Re: [asterisk-users] All circuits are busy now on outgoing trunk call
Yes they did. On 3/26/12, SamyGo govoi...@gmail.com wrote: Good to know, hope our replies did some help :) On Thu, Mar 22, 2012 at 7:39 PM, James Mutuku listmut...@gmail.com wrote: Hi, Thanks for the support. Issue solved. Somehow the routes on the fxo gw were not working. On 3/21/12, James Mutuku listmut...@gmail.com wrote: Hi, I have configured a route on the fxo to send all incoming sip traffic to the fxo ports. I will try set the specific digits and see. On 3/21/12, SamyGo govoi...@gmail.com wrote: 404 NOT FOUND means that they were unable to find any destination/route/rule/prefix match corresponding to your dialled number. See your FXO gateway configuration Web-UI for outbound patterns OR verify that the FXO has its outbound line configured and working properly. On Wed, Mar 21, 2012 at 5:20 PM, James Mutuku listmut...@gmail.com wrote: I am setting up asterisk-fxo gw. 404 Not Found (User not found) means the user is not found, but I don't need to have extensions or authentication on the fxo gw On 3/21/12, Michael L. Young myo...@acsacc.com wrote: [0K --- SIP read from UDP:192.168.9.251:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.9.250:5060 ;rport=5060;branch=z9hG4bK111ef687 To: sip:0722490994@192.168.9.251 ;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06 From: pbxserver sip:Unknown@192.168.9.250;tag=as66c75bd7 CSeq: 102 INVITE Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250 Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO) Content-Length: 0 I think the 404 Not Found being returned from the server is a clue as to what the problem is. Michael L. Young (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [OT] FreePBX + Trunk over VPN + Local LAN
Hi, How about having 2 NIC cards on the PBX(configure the machine as a gw of sorts). On 3/23/12, Sean McMaster sean.mcmas...@msn.com wrote: Hello, First let me apologize for posting about a GUI topic on here. There's a reason why I did that, and it's because the underlying concept of this is connected to Asterisk.Here's my situation: Twenty wifi clients connecting to our wireless router (Cisco Linksys E4200 loaded with Tomato). All these WiFi clients are running eyeBeam (in case you're wondering where the calls come and go from). We're getting a SIP Trunk from a local provider that is poised for 30 lines (ISDN 30) - and don't ask me why, the only way I can configure our FreePBX to connect to them as a trunk is via an IP-VPN provided by them. Anyway, the server is connected by ethernet to our router and has an IP 192.168.1.252 and the other local clients are on 192.168.1.*Problem is I don't know how I can keep the PBX on this subnet, and also connect it via eth to the other vpn modem and give it another IP which is on a 192.168.200.* subnet. Any pointers?Thanks! -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All circuits are busy now on outgoing trunk call
Hi, Thanks for the support. Issue solved. Somehow the routes on the fxo gw were not working. On 3/21/12, James Mutuku listmut...@gmail.com wrote: Hi, I have configured a route on the fxo to send all incoming sip traffic to the fxo ports. I will try set the specific digits and see. On 3/21/12, SamyGo govoi...@gmail.com wrote: 404 NOT FOUND means that they were unable to find any destination/route/rule/prefix match corresponding to your dialled number. See your FXO gateway configuration Web-UI for outbound patterns OR verify that the FXO has its outbound line configured and working properly. On Wed, Mar 21, 2012 at 5:20 PM, James Mutuku listmut...@gmail.com wrote: I am setting up asterisk-fxo gw. 404 Not Found (User not found) means the user is not found, but I don't need to have extensions or authentication on the fxo gw On 3/21/12, Michael L. Young myo...@acsacc.com wrote: [0K --- SIP read from UDP:192.168.9.251:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687 To: sip:0722490994@192.168.9.251 ;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06 From: pbxserver sip:Unknown@192.168.9.250;tag=as66c75bd7 CSeq: 102 INVITE Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250 Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO) Content-Length: 0 I think the 404 Not Found being returned from the server is a clue as to what the problem is. Michael L. Young (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All circuits are busy now on outgoing trunk call
I am still getting the same error On 3/21/12, white hat whitehat...@gmail.com wrote: Just a guess here, but it looks like you are dialing a 10 digit phone number but the dial pattern in your outbound route does not handle that. Try using a different dial pattern in your outbound route such as: 1NXXNXX or 9|1NXXNXX I believe that asterisk is telling you that all circuits are busy because none of the outbound routes can be matched against the pattern you are dialing. On Wed, Mar 21, 2012 at 12:48 AM, SamyGo govoi...@gmail.com wrote: This is not in human readable format, but using my special powers I was able to locate the lines -- Called fxosip/0799490994 -- SIP/fxosip-0015 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) Please enable sip debug for this carrier and then try to send the sip traces in human readable format. From those traces it'd be more clear what is the issue from the carrier end rejecting the calls...Maybe your credit expired !! Regards, Sammy. On Wed, Mar 21, 2012 at 9:36 AM, James Mutuku listmut...@gmail.comwrote: I have setup a trunk on freepbx and the outbound route. Everytime I dial via the trunk, I get all circuits are busy now. Incoming calls are working fine on the trunk. This is my dial 9|XXX. and these are my peer details allow=ulawalaw canredirect=no disallow=all dtmfmode=rfc2833 host=192.168.9.251 insecure=very type=peer Below are the logs. my trunk name is called fxosip Using SIP RTP TOS bits 184 [0K == Using SIP RTP CoS mark 5 [0K-- Called fxosip/0799490994 [0K-- SIP/fxosip-0015 is circuit-busy [0K == Everyone is busy/congested at this time (1:0/1/0) [0K -- Executing [s@macro-dialout-trunk:20][1;36mNoOp [0m( [1;35mSIP/3000-0014 [0m, [1;35mDial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 1 [0m) in new stack [0K-- Executing [s@macro-dialout-trunk:21] [1;36mGoto [0m( [1;35mSIP/3000-0014 [0m, [1;35ms-CONGESTION,1 [0m) in new stack [0K -- Goto (macro-dialout-trunk,s-CONGESTION,1) [0K-- Executing [s-CONGESTION@macro-dialout-trunk:1] [1;36mSet [0m( [1;35mSIP/3000-0014 [0m, [1;35mRC=1 [0m) in new stack [0K-- Executing [s-CONGESTION@macro-dialout-trunk:2] [1;36mGoto [0m( [1;35mSIP/3000-0014 [0m, [1;35m1,1 [0m) in new stack [0K-- Goto (macro-dialout-trunk,1,1) [0K-- Executing [1@macro-dialout-trunk:1] [1;36mGoto [0m( [1;35mSIP/3000-0014 [0m, [1;35mcontinue,1 [0m) in new stack [0K-- Goto (macro-dialout-trunk,continue,1) [0K-- Executing [continue@macro-dialout-trunk:1] [1;36mGotoIf [0m( [1;35mSIP/3000-0014 [0m, [1;35m1?noreport [0m) in new stack [0K -- Goto (macro-dialout-trunk,continue,3) [0K-- Executing [continue@macro-dialout-trunk:3] [1;36mNoOp [0m( [1;35mSIP/3000-0014 [0m, [1;35mTRUNK Dial failed due to CONGESTION HANGUPCAUSE: 1 - failing through to other trunks [0m) in new stack [0K -- Executing [continue@macro-dialout-trunk:4] [1;36mSet [0m( [1;35mSIP/3000-0014 [0m, [1;35mCALLERID(number)=3000 [0m) in new stack [0K-- Executing [90722490994@from-internal:5] [1;36mMacro [0m( [1;35mSIP/3000-0014 [0m, [1;35moutisbusy, [0m) in new stack [0K -- Executing [s@macro-outisbusy:1] [1;36mProgress [0m( [1;35mSIP/3000-0014 [0m, [1;35m [0m) in new stack [0K-- Executing [s@macro-outisbusy:2] [1;36mGotoIf [0m( [1;35mSIP/3000-0014 [0m, [1;35m0?emergency,1 [0m) in new stack [0K -- Executing [s@macro-outisbusy:3] [1;36mGotoIf [0m( [1;35mSIP/3000-0014 [0m, [1;35m0?intracompany,1 [0m) in new stack [0K -- Executing [s@macro-outisbusy:4] [1;36mPlayback [0m( [1;35mSIP/3000-0014 [0m, [1;35mall-circuits-busy-nowpls-try-call-later, noanswer [0m) in new stack [0K-- SIP/3000-0014 Playing 'all-circuits-busy-now.gsm' (language 'en') [0K == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/3000-0014' in macro 'outisbusy' == Spawn extension (from-internal, 90722490994, 5) exited non-zero on 'SIP/3000-0014' Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.phpand find out how our CRM can help you achieve better customer satisfaction and sales _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
Re: [asterisk-users] All circuits are busy now on outgoing trunk call
my sip traces are below Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 Audio is at 192.168.9.250 port 17722 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 192.168.9.251:5060: INVITE sip:0722490994@192.168.9.251 SIP/2.0 Via: SIP/2.0/UDP 192.168.9.250:5060;branch=z9hG4bK111ef687;rport Max-Forwards: 70 From: pbxserver sip:Unknown@192.168.9.250;tag=as66c75bd7 To: sip:0722490994@192.168.9.251 Contact: sip:Unknown@192.168.9.250 Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.6.2.13 Date: Wed, 21 Mar 2012 11:19:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 286 v=0 o=root 1836379524 1836379524 IN IP4 192.168.9.250 s=Asterisk PBX 1.6.2.13 c=IN IP4 192.168.9.250 t=0 0 m=audio 17722 RTP/AVP 0 8 3 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- -- Called 6fxogateway/0722490994 [0K --- SIP read from UDP:192.168.9.251:5060 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687 To: sip:0722490994@192.168.9.251 From: pbxserver sip:Unknown@192.168.9.250;tag=as66c75bd7 CSeq: 102 INVITE Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250 Content-Length: 0 - --- (7 headers 0 lines) --- [0K --- SIP read from UDP:192.168.9.251:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687 To: sip:0722490994@192.168.9.251;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06 From: pbxserver sip:Unknown@192.168.9.250;tag=as66c75bd7 CSeq: 102 INVITE Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250 Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO) Content-Length: 0 - --- (8 headers 0 lines) --- Transmitting (no NAT) to 192.168.9.251:5060: ACK sip:0722490994@192.168.9.251 SIP/2.0 Via: SIP/2.0/UDP 192.168.9.250:5060;branch=z9hG4bK111ef687;rport Max-Forwards: 70 From: pbxserver sip:Unknown@192.168.9.250;tag=as66c75bd7 To: sip:0722490994@192.168.9.251;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06 Contact: sip:Unknown@192.168.9.250 Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250 CSeq: 102 ACK User-Agent: Asterisk PBX 1.6.2.13 Content-Length: 0 -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All circuits are busy now on outgoing trunk call
I am setting up asterisk-fxo gw. 404 Not Found (User not found) means the user is not found, but I don't need to have extensions or authentication on the fxo gw On 3/21/12, Michael L. Young myo...@acsacc.com wrote: [0K --- SIP read from UDP:192.168.9.251:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687 To: sip:0722490994@192.168.9.251;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06 From: pbxserver sip:Unknown@192.168.9.250;tag=as66c75bd7 CSeq: 102 INVITE Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250 Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO) Content-Length: 0 I think the 404 Not Found being returned from the server is a clue as to what the problem is. Michael L. Young (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All circuits are busy now on outgoing trunk call
Hi, I have configured a route on the fxo to send all incoming sip traffic to the fxo ports. I will try set the specific digits and see. On 3/21/12, SamyGo govoi...@gmail.com wrote: 404 NOT FOUND means that they were unable to find any destination/route/rule/prefix match corresponding to your dialled number. See your FXO gateway configuration Web-UI for outbound patterns OR verify that the FXO has its outbound line configured and working properly. On Wed, Mar 21, 2012 at 5:20 PM, James Mutuku listmut...@gmail.com wrote: I am setting up asterisk-fxo gw. 404 Not Found (User not found) means the user is not found, but I don't need to have extensions or authentication on the fxo gw On 3/21/12, Michael L. Young myo...@acsacc.com wrote: [0K --- SIP read from UDP:192.168.9.251:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.9.250:5060;rport=5060;branch=z9hG4bK111ef687 To: sip:0722490994@192.168.9.251 ;tag=1332328154302b4aa3-f15d-4eb7-beee-97782e7cbd06 From: pbxserver sip:Unknown@192.168.9.250;tag=as66c75bd7 CSeq: 102 INVITE Call-ID: 4ce934e47314480b45073a7d768cb0c8@192.168.9.250 Server: Epygi Quadro SIP User Agent/v5.1.16 (QUADRO-FXO) Content-Length: 0 I think the 404 Not Found being returned from the server is a clue as to what the problem is. Michael L. Young (elguero) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Installing asterisk on a server vs appliance(e.g digium mypbx)
