Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Hi Helvio, Could you tell me what is process to setup an environment for IAX. Regards Akhilesh On Fri, Apr 24, 2015 at 4:25 PM, Helvio Junior helvio.lis...@gmail.com wrote: Hi Akhilesh, SIP protocol use port 5060 (default) and many other ports to stablish calls. You need to check if there is AWS firewall rule that allow your communication from your client external IP and your AWS host. Also, think in use IAX intead of SIP, because SIP protocol has many trouble when used with NAT, also IAX protocol use only one port (4569) to everything. When i need allow external clients (throught NAT or not) i used to use IAX. If you want i can help you in your environment (SIP or IAX). Att, Hélvio Junior SafeId - Gestão de identidades e Acessos +55 41 | 9893-2694, single-sign-on.com.br helvio.jun...@safetrend.com.br On 24/04/2015 06:35, akhilesh chand wrote: Hi Guenther, Thanks for ur reply I have concern from long time I'm not able to login through softphone with AWS Cloud.Please let me know is there any document or guide line for the same. 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).
Hi James, Please let me know how could I implement for sip.I will appreciate your help. Regards Akhilesh On Mon, Apr 27, 2015 at 7:01 PM, James Cass jcas...@gmail.com wrote: Akhilesh, I have implemented several ec2 instances with both sip and iax2 and have no problems with xlite or hard phones. Have you already opened the ports in the vpc security group on the Amazon side? Let me know is I can help. --James On Apr 24, 2015 3:34 AM, akhilesh chand omakhileshch...@gmail.com wrote: Hi Thomas, Could you tell how can I change the protocol of corresponding port means 5060 is configured with tcp protocol I want to configured with udp. When I execute nmap -p5060 xx.xx.xx.xx I got below output [root@ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATESERVICE 5060/tcp filtered sip Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds Where I'm seeing 5060 is configured with tcp. Regards Akhilesh On Tue, Apr 21, 2015 at 5:10 PM, Thomas Stein himbe...@meine-oma.de wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Am 21.04.15 um 13:38 schrieb akhilesh chand: Hi Guenther, When I executed nmap -p5060 xx.xx.xx.xx I got below output. [root@ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATE SERVICE 5060/tcp filtered sip Maybe your softphone is trying UDP? cheers t. Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds On Tue, Apr 21, 2015 at 3:20 PM, Guenther Boelter gboel...@gmail.com wrote: On 04/21/2015 04:58 PM, akhilesh chand wrote: Hi Guenther, What did you recommend to me, I did accordingly but there is no log showing in asterisk CLI. I'm getting same problem. Regards Akhilesh Hi Akhilesh, looks like your firewall is blocking it. Have you tried 'nmap -p5060 ip of your asterisk' or something similar? Regards Guenther On Mon, Apr 20, 2015 at 6:05 PM, Guenther Boelter gboel...@gmail.com mailto:gboel...@gmail.com wrote: 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 -- _ -- 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 -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.16 (Darwin) iQEcBAEBAgAGBQJVNjc5AAoJEDoNDMVwb47fa4oH/iG+Y7bJixA/WAJjpydrB5Jr 2lt2qUw9+0rGwn+DiXAQZy5EiA1x3ax4G6/kHJA49XCV7K00w2O69ShVNXwoXAdA yAJxwD6JCExF6BbuAk6xs37wqbMdnhuWjJY42n8GalhUIj/h/I5DQ06PlAaVDlgn aoEcM78JYh2LZg19E7daVcDRc+lfan5werseBSU88Jwo0RGf8zbMb1pM/tACrqkv dzt/CtXjVRK/vCQPjDqkTm20JRCYbp4z2RjL0RtF0Ub60cOdt5fRn0nBrZrn5S5X TvdTvx0vPqmu/KiaA4YybzpjPtfXzBW/gf3+1ZzW/UmkP7smjJewu7hmKKpOVTw= =tvB/ -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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
