similar to: Does outboundproxyport still work in 1.4.4

Looks like outboundproxyport doesn't support in 1.4.4 If you set the port, then it conflit with the one in "To URI" with host. I saw the code for outboundproxyport from the source, but is it a bug? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070625/7aad927d/attachment.htm

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

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

I want to setup a blind transer for auto forwarding to SIP peer. I have context forwarding looks like this in extension.conf [forwarding] ... exten => 511,1,Dial(SIP/sip_proxy-out) ... This will do the re-invite, which is attendance transfer maybe. But I want a blind transfer by REFER method. How can I do that? I know that the transfer() function may be able to do that. But I don't know

Hello, new to asterisk and trying to set it up to work with my voip provider (vbuzzer.com). I am behind a firewall that I don't have access to, to open ports etc. Before using asterisk, I tried vbuzzer's windows client, and linphone and twinklephone which all worked without having to enable nat or stun. However I did have to enter the outboundproxy server to get them to function. Not

Is there a way to let chan_sip skip host lookup? Problem is I have to have a peer host config for every sip message outgoing. For example, I cann't have this in extension.conf exten => 500,n,Dial(SIP/romi at 192.168.1.79) It'll return, chan_sip.c:2738 create_addr: No such host: 192.168.1.79 when call forwarding I have to have a peer in SIP [outgoing] host=192.168.1.79 ... in

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, I'm going to forward SIP request to special outbound SIP proxy with none SIP port. I have this in my sip.conf [sip_proxy-out] type=peer ; we only want to call out, not be called username=408 host=192.168.0.95 outboundproxy=192.168.0.74 port=9097 I want a To: 408 at 192.168.0.95 by proxy 192.168.0.74:9097 but it turns out the "To" also has the port To: 408 at

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 =

I have a repeated error in the asterisk console. Incoming call: Got SIP response 400 "Bad Request" back from 10.0.2.136 It is repeated every few seconds and never stops until I restart Asterisk. It starts after I make a call to a certain extension, 202 which transfers the call to another extension all phones then hang up. The SIP trace looks like:

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

When making an outbound call, the outbound peer return a 301 forwarded with URI to other domain, but asterisk think it's a local domain and try to look it up from extension.conf. How to configure so that a 301 forwarded with URI from other domain thinks it's outgoing to another proxy? thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi there! I am having problems registering my 7970 hardphone with Asterisk 1.4(with FreePBX interface). I had an earlier post about trying to get it to work first with a 7970 emulator (Cisco IP Communicator) on the Asterisk Forum : http://forums.digium.com/viewtopic.php?t=19160 Instead I decided to try the real phone instead, and was able to advance further. The firmware was able to install

I was hoping that someone may be able to shed some light on some issues I'm having on trying to get an Asterisk test server up and running. At the moment I have the basics, two Polycom hard phones (301 & 601 with expansion unit (which oddly will not power up)) that can call each other, log into voicemail (one touch) and have custom directories & buddy lists. Unfortunately some of the

I've tried everything with the * box after this weekend. I have read every document on the problems people are having with them after this weekend as well, but none of them address my problem. I checked my settings in my sips which I have below as well, I have changed the host file a few times, but this was new to me and I never had modified it before. I have and the same results

Hello All! I have a problem with my calls, that drops after 20 - 30 seconds. I got a piece of PAP2-NA log and Asterisk log and there's an error 603 - call declived, as showed. Thanks for any help. McCoy *********** PAP2-NA LOG *********** Feb 9 09:00:56 192.168.4.205 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060

I'm hoping people can suggest some ideas for debugging a problem that I'm having with DTMF. Unlike most of the DTMF problems reported here, it has nothing to do with Asterisk interpreting DTMF. My problem is with the synthesis of DTMF tones on outbound calls on a PRI connected to a TE412P card. I'm running * 1.4.10.1 with Zaptel 1.4.4. It is important to note that these problems

I have been going crazy with this also since Sat. Our server was working perfectly with BV but will now not place calls to BV. Incoming from BV works fine. I felt sad rebooting it today, it had been running for almost 200 days! Here is my error message from the console... Notice I am running today's CVS Connected to Asterisk CVS-03/08/05-14:32:39 currently running on com (pid = 1624)

Hi Everyone, I have setup an OpenVPN tunnel between Server A (running Asterisk) and Server B suppling it's SIP Phones with DHCP pool of IPs. So, the tunnel is established nicely and everyone can ping others. "sip show peers" shows the local subnet of the SIP Phones registered (192.168.100.0/24 ). But there is the old bad one-way audio. Calls also drop after few seconds. In the SIP

i have posted this before but was unable to resolve it. i have some new info so i figured i would try again. the trace from bandwidth.com are below. they are telling me that the ip that is bold should be our ip not bandwidth.com. i have changed every setting that i can see and nothing fixes this. Where would i change this at? they cannot tell me. INVITE sip:+185993133333 at 216.82.224.202