Re: [asterisk-users] queue log realtime mysql
On 04-11-14 11:50, Ishfaq Malik wrote: On 4 November 2014 10:40, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: Hello, I have 5 Asterisk servers all using mysql realtime to store queue log information. There is 1 out of 5 servers which stores the data in 4 columns : 'data1' -- 'data 5'. All other servers store data in 1 column 'data' with the data seperated by pipe. I see no difference in my configuration of extconfig.conf and logger.conf. Maybe a hidden default value ? Can someone tell me which setting makes the mysql realtime driver store data in 1 column or in seperate columns ? Using Asterisk 1.8.12.2 Kind regards, Jonas. Are you using mysql_realtime or odbc with a mysql back end? Using mysql_realtime, not using odbc. Kind regards, Jonas -- _ -- 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] ${HASH(SIP_CAUSE,channel-name)}
Hello,
I read on the wiki :
Asterisk 1.8 will allow to read SIP response codes in the dialplan via
*${HASH(SIP_CAUSE,channel-name)}*. Additionally make sure you're using
the destination channel, not the source channel.
But when I use this in my dialplan, this 'variable' is empty.
Dialplan :
exten = h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,${CHANNEL})})
exten = h,n,NoOp(sip cause = ${HASH(SIP_CAUSE,CHANNEL)})
CLI :
[Oct 30 14:48:03] -- Executing [h@pbx-routing:5]
NoOp(SIP/SipAT01-0015, sip cause = ) in new stack
[Oct 30 14:48:03] -- Executing [h@pbx-routing:6]
NoOp(SIP/SipAT01-0015, sip cause = ) in new stack
Can anyone tell me how this should be used ?
Kind regards,
Jonas.
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[asterisk-users] dialplan reload context
Hello, is it possible to reload just a context in stead of the whole dialplan ? I see this on the tracker : https://issues.asterisk.org/jira/browse/ASTERISK-19934 But is it possible in some Asterisk version ? Kind regards, Jonas. -- _ -- 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] sdp_crypto_process: Crypto life time unsupported: crypto
Hello, I have added the following to the peer definition : ignorecryptolifetime=yes But still Asterisk tells me : [Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:244 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ikW6yFvdVkSaeTuVO1isTQkdaxOjgQjMEMSGUf+K|2^32 [Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:254 sdp_crypto_process: SRTP crypto offer not acceptable [Oct 9 14:02:34] WARNING[31980]: chan_sip.c:9129 process_sdp: Can't provide secure audio requested in SDP offer What else do I need to configure ? Kind regards, Jonas. -- _ -- 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] sdp_crypto_process: Crypto life time unsupported: crypto
On 09-10-14 14:11, Joshua Colp wrote: Jonas Kellens wrote: Hello, Kia ora, I have added the following to the peer definition : ignorecryptolifetime=yes This is not an option within the official tree so unless you've added a patch this won't actually do anything. But still Asterisk tells me : [Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:244 sdp_crypto_process: Crypto life time unsupported: crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ikW6yFvdVkSaeTuVO1isTQkdaxOjgQjMEMSGUf+K|2^32 [Oct 9 14:02:34] NOTICE[31980]: sip/sdp_crypto.c:254 sdp_crypto_process: SRTP crypto offer not acceptable [Oct 9 14:02:34] WARNING[31980]: chan_sip.c:9129 process_sdp: Can't provide secure audio requested in SDP offer What else do I need to configure ? Currently there is no way to turn this off without modifying the source code. I expect this to change in the future based on testing we did at SIPit and stuff we learned. Hello, any idea where and what to change in the source code then ? I am able to change the source code, but to do minimal damage I would like to know where to change what exactly. Using asterisk 1.8.12 Kind regards, Jonas. -- _ -- 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] sdp_crypto_process: Crypto life time unsupported: crypto
On 09-10-14 14:28, Joshua Colp wrote: Jonas Kellens wrote: Hello, any idea where and what to change in the source code then ? I am able to change the source code, but to do minimal damage I would like to know where to change what exactly. Yes. In channels/sip/sdp_crypto.c where the line: ast_log(LOG_NOTICE, Crypto life time unsupported: %s\n, attr); is remove the: continue; Afterwards. Ok this seems to work ! Thanks. Does Asterisk now ignore the SRTP crypto offer ? Or does it just ignore the lifetime (in this case : |2^32) ? It does not seem right that Asterisk now should ignore the whole crypto offer. Kind regards, Jonas. -- _ -- 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] Grandstream GXP2160 + SRTP
On 07-10-14 12:32, Jonas Kellens wrote: Hello, I am trying to setup a Grandstream GXP2160 IP-phone with secure calling (SRTP). Secure signaling SSIP for registration is working great ! I follow this guide : https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial But when I try to make a call with SRTP, I get stuck. There is an initial INVITE which is anwered with a 401. There should follow a new INVITE with a nonce, but this does not happen. Any idea why ? Is it the Grandstream IP-phone ?? --- SIP read from TLS:my.pub.lic.ip:53416 --- INVITE sip:0123123...@ast.ser.ver.ip:5061 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018 To: sip:0123123...@ast.ser.ver.ip:5061 Call-ID: 1695864968-506...@bjc.bgi.b.bae CSeq: 50 INVITE Contact: sips:testacc77005@192.168.1.104:5068;transport=tls X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.2.9 Privacy: none P-Preferred-Identity: sip:testacc77...@ast.ser.ver.ip:5061 Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 522 v=0 o=testacc77005 8004 8000 IN IP4 192.168.1.104 s=SIP Call c=IN IP4 192.168.1.104 t=0 0 m=audio 5020 RTP/SAVP 0 8 18 9 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8m7ZfG+0t3KBFGK40IfDO11SZ6D54glKKIwdgo00|2^32 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:nn+id/sSK7OErMfnZZduKNPLejpscxx1vUQB2seO|2^32 --- Reliably Transmitting (NAT) to my.pub.lic.ip:53416 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;alias;received=my.pub.lic.ip;rport=53416 From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018 To: sip:0123123...@ast.ser.ver.ip:5061;tag=as1e527556 Call-ID: 1695864968-506...@bjc.bgi.b.bae CSeq: 50 INVITE Server: mydomain Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=mydomain.be, nonce=13b47342 Content-Length: 0 --- SIP read from TLS:my.pub.lic.ip:53416 --- ACK sip:0123123...@ast.ser.ver.ip:5061 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018 To: sip:0123123...@ast.ser.ver.ip:5061;tag=as1e527556 Call-ID: 1695864968-506...@bjc.bgi.b.bae CSeq: 50 ACK Content-Length: 0 Hello, I seem to have the same problem with Snom 370 IP-phone. Registration works fine ! But I can not make calls with encrypted rtp. --- SIP read from TLS:my.pub.lic.ip:1068 --- INVITE sip:0123123...@ast.ser.ver.ip;user=phone SIP/2.0 Via: SIP/2.0/TLS 192.168.1.107:1068;branch=z9hG4bK-gxm8w1q7l2co;rport From: sip:testacc77...@ast.ser.ver.ip;tag=zdwiwg10qx To: sip:0123123...@ast.ser.ver.ip;user=phone Call-ID: 3c2679977b67-9j0euqvseh5v CSeq: 1 INVITE Max-Forwards: 70 Contact: sip:testacc77003@192.168.1.107:1068;transport=tls;reg-id=1 X-Serialnumber: 0004132E2809 P-Key-Flags: resolution=31x13, keys=4 User-Agent: snom370/8.4.35 Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE Allow-Events: talk, hold, refer, call-info Supported: timer, 100rel, replaces, from-change Call-Info: sip:ast.ser.ver.ip;appearance-index=1 Session-Expires: 3600;refresher=uas Min-SE: 90 Content-Type: application/sdp Content-Length: 632 v=0 o=root 1052895538 1052895538 IN IP4 192.168.1.107 s=call c=IN IP4 192.168.1.107 t=0 0 m=audio 65418 RTP/SAVP 8 3 18 99 101 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:KiXn5H+mKwavoDNa1PfnBqPoODTnxK6hOlWSNJM7 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv m=audio 65418 RTP/AVP 8 3 18 99 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:99 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv - --- Reliably Transmitting (NAT) to my.pub.lic.ip:1068 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.1.107:1068;branch=z9hG4bK-gxm8w1q7l2co;received=my.pub.lic.ip;rport=1068 From: sip:testacc77...@ast.ser.ver.ip;tag=zdwiwg10qx To: sip:0123123...@ast.ser.ver.ip;user=phone;tag=as1cd819c5 Call-ID: 3c2679977b67-9j0euqvseh5v CSeq: 1 INVITE Server: mydomain Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=mydomain.be, nonce=323823f6 Content-Length: 0 --- SIP read from TLS:my.pub.lic.ip:1068 --- ACK sip:0123123
[asterisk-users] Grandstream GXP2160 + SRTP
Hello, I am trying to setup a Grandstream GXP2160 IP-phone with secure calling (SRTP). Secure signaling SSIP for registration is working great ! I follow this guide : https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial But when I try to make a call with SRTP, I get stuck. There is an initial INVITE which is anwered with a 401. There should follow a new INVITE with a nonce, but this does not happen. Any idea why ? Is it the Grandstream IP-phone ?? --- SIP read from TLS:my.pub.lic.ip:53416 --- INVITE sip:0123123...@ast.ser.ver.ip:5061 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018 To: sip:0123123...@ast.ser.ver.ip:5061 Call-ID: 1695864968-506...@bjc.bgi.b.bae CSeq: 50 INVITE Contact: sips:testacc77005@192.168.1.104:5068;transport=tls X-Grandstream-PBX: true Max-Forwards: 70 User-Agent: Grandstream GXP2160 1.0.2.9 Privacy: none P-Preferred-Identity: sip:testacc77...@ast.ser.ver.ip:5061 Supported: replaces, path, timer Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE Content-Type: application/sdp Accept: application/sdp, application/dtmf-relay Content-Length: 522 v=0 o=testacc77005 8004 8000 IN IP4 192.168.1.104 s=SIP Call c=IN IP4 192.168.1.104 t=0 0 m=audio 5020 RTP/SAVP 0 8 18 9 2 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=ptime:20 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:9 G722/8000 a=rtpmap:2 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:8m7ZfG+0t3KBFGK40IfDO11SZ6D54glKKIwdgo00|2^32 a=crypto:2 AES_CM_128_HMAC_SHA1_32 inline:nn+id/sSK7OErMfnZZduKNPLejpscxx1vUQB2seO|2^32 --- Reliably Transmitting (NAT) to my.pub.lic.ip:53416 --- SIP/2.0 401 Unauthorized Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;alias;received=my.pub.lic.ip;rport=53416 From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018 To: sip:0123123...@ast.ser.ver.ip:5061;tag=as1e527556 Call-ID: 1695864968-506...@bjc.bgi.b.bae CSeq: 50 INVITE Server: mydomain Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=mydomain.be, nonce=13b47342 Content-Length: 0 --- SIP read from TLS:my.pub.lic.ip:53416 --- ACK sip:0123123...@ast.ser.ver.ip:5061 SIP/2.0 Via: SIP/2.0/TLS 192.168.1.104:5068;branch=z9hG4bK60724585;rport;alias From: sip:testacc77...@ast.ser.ver.ip:5061;tag=263162018 To: sip:0123123...@ast.ser.ver.ip:5061;tag=as1e527556 Call-ID: 1695864968-506...@bjc.bgi.b.bae CSeq: 50 ACK Content-Length: 0 -- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
Hello, I have a situation where a call comes in to my Asterisk server B. This call comes from another Asterisk server A. I want to tell to this server A why my server B hangs up. So just before hanging up, I add a custom SIP-header : exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan) exten = s,n,Hangup() But I notice that this extra SIP-header is not send within the SIP-reponse : SIP/2.0 603 Declined Via: SIP/2.0/UDP xx.xx.xx.98:5060;branch=z9hG4bK168884d7;received=xx.xx.xx.98;rport=5060 From: 5006 sip:5...@xx.xx.xx.98;tag=as50c98b4c To: sip:0...@xx.xx.xx.238;tag=as3c6e57b0 Call-ID: 6d1039bb22716c6e6dec69fb3e78a...@xx.xx.xx.98:5060 CSeq: 102 INVITE Server: myasterisk Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 How can I make this work ? Thanks. Jonas. -- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
On 02-09-14 11:34, Steven Howes wrote: On 2 Sep 2014, at 09:03, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: So just before hanging up, I add a custom SIP-header : exten = s,n,SIPAddHeader(X-My-Hangup: MaxChan) exten = s,n,Hangup() SIPAddHeader only works for INVITE as far as I know. Steve OK. Then how can I let another Asterisk server know the custom reason of hangup ? If it is not possible with custom SIP-header, then how ? Regards, Jonas. -- _ -- 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] Custom SIP-header not present in call Asterisk to Asterisk
On 02-09-14 14:22, Eric Wieling wrote: Try Hangup(123) where 123 is whatever hangup cause you want to send back to the caller. The calliing Asterisk server will get the valuse back in HANGUPCAUSE variable. Hello, I have tried sending Hangup(321) on Asterisk server B to Asterisk A but when I read HangupCause on Asterisk A it always is '21'. Good idea, but it does not seem to work. Kind regards, Jonas. -- _ -- 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] NotifyCID to see who is calling for call pickup
Hello, If the phone of my colleague rings, I can see this with BLF-lamps on my Snom IP-phone. I would also like to see *_who_* is calling. I would like to see the external number on my screen so I can choose whether to pickup the call with BLF. Therefore I have in sip.conf : notifycid = yes With this setting on, I see on my screen : 10 -- 10 10 is the internal extension of my colleague. But how can I see the external number that is calling in ? I would expect to see : 10 -- 3221234567 3221234567 being the external number I would like to know about. This is what Asterisk sends to my Snom IP-phone : [Aug 11 16:37:56] Reliably Transmitting (NAT) to my.pub.lic.ip:1024: NOTIFY sip:testacc77003@192.168.1.109:1024 SIP/2.0 Via: SIP/2.0/UDP ip.ast.ser.ver:5060;branch=z9hG4bK5b999cd4;rport Max-Forwards: 70 From: sip:1...@ip.ast.ser.ver;user=phone;tag=as6b302bda To: sip:testacc77...@ip.ast.ser.ver;tag=ashydm1he5 Contact: sip:1...@ip.ast.ser.ver:5060 Call-ID: 3c26b7878939-ri1v0tkqfa2h CSeq: 103 NOTIFY User-Agent: pbx Subscription-State: active Event: dialog Content-Type: application/dialog-info+xml Content-Length: 516 ?xml version=1.0? dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=1 state=full entity=sip:1...@ip.ast.ser.ver dialog id=10 call-id=pickup-3c26b7878939-ri1v0tkqfa2h local-tag=ashydm1he5 remote-tag=as6b302bda direction=recipient remote identity display=10sip:1...@ip.ast.ser.ver/identity target uri=sip:1...@ip.ast.ser.ver/ /remote local identitysip:1...@ip.ast.ser.ver/identity target uri=sip:1...@ip.ast.ser.ver/ /local stateearly/state /dialog /dialog-info Where does Asterisk take the information to put into the dialog-info and how can I change it to display the external number also ? Thanks ! Jonas. -- _ -- 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] Subscription-State always active ?
