[Asterisk-Users] how can i let the user in 1th Asterisk can call the user in 2nd Asterisk?
Hi list, I set up two asterisk servers , 1001 is the first asterisk servers sip user, and 2001 is the second asterisk servers sip user. Each of them work well, but I don't konw how to connect them. I want to let the user in 1th Asterisk can call the user in 2nd Asterisk. First asterisk server ip : 192.168.3.101 Second asterisk server ip : 192.168.3.102 can someone give me some ideas about how to write this configuration in asterisk config files and which conf file should i use? Thanks, Erdem HAKI ___ --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 restrictions
Hello, I want to use call restriction option. For example, there are 3 registered numbers that 100,200 and 300.I want 100 to call 200 but not 300, btw 300 can call both 100 and 200. How can i configure this? Thanks. Erdem HAKI ___ --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] Eyebeam Video+Nat
Try this; nat=yes qualify=yes Erdem HAKI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Guillaume Sent: Tuesday, July 26, 2005 2:15 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Eyebeam Video+Nat Hi, I test the video on asterisk with eyebeam. When I use a public IP for the softphone, the video work. However, when I test eyebeam under nat the video doesnt work. I use a routeur linksys WRT54G. I try also to configure my laptop under DMZ for redirect all the traffic IP and the video doesnt work too. Can you help me please? Sincerely, ___ 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 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 users from Turkey?
Yes, i have some but what kind of experience? Erdem HAKI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ceyhun KIRMIZITAS Sent: Saturday, July 23, 2005 12:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk users from Turkey? Is there any1 who has some experience with Asterisk in Turkey? ___ 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 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] Meet Me - this is not a valid conference number, please try again
Hello; I have already the same problem, i can't solve this. Is there anybody to help me? Thanks Erdem HAKI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erdem HAKİ Sent: Friday, July 15, 2005 4:44 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Meet Me - this is not a valid conference number,please try again I did what you said, but still not working :( [EMAIL PROTECTED] ~]# modprobe ztdummy [EMAIL PROTECTED] ~]# lsmod Module Size Used by ztdummy 3924 0 md5 4161 1 ipv6 259201 20 parport_pc 28421 1 lp 12489 0 . . . . . . . . . Thanks Erdem HAKI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki Sent: Friday, July 15, 2005 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Meet Me - this is not a valid conference number, please try again will the lsmod list show you ztdummy modul? if not, modprobe ztdummy I think without a timer source meetme won't work Erdem HAKİ wrote: Hello, I'm trying Meet Me Feature. I read wiki , searched google and i configured my extension.conf and meetme.conf. But I receive this is not a valid conference number, please try again message, so what could be the problem? Thanks for your interest. Erdem HAKI ___ 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 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 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 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] Meet Me - this is not a valid conference number, please try again
Hello, Im trying Meet Me Feature. I read wiki , searched google and i configured my extension.conf and meetme.conf. But I receive this is not a valid conference number, please try again message, so what could be the problem? Thanks for your interest. Erdem HAKI ___ 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] Meet Me - this is not a valid conference number, please try again
I tried, but it doesnt work. You can see my conf files, is there a problem related to conf files? Could you check it? My meetme.conf file [rooms] conf = 1000 conf = 4000 conf = 9000 conf = 9001,123456 My extensions.conf file exten = 9000,1,MeetMe(9000) Thanks for your help. Erdem HAKI From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, July 15, 2005 2:50 PM To: asterisk-users@lists.digium.com Subject: RES: [Asterisk-Users] Meet Me - this is not a valid conference number,please try again Hello Haki I fixed this problem following the instructions in /usr/src/zaptel-1.0.9/README.udev. Regards Cecília -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]Em nome de Erdem HAKI Enviada em: sexta-feira, 15 de julho de 2005 05:11 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Assunto: [Asterisk-Users] Meet Me - this is not a valid conference number,please try again Hello, Im trying Meet Me Feature. I read wiki , searched google and i configured my extension.conf and meetme.conf. But I receive this is not a valid conference number, please try again message, so what could be the problem? Thanks for your interest. Erdem HAKI ___ 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] Meet Me - this is not a valid conference number, please try again
