RE: [Asterisk-Users] Presence + Eyebeam + Asterisk 1.2
Don't waste your time asterisk does not support presence --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit : Hi, anyone managed to get a Presence Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe this should be paritally supported now in 1.2 ? regards Mark ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Re: Presence + Eyebeam + Asterisk 1.2
I'm sure look at rfc3265 (SUBSCRIBE/NOTIFY) which is not support by asterisk. How can you monitor the states of the buddies ? Harry --- Ben Buxton [EMAIL PROTECTED] a écrit : Are you sure? I've got it working with Eyebeam, showing me just who is available and who isn't. http://www.voip-info.org/wiki-Asterisk+phone+snom A couple of pages down you'll see this: SNOM SUBSCRIBE/NOTIFY support for monitoring extension states The methods and configuration here are also valid for Eyebeam. BB harry gaillac [EMAIL PROTECTED] uttered the following thing: Don't waste your time asterisk does not support presence --- Mark van Kerkwyk [EMAIL PROTECTED] a écrit : Hi, anyone managed to get a Presence Agent configuration with Asterisk 1.2 and X-Ten Eyebeam working. I believe this should be paritally supported now in 1.2 ? regards Mark ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Asterisk dial plan
Hello, When asterisk receive a registration with a private address is it possible to forward the sip request for this agent to a sip proxy ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] harry's project
Hello, I need SER for IM/presence and sip routing. Harry --- Jonathan k. Creasy [EMAIL PROTECTED] a écrit : http://www.automated.it/guidetoasterisk.htm I don't think you even require SER in that case. That will be $100. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac Sent: Thursday, November 24, 2005 7:11 PM To: users@openser.org; asterisk-users@lists.digium.com Subject: [Asterisk-Users] harry's project Hello, here is an other diagram for people who don't yet understand what i expect to do. Look at sip_call_flow.png file i wish to substitute ondo sip server with ser and ondo pbx with asterisk . ondo sip server is able to do far-end near-end nat I guess ser too. I do hope i will find some people who help me to configure that . Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Asterisk doesn't start
Hello, You built asterisk on freebsd ? Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Hello Whan starting astersik(1.2) (asterisk -vvc), I get this message : [res_config_mysql.so] = (MySQL RealTime Configuration Driver) /libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so: Undefined s ymbol ast_config_load What did I forgot to do? Olivier ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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: RE : [Asterisk-Users] Asterisk doesn't start
Try to post your problem to asterisk-dev I guess they could solve or explain this problem better than asterisk'users . Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Yes, beta2 works perfectly, but 1.2 released version gives me this error. Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de harry gaillac Envoyé : vendredi 25 novembre 2005 11:24 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [Asterisk-Users] Asterisk doesn't start Hello, You built asterisk on freebsd ? Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Hello Whan starting astersik(1.2) (asterisk -vvc), I get this message : [res_config_mysql.so] = (MySQL RealTime Configuration Driver) /libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so: Undefined s ymbol ast_config_load What did I forgot to do? Olivier ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] GUI and Asterisk Realtime
Hello, Is there a GUI to manage sip users and voicemail with Asterisk Realtime . Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] What does it mean?
Hello, Read the Makefile in apps. Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Hello, I have compiled asterisk cvs under freebsd, no problems. When starting asterisk, I get : [res_config_mysql.so] = (MySQL RealTime Configuration Driver) /libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so: Undefined symbol ast_config_load What's wrong? Olivier ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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: RE : [Asterisk-Users] What does it mean?
Je ne connais pas la signification de sybillines. Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Tes réponses sont aussi sybillines que tes questions :) Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de harry gaillac Envoyé : jeudi 24 novembre 2005 16:45 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [Asterisk-Users] What does it mean? Hello, Read the Makefile in apps. Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Hello, I have compiled asterisk cvs under freebsd, no problems. When starting asterisk, I get : [res_config_mysql.so] = (MySQL RealTime Configuration Driver) /libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so: Undefined symbol ast_config_load What's wrong? Olivier ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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: RE : RE : [Asterisk-Users] What does it mean?
Merci pour ces précisions. Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : SIBYLLIN, INE. adj. Qui appartient aux sibylles. Il n'est guère usité au sens propre que dans ces locutions : Les oracles, les livres, les vers sibyllins, Les oracles, les livres, les vers des sibylles. Il signifie au figuré Qui est mystérieux obscur, dont le sens est difficile à saisir. Il m'a répondu en termes sibyllins. Des paroles sibyllines. Un langage sibyllin. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de harry gaillac Envoyé : jeudi 24 novembre 2005 17:54 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: RE : [Asterisk-Users] What does it mean? Je ne connais pas la signification de sybillines. Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Tes réponses sont aussi sybillines que tes questions :) Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de harry gaillac Envoyé : jeudi 24 novembre 2005 16:45 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [Asterisk-Users] What does it mean? Hello, Read the Makefile in apps. Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Hello, I have compiled asterisk cvs under freebsd, no problems. When starting asterisk, I get : [res_config_mysql.so] = (MySQL RealTime Configuration Driver) /libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so: Undefined symbol ast_config_load What's wrong? Olivier ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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: RE : RE : [Asterisk-Users] What does it mean?
Je ne donne pas de réponse ! Il me semble t'avoir suggèrer asterisk comme système de messagerie vocale au lieu d'SEMS, avoir fourni quelques fichiers de configuration, ce n'étaient pas des devinettes. Conbien de fois on ma répondu personne n'est obligé de faire ton tavail, tu n'as qu'a payé pour ce que tu demandes. IL me semble même me souvenir avoir lu un développeur te faire la remarque les utilisateurs de nos projets vous ne profitez que de notre travail !. Pour répondre à ton problème configure logger.conf . Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Cela veut simplement dire que tu te plains de ne pas avoir de réponses, mais qu'en fait tu n'en donnes pas non plus, sauf sous forme de devinette. Auquel cas, il est plus simple de ne pas répondre, merci -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de harry gaillac Envoyé : jeudi 24 novembre 2005 17:54 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: RE : [Asterisk-Users] What does it mean? Je ne connais pas la signification de sybillines. Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Tes réponses sont aussi sybillines que tes questions :) Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de harry gaillac Envoyé : jeudi 24 novembre 2005 16:45 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : RE: [Asterisk-Users] What does it mean? Hello, Read the Makefile in apps. Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Hello, I have compiled asterisk cvs under freebsd, no problems. When starting asterisk, I get : [res_config_mysql.so] = (MySQL RealTime Configuration Driver) /libexec/ld-elf.so.1: /usr/lib/asterisk/modules/res_config_mysql.so: Undefined symbol ast_config_load What's wrong? Olivier ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] harry's project
Hello, here is an other diagram for people who don't yet understand what i expect to do. Look at sip_call_flow.png file i wish to substitute ondo sip server with ser and ondo pbx with asterisk . ondo sip server is able to do far-end near-end nat I guess ser too. I do hope i will find some people who help me to configure that . Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.comattachment: sip_call_flow.jpg ___ --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] RE: [Serusers] Re: open letter
Doug, You have ever post this mail. Harry Others have tried to explain it too you, but I don't think you fully understand. Maybe it is a language issue. Your follow-up posts come across as demanding. When I read your posts, I feel like you are criticizing people for not having responded to you. It is like you feel they have done something wrong. This probably isn't what you mean, but that is how it seems. ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] open letter (2)
Dear users, This letter is addressed to the most experienced users for the ser openser and asterisk projects. Advice me and I'll stop to mail my question. How a session between two user agents behind nat could stay in the path ? Harry Kinds Regards |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Re: [Users] open letter (2)
Hi Klaus, Please do not cross post. Split your problems into smaller problems and ask them on the correspondig list. I mail my question to asterisk, openser ser lists After all your emails, I still have no glue what your scenario is. Why do you want to host ser+asterisk+NAT on the same device? pass through I agree my english is not very good sorry i try my best . Asterisk don't provide IM/presence unlike ser however ser don't provide telephony features like MOH ACD call parked IVR and more I want my sip agents to provide these features. Ser handle sip routing asterisk telephony features . Should the Asterisk/ser be reachable also from the public interface? If not, why do you need NAT traversal at all? In fact i have got a single machine for my tests . Ser handle sip routing so incoming or outgoing requests pass through SER not directly to asterisk . I need nat support for sip agents behind nat. Why do you use both? Asterisk can also do NAT traversal. For how many users is the setup? I think asterisk support 255 users klaus harry gaillac wrote: Dear users, This letter is addressed to the most experienced users for the ser openser and asterisk projects. Advice me and I'll stop to mail my question. How a session between two user agents behind nat could stay in the path ? Harry Kinds Regards |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Re: [Users] open letter (2)