Re: [asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
Hi Guenther, Thanks for ur reply I have concern from long time I'm not able to login through softphone with AWS Cloud.Please let me know is there any document or guide line for the same. Regards Akhilesh On Fri, Apr 24, 2015 at 1:26 PM, Guenther Boelter gboel...@gmail.com wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 04/24/2015 03:34 PM, akhilesh chand wrote: Hi Thomas, Could you tell how can I change the protocol of corresponding port means 5060 is configured with tcp protocol I want to configured with udp. When I execute nmap -p5060 xx.xx.xx.xx I got below output [root@ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx Starting Nmap 5.51 ( http://nmap.org http://nmap.org/ ) at 2015-04-21 11:19 UTC Nmap scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com http://ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com/ (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATE SERVICE 5060/tcp filtered sip Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds Where I'm seeing 5060 is configured with tcp. 'nmap -p5060 xx.xx.xx.xx' will show you only tcp-ports, try 'nmap -sU - -p5060 xx.xx.xx.xx insteadt -BEGIN PGP SIGNATURE- Version: GnuPG v2 iQIcBAEBAgAGBQJVOfc+AAoJENexF5oIz3BCEZwQAKW9+S50iDmqpgo1wHLQUQA5 62myd+Xnf4v9wD+jh4Z3vnT+OenZ5U+3BNBTwNurCrQ70yX73fc0acektEYXrgIu /M6OgYkCf4ej3sySIvlFyozerYOqh5SxT0CsQHnPLKnyyz4OUlPbiFoiy+2GbNUF xqXifWdjYhKN3MPGFkmtPzGmaJqAmlntE6+0z8h8XQ4uHzKQJ/u6AxweIDZduG78 84XQlIVBiRyve/zaXIZ0aw6IjBoT1saDYF0X8+CmbG4eFfxtZfKC/LKcFaooTTWw FN5BrXYsIjCQzEU/QwoZxHUCXcbKwjoxqKEsb7p/3jRxTLNDaqVuJpql0CUVNBa/ P5kAIHyk9Cfo+GRFkBQD3LcR8DJENLD3hgghyG13E+F4fNAoLs23eaqIjnIzfrDl a1dU0WkS9QsCUWgSckY1Yusq7uEMlcz5myf5axJDkdbO49fXysSPZMlssS11Q0WN +RbfWrFFY4t5mGt3FKHqAM6iuy+hOy6skkSnfhA8CLFVGWbgpwOTzXOCIzzmPwL6 CvIwBv8QLEgunG6O/WIXh2iDd4r07oLR4LKePXis6N8x1LptatIKNhMwnWLwXFTN rdyRY79Zbif3pmJZBas1Jp8YiF81PnJUO3dfnBHzXoY45cztqX29ypi/nLnZPl1i Pl0tW8EV4Qh+XjGJMIeu =vMtm -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 -- _ -- 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, Could you tell how can I change the protocol of corresponding port means 5060 is configured with tcp protocol I want to configured with udp. When I execute nmap -p5060 xx.xx.xx.xx I got below output [root@ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATESERVICE 5060/tcp filtered sip Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds Where I'm seeing 5060 is configured with tcp. Regards Akhilesh On Tue, Apr 21, 2015 at 5:10 PM, Thomas Stein himbe...@meine-oma.de wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Am 21.04.15 um 13:38 schrieb akhilesh chand: Hi Guenther, When I executed nmap -p5060 xx.xx.xx.xx I got below output. [root@ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATE SERVICE 5060/tcp filtered sip Maybe your softphone is trying UDP? cheers t. Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds On Tue, Apr 21, 2015 at 3:20 PM, Guenther Boelter gboel...@gmail.com wrote: On 04/21/2015 04:58 PM, akhilesh chand wrote: Hi Guenther, What did you recommend to me, I did accordingly but there is no log showing in asterisk CLI. I'm getting same problem. Regards Akhilesh Hi Akhilesh, looks like your firewall is blocking it. Have you tried 'nmap -p5060 ip of your asterisk' or something similar? Regards Guenther On Mon, Apr 20, 2015 at 6:05 PM, Guenther Boelter gboel...