search for: invites

I have a high volume asterisk 1.40 installation and I ran a SIP SHOW CHANNELS. (see partial output below). My questions are: 1. "wc-l" of the output shows 4000 lines. Does this mean 2000 active calls? (2 channels per call) 2. The latter part of the output shows "unkn" for Form column. Why does it not know the codec? Could it be UDPTL? Or are these calls messed up? 3.

INVITATION BETA EMAIL I have in the email that the app send to friend''s email address ------------------------ You are invited to ExampleApp.com click below to signup http://localhost:3000/signup.efweiuvwnjernfwkefwebhsohj ------------------------ But I have a dot in the url beteween http://localhost:3000 and the token I wish the following url

Hi Everyone... I''m following a railscast episode on how to implement an invitation feature. It''s going really well, but i''ve hit a minor snag that I cant get over.. undefined method `generate_token'' for #<Invitation:0x2563bf8> The invite form allows me to check for a user, and whether they already have registered. If they have, the invitation is not

----- Original Message ----- > From: "Joshua Colp" <jcolp at digium.com> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> > Sent: Tuesday, May 12, 2015 5:42:57 PM > Subject: Re: [asterisk-users] "Retransmission Timeout" results in dropped calls after 32 seconds > > Andrew Martin wrote:

Hi, I am working on a scheduling app and I have a perpelextion (new word). I am wondering if the problem is my data model I have Users. Users can create Events. Users can be invited to Events created by other Users. So... user.rb class User < ActiveRecord::Base has_many :invitations # invitations to other users'' events has_many :events, :through => :invitations # all events

Hi, ruby 1.8.4, rails 2.2.2, mongrel 1.5.1, win xp I read in the docs that this : map.resources :articles do |article| article.resources :comments end should result in this lot: article_comments_url(@article) article_comment_url(@article, @comment) article_comments_url(:article_id => @article) article_comment_url(:article_id => @article, :id => @comment) So when I did

I am trying to optimize some methods in my model so they don''t repeat CPU intensive algorithms every time I call the method in the same request/response cycle. Eg. ================ def invitations all_pgm_updates.find_all do |update| update.invited? end end ================ I want to do something like: ================ def invitations if @invitations.nil?

...> INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > > This appears to start out with a successful SIP conversation (ending with the > first ACK), so it is unclear to me why we have two new sets of INVITEs sent > afterwards. Asterisk has sent a re-INVITE to have the media flow directly. The device (seems) to respond with the 200 OK (you can tell based on the CSeq) for the initial INVITE, and not for the re-INVITE. As Asterisk gets no response to its re-INVITE it gives up and terminates the dia...

Andrew Martin wrote: <snip> >> > Joshua, > > As a mitigation for this problem, could I increase the "timerb" option in sip.conf > to a large value, say 1 hour (instead of the default 32 seconds)? What other > consequences would there be from this change? I don't know if chan_sip will allow this, but if it does... it'll keep transmitting over and

...hat I've read.. a bye should be the termination of the session/channel and therefore it should be hungup and removed.. yet it is not. I am using cvs head from 2005-10-08 00:00 .. I can't use the latest cvs head as it's rather ugly with sip right now.. especially on refer/redirect/reinvites.. but that will be left for a different topic. I believe from looking at things that the sip gateway involved with the sip session is re-using a particular call identifier immediately after it believes that call from before is gone.. (possibly a bug on the vendor side as far as that goes) but r...

...ends the invite? Asterisk? A soft-phone? I found sample config's to send an invite with Asterisk but no other method was given. Can only Asterisk send an invite? Why? The article says that it's sent "to set-up a VoIP call," so presumably any reasonable soft-phone sends these invites as a normal process. That's all well and good, but how do send an actual invite and get a response? This can only be done through Asterisk? This is in the context of: Requires IP Authentication to be setup through the portal and associated with LRN under Telephone Data <https://portal...

I'm using "insecure=invite" with two different dial plans, it it working with one dial plan but not with the other. What other parameters could influence "insecure=invite" In sip.conf below "insecure=invite" is working OK [pstn-1270] type=friend secret=spa3k username=voice-1270 mailbox=369 host=dynamic insecure=invite canreinvite=no disallow=all allow=ulaw

On 06/14/2017 at 10:17 PM, Joshua Colp wrote: > On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > <snip> > >> >> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >> Just one exception - and that's the package in question, which can't be >> seen in tcpdump. >> >> I extended the above patch by adding

Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose

....168.4.23:5060, with session description 10 4.046102 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23:5060, with session description I disconnected the call, Receive BYe from Vendor, Asterisk acknowledge Bye and does not send Bye to the client. Few more invites from Asterisk to the client machine. 11 8.046123 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23:5060, with session description 12 16.046179 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23:5060, with session descr...

Hi All, I;m using asterisk 1.4 with FreePBX and a Grandstream 4108. I am noticing a delay calling in and out via the FXO, but calls to local extension are ok. What i noticed when i used ngrep is that, it sends invite but got no response from the server, send another invite but got no response again, then again until it finally gets it. but if you will notice on the 2nd ngrep, the asterisk

...ne! 2. Why does asterisk stop the call completely after the second invite, which is canceled by Teconisy? It should be ignored because it means, that the first invite is already processed by Teconisy. Btw: I can see the same behavior with German Telekom. Asterisk most of the time sends two invites, too, but here I don't get an error back from Telekom, because I have two channels and therefore the channel limit isn't exceeded most of the time. Setting t1min to the VoIP standard of 500ms prevents these unnecessary double invites with German Telekom, too. Therefore: Please use the sta...

Hi, I can't make outgoing calls via Broadvoice. I have tried each and every configuration that was posted to list previously. I am able to receive incoming calls fine. I get the following in asterisk console: ===================================================== asterisk*CLI> show version Asterisk CVS-HEAD-03/10/05-22:51:28 built by vicky@asterisk on a i686 running Linux

Good evening! I was wondering one thing, I'm using freepbx to configure my asterisk server and I have a problem with some inbound calls. When I receive a call to an INVITE sip:01xxxxxx at myip.com I an set an inbound route! It matches a DID number. How can I route an INVITE sip:s at myip.com? The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it

Hi! This is my first post in this forum. I''m learning RoR for two weeks and I''m very interested about how to improve this framework. I was testing one app I''m working in and I had some problems with destroying. The code is simple (and maybe wrong, as I said i''m just learning!). When you destroy a league, you send a message to all the users associated to this