[asterisk-users] Asterisk with Cisco 887M
Hi! I am installing asterisk with my ISP but he give me a Cisco 887M router to use for SIP conection. My problem is that I dont know how to link Asterisk with this device because I dont have user/pass to use. Anybody has a cluee to use CISCO 887M with Asterisk ? Thks! -- _ -- 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] Using a Virtual IP Line
Hello! I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured. I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work. I have traces of xlite for the invite and register this done to see if someone can help me to use this line with my asterisk. These are the traces of my Xllite REGISTER sip:Xlite release 1100l stamp 49022 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-e322ee549824f666-1---d8754z-;rport Max-Forwards: 70 Contact: sip:888777@10.0.0.221:22818;rinstance=570ac597afa82c9a To: 888777sip:888777@Xlite release 1100l stamp 49022 From: 888777sip:888777@Xlite release 1100l stamp 49022;tag=fb1acd4f Call-ID: NjUyN2Q5ODEzZmZiNjRhY2FmNzRlZjljNDRjMjg5Yjk. CSeq: 2 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: Xlite release 1100l stamp 49022 Authorization: Digest username=888777,realm=192.168.50.20,nonce=d999e9471b1d,uri=sip:Xlite release 1100l stamp 49022,response=ba26805d2f0b97a70565c37e81444e44,cnonce=820e1f348b49cd73d92e1bc793be5ad7,nc=0001,qop=auth,algorithm=MD5 Content-Length: 0 REGISTER sip:Xlite release 1100l stamp 49022 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-c605aa61ac248834-1---d8754z-;rport Max-Forwards: 70 Contact: sip:888777@33.33.33.33:22818;rinstance=da357fa09f45cdda To: 888777sip:888777@Xlite release 1100l stamp 49022 From: 888777sip:888777@Xlite release 1100l stamp 49022;tag=fb1acd4f Call-ID: NjUyN2Q5ODEzZmZiNjRhY2FmNzRlZjljNDRjMjg5Yjk. CSeq: 4 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: Xlite release 1100l stamp 49022 Authorization: Digest username=888777,realm=192.168.50.20,nonce=d999e9471b1d,uri=sip:Xlite release 1100l stamp 49022,response=b0858d0b5914f054faf8f0b0eed22400,cnonce=659200e211cc5023724817d04c14cb3a,nc=0003,qop=auth,algorithm=MD5 Content-Length: 0 SUBSCRIBE sip:888777@Xlite release 1100l stamp 49022 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-23698d60215c9f07-1---d8754z-;rport Max-Forwards: 70 Contact: sip:888777@33.33.33.33:22818 To: 888777sip:888777@Xlite release 1100l stamp 49022 From: 888777sip:888777@Xlite release 1100l stamp 49022;tag=f5062e32 Call-ID: Y2Y5MjFjNWFlM2QzNWFiZjgwYWQxYTc5ZmRmZTVhOWE. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO User-Agent: Xlite release 1100l stamp 49022 Event: message-summary Content-Length: 0 INVITE sip:18094713172@Xlite release 1100l stamp 49022 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-8c57153848230175-1---d8754z-;rport Max-Forwards: 70 Contact: sip:888777@33.33.33.33:22818 To: 18094713172sip:18094713172@Xlite release 1100l stamp 49022 From: 888777sip:888777@Xlite release 1100l stamp 49022;tag=0337ad04 Call-ID: NWNlMzIyNDZiNjUxNjA4NjQ4ZjM3ZDhjM2E3NmViNjQ. CSeq: 1 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: Xlite release 1100l stamp 49022 Content-Length: 386 v=0 o=- 3 2 IN IP4 10.0.0.221 s=CounterPath eyeBeam 1.5 c=IN IP4 10.0.0.221 t=0 0 m=audio 48758 RTP/AVP 18 100 106 6 0 105 8 3 5 101 a=fmtp:18 annexb=yes a=fmtp:101 0-15 a=rtpmap:18 G729/8000 a=rtpmap:100 SPEEX/16000 a=rtpmap:106 SPEEX-FEC/16000 a=rtpmap:105 SPEEX-FEC/8000 a=rtpmap:101 telephone-event/8000 a=sendrecv a=x-rtp-session-id:BB752EE94E6C4F5E870B02DB4DA411D5 Any help or any sugestion will be so appreciated. TIA *---* *-Edwin Quijada *- -- _ -- 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] Using files .call or AMI
Thks, now I understand for your cooperation.TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Sat, 12 Feb 2011 23:20:11 + From: ro...@firedrake.org To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using files .call or AMI On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote: This works for me.! but the agent has to dial the number ? How could be the context for do this ? U can give an example ? I'm using this to place calls from local IP-phones over the PSTN. So my script will generate, say: Channel: SIP/lanphone Context: from-lan Extension: 08001234567 taking the 0800... from the list of customer details. SIP/lanphone is the ID of the originating phone. Extension is the sequence the agent would dial if he were placing the call himself. The originating phone rings; when it's picked up, the Asterisk server calls the Extension number and bridges the two calls, so the local agent hears ringing tones from the far end. All the agent has to do is pick up the phone when it rings and put it down when the call is over. -- _ -- 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://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] Using files .call or AMI
How would be the dialplan for this context from-lan ??? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Sat, 12 Feb 2011 23:20:11 + From: ro...@firedrake.org To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using files .call or AMI On Sat, Feb 12, 2011 at 10:19:16PM +, Edwin Quijada wrote: This works for me.! but the agent has to dial the number ? How could be the context for do this ? U can give an example ? I'm using this to place calls from local IP-phones over the PSTN. So my script will generate, say: Channel: SIP/lanphone Context: from-lan Extension: 08001234567 taking the 0800... from the list of customer details. SIP/lanphone is the ID of the originating phone. Extension is the sequence the agent would dial if he were placing the call himself. The originating phone rings; when it's picked up, the Asterisk server calls the Extension number and bridges the two calls, so the local agent hears ringing tones from the far end. All the agent has to do is pick up the phone when it rings and put it down when the call is over. -- _ -- 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://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using files .call or AMI
Hi! I have a script to generate calls from a database using .call files and giving a message. If works great! but now I need to do the same but instead of play a recorded message I need transfer this call to live person in a specfic extension. This is the scenarioI have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer the call to agent if this call is answered.I did the page and everything but when I do the clicktodial I dont know how transfer the call to this agent. I ask the extension and user before login so I know what agent is in each extension to transfer the call to rigth agent. Anybody can give an idea ?TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- 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] Using files .call or AMI
Date: Sat, 12 Feb 2011 21:35:29 + From: ro...@firedrake.org To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using files .call or AMI On Sat, Feb 12, 2011 at 04:23:00PM +, Edwin Quijada wrote: I have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer the call to agent if this call is answered. Usually it's the other way round: the agent's phone rings, and when he picks it up the other end gets dialled. That's trivial with call files: Channel: (local channel ID for agent) Context: (context for calling local channel) Extension: (remote party's phone number) This works for me.! but the agent has to dial the number ? How could be the context for do this ? U can give an example ? TIA -- _ -- 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] Using files .call or AMI
My problem is that I dont know how to do for transfer the call to agentExample, I have this .call Channel: Zap/g1/8652323454MaxRetries: 2 RetryTime: 60 WaitTime: 30 Context: call-file-test Extension: 10 So my context is this [call-file-test ]exten = 10,1,Dial(SIP/2031,tT)exten = 10,2,hangup In this case I call the number 8652323454 if the call is connect this call in the context call-file-test uisng extension 10 for tranfering this call to extension 2031, but this doesnt work. The call file works fine but when I try to transfer the call I get an error Any help ? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* From: l...@lopl.net Date: Sat, 12 Feb 2011 21:22:50 +0330 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Using files .call or AMI as you know you have 2 ways. using ami or .call files. if you have experience, the AMI is more powerful. you must have a context in your extensions.conf to manage agent procedures, it looks like a simple context, that you must have, for managing queues. with .call file or ami dial your customers, () and divert it to the defined context for queue. for example test.call Channel: SIP/customer number@your carrier Context: your queue context. ask if you need more infobest On Sat, Feb 12, 2011 at 7:53 PM, Edwin Quijada listas_quij...@hotmail.com wrote: Hi! I have a script to generate calls from a database using .call files and giving a message. If works great! but now I need to do the same but instead of play a recorded message I need transfer this call to live person in a specfic extension. This is the scenarioI have a webpage with information about a customer so in this page the agent click a phone number and asterisk do the call and transfer the call to agent if this call is answered. I did the page and everything but when I do the clicktodial I dont know how transfer the call to this agent. I ask the extension and user before login so I know what agent is in each extension to transfer the call to rigth agent. Anybody can give an idea ?TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- 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://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://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] Issue with Red Alarm with DAhDi