@gmail.com mailto:gboel...@gmail.com wrote: 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 -- _ -- 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 -BEGIN PGP SIGNATURE- Version: GnuPG/MacGPG2 v2.0.16 (Darwin) iQEcBAEBAgAGBQJVNjc5AAoJEDoNDMVwb47fa4oH/iG+Y7bJixA/WAJjpydrB5Jr 2lt2qUw9+0rGwn+DiXAQZy5EiA1x3ax4G6/kHJA49XCV7K00w2O69ShVNXwoXAdA yAJxwD6JCExF6BbuAk6xs37wqbMdnhuWjJY42n8GalhUIj/h/I5DQ06PlAaVDlgn aoEcM78JYh2LZg19E7daVcDRc+lfan5werseBSU88Jwo0RGf8zbMb1pM/tACrqkv dzt/CtXjVRK/vCQPjDqkTm20JRCYbp4z2RjL0RtF0Ub60cOdt5fRn0nBrZrn5S5X TvdTvx0vPqmu/KiaA4YybzpjPtfXzBW/gf3+1ZzW/UmkP7smjJewu7hmKKpOVTw= =tvB/ -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 -- _ -- 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 Guenther, What did you recommend to me, I did accordingly but there is no log showing in asterisk CLI. I'm getting same problem. Regards Akhilesh On Mon, Apr 20, 2015 at 6:05 PM, Guenther Boelter gboel...@gmail.com wrote: -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 -- _ -- 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 Greg, I moved REJECT rule to last in the list but I'm getting same error. Regards Akhilesh On Mon, Apr 20, 2015 at 5:57 PM, Greg Woods g...@gregandeva.net wrote: 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 -- 0.0.0.0/00.0.0.0/0 reject-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 -- _ -- 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 Guenther, When I executed nmap -p5060 xx.xx.xx.xx I got below output. [root@ip-172-31-32-117 cel]# nmap -p5060 xx.xx.xx.xx Starting Nmap 5.51 ( http://nmap.org ) at 2015-04-21 11:19 UTC Nmap scan report for ec2-xx-xx-xx-xx.us-west-2.compute.amazonaws.com (xx.xx.xx.xx) Host is up (0.00080s latency). PORT STATESERVICE 5060/tcp filtered sip Nmap done: 1 IP address (1 host up) scanned in 0.23 seconds On Tue, Apr 21, 2015 at 3:20 PM, Guenther Boelter gboel...@gmail.com wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 04/21/2015 04:58 PM, akhilesh chand wrote: Hi Guenther, What did you recommend to me, I did accordingly but there is no log showing in asterisk CLI. I'm getting same problem. Regards Akhilesh Hi Akhilesh, looks like your firewall is blocking it. Have you tried 'nmap -p5060 ip of your asterisk' or something similar? Regards Guenther On Mon, Apr 20, 2015 at 6:05 PM, Guenther Boelter gboel...@gmail.com mailto:gboel...@gmail.com wrote: 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 -- _ - -- 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 - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -BEGIN PGP SIGNATURE- Version: GnuPG v2 iQIcBAEBAgAGBQJVNh1pAAoJENexF5oIz3BCeTMQAO50VIcKKQndKmiJEevo+Uzu 4kxTFPg6vLMfV6yOIIC86oxlzqJW6qkuLpt2xb7jr3ppg5VMH88Ozhqpu3E0hZNk VYxXTafWFgHgJLSblxG1ZqA+QZhX/HDdwnHo+OcfeLleJQjxhqvh+8VYOnscv/SP +sfBU2thMckFYaYHSrk9q6hEBKPTnv5oQ8OYwdm92GIvV94N3z7gQ+056dqAOqhs 1dM7D2gLnMZxUPh2aTZStP1V314mFxT/wzIYMRSfwSLJOrclzuzfr4OYPhwJiPsj oD8sCwP9ePwtlBkgFFt5O0FCyR2Uyf5LZN0ClqnSOkizqQi3w9Ssva0y4NzuPOpU aYFD9XYYPDoftuz8kpGqx9Ys8QDf5QONh1HYwx8qkQagvsfkhTnRlb8EvbYkwSDD +pnzC+xysg3qpCBjr/xV31QKJ9q/E9I/qigNiI42GB7gxctkvZ1tf2tEeohZqwzA GaN9kVgKE0oAcJu5kjkKWZFD1uUz7aF8udFR5IH8R10ZvC+PffTZWjDACGZrHM7e q38XH7rb3TKmH+mFwTfAGdoy56FkpJzgjZQYMHie8FPI8uJK/9XStnqb3EF0oFHi 3FJ994MG8iOBS0nlIvJJ2ZtX5TVgBWz9qthETe9/J+eRWueuZv5a7HEfwoOCh8C5 GfIZhiiq9jqjWGRZherc =XKL0 -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 -- _ -- 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
[asterisk-users] I'm not able to register Softphone(X-lite) in asterisk(Which is installed in EC2 Cloud).
