[asterisk-users] PRI span debug out put - failing international calls
I have attached my PRI debug out put when making an international call - hopefully it can shed some light on the situation. I am sorry if this attachment gets to the list twice, I sent one early this morning, but it has yet to appear - i may have sent that one in error. Kind Regards: Gabriel -- Making new call for cr 32774 Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 07 21 80 31 32 33 34 35] Calling Number (len= 9) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '12345' ] [70 0e a1 30 30 33 35 33 31 36 36 30 32 33 31 31] Called Number (len=16) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0035316602311' ] [a1] Sending Complete (len= 1) q931.c:2879 q931_setup: call 32774 on channel 1 enters state 1 (Call Initiated) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3414 q931_receive: call 32774 on channel 1 enters state 3 (Outgoing call Proceeding) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] [1e 02 82 88] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Progress Description: Inband information or appropriate pattern now available. (8) ] -- Processing IE 8 (cs0, Cause) -- Processing IE 30 (cs0, Progress Indicator) q931.c:3549 q931_receive: call 32774 on channel 1 enters state 12 (Disconnect Indication) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2715 q931_release: call 32774 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: RELEASE (77) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: RELEASE COMPLETE (90) q931.c:3489 q931_receive: call 32774 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI span debug out put - failing international calls
- Original Message - From: Tony Mountifield [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, 8 December, 2008 10:17:33 GMT +00:00 GMT Britain, Ireland, Portugal Subject: Re: [asterisk-users] PRI span debug out put - failing international calls In article [EMAIL PROTECTED], Mr Gabriel [EMAIL PROTECTED] wrote: I have attached my PRI debug out put when making an international call - hopefully it can shed some light on the situation. I am sorry if this attachment gets to the list twice, I sent one early this morning, but it has yet to appear - i may have sent that one in error. Kind Regards: Gabriel -- Making new call for cr 32774 Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 6/0x6) (Originator) Message type: SETUP (5) [04 03 80 90 a3] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: A-Law (35) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [6c 07 21 80 31 32 33 34 35] Calling Number (len= 9) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '12345' ] This is not correct - you are presenting an internal number as a Caller-ID with a TON of National. You should set a valid Caller-ID in your dialplan before calling Dial(). Or via whatever GUI you might be using. However, this probably isn't the cause of failure - BT should just ignore the Caller-ID. [70 0e a1 30 30 33 35 33 31 36 36 30 32 33 31 31] Called Number (len=16) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) '0035316602311' ] However, I think this is wrong, and probably the cause of the failure. It is saying that you have pridialplan=national (the default), but you are giving a complete number. This is effectively dialling the number '00035316602311'. The first thing to make sure is that your pridialplan= and xxxprefix= directives in zapata.conf are BEFORE the channels to which they apply. When you have a channel= directive in the file, those channels will be created with the parameters that have ALREADY been seen in the file, and any parameters that come later, won't apply to those channels. Also, don't forget you need to restart Asterisk if you change the details in zapata.conf (perhaps reload might be enough, but I'm never sure). [a1] Sending Complete (len= 1) q931.c:2879 q931_setup: call 32774 on channel 1 enters state 1 (Call Initiated) Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] -- Processing IE 24 (cs0, Channel Identification) q931.c:3414 q931_receive: call 32774 on channel 1 enters state 3 (Outgoing call Proceeding) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 6/0x6) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] This response field is telling you there is no such number as 00035316602311 [remainder snipped] Things you need to try, exactly: 1) Make sure the pri and prefix directives are before the channel list. 2) Change to pridialplan=unknown and try dialling both UK and Ireland. 3) Change to pridialplan=dynamic with nationalprefix=0 and internationalprefix=00, and try dialling both UK and Ireland. If that still doesn't work, and you are happy to give me remote ssh access, email me privately. **Gabriel says** I have made the amendments as advised, but the issues still exists - I have attached another PRI debug of attempted international call, and also the zapata.conf - I am at a lost, because we feel that everything is actually complete - and help will be appreciated. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Making new call for cr 32770 Protocol Discriminator: Q.931 (8) len=41 Call Ref: len= 2 (reference 2/0x2) (Originator) Message type: SETUP (5) [04 03
[asterisk-users] International Calls still failing - Confused!
My international calls are not connecting. [general] pridialplan=dynamic ;prilocaldialplan=unknown internationalprefix=00 nationalprefix=0 localprefix= I have the above in my zapta.conf - yet when I dial an international number, I get a ring, then I get the message the person you are calling, is currently unavailable This is an ubuntu machine, with a sangoma card, with FreePBX setup, on asterisk 1.4. Incoming calls are working fine - outgoing national, mobile, and local calls are also working fine. I cannot understand why international calls are not working. Any pointers, no matter how outrageous are very, very welcome! Kind Regards: Gabriel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Visual Dial Plan application: Recommendations?