Hello, I notice that Asterisk always sends Subscription-State: active, even when the SIP-peer is offline (IP-phone cut from power) : /[Jul 31 11:56:58] NOTICE[32273]: chan_sip.c:26194 sip_poke_noanswer: Peer 'testacc77000' is now UNREACHABLE! Last qualify: 49// //[Jul 31 11:56:58] Really destroying SIP dialog '78b0d1701d3694b1494a0c4b55344d57@ip-sip-server:5060' Method: OPTIONS// //[Jul 31 11:56:58] set_destination: Parsing sip:testacc77003@192.168.1.109:1024 for address/port to send to// //[Jul 31 11:56:58] set_destination: set destination to 192.168.1.109:1024// //[Jul 31 11:56:58] Reliably Transmitting (NAT) to my-public-ip:1024:// //NOTIFY sip:testacc77003@192.168.1.109:1024 SIP/2.0// //Via: SIP/2.0/UDP ip-sip-server:5060;branch=z9hG4bK3afa3dd6;rport// //Max-Forwards: 70// //From: sip:10@ip-sip-server;user=phone;tag=as00df4bee// //To: sip:testacc77003@ip-sip-server;tag=9wdraz254n// //Contact: sip:10@ip-sip-server:5060// //Call-ID: 3c267066aeb1-bv3r703hb93x// //CSeq: 109 NOTIFY// //User-Agent: myasterisk// //Subscription-State: active// //Event: dialog// //Content-Type: application/dialog-info+xml// //Content-Length: 202// // //?xml version=1.0?// //dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=7 state=full entity=sip:10@ip-sip-server// //dialog id=10// //stateterminated/state// ///dialog// ///dialog-info/ It seems to me that this information is not correct. Is this some setting I need to tweak ? Kind regards, Jonas. -- _ -- 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] Subscription-State always active ?
On 31-07-14 12:13, Joshua Colp wrote: Jonas Kellens wrote: I notice that Asterisk always sends Subscription-State: active, even when the SIP-peer is offline (IP-phone cut from power) : /[Jul 31 11:56:58] NOTICE[32273]: chan_sip.c:26194 sip_poke_noanswer: Peer 'testacc77000' is now UNREACHABLE! Last qualify: 49// //[Jul 31 11:56:58] Really destroying SIP dialog '78b0d1701d3694b1494a0c4b55344d57@ip-sip-server:5060' Method: OPTIONS// //[Jul 31 11:56:58] set_destination: Parsing sip:testacc77003@192.168.1.109:1024 for address/port to send to// //[Jul 31 11:56:58] set_destination: set destination to 192.168.1.109:1024// //[Jul 31 11:56:58] Reliably Transmitting (NAT) to my-public-ip:1024:// //NOTIFY sip:testacc77003@192.168.1.109:1024 SIP/2.0// //Via: SIP/2.0/UDP ip-sip-server:5060;branch=z9hG4bK3afa3dd6;rport// //Max-Forwards: 70// //From: sip:10@ip-sip-server;user=phone;tag=as00df4bee// //To: sip:testacc77003@ip-sip-server;tag=9wdraz254n// //Contact: sip:10@ip-sip-server:5060// //Call-ID: 3c267066aeb1-bv3r703hb93x// //CSeq: 109 NOTIFY// //User-Agent: myasterisk// //Subscription-State: active// //Event: dialog// //Content-Type: application/dialog-info+xml// //Content-Length: 202// // //?xml version=1.0?// //dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=7 state=full entity=sip:10@ip-sip-server// //dialog id=10// //stateterminated/state// ///dialog// ///dialog-info/ It seems to me that this information is not correct. Is this some setting I need to tweak ? It is correct. The subscription setup by the subscribing device remains active. The body itself (the XML) conveys the information about the device that has been subscribed to. In the case of the above the stateterminated/state means that it is unavailable. There's a few different body types that implementations use. Hello, I read on Yealink support that Yealink IP-phones expect Subscription-State:terminated for there Presence/BLF-functionality. So how can I get Subscription-State:terminated on Asterisk ? Kind regards, Jonas. -- _ -- 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] Subscription-State always active ?
On 31-07-14 14:28, Joshua Colp wrote: Jonas Kellens wrote: Hello, I read on Yealink support that Yealink IP-phones expect Subscription-State:terminated for there Presence/BLF-functionality. So how can I get Subscription-State:terminated on Asterisk ? That would be a bit strange as the subscription would then be terminated, no more NOTIFY messages would go to it which would defeat the purpose of subscribing to something. The only way to achieve that would be to have the phone unsubscribe or to change the code to force it to terminate the subscription under certain circumstances. This would require knowing the exact specifications and details of what they expect and when. Is there currently a problem you are facing with subscription support? Hello, this Subscription-State:terminated is expected when the IP-phone goes offline (Unregister or cut off from power). At that moment indeed the IP-phone no longer sends NOTIFY messages. Also, Asterisk knows very well the SIP peer becomes unreachable (see my first post). But still Asterisk replies Subscription-State: active to the IP-phones that request the state of the offline SIP peer. Yealink expects Subscription-State:terminated so the Yealink IP-phone can put out the BLF light (in stead of staying in a green mode, which indicates that the SIP peer is still online but not in a call). So I can follow the Yealink logic. Can you ? Kind regards, Jonas. -- _ -- 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] Subscription-State always active ?
On 31-07-14 15:06, Joshua Colp wrote: Jonas Kellens wrote: On 31-07-14 14:28, Joshua Colp wrote: Jonas Kellens wrote: Hello, I read on Yealink support that Yealink IP-phones expect Subscription-State:terminated for there Presence/BLF-functionality. So how can I get Subscription-State:terminated on Asterisk ? That would be a bit strange as the subscription would then be terminated, no more NOTIFY messages would go to it which would defeat the purpose of subscribing to something. The only way to achieve that would be to have the phone unsubscribe or to change the code to force it to terminate the subscription under certain circumstances. This would require knowing the exact specifications and details of what they expect and when. Is there currently a problem you are facing with subscription support? Hello, this Subscription-State:terminated is expected when the IP-phone goes offline (Unregister or cut off from power). I would expect that if the phone that went offline was subscribed to stuff and the subscription expired. At that moment indeed the IP-phone no longer sends NOTIFY messages. Also, Asterisk knows very well the SIP peer becomes unreachable (see my first post). But still Asterisk replies Subscription-State: active to the IP-phones that request the state of the offline SIP peer. Generally IP phones don't send NOTIFY messages. They are sent NOTIFY messages to inform them of the state of things they have subscribed to (such as devices or mailboxes). Yealink expects Subscription-State:terminated so the Yealink IP-phone can put out the BLF light (in stead of staying in a green mode, which indicates that the SIP peer is still online but not in a call). Sending Subscription-State:terminated terminates the subscription. If the device in question comes back online you can't send any new NOTIFY messages because the subscription is gone. The state of what you are subscribed to and the underlying state of the subscription itself are different things. So I can follow the Yealink logic. Can you ? Not really as it doesn't make sense to me. Do you have a link to the documentation for this? I've also done a search on the issue tracker and there have been no issues filed ever about subscriptions and specifically Yealink. Hello, I was reading this post : http://forum.yealink.com/forum/showthread.php?tid=894 Jonas. -- _ -- 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] Subscription-State always active ?
On 31-07-14 16:14, Joshua Colp wrote: Jonas Kellens wrote: Hello, I was reading this post : http://forum.yealink.com/forum/showthread.php?tid=894 http://forum.yealink.com/forum/showthread.php?tid=894pid=4794#pid4794 Has the explanation. Since they are using dialog-info+xml there's nothing different between not in use and unavailable. Hello, the explanation is that is does not work with Asterisk ? I don't understand. Asterisk sends dialog-info+xml, right ?! You can see it in my first post : /?xml version=1.0?// //dialog-info xmlns=urn:ietf:params:xml:ns:dialog-info version=7 state=full entity=sip:10@ip-sip-server// //dialog id=10// //stateterminated/state// ///dialog// ///dialog-info/ Kind regards, Jonas. -- _ -- 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] Dynamic Call parking
Hello, I know now after some testing that there is no dynamic call parking. Also explains why you find no example when searching the internet : no one has a working example. I have now the following working case : features.conf : [general] parkeddynamic = yes [parkinglot_77] findslot = first dialplan : [Jul 3 13:47:50] -- Executing [891@from-77:4] NoOp(SIP/SipT01-, ) in new stack [Jul 3 13:47:50] -- Executing [891@from-77:5] Set(SIP/SipT01-, PARKINGDYNEXTEN=891) in new stack [Jul 3 13:47:50] -- Executing [891@from-77:6] Set(SIP/SipT01-, PARKINGDYNPOS=890-891) in new stack [Jul 3 13:47:50] -- Executing [891@from-77:7] Set(SIP/SipT01-, PARKINGDYNCONTEXT=parked_77) in new stack [Jul 3 13:47:50] -- Executing [891@from-77:8] Park(SIP/SipT01-, 4,parkinglot_77) in new stack [Jul 3 13:47:50] -- Registered extension context 'parked_77'; registrar: features [Jul 3 13:47:50] -- Added extension '891' priority 1 to parked_77 [Jul 3 13:47:50] -- Added extension '890' priority -1 to parked_77 [Jul 3 13:47:50] -- Added extension '891' priority -1 to parked_77 [Jul 3 13:47:50] == Parked SIP/SipT01- on 890 (lot parkinglot_77). Will timeout back to extension [from-77] s, 1 in 40 seconds [Jul 3 13:47:50] -- Added extension '890' priority 1 to parked_77 Remarks : The park position (890) is not announced, so you have no idea. PARKINGDYNEXTEN does nothing, could not find out what it is for. PARKINGDYNPOS creates parking positions, but if you change this in the dialplan, there is no dynamical change in the parkinglot. You need to restart Asterisk for changes to take effect. PARKINGDYNCONTEXT dynamically creates a context for hints, but don't understand fully what else it is for. The function Park() when using it without extra parameters always seems to park in the default (defining PARKINGDYNAMIC, PARKINGEXTEN, PARKINGDYNEXTEN, PARKINGDYNPOS, PARKINGDYNCONTEXT changes nothing) When you use the function Park() with the parking lot parameter (here : parkinglot_77) then this context for parking calls is used ! So this works (as you can see in the dialplan) For every change you make in the dialplan (say you change PARKINGDYNCONTEXT=parked_707070) , you need to restart Asterisk to take effect. Unless you restart Asterisk, calls stay in the context parked_707070. A simple 'reload' makes no changes. Nothing dynamic here. Hints : Hints are automatically created if you use 'parkinghints = yes' but you need to issue a 'dialplan reload' AFTER the first call is parked because before a call has been parked, the context is not created inside the dialplan. So if you include for example the context [parked_77] you will get an error when you issue a 'dialplan reload'' because you try to include a context that does not exist (it will exist once you have at least parked 1 call) My conclusion : Call Parking still remains very static. Creating call parking inside the dialplan is not possible, you still need to use features.conf and restart Asterisk after every change. Kind regards, Jonas. On 03-07-14 03:05, Richard Mudgett wrote: On Wed, Jul 2, 2014 at 4:39 AM, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: Hello, I am trying to create a dynamic call parking lot using https://wiki.asterisk.org/wiki/display/AST/Application_Park But this manual is not enough to fix my problem : Asterisk keeps trying to park the call in the default parking lot : [Jul 2 11:32:14] -- Executing [@from-77:5] Set(SIP/testacc77000-0002, PARKINGDYNAMIC=parkinglot_test) in new stack [Jul 2 11:32:14] -- Executing [@from-77:6] Set(SIP/testacc77000-0002, PARKINGEXTEN=3300) in new stack [Jul 2 11:32:14] -- Executing [@from-77:7] Set(SIP/testacc77000-0002, PARKINGDYNEXTEN=110) in new stack [Jul 2 11:32:14] -- Executing [@from-77:8] Set(SIP/testacc77000-0002, PARKINGDYNPOS=111-120) in new stack [Jul 2 11:32:14] -- Executing [@from-77:9] Set(SIP/testacc77000-0002, PARKINGDYNCONTEXT=contextfromtestpark) in new stack [Jul 2 11:32:14] -- Executing [@from-77:10] Park(SIP/testacc77000-0002, ) in new stack [Jul 2 11:32:14] WARNING[28618]: features.c:1291 park_space_reserve: PARKINGEXTEN=3300 is not in default (701-750). [Jul 2 11:32:14] -- SIP/testacc77000-0002 Playing 'pbx-parkingfailed.alaw' (language 'nl') I have the following in features.conf : [parkinglot_test] context = testparkinglot findslot = next This parking lot is invalid because it does not have any defined parking spaces. As a result it does not exist. Check you log for error messages when Asterisk loaded. Alternately, you
[asterisk-users] Dynamic Call parking
Hello, I am trying to create a dynamic call parking lot using https://wiki.asterisk.org/wiki/display/AST/Application_Park But this manual is not enough to fix my problem : Asterisk keeps trying to park the call in the default parking lot : [Jul 2 11:32:14] -- Executing [@from-77:5] Set(SIP/testacc77000-0002, PARKINGDYNAMIC=parkinglot_test) in new stack [Jul 2 11:32:14] -- Executing [@from-77:6] Set(SIP/testacc77000-0002, PARKINGEXTEN=3300) in new stack [Jul 2 11:32:14] -- Executing [@from-77:7] Set(SIP/testacc77000-0002, PARKINGDYNEXTEN=110) in new stack [Jul 2 11:32:14] -- Executing [@from-77:8] Set(SIP/testacc77000-0002, PARKINGDYNPOS=111-120) in new stack [Jul 2 11:32:14] -- Executing [@from-77:9] Set(SIP/testacc77000-0002, PARKINGDYNCONTEXT=contextfromtestpark) in new stack [Jul 2 11:32:14] -- Executing [@from-77:10] Park(SIP/testacc77000-0002, ) in new stack [Jul 2 11:32:14] WARNING[28618]: features.c:1291 park_space_reserve: PARKINGEXTEN=3300 is not in default (701-750). [Jul 2 11:32:14] -- SIP/testacc77000-0002 Playing 'pbx-parkingfailed.alaw' (language 'nl') I have the following in features.conf : [parkinglot_test] context = testparkinglot findslot = next I read that [parkinglot_test] will be used as a template to create the dynamic call park. So all necessary parameters are given inside the dialplan (PARKINGDYNAMIC, PARKINGEXTEN, PARKINGDYNEXTEN, PARKINGDYNPOS, PARKINGDYNCONTEXT). So why does Asterisk always uses the default ? I clearly create the dynamic call park inside the dialplan. Kind regards, Jonas. -- _ -- 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 announcement only once in a Call Queue after 10 seconds
Hello, how can I create the following scenario : I have a Call Queue and I want to play an announcement, but only once after about 10 seconds. The current option |periodic| |-| |announce| |-| |frequency| keeps on playing the announcement indefinitely. (it should have an option 'once' like the option |announce-holdtime|) Kind regards, Jonas. -- _ -- 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] Changing to the linear strategy currently requires asterisk to be restarted
Hello, using asterisk 1.8.12.2 and realtime architecture with mysql. I get the following message on CLI when changing the value in the strategy /[Mar 26 11:02:24] WARNING[10648]: app_queue.c:2030 queue_set_param: Changing to the linear strategy currently requires asterisk to be restarted.// //[Mar 26 11:02:24] WARNING[10648]: app_queue.c:2030 queue_set_param: Changing to the linear strategy currently requires asterisk to be restarted./ Can this be done without restarting asterisk ? Is this also the case in higher Asterisk versions ? For example Asterisk 1.8.24 ? Kind regards, Jonas. -- _ -- 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] php script in h context makes channel hang : solution ?