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
[asterisk-users] Trying to register Softpone in AWS Cloud
Hi Folks, I'm trying to register softphone(3CX Phone) in AWS Cloud but I'm not able to register I got below screen. [image: Inline image 1] Register Screen for 3CX Phone [image: Inline image 1] 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 install asterisk in AWS cloud
Hi Ajahar, I tried your solution it is working.Thanks a lot man. Regards Akhilesh On Mon, Apr 13, 2015 at 5:43 PM, ajahar mohd azhar5...@gmail.com wrote: Hi Akhilesh, Here is another fix, getting the error, that: make[1]: *** No rule to make target `../main/modules.link’, needed by `asterisk’. Stop. make: *** [main] Error 2 when compile asterisk To get around this, just delete following line in file makeopts.embed_rules EMBED_LDSCRIPTS+=../main/modules.link Source: http://showmyroutes.com/wordpress/?p=500 Sincerely, M azhar http://www.nicacresults.com On Mon, Apr 13, 2015 at 1:10 PM, akhilesh chand omakhileshch...@gmail.com wrote: Hi folks, I'm not able to install asterisk whenever I hit make command I get below error: make[1]: *** No rule to make target `../main/modules.link', needed by `asterisk'. Stop. make: *** [main] Error 2 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
[asterisk-users] I'm not able to install asterisk in AWS cloud
Hi folks, I'm not able to install asterisk whenever I hit make command I get below error: make[1]: *** No rule to make target `../main/modules.link', needed by `asterisk'. Stop. make: *** [main] Error 2 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 install asterisk in AWS cloud
yes I called On Mon, Apr 13, 2015 at 1:27 PM, jg webaccounts...@jgoettgens.de wrote: I'm not able to install asterisk whenever I hit make command I get below error: make[1]: *** No rule to make target `../main/modules.link', needed by `asterisk'. Stop. make: *** [main] Error 2 Just guessing. Did you call ./configure? jg -- _ -- 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 install asterisk in AWS cloud
yes modules.link is existing in pbx/modules.link. On Mon, Apr 13, 2015 at 2:35 PM, Guenther Boelter gboel...@gmail.com wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 04/13/2015 03:40 PM, akhilesh chand wrote: Hi folks, I'm not able to install asterisk whenever I hit make command I get below error: make[1]: *** No rule to make target `../main/modules.link', needed by `asterisk'. Stop. make: *** [main] Error 2 Regards Akhilesh Does `../main/modules.link' exist after running ./configure? - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -BEGIN PGP SIGNATURE- Version: GnuPG v2 iQIcBAEBAgAGBQJVK4bAAAoJENexF5oIz3BC4gsQAMDXbhvpvDlv7pTKcspO0VGF iJkLqRHUsEsrLtNg8WWOT1AvwO5aZE/lrVgY/xA0WYjVdfDwXZK3vT2cgyt1JvDf dyuw9DzHKjeQNoa+HqxfdnnvrQbOdt+vmgJVzV2uVBTbTH1WY3tAonTRsHYD4kZr 0C4P9jkB4Ov45LFLfRbrmjst/Wu/ilYKg5HkqQSKvmIZiP0VockfxFSzMjZej+gX GtERMQe/Ni/1w16Fvwv3vM28DlM4nnx2ujD1XWLoR7UsUvU+f+YhJcOD3ogrBA8Q gwUG4YvLP0yITtHOQy3uy7sGinC4m3Tj41gu32nqfcEvwE+MsGuCzammDKf7VxOk Yw3OQctEi68zja+5j6GBaFB/BIaylovqwVSy3sHZEqU9Gx811X7Y4IK0ml5ydl+v Smy3RG2BnNVZoR6YOlN3L3NSUPEw5btp4mli7YkOtmKAljDdzHXNI6c7X+pr409F eEIppmSii2if7D5asXr9Y+SPI8ErYSEJHzCqxRBS27ATG4zO6Ki5BxQ4oqs9N0Lj JvnkVDKbPW6/Mx2/p1rXSETMhEqnXW741Pm0K5eXEKLfufoqlESw3KaLsrTpzOlP TJ/f+RPWPwR3DX5TLiHHLLyTQ3FJsrvlkI3C57KxzBX36JfSWhGYSGtozvtpxP1h v++czTsnwkwWA8mEHdOX =IFVV -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 -- _ -- 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] Not able to register an Extension
