[asterisk-users] Default extension
Hello, When I get a SIP INVITE as follows: INVITE sip:s@10.1.0.191:5060 SIP/2.0 Max-Forwards: 69 From: 0475XX sip:1053...@sip.domain.com;tag=as7df9ab18 To: sip:02XX@IP:5060 Contact: sip:1053212@IP:5060 Call-ID: 344d42bd16975a54141d11f635bdf...@sip.domain.com CSeq: 102 INVITE Date: Wed, 26 Mar 2014 15:06:01 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 252 Asterisk considers that the extension is 's'. (The Register) How to make the extension number that is shown in the 'To' ?? Thank you, Mickael -- _ -- 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 sends the INTERNAL IP address of my equipments to my SIP friends?!?
Hello Matthew, My version is Asterisk 1.6.2.9. Or have you seen NAT? I have no NAT on my network. Have you seen my little diagram above? Here it is: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) My Asterisk server has two NIC/interfaces. - 1 interface with public IP (109.69.217.6 to talk with SIP friends) - 1 interface with internal ip (10.4.0.1 to talk with SIP gateway's) SIP friend should not even know that the call is routed to the SIP/PSTN gateway. It could be a SIP trunk to a SIP provider Internet, the user does not have to know... Best regards, Mickael 2013/6/13 Matthew J. Roth mr...@imminc.com Mickael MONSIEUR wrote: I have a standard Asterisk configuration: SIP friends (phones) - Asterisk - SIP gateway to PSTN converter 80.236.215.61109.69.217.6 internal IP ( 10.4.0.10/255.255.255.0 ) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:@80.236.215.61:64946;ob SIP/2.0 Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport Max-Forwards: 70 From: sip:@109.69.217.6 ;tag=as15b47581 To: test sip:@109.69.217.6;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Contact: sip:x@109.69.217.6 Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM CSeq: 102 INVITE User-Agent: Asterisk Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 217 v=0 o=root 664087974 664087976 IN IP4 10.4.0.10 s=Asterisk c=IN IP4 10.4.0.10 t=0 0 m=audio 8652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv My equipement IP 10.4.0.10 is visible to the user, why? Mickael, What version of Asterisk are you running? Is the Asterisk server outside and the SIP gateway to PSTN converter inside of a NAT? What are the NAT SUPPORT and MEDIA HANDLING settings in sip.conf? Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] Asterisk sends the INTERNAL IP address of my equipments to my SIP friends?!?
Good morning, or Good afternoon! It depends :-) I have a standard Asterisk configuration: SIP friends (phones)-Asterisk-SIP gateway to PSTN converter 80.236.215.61 109.69.217.6internal IP ( 10.4.0.10/255.255.255.0) When analyzing traffic on a SIP friend/phone I see this: INVITE sip:@80.236.215.61:64946;ob SIP/2.0 Via: SIP/2.0/UDP 109.69.217.6:5060;branch=z9hG4bK52d50250;rport Max-Forwards: 70 From: sip:@109.69.217.6;tag=as15b47581 To: test sip:@109.69.217.6;tag=kp1VwHD80rA9MVdBjTF4jyFIaCkrJcjh Contact: sip:x@109.69.217.6 Call-ID: MSMhw2bsheHWAQgHlae3O7yKQ2P9EcsM CSeq: 102 INVITE User-Agent: Asterisk Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Type: application/sdp Content-Length: 217 v=0 o=root 664087974 664087976 IN IP4 10.4.0.10 s=Asterisk c=IN IP4 10.4.0.10 t=0 0 m=audio 8652 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv My equipement IP 10.4.0.10 is visible to the user, why? Thank you, Mickael -- _ -- 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] Asterisk 1.6 + Cisco AS5300
Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 Bad Extension back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-004d == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-004d' Do you have an explanation? Best regards, Mickael -- _ -- 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 1.6 + Cisco AS5300
