Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 04/20/2015 12:31 PM, akhilesh chand wrote: Hi Folks, I'm trying to register softphone(X-lite) but I'm not able to register softphone whenever I'm trying to register softphone I got below error Inline image 1 Is there any document/guide line where I will get process to register softphone in asterisk(Which is installed in EC2 Cloud). Don't make it to complicated ... Connect to your Asterisk via ssh and run asterisk -rvv. Then let your Phone try to register. Asterisk should show you what's getting wrong. If you can't see anything while trying to register, shutdown your firewall and try it again ... Regards Guenther - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -BEGIN PGP SIGNATURE- Version: GnuPG v2 iQIcBAEBAgAGBQJVNPKpAAoJENexF5oIz3BCP84QAOOwrZRJn27PuGqVYT643pqZ oUSIscdJfrDBcWImIRRWNZc1nbKN8hYpXfVgAvdWYaOzcdqeMS95aLiEN6sX+zni A2aAOMajt5bajam5MYRR6yxo87scEr0ku0uBlAH1IPOJ4Ikv2BSfBB7lUjlh6fW/ tLQYeUdYPq82isc7PXJLNGzdHsZzyaYSfMBeKaSRUIexFShtLfCoFRJtCPgf5fav SRNwpTNmQbYz9/4Fs3oZAn0kDQrEI6LSyQDDxBDSVaCJBTuWwJ2ND+gsYLXaIIUZ iemwN03QMNDDeYhWY5IunvPsmNBw1AbpIH74FzNGuhdrRMlAkiwmOL44WsdF5e++ /7fpROUTmt72+Y/O/RT9rSN25RNo+Bzo9hcts0gRw0IRw+lnh2jq63JCrosNhuC+ Zf1lBX4TEiPnG5n65ipzPAyxl7L9r5AtrTApkKjnNz9E1BfQ7oK3zeUEfIhhgFJs W7t8EveORN8AyTvgoa2C6GW6vpY4lILiJ0xjxffVYSQkjkdocMws5HhwfJG5unyA n7rdJjlFSVe1Dp9QSgzKXpSA4WBBbloq40ZZpToUw9id7zZWy5DMp36KxilbyOMd bqFjCEEwQsCZadnRQzWTOGqDYfa2evg3KwdKGEZVVGEZ4xG1nokTRSZ6PqHVujve xQ7hKnPa+icYn1MUUhpC =o6f4 -END PGP SIGNATURE- -- _ -- 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] Issues with call dropping
Hi guys, have really annoying problem with trunks when I calling over voip provider.. after awhile provider sends INFO packages but for some reason Asterisk doesn’t answer on it. after 8 packagers provider just drops the call, here is the package --- Received SIP request (555 bytes) from UDP:192.168.53.1:5060 --- INFO sip:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9@192.168.53.9:5060 SIP/2.0 Max-Forwards: 69 To: sip:4959810128@192.168.53.9;tag=b3769af4-118b-4467-8c95-042247ff1776 From: sip:84957774888@192.168.53.1;tag=3638518512-132845 Call-ID: cfe34652-14c2-4072-9dea-f0b0c30cb15e CSeq: 2 INFO Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Via: SIP/2.0/UDP 192.168.53.1:5060;branch=z9hG4bK8da58cede20b91eb54dec15ad27f866c Contact: sip:84957774888@192.168.53.1:5060 Content-Length: 0 192.168.53.1 - operator IP 192.168.53.9 - asterisk IP Any idea how to fix this? have 2 Ethernet interfaces: 192.168.1.4 - local network 192.168.53.9 - VOIP Provider network Im using PJSIP, here is config: [udp] type=transport protocol=udp bind=192.168.1.4 local_net=10.0.0.0/24 local_net=10.0.1.0/24 local_net=192.168.1.0/24 external_media_address=195.239.8.122 external_signaling_address=195.239.8.122 [udp_B] type=transport protocol=udp bind=192.168.53.9 [1] type=endpoint aors=1 context=dialmap disallow=all allow=alaw,ulaw transport=udp_B [1] type=aor contact=sip:192.168.53.1:5060 max_contacts=4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Am 20.04.15 um 09:43 schrieb akhilesh chand: Hi Thomas, Hello. Yes I'm able to access asterisk server but there is no logs capture into log file related to softphone.If you want more information regarding configuration means sip.conf and extension.conf I will share. Could you increase the verbose level? # core set verbose 6 # sip set debug on Looking for blocking Firewall Rules is also a valid point. cheers t. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