OpenVox A800P\ 8 port FXO *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Tue, 11 Jan 2011 17:09:51 -0600 From: sruff...@digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Issue with Red Alarm with DAhDi On 1/11/11 2:33 PM, Edwin Quijada wrote: Hi! I have an analog line connected to my asterisk and when I try to answer a call I get this -- Starting simple switch on 'DAHDI/7-1' -- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack -- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en') [Jan 11 16:29:46] WARNING[3411]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' [Jan 11 16:29:47] NOTICE[3348]: chan_dahdi.c:7434 handle_init_event: Alarm cleared on channel 7 -- Starting simple switch on 'DAHDI/7-1' -- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack -- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en') [Jan 11 16:29:52] WARNING[3412]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' [Jan 11 16:29:53] NOTICE[3348]: chan_dahdi.c:7434 handle_init_event: Alarm cleared on channel 7 -- Starting simple switch on 'DAHDI/7-1' -- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack -- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en') [Jan 11 16:29:58] WARNING[3413]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' I checked fisically the card and not red alarm in this. I am using Asterisk 1.4.38 and Dahdi 2.4.0 Any cluees ? TIA What card are you using for your DAHDI channels? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- 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://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with Red Alarm with DAhDi
Hi! I have an analog line connected to my asterisk and when I try to answer a call I get this -- Starting simple switch on 'DAHDI/7-1'-- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack-- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack-- DAHDI/7-1 Playing 'vm-intro' (language 'en')[Jan 11 16:29:46] WARNING[3411]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1'-- Hungup 'DAHDI/7-1'[Jan 11 16:29:47] NOTICE[3348]: chan_dahdi.c:7434 handle_init_event: Alarm cleared on channel 7-- Starting simple switch on 'DAHDI/7-1'-- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack-- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack-- DAHDI/7-1 Playing 'vm-intro' (language 'en')[Jan 11 16:29:52] WARNING[3412]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1'-- Hungup 'DAHDI/7-1'[Jan 11 16:29:53] NOTICE[3348]: chan_dahdi.c:7434 handle_init_event: Alarm cleared on channel 7-- Starting simple switch on 'DAHDI/7-1'-- Executing [...@from-pstn:1] Answer(DAHDI/7-1, ) in new stack-- Executing [...@from-pstn:2] Playback(DAHDI/7-1, vm-intro) in new stack -- DAHDI/7-1 Playing 'vm-intro' (language 'en')[Jan 11 16:29:58] WARNING[3413]: chan_dahdi.c:4283 handle_alarms: Detected alarm on channel 7: Red Alarm == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/7-1' -- Hungup 'DAHDI/7-1' I checked fisically the card and not red alarm in this. I am using Asterisk 1.4.38 and Dahdi 2.4.0 Any cluees ? TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- 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] Polarity Reverseal....with analog line
Can I reverse the polarity from Asterisk to get the call ? I have 5 days with this and I dont know what to do. I changed zaptel for dAHDI now I have Dahdi 2.4 and asterisk 1.4.30 TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Wed, 5 Jan 2011 17:04:02 -0500 From: markm-li...@intellasoft.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Polarity Reversealwith analog line Looks like your telco is sending you polarity reversal on sending you a call. Which is one of the types of setups for analog lines.l From your console output it looks like the call was handled just fine other than the 'weird event' notification, which I'm not familiar with. On 01/05/2011 11:50 AM, Edwin Quijada wrote: Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [...@from-pstn:3] Hangup(Zap/5-1, ) in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1' [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)... [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)... [Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)... -- Executing [...@from-pstn:1] Answer(Zap/5-1, ) in new stack -- Executing [...@from-pstn:2] Playback(Zap/5-1, vm-intro) in new stack -- Zap/5-1 Playing 'vm-intro' (language 'en') [Jan 5 12:45:08] WARNING[2893]: chan_dahdi.c:4550 dahdi_handle_event: Ring/Off-hook in strange state 6 on channel 5 -- Executing [...@from-pstn:3] Hangup(Zap/5-1, ) in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1' -- Hungup 'Zap/5-1' I am using 1.4.30 and zaptel 1.12. Any cluess? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- 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://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://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polarity Reverseal....with analog line
Hi ! I ma having trouble with my PTSN line. When I call to my asterisk I get this.. -- Executing [...@from-pstn:3] Hangup(Zap/5-1, ) in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1'-- Hungup 'Zap/5-1' -- Starting simple switch on 'Zap/5-1'[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)...[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)...[Jan 5 12:45:06] NOTICE[2893]: chan_dahdi.c:6869 ss_thread: Got event 17 (Polarity Reversal)...-- Executing [...@from-pstn:1] Answer(Zap/5-1, ) in new stack-- Executing [...@from-pstn:2] Playback(Zap/5-1, vm-intro) in new stack-- Zap/5-1 Playing 'vm-intro' (language 'en')[Jan 5 12:45:08] WARNING[2893]: chan_dahdi.c:4550 dahdi_handle_event: Ring/Off-hook in strange state 6 on channel 5-- Executing [...@from-pstn:3] Hangup(Zap/5-1, ) in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/5-1'-- Hungup 'Zap/5-1' I am using 1.4.30 and zaptel 1.12. Any cluess?*---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- 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] PostgreSQL is asterisk friendly with it?
I use Postgres always and it is wonderful. Never use mysql so if you want a real DB just use Postgres *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* From: benny+use...@amorsen.dk To: brya...@zktech.com Date: Mon, 13 Sep 2010 20:24:25 +0200 CC: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] PostgreSQL is asterisk friendly with it? Bryant Zimmerman brya...@zktech.com writes: As I look to move our systems to version 1.8 I am looking at making a change from mySQL to PostgreSQL. I love mySQL but am getting very concerned about i'ts new owners. Should I be able to move all my realtime stuff to PostgreSQL is it fully supported with asterisk? Yes. The ODBC drivers don't really care which database you access. Is there any down side to PostgreSQL over mySQL or will it be a big win? The only issue we have with Postgres is the dump/reload cycle when upgrading database version. This is being fixed in the latest versions though. Our database servers are linux but we access them from asterisk as well as windows are there any thing to be concerned with there? It works fine from Windows as well. /Benny -- _ -- 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://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] How to finish an AGI
IMHO, is more easy in Perl that in dialplan but if for you work ..
*---*
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Date: Fri, 3 Sep 2010 10:29:02 +0200
From: ing.diasda...@gmail.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to finish an AGI
Any particular reason you don't want to put the logic of the macro in your AGI?
Yes...i've no idea how to do it...it's a PERL script, i'm already checking how
to do this...but it will be a little complicated :(
2010/9/3 Steve Edwards asterisk@sedwards.com
On Thu, 2 Sep 2010, Danny Dias wrote:
Sorry for my poor explanation...what i'm trying to do is to invoke a Macro from
my AGI, like this:
$agi-exec(Macro,check-call-limit);
If the Macro checks that the group_name is bigger than a number specified for
every peer with setvar it should Hangup the call (frobidden,1 in the Gotoif...)
but this
is not happening, the AGI always continue with is process and it doesn´t play
attention to the Hangup in the macro, the macro is here:
[macro-check-call-limit]
exten = s,1,Set(group_name=out_calls_user_${SIPCHANINFO(peername)})
exten = s,n,Set(GROUP()=${group_name})
exten = s,n,GotoIf($[${GROUP_COUNT(${group_name})}
${MAX_OUT_CALLS_PER_USER}] forbidden,1)
; EXITO:
exten = s,n,MacroExit
; FRACASO:
exten = forbidden,1,NoOp(*** llamada saliente bloqueada: el usuario
${SIPCHANINFO(peername)} tiene actualmente
${MATH(${GROUP_COUNT(${group_name})})-1,int)} llamadas
salientes)
exten = forbidden,n,Hangup(21) ; ISUP 21 = SIP 403 (Forbidden)
The concept of calling a macro from within an AGI seem convoluted, but may
work. I've never tried it.
Any particular reason you don't want to put the logic of the macro in your AGI?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] Video IVR Asterisk ?
