search for: invite

...0224]: udptl.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 172.16.116.29 2897516#15 1db77942648 00102/00000 g729 No Init: INVITE 172.16.116.29 2897516#13 585240b13ef 00102/00000 g729 No Init: INVITE 172.16.116.29 2897516#13 0244b8e668d 00102/00000 g729 No Init: INVITE 172.16.116.29 2897516#16 46d0960e602 00102/00000 g729 No Init: INVITE...

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 http://localhost:3000/signup/efweiuvwnjernfwkefwebhsohj ------------------...

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...

...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...

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...

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? @invitations = all_pgm_updates.find_all do |update| update.invited? end end @invitations end ================ but isn''t the instanc...

...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: > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > SIP/2.0 180 Ringing > SIP/2.0 180 Ringing > SIP/2.0 200 OK > ACK sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:5062 SIP/2.0 > SIP/2.0 200 OK > ACK sip:146 at 10.10.32.96:5062 SIP/2.0 > INVITE sip:146 at 10.10.32.96:...

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 SIP/2.0 / 102 INVITE 11. CancelDestroy 12...

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/si...

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...

...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:195.185.37.60:5060 ... 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 rdat...

...i 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 is ICMP Destination unreachable (Port unreachable). any hints? Thx in advance Xten asterisk HiPath | INVITE | | |---------->| | | TRYING | | |<----------| | |...

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 SIP Status: 100 Trying 3 0.044235 192.168.4.23 -> 192.168.3.222 SIP Status: 183 Session Progress 4 0.046546 192.168.4.23 -> 192.168.3.222 S...

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....

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...

...t; XXXXXXXX@sip.broadvoice.com:PPPPPPPP:XXXXXXXX@sip.broadvoice.com/broadvoice [XXXXXXXX] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com 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

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 (there is only the number to be reached in the To: section) # U 217.36.112.145:5...

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