--- Klaus Darilion [EMAIL PROTECTED] a écrit : Hi Harry! As this emails are on-topic you should cc: to the list. harry gaillac wrote: In fact the problem is in contact sip header field (private ip) agent send ReGISTER to SER (outbound proxy) which one send REGISTER to ASTERISK . Asterisk register agent with AOR sip:[EMAIL PROTECTED] ip When agent send INVITE to an other agent ASTERISK use AOR sip:[EMAIL PROTECTED] ip but the firewall don't allow this Asterisk SHOULD resend INVITE to SER. Does SER is able to rewrite contact field in SIP HF? Which IPaddress:port do you want to have in the REGISTER's Contact: header sent from ser to Asterisk? in fact i wish to replace all private ip in the contact field with the public ip of ASTERISK Harry klaus Regards Thanks for your advices Harry --- Klaus Darilion [EMAIL PROTECTED] a écrit : harry gaillac wrote: Have you ever used SIP clients with presence and IM? I suggest to setup ser (without Asterisk) just to test the IM features. SIP based IM/presence implementations are very poor yet. I've done it And what were your experiences? Which clients do you use? Polycom IP300 In your picture, the NAT router is on the same PC as ser and asterisk. Is this correct? this is correct It would be a good idea to split things. This is a rather complicated setup. what scenario do you have? Are all the users behding the same NAT (in the same subnet) and you provide VoIP within this network (e.g. an enterprise) or do you have external users (e.g. like iptel or freeworlddialup)? in fact both asterisk+ser private net=nathelper ==nat===private net nat box || internet== I suggest: 1. Asterisk, ser and the RTP proxy 8rtpproxy or mediaproxy) should listen only on the public interface (this really must be a routable public IP address, no private). SER asterisk listen on public ip 2. Setup the firewall (e.g. iptables) correctly to allow traffic from/to ser, asterisk and the RTP proxy Done 3. setup ser according the getting started document on onsip.org. AFAIK this document contains hints how to route to a gateway. Reuse this part of the config to route certain calls to the asterisk box. Done 4. Try to solve things step by step: - REGISTER should work fine from Internet and LAN - Calls from Internet clients to Internet clients - Calls from LAN clients to LAN clients - Calls from LAN clients to Internet clients (and vice versa) - now try to add asterisk, e.g. calling a certain number will be routed to asterisk and starts the echo application If all the above works (DO NOT start integrating the asterisk as long as basic SIP call do not work!), you can implement your setup. 5. Do really read every word in the getting started document, if things are unclear read it again. 6. Do not post how to make this setup. Ask small questions addressing particular (small) problems. 7. Post to the related list. - do not post to developer lists - if you use ser, post to ser's list - if you use openser, post to openser's list - if you have an asterisk problem, ask at the asterisk list (e.g. you want to solve NAT traversal and registration with ser. Thus, do not ask this kind of questions at the asterisk list). 8. always remember that this support is voluntary 9. If you don't find the proper english word, look into the dictionary instead of using another word which might also have other meanings. 10. Go and buy an english SIP book. (this will you help to learn the english terms for all the SIP stuff) 11. use ngrep to watch the SIP call flow # ngrep -t -d any port 5060 regards klaus === message truncated === ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] open letter (2)
may be you I agree --- Patrick [EMAIL PROTECTED] a écrit : On Wed, 2005-11-23 at 10:34 +0100, harry gaillac wrote: Advice me and I'll stop to mail my question. That almost sounds like a threat. Do you really think you motivate people to answer you this way? Since you asked this question already so many times perhaps it's time to hire a paid consultant. ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] open letter (2)
What are your prices Harry --- harry gaillac [EMAIL PROTECTED] a écrit : may be you I agree --- Patrick [EMAIL PROTECTED] a écrit : On Wed, 2005-11-23 at 10:34 +0100, harry gaillac wrote: Advice me and I'll stop to mail my question. That almost sounds like a threat. Do you really think you motivate people to answer you this way? Since you asked this question already so many times perhaps it's time to hire a paid consultant. ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] open letter (2)
You should read my mail so you would have an idea of my problem !!! Harry --- Patrick [EMAIL PROTECTED] a écrit : On Wed, 2005-11-23 at 14:36 +0100, harry gaillac wrote: What are your prices Don't have any since I have no idea what your problem is and how to solve it so I can't help you. Looking at the rates/pricing that were mentioned on the lists and elsewhere in the past I guess you can expect to pay around â¬100/hour for a good consultant. ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] open letter (2)
my name is gaillac not giallac Harry --- Steve Totaro [EMAIL PROTECTED] a écrit : New rule for email Sender = harry giallac = deleted -Original Message- From: harry gaillac [mailto:[EMAIL PROTECTED] Sent: Wednesday, November 23, 2005 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] open letter (2) may be you I agree --- Patrick [EMAIL PROTECTED] a écrit : On Wed, 2005-11-23 at 10:34 +0100, harry gaillac wrote: Advice me and I'll stop to mail my question. That almost sounds like a threat. Do you really think you motivate people to answer you this way? Since you asked this question already so many times perhaps it's time to hire a paid consultant. ___ --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 __ _ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] open letter (2)
You've now been asking the same questions on about 5 lists (we're all on) and it doesn't help your cause. three lists. Why do you think i sent and resent my posts just for playing ? This (and the other) lists are free resources provided by the community. Have a look on the wiki (www.voip-info.org) for Asterisk/SER consultants and if you're lucky you might find someone who isn't subscribed to the lists and therefore may help you. I think Consultants have subscribed to these lists They could tell me we have the solution, here is the price Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] open letter
Could you tell me more please ? You understand than with host=dynamic in sip.conf asterisk use contact field in SIP HF Regards Harry --- [EMAIL PROTECTED] [EMAIL PROTECTED] a écrit : Sounds to me as what you want to do require 'a few' code changes to Asterisk. Maybe I am wrong, but this might take some work to get right. Jan harry gaillac wrote: Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] open letter
Could you tell me more please ? You understand than with host=dynamic in sip.conf asterisk use contact field in SIP HF Regards Harry --- [EMAIL PROTECTED] [EMAIL PROTECTED] a écrit : Sounds to me as what you want to do require 'a few' code changes to Asterisk. Maybe I am wrong, but this might take some work to get right. Jan harry gaillac wrote: Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] hello
hello ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] [Asterisk-Dev] hello
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[Asterisk-Users] outbound sip proxy
Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] sip routing
Hello, Can we configure asterisk in order to send sip requests to a outbound proxy when asterisk get AOR of users agents with an private ip ? Asterisk AOR:[EMAIL PROTECTED] ip | | sip proxy/nat box---user agent 192.168.0.0/24 Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] open letter
Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Re: [Serusers] open letter
You lost me here. Was that a question or a statement? I might not be able to help, since my SER usage is totally diffent, but let me see if I got this right: - You want the SER to forward REGISTER messages to the Asterisk. - The user agents use private IP addresses. - You want the SER to perform NAT? (I'm guessing here) How a session between two user agents behind nat could keep in the path That is the question Since you a talking of a session, do you talk of calls now? yes Could you perhaps post the parts of ser.cfg that deal with register requests? I added this in register block rewritehostport(nxs.yi.org:5050); t_relay_to_udp(nxs.yi.org,5050); 5050 = asterisk port Regards, Stefan ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Re: [Users] open letter
Let me get this straight All you are doing is registering the devices with SER (below you have mentioned asterisk, and then you say they goto ser) No to asterisk. Asterisk should handle INVITE, REGISTER via ser. SER should handle IM/presence Once they are registered to ser you wish to send them to asterisk...is this correct If so, this does not seem to hard, NAT ius dealt with in ser, I use mediaproxy, others may use nathelper, so before you send to asterisk take care of NAT issues in SER and then send to asterisk. Paste config, in pastebin, and also a ngrep of the call debug. Iqbal harry gaillac wrote: Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users . ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Re: [Users] open letter
okay, so ALL your users are registering to asterisk...is that correct. Correct via ser as outbound sip proxy If so the problem is howto accept users from behind a NAT into asterisk, or am I confusing things further. the problem is in contact field. when user agents send register we have in sip hf Contact sip:[EMAIL PROTECTED] So asterisk store this AOR and try to contact agent via nat box instead of SER If the above are true, where is SER in this, or are users hitting SER and you are sending the REGISTER from ser into asterisk. SER is an outbound sip proxy which handle IM presence nat Harry One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private | ---- |-- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Re: [Users] open letter
In fact ser should keep nat opened of ua behind nat. Ser just need to keep location for im an presence Asterisk forward requests according to contact field to ser. --- Iqbal [EMAIL PROTECTED] a écrit : Okay, so get ser to fix the NAT part before sending to asterisk. Any is ser just proxying all register commands, why not register in ser, than asterisk, I know you are doing IM in asterisk, and I havent done that, Asterisk do not support IM/presence. but I am using asterisk for features like call pickup and transfer, they might be different in operation but I think its best to find out howto let ser do all the hardwork and let asterisk only work when it needs to. They can work together ! thanks for help harry harry gaillac wrote: not exactly ! something like this : asterisk | ser ua1| | ua2 ua1 and ua2 send registration to asterisk via ser . when ua1 invite ua2 sip INVITE is sent to ser which one forward it to asterisk. asterisk lookup in its AORs so it bridge the call and send INVITE to ua2 via ser. Harry --- Iqbal [EMAIL PROTECTED] a écrit : Okay almost there :-) So UA --- asterisk --- SER --- UA is that it harry gaillac wrote: okay, so ALL your users are registering to asterisk...is that correct. Correct via ser as outbound sip proxy If so the problem is howto accept users from behind a NAT into asterisk, or am I confusing things further. the problem is in contact field. when user agents send register we have in sip hf Contact sip:[EMAIL PROTECTED] So asterisk store this AOR and try to contact agent via nat box instead of SER If the above are true, where is SER in this, or are users hitting SER and you are sending the REGISTER from ser into asterisk. SER is an outbound sip proxy which handle IM presence nat Harry One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private | ---- |-- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com . ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com . ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] RE: [Serusers] open letter
hello, Give me your price to enable my diagram ASAP --- harry gaillac [EMAIL PROTECTED] a écrit : Hello open(ser) asterisk users Here is what i expect to do : Asterisk: registrar with public ip port=5050 open(ser): outbound proxy with public ip port=5060 Asterisk don't support IM and presence so i want to use SER because of it's a good proxy: I want user agents behind nat send registration to asterisk because of it's an ipbx :-) Look at this diagram when user agent behind nat send REGISTER to ser the contact field in sip header has a private address which one is forward to asterisk for registration. When user agent are registered in asterisk AOR is sip:[EMAIL PROTECTED] ip so asterisk query sip:[EMAIL PROTECTED] behind nat (not possible). How a session between two user agents behind nat could keep in the path |register || register | agent1 asterisk| |ser/nat box || | 200 OK ||200 OK | agent2 One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Send me your questions if you don't understand what i expect to do . Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Can not build zaptel with kernel-2.6.12
Hello, I try to compile zaptel . I installed kernel-sources but when i run : make linux26 / serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL _CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL _CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm ir You do not appear to have the sources for the 2.6.12-1-386 kernel installed. make: *** [linux26] Error 1 // Something don't match in makefile with debian sarge 3.1 here linux26: prereq $(BINS) @echo $(KSRC) @if [ -z $(KSRC) -o ! -d $(KSRC) ]; then echo You do not appear to have the sources for the $(KVERS) kernel installed.; exit 1 ; fi $(MAKE) -C $(KSRC) SUBDIRS=$(PWD) modules Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Can not build zaptel with kernel-2.6.12