Thanks for Carlos for the response, I have worked with bare asterisk + freepbx before. the mypbx was just an example but my reference to appliances as a whole. The appliances seem to have lower entry costs. On 12/1/11, Carlos Alvarez car...@televolve.com wrote: At the most basic level, typically an appliance will have a GUI and be geared towards non-tech installation. Loading bare Asterisk on a server is very different. Do you want a GUI or bare Asterisk? BTW, the MyPBX product is not a Digium product, it's from an oriental company named Yeastar. My experience in talking to them about their phones has been so-so. Historically we've had awful experiences with other Chinese phone vendors and have stopped considering products from Chinese companies. We did not actually try Yeastar products. On Wed, Nov 30, 2011 at 3:39 PM, James Mutuku listmut...@gmail.com wrote: Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially. From experience, what would be pro and cons for either option? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing asterisk on a server vs appliance(e.g digium mypbx)
Hi, I am looking into advising a client on the pro's and cons of using Installing asterisk on a server vs appliance(e.g digium mypbx). the appliance seems cheaper initially. From experience, what would be pro and cons for either option? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke www.zetu.co.ke Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Beginners Guide to setting up a Call Centre
http://www.google.co.ke/search?q=asterisk+for+call+centersie=utf-8oe=utf-8aq=trls=org.mozilla:en-US:officialclient=firefox-a -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales -- _ -- 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] Sending call information to handset
I have asterisk and linksys spa 942 phones. Normally If there are missed calls they display on the phones screen. I want to write a script that sends all missed calls to the phones screen, and email for theat extension. I need advice on where to start especially on how to send the informatio to the handset Thanks -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- 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] setting up a IP based voip carrier account
Hellos,
My voip carrier has assigned me a IP based account...where they only give me
the IP to call through. I have setup the dial plan
exten = _7XXX.,1,Answer()
exten = _7XXX.,2,vmauthenticate(${CALLERID(number)})
exten = _7XXX.,3,Dial(SIP/${EXTEN:1...@y.y.y.y)
exten = _7XXX.,4,Hungup()
Where Y.Y.Y.Y is the assigned IP. After Dialing I asterisk logs the error
SIP/Y.Y.Y.Y-35dc is circuit-busy
Are there any settings I am leaving out?
Thanks
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM
can help you achieve better customer satisfaction and sales
___
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[asterisk-users] Freepbx database
Hellos I am using freepbx and asterisk. I am writing an AGI script to edit the values in findmefollow table. The script will enable users to delete and add follow me numbers from their phones. I want it to enable users enable/disable follow me. I can't seem to find a value in the database that deals with enabling/disabling followme. Please help -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Freepbx database
I have tried all that. I just can't trace the value. this maybe the wrong list. I just thought someone might know On Thu, Sep 17, 2009 at 5:39 PM, cov...@ccs.covici.com wrote: Well, why not disable it from the GUI and see what changes -- this is sort of the wrong list, but maybe someone knows more fully. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with dialparties.agi
Hellos, I have asterisk 1.2 and freepbx 2.3. I have edited the agi script(dialparties.agi). Everytime I restart asterisk, the file gets overwritten. How do I make sure my changes are not overwritten? What generates dialparties.agi? Thanks -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help with dialparties.agi
I got my answer dialparties.agi is not generated. It get's copied from the core module when you 'Apply Configuration Settings' which is what allows it to be updated with new core modules. It is also copied when you reload asterisk James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR Reporting
I got it running a few days ago. I am using php4. Itshould run on php5 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and link spa942 provisioning
Hellos, I need to send personal directory from asterisk to the ersonal directory of the linksys spa 942. Is this possible? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk remote calls with low bandwith and high latency
Hello, I have 2 sites. One(Site 1) has an asterisk PBx and the Other(site 2) has 2 remote soft phones. The latency btw both sites is btw 500ms-700ms. I know this is a shot in the dark...but are there ways of improving the voice quality for the remote calls(btw site 1 and site 2), Other than increasing bandwidth? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Freepbx database followme disable/enable value
Hello, I am writing an AGI script to achieve the following - Users can Disable/Enable followme from their extension. They can also change the followme details from their extensions. I have looked at the follow me table for freepbx. I can't see the field for the values enabling/disable followme. Is this value stored in the database? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] passing commands asterisk cli and getting output using PHP AGI
Hellos, I know this might be an easy one but either way I am stuck...I need to execute asterisk cli commands using php agi and get the output via the same script. How to I execute let's say show hints and get the output back to the script? I have tried $agi-exec(show hints); but I am getting the output below on the cli debug AGI Rx EXEC show hints AGI Tx 200 result=-2 AGI Rx VERBOSE EXEC show hints returned -2 1 AGI Tx 200 result=1 From My understanding -2 means failure to find application What am I doing wrong? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] passing commands asterisk cli and getting output using PHP AGI
I have included that but my scripts goes silent at
AGI Rx EXEC Flite Hello 1215, you have dialed 1220.
AGI Tx 200 result=0
Below is my script
#!/usr/bin/php -q
?php
set_time_limit(30);
require('phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
$asm = $agi-new_AsteriskManager();
$agi-answer();
$callext = $agi-get_variable(DNID);
$callext=$callext['data'];
$callid = $agi-get_variable(CALLERID(num));
$callid=$callid['data'];
$agi-exec(Flite,\Hello $callid, you have dialed $callext.\);
$asm-command(show hints);
$agi-exec(flite,\Goodbye\);
$agi-hangup();
?