Hello,
I execute the following php script when a call ends and the h-context is
executed :
/exten = h,n,System(/usr/bin/php
/var/log/asterisk/loggingAST/loggingAST.php
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)/
The script loggingAST.php writes some information in a MySQL database on
a remote webserver.
I have noticed that when the webserver is unreachable, this channel
hangs and Asterisk can not clear the channel and rtp ports.
Is there a way to have the channel cleared, even if it takes some time
to execute the php script ??
Kind regards,
Jonas.
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Re: [asterisk-users] Realtime Call Queues : call members in certain order
On 13-02-14 17:33, Steven Wheeler wrote: *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Thursday, February 13, 2014 7:12 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Realtime Call Queues : call members in certain order On 12-02-14 16:58, Steven Wheeler wrote: *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, February 12, 2014 3:46 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Realtime Call Queues : call members in certain order Hello, I'm using MySQL realtime Call Queues (table /queues/ and table /queue_members/). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy /lineair /and I put the members into the table /queue_members/ in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout | monitor_type | monitor_format | queue_youarenext | queue_thereare | queue_callswaiting | queue_holdtime | queue_minutes | queue_seconds | queue_lessthan | queue_thankyou | queue_reporthold | announce_frequency | announce_round_seconds | announce_holdtime | announce_position | retry | wrapuptime | maxlen | servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus | eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart | periodic_announce | periodic_announce_frequency | ringinuse | +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ | queue6 | default| NULL | | 12 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | 30 | NULL | No | yes | 5 | 10 | 0 | NULL | linear | strict| strict | NULL | NULL| NULL |NULL | NULL | no | | 0 | no| +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ and queue members : +--++++-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++++-++ | 44 | queuemem4 | queue6 | SIP/queuemem4 | 0 | NULL | | 45 | queuemem2 | queue6 | SIP/queuemem2 | 0 | NULL | | 46 | queuemem5 | queue6 | SIP/queuemem5 | 0 | NULL | | 47 | queuemem1 | queue6 | SIP/queuemem1 | 0 | NULL | | 48 | queuemem10 | queue6 | SIP/queuemem10 | 0 | NULL | | 49 | queuemem18 | queue6 | SIP/queuemem18 | 0 | NULL | | 50 | queuemem17 | queue6 | SIP/queuemem17 | 0 | NULL | | 51 | queuemem12 | queue6 | SIP/queuemem12 | 0 | NULL | | 52 | queuemem16 | queue6 | SIP/queuemem16 | 0 | NULL | | 53 | queuemem13 | queue6 | SIP/queuemem13 | 0 | NULL | +--++++-++ You can see that the member /queuemem4/ is first in line to be rang (has the first and lowest uniqueid in the table). But the first member that is being rang, is /queuemem1/. How come ?? Kind regards, Jonas. Jonas, We encountered the same problem. It is a bug in the Queue application. The Queue application actually orders
Re: [asterisk-users] Realtime Call Queues : call members in certain order
On 12-02-14 16:58, Steven Wheeler wrote: *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, February 12, 2014 3:46 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Realtime Call Queues : call members in certain order Hello, I'm using MySQL realtime Call Queues (table /queues/ and table /queue_members/). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy /lineair /and I put the members into the table /queue_members/ in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout | monitor_type | monitor_format | queue_youarenext | queue_thereare | queue_callswaiting | queue_holdtime | queue_minutes | queue_seconds | queue_lessthan | queue_thankyou | queue_reporthold | announce_frequency | announce_round_seconds | announce_holdtime | announce_position | retry | wrapuptime | maxlen | servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus | eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart | periodic_announce | periodic_announce_frequency | ringinuse | +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ | queue6 | default| NULL | | 12 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | 30 | NULL | No| yes | 5 | 10 | 0 | NULL | linear | strict| strict | NULL | NULL| NULL |NULL | NULL | no | | 0 | no| +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ and queue members : +--++++-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++++-++ | 44 | queuemem4 | queue6 | SIP/queuemem4 | 0 | NULL | | 45 | queuemem2 | queue6 | SIP/queuemem2 | 0 | NULL | | 46 | queuemem5 | queue6 | SIP/queuemem5 | 0 | NULL | | 47 | queuemem1 | queue6 | SIP/queuemem1 | 0 | NULL | | 48 | queuemem10 | queue6 | SIP/queuemem10 | 0 | NULL | | 49 | queuemem18 | queue6 | SIP/queuemem18 | 0 | NULL | | 50 | queuemem17 | queue6 | SIP/queuemem17 | 0 | NULL | | 51 | queuemem12 | queue6 | SIP/queuemem12 | 0 | NULL | | 52 | queuemem16 | queue6 | SIP/queuemem16 | 0 | NULL | | 53 | queuemem13 | queue6 | SIP/queuemem13 | 0 | NULL | +--++++-++ You can see that the member /queuemem4/ is first in line to be rang (has the first and lowest uniqueid in the table). But the first member that is being rang, is /queuemem1/. How come ?? Kind regards, Jonas. Jonas, We encountered the same problem. It is a bug in the Queue application. The Queue application actually orders members by their interface value. Here is the bug report I opened https://issues.asterisk.org/jira/browse/ASTERISK-18480 https://issues.asterisk.org/jira/browse/ASTERISK-18480 which was closed as Not A Bug by Digium. We worked around this by prepending an integer (001__, 002__, ...) to the interface in the database table and then removing it later in the dial plan. Hope this helps. Steven Wheeler Hello, thank you for your reply. Is it the membername or the interface that needs to be sorted to have a certain order in the call queue ? How do you
[asterisk-users] Realtime Call Queues : call members in certain order
Hello, I'm using MySQL realtime Call Queues (table /queues/ and table /queue_members/). I would like to ring the members of the call queue in a certain order. Therefore I use ring strategy /lineair /and I put the members into the table /queue_members/ in the order in which they have to be rang. So I have the queue : | name | musicclass | announce | context | timeout | monitor_type | monitor_format | queue_youarenext | queue_thereare | queue_callswaiting | queue_holdtime | queue_minutes | queue_seconds | queue_lessthan | queue_thankyou | queue_reporthold | announce_frequency | announce_round_seconds | announce_holdtime | announce_position | retry | wrapuptime | maxlen | servicelevel | strategy | joinempty | leavewhenempty | eventmemberstatus | eventwhencalled | reportholdtime | memberdelay | weight | timeoutrestart | periodic_announce | periodic_announce_frequency | ringinuse | +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ | queue6 | default| NULL | | 12 | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | NULL | 30 | NULL | No | yes | 5 | 10 | 0 | NULL | linear | strict| strict | NULL | NULL| NULL |NULL | NULL | no | | 0 | no| +++--+-+-+--++--++++---+---+++--+++---+---+---+++--+--+---++---+-++-+++---+-+---+ and queue members : +--++++-++ | uniqueid | membername | queue_name | interface | penalty | paused | +--++++-++ | 44 | queuemem4 | queue6 | SIP/queuemem4 | 0 | NULL | | 45 | queuemem2 | queue6 | SIP/queuemem2 | 0 | NULL | | 46 | queuemem5 | queue6 | SIP/queuemem5 | 0 | NULL | | 47 | queuemem1 | queue6 | SIP/queuemem1 | 0 | NULL | | 48 | queuemem10 | queue6 | SIP/queuemem10 | 0 | NULL | | 49 | queuemem18 | queue6 | SIP/queuemem18 | 0 | NULL | | 50 | queuemem17 | queue6 | SIP/queuemem17 | 0 | NULL | | 51 | queuemem12 | queue6 | SIP/queuemem12 | 0 | NULL | | 52 | queuemem16 | queue6 | SIP/queuemem16 | 0 | NULL | | 53 | queuemem13 | queue6 | SIP/queuemem13 | 0 | NULL | +--++++-++ You can see that the member /queuemem4/ is first in line to be rang (has the first and lowest uniqueid in the table). But the first member that is being rang, is /queuemem//1/. How come ?? Kind regards, Jonas. -- _ -- 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] Mailinglist Digium IP-phones : provisioning Digium D70
Hello, is there a mailinglist where I can post questions regarding Digium IP-phones ? I have the following question : I'm trying to provision a Digium D70 IP-phone from a https provisioning server. The Digium D70 contacts the provisioning server correctly but seems to log in with the wrong credentials : /var/log/ssl_access_log : XX.XX.XX.46 - - [22/Jan/2014:12:15:09 +0100] GET /101001/000fd3068c59.cfg HTTP/1.1 401 481 XX.XX.XX.46 - - [22/Jan/2014:12:15:10 +0100] GET /101001/000FD3068C59.cfg HTTP/1.1 401 481 XX.XX.XX.46 - - [22/Jan/2014:12:15:10 +0100] GET /101001/.cfg HTTP/1.1 401 481 XX.XX.XX.46 - - [22/Jan/2014:12:15:11 +0100] GET /101001 HTTP/1.1 401 481 I am absolutely sure that I have given the correct username and password in the Digium D70 phone. I have tried logging in with my Firefox browser to the provisioning server, and that is succesful ! I get asked for the username and password, and I can see the content of the cfg-file. /var/log/ssl_access_log : XX.XX.XX.46 - 101001 [22/Jan/2014:12:32:26 +0100] GET /101001/000fd3068c59.cfg HTTP/1.1 200 2257 The difference I see is the 101001, which is the username. I see the following in the logs of the https provisioning server : [Wed Jan 22 14:00:34 2014] [error] [client XX.XX.XX.46] Digest: client used wrong authentication scheme `Basic': /101001 So how do I get the Digium IP-phone to use the md5 digest authentication ?? Kind regards, Jonas. -- _ -- 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] Call duration limit ? Calls end after 15 minutes...
Hello, I see the strange behaviour that outgoing calls end after 15 minutes. I didn't knew there is some kind of call duration limit that can be set ? Is there ? Using Asterisk 1.8.12.2 Kind regards, Jonas. -- _ -- 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] Call duration limit ? Calls end after 15 minutes...