Le 7/03/13 11:21, Steven Howes a écrit : On 7 Mar 2013, at 10:12, Mickael Monsieur wrote: Do you have an explanation? Put a SIP debug on and we may be able to find one.. Steve Hello Steve, After checking, I confirm that the call is cut precisely to 900 seconds (15 min). 10.4.0.1 = Asterisk 10.4.0.10 = Cisco AS 5300 Info : debug start at 14min30sec set_destination: Parsing sip:0032487997160@10.4.0.10:5060 for address/port to send to set_destination: set destination to 10.4.0.10, port 5060 Audio is at 10.4.0.1 port 11842 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 10.4.0.10:54789: INVITE sip:0032487997160@10.4.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport Max-Forwards: 70 From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Contact: sip:65939191@10.4.0.1 Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 INVITE User-Agent: isdnbox1.1 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 207 v=0 o=root 1538728127 1538728127 IN IP4 10.4.0.1 s=Asterisk PBX 1.6.2.9-2+squeeze8 c=IN IP4 10.4.0.1 t=0 0 m=audio 11842 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- --- SIP read from UDP:10.4.0.10:5060 --- SIP/2.0 420 Bad Extension Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 INVITE Unsupported: timer Content-Length: 0 - --- (8 headers 0 lines) --- -- Got SIP response 420 Bad Extension back from 10.4.0.10 set_destination: Parsing sip:0032487997160@10.4.0.10:5060 for address/port to send to set_destination: set destination to 10.4.0.10, port 5060 Transmitting (NAT) to 10.4.0.10:5060: ACK sip:0032487997160@10.4.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport Max-Forwards: 70 From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Contact: sip:65939191@10.4.0.1 Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 ACK User-Agent: isdnbox1.1 Content-Length: 0 --- -- Stopped music on hold on SIP/as5300-1-0050 == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-0050' Reliably Transmitting (NAT) to 10.4.0.10:5060: OPTIONS sip:10.4.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport Max-Forwards: 70 From: asterisk sip:asterisk@10.4.0.1;tag=as4eb3efa7 To: sip:10.4.0.10 Contact: sip:asterisk@10.4.0.1 Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 CSeq: 102 OPTIONS User-Agent: isdnbox1.1 Date: Thu, 07 Mar 2013 11:17:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:10.4.0.10:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport From: asterisk sip:asterisk@10.4.0.1;tag=as4eb3efa7 To: sip:10.4.0.10;tag=37A724C-211C Date: Sat, 01 Jan 2000 16:12:32 GMT Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp CSeq: 102 OPTIONS Supported: 100rel Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Accept: application/sdp Allow-Events: telephone-event Content-Length: 154 v=0 o=CiscoSystemsSIP-GW-UserAgent 793 5073 IN IP4 10.4.0.10 s=SIP Call c=IN IP4 10.4.0.10 t=0 0 m=audio 0 RTP/AVP 18 0 8 4 2 15 3 c=IN IP4 10.4.0.10 - --- (14 headers 7 lines) --- Really destroying SIP dialog '6a43ad4b27d870d048e8425077bcc075@10.4.0.1' Method: OPTIONS --- SIP read from UDP:10.4.0.10:54336 --- BYE sip:65939191@10.4.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.10:5060 From: sip:0032487997160@10.4.0.10;tag=36CA05C-167B To: sip:65939191@10.4.0.1;tag=as12acaefb Date: Sat, 01 Jan 2000 16:12:26 GMT Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 6 Timestamp: 946743153 CSeq: 102 BYE Content-Length: 0 - --- (11 headers 0 lines) --- --- Transmitting (NAT) to 10.4.0.10:54336 --- SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.4.0.10:5060;received=10.4.0.10 From: sip:0032487997160@10.4.0.10;tag=36CA05C-167B To: sip:65939191@10.4.0.1;tag=as12acaefb Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 BYE Server: isdnbox1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 15 min (call ended) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300
Le 7/03/13 11:12, Mickael Monsieur a écrit : Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 Bad Extension back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-004d == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-004d' Do you have an explanation? Best regards, Mickael Ok i solved : https://issues.asterisk.org/jira/browse/ASTERISK-15787 -- _ -- 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] Cisco AS5300 - no incoming sound
Hello, If someone has an example of configuration for Cisco AS5300 / Asterisk, I am very interested. Thank you, Mickael Le 28/12/12 00:48, Mickael MONSIEUR a écrit : Hello, I'm trying to connect a Cisco AS5300 has Asterisk, but I have a problem. Sound from POTS - Asterisk does not work. (In the sense Asterisk - POTS it works!!) The problem lies in two directions (call initiated from the Asterisk or POTS) I have no firewall between Asterisk and Cisco. (it's a LAN) Do you have any ideas? Thank you, Mickael -- _ -- 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] Cisco AS5300 - no incoming sound