Hi Alonso, Thanks for your reply but after setting the value of srvlookup=no i got same error. On Sat, Nov 22, 2014 at 1:37 AM, Alonso Genis alo...@planetfone.com.br wrote: - Mensagem original - De: akhilesh chand omakhileshch...@gmail.com Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviadas: Sexta-feira, 21 de novembro de 2014 16:54:35 Assunto: Re: [asterisk-users] Not able to register an Extension Hi Alonso, sip.conf [general] context=hunt_incoming port=5060 bindaddr=0.0.0.0 srvlookup=yes Did you try to set srvlookup=no? http://www.voip-info.org/wiki/view/Asterisk+SIP+srvlookup disallow=all allow=all nat=yes callerid = LITE externip= externhost= autocreatepeer=yes autodomain=yes localnet=xxx.xxx.xxx.xxx/xxx.xxx.xxx.xxx canreinvite=yes language=En allowtransfer=yes realm=telunet domain=192.168.1.5 maxexpiry=3600 defaultexpiry=200 useragent=LITE PBX usereqphone = yes dtmfmode = rfc2833 alwaysauthreject = no regcontext=sipregistrations rtptimeout=3600 rtpholdtimeout=300 rtcachefriends=yes ;--- SIP DEBUGGING --- sipdebug = yes registertimeout=60 registerattempts=5 callgroup=1 pickupgroup=1 callevents=yes ;register = username:password:username@Sip Proxy IP or domain name [authentication] [4001] type=friend context=outbound defaultuser=4001 secret=4001 callerid=EXT1 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4002] type=friend context=outbound defaultuser=4002 secret=4002 callerid=EXT2 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4003] type=friend context=outbound defaultuser=4003 secret=4003 callerid=EXT3 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all On Fri, Nov 21, 2014 at 11:52 PM, Alonso Genis alo...@planetfone.com.br wrote: - Mensagem original - De: akhilesh chand omakhileshch...@gmail.com Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviadas: Sexta-feira, 21 de novembro de 2014 14:36:05 Assunto: [asterisk-users] Not able to register an Extension Hi folk, I'm trying to register an extension through softphone and got stuck.I got below error:- [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: missing sent-by in Via header [Nov 22 07:31:28] ERROR[3522]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo(, (null), ...): Name or service not known [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:16056 check_via: Could not resolve socket address for '' Sending to 192.168.1.2:5060 (NAT) [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: missing sent-by in Via header [Nov 22 07:31:28] ERROR[3522]: chan_sip.c:7897 process_via: error processing via header [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:10420 respprep: error processing via header, will send response to originating address Please let me know how could i solve the same and I will appreciate your suggestion. Please, send us your sip.conf, i suspect is a problem with your bindaddr or name resolution. Alonso. Thanks 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us
[asterisk-users] Not able to register an Extension
Hi folk, I'm trying to register an extension through softphone and got stuck.I got below error:- [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: missing sent-by in Via header [Nov 22 07:31:28] ERROR[3522]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo(, (null), ...): Name or service not known [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:16056 check_via: Could not resolve socket address for '' Sending to 192.168.1.2:5060 (NAT) [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: missing sent-by in Via header [Nov 22 07:31:28] ERROR[3522]: chan_sip.c:7897 process_via: error processing via header [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:10420 respprep: error processing via header, will send response to originating address Please let me know how could i solve the same and I will appreciate your suggestion. Thanks 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] Not able to register an Extension