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[asterisk-users] followme Script
Hello, I am looking for a follow me script, where users can toggle follow me from their extensions and add follow me numbers from their extensions. Thanks -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] problem with agi script not getting variable
I am learning agi scripting using php. I m using phpagi 2.x on asterisk 1.2.
I hve written a simple script that reads out the callerid using flite. My
problem is that I seems the script is not getting the callerID.
Bellow is the script
_
#!/usr/bin/php -q
?php
/**
* @package phpAGI_examples
* @version 2.0
*/
set_time_limit(30);
require('phpagi.php');
error_reporting(E_ALL);
$agi = new AGI();
$agi-answer();
$cid = $agi-parse_callerid();
$agi-exec(Flite,\Hello, {$cid['name']}.\);
$agi-exec(flite,\Goodbye\);
$agi-hangup();
?
___
and below is my agi debug output
-- Launched AGI Script /var/lib/asterisk/agi-bin/hints.php
AGI Tx agi_request: hints.php
AGI Tx agi_channel: SIP/1215-e5b8
AGI Tx agi_language: en
AGI Tx agi_type: SIP
AGI Tx agi_uniqueid: 1251926037.3
AGI Tx agi_callerid: 1215
AGI Tx agi_calleridname: device
AGI Tx agi_callingpres: 0
AGI Tx agi_callingani2: 0
AGI Tx agi_callingtns: 0
AGI Tx agi_dnid: 1220
AGI Tx agi_rdnis: unknown
AGI Tx agi_context: privileged
AGI Tx agi_extension: 1220
AGI Tx agi_priority: 2
AGI Tx agi_enhanced: 0.0
AGI Tx agi_accountcode:
AGI Tx AGI Rx EXEC Flite Hello, .
-- AGI Script Executing Application: (Flite) Options: (Hello, .)
-- Playing '/tmp/flite_buf_VTgzTg' (language 'en')
As you can see, the callerID is not palyed out. What could I be doing wrong?
--
Best Regards,
James Mutuku Ndeti
Agile Systems Limited
+254722490994
www.agile.co.ke
mutuku.wordpress.com
Has your organization implemented a customer relationship management
(CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM
can help you achieve better customer satisfaction and sales
___
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Re: [asterisk-users] Help with call scenario
I am new to AGI. I have written my first php agi script that gets the extension dialed and says it back the caller using flite. I am stuck on how to pass the comand asterisk –rx “core show hints to asterisk and get the data back. This isn’t the recommended way, but it does work: Let’s say extension A is 100 and B is 101. Set up hints for 100 and 101. Then do a quick and dirty agi to parse “asterisk –rx “core show hints” “ for InUse. If any of the 4 lines of 100 are in use, hints will report it as inuse, so you can use that to report back to b (101) that 100 is busy. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] jitterbuffer for chan_sip on asterisk 1.2
Hello, From http://www.voip-info.org/wiki/view/Asterisk+new+jitterbuffer, it says that there For Asterisk 1.2 there was no jitterbuffer in the RTP-based channels (i.e. chan_sip). I am using 1.2 and Ind there is no reason to upgrade. Are there any developments on this? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer for chan_sip on asterisk 1.2
The project I am working on is really big. Unless I upgrade during christmas(by then the project will be several months overdue). Just googled further and saw some patches. I will try them and see. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer for chan_sip on asterisk 1.2
I did am not the one who started the project. the client has been running 1.2 for years and they needed additional features set up ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] jitterbuffer for chan_sip on asterisk 1.2
It's long gone. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help with call scenario
I am running asterisk and I want to achieve the following scenario My goal in the end is to achieve the scenario (example using extension A and Extension B) 1. Extension A has a line apperance of 4(4 calls can ring on it). 2. Extension B calls extension A(which is busy on one of the lines). 3. Extension A sees the second light blinking and hears the beeps (currently working). 4. Extension B is notified that extension A is on another call. Where do I start...I have looked in the following call variables--$dialstatus and $devstatus but I can't get them working -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call variables(dialstatus?)
I am running asterisk 1.2.9.1. I looked into devstate(). I still can't
figure how to use it.
Below is my script. I am dialing 1215,then I get a prompt on the state of
extension 1215. I have set extension 1215 to busy(by making several calls to
other extensions). I still get the prompt 1215 has a state 0. Below is my
script. What am I doing wrong?.
exten = 1215,1,Answer
exten = 1215,2,Flite(${EXTEN} has state ${devstate(${EXTEN})})
exten = 1215,3,Hangup
Thanks in advance.
My goal in the end is to achieve the scenario (example using extension A and
Extension B)
1. Extension A has a line apperance of 4(4 calls can ring on it).
2. Extension B calls extension A(which is busy on one of the lines).
3. Extension A sees the second light blinking and hears the beeps (currently
working).
4. Extension B is notified that extension A is on another call.
___
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Re: [asterisk-users] Call variables(dialstatus?)
...the above post lacked some details
I am running asterisk 1.2.9.1. I looked into devstate(). I still can't
figure how to use it. My dial plan gives 1215 has state 0 on all
scenarios.
Below is my script. I am dialing 1215,then I get a prompt on the state of
extension 1215. I have set extension 1215 to busy(by making several calls to
other extensions). I still get the prompt 1215 has a state 0. Below is my
script. What am I doing wrong?.
exten = 1215,1,Answer
exten = 1215,2,Flite(${EXTEN} has state ${devstate(${EXTEN})})
exten = 1215,3,Hangup
Thanks in advance.
My goal in the end is to achieve the scenario (example using extension A and
Extension B)
1. Extension A has a line apperance of 4(4 calls can ring on it).
2. Extension B calls extension A(which is busy on one of the lines).
3. Extension A sees the second light blinking and hears the beeps (currently
working).
