search for: inviting

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?

Andrew Martin wrote: > ----- Original Message ----- <snip> > > By doing a number of test calls today, I have managed to reproduce this while > sip debugging was on, so I have that information available now as well: > http://pastebin.com/ZJqzdvY3 > > This was a call from 113 to 146 via a queue. Note that the asterisk server is > at 10.10.32.251. I see the following:

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

I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRel INVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx

I don't understand what a SIP invite is. Certainly it's explained as: "This article explains the main fields included in a SIP INVITE, which is sent to set-up a VoIP call. A SIP INVITE message contains typically between 4 and 6 header entries with contact information inside them." http://www.3cx.com/blog/voip-howto/sip-invite-header-fields/ The article enumerates the headers

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

Hello, I have a issue between asterisk and windows based VoIP system (Client). Vendor SIP Server --> My asterisk --> Client Here is ethereal trace between asterisk and client. 1 0.000000 192.168.3.222 -> 192.168.4.23 SIP/SDP Request: INVITE sip:1978525648 at 192.168.4.23 <sip%3A1978525648 at 192.168.4.23>, with session description 2 0.042380 192.168.4.23 -> 192.168.3.222

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

Hello! I encounter the following problem (asterisk 11 and 13) with Teconisy as trunk provider with enabled qualify and default t1min (100ms): Teconisy most often doesn't answer the first invite before asterisk default t1min ended. Therefore asterisk sends one more invite. This second invite is answered by Teconisy with status 486 - Request terminated - Channel limit exceeded. (The second

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