As for MeetMe, I do not have a Zaptel card. My kernel above 2.6. so is ztdummy required? Because I configured conf files but still doesnt work. I think that something is missing. Thanks Erdem HAKI From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, July 15, 2005 3:49 PM To: asterisk-users@lists.digium.com Subject: RES: [Asterisk-Users] Meet Me - this is not a valid conference number,please try again I´ve got just one room configured. These´s my configuration files .. My extensions.conf file ; Or a conference room (you'll need to edit meetme.conf to enable this room) ; exten = 8600,1,Meetme(1234) My meetme.conf file [rooms] ; ; Usage is conf = confno[,pin] ; conf = 1234 Hope it helps Cecília -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]Em nome de Erdem HAKI Enviada em: sexta-feira, 15 de julho de 2005 09:24 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Assunto: RE: [Asterisk-Users] Meet Me - this is not a valid conference number,please try again I tried, but it doesnt work. You can see my conf files, is there a problem related to conf files? Could you check it? My meetme.conf file [rooms] conf = 1000 conf = 4000 conf = 9000 conf = 9001,123456 My extensions.conf file exten = 9000,1,MeetMe(9000) Thanks for your help. Erdem HAKI From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, July 15, 2005 2:50 PM To: asterisk-users@lists.digium.com Subject: RES: [Asterisk-Users] Meet Me - this is not a valid conference number,please try again Hello Haki I fixed this problem following the instructions in /usr/src/zaptel-1.0.9/README.udev. Regards Cecília -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]Em nome de Erdem HAKI Enviada em: sexta-feira, 15 de julho de 2005 05:11 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Assunto: [Asterisk-Users] Meet Me - this is not a valid conference number,please try again Hello, Im trying Meet Me Feature. I read wiki , searched google and i configured my extension.conf and meetme.conf. But I receive this is not a valid conference number, please try again message, so what could be the problem? Thanks for your interest. Erdem HAKI ___ 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] Meet Me - this is not a valid conference number, please try again
I did what you said, but still not working :( [EMAIL PROTECTED] ~]# modprobe ztdummy [EMAIL PROTECTED] ~]# lsmod Module Size Used by ztdummy 3924 0 md5 4161 1 ipv6 259201 20 parport_pc 28421 1 lp 12489 0 . . . . . . . . . Thanks Erdem HAKI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kib Eki Sent: Friday, July 15, 2005 4:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Meet Me - this is not a valid conference number, please try again will the lsmod list show you ztdummy modul? if not, modprobe ztdummy I think without a timer source meetme won't work Erdem HAKİ wrote: Hello, I'm trying Meet Me Feature. I read wiki , searched google and i configured my extension.conf and meetme.conf. But I receive this is not a valid conference number, please try again message, so what could be the problem? Thanks for your interest. Erdem HAKI ___ 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 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 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] Meet Me - this is not a valid conferencenumber, please try again
[EMAIL PROTECTED] ~]# modprobe ztdummy [EMAIL PROTECTED] ~]# lsmod Module Size Used by ztdummy 3924 0 md5 4161 1 ipv6 259201 20 parport_pc 28421 1 lp 12489 0 parport40201 2 parport_pc,lp autofs426181 0 sunrpc164485 1 zaptel208132 1 ztdummy crc_ccitt 2113 1 zaptel Thanks Erdem HAKI -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dave Cotton Sent: Friday, July 15, 2005 5:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Meet Me - this is not a valid conferencenumber, please try again On Fri, 2005-07-15 at 16:44 +0300, Erdem HAKİ wrote: [EMAIL PROTECTED] ~]# modprobe ztdummy [EMAIL PROTECTED] ~]# lsmod Module Size Used by ztdummy 3924 0 md5 4161 1 ipv6 259201 20 parport_pc 28421 1 lp 12489 0 . . . Where's zaptel? modprobe ztdummy should have loaded zaptel. -- Dave Cotton [EMAIL PROTECTED] ___ 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 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 we use asterisk as a SIP Redirect Server?
