similar to: outboundproxy issue

Hi, I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of Asterisk, but for the tests I made, every calls, even internal SIP calls between extensions are sent over the proxy that I have specified with the outboundproxy=xxx.xxx.xxx.xxx in sip.conf. I think this isn't the expected behaviour, right? Only OUTBOUND calls should go through the proxy, right? Am I doing

I managed to upgrade the phone to 8.2.2SR1 after renaming jar70sip.8-2-2ES1.sbn to Jar70sip.8-2-2ES1.sbn but the phone would continually say "Registering" and the red X next to the phone icon. The phone would eventually time out and couldn't make incoming or outgoing calls. Then I disabled registering with the proxy with the following line in the config:

Hi All, I'm connecting to my carrier which requires setting of outboundproxy. There has been few cases where the proxy server failed due to network issues and required us to use a secondary one. Is there a timeout or qualify setting for outboundproxy setting in sip.conf? I do appreciate if anyone can help please. Thank you -Abeed -------------- next part -------------- An HTML attachment

I'm having some trouble turning with trunk monitoring while using an outbound proxy. While all other sip messaging (e.g. calls) respects the outboundproxy setting, Options packets from setting qualify=yes do not. Asterisk tried to send the Options message directly to the "host=" IP, instead of the "outboundproxy=" IP as it should, verified with tcpdump. I've done a

Is it possible to set an outboundproxy for REGISTER from Asterisk? register => xxx:yyy at sip99.foobar.com [foobar] type=peer host=sip99.foobar.com disallow=all allow=g729 canreinvite=no secret=yyy fromuser=xxx port=5099 outboundproxy=xxx.42.149.69 However, SIP REGISTER still goes directly to sip99.foobar.com, not xxx.42.149.69. Thanks, Doug.

continuing discussions of 79xx issues. i've seen referenced and am experiencing difficulty getting a 7970 to work behind NAT to a public asterisk server. i am successful with 7960s. 1. SIP load is 70.8-3-3SR2S 2. config works fine if 7970 is connecting to an asterisk server a local LAN (same subnet) 3. when debugging it in a NAT'd environment I see the register and

In the latest release of chan_sip2 I've added support for SIP Outbound Proxy. I've seen a lot of requests for that lately, so if you can test this and confirm wheather it works for you or not, I'll be grateful. If I get positive reports, we'll try to add this to chan_sip in CVS. It works like this: * Configure outboundproxy in the general section of sip.conf outboundproxy =

Hi, I have been using Cisco 7960's with Asterisk for years. I am trying get a 7961 working and have a problem. In my configuration, not all of my line appearances register to the same Asterisk SIP server. I have an Asterisk server at home and another at work. My Line 1 button registers to the home server and my Line 2 button registers to the work server. This has worked for years

Hi, I've been trying to get asterisk to use an outbound sip proxy. Putting the outboundproxyhost directive in the [general] section of sip.conf, but it doesn't seem to work. My expectation would be that by setting outboundproxy and outboundproxyport in that location, then all dial commans (or at least dial commands of the form Dial(SIP/asdf@asdf.com) or Dial(SIP/asdf@99.99.99.99)) would

I'm running Asterisk on an external static IP address, siproxd on a different server with its own external static IP address, and communicating using a Grandstream behind a NAT firewall configured to register with Asterisk using siproxd as the outbound proxy. Now I'm aware that siproxd is not intended to be used as an outbound proxy but rather as a SIP relay when installed on the same box

Ok, I have a 7960 that's plugged into my laptop. my home network is wireless so I don't have a switch anywhere to plug the phone into directly. I'm running siproxd on my OS X laptop and I can make outbound calls from the 7960 fine (I guess I don't have the phone configured to register inbound calls via SIP), but the phone isn't registering to the asterisk box via siproxd

I decided to give the whole family IP phones for christmas, all hooked into my asterisk server, so all the nephews can have their own lines. However, one of the phones I got was the SNOM 200. That's worked fine for me on my own network, but I'm having bad luck getting it to work behind NAT talking to Asterisk. It talks to my termination/origination provider, which seems to ruthlessly

Le 01/05/2015 00:05, Andrew Martin a ?crit : > ----- Original Message ----- >> From: "Administrator TOOTAI" <admin at tootai.net> >> To: asterisk-users at lists.digium.com >> Sent: Thursday, April 30, 2015 4:43:33 PM >> Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In >> >>> I am running Asterisk 11.12.0 on CentOS

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 05/05/2015 10:59 AM, Andrew Martin wrote: > > > ----- Original Message ----- >> From: "Administrator TOOTAI" <admin at tootai.net> To: >> asterisk-users at lists.digium.com Sent: Friday, May 1, 2015 6:42:38 >> AM Subject: Re: [asterisk-users] OpenVPN Clients Intermittently >> Cannot Call In

----- Original Message ----- > From: "Administrator TOOTAI" <admin at tootai.net> > To: asterisk-users at lists.digium.com > Sent: Thursday, April 30, 2015 4:43:33 PM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > > I am running Asterisk 11.12.0 on CentOS 6.4. The asterisk server and > > internal phones are located on

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Guenther Boelter <gboelter at gmail.com> schrieb: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA256 > > On 05/31/2015 02:31 PM, Luca Bertoncello wrote: > > Hi list! > > > > Now all works as expected, at least in the simulation I did with > > AsteriskNOW. Hopefully it will work later, when Deutsche Telekom

----- Original Message ----- > From: "Administrator TOOTAI" <admin at tootai.net> > To: asterisk-users at lists.digium.com > Sent: Friday, May 1, 2015 6:42:38 AM > Subject: Re: [asterisk-users] OpenVPN Clients Intermittently Cannot Call In > > Le 01/05/2015 00:05, Andrew Martin a ?crit : > > ----- Original Message ----- > >> From:

Hi, My asterisk's version is 1.6.0.26. I've couple sip providers and I've for new SIP provider I need define outbound proxy. Everything is ok in peer section (outboundproxy=192.0.2.1). But what about SIP REGISTER messages? I need send SIP register messages also via outbound proxy. How to write SIP OUTBOUND call register statement and send this to proxy? If I define in general

Hello, I try to move our asterisk installation (3 Asterisk servers in different offices connected using IAX and a lot of SIP phones, as well as ISDN connections using CAPI) to use G.722 instead of G.711. Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one, which solves the gain problem). So SIP-to-SIP and to ISDN there is no problem. G.722 itself works and transconding to G.711 for

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