Hello David, I rewrote the Makefile so I can compile the modules . However I got the same problems with kernel 2.4.I fixed some variables which was not found . Is it a problem with my debian installation !!??? Regards Harry PS: I like to set ! for Mr Pascal :-) --- David Uzzell [EMAIL PROTECTED] a écrit : harry gaillac wrote: Hello, I try to compile zaptel . I installed kernel-sources but when i run : make linux26 / serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL _CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL _CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm ir You do not appear to have the sources for the 2.6.12-1-386 kernel installed. make: *** [linux26] Error 1 // I have to ask the obvious question. Do you have the same source as you have kernel running? Remember if you have run an upgrade it could have updated the kernel but may not have doen the sources and if you have the sources from the installion media then you would have different versions that will cause this exact problem. David Something don't match in makefile with debian sarge 3.1 here linux26: prereq $(BINS) @echo $(KSRC) @if [ -z $(KSRC) -o ! -d $(KSRC) ]; then echo You do not appear to have the sources for the $(KVERS) kernel installed.; exit 1 ; fi $(MAKE) -C $(KSRC) SUBDIRS=$(PWD) modules Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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: RE : [Asterisk-Users] Can not build zaptel with kernel-2.6.12
Hello Olivier, Non je ne suis pas fâché ! Alors ce *b2bua ? En fait je cherche une solution pour intègrer SER+Asterisk sur la même machine. Ser est un bon proxy asterisk un bon ipbx. Je souhaite utilisé ser pour le routage sip avec asterisk et pour fournir les service de téléponie d'entreprise plus l'IM et presence via SIMPLE qu'asterisk ne propose pas ! Mon problème est le champ contact dans le Sip HF avec des clients natés Une idée ? Harry --- Olivier Taylor [EMAIL PROTECTED] a écrit : Salut Harry, plus de nouvelles de toi :( Serais tu faché? Olivier -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de harry gaillac Envoyé : lundi 21 novembre 2005 13:34 À : asterisk-users@lists.digium.com Objet : [Asterisk-Users] Can not build zaptel with kernel-2.6.12 Hello, I try to compile zaptel . I installed kernel-sources but when i run : make linux26 / serveur1:/usr/local/src/ASTERISK/zaptel-1.2.0# make linux26 cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\makefw.c -o makefw ./makefw tormenta2.rbt tor2fw tor2fw.h Loaded 69900 bytes from file ./makefw pciradio.rbt radfw radfw.h Loaded 42096 bytes from file cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o ztcfg.o ztcfg.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL _CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o zonedata.lo zonedata.c cc -c -fPIC -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL _CONFIG=\/etc/zaptel.conf\ -DBUILDING_TONEZONE -o tonezone.lo tonezone.c ar rcs libtonezone.a zonedata.lo tonezone.lo cc -o ztcfg ztcfg.o libtonezone.a -lm cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o torisatool.o torisatool.c cc -o torisatool torisatool.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o ztmonitor.o ztmonitor.c cc -o ztmonitor ztmonitor.o cc -o ztspeed.o -c ztspeed.c cc -o ztspeed ztspeed.o cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\zttest.c -o zttest cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\ /etc/zaptel.conf\ -c -o fxotune.o fxotune.c cc -o fxotune fxotune.o -lm ir You do not appear to have the sources for the 2.6.12-1-386 kernel installed. make: *** [linux26] Error 1 // Something don't match in makefile with debian sarge 3.1 here linux26: prereq $(BINS) @echo $(KSRC) @if [ -z $(KSRC) -o ! -d $(KSRC) ]; then echo You do not appear to have the sources for the $(KVERS) kernel installed.; exit 1 ; fi $(MAKE) -C $(KSRC) SUBDIRS=$(PWD) modules Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Contact field in SIP HF between asterisk + ser
___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ---End Message--- ___ --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] Contact field in SIP HF between asterisk + ser
___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ---End Message--- ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ---End Message--- ___ --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] Contact field in SIP HF between asterisk + ser
Remarque : message transféré en pièce jointe. ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ---End Message--- ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ---End Message--- ___ --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] Contact field in SIP HF between asterisk + ser
Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be adviced. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Contact field in SIP HF between asterisk + ser
Hello, Here is my config : Asterisk as registrar server :public ip:5050 Ser as outbound proxy server :public ip 5060 I wish ser to handle the packets between Nat box (netfilter) and Asterisk However contact field in the sip HF don't change from nat box to asterisk which don't allow to keep the sessions via SER . Ser receive packets with private ip in contact field which one is forward to asterisk . How ser can handle the contact field to establish sip sessions between sip agents and asterisk ? I've been trying mangle and textops modules but i really need to be advice. One box --- | | | | asterisk pbx | | | | ||| | | ---- | | SER ||NAT box | private network | ---- --- Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Polycom Buddy Feature
However, it doesn't work consistently. Sometimes it does, and sometimes it doesn't. There's a thread on the asterisk-dev list titled chan_exosip2 where I am discussing my problems with Olle. Yes i posted the chan_exosip2 thread ! Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Polycom Buddy Feature
Do you configure BLA bridged line appearence for presence in asterisk ? Harry --- Kevin Hanson [EMAIL PROTECTED] a écrit : Michael Araba wrote: I am having the same problems. The polycom phones the 501 or 601 or 301 will list more more than 7 buddies neither will the 601 with an expansion module monitor more than 7 other phones. Is there anyone out there who can explain waht is happening. My reseller can not help. I am surprised no one has reported the reason for the problem or or even a word from the manafacturer. Please someone our ther help. the phones are great but this is a big issue maraba From an earlier thread (9/21/05) Kevin Fleming said: The only issue today with displaying hint status is an artificial limit of eight (8) 'buddies' in the Contact Directory to watch. Once Polycom has released the final firmware for the phone with support for a larger number of watched contacts, the expansion module will be fully usable with Asterisk. My 601/expansion module only allows me 7 buddies. I don't know when Polycom will update the firmware. I am going to contact my supplier and see if they can find out. I'll post my results. Cheers, Kevin - Optimacy Communications, LLC http://www.optimacycomm.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Polycom Buddy Feature
you mean than when the status of your subscribers change they are notified busy, away, ... harry --- Kevin Hanson [EMAIL PROTECTED] a écrit : harry gaillac wrote: Do you configure BLA bridged line appearence for presence in asterisk ? Harry I am using hints in extensions.conf in asterisk combined with buddy lists / buddy watch on the Polycoms. It works pretty well in Asterisk 1.2. I'm having a couple of issues that are outlined in a recent thread in the asterisk-dev list. Cheers, Kevin - Optimacy Communications, LLC http://www.optimacycomm.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Polycom Buddy Feature
you mean than when the status of your subscribers change they are notified busy, away, ... harry --- Kevin Hanson [EMAIL PROTECTED] a écrit : harry gaillac wrote: Do you configure BLA bridged line appearence for presence in asterisk ? Harry I am using hints in extensions.conf in asterisk combined with buddy lists / buddy watch on the Polycoms. It works pretty well in Asterisk 1.2. I'm having a couple of issues that are outlined in a recent thread in the asterisk-dev list. Cheers, Kevin - Optimacy Communications, LLC http://www.optimacycomm.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Polycom Buddy Feature
Hello, Can you monitor the buddies status with you polycom phones ? Harry --- Michael Araba [EMAIL PROTECTED] a écrit : I am having the same problems. The polycom phones the 501 or 601 or 301 will list more more than 7 buddies neither will the 601 with an expansion module monitor more than 7 other phones. Is there anyone out there who can explain waht is happening. My reseller can not help. I am surprised no one has reported the reason for the problem or or even a word from the manafacturer. Please someone our ther help. the phones are great but this is a big issue maraba ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Errors With Hint
Hello How do you configure Polycom for presence please ? Harry --- Alvaro Parres [EMAIL PROTECTED] a écrit : Hi list, i have the next problem: I create 3 hints.. (111 (SIP/111), 112 (SIP/112), and 102 (ZAP/35) ) the SIP/111 is a GrandStream ATA the SIP/112 is a Polycom 301 the ZAP/35 is a Analogic Phone. The SIP/112 hints works great. But the other 2 no. The ZAP/35 is say is always in USE and as you see en the next console output is not in use. any Idea asterisk*CLI -= Registered Asterisk Dial Plan Hints =- 111 : SIP/111 State:Idle Watchers 4 102 : ZAP/35 State:InUse Watchers 5 112 : SIP/112 State:InUse Watchers 2 - 3 hints registered asterisk*CLI show cha channel channels channeltypes asterisk*CLI show channels Channel Location State Application(Data) Zap/34-1 [EMAIL PROTECTED]:1 Up Bridged Call(SIP/112-1f3d) SIP/112-1f3d [EMAIL PROTECTED]: Up Dial(ZAP/34/3338182842|120|Tt) 2 active channels 1 active call And also the SIP/111 is always in Idle any idea of why ??? thanks ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] receive fax with asterisk
http://www.hylafax.org/ Harry --- Doug Lytle [EMAIL PROTECTED] a écrit : Jason Brashear wrote: Receiving faxes with Asterisk. Is there a good resource for learning how to set this up? www.soft-switch.org Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] IM / presence asterisk-1.2-RC1
Sorry, Here are some files Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : This is good debugging info you've listed below, but this isn't a sip debug/trace. To do that, first verify in your logger.conf file you have the following line: full = notice,warning,error,debug,verbose Then, if you needed to add anything to logger.conf, please first restart Asterisk so those new settings take effect. Then, from the CLI issue set verbose 5 and set debug 5 and finally sip debug. The repeat your dialing steps. The sip debug/trace will then be contained in /var/log/asterisk/full if /var/log/asterisk is where your log files are kept. With that, we can have a better idea of what's happening/not happening to give you the issue you're having. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: I did it !? // Connected to Asterisk 1.2.0-rc1 currently running on serveur1 (pid = 1125) Verbosity is at least 4 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 f1682d8d-8f 84 Idle xpidf+xml 192.168.0.21 86 5f32aec-95b 85 Idle xpidf+xml 192.168.0.20 84 cb424ae1-e4 86 Idle xpidf+xml 192.168.0.20 84 715fac66-a9 87 Idle xpidf+xml 4 active SIP subscriptions serveur1*CLI // serveur1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 87/87 192.168.0.21 D N 5060 OK (84 ms) 86/86 192.168.0.21 D N 5060 OK (97 ms) 85/85 192.168.0.20 D N 5060 OK (87 ms) 84/84 192.168.0.20 D N 5060 OK (96 ms) 4 sip peers [4 online , 0 offline] serveur1*CLI /// my sip.conf: [general] context=local ; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=nxs.yi.org; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=1000 ; Default length of incoming/outoing registration allow=all ; First disallow all codecs musicclass=default ; Sets the default music on hold class for all SIP calls language=fr ; Default language setting for all users/peers rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity tpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity useragent=Asterisk PBX ; Allows you to change the user agent string dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com messages.serveur1.home.net Description: 1676272990-messages.serveur1.home.net debug.serveur1.home.net Description: 3484436676-debug.serveur1.home.net ___ --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] IM / presence asterisk-1.2-RC1