4. Extension B is notified that extension A is on another call.
___
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Re: [asterisk-users] Follow me IVR sounds
thanks danny for the reply. I am looking into using flite to read out the prompts. if i ma ask...are there other voices other than the mechanical robotic male voice available for flite.? I have searched over the internet and I can't seem to find any ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Follow me IVR sounds
Hellos, I am looking for the sounds used in this ivr example http://www.voip-info.org/wiki/view/Asterisk+Tips+follow+me. The one with 6900. Any assistance is welcome. -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk followme feature code
Hellos, I have using asterisk 1.2 and freepbx 2.3. I need users to disable and enable followme from there phones. I can't see any support for it. Is this possible/available.? I have googled and I can't get information on it -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Individual PIN Code per Extension
Hellos, I have astersist 1.2 working with freepbx. I want to tie pin codes to extensions(users). How do I do this? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Call variables(dialstatus?)
Hellos, I am running asterisk 1.2 with linksys spa 942. Normally, when a call calls an extension and the extension is on another call, the call is put in line but not notified that the extension is busy. I want to notify the second, third and fourth callet that they are on extension is on another call. Which variable do I use on my dial plan -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call variables(dialstatus?)
Thanks for the reply Patrick. I do not want to limit the second, third...etc call, but I want the caller to be notified that the extension they are calling is on another call and they are waiting. Let me look into DEVSTATE On Tue, Aug 18, 2009 at 12:12 PM, Patrick Plattes patr...@erdbeere.netwrote: hi, you can use call-limit=1 in sip.conf or DEVSTATE() http://www.voip-info.org/wiki/view/Asterisk+func+device_State http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf http://www.voip-info.org/wiki/view/Asterisk+sip+incominglimit bye On Tue, Aug 18, 2009 at 9:03 AM, James Mutukulistmut...@gmail.com wrote: Hellos, I am running asterisk 1.2 with linksys spa 942. Normally, when a call calls an extension and the extension is on another call, the call is put in line but not notified that the extension is busy. I want to notify the second, third and fourth callet that they are on extension is on another call. Which variable do I use on my dial plan -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Niemann + Frey GmbH Bischofstraße 80 47809 Krefeld Tel. +49 2151 5554-263 Geschäftsführer: Gerd Frey Sitz und Registergericht: Krefeld HRB 10851 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk conference error/bug?
Hellos, I am having issues with my meetme conferencing. When I dial the conferencing number, It hangs after a few seconds.I have read somewhere that I need to enable ztdummy, which I have done but still no changes. Here is my log ~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43 =~=~=~=~=~=~=~=~=~=~=~= -- Executing [1;36;40mMacro [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40muser-callerid| [0;37;40m) in new stack -- Executing [1;36;40mNoOp [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40muser-callerid: device 1215 [0;37;40m) in new stack -- Executing [1;36;40mSet [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mAMPUSER=1215 [0;37;40m) in new stack -- Executing [1;36;40mGotoIf [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m0?report [0;37;40m) in new stack -- Executing [1;36;40mGotoIf [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m0?start [0;37;40m) in new stack -- Executing [1;36;40mSet [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mREALCALLERIDNUM=1215 [0;37;40m) in new stack -- Executing [1;36;40mNoOp [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mREALCALLERIDNUM is 1215 [0;37;40m) in new stack -- Executing [1;36;40mSet [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mAMPUSER=1215 [0;37;40m) in new stack -- Executing [1;36;40mSet [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mAMPUSERCIDNAME=Test Remote [0;37;40m) in new stack -- Executing [1;36;40mGotoIf [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m0?report [0;37;40m) in new stack -- Executing [1;36;40mSet [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mAMPUSERCID=1215 [0;37;40m) in new stack -- Executing [1;36;40mSet [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mCALLERID(all)=Test Remote 1215 [0;37;40m) in new stack -- Executing [1;36;40mSet [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mREALCALLERIDNUM=1215 [0;37;40m) in new stack -- Executing [1;36;40mNoOp [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mTTL: ARG1: [0;37;40m) in new stack -- Executing [1;36;40mGotoIf [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m0?continue [0;37;40m) in new stack -- Executing [1;36;40mSet [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m__TTL=64 [0;37;40m) in new stack -- Executing [1;36;40mGotoIf [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m1?continue [0;37;40m) in new stack -- Goto (macro-user-callerid,s,23) -- Executing [1;36;40mNoOp [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mUsing CallerID Test Remote 1215 [0;37;40m) in new stack -- Executing [1;36;40mSet [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mMEETME_ROOMNUM=5502 [0;37;40m) in new stack -- Executing [1;36;40mGotoIf [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m0?READPIN [0;37;40m) in new stack -- Executing [1;36;40mAnswer [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m [0;37;40m) in new stack -- Executing [1;36;40mWait [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m1 [0;37;40m) in new stack -- Executing [1;36;40mSet [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mPINCOUNT=0 [0;37;40m) in new stack -- Executing [1;36;40mRead [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mPIN|enter-conf-pin-number [0;37;40m) in new stack -- User disconnected == Spawn extension (privileged, 5502, 7) exited non-zero on 'SIP/1215-fc5b' -- Executing [1;36;40mMacro [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mhangupcall [0;37;40m) in new stack -- Executing [1;36;40mResetCDR [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40mw [0;37;40m) in new stack -- Executing [1;36;40mNoCDR [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m [0;37;40m) in new stack -- Executing [1;36;40mGotoIf [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m1?skiprg [0;37;40m) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [1;36;40mGotoIf [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m1?skipblkvm [0;37;40m) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [1;36;40mGotoIf [0;37;40m( [1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m1?theend [0;37;40m) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [1;36;40mHangup [0;37;40m( [1;35;40mSIP/1215-fc5b [0;37;40m, [1;35;40m [0;37;40m) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1215-fc5b' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1215-fc5b' -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales
[asterisk-users] meetme conference hangs in silence after dialing