can we use asterisk as a SIP Redirect Server? Thanks Erdem HAKI [EMAIL PROTECTED] ___ 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] Users handbook
I just wonder what can i do with asterisk and its limits. For example i really don't know yet is asterisk used as redirect server? Thanks for your reply, Erdem HAKI - [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew M Stemen Sent: Wednesday, July 06, 2005 1:29 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Users handbook I'm planning on implementing an Asterisk system at a couple of small offices, and a couple of homes, in the near future... and I don't have any documentation yet. What you're suggesting sounds wonderful, to me. I would contribute, if I had anything... but making it an inclusive manual would be a good idea, I think... you can always edit/remove sections. :) Andrew M Stemen [EMAIL PROTECTED] http://www.andrewmstemen.com Chris Mason (Lists) wrote: Mark Phillips wrote: This is somewhat unique to the site installation. For example, I don't have *69 programmed at my site because frankly there's no need for it with the Cisco 7960's. I do however have an automatic conference booking utility and a speaking clock. Not often found in smaller sites. I think you are on your own here. If one is implementing an Asterisk solution in an office scenario, it has to have fairly similar features to another Asterisk installation. It's easy enough to edit and remove the parts that are different. What I am suggesting is a comprehensive Here's everything Asterisk can do out of the box document, change or remove what doesn't apply. Let me know if any of you want to pool the work we have already done, I will compile to a complete document and post on the wiki. ___ 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 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] voice mail problem
Hi, Perhaps Im wrong but if you use g729 with no translation (pass-thru) you cant hear voice mail. Set your codec to gsm or g711 and try again. Erdem HAKI [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Betül Gözlükoğlu Sent: Thursday, June 30, 2005 1:41 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] voice mail problem Hi; Have a BUDGETONE-100 and using it with asteriskProblem occurs when I dial message centerMessage center does not accept tones (password for example) that I dial, Behaves as I do not dial any number and asks for the password againChanged the DTMF Mode from in-audio to RTP(RFC2833) it works but this time, dialing internal numbers over telephony system is denied Does anybody has any idea about correct configuration on Asterisk or Budgetone? Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli olabilir, yasalar ya da anlasmalar geregi ucuncu kisiler ile paylasilmasi mümkün olmayabilir. Mesaji alan kisi, mesajin gönderilmek istendigi kisi veya kurulus degilse, bu mesaji yaymak, dagitmak veya kopyalamak yasaktir. Mesaj tarafiniza yanlislikla ulasmis ise tarafimiza telefon ile derhal bilgi vermenizi ve orijinal mesaji yukarida belirtilen adrese geri gondermenizi ve imha etmenizi rica ederiz. Tesekkürler - Hassangroup Important note : This e-mail transmission is intended only for the use of the individual or entity to which it is addressed, and may contain information that is privileged, confidential and that may not be made public by law or agreement. If the recipient of this message is not the intended recipient or entity, you are hereby notified that any further dissemination, distribution or copying of this information is strictly prohibited. If you have received this communication in error, please notify us immediately by telephone and return the original message to us to the above address or destroy it. Thank you - Hassangroup ___ 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] RTP session between two end users
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Tuesday, June 28, 2005 6:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RTP session between two end users Erdem HAKİ wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Monday, June 27, 2005 8:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RTP session between two end users Erdem HAKİ wrote: Is it possible that a RTP session between two end users (so i want to use asterisk as a signaling proxy and bypass RTP sessions)? I used canreinvite=yes but it didn't work. Description from asterisk conf. File; (canreinvite=yes; allow RTP voice traffic to bypass Asterisk) It's sip.conf. reinvites only work if the codec is the same for the two endpoints and Asterisk does NOT have to listen for DTMF (no t or T on the dial line, no meetme, etc.) *** We use same codec and don't use meetme etc... So what else should i do? How are you determining if RTP audio is going thru Asterisk? Remember, SIP signaling will always go thru Asterisk. Also do a sip show channels during a call to confirm that the codecs are the same. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ___ 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 Hi, I determine signaling with ethereal and i am sure that both sides use the same codec. By the way, i searched forum again and i read something below; In wiki pages it is stated that The audio channels (RTP) may go directly from phone to phone or may go through Asterisk's media bridge. Currently with my settings, I notice that all rtps are passing through my asterisk. How could I achieve that they go directly from phone to phone? I assume this way, my machine will have less load and therefore could handle more calls. As bkw pointed out, use canreinvite=yes for each sip phone definition. But, that will only work if the phones can reach each other directly (the phones and/or asterisk can't be behind a nat/firewall box). Thanks Erdem HAKI [EMAIL PROTECTED] ___ 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] simultaneus calls?
Thanks for your help Bernard, it's realy useful web site, but i also want to know limits which depens on hardware of the box. Any practical experience? Thanks again :-) Erdem HAKI - [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bernard Cresencia Sent: Tuesday, June 28, 2005 8:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] simultaneus calls? I did a google search on 'voip speed test' - the first site is very good. Here's the link: http://www.talkswitch.com/voip/voip_test.php It will test both your download and upload speeds and will let you know how many concurrent calls at different codecs your connection will support. Try it a few times and on different times of the day to get an average. --- Erdem HAKÝ [EMAIL PROTECTED] wrote: Yes it is DSL and outbound speed is aslo 1Mbit, it's a dedicated server and we just use to talk. I look at the web site which you suggested, but i want to learn how many calls supported practically? Any information do you have? Thanks Erdem HAKI - [EMAIL PROTECTED] _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, June 28, 2005 5:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] simultaneus calls? The 1 m internet connection will be the limiting factor in your setup, you did not state what type of internet connection, but given the speed of 1 mbit it must be DSL (or maybe fraction t/e1). Is the outbound speed also 1m? Is there data on the line also? How much? What about voice Qos? You should start here http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erdem HAKI Sent: Tuesday, June 28, 2005 3:04 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] simultaneus calls? Hello, How can i learn my asterisk how many simulyaneus calls support? My configuration: 80 GB HDD, 1 GB Ram, P4 2,8 MHz processor, Fedora Core 3 minimum installation, no digium cards, codecs g729 or gsm, 1Mbit internet connection. Thanks for your interest... Erdem HAKI - [EMAIL PROTECTED] ___ 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 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 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] RTP session between two end users
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Monday, June 27, 2005 8:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RTP session between two end users Erdem HAKİ wrote: Is it possible that a RTP session between two end users (so i want to use asterisk as a signaling proxy and bypass RTP sessions)? I used canreinvite=yes but it didn't work. Description from asterisk conf. File; (canreinvite=yes; allow RTP voice traffic to bypass Asterisk) It's sip.conf. reinvites only work if the codec is the same for the two endpoints and Asterisk does NOT have to listen for DTMF (no t or T on the dial line, no meetme, etc.) *** We use same codec and don't use meetme etc... So what else should i do? Thanks Erdem HAKI *** ___ 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] simultaneus calls?
Hello, How can i learn my asterisk how many simulyaneus calls support? My configuration: 80 GB HDD, 1 GB Ram, P4 2,8 MHz processor, Fedora Core 3 minimum installation, no digium cards, codecs g729 or gsm, 1Mbit internet connection. Thanks for your interest... Erdem HAKI [EMAIL PROTECTED] ___ 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] simultaneus calls?