On 08-01-14 16:47, Markus wrote: Am 08.01.2014 16:07, schrieb Jonas Kellens: Hello, I see the strange behaviour that outgoing calls end after 15 minutes. I didn't knew there is some kind of call duration limit that can be set ? Is there ? Look at session-timers in sip.conf. I had to set it to refuse for a specific provider because they are a little incompetent. Drawback is that a call can show as going on forever if the BYE message is lost due to network problems. Are SIP session timers also present in IP-phones ? Or is this only a setting in a SIP-server and not in a SIP client like an IP-phone ? Jonas. -- _ -- 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 building dahdi from source
Hello, I am getting the following error when compiling dahdi : make[2]: Entering directory `/usr/src/kernels/2.6.32-431.1.2.0.1.el6.x86_64' Building modules, stage 2. MODPOST 0 modules make[2]: Leaving directory `/usr/src/kernels/2.6.32-431.1.2.0.1.el6.x86_64' make -C /lib/modules/2.6.32-431.1.2.0.1.el6.x86_64/build SUBDIRS=/usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux/drivers/dahdi DAHDI_INCLUDE=/usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux/include DAHDI_MODULES_EXTRA= HOTPLUG_FIRMWARE=yes modules DAHDI_BUILD_ALL=m make[2]: Entering directory `/usr/src/kernels/2.6.32-431.1.2.0.1.el6.x86_64' CC [M] /usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux/drivers/dahdi/dahdi-base.o In file included from /usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux/drivers/dahdi/dahdi-base.c:66: /usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux/include/dahdi/kernel.h:1407: error: redefinition of 'PDE_DATA' include/linux/proc_fs.h:328: note: previous definition of 'PDE_DATA' was here make[3]: *** [/usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux/drivers/dahdi/dahdi-base.o] Error 1 make[2]: *** [_module_/usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux/drivers/dahdi] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.32-431.1.2.0.1.el6.x86_64' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.7.0.1+2.7.0.1/linux' make: *** [all] Error 2 I have the right kernel sources installed : [root@sip dahdi-linux-complete-2.7.0.1+2.7.0.1]# uname -a Linux sip 2.6.32-431.1.2.0.1.el6.x86_64 #1 SMP Fri Dec 13 13:06:13 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux So what am I missing ? Kind regards, Jonas. -- _ -- 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] Direct Media and message SIP/SipAgent-00000bf9 requested media update control 26, passing it to SIP/ead14-00000bfb
Hello, I have the following construction : Provider -- SipAgent (asterisk) -- Asterisk Server_A -- IP-phone (Snom 370) If a call comes in from the Provider to my SipAgent, then my SipAgent send the call to the correct Asterisk Server_A (dialplan logic based on number). The Asterisk Server_A takes the call and sends it to the IP-phone. My SipAgent has DirectMedia=yes so there is no audio flowing through this SipAgent. It only stays in the signaling path (SIP). My SipAgent will communicate in a SIP re-INVITE the audio ports of the Asterisk Server_A to the Provider. My SipAgent will communicate in a SIP re-INVITE the audio ports of the Provider to the Asterisk Server_A. Audio will flow directly between Provider and Asterisk Server_A. This works great. On my Asterisk Server_A, I see the following : /SIP/SipAgent-0bf9 requested media update control 26, passing it to SIP/ead14-0bfb/ Mostly this appears one time in a call. This I find normal. But sometimes the CLI is flooded with 100 of these messages... and that I find NOT NORMAL. The flood stops when the call is anwered. This is the SIP INVITE on my SipAgent : INVITE sip:xx32xxx...@xx.xx.xx.199:5060 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.198:5060;branch=z9hG4bK37fc69a2;rport Max-Forwards: 70 From: xx35xx sip:xx35xxx...@xx.xx.xx.198;tag=as3bbe54ca To: sip:xx32xxx...@xx.xx.xx.199;tag=as180f6a04 Contact: sip:xx35xxx...@xx.xx.xx.198:5060 Call-ID: 675c1f3f5141f5ac0e981c27414de...@xx.xx.xx.198:5060 CSeq: 103 INVITE User-Agent: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer X-asterisk-Info: SIP re-invite (External RTP bridge) Content-Type: application/sdp Content-Length: 239 X-Asterisk-Info shows the RTP bridge, which I find normal. And my Asterisk Server_A answers with 100 Trying. Now, what could be the difference between a call where the CLI on Asterisk Server_A tells /requested media update control 26/ one time and where it floods the CLI ?/ / Kind regards, Jonas. / / -- _ -- 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] RTP packets send, but no audio
Hello, What does it mean when rtp set debug ip shows RTP packets that have been send, but there is no audio ? There was no audio on my call in both directions, but rtp set debug shows that there were RTP packets send. There is no firewall active on my Asterisk server : [root@sip asterisk]# /sbin/service iptables status iptables: Firewall not running. Kind regards, Jonas. -- _ -- 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] RTP packets send, but no audio
On 28-11-13 11:45, Jonas Kellens wrote: Hello, What does it mean when rtp set debug ip shows RTP packets that have been send, but there is no audio ? There was no audio on my call in both directions, but rtp set debug shows that there were RTP packets send. There is no firewall active on my Asterisk server : [root@sip asterisk]# /sbin/service iptables status iptables: Firewall not running. Kind regards, Jonas. Sorry, here's the rtp set debug : [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015722, ts 140986608, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006738, ts 1884800, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015723, ts 140986768, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006739, ts 1884960, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015724, ts 140986928, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006740, ts 1885120, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015725, ts 140987088, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006741, ts 1885280, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015726, ts 140987248, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006742, ts 1885440, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015727, ts 140987408, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006743, ts 1885600, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015728, ts 140987568, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006744, ts 1885760, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015729, ts 140987728, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006745, ts 1885920, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015730, ts 140987888, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006746, ts 1886080, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015731, ts 140988048, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006747, ts 1886240, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015732, ts 140988208, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006748, ts 1886400, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015733, ts 140988368, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006749, ts 1886560, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015734, ts 140988528, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq 006750, ts 1886720, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Got RTP packet frommy_ip_address:16448 (type 08, seq 015735, ts 140988688, len 000160) [Nov 28 11:35:47] VERBOSE[8449] res_rtp_asterisk.c: [Nov 28 11:35:47] Sent RTP packet to my_ip_address:16448 (type 08, seq
[asterisk-users] Asterisk uses 105% CPU
Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init What can be causing such a high load of the asterisk proces ?? There are about 35 calls with G711a codec, no translation. Kind regards, Jonas. -- _ -- 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 uses 105% CPU
On 27-11-13 12:26, Jonas Kellens wrote: Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEMTIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init What can be causing such a high load of the asterisk proces ?? There are about 35 calls with G711a codec, no translation. Kind regards, Jonas. I want to add some more information. Maybe someone knows how to help me with this information : sip*CLI core show threads 0x7f98f87fd700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8ae5700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9229700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9609700 netconsole started at [ 1423] asterisk.c listener() 0x7f98f8971700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8ec5700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8e49700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9a65700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f97f9700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8a69700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8dcd700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8d51700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9ae1700 shaun_of_the_deadstarted at [ 2141] app.c ast_safe_fork() 0x7f98f9b5d700 inotify_daemon started at [ 334] stdtime/localtime.c add_notify() 0x7f98f9def700 autoservice_run started at [ 219] autoservice.c ast_autoservice_start() 0x7f98f9ee7700 monitor_sig_flagsstarted at [ 4097] asterisk.c main() 0x7f98f9f63700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 0x7f98f9fdf700 cleanup started at [ 414] pbx_realtime.c load_module() 0x7f98fa05b700 scan_thread started at [ 885] pbx_spool.c load_module() 0x7f98fa0d7700 do_monitor started at [ 4684] chan_unistim.c restart_monitor() 0x7f98fa153700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 0x7f98fa1cf700 process_clearcache started at [ 2265] pbx_dundi.c start_network_thread() 0x7f98fa2c7700 network_thread started at [ 2263] pbx_dundi.c start_network_thread() 0x7f98fa24b700 process_precache started at [ 2264] pbx_dundi.c start_network_thread() 0x7f98fa343700 do_monitor started at [ 1167] chan_phone.c restart_monitor() 0x7f98fa3bf700 lock_broker started at [ 509] func_lock.c load_module() 0x7f98fa43b700 network_thread started at [12310] chan_iax2.c start_network_thread() 0x7f98fa4b7700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa533700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa5af700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa62b700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa6a7700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa723700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa79f700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa81b700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa897700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa913700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa98f700 sched_runstarted at [ 186] sched.c ast_sched_thread_create() 0x7f98faa0b700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 0x7f98faa87700 do_monitor started at [ 3897] chan_mgcp.c restart_monitor() 0x7f98fab03700 do_monitor started at [ 6647] chan_skinny.c restart_monitor() 0x7f98fab7f700 accept_threadstarted at [ 7358] chan_skinny.c config_load() 0x7f98fabfb700 do_monitor started at [12011] chan_dahdi.c restart_monitor() 0x7f98fac77700 do_monitor started at [26669] chan_sip.c restart_monitor() 0x7f992c09a700 do_timingstarted at [ 490] res_timing_pthread.c init_timing_thread() 0x7f992e55f700 do_refresh started at [ 1766] res_calendar.c load_module() 0x7f992f84b700 sched_runstarted at [ 186] sched.c ast_sched_thread_create() 0x7f992f8c7700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 0x7f992f943700 db_sync_thread started at [ 883] db.c astdb_init() 0x7f993c082700 do_parking_threadstarted at [ 8304] features.c
Re: [asterisk-users] Asterisk uses 105% CPU
Server specs : XEON E3-1220V2 4 GB RAM 2 x 500GB HD (RAID0) 1 U HOT-PLUG PSU Linux sip.server.tld 2.6.32-358.18.1.el6.x86_64 #1 SMP Wed Aug 28 17:19:38 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux There is no transcoding. Calls are using G711a. Maybe there is some trancoding when using voicemail... How can I find out if there is trancoding ?? Kind regards, Jonas. On 27-11-13 13:27, Andrew Colin wrote: Are you transcoding? What is your server spec? Regards Andrew Colin-mobile Vsave(PTY)Ltd Original message From: Jonas Kellens Date:27/11/2013 13:48 (GMT+02:00) To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk uses 105% CPU On 27-11-13 12:26, Jonas Kellens wrote: Hello, Using asterisk 1.8.24 on CentOS 6.4 I notice that the asterisk process is using between 105 en 110 % CPU : PID USER PR NI VIRT RES SHR S %CPU %MEM TIME+ COMMAND 1765 root 20 0 2508m 102m 8864 S 105.8 2.7 102:11.55 asterisk 2682 mysql 20 0 627m 29m 6204 S 0.7 0.8 1:59.51 mysqld 1 root 20 0 19228 1508 1220 S 0.0 0.0 0:00.75 init What can be causing such a high load of the asterisk proces ?? There are about 35 calls with G711a codec, no translation. Kind regards, Jonas. I want to add some more information. Maybe someone knows how to help me with this information : sip*CLI core show threads 0x7f98f87fd700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8ae5700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9229700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9609700 netconsole started at [ 1423] asterisk.c listener() 0x7f98f8971700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8ec5700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8e49700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9a65700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f97f9700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8a69700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8dcd700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f8d51700 pbx_thread started at [ 5597] pbx.c ast_pbx_start() 0x7f98f9ae1700 shaun_of_the_deadstarted at [ 2141] app.c ast_safe_fork() 0x7f98f9b5d700 inotify_daemon started at [ 334] stdtime/localtime.c add_notify() 0x7f98f9def700 autoservice_run started at [ 219] autoservice.c ast_autoservice_start() 0x7f98f9ee7700 monitor_sig_flagsstarted at [ 4097] asterisk.c main() 0x7f98f9f63700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 0x7f98f9fdf700 cleanup started at [ 414] pbx_realtime.c load_module() 0x7f98fa05b700 scan_thread started at [ 885] pbx_spool.c load_module() 0x7f98fa0d7700 do_monitor started at [ 4684] chan_unistim.c restart_monitor() 0x7f98fa153700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 0x7f98fa1cf700 process_clearcache started at [ 2265] pbx_dundi.c start_network_thread() 0x7f98fa2c7700 network_thread started at [ 2263] pbx_dundi.c start_network_thread() 0x7f98fa24b700 process_precache started at [ 2264] pbx_dundi.c start_network_thread() 0x7f98fa343700 do_monitor started at [ 1167] chan_phone.c restart_monitor() 0x7f98fa3bf700 lock_broker started at [ 509] func_lock.c load_module() 0x7f98fa43b700 network_thread started at [12310] chan_iax2.c start_network_thread() 0x7f98fa4b7700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa533700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa5af700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa62b700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa6a7700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa723700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa79f700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa81b700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa897700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa913700 iax2_process_thread started at [12288] chan_iax2.c start_network_thread() 0x7f98fa98f700 sched_runstarted at [ 186] sched.c ast_sched_thread_create() 0x7f98faa0b700 tps_processing_function started at [ 468] taskprocessor.c ast_taskprocessor_get() 0x7f98faa87700 do_monitor started at [ 3897] chan_mgcp.c restart_monitor() 0x7f98fab03700 do_monitor started at [ 6647] chan_skinny.c restart_monitor() 0x7f98fab7f700 accept_threadstarted
[asterisk-users] Asterisk 1.8.24 : illegal instruction
Hello, I have installed asterisk 1.8.24 (from source) but I can not start up Asterisk : [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction [root@sip32 admin]# /sbin/service asterisk status asterisk dead but subsys locked [root@sip32 admin]# /sbin/service asterisk restart Stopping safe_asterisk:[ OK ] Shutting down asterisk: [FAILED] Starting asterisk: [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction [root@sip32 admin]# [root@sip32 admin]# /usr/sbin/asterisk -c Illegal instruction Why can I not start Asterisk ? I also notice the following in /var/log/asterisk : [root@sip32 admin-voipcenter]# tail -f /var/log/messages Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000] Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000] Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000] Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000] Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000] Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000] Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000] Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000] Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000] Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000] Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000] Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000] Kind regards, Jonas. -- _ -- 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 1.8.24 : illegal instruction
Hello, I have installed asterisk 1.8.24 (from source) but I can not start up Asterisk : [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction [root@sip32 admin]# /sbin/service asterisk status asterisk dead but subsys locked [root@sip32 admin]# /sbin/service asterisk restart Stopping safe_asterisk:[ OK ] Shutting down asterisk: [FAILED] Starting asterisk: [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction [root@sip32 admin]# [root@sip32 admin]# /usr/sbin/asterisk -c Illegal instruction Why can I not start Asterisk ? I also notice the following in /var/log/messages : [root@sip32 admin-voipcenter]# tail -f /var/log/messages Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000] Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000] Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000] Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000] Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000] Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000] Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000] Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000] Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000] Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000] Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000] Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000] Kind regards, Jonas. -- _ -- 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 1.8.24 : illegal instruction
Hello, how can I mix libraries ? I have installed prerequisites from yum and asterisk from source (make make install). My kernel : [root@sip32 asterisk-1.8.24.0]# uname -a Linux sip32.domain.tld 2.6.32-358.23.2.el6.x86_64 #1 SMP Wed Oct 16 18:37:12 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux Jonas. On 20-11-13 14:11, Ron Wheeler wrote: Is it possible that in your build you mixed 32 bit and 64 bit libraries? Ron On 20/11/2013 8:06 AM, Jonas Kellens wrote: Hello, I have installed asterisk 1.8.24 (from source) but I can not start up Asterisk : [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction [root@sip32 admin]# /sbin/service asterisk status asterisk dead but subsys locked [root@sip32 admin]# /sbin/service asterisk restart Stopping safe_asterisk:[ OK ] Shutting down asterisk: [FAILED] Starting asterisk: [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction [root@sip32 admin]# [root@sip32 admin]# /usr/sbin/asterisk -c Illegal instruction Why can I not start Asterisk ? I also notice the following in /var/log/messages : [root@sip32 admin-voipcenter]# tail -f /var/log/messages Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000] Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000] Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000] Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000] Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000] Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000] Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000] Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000] Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000] Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000] Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000] Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000] Kind regards, Jonas. -- Ron Wheeler President Artifact Software Inc email:rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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 1.8.24 : illegal instruction