Just a question what is the advantage to do a video IVR, really I dont understand? Maybe, I am in the prehistory, in my country there is no bandwith for this, so somebody can explain me this,just for acknowledgement *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* From: j...@sunfone.com To: asterisk-users@lists.digium.com Date: Fri, 16 Jul 2010 14:09:54 -0500 Subject: Re: [asterisk-users] Video IVR Asterisk ? On Sat, 2010-07-17 at 00:08 +0530, Anita Hall wrote: Hi Is it possible to receive video calls using Asterisk and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of your valuable inputs. regards, Anita Hall, Simmortel. We use Grandstream video phones and have noticed that if we record our prompts with these phones, the video is saved with the audio. So we set our main IVR up this way, and without doing anything special (other than enabling video in sip.cfg), we have video IVR for those customers that call with video capable endpoints. j -- _ -- 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://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] Asterisk distribution for a Call Center
The best option JUST ASTERISK without anything else. Maybe you need hire somebody with expereince with callcenter. *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Tue, 22 Jun 2010 15:21:18 -0300 From: aco1...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Asterisk distribution for a Call Center Dear all, I need to build a PBX based on Asterisk for a call center. I have worked with raw Asterisk but it's hard to work for big implementations think. Also I have worked with Trixbox CE for a small bussines and it was prette good, but I have not have many features like ACD. I know there is another version called Trixbox PRO -specially Call Center edition- that's not free but has got more features like ACD and billing. I've heart about AsteriskNow and I know it's free. What distribution/version do you recommend to me in order to implement a call center and taking into account I'm not an expert in programming from Asterisk CLI ??? Thanks a lot Alejandro -- _ -- 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://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] [AGI] What scripting language for embedded hardware?
Uhmmm.. remember for each channel you run perl or php interpreter so with that amount of memory maybe this can be a problem. For that kind of project I'd use C or java as fastagi protocol From: desired@gmail.com Date: Mon, 21 Jun 2010 17:25:09 +0300 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [AGI] What scripting language for embedded hardware? If you can install python or PHP in that machine (in means of storage), you are free to run it there. 64 RAM is really enough to run python, so you have to just try if it suits in the application. If it takes too slow to initialize - try to find some embedded versions. openwrt, for instance, has one, that means it's possible to run python on wrt54gl (16 MB RAM, 200MHz MIPS), so your platform should be really possible. On Mon, Jun 21, 2010 at 3:48 PM, Gilles codecompl...@free.fr wrote: Hello I'm learning how to work with Asterisk on an embedded system (MMU-less Blackfin processor, 64MB RAM and 256MB NAND), and was wondering what people use as scripting language to handle calls through the dialplan and AGI, considering the hardware limitations? Ideally, I'd rather use a rich language like PHP or Python, but can those be fit with even their common modules into such small hardware? I'm also thinking of Lua and modules, provided they can be included in the buildroot. Thank you for any feedback. -- _ -- 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://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://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] own Caller ID
Just is PRI line you can do it.. *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Tue, 8 Jun 2010 12:44:07 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] own Caller ID On Tue, 8 Jun 2010, taimur hasan wrote: I want to use my own caller id, instead of the caller id of PSTN line, for the outbound calls through DAHDI channel. Is there any way ?? It depends on your technology (POTS, PRI, etc) and your provider. Tell your provider you want to set the outgoing caller ID and see what their response is. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 _ -- _ -- 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] Queue with PopUP screen for customer
I installed a queue for a client with 10 officers so far so good. Now the client wants an agent when making a call to this will leave any customer information using the phone as a key. I'm trying to do this app using delphi obviously I will need to connect to the AMI, but do not quite understand how to identify the call you get to a specific agent. Can you give me some point where to start. *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* _ -- _ -- 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] Recording with extension and agent in queue
Hi! I am recording with asterisk and so far so good. Now I need to use in the name of recording wich extension that takes the call and the agent in the queue that takes the call/ Is there a way to know what extension and the agent that take the call in a queue for recording??? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* _ -- _ -- 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] voipmonitor.org
Date: Mon, 10 May 2010 09:39:55 +0200 From: v...@lam.cz To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] voipmonitor.org On 8.5.2010 00:40, Jeff Brower wrote: Martin- checkout new open source voipmonitor.org SIP packet sniffer. I've developed it for my telco company and I've decided to share it. Testing and contributions are welcome! VoIPmonitor is open source live network packet sniffer which analyze SIP and RTP protocol. It can run as daemon or analyzes already captured pcap files. For each detected VoIP call voipmonitor calculates statistics about loss, burstiness, latency and predicts MOS (Meaning Opinion Score) according to ITU-T G.107 E-model. These statistics are saved to MySQL database and each call is saved as pcap dump. Web PHP application (it is not part of open source sniffer) filters data from database and graphs latency and loss distribution. Voipmonitor also detects improperly terminated calls when BYE or OK was not seen. To accuratly transform latency to loss packets, voipmonitor simulates fixed and adaptive jitterbuffer. How many channels can it handle simultaneously? I've not tested limits but capturing 15 voip calls takes 3-4% on Core2 2.40GHz. Complexity in worst case is O(N^2) where N is number of calls. Packets are matched as llinear list of IP and port. If this will be limit, it could be rewriten to hash table O(N) How does it do MOS prediction if low bitrate codecs are being used (G729, AMR, etc)? It is calibrated only to G.711 with PLC for now but I'm planing adding equations for G.729 and iLBC. MV Maybe this question is out little but is the same context. I need read the VoIP packets and order all this packets in another place to get the audio. The idea is can record a call using directly the packets. I know asterisk can record but my problem is that I have Avaya and asterisk working togheter and I can not record by Avaya and somebody tells me this idea to sniff the VoIP packets order after the call. I am seeing the code for VoIp monitor Is it so stupid?? TIA _ -- _ -- 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] AGI, FASTAGI or Windows Voice Server
Now I created the file in Windows and with Sox convert it to Asterisk wav format. I havent tried with samba but if with FTP. My problem is that file created must be played in the same call in progress so I cant wait to finish the call because I need the file. I think the only way to do that is using FastAGI but I have not worked for me yet. I am using Perl to FastAGi but the examples that I find are so confused. I have created a lot of AGI perl but in the same asterisk server. Call -- Asterisk AGI Answer -- Windows Create File - Copy File Asterisk -- Play File -- Finish Call My last option was send the file for the same socket that I create to send the name and text. ASterisk send request with text and file name to windows server in port Windows server respond with the file but using the same socket not FTP or samba, I tried this using FTP and did not work TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Sun, 18 Apr 2010 19:20:56 -0400 From: stot...@first-notification.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server I figured as much. ESOL=English as a Second Language. Apology accepted. Have you tried creating the file on the windows server, running sox to your specifications and then moving the file to a samba share? The key to this is moving the files at different stages. The first sound file is being created while the call is in progress. When the call is finished, move the file to a different location to process, after processing, move it to it's final destination so it can be played. Thanks, Steve T On Sun, Apr 18, 2010 at 2:00 PM, Edwin Quijada listas_quij...@hotmail.com wrote: Sorry if u understood this my english is so limited and not so good , my apologize it was not my intention. *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Sat, 17 Apr 2010 18:19:53 -0400 From: stot...@first-notification.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server On Sat, Apr 17, 2010 at 1:48 PM, Edwin Quijada listas_quij...@hotmail.com wrote: Just a shot in the dark, have you tried ExternalIVR? It was originally developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer teamed up on this one. This option NO. Another option would be FastAGI to your windows server. You write an app for the windows box that interacts with the ATT application and then pipe the audio back to your asterisk box somehow. First thought is app_bridge or meetme. This is the idea just I dont know how to do. You can give any direction to start first. I am looking for information about app_bridge *---* *-Edwin Quijada *-Developer DataBase king *-JQ Microsistemas *-Soporte PostgreSQL*-www.jqmicrosistemas.com*-809-849-8087*---* This option NO. is quite a rude reply when someone is giving you ideas for free. Maybe you can say why it is not an option but your response was rude and makes me not want to help you anymore. I can tell you are an ESOL by the way you write, so maybe you don't understand the best way to communicate. Also, if you tried FTP, then did you not post that first. What else have you tried? Why waste people's time when you have tried things that didn't work but don't convey them? Did you try Samba? As far as app_bridge, there is plenty of documentation, let me waste more of my time.. http://tinyurl.com/y73mp9s Sounds like you should pay for the Linux version or paid Asterisk support. I really appreciate helping you, thanks, Steve Totaro -- _ -- 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://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo
[asterisk-users] Help with FastAGI server in Windows
Hi! I am trying to do a FastAGI server in windows. I am using the example from their page but I dont get anything. Anybody here has experienced with Fastagi in windows and perl that give a rigth direction to do this. I have experience with AGI but fastagi dont *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* _ -- _ -- 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] AGI, FASTAGI or Windows Voice Server
Sorry if u understood this my english is so limited and not so good , my apologize it was not my intention. *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Sat, 17 Apr 2010 18:19:53 -0400 From: stot...@first-notification.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server On Sat, Apr 17, 2010 at 1:48 PM, Edwin Quijada listas_quij...@hotmail.com wrote: Just a shot in the dark, have you tried ExternalIVR? It was originally developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer teamed up on this one. This option NO. Another option would be FastAGI to your windows server. You write an app for the windows box that interacts with the ATT application and then pipe the audio back to your asterisk box somehow. First thought is app_bridge or meetme. This is the idea just I dont know how to do. You can give any direction to start first. I am looking for information about app_bridge*---* *-Edwin Quijada *-Developer DataBase king *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com*-809-849-8087*---* This option NO. is quite a rude reply when someone is giving you ideas for free. Maybe you can say why it is not an option but your response was rude and makes me not want to help you anymore. I can tell you are an ESOL by the way you write, so maybe you don't understand the best way to communicate. Also, if you tried FTP, then did you not post that first. What else have you tried? Why waste people's time when you have tried things that didn't work but don't convey them? Did you try Samba? As far as app_bridge, there is plenty of documentation, let me waste more of my time.. http://tinyurl.com/y73mp9s Sounds like you should pay for the Linux version or paid Asterisk support. I really appreciate helping you, thanks, Steve Totaro _ -- _ -- 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] AGI, FASTAGI or Windows Voice Server
I did using FTP. This is the problem and the solution that I did but doesnt work
1-When the call in to asterisk I play one prompt if this prompt doesnt exist I
create it2-In windows I have a program listen on a port waiting for request
from asterisk 3- I sent by this socket the text and name for the file4- In
windows server create the file and convert to 8khz using sox5- From windows try
to copy this file to asterisk using FTP protocol 6- There is no syncronize
between AGI script and copy to FTP 7- I did a loop to wait for copy of file to
my sound directory but it never happenned because it couldnt create the file 8-
if I put off the loop while (!existfile) { } so it can create the file in
windows I really dont know why this behaviour
My plan was so simple A server waiting request for asterisk and the copy this
file to asterisk to play itbut doesnt work, for this reason i am trying to do
everything using FastAgi in a windows server.
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Date: Sat, 17 Apr 2010 13:23:22 -0400
From: stot...@first-notification.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server
On Fri, Apr 16, 2010 at 5:58 PM, Edwin Quijada listas_quij...@hotmail.com
wrote:
Why don’t you use sox to transform the windows audio file into the asterisk
format – I do this with pretty good results.
I did. But my problem is not conversion my problem is that I dont know how play
the file from windows server or copy this to asterisk without my AGI continue
and desyncronyze it.
Can you explain me exactly what did you do /?
Do you have something like this using AGI ?
I use sox with good results too in windows. The problem is when create the file
and convert it , how send to asterisk
Edwin Jaws
If you just need to transfer a file to a linux box, there are plenty of ways.
FTP, SFTP, TFTP, Samba.
Thanks,
Steve T
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Re: [asterisk-users] AGI, FASTAGI or Windows Voice Server
Just a shot in the dark, have you tried ExternalIVR? It was originally developed for Unwired Buyer (Ebay), I believe Digium and Unwired Buyer teamed up on this one. This option NO. Another option would be FastAGI to your windows server. You write an app for the windows box that interacts with the ATT application and then pipe the audio back to your asterisk box somehow. First thought is app_bridge or meetme. This is the idea just I dont know how to do. You can give any direction to start first. I am looking for information about app_bridge*---* *-Edwin Quijada *-Developer DataBase king *-JQ Microsistemas *-Soporte PostgreSQL*-www.jqmicrosistemas.com*-809-849-8087*---* _ -- _ -- 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] AGI, FASTAGI or Windows Voice Server
Hello! I have developed an IVR using AGI and so far it works great. I'm using Cepstral voices, but now want to use the voices from AT T that are on a Windows server to be heard best. With cepstral what I do is to generate audio files from shipping and this text I reproduce this method it has worked very well. Now, try to do the same by creating the audio file in windows with the voices of AT T, the problem is that there is no way to synchronize the generation of the audio file and step Asterisk to be played, so it occurred to me to use FastAGI to generate all Windows and play in the same window the audio file generated. We buy Linux licenses for the voices but they are very expensive and already bought windows for another project. How do you think would be the best option? If you have another idea, please Tell me because I'm getting crazy with this and can not solve. TIA Edwin _ -- _ -- 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] AGI, FASTAGI or Windows Voice Server
Why don’t you use sox to transform the windows audio file into the asterisk format – I do this with pretty good results. I did. But my problem is not conversion my problem is that I dont know how play the file from windows server or copy this to asterisk without my AGI continue and desyncronyze it. Can you explain me exactly what did you do /? Do you have something like this using AGI ? I use sox with good results too in windows. The problem is when create the file and convert it , how send to asterisk Edwin Jaws _ -- _ -- 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] FastAGiin Windows Server
Date: Wed, 14 Apr 2010 21:09:03 -0400 From: dbackeb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FastAGiin Windows Server On Wed, Apr 14, 2010 at 8:55 PM, Edwin Quijada listas_quij...@hotmail.com wrote: My problem is that I need to execute windows app using IVR in Asterisk so we What is the windows app that you cannot replace on Linux? How about wrapping THAT program with simple inputs and outputs, and build a network interface on top of it, then bounce interface calls back and forth from linux? It is a custom app not mine. _ -- _ -- 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] FastAGiin Windows Server
My problem really is find out how Asterisk::fastagi works. Date: Thu, 15 Apr 2010 13:05:03 -0800 From: s...@inbox.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FastAGiin Windows Server You could always ask someone to rewrite the perl code to something else. -Original Message- From: listas_quij...@hotmail.com Sent: Thu, 15 Apr 2010 20:52:45 + To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FastAGiin Windows Server Date: Wed, 14 Apr 2010 21:09:03 -0400 From: dbackeb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FastAGiin Windows Server On Wed, Apr 14, 2010 at 8:55 PM, Edwin Quijada listas_quij...@hotmail.com wrote: My problem is that I need to execute windows app using IVR in Asterisk so we What is the windows app that you cannot replace on Linux? How about wrapping THAT program with simple inputs and outputs, and build a network interface on top of it, then bounce interface calls back and forth from linux? It is a custom app not mine. _ -- _ -- 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://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FastAGiin Windows Server
Hi! I wanna know if can run my AGI scripts as fastAGI scripts in Windoes server. I need a lot of script done in perl and I wanna move to windows server. I checked Asterisk::fastagi but I see that everything is for Linux. Somebody has idea to do this in perl. I dont want to change the language. TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* _ -- _ -- 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] FastAGiin Windows Server
My problem is that I need to execute windows app using IVR in Asterisk so we need FastAGI using perl. I saw Asterisk::fastagi but everything for this is in Linux and i dont know if it works in windows. I need to know if somebody has used fastagi in windows with perl becuase I have a lot of agi in perl TIA Date: Wed, 14 Apr 2010 10:04:27 -0700 From: asterisk@sedwards.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FastAGiin Windows Server On Wed, 14 Apr 2010, Edwin Quijada wrote: I wanna know if can run my AGI scripts as fastAGI scripts in Windoes server. Seems like a move in the wrong direction to me, but no -- you can't run an AGI script via fastagi() without changes. I need a lot of script done in perl and I wanna move to windows server. I checked Asterisk::fastagi but I see that everything is for Linux. Fastagi is a protocol. You could implement it in most languages on most OSs. (Everything is for Linux because that's where server stuff belongs.) Somebody has idea to do this in perl. I dont want to change the language. Somebody should re-think their ideas :) Saying you need a lot of script done implies you haven't done it yet. I'd suggest changing your language to C. You can execute XXX AGIs written in C in the time it takes to load Perl and parse your script. Maybe you wouldn't even need to use a separate server. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- 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://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with Callfiles
