similar to: what is a SIP invite, and who can issue them?

On Fri, Mar 18, 2016 at 10:49 AM Administrator TOOTAI <admin at tootai.net> wrote: > Le 18/03/2016 16:20, Trey Hilyard a ?crit : > > I am trying to set up my Asterisk server so that it will recognize an > > incoming call to the Asterisk's own Location Routing Number (LRN), > > validating the "rn" in the INVITE and then using the Called Number from >

I am trying to set up my Asterisk server so that it will recognize an incoming call to the Asterisk's own Location Routing Number (LRN), validating the "rn" in the INVITE and then using the Called Number from the INVITE as the extension in the dialplan. The INVITE R-URI looks like: INVITE sip:+19135041291;rn=+19136630000;npdi at 12.4.240.200:5060;user=phone;transport=udp SIP/2.0

I tried to do what I with regular SIP to Transfer a call via 302 Redirect. In asterisk 12 we need to add the Tech, or not, but in any case, there is no transfer done. The call is closed. Here is a trace. How do I do this? [Jul 9 21:39:29] DEBUG[47716][C-00000002]: pbx.c:4869 pbx_extension_helper: Launching 'Transfer' -- Executing [17274428141 at redirect:30]

Telnyx, 382com, voicetel and as others mentioned BandWidth. I have contacts at 382 and voicetel if you want an intro. On Thu, Aug 17, 2023 at 11:50 PM Federico <federico at digitalipvoice.com> wrote: > I am looking for a decent provider of SIP Trunks but it has to pass the > Stir Shaken token to the next carrier. Does anybody know about any? > Sipstation from Sangoma, does not

Having an issue with receiving faxes, but when I pass through the fax. Currently, I receive the fax with Digium's Fax for Asterisk, store it and the initiate an outbound call to our fax server. (XMedius Fax). This works, but we would prefer to have Asterisk simply route the call directly to the fax server and take the store and forward out of the equation. When I do that, however, the

Hello everyone, I use winbind against a Samba DC for nsswich, and on one client it works perfectly (Samba 3.5.15 on all systems). On another client, everything works except SID_TO_UID (i.e. wbinfo -i, -S ... which breaks directory listings, too). I've now tried to narrow down the problem in a level 10 log, but I need some help interpreting. In log.winbindd, I see the following when

My only problem now is this: I have a broadvoice number that I can dial out on, and i ncomming calls word fine. Once the ip phone handset picks up, the audio bi-directionally is perfect. The problem is the audio BEFORE the handset picks up is silent on the broadvoice side. The user calling in over the sip number doesnt hear the menu, or even the ringing. Once the handset picks up then

<http://localhost:3000/hello/world> gives: "Unknown action No action responded to world" I''m following along with <http://wiki.rubyonrails.org/rails/pages/ Tutorial> and what stands out are statements like: "Now take a look at app/controllers/hello_controller.rb." Which doesn''t exist for me, I had to create multiple directories. I

I want to iterate ?nodes? and ?leafs? for a yaml document: thufir@ARRAKIS:~/projects/rss$ thufir@ARRAKIS:~/projects/rss$ ruby user.rb user.rb:6: undefined method `[]'' for nil:NilClass (NoMethodError) from user.rb:5:in `each_key'' from user.rb:5 thufir@ARRAKIS:~/projects/rss$ thufir@ARRAKIS:~/projects/rss$ ruby user2.rb user2.rb:5: undefined method `[]'' for

Looking at: http://appdb.winehq.org/objectManager.php?sClass=application&iId=10591 What are the steps to install this application? Yes, it's a garbage application, but I'd like to at least give it a go. Looks like msiexec apparently isn't the right approach. Should that be through wcmd instead? thufir at ARRAKIS:~/.wine/drive_c$ thufir at ARRAKIS:~/.wine/drive_c$ msiexec

I ran dovecot -a and the blizzard of data seemed ok to my limited knowledge. Is there another log I should look into to trace this error down? Dovecot and system info: thufir at dur:~$ thufir at dur:~$ dovecot --version 2.0.19 thufir at dur:~$ thufir at dur:~$ cat /etc/lsb-release DISTRIB_ID=Ubuntu DISTRIB_RELEASE=12.04 DISTRIB_CODENAME=precise DISTRIB_DESCRIPTION="Ubuntu 12.04.1

I just want to make a SIP call from 192.168.1.3 to 192.168.1.4; or not even a call. Ring? Beep? Ping? Some sort of "hello world" connection. 192.168.1.1 netgear router 192.168.1.2 asterisk (vicidial) 192.168.1.3 ubuntu client 192.168.1.4 mac OSX client (not shown) Do I have a firewall problem which would impact a soft phone from establishing a connection?

To use dovecot-postfix stack with thunderbird, do I put the configuration into /usr/share/dovecot/protocols.d/impad.protocol? That would seem to be how the stack is configured. "Thunderbird To use with Thunderbird, edit the file /etc/dovecot/dovecot.conf: protocol imap { ... login_greeting_capability = yes imap_client_workarounds = tb-extra-mailbox-sep }"

Why is dovecot using Maildir and not IMAP. Or is it using even using Maildir at all? Currently I'm using mailman, postfix and dovecot to manage a mailing list. Mail is sent to thufir at dur.bounceme.net which the "mail server delivery agent stack provided by Ubuntu server team" of dovecot-postfix handles fine, keeping it locally, so far as it goes. The mail ends up in

What's the difference between user "123" and "devries"? Based on the output here, they seem the same..? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes 123

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured out that they are sending the call to an extension that matches my number with them, in the

Can I dial directly from the asterisk console with the Dial() application? or, is channel originate preferred: channel originate SIP/thufir extension 18003569377 at outbound thanks, Thufir

On Fri, 20 Feb 2015 08:46:13 -0500, Andres wrote: > A "sip set debug on" will give you more info on why you are getting the > 404. It probably has to do something with your context/dialplan. on tleilax: tleilax*CLI> tleilax*CLI> sip set debug on SIP Debugging enabled tleilax*CLI> on doge: thufir at doge:~$ thufir at doge:~$ sudo sipsak -vv -s sip:devries at

Hi, I **think** that I have user of thufir101, because I get a 200 response below, but I also get a 404. It seems to depend on how I send the ip address/fqdn? tleilax*CLI> tleilax*CLI> sip show users Username Secret Accountcode Def.Context ACL Forcerport 201 password 201 default No Yes

I''m trying to track down why I''m getting close to desired output, but not quite. It looks like the webpage sorta-kinda is grabbing the right data. For instance, there are three Login objects which should be showing and it looks like the show is grabbing three objects, so that fits, to a degree. However, it''s just grabbing the hexadecimal object id, and not