I am found and application called Visual Dialplan - And the idea seems good, apart from it didn't read the dial plans from a freepbx setup. Are there any other applications that I may try, that you guys can recommend? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Check variables on a live system - Is it possible?
Is it possible to check certain varibles on the live system, for example, what the current setting for pridialplan is? I know what is set in the config files, but the behaviour does not reflect this. Can this be checked? Kind Regards: Gabriel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design]
- Original Message - From: Andrew Thomas [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 5 December, 2008 13:49:59 GMT +00:00 GMT Britain, Ireland, Portugal Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] Address added to spam filter. Please do NOT e-mail me again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 05 December 2008 13:27 To: asterisk-users@lists.digium.com Subject: [asterisk-users] top posting again [was: Re: CDR Design] Top posting strikes again: On Fri, Dec 05, 2008 at 01:39:59PM +0200, [EMAIL PROTECTED] wrote: Quote : Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Who wrote that? [snip the rest of the reply] Andrew Thomas wrote: [snip] Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Why didn't you place your reply here? We have archives of the list. We can spot the original message. [snip more useless quoting resulted from top-posting] -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Which address did you add to the spam filter? Was in the asterisk address? If you don't want to recieve any more mails from the list, then you should unregister, not add it to the spam list :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] top posting again [was: Re: CDR Design] - Or was it top posting?
- Original Message - From: Mr Gabriel [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 5 December, 2008 13:58:09 GMT +00:00 GMT Britain, Ireland, Portugal Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] - Original Message - From: Andrew Thomas [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 5 December, 2008 13:49:59 GMT +00:00 GMT Britain, Ireland, Portugal Subject: Re: [asterisk-users] top posting again [was: Re: CDR Design] Address added to spam filter. Please do NOT e-mail me again. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: 05 December 2008 13:27 To: asterisk-users@lists.digium.com Subject: [asterisk-users] top posting again [was: Re: CDR Design] Top posting strikes again: On Fri, Dec 05, 2008 at 01:39:59PM +0200, [EMAIL PROTECTED] wrote: Quote : Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Who wrote that? [snip the rest of the reply] Andrew Thomas wrote: [snip] Like I said earlier - the CDR's aren't reliable enough for a billing platform (as you've rightly pointed out) but are OK for very basic call logging (something the customer can look at). Why didn't you place your reply here? We have archives of the list. We can spot the original message. [snip more useless quoting resulted from top-posting] ** SNIP ** Which address did you add to the spam filter? Was in the asterisk address? If you don't want to recieve any more mails from the list, then you should unregister, not add it to the spam list :) ** SNIP ** I don't think this one was a TOP POST read the whole message! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] BT - ISDN30 - International Calls not working, everything else is fine :(
Dear All, Thank you for taking the time to read this post - I am *confused!* as to why my asterisk setup does not work as it should. I have an ISDN 30 connection for telephony, a Sangoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for the call • National is 020 will result in 20 being sent and dialled, which works • Mobile is 07x will result in 7x being sent and dialled, which works • International 00x[any number of digits] will result in 00x[any number of digits] which does not work I do not see why this does not work. I do know that for every call, the flag sent is national - how can I make sure the correct flag is sent for the call? By flag, I mean the TON, (type of number) Any assistance will be greatly appreciated. Thank you Mr Gabriel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] BT - ISDN30 - International Calls not working, everything else is fine :(
- Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, 4 December, 2008 12:01:54 GMT +00:00 GMT Britain, Ireland, Portugal Subject: Re: [asterisk-users] BT - ISDN30 - International Calls not working, everything else is fine :( On Thu, Dec 04, 2008 at 11:49:50AM +, Mr Gabriel wrote: Dear All, Thank you for taking the time to read this post - I am *confused!* as to why my asterisk setup does not work as it should. I have an ISDN 30 connection for telephony, a Sangoma card, and asterisk installed. Incoming calls, and outgoing calls work 100%. Making an international call, results in silence, or the error message all circuits are busy Numbers being passed to the trunk for the call • National is 020 will result in 20 being sent and dialled, which works • Mobile is 07x will result in 7x being sent and dialled, which works • International 00x[any number of digits] will result in 00x[any number of digits] which does not work I do not see why this does not work. I do know that for every call, the flag sent is national - how can I make sure the correct flag is sent for the call? By flag, I mean the TON, (type of number) Any assistance will be greatly appreciated. Look into pridialplan in zapata.conf / chan_dahdi.conf . I'm not sure if 'pridialplan = unknown' is applicable. If not: something of the sort of internationalprefixx should help. Gabriel Says The pridialplan is set to pridialplan=unknown, and internationalprefix=00, I have rebooted a few times, so I know this is what is currently loaded. I am using freepbx as a web interface, is it possible that there are conflicting settings? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7910 and Asterisk
Dear All, Has anyone got these phones to work with Asterisk? I have about 20 of them, and not very long to get them working :) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Cisco 7910 Handsets: Skinny protocol?