Hi! I am trying to do a callfiel for autodialing but when I move the callfile to outdialing folder asterisk seems like if did the call but it doesnt. I put here my callfile and that I get when asterisk begins to do the call If anybody has idea, pls. Tell me TIA ;;CallFile- Channel: Zap/g1/8093908270 Callerid: 8093908270 MaxRetries: 2 RetryTime: 300 WaitTime: 45 Context: 1call Extension: s Priority: 1 ;;EXTENSION:: [1call] exten = s,1,Playback(vm-intro) exten = s,2,Playback(vm-goodbye) exten = s,3,Hangup I am getting this when I put the 1.call to outgoing directory. The call never started == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Connected to Asterisk 1.4.30 currently running on ivr-server (pid = 1873) Verbosity is at least 5 Channel Zap/8-1 was answered. -- Executing [...@1call:1] Playback(Zap/8-1, vm-intro) in new stack -- Zap/8-1 Playing 'vm-intro' (language 'en') -- Executing [...@1call:2] Playback(Zap/8-1, vm-goodbye) in new stack -- Zap/8-1 Playing 'vm-goodbye' (language 'en') -- Executing [...@1call:3] Hangup(Zap/8-1, ) in new stack == Spawn extension (1call, s, 3) exited non-zero on 'Zap/8-1' -- Hungup 'Zap/8-1' [Apr 13 00:54:03] NOTICE[2493]: pbx_spool.c:370 attempt_thread: Call completed to Zap/g1/8093908270 I tested the channel doing a call to this and I get this, the call worked -- Starting simple switch on 'Zap/8-1' [Apr 13 00:58:27] NOTICE[2496]: chan_dahdi.c:6869 ss_thread: Got event 18 (Ring Begin)... [Apr 13 00:58:28] NOTICE[2496]: chan_dahdi.c:6869 ss_thread: Got event 2 (Ring/Answered)... [Apr 13 00:58:32] NOTICE[2496]: chan_dahdi.c:6869 ss_thread: Got event 18 (Ring Begin)... -- Executing [...@from-pstn:1] Answer(Zap/8-1, ) in new stack -- Executing [...@from-pstn:2] Playback(Zap/8-1, vm-intro) in new stack -- Zap/8-1 Playing 'vm-intro' (language 'en') -- Executing [...@from-pstn:3] Hangup(Zap/8-1, ) in new stack == Spawn extension (from-pstn, s, 3) exited non-zero on 'Zap/8-1' -- Hungup 'Zap/8-1' *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* _ -- _ -- 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] Callcenter open source program
gNUDIALER *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Sun, 7 Mar 2010 06:21:34 -0800 From: wassimdarwi...@yahoo.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Callcenter open source program HI all: Iam planning to use my asterisk box as callcenter ,any one can advice me with the best callcenter open source program based on asterisk . Any help will be apreciated. _ -- _ -- 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] Avaya with Asterisk
I have a connection of Asterisk with Avaya by H.323 and so far everything worked well because only sent to Avaya. Now, the matter is that from Avaya will send me an IVR calls to capture credit card information, the link is active on Avaya 23 channels which is not how to configure Asterisk for those 23 simultaneous channels of Avaya's collect asterisk. Do not know if I can be with a group or queue, the idea is that all calls go to one place and who answer all calls is the IVR. Any suggestions or ideas? Edwin Quijada *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* _ -- _ -- 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] Open source or low-budget recommendation for call-center software
GnuDialer *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* From: juanch...@gmail.com Date: Mon, 22 Feb 2010 16:37:22 -0500 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Open source or low-budget recommendation for call-center software I think Vicidial, works great. Regards. 2010/2/22 Apa Minerala apaminer...@yahoo.com Hello, We used to recommend a commercial software but client is a small callcenter who cannot afford something big. Would you recommend something open-source which could work for a 40-seater? Thank you, Tudor www.sunabasarabia.com Moldova 11c/min Romania 2c/min $1 de test de la bun inceput -- _ -- 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 -- Juan. _ -- _ -- 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] ivvr with asterisk
Yes, you can using SIP *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* From: qu...@vega.com.vn To: asterisk-users@lists.digium.com Date: Mon, 25 Jan 2010 08:35:31 +0700 Subject: Re: [asterisk-users] ivvr with asterisk Thanks all, Before purchasing any device i want to make some prototype of IVVR, is it possible to use asterisk to build an IVVR with softphones (such as SIP softphone)? and Is there any example about these? Quyps On Sat, 2010-01-23 at 11:44 +0530, mtha...@gmail.com wrote: Quyps, It looks like you mis-read the picture. Asterisk is the core, it need to be there regardless you use FreePBX or Tribox. FreePBX is a GUI web interface to manage asterisk. Itself is not an IP-PBX. Trixobx, still based on the Asterisk + freePBX, adds some more additional applications based on the community feed back and requirement. Trixbox is an easy go, but there may be some unwanted stuff with it. elastix.org is also a nice package, give it a try. Regards MT Kondela kevesystems.com On Sat, Jan 23, 2010 at 7:32 AM, Pham Quy qu...@vega.com.vn wrote: Hi all, First I'm very new. I want to build an Interactive Video-voice Response system. There is number of choice I have found so far: FreePBX, TriBox, Asterisk. Which is the best in my case? and what do i need to build such IVVR system? Thanks. Quyps -- _ -- 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] Problem with my dialplan
U alrigth! The number begins with 8 the TELCO sent this number like DID *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* Date: Sun, 10 Jan 2010 15:35:40 -0800 From: doctor.w...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Problem with my dialplan The 8 probably comes from the T1, does the telephone number end with an 8? The playback of ss-noservice might be a fallback ensuring that *something* happens when a call comes in On Sun, Jan 10, 2010 at 1:31 PM, Edwin Quijada listas_quij...@hotmail.com wrote: Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' == Unknown extension '8' in context 'from-ptsn' requested -- Zap/1-1 Playing 'ss-noservice' (language 'en') -- Hungup 'Zap/1-1' ivr-server*CLI ivr-server*CLI dialplan show [ Context 'defaults' created by 'pbx_config' ] Include ='from-ptsn' [pbx_config] [ Context 'from-ptsn' created by 'pbx_config' ] 's' =1. Answer() [pbx_config] 2. Playback(vm-Work) [pbx_config] 3. Hangup() [pbx_config] [ Context 'parkedcalls' created by 'res_features' ] '700' = 1. Park() [res_features] -= 2 extensions (4 priorities) in 3 contexts. =- ivr-server*CLI El extension es este [general] language=en [from-ptsn] exten = s,1,Answer() exten = s,2,Playback(vm-Work) exten = s,3,Hangup() [defaults] include = from-ptsn *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* -- _ -- 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] Problem with my dialplan
Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' == Unknown extension '8' in context 'from-ptsn' requested -- Zap/1-1 Playing 'ss-noservice' (language 'en') -- Hungup 'Zap/1-1' ivr-server*CLI ivr-server*CLI dialplan show [ Context 'defaults' created by 'pbx_config' ] Include ='from-ptsn' [pbx_config] [ Context 'from-ptsn' created by 'pbx_config' ] 's' =1. Answer() [pbx_config] 2. Playback(vm-Work) [pbx_config] 3. Hangup() [pbx_config] [ Context 'parkedcalls' created by 'res_features' ] '700' = 1. Park() [res_features] -= 2 extensions (4 priorities) in 3 contexts. =- ivr-server*CLI El extension es este [general] language=en [from-ptsn] exten = s,1,Answer() exten = s,2,Playback(vm-Work) exten = s,3,Hangup() [defaults] include = from-ptsn *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* _ -- _ -- 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] Asterisk Autodialer
I did with Gnudialer. *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* From: sanjoy_r...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 25 Aug 2009 17:19:52 + Subject: Re: [asterisk-users] Asterisk Autodialer Thanks Miguel. Have your configured GNUDialer before? Date: Tue, 25 Aug 2009 11:22:16 -0500 From: mmol...@millenium.com.co To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Autodialer Sanjoy Rath escribió: Anyways I checked VOIP-info.org the information there was pretty basic. I was trying to get some more insight to this autodialer stuff. If there is something I can take leverage of that will be great (because I do not want reinvent the wheel) or else (as sadi before) I will figure out. If you for example manage to configure and test GNUdialer (http://www.gnudialer.org/ , http://dynx.net/ASTERISK/gnudialer/) by yourself, that would be a good start into knowing how does a basic dialer works. Maybe VICIDIAL has better documentation but its internals and initial setup are far away difficult to understand (IMO). More than that, you won't find anything else on the scope of Open Source dialers for asterisk (AACC - Hanashi Dialer is in a very alpha stage). Anything else is closed and/or commercial. Cheers, -- Ing. Miguel Molina Grupo de Tecnología Millenium Phone Center ___ -- 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 Stay on top of things, check email from other accounts! Check it out. _ ___ -- 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] Server linux requirements
It depends about your traffic. But myabe and I guess Core 2Duo 4Gb ram , Sata 160Gb +_ *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-Soporte PostgreSQL *-www.jqmicrosistemas.com *-809-849-8087 *---* From: clubtorr...@hotmail.com To: asterisk-users@lists.digium.com Date: Tue, 4 Aug 2009 15:37:50 + Subject: [asterisk-users] Server linux requirements Hello to all. I am about to initiate to prove asterisk, but that I need a Server linux, that requirements recommend to me that it has my Server? we will use, it for recording of calls. and reports of calls saludos ___ Carlos Rodriguez ¿Quieres un regalo de cumpleaños? Messenger te lo da _ ___ -- 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] Asterisk, SQL Database Update