On Mon, Apr 20, 2015 at 1:58 AM, akhilesh chand omakhileshch...@gmail.com wrote: Chain INPUT (policy ACCEPT) target prot opt source destination ACCEPT all -- 0.0.0.0/00.0.0.0/0state RELATED,ESTABLISHED ACCEPT icmp -- 0.0.0.0/00.0.0.0/0 ACCEPT all -- 0.0.0.0/00.0.0.0/0 ACCEPT tcp -- 0.0.0.0/00.0.0.0/0state NEW tcp dpt:22 REJECT all -- 0.0.0.0/00.0.0.0/0reject-with icmp-host-prohibited ACCEPT udp -- 0.0.0.0/00.0.0.0/0udp spt:5060 ACCEPT udp -- 0.0.0.0/00.0.0.0/0udp spt:5083 ACCEPT udp -- 0.0.0.0/00.0.0.0/0udp spt:1 It looks like youre REJECT rule is getting hit before the accept rules for asterisk. Try moving the REJECT rule to last in the list. I think your firewall is blocking asterisk. --Greg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Hi Thomas, Yes I'm able to access asterisk server but there is no logs capture into log file related to softphone.If you want more information regarding configuration means sip.conf and extension.conf I will share. Regards Akhilesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Hi Karthik, Asterisk is running the output of above command is given below Active Internet connections (only servers) Proto Recv-Q Send-Q Local Address Foreign Address State PID/Program name udp0 0 0.0.0.0:50000.0.0.0:* 10340/asterisk udp0 0 0.0.0.0:45200.0.0.0:* 10340/asterisk udp0 0 0.0.0.0:50600.0.0.0:* 10340/asterisk udp0 0 0.0.0.0:45690.0.0.0:* 10340/asterisk On Mon, Apr 20, 2015 at 1:11 PM, Karthik Kondapaneni karthik.kondapan...@gmail.com wrote: Check if asterisk is running or not first . If asterisk is running check iptables ( firewall ) might be blocking the connection . You can see listening ports with netstat -uplncommand On Mon, Apr 20, 2015 at 10:01 AM, akhilesh chand omakhileshch...@gmail.com wrote: Hi Folks, I'm trying to register softphone(X-lite) but I'm not able to register softphone whenever I'm trying to register softphone I got below error [image: Inline image 1] Is there any document/guide line where I will get process to register softphone in asterisk(Which is installed in EC2 Cloud). Regards Akhilesh -- _ -- 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] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Hi Thomas, I followed your recommended command in asterisk CLI which is mentioned in above chain mail but I'm not able capture any log related to softphone. Chain INPUT (policy ACCEPT) target prot opt source destination ACCEPT all -- 0.0.0.0/00.0.0.0/0state RELATED,ESTABLISHED ACCEPT icmp -- 0.0.0.0/00.0.0.0/0 ACCEPT all -- 0.0.0.0/00.0.0.0/0 ACCEPT tcp -- 0.0.0.0/00.0.0.0/0state NEW tcp dpt:22 REJECT all -- 0.0.0.0/00.0.0.0/0reject-with icmp-host-prohibited ACCEPT udp -- 0.0.0.0/00.0.0.0/0udp spt:5060 ACCEPT udp -- 0.0.0.0/00.0.0.0/0udp spt:5083 ACCEPT udp -- 0.0.0.0/00.0.0.0/0udp spt:1 Chain FORWARD (policy ACCEPT) target prot opt source destination REJECT all -- 0.0.0.0/00.0.0.0/0reject-with icmp-host-prohibited Chain OUTPUT (policy ACCEPT) target prot opt source destination On Mon, Apr 20, 2015 at 1:16 PM, Thomas Stein himbe...@meine-oma.de wrote: Am 20.04.15 um 09:43 schrieb akhilesh chand: Hi Thomas, Hello. Yes I'm able to access asterisk server but there is no logs capture into log file related to softphone.If you want more information regarding configuration means sip.conf and extension.conf I will share. Could you increase the verbose level? # core set verbose 6 # sip set debug on Looking for blocking Firewall Rules is also a valid point. cheers t. -- _ -- 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] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Am 20.04.15 um 06:31 schrieb akhilesh chand: Hi Folks, I'm trying to register softphone(X-lite) but I'm not able to register softphone whenever I'm trying to register softphone I got below error Do you have access to the asterisk server? If so, what's in the logs? cheers t. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Check if asterisk is running or not first . If asterisk is running check iptables ( firewall ) might be blocking the connection . You can see listening ports with netstat -uplncommand On Mon, Apr 20, 2015 at 10:01 AM, akhilesh chand omakhileshch...