Here are some other files. Why asterisk send sip OPTION message to agents ? Harry 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x81cf940 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted Retransmitting #2 (NAT) to 192.168.0.20:5060: OPTIONS sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.119.11.222:5060;branch=z9hG4bK4a119599;rport From: asterisk sip:[EMAIL PROTECTED];tag=as747a6ef0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Nov 2005 10:23:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8194ea0 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted /// --- harry gaillac [EMAIL PROTECTED] a écrit : Sorry, Here are some files Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : This is good debugging info you've listed below, but this isn't a sip debug/trace. To do that, first verify in your logger.conf file you have the following line: full = notice,warning,error,debug,verbose Then, if you needed to add anything to logger.conf, please first restart Asterisk so those new settings take effect. Then, from the CLI issue set verbose 5 and set debug 5 and finally sip debug. The repeat your dialing steps. The sip debug/trace will then be contained in /var/log/asterisk/full if /var/log/asterisk is where your log files are kept. With that, we can have a better idea of what's happening/not happening to give you the issue you're having. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: I did it !? // Connected to Asterisk 1.2.0-rc1 currently running on serveur1 (pid = 1125) Verbosity is at least 4 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 f1682d8d-8f 84 Idle xpidf+xml 192.168.0.21 86 5f32aec-95b 85 Idle xpidf+xml 192.168.0.20 84 cb424ae1-e4 86 Idle xpidf+xml 192.168.0.20 84 715fac66-a9 87 Idle xpidf+xml 4 active SIP subscriptions serveur1*CLI // serveur1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 87/87 192.168.0.21 D N 5060 OK (84 ms) 86/86 192.168.0.21 D N 5060 OK (97 ms) 85/85 192.168.0.20 D N 5060 OK (87 ms) 84/84 192.168.0.20 D N 5060 OK (96 ms) 4 sip peers [4 online , 0 offline] serveur1*CLI /// my sip.conf: [general] context=local ; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=nxs.yi.org; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=1000 ; Default length of incoming/outoing registration allow=all ; First disallow all codecs musicclass=default ; Sets the default music on hold class for all SIP calls language=fr ; Default language setting for all users/peers rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity tpholdtimeout=300 ; Terminate call if 300 seconds of no RTP activity useragent=Asterisk PBX ; Allows you to change the user agent string dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833 -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users
Re: [Asterisk-Users] IM / presence asterisk-1.2-RC1
When the polycom ip300 phone (1.6.2) send registration SUBSCRIBE message is sent to buddies from directory file so NOTIFY is received from these one. When I want to change status the ip phone don't send NOTIFY to subscriber unlike SER which is a proxy!!! Why? Harry --- harry gaillac [EMAIL PROTECTED] a écrit : Here are some other files. Why asterisk send sip OPTION message to agents ? Harry 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x81cf940 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted Retransmitting #2 (NAT) to 192.168.0.20:5060: OPTIONS sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.119.11.222:5060;branch=z9hG4bK4a119599;rport From: asterisk sip:[EMAIL PROTECTED];tag=as747a6ef0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Nov 2005 10:23:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8194ea0 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted /// --- harry gaillac [EMAIL PROTECTED] a écrit : Sorry, Here are some files Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : This is good debugging info you've listed below, but this isn't a sip debug/trace. To do that, first verify in your logger.conf file you have the following line: full = notice,warning,error,debug,verbose Then, if you needed to add anything to logger.conf, please first restart Asterisk so those new settings take effect. Then, from the CLI issue set verbose 5 and set debug 5 and finally sip debug. The repeat your dialing steps. The sip debug/trace will then be contained in /var/log/asterisk/full if /var/log/asterisk is where your log files are kept. With that, we can have a better idea of what's happening/not happening to give you the issue you're having. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: I did it !? // Connected to Asterisk 1.2.0-rc1 currently running on serveur1 (pid = 1125) Verbosity is at least 4 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 f1682d8d-8f 84 Idle xpidf+xml 192.168.0.21 86 5f32aec-95b 85 Idle xpidf+xml 192.168.0.20 84 cb424ae1-e4 86 Idle xpidf+xml 192.168.0.20 84 715fac66-a9 87 Idle xpidf+xml 4 active SIP subscriptions serveur1*CLI // serveur1*CLI sip show peers Name/username HostDyn Nat ACL Port Status 87/87 192.168.0.21 D N 5060 OK (84 ms) 86/86 192.168.0.21 D N 5060 OK (97 ms) 85/85 192.168.0.20 D N 5060 OK (87 ms) 84/84 192.168.0.20 D N 5060 OK (96 ms) 4 sip peers [4 online , 0 offline] serveur1*CLI /// my sip.conf: [general] context=local ; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=nxs.yi.org; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpirey=3600 ; Max length of incoming registration we allow defaultexpirey=1000 ; Default length of incoming/outoing registration allow=all ; First disallow all codecs musicclass=default ; Sets the default music on hold class for all SIP calls language=fr ; Default language setting for all users/peers rtptimeout=60 ; Terminate call
[Asterisk-Users] ASTERISK + POLYCOM IP PHONES
Hello, I try to setup presence with polycom ip phones ip300 (1.6.2) . I added buddies in directory files all is right for registration subscription notification but when i want to change status notify message is not sent to subscribers ? I don't understand ! Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] IM / presence asterisk-1.2-RC1
Hello, Asterisk don't support IM presence because of no proxy function in chan_sip ! Regards Harry --- harry gaillac [EMAIL PROTECTED] a écrit : When the polycom ip300 phone (1.6.2) send registration SUBSCRIBE message is sent to buddies from directory file so NOTIFY is received from these one. When I want to change status the ip phone don't send NOTIFY to subscriber unlike SER which is a proxy!!! Why? Harry --- harry gaillac [EMAIL PROTECTED] a écrit : Here are some other files. Why asterisk send sip OPTION message to agents ? Harry 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x81cf940 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted Retransmitting #2 (NAT) to 192.168.0.20:5060: OPTIONS sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 80.119.11.222:5060;branch=z9hG4bK4a119599;rport From: asterisk sip:[EMAIL PROTECTED];tag=as747a6ef0 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 11 Nov 2005 10:23:08 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- 2005-11-11 11:23:10 WARNING[1639]: chan_sip.c:1045 __sip_xmit: sip_xmit of 0x8194ea0 (len 477) to 192.168.0.20:-1 returned 5060: Operation not permitted /// --- harry gaillac [EMAIL PROTECTED] a écrit : Sorry, Here are some files Harry --- BJ Weschke [EMAIL PROTECTED] a écrit : This is good debugging info you've listed below, but this isn't a sip debug/trace. To do that, first verify in your logger.conf file you have the following line: full = notice,warning,error,debug,verbose Then, if you needed to add anything to logger.conf, please first restart Asterisk so those new settings take effect. Then, from the CLI issue set verbose 5 and set debug 5 and finally sip debug. The repeat your dialing steps. The sip debug/trace will then be contained in /var/log/asterisk/full if /var/log/asterisk is where your log files are kept. With that, we can have a better idea of what's happening/not happening to give you the issue you're having. On 11/10/05, harry gaillac [EMAIL PROTECTED] wrote: I did it !? // Connected to Asterisk 1.2.0-rc1 currently running on serveur1 (pid = 1125) Verbosity is at least 4 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 f1682d8d-8f 84 Idle xpidf+xml 192.168.0.21 86 5f32aec-95b 85 Idle xpidf+xml 192.168.0.20 84 cb424ae1-e4 86 Idle xpidf+xml 192.168.0.20 84 715fac66-a9 87 Idle xpidf+xml 4 active SIP subscriptions serveur1*CLI // serveur1*CLI sip show peers Name/username Host Dyn Nat ACL Port Status 87/87 192.168.0.21 D N 5060 OK (84 ms) 86/86 192.168.0.21 D N 5060 OK (97 ms) 85/85 192.168.0.20 D N 5060 OK (87 ms) 84/84 192.168.0.20 D N 5060 OK (96 ms) 4 sip peers [4 online , 0 offline] serveur1*CLI /// my sip.conf: [general] context=local ; Default context for incoming calls ; if asterisk was compiled with OSP support. realm=nxs.yi.org; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RFC 3261 ; Set this to your host name or domain name bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls tos=lowdelay; lowdelay,throughput,reliability,mincost,none maxexpirey=3600 ; Max length of incoming registration we allow === message truncated
Re: [Asterisk-Users] groupware + unified messagerie +Asterisk
hello, http://www.egroupware.org/ would be a good choice ( open source). --- Patrick [EMAIL PROTECTED] a écrit : On Wed, 2005-11-09 at 12:45 +, Are wrote: We want to intergrate AstBill with a Groupeware or CRM but want input what people will prefeer. On our list today we have http://www.sugarcrm.com/crm/ http://www.vtiger.com/ http://www.egroupware.org/ A couple more worth looking at. Don't remember which one but one of these projects is planning or working on Asterisk integration. CentraView - http://www.centraview.com Centric CRM - http://www.centriccrm.com Regards, Patrick ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] groupware + unified messagerie +Asterisk
Yes I know this project however my goal would be something like this : [FAX] PSTN--[VOICE]--ASTERISK--(e)groupware [SMS] | Mail Server So (e)groupware' clients should be able to send/receive voice messages fax and sms from/to e-mail click to dial contacts in address book and more :) What do you think of this project ? Regards Harry --- Robert Rozman [EMAIL PROTECTED] a écrit : Hi, I guess you know this project, but just in case: http://jivesoftware.org/asterisk-im/ IMHO, Egroupware would be best groupware solution to start on, but they have little interest in doing that (searching their mailing list for voip returned 2 hits...). We will gradually start working on merging java sip client with Asterisk-IM client and see what will come out Regards, Rob. - Original Message - From: Matt Riddell [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, November 10, 2005 5:25 AM Subject: Re: [Asterisk-Users] groupware + unified messagerie +Asterisk harry gaillac wrote: it's no what i expect the easier solution you provide the more customers you get ! Indeed. However, I tend to be of the opinion that you should have enough money in the bank for a full year of wages for someone if you take on extra staff. While this may make my growth slower, at least I can honestly guarantee my staff's continued employment! So, to cut a long story short, I don't have enough staff to write an infinitely configurable one, as I currently have my books pretty crammed with jobs. If you have any questions though and want to develop one yourself, I'm more than happy to help you! :D -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] groupware + unified messagerie +Asterisk
Thanks for your advises it's no what i expect the easier solution you provide the more customers you get ! I don't agree you ! the best solution you provide the more customers you get (apache projects) ! Indeed. However, I tend to be of the opinion that you should have enough money in the bank for a full year of wages for someone if you take on extra staff. A commercial solution would be a better choice ! While this may make my growth slower, at least I can honestly guarantee my staff's continued employment! So, to cut a long story short, I don't have enough staff to write an infinitely configurable one, as I currently have my books pretty crammed with jobs. I agree you I don't ask you to write this project . asterisk hylafax (e)groupware have been written why not provide an open source solution to improve the use of asterisk for the users . If you have any questions though and want to develop one yourself, I'm more than happy to help you! thank you for your assistance Regards Harry PS: What about presence/IM may i load the lastest asterisk on cvs ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] IM / presence asterisk-1.2-RC1
Hello, Does asterisk's team will improve IM and presence in asterisk-1.2 ! Send Sip MESSAGE is impossible. When the buddies status change nothing is happened. How asterisk's team plan to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] IM / presence asterisk-1.2-RC1
I did it !?
