Tielin Xu
2005-Oct-25 11:15 UTC
[Asterisk-Users] How to configure the communication between two Asterisk servers
Hi All: I have special set up to be done. See anyone can help me some ideas. Two Asterisk servers, server A trunks to PSTN, server B works as call routing engine. All sip phones are registered in server B. I have scenario like following: 1. A call comes to server A, server A sends the call related information to server B, assume that uses fast AGI. 2. Server B receives the message from server A, and look up dial plan for call routing, 3. Serve B sends the extension number back to server A, 4. Server A routes the call to the assigned agent. How does server B receive the message from server A? Many thanks for your help. Tielin Xu CTI Analyst Nintendo of America
I have downloaded SJPhone - and well.. it does connect to my system, however popping audio is heard when i dial my music on hold extension... the quality is really really bad.. i have a WLAN-SDIO utility, the signal strength is at 11mb/s. however. is that sufficient? The codecs for sjphone are fixed at 64000.. i could not change those values. has anyone had successful attempts with something better? Thanks...
Jesse Keating
2005-Oct-25 11:28 UTC
[Asterisk-Users] How to configure the communication between two Asterisk servers
On Tue, 2005-10-25 at 11:15 -0700, Tielin Xu wrote:> How does server B receive the message from server A? > > Many thanks for your help.Nintendo eh? The Redmond office? Thats near where I live. So let me make sure I understand the problem. Server A needs to get information from Server B about where to send the call to, which will most likely be somewhere from Server B, since all SIP phones go to server B? Why not use switch? We do something like that. We have 'Pandora' which is at a remote location connected to PSTN. We have 'Asterisk' which is local and all sip phones are connected to. 'Asterisk' has a context in dialplan that lists all the sip extensions and how to dial them and whatnot. 'Pandora' has a line within the context of the incomign PSTN calls that says: switch => IAX2/Asterisk/sipphones thats it! Basically it 'includes' the sipphones context on Asterisk into the call plan for Pandora. Works great. Does this help you? -- Jesse Keating GameHouse -- Systems Engineer
astgroups
2005-Oct-25 11:28 UTC
[Asterisk-Users] How to configure the communication between two Asterisk servers
Tielin Xu wrote:>Hi All: > >I have special set up to be done. See anyone can help me some ideas. >Two Asterisk servers, server A trunks to PSTN, server B works as call >routing engine. >All sip phones are registered in server B. > >I have scenario like following: >1. A call comes to server A, server A sends the call related >information to server B, > assume that uses fast AGI. >2. Server B receives the message from server A, and look up dial plan >for call routing, >3. Serve B sends the extension number back to server A, >4. Server A routes the call to the assigned agent. > >How does server B receive the message from server A? > >Many thanks for your help. > >Tielin Xu >CTI Analyst >Nintendo of America >_______________________________________________ >--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 > > >You can trunk your two servers through IAX or similar but I sense you are looking for something that goes beyond that though it's not too easy to discern from your message....Why not have server B route the calls to the SIP agents registered on the same server B?
64000 codec sounds like 64Kbps. A 11Mbps WLAN will give roughly 3-4Mbps above the MAC layer so in theory it shall be perfectly fine. In practice you could suffer from RF interferers or ioverloaded WLAN. I would try and run some pings from the PDA to the * and check the quality of the wireless link. Ping will show delay and packet loss. - Arnaud On 10/25/05, pbx@itsngroup.com <pbx@itsngroup.com> wrote:> > I have downloaded SJPhone - and well.. it does connect to my system, > however popping audio is heard when i dial my music on hold extension... > > the quality is really really bad.. > > i have a WLAN-SDIO utility, the signal strength is at 11mb/s. however. is > that sufficient? The codecs for sjphone are fixed at 64000.. i could not > change those values. > > has anyone had successful attempts with something better? > > Thanks... > > _______________________________________________ > --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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051025/58a9816c/attachment.htm
Paul
2005-Oct-25 13:25 UTC
[Asterisk-Users] How to configure the communication between twoAsterisk servers
I would like to know why you are using two servers? Are you out of interface slots? You stated, Sip phones registered to server B, then you say Server A routes the call to an agent. It can not be done that way. If you want server A to receive all calls and forward them to Server B, this is done using an IAX link & inbound routing. Server B receives all line information from the IAX inbound call and server B routes it to the SIP phone based on the dialplan. :) Paul -----Original Message----- From: Tielin Xu [mailto:TIELXU01@noa.nintendo.com] Sent: Tuesday, October 25, 2005 2:16 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] How to configure the communication between twoAsterisk servers Hi All: I have special set up to be done. See anyone can help me some ideas. Two Asterisk servers, server A trunks to PSTN, server B works as call routing engine. All sip phones are registered in server B. I have scenario like following: 1. A call comes to server A, server A sends the call related information to server B, assume that uses fast AGI. 2. Server B receives the message from server A, and look up dial plan for call routing, 3. Serve B sends the extension number back to server A, 4. Server A routes the call to the assigned agent. How does server B receive the message from server A? Many thanks for your help. Tielin Xu CTI Analyst Nintendo of America