Perl and AGI Piece of cake.!!! *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* From: torinti...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 25 May 2009 22:28:54 +0300 Subject: Re: [asterisk-users] Asterisk, SQL Database Update Thanks for your helpful reply. I am not so good in coding. simply all i need is as follow When a call comes, goes into an IVR, and then depending on the entry option it will connect to a remote SQL Database, to check the pre-existed data, and in the end of the IVR the caller will enter an option that will need to be written in the SQL Database. Can you please give me a general scenrio how this will be achieved. and which files that i will need to modify. Thanks a lot. Date: Sun, 24 May 2009 22:15:31 +0200 From: philipp.kemp...@amooma.de To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk, SQL Database Update Torintino T schrieb: Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Depending on what you are trying to do there are various solutions: Channel Event Logging (CEL) - http://www.asterisk.org/node/48358 AGI System() ODBC_*() functions Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- 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 check out the rest of the Windows Live™. More than mail–Windows Live™ goes way beyond your inbox. More than messages _ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008 ___ -- 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] Asterisk, SQL Database Update
Thanks for your helpful reply. I am not so good in coding. simply all i need is as follow When a call comes, goes into an IVR, and then depending on the entry option it will connect to a remote SQL Database, to check the pre-existed data, and in the end of the IVR the caller will enter an option that will need to be written in the SQL Database. Can you please give me a general scenrio how this will be achieved. and which files that i will need to modify. I think that if you are not good coding you will have a few problems. Maybe, the best solution 4u is hire external to do that. It is simple but just in dialplan it is so difficult with AGI it is so easy but you dont want coding. *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Windows Live Hotmail now works up to 70% faster. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster_112008 ___ -- 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] How to Integrate Neospeech with Asterisk
Can You post your solution? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* Date: Fri, 27 Mar 2009 07:55:45 -0500 From: deric.p...@nisc.coop To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How to Integrate Neospeech with Asterisk I’ve used NeoSpeech’s Java API to build a custom TTS interface that creates sound files. I call that from Asterisk using AGI. Then I just have Asterisk play the file I created. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of msp Sent: Friday, March 27, 2009 5:31 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to Integrate Neospeech with Asterisk Hi all, I was wondering if anyone knows how to integrate the Neospeech Text to Speech engine with asterisk. I have scoured the web and haven't found anything. I think it's possible, I just don't know how to do it. If Any body tried Neospeech with Asterisk then kindly share the experience or comment. Thanks, msp _ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008___ -- 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] check if not human
NVLineDetect , I dont find it in the web for asterisk 1.4 Anybody has a link that works? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* Date: Fri, 20 Feb 2009 09:31:41 -0800 From: nt_aster...@yahoo.com To: asterisk-users@lists.digium.com CC: nt_jnew...@yahoo.com Subject: Re: [asterisk-users] check if not human NVGenderDetect is new, but you can find NVLineDetect on the web. From: David fire ddf...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, February 19, 2009 3:00:14 PM Subject: Re: [asterisk-users] check if not human NVLineDetect, NVGenderDetect what is that? amd info voip-info.org or asterisk.org support asterisk book. i bougth one to support the cause!!! David 2009/2/19 Asterisk Asterisk nt_aster...@yahoo.com You can probably use combo of NVLineDetect, NVGenderDetect, and AMD (NVMachineDetect). From: Edwin Quijada listas_quij...@hotmail.com To: Asterisk Asterisk asterisk-users@lists.digium.com Sent: Thursday, February 19, 2009 12:55:05 PM Subject: Re: [asterisk-users] check if not human How can I detect how many ring a call to hangup? Where I can find info about AMD? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* Get Windows Live and get whatever you need, wherever you are. Start here. ___ -- 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 -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your ()_()signature to help him gain world domination. _ See how Windows® connects the people, information, and fun that are part of your life http://clk.atdmt.com/MRT/go/119463819/direct/01/___ -- 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] Credit Card processing machines
Date: Wed, 18 Feb 2009 09:50:18 -0800 From: bilmar...@yahoo.com Subject: Re: Credit Card processing machines To: asterisk-users@lists.digium.com CC: listas_quij...@hotmail.com And is there a bank accept to give such kind of communication? The user was able to dial his card number and the amount from his phone (or IP Phone registered with Asterisk), and Asterisk communicate with the bank or company credit card provider? Yes! WEll, no asterisk exactly, we can do an interface to talk with verifone by RS232 and send commands How the user will enter $50.25? What about expiration date of the credit card? You can use *, key, for period and finish the value with # 50*25# the AGI validate the data Regards Bilal *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Get more from your digital life. Find out how. http://www.windowslive.com/default.html?ocid=TXT_TAGLM_WL_Home2_082008___ -- 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] Credit Card processing machines
Date: Thu, 19 Feb 2009 11:25:50 -0800 From: bilmar...@yahoo.com Subject: RE: Credit Card processing machines To: asterisk-users@lists.digium.com; listas_quij...@hotmail.com Why not Asterisk? And if need to use RS232, then ethernet is not possible? So how u will use AGI with RS232? Yes, you can but I dont know how *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Reveal your inner athlete and share it with friends on Windows Live. http://revealyourinnerathlete.windowslive.com?locale=en-usocid=TXT_TAGLM_WLYIA_whichathlete_us___ -- 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] check if not human
How can I detect how many ring a call to hangup? Where I can find info about AMD? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Get Windows Live and get whatever you need, wherever you are. Start here. http://www.windowslive.com/default.html?ocid=TXT_TAGLM_WL_Home_082008___ -- 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] Credit Card processing machines
Our creditcard company's small print _insists_ on a direct analog exchange line with no other devices in between. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk You can do it an interface using AGI to comunicate with equipment or verifone. I did it once *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Got Game? Win Prizes in the Windows Live Hotmail Mobile Summer Games Trivia Contest http://www.gowindowslive.com/summergames?ocid=TXT_TAGHM___ -- 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] Avaya and Asterisk sound one-way
Hi! I have connected an Avaya System with my asterisk but when I call to avaya extension I can hear everything but when I speak from Aterisk extension the person in AVaya cant hear me. I have seen this issue so much in internet but any solution. Any help or any cuees?? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008 ___ -- 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] Avaya and Asterisk sound one-way
You say Connected but do not specify in what fashions you are connecting. That piece of info will be the solution. I have done this many times in many fashions. OK. My Avaya is a definity 87000 , Asterisk 1.4.21 *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Color coding for safety: Windows Live Hotmail alerts you to suspicious email. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_safety_112008 ___ -- 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] Avaya and Asterisk sound one-way
You say Connected but do not specify in what fashions you are connecting. That piece of info will be the solution. I have done this many times in many fashions. Sorry for my bad english but I dont understand what info you need to know. You can ask me for anything that u need. Conection by H323 protocol Using The NuFone Network's Open H.323 driver configuration Pwlib for asterisk \ If u need something else, just ask me.! *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/ ___ -- 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] Avaya and Asterisk sound one-way
Well, thks anyway :) Maybe in another ocasion with T1 trunk :) *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* Date: Sat, 31 Jan 2009 18:58:09 -0500 From: stot...@first-notification.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Avaya and Asterisk sound one-way On Sat, Jan 31, 2009 at 6:42 PM, Steve Totaro wrote: On Sat, Jan 31, 2009 at 6:25 PM, Edwin Quijada wrote: You say Connected but do not specify in what fashions you are connecting. That piece of info will be the solution. I have done this many times in many fashions. Sorry for my bad english but I dont understand what info you need to know. You can ask me for anything that u need. Conection by H323 protocol Using The NuFone Network's Open H.323 driver configuration Pwlib for asterisk \ If u need something else, just ask me.! *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/ ___ -- 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 H323 is a dog in Asterisk, JerJer can probaby help you out ot this one. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) Bye the way, JerJer is the man at NuFone. I don't use H323 but I believe there are other H323 implementations other than JerJer's you might want to try. JerJer was never a help to me and purposely ripped me off for ~$40, this was years ago, I prepaid and the service was down with no tech support, so I asked for a refund that I never got. It is in the archives somewhere. JerJer/NuFone=bad at least at the beginning. I would look elsewhere. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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 _ Get 5 GB of storage with Windows Live Hotmail. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_5gb_112008 ___ -- 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 with Avaya
Hi !