//
Connected to Asterisk 1.2.0-rc1 currently running on
serveur1 (pid = 1125)
Verbosity is at least 4
serveur1*CLI sip show subscriptions
Peer UserCall ID Extension
Last state Type
192.168.0.21 86 f1682d8d-8f 84
Idle xpidf+xml
192.168.0.21 86 5f32aec-95b 85
Idle xpidf+xml
192.168.0.20 84 cb424ae1-e4 86
Idle xpidf+xml
192.168.0.20 84 715fac66-a9 87
Idle xpidf+xml
4 active SIP subscriptions
serveur1*CLI
//
serveur1*CLI sip show peers
Name/username HostDyn Nat ACL
Port Status
87/87 192.168.0.21 D N
5060 OK (84 ms)
86/86 192.168.0.21 D N
5060 OK (97 ms)
85/85 192.168.0.20 D N
5060 OK (87 ms)
84/84 192.168.0.20 D N
5060 OK (96 ms)
4 sip peers [4 online , 0 offline]
serveur1*CLI
///
my sip.conf:
[general]
context=local ; Default context for incoming calls
; if asterisk was compiled with OSP support.
realm=nxs.yi.org; Realm for digest authentication
; defaults to asterisk
; Realms MUST be globally unique according to
RFC
3261
; Set this to your host name or domain name
bindport=5060 ; UDP Port to bind to (SIP standard
port is 5060)
bindaddr=nxs.yi.org ; IP address to bind to (0.0.0.0
binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound
calls
tos=lowdelay;
lowdelay,throughput,reliability,mincost,none
maxexpirey=3600 ; Max length of incoming
registration we allow
defaultexpirey=1000 ; Default length of
incoming/outoing registration
allow=all ; First disallow all codecs
musicclass=default ; Sets the default music on hold
class for all SIP calls
language=fr ; Default language setting for all
users/peers
rtptimeout=60 ; Terminate call if 60 seconds of no
RTP activity
tpholdtimeout=300 ; Terminate call if 300 seconds of
no RTP activity
useragent=Asterisk PBX ; Allows you to change the
user agent string
dtmfmode = rfc2833 ; Set default dtmfmode for sending
DTMF. Default: rfc2833
promiscredir = no ; If yes, allows 302
or REDIR to non-local SIP address
nat=yes
qualify=500
[84]
type=friend
secret=84
context=local
host=dynamic
mailbox=84
allow=all
[85]
type=friend
secret=85
context=local
host=dynamic
mailbox=85
allow=all
[86]
type=friend
secret=86
context=local
host=dynamic
mailbox=86
allow=all
[87]
type=friend
secret=87
context=local
host=dynamic
mailbox=87
allow=all
//
my extension.conf
;
[general]
;
static=yes
writeprotect=no
switch = Realtime/[EMAIL PROTECTED]
;
[globals]
;
[local]
exten = 80,1,Answer
exten = 80,2,Dial(Zap/g2,14)
exten = 80,3,VoiceMail(u80)
exten = 80,103,VoiceMail(b80)
exten = 84,hint,Sip/84
exten = 84,1,Answer
exten = 84,2,Dial(Sip/84,10)
exten = 84,3,VoiceMail(u84)
exten = 84,103,VoiceMail(b84)
exten = 85,hint,Sip/85
exten = 85,1,Answer
exten = 85,2,Dial(Sip/85,10)
exten = 85,3,VoiceMail(u85)
exten = 85,103,VoiceMail(b85)
exten = 86,hint,Sip/86
exten = 86,1,Answer
exten = 86,2,Dial(Sip/86,10)
exten = 86,3,VoiceMail(u86)
exten = 86,103,VoiceMail(b86)
exten = 87,hint,Sip/87
exten = 87,1,Answer
exten = 87,2,Dial(Sip/87,10)
exten = 87,3,VoiceMail(u87)
exten = 87,103,VoiceMail(b87)
include = mailbox
include = apps
include = pstn
[mailbox]
exten = 700,1,VoiceMailMain()
[pstn]
exten = s,1,Answer
exten = s,2,Goto(local,84,1)
include = outgoing-pstn
[outgoing-pstn]
ingnorepat = 0
exten = _0,1,Dial(Zap/g1/${EXTEN:1})
exten = _0.,1,Dial(Zap/g1/${EXTEN:1})
exten = _0.,3,Hangup
//
Regards
Harry
--- BJ Weschke [EMAIL PROTECTED] a écrit :
Harry,
The monitoring of buddies on Polycom phones is
possible with the
release candidate for v1.2. We've asked for a sip
debug/trace from you
to try and troubleshoot your problem, and you
haven't provided that to
date.
On 11/10/05, harry gaillac [EMAIL PROTECTED]
wrote:
Hello,
Does asterisk's team will improve IM and presence
in
asterisk-1.2 !
Send Sip MESSAGE is impossible.
When the buddies status change nothing is
happened.
How asterisk's team plan to solve this problem ?