Yes it is DSL and outbound speed is aslo 1Mbit, its a dedicated server and we just use to talk. I look at the web site which you suggested, but i want to learn how many calls supported practically? Any information do you have? Thanks Erdem HAKI [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Damon Estep Sent: Tuesday, June 28, 2005 5:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] simultaneus calls? The 1 m internet connection will be the limiting factor in your setup, you did not state what type of internet connection, but given the speed of 1 mbit it must be DSL (or maybe fraction t/e1). Is the outbound speed also 1m? Is there data on the line also? How much? What about voice Qos? You should start here http://www.voip-info.org/tiki-index.php?page=Bandwidth%20consumption From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erdem HAKI Sent: Tuesday, June 28, 2005 3:04 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] simultaneus calls? Hello, How can i learn my asterisk how many simulyaneus calls support? My configuration: 80 GB HDD, 1 GB Ram, P4 2,8 MHz processor, Fedora Core 3 minimum installation, no digium cards, codecs g729 or gsm, 1Mbit internet connection. Thanks for your interest... Erdem HAKI [EMAIL PROTECTED] ___ 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] RTP session between two end users
Is it possible that a RTP session between two end users (so i want to use asterisk as a signaling proxy and bypass RTP sessions)? I used canreinvite=yes but it didnt work. Description from asterisk conf. File; (canreinvite=yes ; allow RTP voice traffic to bypass Asterisk) Thanks Erdem HAKI [EMAIL PROTECTED] ___ 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] RTP session between two end users
Is it possible that a RTP session between two end users (so i want to use asterisk as a signaling proxy and bypass RTP sessions)? I used canreinvite=yes but it didnt work. Description from asterisk conf. File; (canreinvite=yes ; allow RTP voice traffic to bypass Asterisk) Thanks Erdem HAKI [EMAIL PROTECTED] ___ 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 divert to TRUNK , if one number is unregistered?
I have a question. I have two numbers on Asterisk like 902121234567 and 902123645789 and i want to divert first numbers call to Trunk if second number is unregistered. Is it possible? İf yes, how? Flow Diagram: *Two numbers are registered on Asterisk 902121234567 registered to Asterisk 902123645789 registered to Asterisk *One number is registered, other one is not registered 902121234567 registered to Asterisk 902123645789-x not registered to Asterisk *So first number want to make a call second one (desired situation) 902121234567 Asterisk à Trunk Thanks for your interest. Erdem HAKI [EMAIL PROTECTED] ___ 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] call divert to TRUNK , if one number is unregistered?
Yes it works :) but i need to add voicemail option, how can i do this? I want a configuration like this exten = XXX,1, Dial(sip/,20,r) exten = XXX,2,Dial(zap/) exten = XXX,3,Voicemail(u) exten = XXX,103,Voicemail(b) -what should order number be? exten = XXX,104,Hangup From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mohamed A. Gombolaty Sent: Wednesday, June 22, 2005 2:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] call divert to TRUNK ,if one number is unregistered? Hi Erdem, Can you try to put another dial command that points to the trunk afetr the dial command to the SIP? fro example: exten = XXX,1, dial(sip/,20,r) exten = XXX,2,dial(zap/) - note here that I am not sure if the order number should be 2 or 102 but if this didn't work try the other one. Thx MAG Erdem HAK] wrote: I have a question. I have two numbers on Asterisk like 902121234567 and 902123645789 and i want to divert first number's call to Trunk if second number is unregistered. Is it possible? ]f yes, how? Flow Diagram: *Two numbers are registered on Asterisk 902121234567 registered to Asterisk 902123645789 registered to Asterisk *One number is registered, other one is not registered 902121234567 registered to Asterisk 902123645789-x not registered to Asterisk *So first number want to make a call second one (desired situation) 902121234567 Asterisk` Trunk Thanks for your interest. Erdem HAKI - [EMAIL PROTECTED] ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --ThxMAG ___ 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] help for making several calls at the same time..