Hellos, I am having issues with my meetme conferencing. When I dial the conferencing number, It hangs after a few seconds.I have read somewhere that I need to enable ztdummy, which I have done but still no changes. Here is my log ~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43 =~=~=~=~=~=~=~=~=~=~=~= -- Executing [1;36;40mMacro[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40muser-callerid|[0;37;40m) in new stack -- Executing [1;36;40mNoOp[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40muser-callerid: device 1215[0;37;40m) in new stack -- Executing [1;36;40mSet[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mAMPUSER=1215[0;37;40m) in new stack -- Executing [1;36;40mGotoIf[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m0?report[0;37;40m) in new stack -- Executing [1;36;40mGotoIf[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m0?start[0;37;40m) in new stack -- Executing [1;36;40mSet[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mREALCALLERIDNUM=1215[0;37;40m) in new stack -- Executing [1;36;40mNoOp[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mREALCALLERIDNUM is 1215[0;37;40m) in new stack -- Executing [1;36;40mSet[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mAMPUSER=1215[0;37;40m) in new stack -- Executing [1;36;40mSet[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mAMPUSERCIDNAME=Test Remote[0;37;40m) in new stack -- Executing [1;36;40mGotoIf[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m0?report[0;37;40m) in new stack -- Executing [1;36;40mSet[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mAMPUSERCID=1215[0;37;40m) in new stack -- Executing [1;36;40mSet[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mCALLERID(all)=Test Remote 1215[0;37;40m) in new stack -- Executing [1;36;40mSet[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mREALCALLERIDNUM=1215[0;37;40m) in new stack -- Executing [1;36;40mNoOp[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mTTL: ARG1: [0;37;40m) in new stack -- Executing [1;36;40mGotoIf[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m0?continue[0;37;40m) in new stack -- Executing [1;36;40mSet[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m__TTL=64[0;37;40m) in new stack -- Executing [1;36;40mGotoIf[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m1?continue[0;37;40m) in new stack -- Goto (macro-user-callerid,s,23) -- Executing [1;36;40mNoOp[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mUsing CallerID Test Remote 1215[0;37;40m) in new stack -- Executing [1;36;40mSet[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mMEETME_ROOMNUM=5502[0;37;40m) in new stack -- Executing [1;36;40mGotoIf[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m0?READPIN[0;37;40m) in new stack -- Executing [1;36;40mAnswer[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m[0;37;40m) in new stack -- Executing [1;36;40mWait[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m1[0;37;40m) in new stack -- Executing [1;36;40mSet[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mPINCOUNT=0[0;37;40m) in new stack -- Executing [1;36;40mRead[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mPIN|enter-conf-pin-number[0;37;40m) in new stack -- User disconnected == Spawn extension (privileged, 5502, 7) exited non-zero on 'SIP/1215-fc5b' -- Executing [1;36;40mMacro[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mhangupcall[0;37;40m) in new stack -- Executing [1;36;40mResetCDR[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40mw[0;37;40m) in new stack -- Executing [1;36;40mNoCDR[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m[0;37;40m) in new stack -- Executing [1;36;40mGotoIf[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m1?skiprg[0;37;40m) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [1;36;40mGotoIf[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m1?skipblkvm[0;37;40m) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [1;36;40mGotoIf[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m1?theend[0;37;40m) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [1;36;40mHangup[0;37;40m([1;35;40mSIP/1215-fc5b[0;37;40m, [1;35;40m[0;37;40m) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1215-fc5b' in macro 'hangupcall' == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/1215-fc5b' -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales
[asterisk-users] Second voice caller notification
Hello, I have asterisk and linksys ip phones setup. If an ext. is busy on phone, the ext. user is notified by a beep. I want to configure asterisk to notify(voice) the caller that the extension is busy on a call. Is this possible? -- Best Regards, James Mutuku Ndeti Agile Systems Limited +254722490994 www.agile.co.ke mutuku.wordpress.com Has your organization implemented a customer relationship management (CRM)system? visit http://www.agile.co.ke/crm.php and find out how our CRM can help you achieve better customer satisfaction and sales ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk+a2billing for over 10,000 ext
Hellos, I want to setup Asterisk+a2billing for over 10,000 extensions for voip resale. Has anyone done this before. What are the hardware requirements and challenges? James ___ -- 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 routine maintenance activities
Hello(s), I know this might be test book question or one best suited for google but I will take the risk of asking. Here I go. What common routine maintenance tasks do you run on your asterisk box? Thanks James. ___ -- 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] Configuring asterisk to revert call back to forwarder if exten is busy
I manged to get something working but It's only working when a
grandstream ip phones is the one tranfering calls. With linksys IP
phones I get a busy yone
I edited the extensions.conf
last lines of [macro-exten-vm]:
; Extensions with no Voicemail box reporting BUSY come here
exten = s-BUSY,1,NoOp(Extension is reporting BUSY and not passing to Voicemail)
; This should recover failed transfers.
exten = s-BUSY,n,GotoIf($[${LEN(${BLINDTRANSFER})}
0]?custom-MANAGE_LOST_TRANSFERS,s,1)
exten = s-BUSY,n,Playtones(busy)
exten = s-BUSY,n,Busy(20)
; Anything but BUSY comes here
; This should recover failed transfers.
exten = _s-.,1,GotoIf($[${LEN(${BLINDTRANSFER})}
0]?custom-MANAGE_LOST_TRANSFERS,s,1)
exten = _s-.,n,Playtones(congestion)
exten = _s-.,n,Congestion(10)
and added a new context on extensions_custom.conf
[custom-MANAGE_LOST_TRANSFERS]
exten = s,1,Answer()
exten = s,n,Playback(please-wait-bouncing-back)
; Supposing there are 4-digit extensions here - no error checking
exten = s,n,Set(RETURN_EXT=${BLINDTRANSFER:4:3})
exten = s,n,Goto(from-internal,${RETURN_EXT},1)
exten = s,n,HangUp()
What could be the problem with linksys phones?
_
Paul Hales wrote:
Can I assume that you want this only for blind transfers?
I have done this previously, but I lost my copy of the work (and it was
a proof of concept only)
It involved the ${BLINDTRANSFER} variable, which catches the number that
made the blind transfer and making macro-stdexten (or your equivalent)
dial that variable in the case of the dial status being treated as BUSY.
To get a 'busy' will involve single line phones, or disabling call
waiting on the phone receiving the call.
regards,
PaulH
James Mutuku wrote:
Hellos,
I want to configure asterisk so that if exten A transfers a call to
exten B, and B is either busy or the call is not answered, the call
returns back to A. Is this possible?
