similar to: proxy howto

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

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:

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

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

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

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 List I work at an SIP Provider and we have added and SBC in front of our Voice Switch to protect it. This requires all our SIP Trunk customers to register via a 'proxy'. I struggle with Asterisk to work over a proxy. This is what I have done so far. register => username at sip.example.com:password at sbc.example.com This works fine, asterisk is sending registrations via the

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

I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using "X-Lite" I have no issue with settings as follows: Display Name: Any Name User name: 00575000010XXXX Password: 00575000010XXXX Authorization user name: <blank> Domain: directnationalloan.com Checked "Register with domain" and "Send outbound

Hi, I have asterisk register two users (client-1, client-2) with a SIP proxy. I have the same two SIP client registered with asterisk. Now my dial plan setup is such that any call from client-1/client-2 is forwarded to the SIP proxy and the SIP proxy then takes the routing decision. Calls coming from SIP proxy will dial out the respective user. Asterisk is required to stay in the signaling as

I've noticed what I think is curious behavior in using 'sub(fixed = TRUE)' and was wondering if my expectation is incorrect. Here is one example: v <- paste(0:10, "asdf", sep = ".") sub(".asdf", "", v, fixed = TRUE) The results I get are > sub(".asdf", "", v, fixed = TRUE) [1] "0"

Hi list! Take this markdown input: * asdf * asdf * asdf asdf What I want: <ul> <li>asdf <ul> <li>asdf</li> <li>asdf</li> </ul> <p>asdf</p></li> </ul> But markdown puts the first asdf into a paragraph. There is nothing I can do

-----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

Hi, I am generating a nice document with some inline code blocks and came around the following error: 1. asdf - \` asdf `` `asdf` `` produces: <ol> <li>asdf <ul><li>` asdf <code><code>asdf</code></code> </li></ul></li> </ol> instead of: <ol> <li>asdf <ul><li>` asdf

Hi, I specific outboundproxyport=9097 in version 1.4.4, but it doesn't work. It still connects sip port 5060. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070625/3fbf28e8/attachment.htm

This does not produce the expected output, yet does work on the dingus [1]: * asdf * asdf >> require ''maruku'' => true >> doc = Maruku.new("* asdf\n * asdf\n") => md_el(:document,[ md_el(:ul,[md_el(:li_span,["asdf * asdf"],{:want_my_paragraph=>false},[])],{},[]) ],{},[]) >> doc.to_html => "<ul>\n<li>asdf *

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

$ echo ${PWD##*/} somefolder $ if "${PWD##*/}" -eq "asdf" > /dev/null; then echo "this is the asdf folder"; else exit 1; fi bash: notthatfolder: command not found... this is the asdf folder $ So i just want to check that i'm in an exact folder. e.g.: "asdf" What's wrong with my one-liner? I just want to check, that a script is in a folder,

This is a dumb question, but I'm having trouble finding the answer to this. I'd like to do the following: x<-"asdf" and then have the object x.y become automatically converted/represented as asdf.y (sort of akin to macro variables in SAS where you would do: %let x=asdf and do &x..y) What is the syntax for having x represented as "asdf" in x.y ? Thanks,