search for: invitation

Displaying 20 results from an estimated 3986 matches for "invitation".

2007 Feb 27
1
Help understanding SIP SHOW CHANNELS
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.
2011 Mar 17
7
Beta Invitation in Rails 3, little problem
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...
2010 Feb 09
2
undefined method `generate_token'
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 i...
2015 May 13
0
"Retransmission Timeout" results in dropped calls after 32 seconds
----- 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:
2006 Jul 23
4
has_many AND has_many :through ?
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 the user is invited to #HERE IS THE PROBLEM has_many :events # the events that the user is the owner of end event.rb class Event < ActiveRecord::Base has_many :invitations belon...
2008 Dec 31
1
resource api docs not working for me
...nts_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 my own idea of the above : map.resource :people,:singular=>:person do |person| person.resources :invitations; end I was surprised to find no method such as person_invitations (with or without the :singular=>true): all the methods are pluralised for people. And for those methods that do exist none take the person id, even the index method (which makes it useless to me). Here is what rake is telling me...
2015 May 11
2
"Retransmission Timeout" results in dropped calls after 32 seconds
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
2006 Mar 17
2
Temporary Model Data
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? @invitations = all_pgm_updates.find_all do |update| update.invited? end end @in...
2015 May 12
2
"Retransmission Timeout" results in dropped calls after 32 seconds
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
2005 Oct 18
1
sip rfc bye violated?
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
2016 Jun 29
2
what is a SIP invite, and who can issue them?
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
2010 Feb 15
2
insecure=invite - not working for different dial plan
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
2017 Jun 15
2
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
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
2007 Apr 16
2
sip tcp support
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
2009 Sep 05
2
Need some help/Suggestions for multiple invites from Asterisk
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 ->
2011 Jan 11
0
slow response to INVITE
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
2016 Apr 25
2
Second invite after 100ms (with default t1min=100) --> canceled call problem!
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
2005 Mar 10
5
asterisk and Broadvoice Outgoing Again :(
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
2007 Nov 13
1
route INVITE sip:s@sip.test.fr
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
2008 Feb 15
3
Destroy, dependent and performance
...as deleted, you are a criminal!" redirect_to route(''/leagues'') else flash[:error] = ''Something went wrong creating this league!'' redirect_to route(''/leagues'') end end class League < ActiveRecord::Base has_many :invitations, :dependent => :destroy has_many :participants,:through => :invitations,:source => :user,:conditions => "accepted == ''t''" end First of all: Leagues and invitations are dependent as you can see in the model. When I destroy a League the SQL generated is thi...