Hello, same problem with asterisk-1.8.23.1 So how do I check if I have 32 bit libs installed ? I always install with yum, so on a 64bit CentOS 6.4 there should only be 64bit libs installed... Jonas. On 20-11-13 14:26, Ron Wheeler wrote: I am not sure that this is the cause of your problem but I think that the message that you are getting can be caused by that. You might want to check the build logs to be sure that you do not have a 32 bit library installed. 32 bit libraries will work on 64 bit Linux but not when mixed with 64 bit applications. Ron On 20/11/2013 8:15 AM, Jonas Kellens wrote: Hello, how can I mix libraries ? I have installed prerequisites from yum and asterisk from source (make make install). My kernel : [root@sip32 asterisk-1.8.24.0]# uname -a Linux sip32.domain.tld 2.6.32-358.23.2.el6.x86_64 #1 SMP Wed Oct 16 18:37:12 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux Jonas. On 20-11-13 14:11, Ron Wheeler wrote: Is it possible that in your build you mixed 32 bit and 64 bit libraries? Ron On 20/11/2013 8:06 AM, Jonas Kellens wrote: Hello, I have installed asterisk 1.8.24 (from source) but I can not start up Asterisk : [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction [root@sip32 admin]# /sbin/service asterisk status asterisk dead but subsys locked [root@sip32 admin]# /sbin/service asterisk restart Stopping safe_asterisk: [ OK ] Shutting down asterisk: [FAILED] Starting asterisk: [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction [root@sip32 admin]# [root@sip32 admin]# /usr/sbin/asterisk -c Illegal instruction Why can I not start Asterisk ? I also notice the following in /var/log/messages : [root@sip32 admin-voipcenter]# tail -f /var/log/messages Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000] Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000] Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000] Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000] Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000] Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000] Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000] Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000] Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000] Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000] Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000] Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000] Kind regards, Jonas. -- Ron Wheeler President Artifact Software Inc email:rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- Ron Wheeler President Artifact Software Inc email:rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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 1.8.24 : illegal instruction
Hello, I think there are no 32bit libs installed : [root@sip32 src]# rpm -qa | grep 'i[6543]86' [root@sip32 src]# yum list installed *.i*86 Loaded plugins: downloadonly, fastestmirror Loading mirror speeds from cached hostfile * base: mirror.muntinternet.net * epel: mirror.muntinternet.net * extras: mirror.muntinternet.net * rpmforge: nl.mirror.eurid.eu * updates: mirror.muntinternet.net Error: No matching Packages to list Jonas. On 20-11-13 14:26, Ron Wheeler wrote: I am not sure that this is the cause of your problem but I think that the message that you are getting can be caused by that. You might want to check the build logs to be sure that you do not have a 32 bit library installed. 32 bit libraries will work on 64 bit Linux but not when mixed with 64 bit applications. Ron On 20/11/2013 8:15 AM, Jonas Kellens wrote: Hello, how can I mix libraries ? I have installed prerequisites from yum and asterisk from source (make make install). My kernel : [root@sip32 asterisk-1.8.24.0]# uname -a Linux sip32.domain.tld 2.6.32-358.23.2.el6.x86_64 #1 SMP Wed Oct 16 18:37:12 UTC 2013 x86_64 x86_64 x86_64 GNU/Linux Jonas. On 20-11-13 14:11, Ron Wheeler wrote: Is it possible that in your build you mixed 32 bit and 64 bit libraries? Ron On 20/11/2013 8:06 AM, Jonas Kellens wrote: Hello, I have installed asterisk 1.8.24 (from source) but I can not start up Asterisk : [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction [root@sip32 admin]# /sbin/service asterisk status asterisk dead but subsys locked [root@sip32 admin]# /sbin/service asterisk restart Stopping safe_asterisk: [ OK ] Shutting down asterisk: [FAILED] Starting asterisk: [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction [root@sip32 admin]# [root@sip32 admin]# /usr/sbin/asterisk -c Illegal instruction Why can I not start Asterisk ? I also notice the following in /var/log/messages : [root@sip32 admin-voipcenter]# tail -f /var/log/messages Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000] Nov 20 14:04:31 sip32 kernel: asterisk[2034] trap invalid opcode ip:530b18 sp:7fffa6051a60 error:0 in asterisk[40+1d7000] Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000] Nov 20 14:04:35 sip32 kernel: asterisk[2041] trap invalid opcode ip:530b18 sp:7fff11d96bf0 error:0 in asterisk[40+1d7000] Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000] Nov 20 14:04:39 sip32 kernel: asterisk[2047] trap invalid opcode ip:530b18 sp:7fff7913f1a0 error:0 in asterisk[40+1d7000] Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000] Nov 20 14:04:43 sip32 kernel: asterisk[2053] trap invalid opcode ip:530b18 sp:7fff663f32c0 error:0 in asterisk[40+1d7000] Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000] Nov 20 14:04:47 sip32 kernel: asterisk[2059] trap invalid opcode ip:530b18 sp:7fffb5200b90 error:0 in asterisk[40+1d7000] Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000] Nov 20 14:04:51 sip32 kernel: asterisk[2066] trap invalid opcode ip:530b18 sp:7fffc19af630 error:0 in asterisk[40+1d7000] Kind regards, Jonas. -- Ron Wheeler President Artifact Software Inc email:rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- Ron Wheeler President Artifact Software Inc email:rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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 1.8.24 : illegal instruction
On 20-11-13 14:43, A J Stiles wrote: On Wednesday 20 November 2013, Jonas Kellens wrote: Hello, I have installed asterisk 1.8.24 (from source) but I can not start up Asterisk : [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction Are you using a VIA C6/C7 processor (often found soldered to tiny motherboards), by any chance? This family of processors falsely report as i686 when they lack some of the instructions for this family. The fix is to build for a target architecture of i586. No, this is a Xen VPS. Jonas. -- _ -- 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 1.8.24 : illegal instruction
The information requested : [root@sip32 src]# file /usr/sbin/asterisk /usr/sbin/asterisk: ELF 64-bit LSB executable, x86-64, version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux 2.6.18, not stripped [root@sip32 src]# ldd /usr/sbin/asterisk linux-vdso.so.1 = (0x7fff677ff000) libssl.so.10 = /usr/lib64/libssl.so.10 (0x7fde449fc000) libcrypto.so.10 = /usr/lib64/libcrypto.so.10 (0x7fde44662000) libc.so.6 = /lib64/libc.so.6 (0x7fde442ce000) libxml2.so.2 = /usr/lib64/libxml2.so.2 (0x7fde43f7c000) libz.so.1 = /lib64/libz.so.1 (0x7fde43d66000) libm.so.6 = /lib64/libm.so.6 (0x7fde43ae1000) libdl.so.2 = /lib64/libdl.so.2 (0x7fde438dd000) libpthread.so.0 = /lib64/libpthread.so.0 (0x7fde436c) libtinfo.so.5 = /lib64/libtinfo.so.5 (0x7fde4349e000) libresolv.so.2 = /lib64/libresolv.so.2 (0x7fde43284000) libgssapi_krb5.so.2 = /lib64/libgssapi_krb5.so.2 (0x7fde4304) libkrb5.so.3 = /lib64/libkrb5.so.3 (0x7fde42d59000) libcom_err.so.2 = /lib64/libcom_err.so.2 (0x7fde42b55000) libk5crypto.so.3 = /lib64/libk5crypto.so.3 (0x7fde42929000) /lib64/ld-linux-x86-64.so.2 (0x7fde44c5f000) libkrb5support.so.0 = /lib64/libkrb5support.so.0 (0x7fde4271d000) libkeyutils.so.1 = /lib64/libkeyutils.so.1 (0x7fde4251a000) libselinux.so.1 = /lib64/libselinux.so.1 (0x7fde422fa000) Kind regards, Jonas. On 20-11-13 15:00, Asghar Mohammad wrote: Hello, you can check the asterisk binary with. file /usr/sbin/asterisk and linked library ldd /usr/sbin/asterisk On Wed, Nov 20, 2013 at 2:51 PM, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: On 20-11-13 14:43, A J Stiles wrote: On Wednesday 20 November 2013, Jonas Kellens wrote: Hello, I have installed asterisk 1.8.24 (from source) but I can not start up Asterisk : [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction Are you using a VIA C6/C7 processor (often found soldered to tiny motherboards), by any chance? This family of processors falsely report as i686 when they lack some of the instructions for this family. The fix is to build for a target architecture of i586. No, this is a Xen VPS. Jonas. -- _ -- 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 1.8.24 : illegal instruction
Hello, problem is solved by compiling Asterisk as follow : [root@sip32 asterisk]# ./configure CFLAGS=-mtune=native Now Asterisk starts normally, without any error message. Is this a problem of Asterisk or a problem of gcc ?? Kind regards, Jonas. On 20-11-13 15:00, Asghar Mohammad wrote: Hello, you can check the asterisk binary with. file /usr/sbin/asterisk and linked library ldd /usr/sbin/asterisk On Wed, Nov 20, 2013 at 2:51 PM, Jonas Kellens jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote: On 20-11-13 14:43, A J Stiles wrote: On Wednesday 20 November 2013, Jonas Kellens wrote: Hello, I have installed asterisk 1.8.24 (from source) but I can not start up Asterisk : [root@sip32 admin]# /usr/sbin/asterisk -r Illegal instruction Are you using a VIA C6/C7 processor (often found soldered to tiny motherboards), by any chance? This family of processors falsely report as i686 when they lack some of the instructions for this family. The fix is to build for a target architecture of i586. No, this is a Xen VPS. Jonas. -- _ -- 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] Add SIP Header for 1 SIP peer when calling a group of SIP peers
Hello, when calling a group of SIP peers like this : Dial( SIP/inno0SIP/inno4SIP/inno6,30) is it possible to have a SIP header added for just 1 of these SIP peers, like only for SIP/inno0 but not for SIP/inno4 and SIP/inno6 ?? I know the function SipAddHeader(), but when I use this in the dialplan before the Dial()-command, then the header is added for all the SIP peers that are being called. So when calling a group of SIP peers, how can I add an extra SIP header for just one of the SIP peers ? Kind regards, Jonas. -- _ -- 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] VoIP sound quality : highroad sound
On 11/13/2013 11:48 AM, Johan Wilfer wrote: 2013-11-12 17:42, Jonas Kellens skrev: X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 A lot of packetloss for theese calls. I would do packetdumps with tcpdump and then analyze it with wireshark. I use voipmonitor to do this (it gives you a pcap for each call), but tcpdump works fine also. This could be a congested link, a broken media gateway, or anything I have already used tcpdump and analyzed the calls with wireshark. When I listen to the call, I clearly hear the highroad sound (always on the upload side). What else can wireshark tell me ? How can wireshark further tell me about the cause of this poor sound quality ? Kind regards, Jonas. -- _ -- 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] calendar.conf include
Hello, can I use include-statements in the calendar.conf configuration file ? Kind regards, Jonas. -- _ -- 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] VoIP sound quality : highroad sound
Hello, what could be causing the issue of poor sound quality ? Some calls, certainly not all of them, sound like if the caller is standing next to a very busy road with lots of cars passing. To be clear : the person calling is not standing next to a highway. But there seems to be a noise on the line of busy highroad that makes that the caller can not be understood. What can be causing this kind of poor quality ? Is it lack of resources on the Asterisk-server (codec translation ?) Is it lack of bandwith ? Is it a problem of CentOS (the underlying OS) ? Is it a physical problem of the server components (network interface ?) ? Kind regards, Jonas. -- _ -- 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] VoIP sound quality : highroad sound
On 11/12/2013 04:29 PM, jg wrote: Did you have a look at the codecs that are involved? There are about 40 Ã 45 simultaneous calls (using G711a). Jonas. -- _ -- 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] VoIP sound quality : highroad sound
Current situation : sip1*CLI sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter X.X.X.133 4d7b0a7f337 00:05:59 000243 00 ( 0.00%) 0. 000576 046854 (8134.38%) 0.0002 X.X.X.42 3c8956648ce 00:02:27 007301 00 ( 0.00%) 0. 007318 01 ( 0.01%) 0.0020 X.X.X.224 684333f5650 00:00:03 00 00 ( 0.00%) 0. 000178 00 ( 0.00%) 0. X.X.X.98 5eceb3a5624 00 00 ( 0.00%) 0. 00 00 ( 0.00%) 0. X.X.X.9825ae26ee564 00:00:03 000179 00 ( 0.00%) 0. 00 00 ( 0.00%) 0. X.X.X.986b26738a0c4 00:00:43 000137 00 ( 0.00%) 0. 000137 00 ( 0.00%) 0.0001 X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 X.X.X.1846893e957-fa 00:05:59 000576 00 ( 0.00%) 0. 000243 00 ( 0.00%) 0.0027 9 active SIP channels Thanks. Jonas. On 11/12/2013 05:32 PM, jg wrote: Are these all SIP-channels? If yes, or if one endpoint is always a SIP-device then you could issue a sip show channelstats in the cli. This is not exact, but it shows if you have any network or timing problems. I could say more about network problems, but first let's see what channelstats says. jg Am 12.11.2013 16:34, schrieb Jonas Kellens: On 11/12/2013 04:29 PM, jg wrote: Did you have a look at the codecs that are involved? There are about 40 Ã 45 simultaneous calls (using G711a). Jonas. -- _ -- 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] VoIP sound quality : highroad sound
Yes, all SIP. Current situation : sip1*CLI sip show channelstats Peer Call ID Duration Recv: Pack Lost ( %) Jitter Send: Pack Lost ( %) Jitter X.X.X.133 4d7b0a7f337 00:05:59 000243 00 ( 0.00%) 0. 000576 046854 (8134.38%) 0.0002 X.X.X.42 3c8956648ce 00:02:27 007301 00 ( 0.00%) 0. 007318 01 ( 0.01%) 0.0020 X.X.X.224 684333f5650 00:00:03 00 00 ( 0.00%) 0. 000178 00 ( 0.00%) 0. X.X.X.98 5eceb3a5624 00 00 ( 0.00%) 0. 00 00 ( 0.00%) 0. X.X.X.9825ae26ee564 00:00:03 000179 00 ( 0.00%) 0. 00 00 ( 0.00%) 0. X.X.X.986b26738a0c4 00:00:43 000137 00 ( 0.00%) 0. 000137 00 ( 0.00%) 0.0001 X.X.X.100 2f9a96ec3e1 00:00:42 000138 049741 (99.72%) 0. 000136 00 ( 0.00%) 0.0031 X.X.X.70 68289fc05ff 00:02:27 007318 060143 (89.15%) 0. 007301 00 ( 0.00%) 0.0001 X.X.X.1846893e957-fa 00:05:59 000576 00 ( 0.00%) 0. 000243 00 ( 0.00%) 0.0027 9 active SIP channels Thanks. Jonas. On 11/12/2013 05:32 PM, jg wrote: Are these all SIP-channels? If yes, or if one endpoint is always a SIP-device then you could issue a sip show channelstats in the cli. This is not exact, but it shows if you have any network or timing problems. I could say more about network problems, but first let's see what channelstats says. jg Am 12.11.2013 16:34, schrieb Jonas Kellens: On 11/12/2013 04:29 PM, jg wrote: Did you have a look at the codecs that are involved? There are about 40 Ã 45 simultaneous calls (using G711a). Jonas. -- _ -- 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] Question about how Asterisk works with RTP ports
Hello, short question : does Asterisk reserve RTP ports for every IP-phone that is being called ? If for instance an incoming call makes 10 IP-phones ring, does this mean that Asterisk preserves 10 x 2 RTP ports for audio ? I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port number for audio ? If this is the case for the 10 IP-phones to which an INVITE is send to, this means at least 10 RTP ports are reserved for incoming audio, correct ??? Thanks. Jonas. -- _ -- 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] Question about how Asterisk works with RTP ports
On 10/29/2013 05:14 PM, Joshua Colp wrote: Jonas Kellens wrote: Hello, short question : does Asterisk reserve RTP ports for every IP-phone that is being called ? It uses 2 ports per channel under normal circumstances, 1 for RTP and 1 for RTCP. If for instance an incoming call makes 10 IP-phones ring, does this mean that Asterisk preserves 10 x 2 RTP ports for audio ? Yes. I guess Asterisk sends in the SIP INVITE an SDP body with an RTP port number for audio ? If this is the case for the 10 IP-phones to which an INVITE is send to, this means at least 10 RTP ports are reserved for incoming audio, correct ??? Yes. So if I understand correct, you don't need to look at the amount of concurrent calls to calculate the RTP range in rtp.conf, you need to look at the amount of INVITES that are being send at one moment ? Kind regards, Jonas. -- _ -- 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] Use Asterisk Realtime Extensions with Switch-statement and include-statement
Hello, Is it possible to use the switch = statement in extensions.conf (http://www.voip-info.org/wiki/view/Asterisk+RealTime+Extensions) to point to a database and in the database use the include-statement ? In extconfig.conf I would have : extensions = mysql,asterisk,extensions_table In extensions.conf I would then have : [includecontext] switch = Realtime/includecontext@realtime_ext in database : INSERT INTO `my_extensions_table` VALUES (NULL, 'includecontext', 'include', =, 'context1', ''); INSERT INTO `my_extensions_table` VALUES (NULL, 'includecontext', 'include', =, 'context2', ''); INSERT INTO `my_extensions_table` VALUES (NULL, 'includecontext', 'include', =, 'context3', ''); This would then replace the following in extensions.conf : [includecontext] include = context1 include = context2 include = context3 Possible or not ? Thanks, Jonas Kellens -- _ -- 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] Exit Call Queue by pressing digit
Hello, I want a caller who is waiting in the queue to be able to exit this queue (and the waiting) by pressing a digit. I read in the wiki : /Context// //; A context may be specified, in which if the user types a SINGLE digit extension while they are in the queue, they will be taken out of the queue and sent to that extension in this context.// //context=context// //This is the context that is used to allow the caller to exit with a key for further action. For example, press 1 to leave a message/ So I fill in the 'context'-parameter with a value 'queueexitdigit'. In extensions.conf I have a context [queueexitdigit]. But when I test this and press a key (for example 5) while I'm waiting in the call queue, nothing happens ! I'm still in the call queue... waiting. Which part of the configuration am I missing ? Kind regards, Jonas. -- _ -- 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] RTP port ranges
Hello, I have defined that I want to receive audio (RTP) on port 11500 till 11954 (rtp.conf). The same range I have defined in my firewall. I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. How come the client sends audio on port 11955 when I clearly define in my SDP-body that I want to receive audio on port range 11500 till 11954 ? What makes the client choose this port number when it is not allowed ? Kind regards, Jonas. -- _ -- 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] RTP port ranges
On 09/13/2013 11:41 AM, Andrew Colin wrote: Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Jonas. -- _ -- 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] RTP port ranges
Hello, and when I define 11500 - 11954 it should use a random port in this range. Where is it stated that you MUST use 1-2 ??? Someone else please ? Jonas. On 09/13/2013 11:46 AM, Andrew Colin wrote: Because normally it will use a random port between them On 9/13/2013 11:43 AM, Jonas Kellens wrote: On 09/13/2013 11:41 AM, Andrew Colin wrote: Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Jonas. -- _ -- 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] RTP port ranges
Could be... is there no way to be sure ? Is there no way to calculate this ? Thanks, Jonas. On 09/13/2013 12:11 PM, Johann Steinwendtner wrote: Maybe you should open 11955 on you fw as well. This could be the rtcp port. Regards Hans On 2013-09-13 11:49, Jonas Kellens wrote: Hello, and when I define 11500 - 11954 it should use a random port in this range. Where is it stated that you MUST use 1-2 ??? Someone else please ? Jonas. On 09/13/2013 11:46 AM, Andrew Colin wrote: Because normally it will use a random port between them On 9/13/2013 11:43 AM, Jonas Kellens wrote: On 09/13/2013 11:41 AM, Andrew Colin wrote: Normally you should open ports 1-2 udp On 9/13/2013 11:37 AM, Jonas Kellens wrote: I now see that an IP-address gets blocked by my firewall because there are packets coming onto port 11955. Why do I need such a big range ? That's like for 250 concurrent calls ! Jonas. -- _ -- 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] Cut off last character of EXTEN
Hello,
how can I cut off the last character of the EXTEN-variable with
variating length ?