Please help
James
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begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
email;internet:jnmut...@gmail.com,jmut...@agile.co.ke
title:Lead Consultant
tel;work:+254-722-490994
tel;home:+254-722-490994
tel;cell:+254-722-490994
url:www.agile.co.ke
version:2.1
end:vcard
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[asterisk-users] Configuring asterisk to revert call back to forwarder if exten is busy
Hellos, I want to configure asterisk so that if exten A transfers a call to exten B, and B is either busy or the call is not answered, the call returns back to A. Is this possible? Please help James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:jnmut...@gmail.com,jmut...@agile.co.ke title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- 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] Configuring asterisk to revert call back to forwarder if exten is busy
Hellos, I want to configure asterisk so that if exten A transfers a call to exten B, and B is either busy or the call is not answered, the call returns back to A. Is this possible? Please help James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:jnmut...@gmail.com,jmut...@agile.co.ke title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- 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] Intergrating vicidial with trixbox
Hello, I am searched the net for tutorials on how I can Integrate vicidial with trixbox. I can't find any. Anyone who knows where I can get one? James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED] title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- 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 appliance A50 vs asterisk open source + fxo cards
Hi list, I need advice on which solution to implement, asterisk appliance A50 or just install linux on a pc and get tdm cards. Any comments? James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED] title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk appliance A50 vs asterisk open source + fxo cards
Hi, Thanks for the response.Would you rather install asterisk or use the appliance? James Grygoriy Dobrovolskyy wrote: As Drew Gibson wrote at anotehr topic: [quote] The internal card should give you higher reliability as there are fewer parts and cables although the external gateways could allow you to have redundant servers. External gateways would also be easier to scale when you need more lines. [/quote] 2008/8/1 James Mutuku [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Hi list, I need advice on which solution to implement, asterisk appliance A50 or just install linux on a pc and get tdm cards. Any comments? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED] title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help With dial plan
Thanks for the wild guess. But The user(who is myself) is dialing 3000.
It only failes to work when I use patterns. So I thought I am making a
mistake on the syntax, I have checked all the books I have and the
internet and I can't see anything wrong. :-\
Rizwan Hisham wrote:
maybe the user is dialing something other than 3000 and that extension
is not registered on your asterisk. just a wild guess.
On Tue, Jul 22, 2008 at 10:41 AM, James Mutuku [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Hi list,
Have installed trixbox and I am working with a fxo gateway to get
fxo calls to trixbox. I am using sip to send the calls from the
gateway to trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial
plan below.
exten = 3000,1,dial(sip/3000)
exten= 3000,2,answer()
exten = 3000,3,congestion()
exten= 3000,4,hangup()
this works fine. But I when I put it in the form
exten = _3XXX,1,dial(sip/${EXTEN})
exten= _3XXX,2,answer()
exten =_3XXX,3,congestion()
exten= _3XXX,4,hangup()
the call goes into congestion and I get a busy tone. What could I
be doing wrong?
James
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Best Regards
Rizwan Hisham
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begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED]
title:Lead Consultant
tel;work:+254-722-490994
tel;home:+254-722-490994
tel;cell:+254-722-490994
url:www.agile.co.ke
version:2.1
end:vcard
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[asterisk-users] Help with dial plan
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo
calls to trixbox. I am using sip to send the calls from the gateway to
trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten = 3000,1,dial(sip/3000)
exten= 3000,2,answer()
exten = 3000,3,congestion()
exten= 3000,4,hangup()
this works fine. But I when I put it in the form
exten = _3XXX,1,dial(sip/${EXTEN})
exten= _3XXX,2,answer()
exten =_3XXX,3,congestion()
exten= _3XXX,4,hangup()
the call goes into congestion and I get a busy tone. What could I be
doing wrong?
James
begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED]
title:Lead Consultant
tel;work:+254-722-490994
tel;home:+254-722-490994
tel;cell:+254-722-490994
url:www.agile.co.ke
version:2.1
end:vcard
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[asterisk-users] Help With dial plan
Hi list,
Have installed trixbox and I am working with a fxo gateway to get fxo
calls to trixbox. I am using sip to send the calls from the gateway to
trixbox. I have an extension 3000 on trixbox
on [from-sip-external] on extensions.conf ,I have put the dial plan below.
exten = 3000,1,dial(sip/3000)
exten= 3000,2,answer()
exten = 3000,3,congestion()
exten= 3000,4,hangup()
this works fine. But I when I put it in the form
exten = _3XXX,1,dial(sip/${EXTEN})
exten= _3XXX,2,answer()
exten =_3XXX,3,congestion()
exten= _3XXX,4,hangup()
the call goes into congestion and I get a busy tone. What could I be
doing wrong?
