Re: [Asterisk-Users] Delayed ringing on some SIP phones
Hi :) Chris Bagnall wrote: Hello all, What I'd like to do is implement a delayed ringing strategy - i.e. if the call comes in for Company 1, only their SIP phones will ring for the first 15 seconds, then if there's not been an answer, company 2's SIP phones will also start ringing. Is there any way to do this without stopping Company 1's phones ringing (i.e. timing out the dial statement after 15 seconds)? Well, I asked this, too and the solution was: exten => s,1,Dial(SIP/company1,15) exten => s,2,Dial(SIP/company1&SIP/company2,30) Thanks in advance. Regards, Chris HTH and regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] oh323 problem
[EMAIL PROTECTED] wrote: WARNING[12461]: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: _ZNK8PChannel7IsClassEPKc Oct 10 14:29:20 WARNING[12461]: Loading module chan_oh323.so failed! Does anybody know which is the problem ? It seems Asterisk source and binary version do not fit. Andrea HTH, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to send error codes to connected phone?
Hi all :) Is it possible to send a specific error code (e.g. 500) to a connected phone? If AGI can do it that would be great :) Thanks and kind regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is realtime meetme supported by Asterisk
Hi :) Am Donnerstag, 29. September 2005 09:03 schrieb Voice over IP: > Hi all, > > Is realtime meetme conference supported by Asterisk? > Yes and no. I wrote a patch for an older CVS-Version and will port it to the latest CVS version. Will take 2 or 3 weeks ;) So current versions do not support MeetMe with Realtime. > > Regards. Kind regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Early Media in 180 Ringing
Hi :) Am Montag, 26. September 2005 19:48 schrieb Ronald Voermans: > Hello, > > As you can see below, the SIP message from 10.254.254.1 (the PSTN > Gateway) has SDP, while * (with 192.168.0.173) removes the SDP content. > > How can this be solved? > Well, I am not that expert but AFAIK your PSTN gateway should send a 183 (Session progress) than a simple 180. Do you use Dial(SIP/blah|30|m(moh_class)) to start early media? Regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Early Media with Asterisk
Hi :) Am Donnerstag, 22. September 2005 12:48 schrieb Andreas Sikkema: > [EMAIL PROTECTED] wrote: > > Now, I traced RTP packets and see how sip2.provider1.de sends > > packets to my Asterisk but the port seems closed on my server so the > > inquiring server of > > provider1 will never get an answer and sends a "port unreachable". > > Did provider1 send the exact same SIP message types to you > as provider2? It looks to me like provider1 is not sending > a 183 Session Progress message. Which is usually used for > this kind of functionality I think. Oops! Mistake by myself: They start with INVITE (sure!) with all the SDP stuff. I answer 100 followed by 183. So I do send the 183 message to provider1 and provider2. Oh, I've forgotten to tell: My problem are incoming calls with early media from PSTN to my server. Sorry for that :) Regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Early Media with Asterisk
Hi :) I hope someone has a hint concerning Early Media. The situation: My Asterisk is connected to small local carrier who works with several SIP servers. I traced some SIP headers and find something like this: Via: SIP/2.0 UDP sip1.provider1.de In the SDP part I found something like this: o=- 2268929 0 IN IP4 sip2.provider1.de c=IN IP4 sip2.provider1.de If I send Dial(SIP/number|10|m(number)) I have silence on the line. No ringtone, nothing. Now contacting a friend whose Asterisk is connected to another provider (let's give him domain provider2) traced this: Via: SIP/2.0 UDP sip1.provider2.de and its SDP looks like this: o=- 2096205915 2096205915 IN IP4 sip1.provider2.de c=IN IP4 sip1.provider2.de and his early media works fine which means Dialing like the dial above works. The caller can listen to music :) Btw: I wrote hostnames because it's an example. Originally there are IP addresses in SDP part. Now, I traced RTP packets and see how sip2.provider1.de sends packets to my Asterisk but the port seems closed on my server so the inquiring server of provider1 will never get an answer and sends a "port unreachable". So I think Asterisk has a problem if another gateway than the original SIP server tries to connect. Is this correct? Hope someone has a hint, but maybe it is an error in my provider's routing or configuration? Thanks and kind regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 and Asterisk: Calls always hang up
Hi :)
I hope someone can help me (google cannot):
My little asterisk receives calls via h323 from PSTN. I connected a Sipura
phone to my asterisk. oh323 is installed and calls go into the right context
but immediately after the phone is picked up a hangup is signalled and the
call ends :(
This is what I get:
Inbound H.323 call 'ip$213.30.225.5:42873/1893' detected.
Channel OH323/[EMAIL PROTECTED] created and attached for inbound H.323
call 'ip$213.30.225.5:42873/1893'.
-- Executing NoOp("OH323/[EMAIL PROTECTED]", "h323 Call an
4999663-99!") in new stack
-- Executing Playback("OH323/[EMAIL PROTECTED]", "tt-monkeysintro")
in new stack
Channel OH323/[EMAIL PROTECTED] answered.
-- Playing 'tt-monkeysintro' (language 'en')
Call 'ip$213.30.225.5:42873/1893' cleared.
-- H.323 call 'ip$213.30.225.5:42873/1893' cleared, reason 24 (Call ended
with Q.931 cause)
Sep 14 10:30:42 WARNING[14895]: file.c:970 ast_waitstream: Unexpected control
subclass '5'
Call 'ip$213.30.225.5:42873/1893' with owner has already been cleared (2).
Call 'ip$213.30.225.5:42873/1893' has been hungup.
-- Hungup 'OH323/[EMAIL PROTECTED]'
Call 'ip$213.30.225.5:42873/1893' without owner has already been cleared (2).
Any ideas?
Thanks and kind regards,
Hauke
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for better "Follow Me"
Hi Michiel :)
Am Freitag, 2. September 2005 11:25 schrieb Michiel van Baak:
>
> We do something like this:
>
> exten => michiel,1,Dial(SCCP/michiel,15,t)
> exten =>
> michiel,2,Dial(SCCP/michiel&IAX2/vanbaak/${CELL_MICHIEL},40,t)
Aehhh...SCCP? Never heard about that...but I will google for this so thanks
for your answer :)
>
> Maybe thats easier then writing C code ;)
I hope so!
Regards,
Hauke
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for better "Follow Me"
Hi everybody :) I am a new member here and hope that someone gives me a hint for my problem: Let's say I am at work and my SIP phone (KPhone in my case) is connected to my private Asterisk. I want to call my wife at home so her SIP phone rings. She does not pick up the phone (maybe she is somewhere in the house and has to run to the phone) so after 15 seconds her cell phone should ring. Until now it is a classic "follow me" but what I want: I want both phones (SIP and cell) ringing and if one phone is picked up the other phone should stop ringing. My current idea: A C program is called via AGI. The program forks into two children. One of the children sleeps 15 seconds while the other drops a call file into /tmp and finally moves it into spool/outgoing. After 15 seconds the second child wakes up and also drops such a call file (but with another destination). Now the point is: How do I connect (or bridge) the source and the destination call? Another idea was using libagi (or something like that) but it seems that one child blocks stdin or stdout (don't know yet) so the DIAL command cannot send its result back to the child. I tried to understand app_queue.c and app_meetme.c from Asterisk source but unfortunately after reading the codes I was _really_ puzzled :( Thanks for a little hint :) Kind regards, Hauke ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