Dear all. I have about 30 Cisco 7910 handsets, and my basic research has told me that they are not SIP based handsets. Not to worry for now, I just need them to connect to my asterisk server. They are giving me a bit of a hard time. Has anyone here had any experience on how to do this? Documentation on the internet is seriously lacking. Best regards, Mr Gabriel Ogunleye IT Administrator Fusis Group Fusis House, 4 Maple Grove Business Centre Lawrence Road Hounslow, TW4 6DR, UK T: +44(0)845 9000 375 F: +44(0)845 9000 376 M: +44(0)7956 540 134 E: [EMAIL PROTECTED] W: www.fusis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Linksys PAP2 NA
Dear all, I have managed to connect this device to my asterisk box, but it is giving me a bit of a hard time. I can call other extensions from this box, but I am not able to call this one. It seems to permanently remain engaged. When I dial it, this is the message I get. Is there a know issue in this regard? What can I do to test the actual current status of the line? Best regards, Mr Gabriel Ogunleye IT Administrator Fusis Group Fusis House, 4 Maple Grove Business Centre Lawrence Road Hounslow, TW4 6DR, UK T: +44(0)845 9000 375 F: +44(0)845 9000 376 M: +44(0)7956 540 134 E: [EMAIL PROTECTED] W: www.fusis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys PAP2 NA
There is no NAT involved, just a straight connection Mr Gabriel Ogunleye IT Administrator -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Joakimsen Sent: 18 January 2008 05:23 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Linksys PAP2 NA What message? NAT? On Jan 17, 2008 8:18 PM, Mr Gabriel Ogunleye [EMAIL PROTECTED] wrote: Dear all, I have managed to connect this device to my asterisk box, but it is giving me a bit of a hard time. I can call other extensions from this box, but I am not able to call this one. It seems to permanently remain engaged. When I dial it, this is the message I get. Is there a know issue in this regard? What can I do to test the actual current status of the line? Best regards, Mr Gabriel Ogunleye IT Administrator Fusis Group Fusis House, 4 Maple Grove Business Centre Lawrence Road Hounslow, TW4 6DR, UK T: +44(0)845 9000 375 F: +44(0)845 9000 376 M: +44(0)7956 540 134 E: [EMAIL PROTECTED] W: www.fusis.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Linksys PAP2 NA
There is no NAT involved. I think I will try to sip set debug. What exactly should I be looking for? How did you configure these devices - maybe something I missed in the config? Mr Gabriel Ogunleye IT Administrator -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Atis Lezdins Sent: 18 January 2008 01:33 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Linksys PAP2 NA On 1/18/08, Mr Gabriel Ogunleye [EMAIL PROTECTED] wrote: Dear all, I have managed to connect this device to my asterisk box, but it is giving me a bit of a hard time. I can call other extensions from this box, but I am not able to call this one. It seems to permanently remain engaged. When I dial it, this is the message I get. Is there a know issue in this regard? What can I do to test the actual current status of the line? What do you get? Enable sip set debug in CLI. Is it behind NAT? We have a lot of them successfully working. Sometimes they crash and needs reboot, but generally they are ok. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Now Beta 6 and CISCO IP 7910
Dear All, Thank you for taking the time to read my message. I have just installed Asterisk Now, and it seems to be up and running with no issues on my system. The problem I am facing, is that I cannot find anywhere in the web interface, to assign phones. I have a CISCO IP phone 7910 series, and I wish for this to connect to the asterisk box. What's the best way of going about this? Best regards, Mr Gabriel Ogunleye IT Administrator ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Incoming Lines Confusion
First off, please, for the love of God, don't cremate me, if I should already know the answer to this! I've installed a small setup for an office who wanted to be able to talk to each other instead of having to rely on MSN to communicate. Weird request, I know, but hey, we do what we need to do to get paid. I installed soft phones, gave everyone an extension, and bingo, they can call and talk over their PCs happy as hell. Which brings me to my problem - they loved the system so much, that now, they want it for ALL their calls, that is their calls that involve the real world. ATM, they all have separate independent land lines, which is why they had a problem in the first place, large bills for calling each other, now they want a VoIP solution, that would have calls coming in over their broadband connection, and automatically route to each of their phones, depending on which line has called them. I just got out of a meeting with them, and what they want, goes as such... Bob has a VoIP number 020-xxx-xxx - when this number rings, the box answers the call, plays some music, while it waits for Bob to answer. This call should only go to Bobs extension. If he's not there, it routes the call to his voicemail Mary has a different number. When this number rings, it gets routed straight to her extension, in the same manner as Bobs, but if she's not available, it looks for who is, and rings their phones, and if no one answers, then goes to voice mail. Basically, there are 2 types of behaviours that they would like on their lines. My problem, is how to implement it! I'm an asterisk virgin, and getting them to be able to talk to each other across their office network and 12 extensions, took the best part of 2 hours - I don't want to have to spend a whole day working on this one. The VoIP numbers have already been purchased, and are ready to go - i just need to configure it all - Can it be done!? ___ --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