Hello, I'm trying to connect a Cisco AS5300 has Asterisk, but I have a problem. Sound from POTS - Asterisk does not work. (In the sense Asterisk - POTS it works!!) The problem lies in two directions (call initiated from the Asterisk or POTS) I have no firewall between Asterisk and Cisco. (it's a LAN) Do you have any ideas? Thank you, Mickael -- _ -- 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 1.6.1 Realtime SIP Users
Thank you I'll watch. Support for Asterisk-Mysql is a bit minimal ... :-( 2011/7/1 Mickael MONSIEUR mickael.monsi...@gmail.com Hello, I just implement the SIP Peers with MySQL. In the structure mySQL missing the following fields: nat = yes notransfer = yes dtmfmode = rfc2833 call-limit = 2 canreinvite = no subscribecontext = blf subscribecontext (BLF) and call-limit (Protection) are very important ... Can you help me? Best, Mickael -- _ -- 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] Asterisk 1.6.1 Realtime SIP Users
Hello, I just implement the SIP Peers with MySQL. In the structure mySQL missing the following fields: nat = yes notransfer = yes dtmfmode = rfc2833 call-limit = 2 canreinvite = no subscribecontext = blf subscribecontext (BLF) and call-limit (Protection) are very important ... Can you help me? Best, Mickael -- _ -- 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] sendrpid does not work!
Hello, I have Asterisk 1.6.2.9-2, the directive sendrpid does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line Remote-Party-ID does not appear in my sip debug! Please help me, Mickael. -- _ -- 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] sendrpid does not work!
Thank you, Andrew. So, with Asterisk 1.6, I have no alternative but to use SIPAddHeader? 2011/1/10 Andrew Latham lath...@gmail.com On Mon, Jan 10, 2011 at 9:58 AM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hello, I have Asterisk 1.6.2.9-2, the directive sendrpid does not work! I placed this in my peer: (sip.conf) sendrpid=yes trustrpid=yes or sendrpid=yes trustrpid=no (and restarted Asterisk) and the line Remote-Party-ID does not appear in my sip debug! Please help me, Mickael. This functionality is supported in Asterisk 1.8. Read more at: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ -- 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] MixMonitor
Hi, After disabling MixMonitor, I realize that my CPU saturates as always! What my script PHP-AGI is fairly simple! - I answer a call - Some menus - I send the call to another line $this-exec_dial (SIP/provider/NUMBER, ...) And I was 75-80% using an e4...@2.40ghz! It is not logic ! Please help ! 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com Hi, marked - noticed. I do not know where it comes from, my CPU goes from 2% to 60-70% at a command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU e4...@2.40ghz 2010/11/5 Norbert Zawodsky norb...@zawodsky.at Am 05.11.2010 10:16, schrieb Mickael MONSIEUR: none ? 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com mailto:mickael.monsi...@gmail.com Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. Obviously, if the box has more to do, CPU load will increase. What do you mean with marked ?? Norbet -- _ -- 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] MixMonitor
You think of a loop? This is possible because I use AGISIGHUP=no .. exten = s,1,set(AGISIGHUP=no); exten = s,2,AGI(myapp.agi) ; I will put lines and debug log file ... I do not think that Asterisk archive errors AGI script? 2010/11/9 Marino Punturieri map...@gmail.com So it seems not related to MixMonitor. Are you 100% sure that your PHP-AGi script is not looping somewhere? You should try to understand which is the process that is taken you CPU. On Tue, Nov 9, 2010 at 2:32 PM, Mickael MONSIEUR mickael.monsi...@gmail.com wrote: Hi, After disabling MixMonitor, I realize that my CPU saturates as always! What my script PHP-AGI is fairly simple! - I answer a call - Some menus - I send the call to another line $this-exec_dial (SIP/provider/NUMBER, ...) And I was 75-80% using an e4...@2.40ghz! It is not logic ! Please help ! 