Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
I agree 100%, it's too early. There is a lot of businnes out of there based on 1.4 (even still 1.2), and my feelings is that a lot of people is not going to upgrade the asterisk version, they are going to stay with 1.4 for a long time yet. Also i wanna add another little consideration. Voip is not only a software matter, is a Telecomunication matter. And into the Telecomunication world the first priority is the reliability and reliability and reliability without forget that usually the lifetime of a telecomunicaton product is much more than 4 years. I'm not a code writer so I can't put my effort in maintaince stuff. I think 1.4 should be open at least for some critical bug like for example segmentation fault or memory leack. Matteo Il 27/04/2011 21:34, Olle E. Johansson ha scritto: Friends, We have a discussion on asterisk-dev about the maintenance of the 1.4 branch. According to the release plans, support for 1.4 was scheduled to close in April 2011 - basically now. After that, only security patches would be committed. This is already a delay from the original plan published by Russell Bryant. Unfortunately, I think this is way too early. My feeling and experience is that 1.8 is not ready for production in the environments I work in - large scale installations. Customers are not planning migration and all new installs are still 1.4. Tests we've been doing with 1.8 has failed within just a short time and so badly that customers has not paid me to spend any further time with 1.8. Last time we went through this process with a LTS release (which we did not know then) it took over one year before we had a stable product to migrate away from 1.2 and jump on the 1.4 track. Hopefully, with the help of community, we can move up to 1.8 late this year or early next year. For me 1.8 is the focus, it's the LTS release. Not having a supported 1.4 version from the Digium-hosted repositories will mean that we will have to move to separate repositories or branch off from the main track. I already maintain a ton of subversion branches with various patches to 1.4 It takes a lot of time to manage this version that is a fork from the main 1.4 branch. I will soon have to start working with subversion branches for 1.8 to create a compatible version for my customers to test, since most of the patches is not part of 1.8. After a few years of doing this, I know the work involved with managing code myself. The Digium team wants to go ahead and not support 1.4 any more, I want to keep 1.4 open for normal bug fixes. What do you think? Kevin proposed that the community maintains the 1.4 branch without support from the Digium team. I don't think that's a good solution, but it may be the only solution. I haven't got the resources to manage the 1.4 code myself, so I won't step forward as a maintainer if I can't get proper funding. Anyone else out there that has the time and resources to manage the code? Feel free to send me mail off list if you have ideas or suggestions on how to solve this - or continue the discussion here. Regards, /Olle PS. Please don't start a discussion about 1.8 quality in this thread, that's a separate issue. I just want to know what you think about closing 1.4 support now. If you want to discuss 1.8 quality, start a new thread. Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Unable to set callerid for incoming skype calls
HI,
I'm using the usual Set(Callerid(num) function to change the incoming
from skype callerid but it's not working.
Asterisk 1.4.31 and last release of skype channels
This is the dialplan
exten = _0X.,1,NoOP(${CALLERID(num)} - ${CALLERID(name)})
exten = _0X.,n,Set(STRINGA=Skype)
exten = _0X.,n,NoOP(${STRINGA})
exten = _0X.,n,Set(CALLERID(num) = ${STRINGA})
exten = _0X.,n,NoOP(${CALLERID(num)} - ${CALLERID(name)})
and is the output
NoOp(Skype/lab.trentinonetwork.it-08a32278, mapiazza - Matteo
Piazza) in new stack
-- Executing [0461020...@dial-to-openser:2]
Set(Skype/lab.trentinonetwork.it-08a32278, STRINGA=Skype) in new stack
-- Executing [0461020...@dial-to-openser:3]
NoOp(Skype/lab.trentinonetwork.it-08a32278, Skype) in new stack
-- Executing [0461020...@dial-to-openser:4]
Set(Skype/lab.trentinonetwork.it-08a32278, CALLERID(num) = Skype) in
new stack
-- Executing [0461020...@dial-to-openser:5]
NoOp(Skype/lab.trentinonetwork.it-08a32278, mapiazza - Matteo
Piazza) in new stack
As you can see the incoming callerid(num) mapiazza doesn't change.
Is there any limitation on skype channel ?
Matteo
--
==
Ing. Matteo Piazza
Trentino Network s.r.l.
