Re: [asterisk-users] Error in ubuntu dapper
El vie, 21-07-2006 a las 18:53 -0400, Russell Bryant escribió: On Fri, 2006-07-21 at 12:37 -0400, don Paolo Benvenuto wrote: Jul 21 12:31:51 WARNING[6333]: chan_sip.c:12637 reload_config: Failed to bind to 10.152.58.9:5060: Address already in use It looks like another application on your system is using port 5060. Did you install any new software such as a soft phone? If you are now using another application that wants to use port 5060, you will need to configure one of them to use a different port. But, at this point, is the issue worth a bug? I think asterisk should detect the unavailability of the port, and stop with an error message. What can an asterisk running this way help? -- Buon Cammino! don Paolo Benvenuto Vuoi sapere di più su quello che succede qui? leggi il mio diario a http://www.chiesamissionaria.it/diario Visita l'enciclopedia libera, dove puoi contribuire anche tu: http://it.wikipedia.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error in ubuntu dapper
) to 213.91.9.213:5060 returned -1: Bad file descriptor Jul 21 12:31:52 WARNING[6333]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x81607f0 (len 467) to 83.143.18.16:5060 returned -1: Bad file descriptor Jul 21 12:31:52 WARNING[6333]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x8163510 (len 471) to 83.138.130.145:5060 returned -1: Bad file descriptor Jul 21 12:31:53 WARNING[6333]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x815d4a0 (len 527) to 10.152.58.8:5065 returned -1: Bad file descriptor Jul 21 12:31:53 WARNING[6333]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x814c6f8 (len 477) to 10.152.58.18:5061 returned -1: Bad file descriptor Jul 21 12:31:53 WARNING[6333]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x8144358 (len 495) to 10.152.58.17:5060 returned -1: Bad file descriptor Jul 21 12:31:53 WARNING[6333]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x814c380 (len 497) to 10.152.58.27:5061 returned -1: Bad file descriptor Jul 21 12:31:53 WARNING[6333]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x81591f0 (len 491) to 10.152.58.27:5060 returned -1: Bad file descriptor Jul 21 12:31:53 WARNING[6333]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x8165738 (len 475) to 10.152.58.17:5061 returned -1: Bad file descriptor Jul 21 12:31:53 WARNING[6333]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x8164058 (len 499) to 10.152.58.26:5061 returned -1: Bad file descriptor Jul 21 12:31:53 WARNING[6333]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x8161338 (len 503) to 10.152.58.26:5060 returned -1: Bad file descriptor Jul 21 12:31:53 WARNING[6333]: chan_sip.c:1066 __sip_xmit: sip_xmit of 0x815fca8 (len 489) to 10.152.58.16:5060 returned -1: Bad file descriptor 10.152.58.15,16,17 are the pap2 10.152.58.26,27 are sipura spa3000 10.152.58.9 is the IP of the server I trying putting bindaddress=0.0.0.0 in sip.conf, but nothing changed. Besides that, I saw that asterisk couldn't reach the pap2's nor the sipura's, while they were pingable and running normally, like the server. What is happening??? -- Buon Cammino! don Paolo Benvenuto Vuoi sapere di più su quello che succede qui? leggi il mio diario a http://www.chiesamissionaria.it/diario Visita l'enciclopedia libera, dove puoi contribuire anche tu: http://it.wikipedia.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error in ubuntu dapper
Yeah, ekiga was using port 5060, althoug netstat -a didn't say it. I might issue netstat -alpn The weird thing is that I had configured ekiga so that it used port 5061, but unfortunatly if ekiga is run before asterisk it catchs port 5060 too. El vie, 21-07-2006 a las 18:13 -0500, brandon kruz escribió: in addition to russel use (in ubuntu) sudo netstat or man netstat for further, more precise methods look for your specific port eg sudo netstat -a | grep 5060 and it shoudl tell you the process name, and what directory it is comming from shut it off and do that sudo netstat -a | grep 5060 again it should be clear then start asterisk :] keep us updated. the fact that it is from module chan_sip, as russel said i believe its a sip phone indeed(softphone, X-lite, SJ-phone, etc) On Fri, 2006-07-21 at 12:37 -0400, don Paolo Benvenuto wrote: Jul 21 12:31:51 WARNING[6333]: chan_sip.c:12637 reload_config: Failed to bind to 10.152.58.9:5060: Address already in use It looks like another application on your system is using port 5060. Did you install any new software such as a soft phone? If you are now using another application that wants to use port 5060, you will need to configure one of them to use a different port. -- Buon Cammino! don Paolo Benvenuto Vuoi sapere di più su quello che succede qui? leggi il mio diario a http://www.chiesamissionaria.it/diario Visita l'enciclopedia libera, dove puoi contribuire anche tu: http://it.wikipedia.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transferring calls from ekiga to asterisk
I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip debug, I can see that the call arrives to asterisk from [EMAIL PROTECTED] (skypho provider) via something containing my external IP address, and asterisk tries to communicate with a host on my external IP address, obviously unsuccessfully, and in ekiga I get a occupied tone. Note that in ekiga I have an account which is in sip.conf, and ekiga registers without problems with that account to my asterisk server. However, the problem I have is how to transfer to asterisk a call which is managed with another account, specifically a external voip provider account: the call arrives to asterisk with the data of that external voip provider. Anyone could help me? Thank you! -- Buon Cammino! don Paolo Benvenuto Vuoi sapere di più su quello che succede qui? leggi il mio diario a http://www.chiesamissionaria.it/diario Visita l'enciclopedia libera, dove puoi contribuire anche tu: http://it.wikipedia.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] transferring calls from ekiga to asterisk