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com Hi, marked - noticed. I do not know where it comes from, my CPU goes from 2% to 60-70% at a command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU e4...@2.40ghz 2010/11/5 Norbert Zawodsky norb...@zawodsky.at Am 05.11.2010 10:16, schrieb Mickael MONSIEUR: none ? 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com mailto:mickael.monsi...@gmail.com Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. Obviously, if the box has more to do, CPU load will increase. What do you mean with marked ?? Norbet -- _ -- 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 -- Se l'è vera che te me voeuret ben cara Ninin biribimpinpin vegn giò a derví el portell famm pú penà, parabappappà se ti te gh'hee l'amor del tò Marcell che l'è inscí bell vegn giò a derví el portell famm pú penà, parabappappà! -- _ -- 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] MixMonitor
none ? 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. -- _ -- 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] MixMonitor
Hi, marked - noticed. I do not know where it comes from, my CPU goes from 2% to 60-70% at a command Dial (sip) + MixMonitor. I have an Intel (R) Core (TM) 2 Duo CPU e4...@2.40ghz 2010/11/5 Norbert Zawodsky norb...@zawodsky.at Am 05.11.2010 10:16, schrieb Mickael MONSIEUR: none ? 2010/11/5 Mickael MONSIEUR mickael.monsi...@gmail.com mailto:mickael.monsi...@gmail.com Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. Obviously, if the box has more to do, CPU load will increase. What do you mean with marked ?? Norbet -- _ -- 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] MixMonitor
Hi, Have you noticed a marked increase in CPU load when using MixMonitor? I use PHPAgi and Asterisk 1.6.2.9-2. Mickael. -- _ -- 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] CDR display in minute
Hello, I want to graphically display the number of calls per minute to an extension. The programs I have found it possible to do so but the average is done on time or day ... I use Mysql CDR Thank you, Mickael -- _ -- 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] CDR display in minute
http://forums.cacti.net/viewtopic.php?p=111317 Thank you. 2010/9/23 Faisal Hanif fai...@vopium.com use CACTI On 9/23/2010 3:10 PM, Mickael MONSIEUR wrote: Hello, I want to graphically display the number of calls per minute to an extension. The programs I have found it possible to do so but the average is done on time or day ... I use Mysql CDR Thank you, Mickael -- _ -- 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] chan_local - Asterisk 1.6.2.6
Hi, My Asterisk is not running on a virtual machine, and Debian does not have an X Server. I have no value with Kernel Timing enabled. Do you think it may be bound for the proper functioning of chan_local? I have no problem with the Dial (SIP/XX), but only with the Dial (Local/XX) :-( Do you have good documentation for the modification of kernel 2.6.x? I have tried in the past but all I had was the kernel panic ... Mickael. 2010/7/20 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de Hi! Nobody uses chan_local Absolutely nobody. Except you. ;- Maybe this will help you: Search for Asterisk timing, consider to not run Asterisk in a virtual environment, and do not run X on the same box. Makre sure to turn off silence suppression in your SIP client(s). Search for choppy audio. Check if earlier Asterisk versions behave better. Philipp -- _ -- 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] chan_local - Asterisk 1.6.2.6
2.6.30-2-686 (Debian) 2010/7/21 Tzafrir Cohen tzafrir.co...@xorcom.com On Wed, Jul 21, 2010 at 10:58:34AM +0200, Mickael Monsieur wrote: Hi, My Asterisk is not running on a virtual machine, and Debian does not have an X Server. I have no value with Kernel Timing enabled. Do you think it may be bound for the proper functioning of chan_local? I have no problem with the Dial (SIP/XX), but only with the Dial (Local/XX) :-( Do you have good documentation for the modification of kernel 2.6.x? I have tried in the past but all I had was the kernel panic ... I got some reports of (Debian Testing/Unstable) systems where the timerfd timing didn't work properly and the workaround was reverting to the pthreads one. I have not yet managed to reproduce them here. I wonder if this is the issue. What kernel do you use? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] chan_local - Asterisk 1.6.2.6