James
begin:vcard
fn:James Mutuku
n:Mutuku;James
org:Agile Systems Limited;Technical Department
adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya
email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED]
title:Lead Consultant
tel;work:+254-722-490994
tel;home:+254-722-490994
tel;cell:+254-722-490994
url:www.agile.co.ke
version:2.1
end:vcard
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[asterisk-users] question about fxo cards
Hi, has anyone worked with nxtvox(www.nxtvox.com) fxo cards? What is their quality? James ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] has anyone worked with nxtvox fxo cards
Hi, has anyone worked with nxtvox fxo cards? What is their quality? James begin:vcard fn:James Mutuku n:Mutuku;James org:Agile Systems Limited;Technical Department adr:;;P.O Box 55686-00200;Nairobi;;00200;Kenya email;internet:[EMAIL PROTECTED],[EMAIL PROTECTED] title:Lead Consultant tel;work:+254-722-490994 tel;home:+254-722-490994 tel;cell:+254-722-490994 url:www.agile.co.ke version:2.1 end:vcard ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please Advice on Best High traffic fxo gateway/cards
Hi, Probably I did not explain my situation before asking the question. I have been working with epygi fxo gateways(www.epygi.com) for some time. They are 6 port fxo gateways, but they fail(hang) when it comes to high traffic on the 6 POTS, resulting to not-so-happy clients. In my country, the main telco operator does not follow any standards. Different lines(POTs) from the same telco might have different disconnect settings*. With the epygi fxo gateways, you can only set a system wide disconnect settings, not for individual lines(POTs). What happens is when you configure the disconnect settings from one of the lines, you get disconnect problems(call never disconnects with time, the whole system hangs). *disconnect settings - (the way I understand the term)frequency values that enable an fxo gw to detect the call disconnection from the line(POT). Steve Totaro wrote: Some customers are locked into two year contracts. That was the answer I got when adding four POTS lines to a system with four BRIs... Thanks, Steve Totaro On Tue, Jun 17, 2008 at 1:39 PM, James Mutuku [EMAIL PROTECTED] wrote: Michael, I agree. Here we use e1s(which have even more channels). Some clients just don't want to change some if their old infrastructure. Thanks Michael Graves wrote: I just hafta ask, why does one face down a requirement for 48 FXOs? Would it not be more practical to have 2 T-1s dropped into the location? Michael On Mon, 16 Jun 2008 17:35:35 -0400, Steve Totaro wrote: Adit 600 48 FXO. On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku [EMAIL PROTECTED] wrote: Steve, Thanks for the responses. I am talking of 45 POTS Thanks Steve Totaro wrote: Sorry, Quantify High Traffic How many POTS lines are we talking about? Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: I use Adtran or Adit, I think Rhino has a pretty low priced one but I have never used so cannot comment. I can tell you that the Adtran or Adit is rock solid. Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku [EMAIL PROTECTED] wrote: Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- 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 -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- 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 ___ -- 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] Please Advice on Best High traffic fxo gateway/cards
Michael, I agree. Here we use e1s(which have even more channels). Some clients just don't want to change some if their old infrastructure. Thanks Michael Graves wrote: I just hafta ask, why does one face down a requirement for 48 FXOs? Would it not be more practical to have 2 T-1s dropped into the location? Michael On Mon, 16 Jun 2008 17:35:35 -0400, Steve Totaro wrote: Adit 600 48 FXO. On Mon, Jun 16, 2008 at 12:11 PM, James Mutuku [EMAIL PROTECTED] wrote: Steve, Thanks for the responses. I am talking of 45 POTS Thanks Steve Totaro wrote: Sorry, Quantify High Traffic How many POTS lines are we talking about? Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: I use Adtran or Adit, I think Rhino has a pretty low priced one but I have never used so cannot comment. I can tell you that the Adtran or Adit is rock solid. Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku [EMAIL PROTECTED] wrote: Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- 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 -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves [EMAIL PROTECTED] ___ -- 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] Please Advice on Best High traffic fxo gateway/cards
Brian, I agree with you. Google has all the answers, but not the experience. The reason I use lists is to get opinions 'experienced' users. From experience, product manuals say one thing and when the rubber meets the road, its a different story. Thanks though for your kind comments Brian J. Murrell wrote: On Sun, 2008-06-15 at 11:03 -0400, Steve Totaro wrote: On Sun, Jun 15, 2008 at 10:53 AM, Brian J. Murrell [EMAIL PROTECTED] wrote: On Sun, 2008-06-15 at 17:43 +0300, James Mutuku wrote: Please advice on channel bank Dude. There's the cool new website you should check out. It's www.google.com. Seriously. This list is not full of people waiting to do the simplest research at your request. Spend a few minutes and do some self-help before coming here asking the simplest, most general questions. You are more likely to get answers to interesting questions rather than mundane-google-would-have-told-you-all-you-need-to-know-in-5-minutes questions. b/ While true to some degree, I assumed he was looking for someone to recommend a certain product based on good experiences in the Asterisk World. See, I saw the quotes around channel bank more as the follow question what is a channel bank. Maybe it's a language thing and perhaps the OP can take as constructive criticism to be more to one's actual point when asking a question. If he really did understand what a channel bank is and was looking for recommendations, something more direct like Any recommendations on which channel bank(s) I should consider using? would have been much more fruitful I suspect. Google may be good for getting information but will turn up a good many ads too. Most of these ads/sites all claim to be the best. We are the leaders of (such and such) Sure, but all of them will give him a good idea of what one actually is, which is really what I suspect the question was. An obvious pitfall I met was Citel gateways. Maybe they have improved for the Definity line, but going that route a year and a half ago made me look very bad. I wish I had asked on the list and got someone with some experience to say, think twice. Agreed. I wholeheartedly agree with soliciting for and giving product recommendations and experiences, but questions like what is ... most likely can almost always be answered from google with a little effort on one's own behalf. b. ___ -- 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] Please Advice on Best High traffic fxo gateway/cards
Steve, Thanks for the responses. I am talking of 45 POTS Thanks Steve Totaro wrote: Sorry, Quantify High Traffic How many POTS lines are we talking about? Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:47 AM, Steve Totaro [EMAIL PROTECTED] wrote: I use Adtran or Adit, I think Rhino has a pretty low priced one but I have never used so cannot comment. I can tell you that the Adtran or Adit is rock solid. Thanks, Steve Totaro On Sun, Jun 15, 2008 at 10:43 AM, James Mutuku [EMAIL PROTECTED] wrote: Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- 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] Please Advice on Best High traffic fxo gateway/cards
Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- 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] Please Advice on Best High traffic fxo gateway/cards
Please advice on channel bank Steve Totaro wrote: I would suggest a channel bank populated with FXO cards muxing to a T1. Thanks, Steve T On Sun, Jun 15, 2008 at 10:06 AM, James Mutuku [EMAIL PROTECTED] wrote: Hi, I need to get an fxo gateway/card for a high traffic asterisk installation. Please advice on which gateway/ fxo cards Thanks ___ -- 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
[asterisk-users] Asterisk Unified communication features
Hi, I need to know if the following features are available on asterisk and their quality -SMS -Call control, budgeting and monitoring -Video conferencing -support for 500 extensions -fax -audio and video conferencing and 1. Call accounting showing calls made 2. Call budgeting which bills the calls 3. Web access for all users 4. Centralized management and administration 5. Call barring when budget is exhausted 6. Budget utilization alerts to e-mail 7. Reports a) Per extension b) Per trunk c) Per unit (business area) d) Percentage utilization of the total budget Thanks ___ -- 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
