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Lane Hoskins <> wrote:> I have come to a stumbling block. > > We have 8 lines coming into an ADTRAN channelbank that then goes to > the * server via a T100P card. I need to route lines 1 and 2 to > everyone when a call comes in on either of them. I also need lines 3 > - 8 to ring first at specific sip extensions (direct dials for staff > here) and then to go to voicemail or fwd to a cellphone after that if > the extension is not answered. Has anyone done this that could > provide an example for me or point me to better documentation? We > have searched extensively and not found anything yet. > > Lane Hoskins, MCP > Network Engineer > 540.767.7626I have not done it yet, but it would seem to me that the key to this exercise would be having 7 contexts: 1 for lines 1+2 (which rings all lines or a queue or IVR/ACD) and then one for each line 3-8. This means that each of your incoming lines can have their very own s extension. You can define each line's context in the .conf in Asterisk's etc directory. Hope this helps, David Gomillion
On Tue, 2004-01-13 at 07:52, Lane Hoskins wrote:> We have 8 lines coming into an ADTRAN channelbank that then goes to > the * server via a T100P card. I need to route lines 1 and 2 to > everyone when a call comes in on either of them. I also need lines 3 ? > 8 to ring first at specific sip extensions (direct dials for staff > here) and then to go to voicemail or fwd to a cellphone after that if > the extension is not answered. Has anyone done this that could > provide an example for me or point me to better documentation? We have > searched extensively and not found anything yet.You need to understand more about contexts. If you put lines 1 and 2 in a context (let's call it [everyone]) and each of the other lines in it's own context (let's say [line3], [line4], etc.), then you can control what happens in each context. If you haven't figured out where to assign a context to each line, it's in your /etc/asterisk/zapata.conf file. After setting those in zapata.conf, your (very simplified) extensions.conf file will look something like this: [everyone] ; ring everyone exten=>s,1,Answer() exten=>s,2,Dial(SIP/John&SIP/Mary&SIP/Fred&SIP/Bob) [line3] exten=>s,1,Answer() exten=>s,2,Dial(SIP/John,20,r) exten=>s,3,Dial(<John's cellphone goes here>,10,r) exten=>s,4,VoiceMailMain(<John's mailbox>) exten=>s,5,Hangup() exten=>s,103,Dial(<John's cellphone goes here>,10,r) exten=>s,104,VoiceMailMain(<John's mailbox>) exten=>s,105,Hangup() exten=>s,204,VoiceMailMain(<John's mailbox>) exten=>s,205,Hangup() [line4] exten=>s,1,Answer() exten=>s,2,Dial(SIP/Mary,20,r) exten=>s,3,Dial(<Mary's cellphone goes here>,10,r) exten=>s,4,VoiceMailMain(<Mary's mailbox>) exten=>s,5,Hangup() exten=>s,103,Dial(<Mary's cellphone goes here>,10,r) exten=>s,104,VoiceMailMain(<Mary's mailbox>) exten=>s,105,Hangup() exten=>s,204,VoiceMailMain(<Mary's mailbox>) exten=>s,205,Hangup() ... etc., etc. ... Hope that gets you started... While this should work, I take no responsibility for typos and or stupid mistakes I may have made while writing this in a hurry... Jared Smith
On Tue, 13 Jan 2004, Lane Hoskins waxed:> We have 8 lines coming into an ADTRAN channelbank that then goes to the > * server via a T100P card. I need to route lines 1 and 2 to everyone > when a call comes in on either of them. I also need lines 3 - 8 to ring > first at specific sip extensions (direct dials for staff here) and then > to go to voicemail or fwd to a cellphone after that if the extension is > not answered. Has anyone done this that could provide an example for me > or point me to better documentation? We have searched extensively and > not found anything yet.Here's Rich Adamson's "A WORKING EXAMPLE" from September: http://lists.digium.com/pipermail/asterisk-users/2003-September/020944.html I see SIP and Voicemail in there, but I haven't tried it myself. --Chris -- Chris Maj <cmaj_hat_freedomcorpse_hot_info> Pronunciation Guide: Maj == May Fingerprint: 43D6 799C F6CF F920 6623 DC85 C8A3 CFFE F0DE C146
Thanks David, That is exactly what we had to do. We got some help from Digium as well and have it taken care of. Lane Hoskins, MCP Network Engineer 540.767.7626 -----Original Message----- From: David Gomillion [mailto:dgomillion@eyecarenow.com] Sent: Tuesday, January 13, 2004 10:33 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] inbound call routing problem Lane Hoskins <> wrote:> I have come to a stumbling block. > > We have 8 lines coming into an ADTRAN channelbank that then goes to > the * server via a T100P card. I need to route lines 1 and 2 to > everyone when a call comes in on either of them. I also need lines 3 > - 8 to ring first at specific sip extensions (direct dials for staff > here) and then to go to voicemail or fwd to a cellphone after that if > the extension is not answered. Has anyone done this that could > provide an example for me or point me to better documentation? We > have searched extensively and not found anything yet. > > Lane Hoskins, MCP > Network Engineer > 540.767.7626I have not done it yet, but it would seem to me that the key to this exercise would be having 7 contexts: 1 for lines 1+2 (which rings all lines or a queue or IVR/ACD) and then one for each line 3-8. This means that each of your incoming lines can have their very own s extension. You can define each line's context in the .conf in Asterisk's etc directory. Hope this helps, David Gomillion _______________________________________________ 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