[Asterisk-Users] cell mobile network (GSM) to Asterisk
Hello, currently I am solving the problem of having GSM gateway - can you please document how any of you solved the Asterisk to GSM thing? I would appreciate if you write what HW did you use and what was the price of it, pros and cons. I have limited budget, but I'm opened to every idea. Thank you very much. Marcel ___ --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] Hi...Please help me
Yes it is possible - check out the Asterisk manual or nice book from O'Reilly - Asterisk PBX (The Furute of telephony) Marcel Crazy Boy wrote: Hi Friends, I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is: PSTN (Phone call) --- VOIP phone --- Server system --- --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using Asterisk software? If It can be implemented using Asterisk software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank you Thanks Regards, Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.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
Re: [Asterisk-Users] Hi...Please help me
For hardware check out this page: http://www.digium.com/en/products/hardware/ Marcel Crazy Boy wrote: Hi Friends, I want to implement VOIP PBX service in my office. I have 10 computers and a server. All computers are Pentium IV processors with 512 MB RAM. All employee computers have Windows 2000 Professional OS and Server computer Windows 2000 Professional and Fedora Core 5 Linux OS. I have a VOIP phone and have registered with VoIP service provider. Now, I want to implement VOIP PBX facility to all of my systems. The structure for the same is: PSTN (Phone call) --- VOIP phone --- Server system --- --- Employee 1 PC (Softphone i.e., Headphones with Mic) --- Employee 2 PC (Softphone i.e., Headphones with Mic) --- Employee 3 PC (Softphone i.e., Headphones with Mic) ----- ----- --- Employee 10 PC (Softphone i.e., Headphones with Mic) and vice versa. How can I implement this? Is it possible to implement this using Asterisk software? If It can be implemented using Asterisk software, What softwares I should install in Server and Employee PC's? Is there any need of buying extra hardware? Please reply me. Thank you Thanks Regards, Chandra. Talk is cheap. Use Yahoo! Messenger to make PC-to-Phone calls. Great rates starting at 1¢/min. http://us.rd.yahoo.com/mail_us/taglines/postman7/*http://us.rd.yahoo.com/evt=39666/*http://beta.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
[Asterisk-Users] X100P support on POTS around the world (Slovakia)
Hello everybody, does anybody use P100P FXO card on POTS lines in Slovakia, Bohemia (Czech rep.), Poland, Hungary...? I need to know if those cards work especially in Slovakia or if you can reccomend FXO cards for Slovak POTS lines. Thanks, Marcel ___ --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] X100P support on POTS around the world (Slovakia)
If there any way to find out if they will work? (If I ask the local telco) Or just buy and test the card out? :) Marcel Krzysztof Drewicz wrote: Marcel Hecko napisał(a): Hello everybody, does anybody use P100P FXO card on POTS lines in Poland, I need to know if those cards work especially in Slovakia or if you can reccomend FXO cards for Slovak POTS lines. For Slovak I don't know. For POLAND, I use them with TPSA (Main, the biggest one, Telco operator) aka France Telecom -- Poland division. Btw: everything depends on you Area PABX, so you experience in any location may (and i proablly will) be diffrent in diffrent cities. kd, ___ --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] Grandstream Budge Tone 101 keeps deregistering
Hello, I have a problem with one of three [topic] phones. The phone, which is on the LAN in the same subnet as Asterisk, keeps unregistering from the Asterisk server. Whan it is unregistered there is no way to make a phone call from it, but once it is rang by any other of the phones it registers to Asterisk again. The other two are absolutely fine. The problematic one [ecco] puts this messages into messages log file: Apr 21 15:27:23 NOTICE[1099] chan_sip.c: Auto-congesting SIP/ecco-9091 Apr 21 15:27:23 WARNING[1077] channel.c: Avoided initial deadlock for '0x8129f38', 10 retries! sip.conf: [ecco] type=friend username=ecco defaultip=10.10.129.31 host=dynamic nat=no canreinvite=no qualify=yes dtmfmode=rfc2833 Can somebody please explain what these messages mean? What is '0x8129f38'? Can somebody please post working sip.conf (full) for Grandstream (101) phones. I have discussed this topic with Google for quite a long time, but with no results. Thank you very much. Marcel ___ --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