Hi Alonso, sip.conf [general] context=hunt_incoming port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=all nat=yes callerid = LITE externip= externhost= autocreatepeer=yes autodomain=yes localnet=xxx.xxx.xxx.xxx/xxx.xxx.xxx.xxx canreinvite=yes language=En allowtransfer=yes realm=telunet domain=192.168.1.5 maxexpiry=3600 defaultexpiry=200 useragent=LITE PBX usereqphone = yes dtmfmode = rfc2833 alwaysauthreject = no regcontext=sipregistrations rtptimeout=3600 rtpholdtimeout=300 rtcachefriends=yes ;--- SIP DEBUGGING --- sipdebug = yes registertimeout=60 registerattempts=5 callgroup=1 pickupgroup=1 callevents=yes ;register = username:password:username@Sip Proxy IP or domain name [authentication] [4001] type=friend context=outbound defaultuser=4001 secret=4001 callerid=EXT1 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4002] type=friend context=outbound defaultuser=4002 secret=4002 callerid=EXT2 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4003] type=friend context=outbound defaultuser=4003 secret=4003 callerid=EXT3 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all On Fri, Nov 21, 2014 at 11:52 PM, Alonso Genis alo...@planetfone.com.br wrote: - Mensagem original - De: akhilesh chand omakhileshch...@gmail.com Para: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Enviadas: Sexta-feira, 21 de novembro de 2014 14:36:05 Assunto: [asterisk-users] Not able to register an Extension Hi folk, I'm trying to register an extension through softphone and got stuck.I got below error:- [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: missing sent-by in Via header [Nov 22 07:31:28] ERROR[3522]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo(, (null), ...): Name or service not known [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:16056 check_via: Could not resolve socket address for '' Sending to 192.168.1.2:5060 (NAT) [Nov 22 07:31:28] ERROR[3522]: sip/reqresp_parser.c:2265 parse_via: missing sent-by in Via header [Nov 22 07:31:28] ERROR[3522]: chan_sip.c:7897 process_via: error processing via header [Nov 22 07:31:28] WARNING[3522]: chan_sip.c:10420 respprep: error processing via header, will send response to originating address Please let me know how could i solve the same and I will appreciate your suggestion. Please, send us your sip.conf, i suspect is a problem with your bindaddr or name resolution. Alonso. Thanks 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
[asterisk-users] file.c:1160 ast_writefile: Unable to open file /var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav: No such file or directory
Dear Folks,
[Test_Context]
exten = _911.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _911.,2,Set(CALLERID(num)=xxx)
exten =
_911.,3,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})
exten = _911.,4,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})
exten = _911.,5,Set(${CALLERID}=${CALLERID(num)})
exten = _911.,6,Set(FILENAME=${CALLERID}_${CALLTIME}.wav)
exten = _911.,7,Set(RECORDFILENAME=${RECSUBDIR}/${FILENAME})
;exten = _911.,8,Set(SOUND_PATH=${RECORDING_KINREP}/${RECORDFILENAME})
exten = _911.,8,MixMonitor(${RECORDFILENAME},b)
exten = _911.,9,Dial(${TRUNK}/${EXTEN:3},,To)
exten = _911.,10,Hangup
Mixmont is not working ,Whenever my give code is executing i got following
error:
file.c:1160 ast_writefile: Unable to open file
/var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav: No such
file or directory
app_mixmonitor.c:286 mixmonitor_thread: Cannot open
/var/spool/asterisk/monitor/11Feb2014/_11-Feb-2014-17-44-01.wav
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
[asterisk-users] I'm not able hearing the voice.
