similar to: SIP Reinvites

Hi all, Love tinc by the way. It's a great VPN. I'm having issues with 2 nodes always talking through an intermediate node. My set up is a VPS in a cloud somewhere that's running tinc and 2 other nodes - one a roaming laptop (always NAT'd) and the other a server behind a dynamic IP home broadband connection (Not NAT'd but firewalled). Neither the laptop nor the home

Sipura does include STUN as an option. It has for quite some time. We are using it with all of our Sipuras behind NAT'd gateways and it works great! Try upgrading to the latest Sipura firmware rev. Darren Nay > -----Original Message----- > From: John Todd [mailto:jtodd@loligo.com] > Sent: Saturday, May 22, 2004 1:57 PM > To: asterisk-users@lists.digium.com > Subject:

I have 20-30 GXP2000's connected to * over a T1 line. Neither end is NAT'd and there is plenty of bandwidth available over the line. The GXP's are 1.1.5.15, which is the latest. I have a problem where the phones keep dropping off of * and I get a "failed to register" message in the log of *. Sometimes they eventually connect and sometimes, I have to reboot them to

I don't know if this is possible, and I can't quite get my head around how to do it... If I am using DUNDi for redundancy in a cluster, when Phone1 makes a call to Phone2, both Asterisk A and B will be in the RTP stream: +---+ +---+ | A |-----| B | /+---+ +---+\ / \ Phone1 Phone2 Is there a way configure re-invites

I'm using Asterisk (1.2.14, RedHat 9) but I've been having trouble with SIP re-invites. I have a DiD from an ITSP and when someone calls in, Asterisk plays a menu recording and transfers the call to the external line the caller selects. Since both sides of the call are external, I want to use re-invite to avoid the rtp packets from going through my server after the call is bridged. I

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Wednesday, August 22, 2007 10:51 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 37, Issue 88 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com

I've been wondering for a long time why my reinvite option is not working with Broadvoice anymore. It happend during the time Broadvoice was having a lot of issues, so I decided to wait. Recently I decided to test the same sip.conf with another VSP (SIPphone) and it worked fine! No issues on the reinvite. Note: clients and server using ULAW (only), no NAT or firewalls, public ip address and

is there a trick to get asterisk (1.6.2.13) to propagate codec-changing sip reinvites when directrtpsetup=yes? i'm trying to route calls to a gateway without keeping asterisk in the rtp stream. the gateway is first routing the call to a media server. when connecting the call to the downstream carrier a different codec is selected. the reinvite makes it to asterisk but asterisk isn't

On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote: > Hello > > I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom > Germany. We have sometimes problems with incoming and outgoing calls. > I hope I can explain it understandable. Hello Daniel, I'll find myself in the same situation a few weeks from now :-) > > For example, Asterisk sends a

Hello I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom Germany. We have sometimes problems with incoming and outgoing calls. I hope I can explain it understandable. For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de <http://tel.t-online.de/>), the message is answered with OK and the peer is registered. Usually INVITES comes now from this ip address. All

Guys. Im reading a lot about ser, nat, stun, etc. And I noticed there are a lot of ways to get around nat but I would like to hear some success stories about handling nat users with multiple voip phones behind nat. I have my asterisk box behind but ports are forwarded (5060 5004 10000-20000 for rtp and 4569 for iax2) but still.. I can quite figure out what ser and stund have to do on this

Maybe someone could elaborate on my first question again. If the ip address changes while a REGISTER period, the ip address of the peer isn't been updated. How can asterisk update the ip address of the peer? > Am 31.03.2015 um 12:36 schrieb Daniel Heckl <daniel.heckl at gmail.com>: > > Hello Sebastian, > > I had already seen this list of the hosts, but it is not

Scott, thank you four your reply. I had already though about both options, but the problem is, that after an ip change AND a new registration the ip address of the peer is not updated automatically. INVITES are answered with 401. Only after a sip reload the peer works again. That can't be normal... Daniel > Am 31.03.2015 um 22:45 schrieb Scott Griepentrog <sgriepentrog at

I'm at my wit's end! I've spent 2 days now trying to get what I thought was a very simply SIP + NAT arrangement working. I've trawled the web and picked brains, but nothing anyone suggests work. My setup is very simple. I have a * server in a datacentre, with a public IP address. There is no firewall in place, it's completely open (at least, as far as I'm concerned). I

I'm confronted with an issue that I am sure many others are too with Asterisk and scalability. I'd like to be able to build a cluster of Asterisk boxes to handle a large volume of simultaneous calls but have the feeling that the hardware requirements to handle large volumes of RTP streams would be too vast. So with that assumption I imagine a platform that would not get involved with the

Hi all, My Asterisk server is working fine, although every time that in the middle of any call there is a reinvite, the user hears a glitch. Why is this happening? How can I solve this problem? Thanks in advance, Ricardo Carvalho.

For a sip-only application, what exactly is required to ensure that calls completed via followme are reinvited? Can it at all? The code after outbound = findmeexec(targs, chan) calls ast_bridge_ call(). I don't see anything there which can cause a reinvite, yes? When the same peer is used for both the incoming and outgoing legs, it is a bit of a waste to proxy the rtp. And even when the

I am wondering if the SIP protocol and its implementation in * allows for changing codecs mid-connection. I've seen some questions regarding this on the list, but I've not found any clear answers. I've also seen the SIP_CODEC variable, but it's not clear that it will change the codec on an existing call. Also, there are mentions of needing a reinvite to make the change, but most

Is there a way in iax to have to endpoints talk to each other directly (after the call is setup by *) without going through *. In sip, with * you can do it by configuring sip.conf with canreinvite = yes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030728/0c711d05/attachment.htm

We all love Asterisk's ability to Dial(chan1&chan2) and take the first that answers. However, I have been encountering a problem when one of the channels is an external phone behind NAT and another is a local phone on the same net as the asterisk server. All have canreinvite=yes, and the phone behind NAT is correctly using Stun to give its external ports, which are opened to it in the