Nobody uses chan_local 2010/7/16 Mickael Monsieur mickael.monsi...@gmail.com Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) - OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extens...@context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... You have No messa ... or You have ... The sound stops but the call continues. Please help! Debian 5.0 - Asterisk 1.6.2.6-1 Mickael. -- _ -- 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] chan_local - Asterisk 1.6.2.6
Hello I just coding a AGI script for billing. - For external calls, I pass the call directly on a trunk. I do : Dial(trunk1/extension) - OK ! - For internal calls (shortcode, others users ...) I am Dial(Local/extens...@context/n) The problem is that through chan_local.so, I sound as it cut! Example if I call the voicemail ... You have No messa ... or You have ... The sound stops but the call continues. Please help! Debian 5.0 - Asterisk 1.6.2.6-1 Mickael. -- _ -- 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] I look ARI (Asterisk Recording Interface)
Hello Bruce, This module is not reliable on FreePBX? You know if there is a open source web-voicemail for Asterisk? Best regards, Mickael. 2010/6/23 bruce bruce bruceb...@gmail.com It's one of the bad modules that goes with FreePBX anyhow. The moment you go over 3000 recordings you are already in trouble. It's about time someone come up with a better moduel. On Wed, Jun 23, 2010 at 11:05 AM, Mickael Monsieur mickael.monsi...@gmail.com wrote: Hello, I look ARI (Asterisk Recording Interface) the publisher site is closed... http://www.littlejohnconsulting.com/ari Thank you, Mickael -- _ -- 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] I look ARI (Asterisk Recording Interface)
Hello, I look ARI (Asterisk Recording Interface) the publisher site is closed... http://www.littlejohnconsulting.com/ari Thank you, Mickael -- _ -- 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] bug with Moh on MeetMe ?
Hello, The MeetMe application refuses MusicOnHold personalized and skip always in the default! Have you any idea how to fix this? -- Executing [028883...@default:1] Set(SIP/109.10.214.1-0002, CHANNEL(language)=fr) in new stack -- Executing [028883...@default:2] Answer(SIP/109.10.214.1-0002, ) in new stack -- Executing [028883...@default:3] Playback(SIP/109.10.214.1-0002, welcome) in new stack -- SIP/109.10.214.1-0002 Playing 'welcome.alaw' (language 'fr') [Jun 13 12:30:00] NOTICE[13437]: channel.c:3012 __ast_read: Dropping incompatible voice frame on SIP/109.10.214.1-0002 of format ulaw since our native format has changed to 0x8 (alaw) -- Executing [028883...@default:4] MeetMeCount(SIP/109.10.214.1-0002, 100,COUNT) in new stack == Parsing '/etc/asterisk/meetme.conf': == Found -- Executing [028883...@default:5] GotoIf(SIP/109.10.214.1-0002, 0?100) in new stack -- Executing [028883...@default:6] MeetMe(SIP/109.10.214.1-0002, 100,1pdM(*personnalised*)) in new stack -- Created MeetMe conference 1023 for conference '100' -- *Started music on hold*, class *'personnalised*', on SIP/109.10.214.1-0002 -- *Stopped music on hold* on SIP/109.10.214.1-0002 -- *Started music on hold, class 'default',* on SIP/109.10.214.1-0002 Thank you, Mickael. -- _ -- 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] contacting
Hello Steve, Thank you for your response. The application ConfBridge is perfect and rapid to setup! The only inconvenience of this application, is that we do not know how to parametrize of key(touch) to go out of the conference, and I absolutely need it! Have you an idea? Thank you. 2010/6/11 Steve Edwards asterisk@sedwards.com On Fri, 11 Jun 2010, Mickael Monsieur wrote: Is it possible to connect two callers without going through a conference (meetme) ? 0) A better subject may attract the interest of someone with relevant experience. Contacting means nothing. 1) More details will yield better responses. What version of Asterisk? 1.6+ has the new confbridge feature that may be of use. I'm a 1.2 Luddite, so I can't tell you more about it. 2) Why is meetme unacceptable? 3) Why is parking unacceptable? -- 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