Dear Folks, I'm not able hearing the voice of client but on other hand client able to hearing my voice.I'm not able to find out the problem where is i'm wrong. I'm getting continues following error: chan_sip.c:10391 check_via: '' is not a valid host Configuration DAHDI Tools Version - 2.9.0.1 DAHDI Version: 2.9.0 Regards akihlesh -- _ -- 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] DAHDI_SPANCONFIG failed on span 1: No such device or address (6)
Dear Folks, whenever I'm executing following command : dahdi_cfg -vvv I got following error: DAHDI_SPANCONFIG failed on span 1: No such device or address (6) 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] DAHDI_SPANCONFIG failed on span 1: No such device or address (6)
I had just upgrade the dahdi drivers. *[root@XX ~]# dahdi_scan*[1] active=yes alarms=OK description=Wildcard TE131/TE133 Card 0 name=WCT13x/0 manufacturer=Digium devicetype=Wildcard TE131/TE133 (VPMOCT032) location=PCI Bus 01 Slot 01 basechan=1 totchans=31 irq=0 type=digital-E1 syncsrc=1 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=AMI,HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS *[root@XX ~]# cat /proc/dahdi/1* Span 1: WCT13x/0 Wildcard TE131/TE133 Card 0 (MASTER) CCS/HDB3 ClockSource Timing slips: 1 1 WCT13x/0/1 Clear (In use) (EC: VPMOCT032 - INACTIVE) 2 WCT13x/0/2 Clear (In use) (EC: VPMOCT032 - INACTIVE) 3 WCT13x/0/3 Clear (In use) (EC: VPMOCT032 - INACTIVE) 4 WCT13x/0/4 Clear (In use) (EC: VPMOCT032 - INACTIVE) 5 WCT13x/0/5 Clear (In use) (EC: VPMOCT032 - INACTIVE) 6 WCT13x/0/6 Clear (In use) (EC: VPMOCT032 - INACTIVE) 7 WCT13x/0/7 Clear (In use) (EC: VPMOCT032 - INACTIVE) 8 WCT13x/0/8 Clear (In use) (EC: VPMOCT032 - INACTIVE) 9 WCT13x/0/9 Clear (In use) (EC: VPMOCT032 - INACTIVE) 10 WCT13x/0/10 Clear (In use) (EC: VPMOCT032 - INACTIVE) 11 WCT13x/0/11 Clear (In use) (EC: VPMOCT032 - INACTIVE) 12 WCT13x/0/12 Clear (In use) (EC: VPMOCT032 - INACTIVE) 13 WCT13x/0/13 Clear (In use) (EC: VPMOCT032 - INACTIVE) 14 WCT13x/0/14 Clear (In use) (EC: VPMOCT032 - INACTIVE) 15 WCT13x/0/15 Clear (In use) (EC: VPMOCT032 - INACTIVE) 16 WCT13x/0/16 HDLCFCS (In use) (EC: VPMOCT032 - INACTIVE) 17 WCT13x/0/17 Clear (In use) (EC: VPMOCT032 - INACTIVE) 18 WCT13x/0/18 Clear (In use) (EC: VPMOCT032 - INACTIVE) 19 WCT13x/0/19 Clear (In use) (EC: VPMOCT032 - INACTIVE) 20 WCT13x/0/20 Clear (In use) (EC: VPMOCT032 - INACTIVE) 21 WCT13x/0/21 Clear (In use) (EC: VPMOCT032 - INACTIVE) 22 WCT13x/0/22 Clear (In use) (EC: VPMOCT032 - INACTIVE) 23 WCT13x/0/23 Clear (In use) (EC: VPMOCT032 - INACTIVE) 24 WCT13x/0/24 Clear (In use) (EC: VPMOCT032 - INACTIVE) 25 WCT13x/0/25 Clear (In use) (EC: VPMOCT032 - INACTIVE) 26 WCT13x/0/26 Clear (In use) (EC: VPMOCT032 - INACTIVE) 27 WCT13x/0/27 Clear (In use) (EC: VPMOCT032 - INACTIVE) 28 WCT13x/0/28 Clear (In use) (EC: VPMOCT032 - INACTIVE) 29 WCT13x/0/29 Clear (In use) (EC: VPMOCT032 - INACTIVE) 30 WCT13x/0/30 Clear (In use) (EC: VPMOCT032 - INACTIVE) 31 WCT13x/0/31 Clear (In use) (EC: VPMOCT032 - INACTIVE) On Wed, Feb 5, 2014 at 2:55 AM, Shaun Ruffell sruff...@digium.com wrote: On Wed, Feb 05, 2014 at 02:46:34AM +0530, akhilesh chand wrote: Dear Folks, whenever I'm executing following command : dahdi_cfg -vvv I got following error: DAHDI_SPANCONFIG failed on span 1: No such device or address (6) Do you have any dahdi devices loaded? What is the output of dahdi_scan or cat /proc/dahdi/1? -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to get ringing sound in outbound call in asterisk
I have two server
Server_A(outbound call) for agent login and agent make a outbound call from
here and pass into server Server_B call
extension.conf
exten = _91XX.,1,AGI(agi://127.0.0.1:4577/call_log)
exten = _91XX.,n,Dial(SIP/${EXTEN}@192.168.53.197,,tToR)
exten = _91XX.,n,hangup()
Server_B[192.168.53.197] for call forwarding
extension.conf
exten = _911X.,1,ChanisAvail(${TRUNK_GRP3})
exten = _911X.,2,gotoif($[${AVAILCHAN} = ]?lbl_busy:)
exten =
_911X.,n,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})