El mar, 13-06-2006 a las 07:33 +, [EMAIL PROTECTED] escribió: When you configured the incoming line in sip.conf, you gave it a context. I think my problem is: how do I configure sip.conf in order to receive those call redirects? In Twinklephone I have two accounts: - an account in which twinkle is a peer of asterisk, which has this settings in sip.conf: [pablopctwinkle] type=friend secret=xx callerid=Pablo PC Twinklephone 619 host=dynamic context=todo nat=no qualify=yes twinkle registers with asterisk without problems with these settings. It sends and receives calls, it's a normal asterisk's extension - another with the voip provider (voip.eutelia.it) This account is the one that receives the calls and redirects it to asterisk. I want to transfer a call from this account to asterisk. That's equivalent, I think, to connecting to asterisk from that account. 196.3.84.214 my routers external address 5062 is the port that twinklephone uses 10.152.58.1=misiongenovesa is the server asterisk and twinklephone are running on 0108937227 is my username with voip provider voip.eutelia.it I must configure sip.conf and extensions.conf in order to receive calls from that account. If I try to call asterisk from that account I get in asterisk's console (sip debug): ---BEGIN- -- SIP read from 10.152.58.1:5062: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm Max-Forwards: 70 To: sip:[EMAIL PROTECTED] From: don Paolo Benvenuto sip:[EMAIL PROTECTED];tag=cfpll Call-ID: [EMAIL PROTECTED] CSeq: 979 INVITE Contact: sip:[EMAIL PROTECTED]:5062 Content-Type: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, REFER, NOTIFY, SUBSCRIBE Supported: 100rel User-Agent: Twinkle/0.7.1 Content-Length: 311 v=0 o=0108937227 1395986944 491937694 IN IP4 196.3.84.214 s=- c=IN IP4 196.3.84.214 t=0 0 m=audio 8000 RTP/AVP 3 98 97 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:98 speex/16000 a=rtpmap:97 speex/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 --- (13 headers 14 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 196.3.84.214 : 5062 (NAT) Found peer 'pablopctwinkle' Reliably Transmitting (no NAT) to 196.3.84.214:5062: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1 From: don Paolo Benvenuto sip:[EMAIL PROTECTED];tag=cfpll To: sip:[EMAIL PROTECTED];tag=as111cfbda Call-ID: [EMAIL PROTECTED] CSeq: 979 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=78099a8f Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms Retransmitting #1 (no NAT) to 196.3.84.214:5062: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1 From: don Paolo Benvenuto sip:[EMAIL PROTECTED];tag=cfpll To: sip:[EMAIL PROTECTED];tag=as111cfbda Call-ID: [EMAIL PROTECTED] CSeq: 979 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=78099a8f Content-Length: 0 --- Retransmitting #2 (no NAT) to 196.3.84.214:5062: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1 From: don Paolo Benvenuto sip:[EMAIL PROTECTED];tag=cfpll To: sip:[EMAIL PROTECTED];tag=as111cfbda Call-ID: [EMAIL PROTECTED] CSeq: 979 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=78099a8f Content-Length: 0 --- Retransmitting #3 (no NAT) to 196.3.84.214:5062: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1 From: don Paolo Benvenuto sip:[EMAIL PROTECTED];tag=cfpll To: sip:[EMAIL PROTECTED];tag=as111cfbda Call-ID: [EMAIL PROTECTED] CSeq: 979 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=78099a8f Content-Length: 0 --- Retransmitting #4 (no NAT) to 196.3.84.214:5062: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 196.3.84.214:5062;rport;branch=z9hG4bKqvdrjtsm;received=10.152.58.1 From: don Paolo Benvenuto sip:[EMAIL PROTECTED];tag=cfpll To: sip:[EMAIL PROTECTED];tag=as111cfbda Call-ID: [EMAIL PROTECTED] CSeq: 979 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=78099a8f Content-Length: 0 --- Retransmitting #5 (no NAT) to 196.3.84.214:5062: SIP/2.0 407
[Asterisk-Users] transferring calls from ekiga to asterisk
I have ekiga registering to a voip provider (skypho) and receiving external call through the stun server. I want to redirect inconditionally all these calls to my asterisk server, but I can't understand how and what should I configure in asterisk in order to accept the redirected call. In asterisk console I can't see nothing when ekiga passes the call. If I turn asterisk's sip debug, I can see that the call arrives to asterisk from [EMAIL PROTECTED] (skypho provider) via something containing my external IP address, and asterisk tries to communicate with a host on my external IP address, obviously unsuccessfully, and in ekiga I get a occupied tone. Note that in ekiga I have an account which is in sip.conf, and ekiga registers without problems with that account to my asterisk server. However, the problem I have is how to transfer to asterisk a call which is managed with another account, specifically a external voip provider account: the call arrives to asterisk with the data of that external voip provider. Anyone could help me? Thank you! -- Buon Cammino! don Paolo Benvenuto Vuoi sapere di più su quello che succede qui? leggi il mio diario a http://www.chiesamissionaria.it/diario Visita l'enciclopedia libera, dove puoi contribuire anche tu: http://it.wikipedia.org/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