Regards
Harry
___
Appel audio GRATUIT
Re: [Asterisk-Users] asterisk-1.2-bêta2 | pre sence/subscription support in the SIP channel driver
Does asterisk support RFC3265 ? Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] sip_message_support.patch
Hello Matt, In fact I look for messaging an presence between sip phones . http://www.voip-forum.com/news.php?p=184c=1 I use polycom ip phone with presence (rfc3265) and IM (SIMPLE). Do you you think the job of Joshua Colp could help me to use presence/IM with asterisk ? Regards Harry http://www.voip-forum.com/news.php?p=184c=1 --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: Hello, Does sip_message_support.patch is available for asterisk-1.2-bêta2 ? Is there an other solution for Sip message ? pabx*CLI show agi send text Usage: SEND TEXT text to send Sends the given text on a channel. Most channels do not support the transmission of text. Returns 0 if text is sent, or if the channel does not support text transmission. Returns -1 only on error/hangup. Text consisting of greater than one word should be placed in quotes since the command only accepts a single argument. pabx*CLI show agi receive text Usage: RECEIVE TEXT timeout Receives a string of text on a channel. Specify timeout to be the maximum time to wait for input in milliseconds, or 0 for infinite. Most channels do not support the reception of text. Returns -1 for failure or 1 for success, and the string in parentheses. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] groupware + unified messagerie +Asterisk
Somebody would be interested in a such project ? Harry --- Kristof Hardy [EMAIL PROTECTED] a écrit : harry gaillac wrote: Is it possible to add a frontend groupware with All is possible, you're only limited by your imagination. (always wanted to say this :p) I'm not sure there's a(n Open-source) project like this already. Cheers.. ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] sip_message_support.patch
the asterisk's answer ! // Connected to Asterisk 1.2.0-beta2 currently running on serveur1 (pid = 2729) Remote UNIX connection Verbosity is at least 3 Nov 9 11:48:21 WARNING[2926]: chan_sip.c:7251 receive_message: Received message to sip:[EMAIL PROTECTED] from bob sip:[EMAIL PROTECTED];tag=F71E8D2E-67C04697, dropped it... Content-Type:text/plain Message: Call me. serveur1*CLI // a part of my extension.conf file exten = 84,hint,Sip/84 exten = 84,1,Answer exten = 84,2,SendText() exten = 84,3,Dial(Sip/84,10) exten = 84,4,VoiceMail(u84) exten = 84,103,VoiceMail(b84) exten = 85,hint,Sip/85 exten = 85,1,Answer exten = 85,2,SendText() exten = 85,3,Dial(Sip/85,10) exten = 85,4,VoiceMail(u85) exten = 85,103,VoiceMail(b85) exten = 86,hint,Sip/86 exten = 86,1,Answer exten = 86,2,SendText() exten = 86,3,Dial(Sip/86,10) exten = 86,4,VoiceMail(u86) exten = 86,103,VoiceMail(b86) exten = 87,hint,Sip/87 exten = 87,1,Answer exten = 87,2,SendText() exten = 87,3,Dial(Sip/87,10) exten = 87,4,VoiceMail(u87) exten = 87,103,VoiceMail(b87) / neither SUBSCRIBE, NOTIFY, MESSAGE sip method are ok :( Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: Hello Matt, In fact I look for messaging an presence between sip phones . http://www.voip-forum.com/news.php?p=184c=1 Should work with current CVS HEAD version. I use polycom ip phone with presence (rfc3265) and IM (SIMPLE). Do you you think the job of Joshua Colp could help me to use presence/IM with asterisk ? Should also do :) -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] asterisk-1.2-bêta2 | presen ce/subscription support in the SIP channel dri ver
I'm not a developper ! What do you mean Some parts of it, yes. harry --- BJ Weschke [EMAIL PROTECTED] a écrit : Some parts of it, yes. On 11/9/05, harry gaillac [EMAIL PROTECTED] wrote: Does asterisk support RFC3265 ? Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] groupware + unified messagerie +Asterisk
it's no what i expect the easier solution you provide the more customers you get ! --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: What about egroupware ! We use it, although there is no simple click to install installation package for Asterisk integration. The idea is to use flash operator panel to load a url when each extension is dialed. And for click to dial, I use call files. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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: RE : [Asterisk-Users] asterisk-1.2-bêta2 | presence/subscription support in the SIP channe l driver
Olivier, Oui !!! pour asterisk ou openpbx. SER est un excellent proxy sip ! Il est evident qu' SER n'offre pas les fonctionalités d'un ipbx. je ne pense pas que toneec soit viable , combien d'opérateurs offrent ces services (Skype)... Votre pojet stagne! Vous avez fait le choix de beacoups d'ITSPs. Je regrette le mépris de votre part à mon égard ! économiquement une solution non open source serait souhaitable ! Je parviendrai a mes objétifs !! Cordialement Harry- --- Olivier Taylor [EMAIL PROTECTED] a écrit : Salut Harry, Tu quittes Ser pour asterisk? Olivier ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] SIP domain support for authentication and virtual hosting
nobody has an answer here !! --- harry gaillac [EMAIL PROTECTED] a écrit : Hello, Where may i find documentation about SIP domain support and dnsmgr.conf , Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] asterisk-1.2-bêta2 | pre sence/subscription support in the SIP channel driver
nobody has an answer here! --- harry gaillac [EMAIL PROTECTED] a écrit : Hello, I configure Polycom ip300 for presence but when status change notify is no sent to subscriber !? Why ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] asterisk-1.2-bêta2 | pre sence/subscription support in the SIP channel driver
Hello, Sorry here are my sip.conf and extensions.conf in fact when polycom ip300 send subscribe to buddies these one send back notify but nothing else when status change Regards Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com sip.conf Description: 3455877249-sip.conf extensions.conf Description: 3949034846-extensions.conf ?xml version=1.0 standalone=yes? directory item_list item lnbob/ln fnSINCLAR/fn ct86/ct sd1/sd bw1/bw /item /item_list /directory ?xml version=1.0 standalone=yes? directory item_list item lnalice/ln fnSPRING/fn ct84/ct sd1/sd bw1/bw /item /item_list /directory___ --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] groupware + unified messagerie +Asterisk
Hello, Is it possible to add a frontend groupware with asterisk in order to Provide send receive fax to mail, sms to mail, voice messages . Asterisk or openpbx could be the server of the unified messagerie . click to dial contact in address book ,... Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] groupware + unified messagerie +Asterisk
What about egroupware ! Harry --- Kristof Hardy [EMAIL PROTECTED] a écrit : harry gaillac wrote: Is it possible to add a frontend groupware with All is possible, you're only limited by your imagination. (always wanted to say this :p) I'm not sure there's a(n Open-source) project like this already. Cheers.. ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] asterisk-1.2-bêta2 | presen ce/subscription support in the SIP channel dri ver
Connected to Asterisk 1.2.0-beta2 currently running on serveur1 (pid = 1553) Verbosity is at least 3 serveur1*CLI sip show subscriptions Peer UserCall ID Extension Last state Type 192.168.0.21 86 2127e5fd-5f 84 Idle xpidf+xml 192.168.0.20 84 61c23b4e-3d 86 Idle xpidf+xml 2 active SIP subscriptions --- BJ Weschke [EMAIL PROTECTED] a écrit : Ok. What does sip show subscriptions from the CLI show you? On 11/8/05, harry gaillac [EMAIL PROTECTED] wrote: Hello, Sorry here are my sip.conf and extensions.conf in fact when polycom ip300 send subscribe to buddies these one send back notify but nothing else when status change Regards Harry --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here! Actually someone asked for you config details. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] SIP domain support for authentication and virtual hosting
thanks Matt for your answer Does asterisk-1.2-stable will provide this features ? Harry PS: Who are the main developpers for the sip channels ? --- Matt Riddell [EMAIL PROTECTED] a écrit : harry gaillac wrote: nobody has an answer here !! Where may i find documentation about SIP domain support and dnsmgr.conf , The problem is that dnsmgr is new and not finished, so there is not much documentation yet. Re the SIP domain support, I don't know, there is the announcement here ( http://www.voip-forum.com/news.php?p=183 ), but it doesn't really have that much info. -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] SIP domain support for authentication and virtualhosting
Hello, I would have wanted to help you but i have no more informations :( Harry PS: The problem is that dnsmgr is new and not finished, so there is not much documentation yet. Re the SIP domain support, I don't know, there is the announcement here ( http://www.voip-forum.com/news.php?p=183 ), but it doesn't really have that much info. --- B. J. Bomar [EMAIL PROTECTED] a écrit : I've tried it in 1.2, and maybe I'm just not smart enough to get it to work. Do you have a working example? What I am looking for is [EMAIL PROTECTED] to be different that [EMAIL PROTECTED] As far as I can tell, currently for registrations asterisk only looks at everything to the left of the @ sign. It also looks like according to the docs I have come across that the above works for routing, but not registering. B. J. -Original Message- From: Matt Riddell [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 08, 2005 12:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP domain support for authentication and virtualhosting harry gaillac wrote: thanks Matt for your answer Does asterisk-1.2-stable will provide this features ? Heh, that's a hard one to answer! 1.2 is only released in Beta at the moment although: The current plan is release 1.2 early next week and get the new development tree open later that week. So, yes it is available in 1.2, but it is not in the current STABLE - 1.0.9 (at least until early next week - when 1.2 becomes the new STABLE version). :D -- Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] sip_message_support.patch
Hello, Does sip_message_support.patch is available for asterisk-1.2-bêta2 ? Is there an other solution for Sip message ? http://juraj.bednar.sk/work/software/asterisk/messaging/sip_message_support.patch ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] asterisk-1.2-bêta2 | pres ence/subscription support in the SIP channel driver
Hello, I configure Polycom ip300 for presence but when status change notify is no sent to subscriber !? Why ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] SIP domain support for authentication and virtual hosting
Hello, Where may i find documentation about SIP domain support and dnsmgr.conf , Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] SER+ASTERISK
No ! Asterisk should send the invite request to sip proxy . Harry --- Walter Willis [EMAIL PROTECTED] a écrit : the ser an asterisk run in the same box??? redirect host + port :) 2005/11/4, harry gaillac [EMAIL PROTECTED]: Hello, I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phones asterisk 5050-- |firewall+nat|-192.168. ser 5060--- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com http://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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] SER+ASTERISK
Hello, I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phones asterisk 5050-- |firewall+nat|-192.168. ser 5060--- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] SER+ASTERISK
