[Asterisk-Users] MeetMe - Party's are not exchanging Audio - Is this BUG?
Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Application: MeetMe Data: |edwx ActionID: ffe4563 When I use the above, Incoming call will be generated to SIP/111 and when it accepts the call, a message shall be played from asterisk which is like - " You are entering conference number 0 and conference will begin as soon as leader arrives". This is fine. Now I shall give another command - Action: Originate Channel: SIP/101 Application: MeetMe Data: 0|aEp ActionID: ffe4563 As soon as I do the above, Incoming call is generated to SIP/101 (leader) and when he accepts the call it plays the message - "You are joinin g the conference number 0". This is fine. But now, when SIP/111 talks SIP/101 (leader) is able to hear. But when SIP/101(leader) talks, SIP/111 is not able to hear anything... Is this a BUG in MeetMe? Please clarify the same. I am using asterisk-1.2.0 and zaptel - ztdummy are installed. Regards, Somesh S. Shanbhag Bring words and photos together (easily) with PhotoMail - it's free and works with your Yahoo! Mail.___ --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] Re: MeetMe - Party's are not exchanging Audio - Is this BUG?
Hi All, Please help me solving this problem. Thanks Somesh S. ShanbhagSomesh S Shanbhag [EMAIL PROTECTED] wrote: Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Application: MeetMe Data: |edwx ActionID: ffe4563 When I use the above, Incoming call will be generated to SIP/111 and when it accepts the call, a message shall be played from asterisk which is like - " You are entering conference number 0 and conference will begin as soon as leader arrives". This is fine. Now I shall give ano ther command - Action: Originate Channel: SIP/101 Application: MeetMe Data: 0|aEp ActionID: ffe4563 As soon as I do the above, Incoming call is generated to SIP/101 (leader) and when he accepts the call it plays the message - "You are joining the conference number 0". This is fine. But now, when SIP/111 talks SIP/101 (leader) is able to hear. But when SIP/101(leader) talks, SIP/111 is not able to hear anything... Is this a BUG in MeetMe? Please clarify the same. I am using asterisk-1.2.0 and zaptel - ztdummy are installed. Regards, Somesh S. Shanbhag Bring words and photos together (easily) with PhotoMail - it's free and works with your Ya hoo! Mail. Bring words and photos together (easily) with PhotoMail - it's free and works with your Yahoo! Mail.___ --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: Configuring Meeting Room from Asterisk Manager API
Hi Alexander, Thanks for the quick response. Actually I tried out this. I tried like - Action: Originate Channel: SIP/111 Application: MeetMe Data: |qdwx ActionID: ffe56637 But actually, it invites 111 and when 111 accepts the call, it will ask for conference number and places 111 into conference with confno which the user keys in. But my requirement is slightly different. 111(user)ServerAsterisk Actually, Server must be able to setup the conference room for 111 without having 111 bothering with keying in confnum. How can I do that from Manager API's? Do I have to use some DialPlan's? Please solve the doubt. Regds, Somesh S. Shanbhag Alexander Chemeris [EMAIL PROTECTED] wrote: Somesh,On 2/3/06, Somesh S Shanbhag wrote: I want to do a three-party conferencing using manager api. But I found out from the asterisk-users list that I *MUST* use the meeting room concept. I wanted to know wheather meeting room can be configured dynamically? on the fly? Otherwise, configuring meeting room statically is not scalable.First search for 'dynamic conferences' on voip-info.org. There you'llfind macro to create dynamic conferences on the fly. Main idea is toenable dynamic creation of meetme rooms and create them according touser phone number.See also Originate command in manager actions reference.You may use command similar to this:Action: OriginateChannel: SIP/4Application: MeetMeData: 41|adEpqActionID: MeetMe-idCallerID: MeetMe-caller-idUse 'Channel' to specify user you want to add, and you may use'CallerID' to track following events. Bring words and photos together (easily) with PhotoMail - it's free and works with your Yahoo! Mail.___ --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] Configuring Meeting Room from Asterisk Manager API
Hi All, I want to do a three-party conferencing using manager api. But I found out from the asterisk-users list that I *MUST* use the meeting room concept. I wanted to know wheather meeting room can be configured dynamically? on the fly? Otherwise, configuring meeting room statically is not scalable. Thanks Regards, Somesh S. Shanbhag Bring words and photos together (easily) with PhotoMail - it's free and works with your Yahoo! Mail.___ --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] Re: Configuring Meeting Room from Asterisk Manager API