@gmail.com wrote: Hi Folks, I'm trying to register softphone(X-lite) but I'm not able to register softphone whenever I'm trying to register softphone I got below error [image: Inline image 1] Is there any document/guide line where I will get process to register softphone in asterisk(Which is installed in EC2 Cloud). Regards Akhilesh -- _ -- 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] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Am 20.04.15 um 09:58 schrieb akhilesh chand: Hi Thomas, I followed your recommended command in asterisk CLI which is mentioned in above chain mail but I'm not able capture any log related to softphone. Well then i guess you have the softphone somehow misconfigured. You have to double check your settings. IP / Port / Protokoll and so on. best regards t. Chain INPUT (policy ACCEPT) target prot opt source destination ACCEPT all -- 0.0.0.0/00.0.0.0/0state RELATED,ESTABLISHED ACCEPT icmp -- 0.0.0.0/00.0.0.0/0 ACCEPT all -- 0.0.0.0/00.0.0.0/0 ACCEPT tcp -- 0.0.0.0/00.0.0.0/0state NEW tcp dpt:22 REJECT all -- 0.0.0.0/00.0.0.0/0reject-with icmp-host-prohibited ACCEPT udp -- 0.0.0.0/00.0.0.0/0udp spt:5060 ACCEPT udp -- 0.0.0.0/00.0.0.0/0udp spt:5083 ACCEPT udp -- 0.0.0.0/00.0.0.0/0udp spt:1 Chain FORWARD (policy ACCEPT) target prot opt source destination REJECT all -- 0.0.0.0/00.0.0.0/0reject-with icmp-host-prohibited Chain OUTPUT (policy ACCEPT) target prot opt source destination On Mon, Apr 20, 2015 at 1:16 PM, Thomas Stein himbe...@meine-oma.de wrote: Am 20.04.15 um 09:43 schrieb akhilesh chand: Hi Thomas, Hello. Yes I'm able to access asterisk server but there is no logs capture into log file related to softphone.If you want more information regarding configuration means sip.conf and extension.conf I will share. Could you increase the verbose level? # core set verbose 6 # sip set debug on Looking for blocking Firewall Rules is also a valid point. cheers t. -- _ -- 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] Kamallio registration
On 20 April 2015 at 16:02, Andrew Colin and...@convergedgroup.net wrote: Hi Guys Is it possible to register Kamallio directly to our SIP provider then load balance the RTP to 2 asterisk servers? We cant do the registration from the asterisk boxes as we want to do it directly from Kamallio. Is this possible? Yes, it would be possible. You can use the Kamailio UAC module ( http://kamailio.org/docs/modules/stable/modules/uac.html) to register with your remote peer, then when INVITES come in from that peer load balance them between your Asterisk boxes. It would be a more simple set up if you allow the Asterisk boxes to send RTP directly to the remote peer, using Kamalio just for SIP signalling and load balancing. If you need to proxy the RTP then you'll have to look at one of the RTP proxy Kamailio modules as well. Probably a better question for the Kamailio mailing list though. Hope this helps. -Barry Flanagan -- _ -- 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] Kamallio registration
Hi Guys Is it possible to register Kamallio directly to our SIP provider then load balance the RTP to 2 asterisk servers? We cant do the registration from the asterisk boxes as we want to do it directly from Kamallio. Is this possible? -- _ -- 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] Issues with call dropping
Nick Awesome wrote: Hi guys, have really annoying problem with trunks when I calling over voip provider.. after awhile provider sends INFO packages but for some reason Asterisk doesn’t answer on it. after 8 packagers provider just drops the call, here is the package --- Received SIP request (555 bytes) from UDP:192.168.53.1:5060 --- INFO sip:c2e74184-4d23-43a3-8fd9-26ba2c5ef6c9@192.168.53.9:5060 SIP/2.0 Max-Forwards: 69 To:sip:4959810128@192.168.53.9;tag=b3769af4-118b-4467-8c95-042247ff1776 From:sip:84957774888@192.168.53.1;tag=3638518512-132845 Call-ID: cfe34652-14c2-4072-9dea-f0b0c30cb15e CSeq: 2 INFO Allow: CANCEL, ACK, INVITE, BYE, OPTIONS, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH Via: SIP/2.0/UDP 192.168.53.1:5060;branch=z9hG4bK8da58cede20b91eb54dec15ad27f866c Contact:sip:84957774888@192.168.53.1:5060 Content-Length: 0 Looks like they are using INFO as a keepalive mechanism. Since we don't answer this it'd be a bug. File an issue on the issue tracker[1]. [1] https://issues.asterisk.org/jira -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