Hi I surmounted the problem by myself :), when i add user who has 12 digits number like 902121112233 ,everything works fine. Erdem HAKI [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Erdem HAKİ Sent: Monday, June 20, 2005 9:57 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] help for making several calls at the same time.. Hi, I have installed latest stable version of Asterisk. I registered 1 Quintum Gateway and 5 eyeBeam SoftPhone. But I can make just one call at the same time, if i try to make calls from 2 softphones to anotherone, second caller listens the person have extension is on the phone . So we couldnt make two or more calls at the same time for a SoftPhone. What should we do to make several calls at the same time? Thanks for your interest. Erdem HAKI [EMAIL PROTECTED] ___ 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] help for making several calls at the same time..
Hi, I have installed latest stable version of Asterisk. I registered 1 Quintum Gateway and 5 eyeBeam SoftPhone. But I can make just one call at the same time, if i try to make calls from 2 softphones to anotherone, second caller listens the person have extension is on the phone . So we couldnt make two or more calls at the same time for a SoftPhone. What should we do to make several calls at the same time? Thanks for your interest. Erdem HAKI [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OH323 with g723
Hi, Visit http://aussievoip.com.au/wiki-G723-1-Install you'll find how to install g723, but first you have to install g729 http://aussievoip.com.au/wiki-G729-Install I have tested it with Quintum, it works Enjoy :) Erdem HAKI - [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bashir Ullah - www.Lamsre.Com Sent: Monday, June 20, 2005 11:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] OH323 with g723 hi is there anybody using g723 with oh323 and sending call by asterisk. if so please let me know how i can use this same, i need to call quintum by g723 . Thanks Bashir ___ 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 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] astpp database creation failed!
Hello, I'm setting up AST Post Paid application, is there anybody who set up astpp ? I followed the directions, i visited the astpp admin page in my web browser. But i couldn't setup the brands and routes etc. "Database unavailable -- please check configuration" appeared on the top of the page, so i went to "configure" section, I filled in the blanks according to my username,pass etc.. but I got "Database creation failed!" message... How can i achieve this problem? ___ 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: astpp database creation failed...please help...
so what should astpp db be exactly, where can i find its name? what should i write there? Thanks again.. The Database field should contain the name of the astpp db, something along the lines of astpp is what I would put in there. Here is a fixed version of the script. It did not post properly to the wiki: ___ 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] Areski Calling Card
Does anyone have complete Areski Calling Card directions? I think Areski Calling Card Idiots guide is not complete :( any other guide? ___ 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] video conference feature
Hi, Is there anybody who has a working video conference config? I use [EMAIL PROTECTED] 1.0.7a , I couldn't use Video Conferencing feature of eyeBeam. Thanks Erdem HAKI ___ 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] VIDEO ON 1.0.7 stable
- Original Message - From: Nardis Dome [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, May 26, 2005 1:59 PM Subject: Re: [Asterisk-Users] VIDEO ON 1.0.7 stable --- listas iPfone [EMAIL PROTECTED] wrote: Hi all I need to know if the video support for h.263 is active in version stable 1.0.7 to use with eyeBeam in asterisk it works for me... [] type=friend secret= auth=md5 callerid=myCallerId canreinvite=no host=dynamic disallow=all context=default allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 Thanks Nardis Dome, it shows the way. I use [EMAIL PROTECTED] ,eyeBeam video feature on asterisk didn't work first, but after adding allow=h263p , it has worked properly. [2001] username=2001 type=friend secret=** qualify=no port=5060 pickupgroup= nat=never [EMAIL PROTECTED] host=dynamic dtmfmode=rfc2833 disallow=all context=from-internal canreinvite=no callgroup= callerid=Erdem HAKI 2001 allow=alaw allow=ulaw allow=speex allow=gsm allow=h261 allow=h263 allow=h263p ___ 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