Hi All, Please help in this regard. Regds, Somesh S. ShanbhagSomesh S Shanbhag [EMAIL PROTECTED] wrote: Hi All, I want to do a three-party conferencing using manager api. But I found out from the asterisk-users list that I *MUST* use the meeting room concept. I wanted to know wheather meeting room can be configured dynamically? on the fly? Otherwise, configuring meeting room statically is not scalable. Thanks Regards, Somesh S. ShanbhagBring words and photos together (easily) with PhotoMail - it's free and works with your Yahoo! Mail. Bring words and photos together (easily) with PhotoMail - it's free and works with your Yahoo! Mail.___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with Action:Originate with ASterisk Manager
Hi Asterisk-users, I am working with Aterisk Manager API's. I can login successfuly with the following. char buff[256]; strcpy(buff, "Action: Login\r\nUsername: admin\r\nSecret: unix\r\n\r\n"); send(msock, buff, 255); Now I want to try Action: Originate, therefore I tried the following char buff1[256]; strcpy(buff1, "Action: Originate\r\nChannel: SIP/101\r\nExten: 102\r\nPriority: 1\r\nContext: default\r\n\r\n"); send(msock, buff1, 255); But I get the following error response from Asterisk-Manager Response: Error Message: Missing action in request Later I enabled the DEBUG Log in Asterisk I can see the following - During the Login *CLI Jan 10 12:26:33 DEBUG[24230]: manager.c:1253 process_message: Manager received command 'Login' == Parsing '/etc/asterisk/manager.conf': Found == Manager 'admin' logged on from 172.16.25.17 During the Action Originate Jan 10 12:26:33 DEBUG[24230]: manager.c:1253 process_message: Manager received command ''My sip.conf [101] type=friend username=101 host=dynamic nat=yes canreinvite=yes disallow=all allow=ulaw allow=alaw dtmfmode=inband [102] type=friend username=102 host=dynamic nat=yes canreinvite=yes disallow=all allow=ulaw allow=alaw dtmfmode=inband My extesions.conf [default] exten=101,1,Dial(SIP/101,30,Ttrf) exten=102,1,Dial(SIP/102,30,Ttrf) What is the actual problem here? Am I doing some mistake? Please help me in this regard. Thanks regards, Somesh S. Shanbhag Yahoo! Photos Showcase holiday pictures in hardcover Photo Books. You design it and well bind it!___ --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] How to get To-header and From-header display name
Hi All, How can I access the To-header and From-header display name in Asterisk? Example: To: Carol [EMAIL PROTECTED] From: Alice [EMAIL PROTECTED] How to access the strings Carol / Alice from Asterisk? I have to access the same, may be, in extensions.conf. But How do I read the display names? Please help me with this topic. Thanks for your valuable time Regards, Somesh S. Shanbhag __ Yahoo! DSL Something to write home about. Just $16.99/mo. or less. dsl.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] Asterisk as Softswitch
Hi, I had doubt like can asterisk talk ISUP over SS7 which the normal PSTN softswitches talk with other switches? It becomes *necessary* that asterisk *should* talk with other softswitches in PSTN using ISUP/SS7 ?? Regards, Somesh S. Shanbhag --- Olle E. Johansson [EMAIL PROTECTED] wrote: Somesh S Shanbhag wrote: Dear All, Can I use Asterisk IP-PBX as Softswitch? If not, what is lacking in asterisk from not *becoming* softswitch? What is your definition of a softswitch? /O __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.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 as Softswitch
Dear All, Can I use Asterisk IP-PBX as Softswitch? If not, what is lacking in asterisk from not *becoming* softswitch? Thanks Regards, Somesh S. Shanbhag __ Yahoo! Music Unlimited Access over 1 million songs. Try it free. http://music.yahoo.com/unlimited/ ___ --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: R: [Asterisk-Users] Problem setting up TDM22B card
modprobe zaptel is successful. When I do lsmod zaptel is loaded. Regards, Somesh S. Shanbhag --- Lyle Giese [EMAIL PROTECTED] wrote: I have not seen the output of modprob zaptel in this thread, which has to take place before loading the other kernel drivers. Lyle so mesh s wrote: Hi, I changed the mother board (MB) but it is giving still the same problem. ouput of dmesg|tail f6 != 58 f7 != 59 f8 != 58 f9 != 59 fa != 58 fb != 59 fc != 58 fd != 59 fe != 58 Freshmaker failed register test and I have also configured zaptel.conf correctly. Whatz next? Can I assume that it is a hardware problem? Regards, Somesh S. Shanbhag --- John Novack [EMAIL PROTECTED] wrote: somesh s wrote: Hi, I didn't get any solution in the mailing list. [http://asterisk.linkx.net/asteriskusers/200409/msg01167] What should be the next step? Changing the machine??? Is it machine dependent?... Regards, Somesh S. Shanbhag Have you talked with Digium support? Their answer almost always is: Try another Motherboard They won't supply a list that is known to work, only ones that are known NOT to work. From my limited experience, even if the MB says it is PCI 2.2, the TDM card may or may not work. If you don't want to change machines, then use an ATA or two Sipura's work great. John Novack ___ --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 __ Yahoo! Mail - PC Magazine Editors' Choice 2005 http://mail.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
