060 output asterisk 2878 asterisk

Should it work to have a multi-homed asterisk server with grandstream phones on the internal network and another grandstream phone on the internet and be able to call between them? I set the bindaddr to the external IP and pointed the internal and external grandstream phones to that address. The signalling works fine to call between phones, but when you pick up the ringing phone you get a

Gentlemen, Forgive me if I am posting at the wrong place! I was going to test the "new" chan_ooh323 driver so I did install: debian: Linux sip2 2.6.26-2-686 #1 SMP dahdi-linux-complete-2.2.0.2+2.2.0 Asterisk SVN-trunk-r231692 Did enable chan_ooh323, everything compiled without any problems. Hardware setup: Phone (975) - Avaya CM - H.323 - Asterisk - X-Lite (0317998975) X-Lite can

Hi all, Still having the one way sound problem. Any suggestions how to hunt the problem down ? Regards, Thorsten --------------------------------------------------------------- Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26

Hello, Please forgive me if this has been addressed previously. I have been searching the archives and have not come across what I thought was a solution. My * server is behind a DSL router using a NAT IP address of 10.0.0.9. A colleague running XP and X-Lite can register with * from his home, specifying my public IP as the SIP proxy in X-Lite (however this is only true if I have the NAT flag

Hi all, OK. I've tried trawling the archives, but I'm not getting very far. I've got an Asterisk box behind a NAT which I want to register with a SIP provider. In my sip.conf I have (edited to protect the innocent): ----- [general] port = 5060 bindaddr = 0.0.0.0 disallow = all allow = alaw allow = ulaw allow = gsm context = bogus-calls tos = lowdelay nat = yes register =>

in sip.conf [general] port = 5060 ; The TCP/IP port for SIP communiations bindaddr = 0.0.0.0 ; Address to bind to. 0.0.0.0 all addresses on server. context=other ; Default for incoming calls disallow=all allow=ulaw allow=gsm in extensions.conf [general] static=yes ; These two lines prevent the command-line interface writeprotect=yes ; from overwriting the config file. Leave them here

I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Below is my sip.conf [general] port = 8060 ; Port to bind to (SIP is 5060) bindaddr = 0.0.0.0 ; Address to bind to (all

I am having difficulty configuring SIP behind NAT (using latest CVS). Using sip.conf: [general] port=5060 ; Port to bind to externip=ww.xx.yy.zz bindaddr=0.0.0.0 nat=yes register=>[userid]:[password]@voiptalk.org/2000 [voiptalk.org] nat=yes externip=ww.xx.yy.zz type=friend secret=[password] nat=yes reinvite=no canreinvite=no I fail to register. SIP Debug gives: SIP

We are trying to use a softphone from a location that is behind a firewall. We are using x-lite as the softphone. So far, we've been able to get the phone to register with the asterisk server, and it can receive voice from the asterisk server (IE, voicemail, etc). However, asterisk can't hear anything from the softphone. We have used 2 different machines to test this on. We are watching

Hi List, Additional with the latest tries from the below I get a nice random seg fault when I hangup on PSTN. (With obviously no sound on x-lite, still!) asterisk -vvvvgc results after hanging up the pstn line in: -- Executing Hangup("SIP/1087997-d79f", "") in new stack == Spawn extension (sip-phone-out, h, 2) exited non-zero on 'SIP/phonenumber-d79f' Segmentation

Hi all, We have a very basic * installation for testing purposes. The * is connected to PSTN with BRI and setup with X-Lite over plain lan. (local IP's) OS: Linux/Debian unstable. Asterisk CVS-10/29/03-23:46:26 chan_capi On the IP side: X-lite (build: 1084) Calling and get calls on PSTN from X-Lite is no problem. We only get sound from PSTN to X-lite. Never from X.-lite to PSTN. The

I've been googling and talking with Libretel about my setup and the fact that incoming calls to my asterisk box through the Libretel number reach my box (I hear the greeting being played) but then don't accept DTMF. Here is a rough diagram of my setup: Asterisk | server | NAT <------------ Libretel | router | Note that there are NO SIP

Hi, I am playing around with SIP extensions on my local lan using X-Lite but I am having a bit of difficulty, I have set up X-Lite and my sip.conf accordingly, but when I start it I get the following message: "Login failed! Contact Network Admin" I am still able to dial local extensions on my * with x-lite even though it is in this state, although trying to dial my sip extension from