Area Ricerca Sviluppo
Via Gilli, 2 - 38121 TRENTO
Tel (+39) 0461.020224
Mob (+39) 335.5378482
Fax (+39) 0461.020201
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Società soggetta a direzione e controllo da parte della Provincia
Autonoma di Trento. C.F. e P. IVA 00337460224
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[asterisk-users] SRV DNS failover - dial to proxy list
I'm looking for some documentationof the implementation of the SRV DNS on asterisk. I'm using asterisk 1.4.22. 1) From voip-info.org: Currently, Asterisk only reads the first SRV entry without bothering with priorities and weights. IS It still true? from my first tests I can see that asterisk is able to manage more than only the first SRV 2) I wanna configure the asterisk in this way: if the proxy with the highest priority is not reachable, IT tries to dial through the next proxy defined in the zone with a SRV record. In my dialplan I have: exten = _0461.,1,Dial(SIP/[EMAIL PROTECTED]) the asterisk recieve this answer at the DNS query Domain Name System (response) [Request In: 1] [Time: 0.000352000 seconds] Transaction ID: 0x989a Flags: 0x8580 (Standard query response, No error) Questions: 1 Answer RRs: 2 Authority RRs: 1 Additional RRs: 3 Queries _sip._udp.gorizia.tnet.it: type SRV, class IN Name: _sip._udp.gorizia.tnet.it Type: SRV (Service location) Class: IN (0x0001) Answers _sip._udp.gorizia.tnet.it: type SRV, class IN, priority 0, weight 0, port 5060, target load.gorizia.tnet.it _sip._udp.gorizia.tnet.it: type SRV, class IN, priority 1, weight 0, port 5060, target failover.gorizia.tnet.it Authoritative nameservers gorizia.tnet.it: type NS, class IN, ns ns1.gorizia.tnet.it Additional records load.gorizia.tnet.it: type A, class IN, addr 172.25.18.68 failover.gorizia.tnet.it: type A, class IN, addr 172.25.18.65 ns1.gorizia.tnet.it: type A, class IN, addr 172.25.18.68 From the asterisk console I have: [2008-10-08 13:34:15] -- Executing [EMAIL PROTECTED]:2] Dial(SIP/tnn2678-08277a20, SIP/[EMAIL PROTECTED]) in new stack [2008-10-08 13:34:15] -- ast_get_srv: SRV lookup for '_sip._udp.gorizia.tnet.it' mapped to host load.gorizia.tnet.it, port 5060 [2008-10-08 13:34:15] -- Called [EMAIL PROTECTED] the load.gorizia.tnet.it is not reachable but asterisk is stuck here. Any one have any clue? Thanks for your time. Matteo ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Max Number of Extensions
Hi, I have a problem with the extensions file, asterisk load only 412 extensions. I use 1.2.4. version of Asterisk, I have try to modify the extensions but asterisk finish to load the extensions when arrived at the 412th extension. Matteo Piazza ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with poud key (#)
I not understand why my asterisk send the tone of pound key (#) only when i click twice time. I deactivate the transfer function. Matteo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-941 hint
Hi Have someome a solution to use the hint function to have the signalling of the status of a extension on the SPA-941 phone ? Matteo ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] double ringing tone on asterisk 1.2 (workaround)
You must change in the indication.conf the country [general] country=it ; default location Simone Cittadini wrote: After reading a description of apparently the same problem by Juan J. Sierralta more detailed than mine tuuu tuuu instead of tuuu we've solved the problem changing the call progress tone of sip phones to something not udible. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] messages of Mobile Operator
Hi,
I have this problem. When I call a GSM number with the IDSN line if the
GSM phone I not hear the messages of operator but I hear the ring.
I suppose that the problem is that asterisk waits a response that is yet
arrived.
Any idea?
This is my extension:
exten = _0XX.,1,Wait,1
exten = _0XX.,2,Dial(Zap/g1/${EXTEN:1},60,tT)
exten = _0XX.,3,Hangup
Matteo
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[Asterisk-Users] remove asterisk?
Is there a command to remove completely asterisk? I want clean the server before the installation of 1.2 version. Matteo ___ --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] remove asterisk?
Are you sure that with make unistall all asterisk's three is cancelled? Martin Vit wrote: make uninstall? Matteo Piazza wrote: Is there a command to remove completely asterisk? I want clean the server before the installation of 1.2 version. Matteo ___ --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 ___ --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 -- === Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: [EMAIL PROTECTED] Tel: +39-0461-408400ext:308 www.create-net.it === ___ --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] MeetMe invite another user
If I use meetme conference room, can I invite another user during a conversation? In which way? Matteo === Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: [EMAIL PROTECTED] Tel: +39-0461-408400ext:308 www.create-net.it === ___ --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