Hello Walter, The ser an asterisk run in the same box. What do you mean redirect host + port :) Sip agents send sip requests to ser (outbound proxy) and this one to asterisk ! sip agents are both registered on ser and asterisk. Please to explain me how asterisk redirect the requests. Regards Harry --- Walter Willis [EMAIL PROTECTED] a écrit : the ser an asterisk run in the same box??? redirect host + port :) 2005/11/4, harry gaillac [EMAIL PROTECTED]: Hello, I wish to setup this scheme: ser-0.9.4 asterisk-1.2-bêta polycom ip300 phones asterisk 5050-- |firewall+nat|-192.168. ser 5060--- My ip phones use ser as outbound sip proxy and asterisk as sip registrar server. Ser Forward REGISTER requests to asterisk however when a phone try to send an invite message then asterisk send icmp to private ip (host=dynamic in sip.conf) Is it possible to solve this problem ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation sponsored by Easynews.com http://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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Re: [Serusers] Accounting
Hello, I've been waiting ASTERISK-B2BUA for asterisk-1.2 Regards Harry --- Rafael R. GV [EMAIL PROTECTED] a écrit : try this: ASTERISK-B2BUA http://lists.berlios.de/pipermail/b2bua-users/2005-November/000155.html Features: full vovida's b2bua radius emulation, extended radius attributes, radius failover, LCR, Call failover, Codec based routing and other useful things. rafael On 11/2/05, harry gaillac [EMAIL PROTECTED] wrote: Ok I trust you but does asterisk support radius ? Harry --- Daryl Sanders [EMAIL PROTECTED] a écrit : Asterisk works fine as a B2BUA for accounting. - Daryl On 11/2/05, harry gaillac [EMAIL PROTECTED] wrote: Sorry My question is: which utility for this module if it cannot calculate acc-session-time ? which b2bua is available for accounting (not vovida) ! Harry --- Jan Janak [EMAIL PROTECTED] a écrit : I already sent you a reply and explained that SER does not support Acct-Session-Time, so I do not understand why did you post the same question again to the list. I also told you that SER does not send Stop accounting request when it recieves BYE but when it receives a final reply to the BYE. Jan. On 02-11-2005 00:33, harry gaillac wrote: Hello, Does acc module allow these requests to a radius server according to rfc2866 ? A Talk Start information is sent when SIP Server receives 200 OK for a INVITE request. A TAlK Stop information when SIP Server receives BYE. How may i configure it ? if acc module can't calculate acct-session-time why does this module provide radius support ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Serusers mailing list [EMAIL PROTECTED] http://mail.iptel.org/mailman/listinfo/serusers -- rrgv ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Asterisk + Ser + Music on hold
Hello asterisk users, I want to register sip agents (polycom ip 300) and asterisk on Ser (sip express router) sip user1--SERsip user2 | | Asterisk How may i configure Ser+Asterisk in order to provide Moh to sip agents when hold key is pressed (rfc3264) ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Forward sip messages to a proxy
Hello, Is this possible to send SIP messages (MESSAGE, SUBSCRIBE, NOTIFY) to a sip proxy from asterisk ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] chan_exosip2
Hello, I read roadmap on www.openpbx.org. Does chan_exosip2 will be able to provide a real sip proxy ? What about asterisk solutions ? Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] chan_exosip2
Thanks for reply, Does Chan_exosip2 is stable where may i find help ? Chan_sip from asterisk doesn't support IM and presence via SIMPLE. I whish to use either asterisk or openpbx to provide telephony features with SER to relay IM and presence SIMPLE openpbx or asterisk || SER--- || Sip agent sip agent What do you advise me ? Regards Harry --- Joshua Colp - Asterlink [EMAIL PROTECTED] a écrit : Hello Harry, This is rather the wrong list to ask this... since this is Asterisk, not OpenPBX.org Chan_exosip2 though is something I'm basically designing to have 3 operating modes. Full server: Most closely resembles chan_sip in that it acts as a B2BUA Partial proxy: Extensions are mapped to SIP URIs and it acts as a proxy. Gateway: No authentication occurs (this is presumably done outside by a SIP proxy), incoming calls just get thrown into a context. Outgoing calls are down via SIP URI. Joshua Colp -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of harry gaillac Sent: Tuesday, November 01, 2005 10:19 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] chan_exosip2 Hello, I read roadmap on www.openpbx.org. Does chan_exosip2 will be able to provide a real sip proxy ? What about asterisk solutions ? Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] chan_exosip2
In fact I want to forward SIP MESSAGES to any sip proxy Unless chan_exosip2 is able to relay IM presence via SIMPLE . Harry --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] a écrit : I read roadmap on www.openpbx.org. Does chan_exosip2 will be able to provide a real sip proxy ? What about asterisk solutions ? I guess you can use chan_exosip2 with asterisk if you hack it in yourself. Also, as soon as asterisk is released in one single GPL license, it may as well be included in the source :) roy ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] SS7 with Asterisk
Hello, I think you have to contact digium for software to support SS7 with digium cards. Harry --- Usman [EMAIL PROTECTED] a écrit : anyone running SS7 with Asterisk ? Please help me out. I need to know the hardware used for SS7 with Digium E1 cards... Thanks, ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] SS7 with Asterisk
Look at http://www.ss7box.com/ too. Harry --- Goran Skular [EMAIL PROTECTED] a écrit : anyone running SS7 with Asterisk ? Please help me out. I need to know the hardware used for SS7 with Digium E1 cards... I can point you to one company in Austria. They deployed SS7 on Asterisk, but not with Digium cards for one smaller telco. ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Where to get the latest SIP Firmware for Polycom Phones?
Hello, I 'll ask to my reseller Harry --- [EMAIL PROTECTED] a écrit : thanks for that, i knew already but it misses the actual version Jesse Keating wrote: On Fri, 2005-10-07 at 11:17 +0200, Kib Eki wrote: Hello, can anybody tell me where to get the latetest SIP Firmware 1.6.2 for the Polycom phones? http://www.freedomphones.net/polycom/files/ ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] www.openpbx.org
Hello, What do you think of this project www.openpbx.org ? Something like ser and openser ! Kinds Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] I got 403, Forbidden... please help
Hello, Try insecure=very in [sip.philonline.com] Harry --- Ryan Pagquil [EMAIL PROTECTED] a écrit : Hi, I'm setting up Asterisk as a voicemail with SER. My problem is, when a caller that is not registered with asterisk (no username and password in sip.conf) it prompts 403, Forbidden . I need all calls from outside of my network to reach asterisk for my users' voicemails, because anonymous users will surely reach voicemail of my users to leave messages. What do I need to do to make those anonymous callers to reach the voicemails of my users? here is my sip.conf. [general] port = 5060 bindaddr = 202.84.24.47 context = sip disallow=all allow=ulaw allow=alow ;register=me:[EMAIL PROTECTED]/1000 [sip.philonline.com] type=friend host=sip.philonline.com fromuser=rpagquil secret=test123 fromdomain=sip.philonline.com [phone1] type = friend username = phone1 secret = test123 host = dynamic context = sip mailbox = callerid=Test1 [acjeff] type=friend username=acjeff host=dynamic defaultip=10.0.1.236 nat=yes context=sip mailbox= callerid=Test2 [usser1] type = friend username = usser1 secret = test123 nat=yes host = dynamic context = sip mailbox = 111 callerid=User1 Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] call tests
Hello, In order I try to fix my configuration please to call me at : sip:[EMAIL PROTECTED] or sip:[EMAIL PROTECTED] or sip:[EMAIL PROTECTED] or sip:[EMAIL PROTECTED] Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] hints and polycom IP 300 phones
Hello, I have two polycom ip300. I patched Asterisk However it don't show status of phones when I press busy, Away, ... So I use Sip Express Router (proxy sip) for IM and Presence SIMPLE. Harry --- Adam Goryachev [EMAIL PROTECTED] a écrit : Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600 Is there any additional debug apart from show hints to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =- 655 : SIP/gs102_1 State 0 Watchers 0 605 : Zap/127 State 0 Watchers 3 604 : SIP/ata186_2 State 0 Watchers 0 603 : SIP/ata186_1 State 0 Watchers 0 602 : Zap/129 State 0 Watchers 0 601 : SIP/polycom_b State 0 Watchers 1 600 : SIP/polycom_a State 1 Watchers 2 The IP600 is watching 605 and 600 and working nicely for both, the IP300 is watching 601, but isn't working Has anyone got a IP300 phone to display the status ?? Any suggestions for things to look at/etc ?? PS, of course, the current state is that 600 is off-hook and all others are on-hook. Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 8304 0001 www.websitemanagers.com.au ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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
Tr: [Asterisk-Users] MWI - message waiting indication
Remarque : message transféré en pièce jointe. ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com---BeginMessage--- hello, I read http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large anybody could tell me more about this ? Is it available with ARA ? Regards Harry Method 3 Q: If you have your SIP phones registered with SER but your voicemail is handled by asterisk, how do you get the MWI (Message Waiting Indicator) light to function on the phone? A: In sip.conf create a section pointing at your SER router. [ser] type=friend; We allow incoming and outgoing calls. Use peer if you are only doing MWI context=ser; This is the context incoming calls land in host=ser.server.tld; This is the hostname or IP address of your SER server fromdomain=ser.server.rld ; This is your SER_DOMAIN insecure=very ; This allows incoming calls from the phones routing through ser to be passed into asterisk [EMAIL PROTECTED] ; This is where you list the voicemail boxes to monitor This tells asterisk that if a voicemail comes in to user then it needs to send a SIP NOTIFY message to the ser.server.tld phone. Well this is all well and good except how does SER deliver this NOTIFY to the phones? First thing is that you need to make a tiny change to the asterisk code to pass the mailbox user in the SIP NOTIFY packet. --- channels/chan_sip.c.origThu Jul 14 12:03:18 2005 +++ channels/chan_sip.c Thu Jul 14 12:05:26 2005 @@ -9710,6 +9710,7 @@ /* Called with peerl lock, but releases it */ struct sip_pvt *p; int newmsgs, oldmsgs; + char *s; /* Check for messages */ ast_app_messagecount(peer-mailbox, newmsgs, oldmsgs); @@ -9735,6 +9736,10 @@ /* Recalculate our side, and recalculate Call ID */ if (ast_sip_ouraddrfor(p-sa.sin_addr,p-ourip)) memcpy(p-ourip, __ourip, sizeof(p-ourip)); + strcpy(p - username, peer - mailbox); /* Username = Mailbox name */ + s = strchr(p - username, '@'); /* Remove the context part */ + if (s != NULL) +*s = 0; build_via(p, p-via, sizeof(p-via)); build_callid(p-callid, sizeof(p-callid), p-ourip, p-fromdomain); /* Send MWI */ After this patch is applied, the MWI NOTIFY messages coming from asterisk will have the URI [EMAIL PROTECTED] This can be then routed with ser to the correct phone with normal SER routing rules. ie. SER does a lookup(location) and then a t_relay(). I don't believe this patch should effect any non-ser controlled sip phones. For me, this method was a lot easier then Method 2 listed above. You can add as may mailbox's as you like into the mailbox= line in the asterisk sip.conf file. One possible problem is if you have a mailbox called [EMAIL PROTECTED] and another called [EMAIL PROTECTED], this patch will make the MWI indicator light up for phone [EMAIL PROTECTED] when either mailbox gets a message. A simple modification to the patch and SER could be used to handle multiple contexts if required however this simplification is sufficient for me. ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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 ---End Message--- ___ --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] Asterisk Community Participant; Katrina Refugee
