search for: invite

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

2007 Feb 27
Help understanding SIP SHOW CHANNELS
....c:819 ast_udptl_new_with_bindaddr: No UDPTL ports remaining" errors - is this related to number 2 above? Thanks, MD Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last Message 2897516#15 1db77942648 00102/00000 g729 No Init: INVITE 2897516#13 585240b13ef 00102/00000 g729 No Init: INVITE 2897516#13 0244b8e668d 00102/00000 g729 No Init: INVITE 2897516#16 46d0960e602 00102/00000 g729 No Init: INVITE...
2011 Mar 17
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 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 http://localhost:3000/signup/efweiuvwnj...
2010 Feb 09
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 is not sent, and a flash message is printed. If the email does not exist, the invite count (starts at 5) decrements until you have none left. When i click invite, i get the error above. It...
2015 May 13
"Retransmission Timeout" results in dropped calls after 32 seconds
...cket goes. That's the only thing left really. It's also > possible something got fixed in relation to directmedia between your > version and latest 11. > Joshua, Looking at the packet capture from the asterisk server during this time, I see the following sequence of SIP packets: INVITE (102) - initial call connecting RINGING (102) - initial call connecting RINGING (102) - initial call connecting OK (102) - initial call connecting ACK (102) - initial call connecting OK (102) - initial call connecting (seems like a duplicate OK) INVITE (103) - re-INVITE to go to bypass mode ACK (10...
2006 Jul 23
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 tha...
2008 Dec 31
resource api docs not working for me
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
2015 May 11
"Retransmission Timeout" results in dropped calls after 32 seconds
..., I have managed to reproduce this while > sip debugging was on, so I have that information available now as well: > > > This was a call from 113 to 146 via a queue. Note that the asterisk server is > at I see the following: > INVITE sip:146 at SIP/2.0 > SIP/2.0 180 Ringing > SIP/2.0 180 Ringing > SIP/2.0 200 OK > ACK sip:146 at SIP/2.0 > INVITE sip:146 at SIP/2.0 > SIP/2.0 200 OK > ACK sip:146 at SIP/2.0 &gt...
2006 Mar 17
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 @invitations end ================ but isn''t the instanc...
2015 May 12
"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
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 SIP/2.0 10. Rx SIP...
2016 Jun 29
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."
2010 Feb 15
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=d...
2017 Jun 15
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
...e scenario and digging deep into it. Nothing > immediately springs to mind. > After enabling pjsip specific debug log in asterisk, I can see, the following behavior: Incoming packages from network are always signaled like this (e.g.): sip_endpoint.c Processing incoming message: Request msg INVITE/cseq=10 (rdata0x7f5f1801a758) <--- Received SIP request (918 bytes) from UDP: ... The path of the critical not existing package is like this and can't be seen elsewhere (there wasn't any corresponding incoming message entry before): sip_endpoint.c Distributing...
2007 Apr 16
sip tcp support
...l, 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 is ICMP Destination unreachable (Port unreachable). any hints? Thx in advance Xten asterisk HiPath | INVITE | | |---------->| | | TRYING | | |<----------| | |...
2009 Sep 05
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 -> SIP/SDP Request: INVITE sip:1978525648 at <sip%3A1978525648 at>, with session description 2 0.042380 -> SIP Status: 100 Trying 3 0.044235 -> SIP Status: 183 Session Progress 4 0.046546 -> S...
2011 Jan 11
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 replied to all the INVITE's it received before it says Ringing. Really need help on this badly, anyone has an idea....
2016 Apr 25
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 invite obviously is interpreted as second, new call). Asterisk therefore cancels the first(!!) call...
2005 Mar 10
asterisk and Broadvoice Outgoing Again :( [XXXXXXXX] type=peer user=phone fromuser=XXXXXXXX secret=PPPPPPPP username=XXXXXXXX insecure=very context=default authname=XXXXXXXX dtmfmode=inband dtmf=inband canreinvite=no ================================================ -- With regards, Vicky Shrestha System Director WorldLink Communications Jawalakhel , Kathmandu, Nepal
2007 Nov 13
route INVITE
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 I an set an inbound route! It matches a DID number. How can I route an INVITE sip:s at The number only appear in the To: Section. Thanks! Example: With this one, I cannot route it (there is only the number to be reached in the To: section) # U
2008 Feb 15
Destroy, dependent and performance
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