Re: [asterisk-users] MeetMe
because... I use it! But I do not use MeetMe with! What is the importance of providing binary packets if the conference (MeetMe app) is impossible without compiling ?? 2010/6/12 Tzafrir Cohen tzafrir.co...@xorcom.com On Fri, Jun 11, 2010 at 04:39:46PM +0200, Mickael Monsieur wrote: What is the interest to supply binary of Asterisk, under debian for example, while to use MeetMe it is necessary to COMPILE Asterisk ??? :-)) Mickael. And you don't use the existing DEB package because? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.comjabber%3atzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] MeetMe
What is the interest to supply binary of Asterisk, under debian for example, while to use MeetMe it is necessary to COMPILE Asterisk ??? :-)) Mickael. -- _ -- 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] contacting
Hello, Is it possible to connect two *callers* without going through a conference (meetme) ? Example: 06:50pm - User 1 call extension 600 and musiconhold / parked call .. 06:51pm - User 2 call extension 600 and connect to User 1. Thank you in advance, Mickael. -- _ -- 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] run script after completed
DeadAGI is deprecated in Asterisk 1.6.x ! 2010/4/9 Danny Nicholas da...@debsinc.com Do the call in a context and have the context run the script as a DeadAGI. [call_and_do] - exten = s,1,Dial… - exten = h,1,Deadagi(…) -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Necati Demir *Sent:* Friday, April 09, 2010 7:34 AM *To:* asterisk-users@lists.digium.com *Subject:* [asterisk-users] run script after completed Hello, I am creating a call file with parameter Archive: yes. When it is completed it is moved to directory outgoing_done. It works. Now i want to execute a script when it is completed. Is there a parameter/configuration for this? -- Necati DEMİR http://blog.demir.web.tr http://friendfeed.com/ndemir ndemir ~ demir.web.tr --- -- _ -- 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] play a sound from the callee before putting it in connection.
Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/11, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -- _ -- 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] play a sound from the callee before putting it in connection.
Perfect! Thank you! Dan Journo a écrit : Look at option A(x) on this page:- *A(*/x/*)*: Play an announcement (/x/.gsm) to the called party. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial Dial(SIP/11,mA(soundfile)) *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Mickael MONSIEUR *Sent:* 26 April 2010 11:22 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Cc:* t...@zilok.com *Subject:* [asterisk-users] play a sound from the callee before putting it in connection. Hello ! I want to call a line and play a sound from the callee before putting it in connection with the caller. Is this possible? Example: Dial(SIP/11, m) // Ring or Music... if(call==ANSWERED) Play(announce) // Play 'announce' to the called // To connect caller and called ? Best regards, Mickael. -- _ -- 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 + Dial + stream file ?
Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -- _ -- 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 + Dial + stream file ?
Thank you Godson Zeeshan ! :-) Mickael. Zeeshan Zakaria a écrit : There is a parameter L which you can use in the dial command. More about it you can see on voip-info.org http://voip-info.org, but it'll be something like this: Dial(SIP/223,60,L(11000:1)) The first 11000 means 11 minutes allowed duration of the call and after 10 minutes it'll play message You have one minute. Zeeshan A Zakaria -- Sent from my Android phone with K-9 Mail. On 2010-04-07 9:39 AM, Mickael MONSIEUR mickael.monsi...@gmail.com mailto:mickael.monsi...@gmail.com wrote: Hi all, I am running an AGI script in a command dial, or call a SIP trunk. I want to execute after 10 minutes a voice message (stream file) on the channel to warn the person that the call is about to end. How to do that? Thank you, Mickael. -- _ -- 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