Mr Richardson, I sympathize with american people after this disaster. However If i was God I would feel remorse for all people in the world in destitution because of diseases, wars, starvation, ... God should really feel remorse . Thinks to all people in destitution in the world . Harry --- JR Richardson [EMAIL PROTECTED] a écrit : Hi All, My family and I are doing well. Thank you all for your prayers. We are using this as an opportunity to rebuild. I didn't think I really needed to but God knows best and we will obey. My family and I will temporarily be in Lafayette, Louisiana for a while but will probably relocate to Houston, TX in the future. We already have my Daughter registered in school here. Lafayette is my old stomping ground so I'm already at home. My Wife is having a time with directions though. She went half way to Lake Charles (wrong direction) yesterday when she was coming back home from shopping. My house, office, lab and 2 vehicles back in Chalmette, LA, St Bernard Parish are swimming with the fishes, snakes and alligators along with all my computers and Asterisk application development. 100% loss, but hey, we have our health. I have both homeowners and flood insurance so I should recoup most of my losses, it will take a while to get back on track. Insurance adjusters will not be able to enter the Parish till the water is out which could take several weeks if not a few months. I was planning on speaking at this years Astricon conference in Anaheim, CA on Embedded Asterisk Systems but have to resend the invitation at this time. As you can imagine, I have other priorities. I will miss this opportunity to collaborate and share my work with this community. My FTP server is 8 feet under Lake Ponchatrain at this time and foreseeable future. My Internet provider is not online anyway but I am committed and will get my work on-line as soon as possible. I will keep up with Asterisk development as I can and will jump back into the community when available to contribute with focus and vigor. I have bought and collected equipment since being in Telecommunications, VoIP and Internet Technologies for 15 years that are irreplaceable but I will re-build my VoIP laboratory bigger and better than ever. If anyone has any trade secrets on successfully recovering waterlogged electronic equipment, please let me know. God Bless. JR Richardson ___ --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 ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] MWI - message waiting indication
hello, I read http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large anybody could tell me more about this ? Is it available with ARA ? Regards Harry Method 3 Q: If you have your SIP phones registered with SER but your voicemail is handled by asterisk, how do you get the MWI (Message Waiting Indicator) light to function on the phone? A: In sip.conf create a section pointing at your SER router. [ser] type=friend; We allow incoming and outgoing calls. Use peer if you are only doing MWI context=ser; This is the context incoming calls land in host=ser.server.tld; This is the hostname or IP address of your SER server fromdomain=ser.server.rld ; This is your SER_DOMAIN insecure=very ; This allows incoming calls from the phones routing through ser to be passed into asterisk [EMAIL PROTECTED] ; This is where you list the voicemail boxes to monitor This tells asterisk that if a voicemail comes in to user then it needs to send a SIP NOTIFY message to the ser.server.tld phone. Well this is all well and good except how does SER deliver this NOTIFY to the phones? First thing is that you need to make a tiny change to the asterisk code to pass the mailbox user in the SIP NOTIFY packet. --- channels/chan_sip.c.origThu Jul 14 12:03:18 2005 +++ channels/chan_sip.c Thu Jul 14 12:05:26 2005 @@ -9710,6 +9710,7 @@ /* Called with peerl lock, but releases it */ struct sip_pvt *p; int newmsgs, oldmsgs; + char *s; /* Check for messages */ ast_app_messagecount(peer-mailbox, newmsgs, oldmsgs); @@ -9735,6 +9736,10 @@ /* Recalculate our side, and recalculate Call ID */ if (ast_sip_ouraddrfor(p-sa.sin_addr,p-ourip)) memcpy(p-ourip, __ourip, sizeof(p-ourip)); + strcpy(p - username, peer - mailbox); /* Username = Mailbox name */ + s = strchr(p - username, '@'); /* Remove the context part */ + if (s != NULL) +*s = 0; build_via(p, p-via, sizeof(p-via)); build_callid(p-callid, sizeof(p-callid), p-ourip, p-fromdomain); /* Send MWI */ After this patch is applied, the MWI NOTIFY messages coming from asterisk will have the URI [EMAIL PROTECTED] This can be then routed with ser to the correct phone with normal SER routing rules. ie. SER does a lookup(location) and then a t_relay(). I don't believe this patch should effect any non-ser controlled sip phones. For me, this method was a lot easier then Method 2 listed above. You can add as may mailbox's as you like into the mailbox= line in the asterisk sip.conf file. One possible problem is if you have a mailbox called [EMAIL PROTECTED] and another called [EMAIL PROTECTED], this patch will make the MWI indicator light up for phone [EMAIL PROTECTED] when either mailbox gets a message. A simple modification to the patch and SER could be used to handle multiple contexts if required however this simplification is sufficient for me. ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] SER+ASTERISK voicemail
Hello, I set SER as sip proxy and ASTERISK as voicemail server (ARA) and serweb as TUI (telephone user interface) . Serweb | Ua---ser---asterisk voicemail | | Mysql DB I add user agents with address sip:[EMAIL PROTECTED] + aliases sip:[EMAIL PROTECTED] where 123 is mailbox I can forward voice messages to Asterisk with failure route for status 408 or 486. However I can't do it for offline users because of SER look for addresses like sip:[EMAIL PROTECTED] not sip:[EMAIL PROTECTED] where 123 is mailbox How could I solve this problem if possible ? Regards Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Re: [Asterisk-Dev] voicemessages table
I agree you however i solved my problem with app_voicemail.c The table scheme provide in doc/README.odbcstorage don't match to sql queries in app_voicemail.c I think developpers who has written app_voicemail.c for ARA provide a useable table ! Regards Harry --- Steve McMahon [EMAIL PROTECTED] a écrit : These questions should be sent to Asterisk-Users this is not a developer issue. Cheer's Steve McMahon - Original Message - From: harry gaillac [EMAIL PROTECTED] To: asterisk-dev@lists.digium.com Sent: Monday, August 29, 2005 9:18 AM Subject: [Asterisk-Dev] voicemessages table hello, I got some errors with voicemessages table in docs/README.odbcstorage : ++-+--+-+-+---+ | Field | Type| Null | Key | Default | Extra | ++-+--+-+-+---+ | msgnum | int(11) | YES | | NULL | | | dir| varchar(80) | YES | MUL | NULL | | | context| varchar(80) | YES | | NULL | | | macrocontext | varchar(80) | YES | | NULL | | | callerid | varchar(40) | YES | | NULL | | | origtime | varchar(40) | YES | | NULL | | | duration | varchar(20) | YES | | NULL | | | mailboxuser| varchar(80) | YES | | NULL | |* | mailboxcontext | varchar(80) | YES | | NULL | |* | recording | longblob| YES | | NULL | | ++-+--+-+-+---+ according to app_voicemail.c i set mysql desc voicemessages; ++-+--+-+-+---+ | Field | Type| Null | Key | Default | Extra | ++-+--+-+-+---+ | dir| varchar(80) | YES | MUL | NULL | | | msgnum | int(11) | YES | | NULL | | | recording | longblob| YES | | NULL | | | context| varchar(80) | YES | | NULL | | | macrocontext | varchar(80) | YES | | NULL | | | callerid | varchar(40) | YES | | NULL | | | origtime | varchar(40) | YES | | NULL | | | duration | varchar(20) | YES | | NULL | | | mailboxuser| varchar(80) | YES | | NULL | | | mailboxcontext | varchar(80) | YES | | NULL | | ++-+--+-+-+---+ 10 rows in set (0.00 sec) Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] Re: [Asterisk-Dev] voicemessages table
I agree you however i solved my problem with app_voicemail.c The table scheme provide in doc/README.odbcstorage don't match to sql queries in app_voicemail.c I think developpers who has written app_voicemail.c for ARA provide a useable table ! Regards Harry --- Steve McMahon [EMAIL PROTECTED] a écrit : These questions should be sent to Asterisk-Users this is not a developer issue. Cheer's Steve McMahon - Original Message - From: harry gaillac [EMAIL PROTECTED] To: asterisk-dev@lists.digium.com Sent: Monday, August 29, 2005 9:18 AM Subject: [Asterisk-Dev] voicemessages table hello, I got some errors with voicemessages table in docs/README.odbcstorage : ++-+--+-+-+---+ | Field | Type| Null | Key | Default | Extra | ++-+--+-+-+---+ | msgnum | int(11) | YES | | NULL | | | dir| varchar(80) | YES | MUL | NULL | | | context| varchar(80) | YES | | NULL | | | macrocontext | varchar(80) | YES | | NULL | | | callerid | varchar(40) | YES | | NULL | | | origtime | varchar(40) | YES | | NULL | | | duration | varchar(20) | YES | | NULL | | | mailboxuser| varchar(80) | YES | | NULL | |* | mailboxcontext | varchar(80) | YES | | NULL | |* | recording | longblob| YES | | NULL | | ++-+--+-+-+---+ according to app_voicemail.c i set mysql desc voicemessages; ++-+--+-+-+---+ | Field | Type| Null | Key | Default | Extra | ++-+--+-+-+---+ | dir| varchar(80) | YES | MUL | NULL | | | msgnum | int(11) | YES | | NULL | | | recording | longblob| YES | | NULL | | | context| varchar(80) | YES | | NULL | | | macrocontext | varchar(80) | YES | | NULL | | | callerid | varchar(40) | YES | | NULL | | | origtime | varchar(40) | YES | | NULL | | | duration | varchar(20) | YES | | NULL | | | mailboxuser| varchar(80) | YES | | NULL | | | mailboxcontext | varchar(80) | YES | | NULL | | ++-+--+-+-+---+ 10 rows in set (0.00 sec) Harry ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --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] RE: [Asterisk-Dev] voicemessages table
Am i alone with this problem ? I just rewrote voicemessages table because of errors. I read app_voicemail.c to fix my problem. However app_voicemail.c support many schemes to query the tables. Harry --- Jerris, Michael MI [EMAIL PROTECTED] a écrit : harry gaillac I agree you however i solved my problem with app_voicemail.c don't match to sql queries in app_voicemail.c I think developpers who has written app_voicemail.c for ARA provide a useable table ! Please provide a patch to the readme to resolve this through bugs.digium.com. Thanks Mike ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ --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] RE: [Asterisk-Dev] voicemessages table
Am i alone with this problem ? I just rewrote voicemessages table because of errors. I read app_voicemail.c to fix my problem. However app_voicemail.c support many schemes to query the tables. Harry --- Jerris, Michael MI [EMAIL PROTECTED] a écrit : harry gaillac I agree you however i solved my problem with app_voicemail.c don't match to sql queries in app_voicemail.c I think developpers who has written app_voicemail.c for ARA provide a useable table ! Please provide a patch to the readme to resolve this through bugs.digium.com. Thanks Mike ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com ___ Asterisk-Dev mailing list Asterisk-Dev@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --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
