similar to: More IP address in bindaddr directive

I'm trying to make H.323 trunk between AVAYA&Asterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob.

Hi all, I have a trouble when I try to configure asterisk to make calls between IAX and SIP. IAX I'm using to connect between asterisks a on SIP I have phones. The calls come from higher asterisk to my on IAX, SIP phone is ringing and when I hang up then dial command ends and connection is loss. When I'll make connection between asterisks on SIP then all work fine. Does anybody has any

Hi all, is it possible to monitor RTP protocol (latency, errors, ...) by Asterisk or other software. Thanks for answer, Bob.

Could you kick me, I can't dial more then 9 digits. Is anyone some default length of extensions or dialed number. Thanks, Bob.

Anybody doing email to fax using spandsp?

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, 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

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

I have a configuration where RTP traffic is going out interface pub0, and coming back into through pub1. I have bindaddr=0.0.0.0 in sip.conf, and a netstat -an shows: udp 0 788 0.0.0.0:5060 0.0.0.0:* which means that Asterisk is listening on all addresses (on all interfaces?). Anyway, when the RTP traffic comes back in on interface pub0, Asterisk does nothing with it. A

Hello, I am doing some tests with asterisk on a dual-stack environment. I have some doubts regarding asterisk binding addresses on a server with 2 network cards. According to asterisk documentation: /; With the current situation, you can do one of four things:/ /; a) Listen on a specific IPv4 address. Example: bindaddr=192.0.2.1/ /; b) Listen on a specific IPv6 address.

I just see warning? 2016-08-29 11:30 GMT-03:00, Telium Technical Support <support at telium.ca>: > This shows that asterisk's IAX is already bound to all adapters - so that's > good. Symptomatically does your IAX stop working? Or do you just see a > warning? > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com >

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 > > > > > -- > _____________________________________________________________________

Oh! In that case ignore it. Asterisk won't rebind the adapter if you've only changed parameters. The message is misleading -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vitor Mazuco Sent: Monday, August 29, 2016 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

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

Hey everyone, I'm an * newbie and I'm trying to get things going here, and please forgive me if you've heard this problem before. I have my * server on OS X behind a NAT, and I'm trying to connect to SIP soft phones (X-Lite) on various computers, some behind the same NAT and others with public addresses. I have successfully configured port forwarding and a few minutes ago I was

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

Hi, see the log below root at AsteriskSlave:~# ip addr 1: lo: <LOOPBACK,UP,LOWER_UP> mtu 65536 qdisc noqueue state UNKNOWN group default link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00 inet 127.0.0.1/8 scope host lo valid_lft forever preferred_lft forever inet6 ::1/128 scope host valid_lft forever preferred_lft forever 2: p3p1: <BROADCAST,MULTICAST> mtu

Trying to configure IAX for use I think I have everything set right. But my IAX phone wont connect. When I run wireshark I'm seeing this Note if above screenshot from wireshark does not show here is a link for it: http://img402.imageshack.us/i/tempe.jpg/ I've tried a variety of setups in my IAX.conf (they all end up with the same issue, tried just bindaddr=0.0.0.0 with

I have a large file that has a line like: bindaddr=0.0.0.0 ; some other text I want to replace the 0.0.0.0 with my address 192.168.1.8 and remove everything else on the line to get: bindaddr=192.168.1.8 How can I do that? Thanks, jerry

I couldn't be happier with the simplicity of this - but it's a hack! Hi all, I'm currently using a SIP client (BT101) to connect via DSL to a remote instance of Asterisk. - Asterisk has a private IP behind my OFFICE router. - The SIP client has a private IP behind my HOME router. I'm doing this _without_ the use of STUN or proxy servers. Here's how it works: -