I am trying to connect Asterisk with Avaya Definity.
I use this tutorial to do this
http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html
The comunication between avaya and asterisk is fine but without sound. I can
call from Asterisk to Avaya and extension ring or Avaya to Asterisk and
extension ring too but I cant hear anything
Example
Asterisk --- Avaya
-- Executing [73...@internal:1] Dial(SIP/59000-08203708, H323/73...@avaya)
in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called 73...@avaya
-- H323/Avaya-1 is making progress passing it to SIP/59000-08203708
-- H323/Avaya-1 is ringing
-- H323/Avaya-1 answered SIP/59000-08203708
== Spawn extension (internal, 73133, 1) exited non-zero on
'SIP/59000-08203708'
Everything good but I cant hear anything.
Avaya Asterisk
-- Executing [59...@internal:1] Answer(H323/ip$10.200.1.47:23924/18397,
) in new stack
-- Executing [59...@internal:2] Playback(H323/ip$10.200.1.47:23924/18397,
vm-intro) in new stack
-- Playing 'vm-intro' (language 'en')
-- Executing [59...@internal:3] Playback(H323/ip$10.200.1.47:23924/18397,
vm-goodbye) in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing [59...@internal:4] Playback(H323/ip$10.200.1.47:23924/18397,
vm-intro) in new stack
-- Playing 'vm-intro' (language 'en')
-- Executing [59...@internal:5] Wait(H323/ip$10.200.1.47:23924/18397,
2) in new stack
-- Executing [59...@internal:6] Hangup(H323/ip$10.200.1.47:23924/18397,
) in new stack
== Spawn extension (internal, 59000, 6) exited non-zero on
'H323/ip$10.200.1.47:23924/18397'
In this case just play a message but I cant hear anything again.
This is my conf files
==EXTENSION===
[general]
language=en
static=yes
autofallthrough=yes
[internal]
;My extension 59xxx
;exten = 59000,1,Dial(SIP/59000)
;exten = 59000,2,VoiceMail(59...@118218)
;exten = 59000,3,PlayBack(vm-goodbye)
;exten = 59000,4,Wait(2)
;exten = 59000,5,HangUp()
exten = 59000,1,Answer
exten = 59000,2,PlayBack(vm-intro)
exten = 59000,3,PlayBack(vm-goodbye)
exten = 59000,4,PlayBack(vm-intro)
exten = 59000,5,Wait(2)
exten = 59000,6,HangUp()
exten = _7,1,Dial(H323/${ext...@avaya); Avaya Extension
exten = _7X,1,Dial(H323/${ext...@avaya); Avaya Extension
exten = _5,1,Dial(H323/${ext...@avaya); to call on SIP Extension
exten = _4,1,Dial(H323/${ext...@avaya); Your extension on Avaya
exten = _006,1,Dial(H323/${ext...@avaya); to call on mobile
exten = _00X,1,Dial(H323/${ext...@avaya); to call on National
===H323==
[general]
port = 1720
bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for
this machine
amaflags = AVAYA
progress_setup = 8
progress_alert = 8
faststart=yes
h245tunneling=yes
gatekeeper = DISABLE
;We need to conserve the main parameters to allow the h323 to call to the SIP
phone
disallow=all
allow=ulaw
allow=alaw
dtmfmode=inband
context=internal ; name of your context
[Avaya]
type=friend
context=internal
host=10.200.1.47 ; IP Address of your CLAN
port=1720; port used to connect on CLAN it could be some others port regarding
your configuration in signalling gr$
disallow=all
allow=ulaw ;alaw
allow=alaw
canreinvite=no
dtmfmode=inband
SIP==
[general]
;context=default
context=internal
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
videosupport=yes ; if you want activate video support
canreinvite=no
[59000]
type=friend
secret=1234 ;your password
host=dynamic
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
allow=h263 ; to use a video codec if needed
callerid=Cyril CONSTANTIN
nat=yes
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-849-8087
* Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun
*---*
_
Color coding for safety: Windows Live Hotmail alerts you to suspicious email.
http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_safety_112008
___
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Re: [asterisk-users] Asterisk with Avaya
Recently I had the same problem using H323 with Cisco and I solved it by changing bindaddr = 0.0.0.0 to the IP address of the Asterisk server. You are my HERO! This was the error!!! -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 _ Windows Live Hotmail now works up to 70% faster. http://windowslive.com/Explore/Hotmail?ocid=TXT_TAGLM_WL_hotmail_acq_faster_112008 ___ -- 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] Asterisk with SC440 Dell(Big Problem)
Date: Fri, 31 Oct 2008 11:39:43 +0200 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk with SC440 Dell(Big Problem) Hi On Fri, Oct 31, 2008 at 03:35:23AM +, Edwin Quijada wrote: I have a Dell SC440 with Centos and Asterisk 1.4.21 and a card openvox D110PG, T1, when a person calling from the PTSN will listen to them but then begins to distort the voice I heard that name. This symptom is not clear to me at all. Calls from PSTN to where, exactly? What devices? Directly to Asterisk (voicemail, echo test)? To a SIP device? What about calls that don't go through the card? One person call from PTSN the extension redirect the call to says a few messages just for test. I tested to passing the call to SIP extension I could heard the person fine in the SIP extension but the person heard so much noise like echo but no echo. He heard the voice like a robot. What do you see on zttest -v ? [EMAIL PROTECTED] ~]# zttest -v Opened pseudo zap interface, measuring accuracy... 8192 zaptel samples in 8190.848 system clock sample intervals (99.986%) 8192 zaptel samples in 8190.328 system clock sample intervals (99.980%) 8192 zaptel samples in 8190.705 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.744 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.704 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.736 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.760 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.728 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.728 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.728 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.736 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.647 system clock sample intervals (99.983%) 8192 zaptel samples in 8190.752 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.728 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.735 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.720 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.720 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.720 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.728 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.728 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.728 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.695 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.729 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.745 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.720 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.720 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.728 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.712 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.736 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.712 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.712 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.720 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.744 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.711 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.759 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.721 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.736 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.728 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.736 system clock sample intervals (99.985%) 8192 zaptel samples in 8190.711 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.656 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.672 system clock sample intervals (99.984%) 8192 zaptel samples in 8190.840 system clock sample intervals (99.986%) 8192 zaptel samples in 8190.744 system clock sample intervals (99.985%) --- Results after 44 passes --- Best: 99.986 -- Worst: 99.980 -- Average: 99.984363, Difference: 99.984363 What version of zaptel do you use? 1.4.12.1 I probe the card in another computer and it works perfectly. Anyone has any idea or help. Install Debian on this server and the same thing happened to me. Both Centos and Debian Etch have a kernel based on 2.6.18 . OTOH, latest versions of Etch also include kernels based on 2.6.24 (the etchanahalf kernel, if I spell that correctly). Maybe also try that? I tested with Debian Lenny 2.6.26 kernel and I get the same. This is my kernel now 2.6.9-78.0.5.ELsmp I tried both and nothing! :( -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http
[asterisk-users] Asterisk with SC440 Dell(Big Problem)
I have a Dell SC440 with Centos and Asterisk 1.4.21 and a card openvox D110PG, T1, when a person calling from the PTSN will listen to them but then begins to distort the voice I heard that name. I probe the card in another computer and it works perfectly. Anyone has any idea or help. I'm going crazy with this problem. Install Debian on this server and the same thing happened to me. I bought this server and now it doesnt work with asterisk. I will appreciate if somebody has any cluee or idea about this. If anybody has this server i'd like to know everything about your config. TIA *---* *-Edwin Quijada *-Developer DataBase _ Get Windows Live and get whatever you need, wherever you are. Start here. http://www.windowslive.com/default.html?ocid=TXT_TAGLM_WL_Home_082008 ___ -- 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] Gnudialer runninig
Hi! I wanna know if here somebody has installed gnudialer ? I installed but i dont know how to run it Anybody has a cluee? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Get Windows Live and get whatever you need, wherever you are. Start here. http://www.windowslive.com/default.html?ocid=TXT_TAGLM_WL_Home_082008 ___ -- 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 AGI and php problem....