So I have :
112233#
123#
123456789#
I want to cut off the last character.
${EXTEN:-1} gives me #, but that is the character I want to cut off.
Kind regards,
Jonas.
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Re: [asterisk-users] Cut off last character of EXTEN
On 08/20/2013 10:47 AM, jg wrote:
How about ${EXTEN:-1:1}?
The Definitive Guide has a special paragraph with the title *More
Advanced Digit Manipulation.*
jg
Same result : #
Jonas.
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Re: [asterisk-users] Cut off last character of EXTEN
On 08/20/2013 10:40 AM, Gareth Blades wrote:
On 20/08/13 09:29, Jonas Kellens wrote:
Hello,
how can I cut off the last character of the EXTEN-variable with
variating length ?
So I have :
112233#
123#
123456789#
I want to cut off the last character.
${EXTEN:-1} gives me #, but that is the character I want to cut off.
Set(variable=${EXTEN:0:$[LEN(${EXTEN})-1]})
Hello,
this works !
Thanks.
Jonas.
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[asterisk-users] Dialplan MySQL inserted ID
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
Kind regards,
Jonas.
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Re: [asterisk-users] Dialplan MySQL inserted ID
On 08/20/2013 06:03 PM, Gergo Csibra wrote:
Tuesday, August 20, 2013, 5:47:24 PM, Gareth wrote:
On 20/08/13 14:53, Jonas Kellens wrote:
Hello,
how can I obtain the inserted ID after having inserted a row with
MySQL in the dialplan ?
exten = s,n,MYSQL(Query resultid ${connid} INSERT INTO myTable SET
C1=${ARG1}, C2=${ARG2},
timestamp=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
I need to know the ID of the newly inserted row.
You could write an AGI script in something like php or perl and get it
to write to the mysql database instead. It can then set a variable which
the dialplan can pick up.
meh...
SELECT LAST_INSERT_ID()
Hello,
can I echo this variable ?
Like : exten = s,n,NoOp(${LAST_INSERT_ID()})
Kind regards,
Jonas.
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[asterisk-users] include directory with multiple files in it
Hello, is it possible to use the #include - syntax to include several configuration files situated in one directory ? Something like : extensions.conf : #include extra/* #include addons/* Is this possible ? Using asterisk 1.8 Thanks. Jonas. -- _ -- 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] Why does it take several seconds to interpret DTMF-input ?
Hello, I notice that it takes 4 to 6 seconds between someone pressing a cipher and Asterisk continuing inside the dialplan. How come ??? Taken from verbose logfile : (attempt 1) [Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin '1' received on SIP/SipAgenT01-1eb0 [Jun 11 15:29:25] DTMF[18549] channel.c: DTMF begin ignored '1' on SIP/SipAgenT01-1eb0 [Jun 11 15:29:25] DTMF[18549] channel.c: DTMF end '1' received on SIP/SipAgenT01-1eb0, duration 180 ms [Jun 11 15:29:25] DTMF[18549] channel.c: DTMF end passthrough '1' on SIP/SipAgenT01-1eb0 [Jun 11 15:29:30] VERBOSE[18549] pbx.c: [Jun 11 15:29:30] == CDR updated on SIP/SipAgenT01-1eb0 [Jun 11 15:29:30] VERBOSE[18549] pbx.c: [Jun 11 15:29:30] -- Executing [1@pbx-routing:1] Set(SIP/SipAgenT01-1eb0, choice=1) in new stack [Jun 11 15:29:30] VERBOSE[18549] pbx.c: [Jun 11 15:29:30] -- Executing [1@pbx-routing:2] System(SIP/SipAgenT01-1eb0, echo '418','IVR','1','','SipAgenT01-1eb0','$(date +%s)' /var/log/asterisk/loggingAST/SipAgenT01-1eb0.csv) in new stack (attempt 2) [Jun 11 15:30:21] DTMF[18780] channel.c: DTMF begin '8' received on SIP/SipAgenT01-1ec1 [Jun 11 15:30:21] DTMF[18780] channel.c: DTMF begin ignored '8' on SIP/SipAgenT01-1ec1 [Jun 11 15:30:21] DTMF[18780] channel.c: DTMF end '8' received on SIP/SipAgenT01-1ec1, duration 160 ms [Jun 11 15:30:21] DTMF[18780] channel.c: DTMF end passthrough '8' on SIP/SipAgenT01-1ec1 [Jun 11 15:30:27] VERBOSE[18780] pbx.c: [Jun 11 15:30:27] == CDR updated on SIP/SipAgenT01-1ec1 [Jun 11 15:30:27] VERBOSE[18780] pbx.c: [Jun 11 15:30:27] -- Executing [8@pbx-routing:1] Set(SIP/SipAgenT01-1ec1, choice=8) in new stack [Jun 11 15:30:27] VERBOSE[18780] pbx.c: [Jun 11 15:30:27] -- Executing [8@pbx-routing:2] System(SIP/SipAgenT01-1ec1, echo '418','IVR','8','','SipAgenT01-1ec1','$(date +%s)' /var/log/asterisk/loggingAST/SipAgenT01-1ec1.csv) in new stack Why doesn't Asterisk continue immediately inside the dialplan after having received the DTMF-input ? Kind regards, Jonas. -- _ -- 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] Why does it take several seconds to interpret DTMF-input ?
On 06/11/2013 04:12 PM, Matthew J. Roth wrote:
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between someone pressing a cipher and
Asterisk continuing inside the dialplan. How come ???
...
Why doesn't Asterisk continue immediately inside the dialplan after having
received the DTMF-input ?
Jonas,
Please provide the version of Asterisk you are using and the part of the
dialplan
that receives the DTMF input.
Regards,
Matthew Roth
Hello,
using Asterisk 1.8.12.2.
Dialplan :
exten = ivr,1,NoOp()
exten =
ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivr,n,NoOp(${BACKGROUNDSTATUS})
exten = ivr,n,WaitExten(15)
exten = ivr,n,GoTo(restartprompt)
exten = _X,1,Set(choice=${EXTEN})
exten = _X,n,System(echo
'${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date +%s)'
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X,n,other_stuff_I_do
exten = _X.,1,Set(choice=${EXTEN})
exten = _X.,n,System(echo
'${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date +%s)'
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X.,n,other_stuff_I_do
Kind regards,
Jonas.
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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?
On 06/11/2013 04:39 PM, Richard Mudgett wrote:
On Tue, Jun 11, 2013 at 9:29 AM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
On 06/11/2013 04:12 PM, Matthew J. Roth wrote:
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between someone pressing a cipher and
Asterisk continuing inside the dialplan. How come ???
...
Why doesn't Asterisk continue immediately inside the dialplan after having
received the DTMF-input ?
snip
Dialplan :
exten = ivr,1,NoOp()
exten =
ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivr,n,NoOp(${BACKGROUNDSTATUS})
exten = ivr,n,WaitExten(15)
exten = ivr,n,GoTo(restartprompt)
exten = _X,1,Set(choice=${EXTEN})
exten = _X,n,System(echo
'${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date +%s)'
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X,n,other_stuff_I_do
exten = _X.,1,Set(choice=${EXTEN})
exten = _X.,n,System(echo
'${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date +%s)'
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X.,n,other_stuff_I_do
It is waiting for more digits because you have asked it for a possible
multi-digit exten and it needs to distinguish between the _X and _X.
patterns.
Richard
Ok thanks.
Any idea how I can resolve this ?
Even if there *can* be more than 1 digit, in case there is only 1 digit
it should go faster.
Could this dialplan logic be a good solution :
[my-context]
exten = ivr,1,NoOp()
exten =
ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivr,n,NoOp(${BACKGROUNDSTATUS})
exten = ivr,n,WaitExten(15)
exten = ivr,n,GoTo(restartprompt)
exten = _X,1,Set(choice=${EXTEN})
exten = _X,n,System(echo
'${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date +%s)'
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X,n,other_stuff_I_do
exten = ivradvanced,1,NoOp()
exten =
ivradvanced,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivradvanced,n,NoOp(${BACKGROUNDSTATUS})
exten = ivradvanced,n,WaitExten(15)
exten = ivradvanced,n,GoTo(restartprompt)
exten = _X.,1,Set(choice=${EXTEN})
exten = _X.,n,System(echo
'${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date +%s)'
/var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X.,n,other_stuff_I_do
[another-context]
...
...
Kind regards,
Jonas.
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Re: [asterisk-users] Why does it take several seconds to interpret DTMF-input ?
On 06/11/2013 04:46 PM, Eric Wieling wrote:
The only way to resolve this is to redesign your dialplan so you do not have
ambiguous matching, This is not an Asterisk issue, this is an issue with the
way you designed your dialplan and would apply to any IVR on any system.
I understand that I need to re-design my dialplan logic.
I gave an example of my re-design in my last post. Would that have been
a good re-design ?? Or is it still ambiguous ?
I will post it again :
[my-context]
exten = ivr,1,NoOp()
exten =
ivr,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivr,n,NoOp(${BACKGROUNDSTATUS}) exten = ivr,n,WaitExten(15) exten =
ivr,n,GoTo(restartprompt)
exten = _X,1,Set(choice=${EXTEN})
exten = _X,n,System(echo '${klantID}','IVR','${choice}','','${CHANNEL:4}','$(date
+%s)' /var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X,n,other_stuff_I_do
exten = ivradvanced,1,NoOp()
exten =
ivradvanced,n(restartprompt),Background(/var/lib/asterisk/sounds/vprompts/${KNUMMER}/${ASTPROMPT})
exten = ivradvanced,n,NoOp(${BACKGROUNDSTATUS})
exten = ivradvanced,n,WaitExten(15)
exten = ivradvanced,n,GoTo(restartprompt)
exten = _X.,1,Set(choice=${EXTEN})
exten = _X.,n,System(echo '${klantID}','IVR','${keuzeID}','','${CHANNEL:4}','$(date
+%s)' /var/log/asterisk/loggingAST/${CHANNEL:4}.csv)
exten = _X.,n,other_stuff_I_do
[another-context]
...
...
Kind regards,
Jonas.