Chechk permissions *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-849-8087 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* Date: Sat, 16 Aug 2008 13:20:18 -0400 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk AGI and php problem '/var/lib/asterisk/agi-bin/cid-to-acct.php': No such file or directory 2 == cid-to-acct.php: Failed to execute It is not complaining about the lack of /usr/bin/php, but about the fact that the file /var/lib/asterisk/agi-bin/cid-to-acct.php is nowhere to be found. Probably asking the obvious but... Did you place the file in the agi-bin folder ? Is it really named cid-to-acct.php ? Is it executable ? Does the user under which asterisk is running as the right to execute it ? hth ___ -- 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 _ Your PC, mobile phone, and online services work together like never before. http://clk.atdmt.com/MRT/go/108587394/direct/01/ ___ -- 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 cant play sounds from AGI
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Date: Fri, 11 Jul 2008 08:10:38 -0700
Subject: Re: [asterisk-users] Asterisk cant play sounds from AGI
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
Hi! I am a newbie using Asterisk. I am developing an IVR using perl
from AGI and Cepstral as voices
The AGI is this
[snip]
My problem is that i cant hear anything when play the file sound
using $AGI-stream_file($filename);
I put asterisk in verbose mode but just see that it plays the sound
but I cant hear anything.
I thought maybe was the codec but asterisk can play .wav
But this works
$AGI-say_number('9865');
If Asterisk says it is playing the file, then I would suspect the file
itself has nothing to say. Try copying the file to your computer and
playing it. If it does indeed play locally on your computer with
audio, double check to make sure it is in the right format. I use AGI
to play files all the time. Actually, I use an AGI script as my whole
menu and dialing system to replace having to do it in AEL (so much
nicer to add a single MySQL record and suddenly have voicemail and
direct dial work instantly).
Daniel
I tested the files playing in other app, Winamp, and the file play fine.
I tested with other files ,sounds from asterisk, and I get the same thing.
In my spftphone doesnt hear anything
But this works
$AGI-say_number('9865')
so fine.
??
*---*
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*-809-849-8087
* Si deseas lograr cosas excepcionales debes de hacer cosas fuera
de lo comun
*---*
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Re: [asterisk-users] Asterisk cant play sounds from AGI
Date: Fri, 11 Jul 2008 11:29:58 -0700
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk cant play sounds from AGI
On Fri, 11 Jul 2008, Tilghman Lesher wrote:
On Jul 10, 2008, at 7:54 PM, Edwin Quijada wrote:
My problem is that i cant hear anything when play the file sound
using $AGI-stream_file($filename); I put asterisk in verbose mode
but just see that it plays the sound but I cant hear anything.
Check the format of the file. In most cases, the file should be 8000Hz,
single channel, uncompressed, signed linear, 16-bit samples format.
Winamp can play a great many different formats, but Asterisk is limited
to the formats for which it has a translator.
If the file is a wav, it should look something like this:
-t2::sedwards:~$ file example.wav
example.wav: RIFF (little-endian) data, WAVE audio, Microsoft\
PCM, 16 bit, mono 8000 Hz
Also, just in case you trip over this, you pass a file name to Asterisk,
not a file type -- the bit after the period. Asterisk chooses the best
type from files of the same name based on the codecs available to the
channel.
vm-debian#file tts-hello
example.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono
8000 Hz
I recorded the sound using Cepstral. This is my AGI
I thought maybe was my sound card but this works fine
$AGI-say_number('9865');
$AGI-say_digits('873746');
and I can hear it in my SIP phone
use Asterisk::AGI;
use File::Basename;
use Digest::MD5 qw(md5_hex);
$AGI = new Asterisk::AGI;
%input = $AGI-ReadParse();
#
$AGI-say_number('9865');
$AGI-say_digits('873746');
speak(Hello World);
sub speak
{
$text = $_[0];
my $hash = md5_hex($text);
my $ttsdir = /var/lib/asterisk/sounds/tts;
my $cepoptions = -p audio/sampling-rate=8000,audio/channels=1;
my $wavefile = $ttsdir/tts-$hash.wav;
unless (-f $wavefile)
{
open(fileOUT, /var/lib/asterisk/sounds/tts/say-text-$hash.txt);
print fileOUT $text;
close(fileOUT);
my $execf=/opt/swift/bin/swift -f $ttsdir/say-text-$hash.txt -o
$wavefile $cepoptions;
system($execf);
unlink($ttsdir/say-text-$hash.txt);
}
$filename = 'tts/'.basename('tts/'.basename($wavefile,.wav));
$AGI-stream_file($filename);
# unlink($wavefile);
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Asterisk cant play sounds from AGI
Hi! I am a newbie using Asterisk. I am developing an IVR using perl from AGI
and Cepstral as voices
The AGI is this
use Asterisk::AGI;
use File::Basename;
use Digest::MD5 qw(md5_hex);
$AGI = new Asterisk::AGI;
%input = $AGI-ReadParse();
#
$AGI-say_number('9865');
$AGI-say_digits('873746');
speak(Hello World);
sub speak
{
$text = $_[0];
my $hash = md5_hex($text);
my $ttsdir = /var/lib/asterisk/sounds/tts;
my $cepoptions = -p audio/sampling-rate=8000,audio/channels=1;
my $wavefile = $ttsdir/tts-$hash.wav;
unless (-f $wavefile)
{
open(fileOUT, /var/lib/asterisk/sounds/tts/say-text-$hash.txt);
print fileOUT $text;
close(fileOUT);
my $execf=/opt/swift/bin/swift -f $ttsdir/say-text-$hash.txt -o
$wavefile $cepoptions;
system($execf);
unlink($ttsdir/say-text-$hash.txt);
}
$filename = 'tts/'.basename('tts/'.basename($wavefile,.wav));
$AGI-stream_file($filename);
# unlink($wavefile);
This function I took from internet where i found it
My problem is that i cant hear anything when play the file sound using
$AGI-stream_file($filename);
I put asterisk in verbose mode but just see that it plays the sound but I cant
hear anything.
I thought maybe was the codec but asterisk can play .wav
But this works
$AGI-say_number('9865');
Any help or cluees will be so appreciate~!
Thks!
*---*
*-Edwin Quijada
*-Developer DataBase
*-JQ Microsistemas
*-809-849-8087
* Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun
*---*
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[Asterisk-Users] AP200B or C
Hi! I wanna know if somebody knows where I can buy this kind of VoIP phone here USA? TIA *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-747-2787 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Consigue aquí las mejores y mas recientes ofertas de trabajo en América Latina y USA: http://latam.msn.com/empleos/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AP200B Phones
Hi! Somebody knows where can I buy this kind of VoIp Phone? *---* *-Edwin Quijada *-Developer DataBase *-JQ Microsistemas *-809-747-2787 * Si deseas lograr cosas excepcionales debes de hacer cosas fuera de lo comun *---* _ Charla con tus amigos en línea mediante MSN Messenger: http://messenger.latam.msn.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Newbie with new Project VOIp
Hi! I am a newbie in VoIp. Looking for in the net I get this product to work for Linux, now I have a few questions I have a customer that wants implement VoIP using phones VOiP and analog and integrate it into network voice/data. 1-Using * can integrate VOIP phone with analog phone and what that I need? 2-Which VOIP phones Can I use with *? 3-I can call from a VOIP phone to analog phone localy in my company and viceversa, what that I need to do that? 4-* support SIP protocol besides H.323? 5-What about the performance using this? 6-What points I must take a count to use thisn product? 7. 8-If I use * I dont need any hardware to communicate with Phone, except the phone , of course.? I am a newbie in this but I have a few years working with Linux -Any ideas , cluees will be appreciate. TIA Edwin Quijada _ Charla con tus amigos en línea mediante MSN Messenger: http://messenger.latam.msn.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