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[asterisk-users] Hangup cause 111 after call pickup
Hello, when picking up an incoming call from one ip phone on another ip phone, the call terminates after about 5 to 10 seconds. When reading out the hangup cause variable in the h-extention of the dialplan, the hangup cause seems to be 111. In the dialplan output, you can see that SIP-peer sipacc3 picks up the incoming channel SipAgenT01-1454, and the call is answered. After 7 seconds, the conversation is terminated. /[Jun 6 10:13:15] VERBOSE[21118] pbx.c: [Jun 6 10:13:15] -- Executing [120@sub-pickup:25] Pickup(SIP///sipacc//3-147c, SIP/SipAgenT01-1454@PICKUPMARK) in new stack// //[Jun 6 10:13:15] VERBOSE[20788] app_queue.c: [Jun 6 10:13:15] -- SIP///sipacc3//-147c answered SIP/SipAgenT01-1454// // //[Jun 6 10:13:22] VERBOSE[20788] pbx.c: [Jun 6 10:13:22] -- Executing [h@pbx-routing:3] NoOp(SIP/SipAgenT01-1454, hangup cause = 111) in new stack/ Questions : 1. what can cause a hangup cause 111 ? What is the meaning of hangup cause 111 ? 2. on voip-info.org I read /111 protocol error 500 Server internal error/. How can I fix this ?? Using Asterisk 1.8.12.2 on CentOS. Kind regards, Jonas. -- _ -- 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] CLI flood : requested media update control 26
On 04/02/2013 05:42 PM, Matthew Jordan wrote: On 04/02/2013 06:37 AM, Jonas Kellens wrote: On 04/02/2013 12:50 PM, A J Stiles wrote: (Message re-ordered for readability. The beginning is *not* the right place for your response -- answers come *after* questions, or *between* points.) On Tuesday 02 April 2013, Jonas Kellens wrote: On 04/02/2013 12:35 PM, A J Stiles wrote: On Tuesday 02 April 2013, Jonas Kellens wrote: Hello, any idea why the Asterisk CLI gets flooded by these messages ? How can the SIP peer /vita3/ cause this flood ? First question: What is vita3 ? A hardware SIP phone, a softphone, an ATA or something else? The SIP peer vita3 is a realtime sip peer, installed in a hardware IP-phone (Siemens Gigaset N510 pro). Have you any other Siemens Gigaset N510 pro phones in your setup? Yes there are. But I want to know what these messages on the CLI mean ? The device communicating with Asterisk over SIP channel SIP/vita3-10af had a change in the media source (26 == AST_CONTROL_SRCCHANGE). This occurs when the SSRC in an RTP packet sent by that device changed. When in the middle of a dialling operation, we tend to log out when one of the parties passes information to the other party. In general, this wouldn't flood the CLI, as a party shouldn't be passing much information off to the other parties involved in the dial. I'm not sure why a device in the middle of a 'normal' dialling operation (regardless of it being either the caller/peer) would switch its SSRC rapidly in such a fashion. A pcap should show the changes in SSRC and might illustrate what's occurring. Matt Hello, I don't think it's related to the IP-phone because I notice my Asterisk-server also gets these messages from my SIP-provider. The call goes : IPphone -- Asterisk -- SIP-provider It does not occur always when calling from the same IP-phone. It can be any IP-phone and phone type. It can also occur at any time : when there are few calls and when there are many calls. The negative side when this occurs is that there is no audio when the calls gets answered. These messages flood the CLI untill the call gets answered. Then it stops, but there is no-way-audio. I have a second Asterisk-server (same version : 1.8.12.2) and there I see that this messages occurs just 1 time in a call. Could it be an issue of Asterisk ? Timing issue ? Any idea which issue and how to tune it ? Kind regards, Jonas. -- _ -- 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] erro compiling dahdi
Hello, when compiling dahdi (CentOS 2.6.18-348.3.1.el5) I get the following error : In file included from /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xpd.h:26, from /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/card_bri.c:29: /usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/xdefs.h:152: error: conflicting types for 'bool' include/linux/types.h:36: error: previous declaration of 'bool' was here make[4]: *** [/usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp/card_bri.o] Error 1 make[3]: *** [/usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi/xpp] Error 2 make[2]: *** [_module_/usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux/drivers/dahdi] Error 2 make[2]: Leaving directory `/usr/src/kernels/2.6.18-348.3.1.el5-x86_64' make[1]: *** [modules] Error 2 make[1]: Leaving directory `/usr/src/dahdi-linux-complete-2.6.1+2.6.1/linux' make: *** [all] Error 2 What is wrong ? Kind regards, Jonas. -- _ -- 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] Progress() on outgoing calls
Hello, can you use Progress() in the dialplan for outgoing calls ? For example just before the Dial()-command ? Is there a risk involved when using the Progress()-command ? Kind regards, Jonas. -- _ -- 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] CLI flood : requested media update control 26
Hello, any idea why the Asterisk CLI gets flooded by these messages ? How can the SIP peer /vita3/ cause this flood ? [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:48] VERBOSE[17029] app_dial.c: [Apr 2 11:45:48] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029] app_dial.c: [Apr 2 11:45:49] -- SIP/vita3-10af requested media update control 26, passing it to SIP/708708-10b3 [Apr 2 11:45:49] VERBOSE[17029]
Re: [asterisk-users] CLI flood : requested media update control 26
The SIP peer vita3 is a realtime sip peer, installed in a hardware IP-phone (Siemens Gigaset N510 pro). Jonas. On 04/02/2013 12:35 PM, A J Stiles wrote: On Tuesday 02 April 2013, Jonas Kellens wrote: Hello, any idea why the Asterisk CLI gets flooded by these messages ? How can the SIP peer /vita3/ cause this flood ? First question: What is vita3 ? A hardware SIP phone, a softphone, an ATA or something else? -- _ -- 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] CLI flood : requested media update control 26
On 04/02/2013 12:50 PM, A J Stiles wrote: (Message re-ordered for readability. The beginning is *not* the right place for your response -- answers come *after* questions, or *between* points.) On Tuesday 02 April 2013, Jonas Kellens wrote: On 04/02/2013 12:35 PM, A J Stiles wrote: On Tuesday 02 April 2013, Jonas Kellens wrote: Hello, any idea why the Asterisk CLI gets flooded by these messages ? How can the SIP peer /vita3/ cause this flood ? First question: What is vita3 ? A hardware SIP phone, a softphone, an ATA or something else? The SIP peer vita3 is a realtime sip peer, installed in a hardware IP-phone (Siemens Gigaset N510 pro). Have you any other Siemens Gigaset N510 pro phones in your setup? Yes there are. But I want to know what these messages on the CLI mean ? Jonas. -- _ -- 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 and MEETME_EXIT_CONTEXT
Hello, what is the equivalent parameter of X in the ConfBridge()-command ? How can you exit ConfBridge by pressing a digit ? Concerning MeetMe() : Verbosity is 25 and I still don't see anything on the console or in the logs when pressing '0' (zero). Kind regards, Jonas. On 02/20/2013 03:32 PM, Rusty Newton wrote: - Original Message - From: Jonas Kellensjonas.kell...@telenet.be But nothing happens when pressing 0 (zero). Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type messages going to it. You can also push those to the console and watch what happens when you press zero. On the console be sure to turn up verbosity with core set verbose 5 If you can't tell what is happening, post a pastebin link to the log and point out (via timestamp or otherwise) where you would expect to see the DTMF digit. Maybe someone will be able to take a look. I'd also really recommend using ConfBridge which is newer than MeetMe. If you switch to ConfBridge I'd recommend an upgrade to the latest 1.8.X as well. Please don't top post (https://www.asterisk.org/community/discuss). Also, you didn't pastebin any debug, so I can't confirm that there is not some other issue upon a possible DTMF reception. If it is the case that Asterisk doesn't detect a DTMF 0 when you send it from the endpoint, then you probably want to look at a SIP packet capture to verify the endpoint is actually sending the DTMF to Asterisk. What you look for in the capture or audio will depend on what kind of DTMF you are sending with the endpoint. Does Asterisk detect the digit 0 at any other time outside of MeetMe? Can you setup an extension matching for 1234567890 and dial that? Do you see DTMF debug for all those digits? If you do end up trying ConfBridge - I've never used it in 1.8. Others have made me aware that ConfBridge wasn't the best in 1.8, and that it's much better in 10 or preferably 11. -- Rusty Newton OS Community Support Manager | Digium, Inc. | www.digium.com Office/Cell/Fax: 256-428-6200 Hello, I've tried now from Cisco SPA 508G and from Yealink T-28 to exit Meetme() by pressing '0' (zero) but no success. As I said, to log in I need to give password 12340 and that goes very well ! Once inside the conference room, I can press any digit : nothing happens. Nothing in the logs about DTMF being received. To exit the whole conferencing thing I can press # and that also succeeds ! So I don't think it has anything to do with DTMF-troubles. I've taken a pcap trace on the Yealink T-28. Where can I find the DTMF ? I can filter SIP, but no DTMF. To check if they were well send... Kind regards, Jonas. -- _ -- 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 and MEETME_EXIT_CONTEXT
Hello,
using Asterisk 1.8.12.2
I am having trouble with exiting the conference room by entering a
single digit.
option X of the Meetme()-application should do this.
I have following in extensions.conf :
/exten = _1000X,n,Set(MEETME_EXIT_CONTEXT=dynamic-nway-invite)//
//exten = _1000X,n,MeetMe(${CONFNO},dMX)//
//
//
//[dynamic-nway-invite]//
//exten = 0,1,NoOp(confno = ${CONFNO})//
//exten = 0,n,Read(DEST,dial,,i)//
//exten = 0,n,Set(DYNAMIC_FEATURES=nway-inv#nway-noinv)//
//exten = 0,n,Dial(Local/${DEST}@${LocalContext},,g)//
//exten = 0,n,Set(DYNAMIC_FEATURES=)//
//exten = 0,n,NoOp(tralalala)//
//exten = 0,n,Goto(dynamic-nway1,${CONFNO},1)//
//exten = i,1,Goto(dynamic-nway1,${CONFNO},1)//
/
So by pressing 0 (zero) while in the conference room, I should be able
to exit and continue in the context [dynamic-nway-invite] . Correct ?
But nothing happens when pressing 0 (zero).
What am I missing ??
Kind regards,
Jonas.
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Re: [asterisk-users] Meetme and MEETME_EXIT_CONTEXT
Hello, I don't really see anything when pressing '0' (zero). It's like the '0' (zero) does not reach Asterisk. However the password to enter the conference does reach Asterisk well. Kind regards, Jonas. On 02/20/2013 03:32 PM, Rusty Newton wrote: - Original Message - From: Jonas Kellens jonas.kell...@telenet.be But nothing happens when pressing 0 (zero). Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type messages going to it. You can also push those to the console and watch what happens when you press zero. On the console be sure to turn up verbosity with core set verbose 5 If you can't tell what is happening, post a pastebin link to the log and point out (via timestamp or otherwise) where you would expect to see the DTMF digit. Maybe someone will be able to take a look. I'd also really recommend using ConfBridge which is newer than MeetMe. If you switch to ConfBridge I'd recommend an upgrade to the latest 1.8.X as well. -- _ -- 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 and MEETME_EXIT_CONTEXT
Hello, what is the equivalent parameter of X in the ConfBridge()-command ? How can you exit ConfBridge by pressing a digit ? Concerning MeetMe() : Verbosity is 25 and I still don't see anything on the console or in the logs when pressing '0' (zero). Kind regards, Jonas. On 02/20/2013 03:32 PM, Rusty Newton wrote: - Original Message - From: Jonas Kellens jonas.kell...@telenet.be But nothing happens when pressing 0 (zero). Why not check the logs in /var/log/asterisk/full ?. Make sure you have the full log enabled in logger.conf and that you have VERBOSE,DEBUG,DTMF type messages going to it. You can also push those to the console and watch what happens when you press zero. On the console be sure to turn up verbosity with core set verbose 5 If you can't tell what is happening, post a pastebin link to the log and point out (via timestamp or otherwise) where you would expect to see the DTMF digit. Maybe someone will be able to take a look. I'd also really recommend using ConfBridge which is newer than MeetMe. If you switch to ConfBridge I'd recommend an upgrade to the latest 1.8.X as well. -- _ -- 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] CallerID external call after Attended Transfer
Hello, using Asterisk 1.8.12.2 case : I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the colleague sees on his IP-phone. In step A the colleague sees the CallerID of the receptionist, which I normal. In step B, after I am connected to my colleague, the colleague still sees the CallerID of the receptionist (and not my cellphone number). How come my colleague does not see my cellphone number ? What is the correct setting ( IP-phone ? Asterisk ? ) to obtain this functionality. Thanks. Jonas. -- _ -- 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] CallerID external call after Attended Transfer
Hello, thanks you for your answer. The IP-phones in this case are Yealink T32G. What setting is needed in this IP-phone ? Jonas. On 02/04/2013 02:29 PM, Steven Howes wrote: On 4 Feb 2013, at 12:53, Jonas Kellens wrote: I call with my cellphone to our public telephone number Our receptionist answers the incoming call and does an attended transfer to my colleague ( A ) Colleague answers and the receptionist tells him that I am on the other side. Receptionist transfers the call and I am connected to my colleague ( B ) My question is about the CallerID that the colleague sees on his IP-phone. In step A the colleague sees the CallerID of the receptionist, which I normal. In step B, after I am connected to my colleague, the colleague still sees the CallerID of the receptionist (and not my cellphone number). How come my colleague does not see my cellphone number ? What is the correct setting ( IP-phone ? Asterisk ? ) to obtain this functionality. It's called connected line ID (it sends clid updates when things change). Asterisk supports it in recent versions (i believe 1.8 is sufficient) - your handsets may or may not (their method of transfer, and their ability to process the updates can affect it's workability). Given you've not mentioned your handsets, we cant make that judgement for you. Steve -- _ -- 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] CallerID external call after Attended Transfer
Hello, and is there any setting in Asterisk to turn this functionality on/off ? Maybe mine is not enabled. Jonas On 02/04/2013 03:30 PM, Steven Howes wrote: On 4 Feb 2013, at 13:45, Jonas Kellens wrote: The IP-phones in this case are Yealink T32G. What setting is needed in this IP-phone ? Quick google doesn't turn up any results. Handsets probably dont support it. Steve -- _ -- 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] Execute a script outside Asterisk
Hello, at certain time inside my dialplan I would like to have an external php script executed. Asterisk should not wait for the end of the execution to continue with the rest of the dialplan. It should just start the execution of the php script (which inserts an entry into a remote mysql-DB). What is the best way to work ? - with AGI inside the dialplan ? - with the system()-command inside the dialplan ? Kind regards, Jonas. -- _ -- 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] Execute a script outside Asterisk
Hello, thank you for your answer. The most important here is that Asterisk continues with the rest of the dialplan, in case the database-connection fails or hangs or ... I don't think the System()-command makes this true. Jonas. On 01/23/2013 03:27 PM, Danny Nicholas wrote: I would vote for system() on two accounts. #1 AGI requires more overhead and protocol #2 you are not expecting a result to return to the dialplan. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, January 23, 2013 4:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Execute a script outside Asterisk Hello, at certain time inside my dialplan I would like to have an external php script executed. Asterisk should not wait for the end of the execution to continue with the rest of the dialplan. It should just start the execution of the php script (which inserts an entry into a remote mysql-DB). What is the best way to work ? - with AGI inside the dialplan ? - with the system()-command inside the dialplan ? Kind regards, Jonas. -- _ -- 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] Execute a script outside Asterisk
Hello, will this : Exten = 2,n,playback(vm-goodbye) be executed even when Exten = 2,1,system(Jonas.php) is still executing ?? The exact snippet would be : Exten = s,1,answer() Exten = s,n,system(Jonas.php) ; script that may take a minute Exten = s,n,do something Exten = s,n,Dial(SIP/peer1,,10) ; dial peer 1 Exten = s,n,system(Jonas.php) ; script that may take a minute Exten = s,n,do something Exten = s,n,Dial(SIP/peer2,,10) ; dial peer 2 Exten = s,n,system(Jonas.php) ; script that may take a minute Exten = s,n,do something Exten = s,n,Dial(SIP/peer3,,10) ; dial peer 3 Exten = s,n,hangup() The peer MUST be dialed even if the script Jonas.php is still running. Jonas. On 01/23/2013 03:44 PM, Danny Nicholas wrote: Let's assume you're using this snippet [default] Exten = s,1,answer() Exten = s,n,playback(tt-monkeys) Exten = s,n,waitexten(6) Exten = s,n,hangup() Exten = 1,1,AGI(Jonas.php) Exten = 1,n,playback(vm-goodbye) Exten = 1,n,hangup() Exten = 2,1,system(Jonas.php) Exten = 2,n,playback(vm-goodbye) Exten = 2,n,hangup() Both of these do the exact same thing -- pick up the line, play tt-monkeys, run Jonas.php if you press 1 or 2, play vm-goodbye and hangup. The failure of Jonas.php due to database or any other problem would not affect the execution of the dialplan. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, January 23, 2013 8:32 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Execute a script outside Asterisk Hello, thank you for your answer. The most important here is that Asterisk continues with the rest of the dialplan, in case the database-connection fails or hangs or ... I don't think the System()-command makes this true. Jonas. On 01/23/2013 03:27 PM, Danny Nicholas wrote: I would vote for system() on two accounts. #1 AGI requires more overhead and protocol #2 you are not expecting a result to return to the dialplan. *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, January 23, 2013 4:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Execute a script outside Asterisk Hello, at certain time inside my dialplan I would like to have an external php script executed. Asterisk should not wait for the end of the execution to continue with the rest of the dialplan. It should just start the execution of the php script (which inserts an entry into a remote mysql-DB). What is the best way to work ? - with AGI inside the dialplan ? - with the system()-command inside the dialplan ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided byhttp://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] Execute a script outside Asterisk
Hello, the behind the command to execute in the background is a great idea ! Jonas. On 01/23/2013 04:29 PM, Danny Nicholas wrote: Here is the way I got it to do what I think you want. '1250' = 1. answer() [pbx_config] 2. setMusiconhold(jazz) [pbx_config] 3. AGI(wait10.sh) [pbx_config] 4. playback(vm-goodbye) [pbx_config] 5. setMusiconhold(monkey) [pbx_config] 6. system(/var/lib/asterisk/agi-bin/wait10.sh ) [pbx_config] 7. playback(vm-goodbye) [pbx_config] 8. hangup() [pbx_config] Without the , AGI and system both execute and wait for completion of wait10.sh. with the , the system command returns control to the dialplan immediately. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, January 23, 2013 8:54 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Execute a script outside Asterisk Hello, will this : Exten = 2,n,playback(vm-goodbye) be executed even when Exten = 2,1,system(Jonas.php) is still executing ?? The exact snippet would be : Exten = s,1,answer() Exten = s,n,system(Jonas.php) ; script that may take a minute Exten = s,n,do something Exten = s,n,Dial(SIP/peer1,,10) ; dial peer 1 Exten = s,n,system(Jonas.php) ; script that may take a minute Exten = s,n,do something Exten = s,n,Dial(SIP/peer2,,10) ; dial peer 2 Exten = s,n,system(Jonas.php) ; script that may take a minute Exten = s,n,do something Exten = s,n,Dial(SIP/peer3,,10) ; dial peer 3 Exten = s,n,hangup() The peer MUST be dialed even if the script Jonas.php is still running. Jonas. On 01/23/2013 03:44 PM, Danny Nicholas wrote: Let's assume you're using this snippet [default] Exten = s,1,answer() Exten = s,n,playback(tt-monkeys) Exten = s,n,waitexten(6) Exten = s,n,hangup() Exten = 1,1,AGI(Jonas.php) Exten = 1,n,playback(vm-goodbye) Exten = 1,n,hangup() Exten = 2,1,system(Jonas.php) Exten = 2,n,playback(vm-goodbye) Exten = 2,n,hangup() Both of these do the exact same thing -- pick up the line, play tt-monkeys, run Jonas.php if you press 1 or 2, play vm-goodbye and hangup. The failure of Jonas.php due to database or any other problem would not affect the execution of the dialplan. *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, January 23, 2013 8:32 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Execute a script outside Asterisk Hello, thank you for your answer. The most important here is that Asterisk continues with the rest of the dialplan, in case the database-connection fails or hangs or ... I don't think the System()-command makes this true. Jonas. On 01/23/2013 03:27 PM, Danny Nicholas wrote: I would vote for system() on two accounts. #1 AGI requires more overhead and protocol #2 you are not expecting a result to return to the dialplan. *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Wednesday, January 23, 2013 4:10 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Execute a script outside Asterisk Hello, at certain time inside my dialplan I would like to have an external php script executed. Asterisk should not wait for the end of the execution to continue with the rest of the dialplan. It should just start the execution of the php script (which inserts an entry into a remote mysql-DB). What is the best way to work ? - with AGI inside the dialplan ? - with the system()-command inside the dialplan ? Kind regards, Jonas. -- _ -- Bandwidth and Colocation Provided byhttp://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 byhttp://www.api-digital.com -- New to Asterisk? Join us
[asterisk-users] param sayduration of mailbox
Hello, what exactly is the function of the parameter 'sayduration' in the voicemail box configuration ? Whether I put this to 'yes' or to 'no', nothing changes. I do not get the announcement of duration at the beginning of the voicemail message. Kind regards, Jonas. -- _ -- 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] Set Language for VoiceMailMain
Hello, how do I set the language for the VoiceMailMain()-command ? How do I set the language per voicemail-box ? Thanks, Jonas. -- _ -- 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] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of mailbox
and VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted
to answer the phone in English, then do voicemails in different
languages, this should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) -- redundant since default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Friday, January 11, 2013 5:36 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
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Re: [asterisk-users] Set Language for VoiceMailMain
Hello,
are you sure that the language-parameter of the SIP peer will
influence the language used by VoiceMailMain() ?
Jonas.
On 01/11/2013 04:07 PM, Danny Nicholas wrote:
No. It is purposely set from the dialplan. In Asterisk 11.X you have
the [zonemessage] section in voicemail.conf that could probably be
tweaked to change the language without dialplan changes. Also in
sip.conf you can set language by peer so you could have something like
[London]
Type = peer
Language=en
[Madrid]
Type=peer
Language=es
[paris]
Type=peer
Language=fr
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Friday, January 11, 2013 9:00 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of
mailbox and VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you
wanted to answer the phone in English, then do voicemails in
different languages, this should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) -- redundant since default
is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Jonas Kellens
*Sent:* Friday, January 11, 2013 5:36 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
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Re: [asterisk-users] Set Language for VoiceMailMain
Well, I thought you had tried it and thus could tell it with 100% certainty.
Thanks for your help.
Jonas.
On 01/11/2013 04:16 PM, Danny Nicholas wrote:
Since the peer language sets CHANNEL(language), I can say yes with
reasonable certainly. Like anything else here, you don't really know
until you try it on your box.
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Friday, January 11, 2013 9:15 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Set Language for VoiceMailMain
Hello,
are you sure that the language-parameter of the SIP peer will
influence the language used by VoiceMailMain() ?
Jonas.
On 01/11/2013 04:07 PM, Danny Nicholas wrote:
No. It is purposely set from the dialplan. In Asterisk 11.X you
have the [zonemessage] section in voicemail.conf that could
probably be tweaked to change the language without dialplan
changes. Also in sip.conf you can set language by peer so you
could have something like
[London]
Type = peer
Language=en
[Madrid]
Type=peer
Language=es
[paris]
Type=peer
Language=fr
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Jonas Kellens
*Sent:* Friday, January 11, 2013 9:00 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Set Language for VoiceMailMain
Thanks you for your answer.
There is no language-parameter that can define the language of
mailbox and VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you
wanted to answer the phone in English, then do voicemails in
different languages, this should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) -- redundant since
default is en
Exten = s,n,background(welcome)
; prompt for voicemail in French
Exten = s,n,Set(CHANNEL(language)=fr)
Exten = s,n,VoiceMailMain(100@default)
; Line 200 is in Spain
Exten = s,n,Set(CHANNEL(language=es)
Exten = s,n,VoiceMailMain(200@default)
*From:*asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of
*Jonas Kellens
*Sent:* Friday, January 11, 2013 5:36 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Set Language for VoiceMailMain
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
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Re: [asterisk-users] Queue joinempty, even after AddQueueMember
Hello, using Asterisk 1.8.12.2 Jonas. On 09-12-12 09:15, Jonathan Rose wrote: I was poking around with the Add/Remove QueueMember code a while back. From the sound of what you are saying I might have just missed something critical. for your case. It'd be good to know what version you are using so that I can verify whether or not my changes could have affected you. - Original Message - From: Jonas Kellens jonas.kell...@telenet.be To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 8, 2012 5:55:39 AM Subject: [asterisk-users] Queue joinempty, even after AddQueueMember Hello, I add a member to a queue with AddQueueMember, but the Queue still indicates joinempty : Add member to queue : -- Executing [queueadd@sub-GetParams:2] AddQueueMember(SIP/sip17-5c1e, myqueue11,member3) in new stack -- Executing [queueadd@sub-GetParams:3] NoOp(SIP/sip17-5c1e, AQMSTATUS = ADDED) in new stack ... but JOINEMPTY when entering the Call Queue : -- Executing [queue@pbx-routing:4] Queue(SIP/SipIncoming-5da9, myqueue1160) in new stack -- Executing [queue@pbx-routing:5] NoOp(SIP/SipIncoming-5da9, queuestatus == JOINEMPTY) in new stack How is this possible ? Kind regards, Jonas. -- _ -- 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] Queue joinempty, even after AddQueueMember
it might work... How come app_queue is suddenly so unstable ? Which version has a stable app_queue ? I thought unstable versions are released with rc- added ? Kind regards, Jonas. On 09-12-12 19:19, Jonathan Rose wrote: Jonas Kellens wrote: Hello, using Asterisk 1.8.12.2 I think that was tagged before any of my recent app_queue patches. In that case, it might work if you just update to the latest 1.8 release. If it doesn't, go ahead and file an issue on JIRA. -- Jonathan R. Rose Digium, Inc. | Software Engineer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct +1 256 428 6139 Check us out at: http://digium.com http://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
Re: [asterisk-users] Queue joinempty, even after AddQueueMember
On 09-12-12 19:49, Joshua Colp wrote: Jonas Kellens wrote: it might work... Without labbing things up with your exact scenario Jonathan can't confirm it. I did a quick search of the issue tracker for anything open similar to the issue you specified and nothing came up. The functionality you are using is commonly used so either it's something specific to how you are using it or was an issue in the version you are using and is not in recent versions. As well - if the log you provided has not been altered then you are attempting to add an interface member3 to the queue. While this will succeed it is ultimately not a valid interface and would not be considered as available. This would explain why it does not work. Hello, what is then a correct interface ? SIP/member3 maybe is more correct ? Thanks for your help. Jonas. -- _ -- 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] Queue joinempty, even after AddQueueMember
On 09-12-12 20:10, Joshua Colp wrote: Jonas Kellens wrote: On 09-12-12 19:49, Joshua Colp wrote: As well - if the log you provided has not been altered then you are attempting to add an interface member3 to the queue. While this will succeed it is ultimately not a valid interface and would not be considered as available. This would explain why it does not work. Hello, Hola, what is then a correct interface ? SIP/member3 maybe is more correct ? That is correct. That type of string is what interface refers to in the AddQueueMember documentation. SIP/member3, IAX2/joe, etc. Cheers, Hello, I will try that. It might be the solution... Jonas. -- _ -- 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 joinempty, even after AddQueueMember
Hello, I add a member to a queue with AddQueueMember, but the Queue still indicates joinempty : Add member to queue : /-- Executing [queueadd@sub-GetParams:2] AddQueueMember(SIP/sip17-5c1e, myqueue11,member3) in new stack -- Executing [queueadd@sub-GetParams:3] NoOp(SIP/sip17-5c1e, AQMSTATUS = ADDED) in new stack/ ... but JOINEMPTY when entering the Call Queue : /-- Executing [queue@pbx-routing:4] Queue(SIP/SipIncoming-5da9, myqueue1160) in new stack -- Executing [queue@pbx-routing:5] NoOp(SIP/SipIncoming-5da9, queuestatus == JOINEMPTY) in new stack/ How is this possible ? Kind regards, Jonas. -- _ -- 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 logging
Hello, at the moment I am logging queues into a MySQL DB, but this can quickly become a lot of information. Is there a way to exclude certain queues from being logged into the queue log ? Thanks, Jonas. -- _ -- 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] Queue logging
Hello, I am not using triggering (what is this ?). Just using extconfig.conf Asterisk 1.8.12.2 Kind regards, Jonas. On 27-11-12 17:28, Danny Nicholas wrote: Are you using triggering? If so, perhaps you could modify the trigger values. PS asterisk version? *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, November 27, 2012 10:21 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Queue logging Hello, at the moment I am logging queues into a MySQL DB, but this can quickly become a lot of information. Is there a way to exclude certain queues from being logged into the queue log ? Thanks, Jonas. -- _ -- 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] Queue logging
Ah OK, that triggering I know. I though maybe there was some kind of setting on a per queue base that could control the logging, like there is amaflags on a peer. Jonas. On 27-11-12 20:53, Danny Nicholas wrote: Triggering is a MYSQL mechanism that forces database action on specified conditions. My best guess is that you would have to tweak addons/res_config_mysql.c to be able to filter logs. It would probably be easier to write a daemon to clear the unwanted data on a periodic basis. *From:*asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, November 27, 2012 12:27 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Queue logging Hello, I am not using triggering (what is this ?). Just using extconfig.conf Asterisk 1.8.12.2 Kind regards, Jonas. On 27-11-12 17:28, Danny Nicholas wrote: Are you using triggering? If so, perhaps you could modify the trigger values. PS asterisk version? *From:*asterisk-users-boun...@lists.digium.com mailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas Kellens *Sent:* Tuesday, November 27, 2012 10:21 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Queue logging Hello, at the moment I am logging queues into a MySQL DB, but this can quickly become a lot of information. Is there a way to exclude certain queues from being logged into the queue log ? Thanks, Jonas. -- _ -- Bandwidth and Colocation Provided byhttp://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] What exactly does hangupcause 111 mean ?
Hello, what exactly does hangupcause 111 mean ? I read on the wiki : 111 protocol error 500 Server internal error Is the the SIP response that was received form the other end ? Or is this an internal server (Asterisk) error ? Kind regards, Jonas. -- _ -- 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