exten = _911X.,n,Gotoif($[${RECORDING_ENABLED}=Y]?lbl_dbc:lbl_dial)
exten =
_911X.,n(lbl_dbc),Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})
exten = _911X.,n,Set(CALLERID(num)=${IDGCLI})
exten = _911X.,n,Set(FILENAME=${IDGTERMINAL}_${EXTEN:1}_${CALLTIME}.WAV)
exten = _911X.,n,Set(RECORDFILENAME=${RECSUBDIR}/${FILENAME})
exten = _911X.,n,Gotoif($[${IDGCALL}=]?lbl_setcall:lbl_sendevent)
exten = _911X.,n(lbl_setcall),Set(IDGCALL=0)
exten = _911X.,n(lbl_sendevent),Gotoif($[${DBTYPE}=SQL]?lbl_sql:)
exten = _911X.,n,Gotoif($[${DBTYPE}=MYSQL]?lbl_mysql:lbl_record)
exten = _911X.,n(lbl_sql),UserEvent(${CHANNEL}$DBEXEC$EXEC
udsp_vlog_start_record
'${CHANNEL}'#'${IDGTERMINAL}'#'${EXTEN:1}'#'${FILENAME}'#'${UNIQUEID}'#'${CALLERID(num)}'#'O'#${VLOGSERVER}#'${HTTPPATH}${RECORDFILENAME}'#${IDGCALL}$)
exten = _911X.,n,Goto(lbl_record)
exten = _911X.,n(lbl_mysql),UserEvent(${CHANNEL}$DBEXEC$CALL
udsp_vlog_start_record
('${CHANNEL}'#'${IDGTERMINAL}'#'${EXTEN:1}'#'${FILENAME}'#'${UNIQUEID}'#'${CALLERID(num)}'#'O'#${VLOGSERVER}#'${HTTPPATH}${RECORDFILENAME}'#${IDGCALL})$)
exten =
_911X.,n(lbl_record),MixMonitor(${RECORDING_PATH_OUT_SREI}${RECORDFILENAME})
exten = _911X.,n(lbl_dial),Set(ChanLength=${LEN(${AVAILCHAN})})
exten = _911X.,n,Set(NewChannel=${AVAILCHAN:0:$[${ChanLength}-2]})
exten = _911X.,n,Dial(${NewChannel}/${EXTEN:3},,tToR)
exten = _911X.,n,hangup()
exten = _911X.,n(lbl_busy),Busy()
I'm not able listened ringing sound when i make outbound call.
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] Communicate with barge agent
thanks a lot Satish and Shahbaz for ur precious reply On Tue, Nov 19, 2013 at 2:59 PM, Shahbaz Afzal pices...@gmail.com wrote: Hi Akhilesh, Yes it is possible using application chanspy in asterisk, use it as following example below. in this example when you press 01 you can listen and also whisper to agent using extension SIP/301 [spy] exten = NoCDR() exten = 01,1, ChanSpy(SIP/301,qw) Regards, Shahbaz On Tue, Nov 19, 2013 at 12:32 PM, akhilesh chand omakhileshch...@gmail.com wrote: HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. 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
[asterisk-users] Communicate with barge agent
HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. 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
[asterisk-users] Make phone ring through webserver using Asterisk
What is the easiest way? And how can it be implemented? I thought to something like: 1. I request a page to the webserver 2. Perl sends to asterisk a number to dial (Perl and asterisk are running in the same machine) 3. Asterisk calls the phone or 1. A Perl sip client registers to remote asterisk server 2. Perl sip client sends to asterisk the number to dial 3. Phone rings i don't care if i can hear something, it's enough that it rings -- _ -- 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] two steps when calling from web!
I'm making a call from web(click to dial) and able to successfully dial to number but problem with when i dial a number call goes to first client and after that call come into my softphone show me Answer and Decline bottoms, and then I have to click Answer to call the number. it seems it is two step to calling the number. If I type the number direct to my client softphone, it calls directly the number without show me to choose Answer to calling. I want integrate web application with softphone suppose to I will click on dial button on web call goes to via my softphone or i will get ringing(or ISDN status of call) sound. On Fri, Nov 8, 2013 at 2:37 PM, A J Stiles asterisk_l...@earthshod.co.ukwrote: On Friday 08 November 2013, akhilesh chand wrote: When I calling a number from web, my softphone show me Answer and Decline bottoms, and then I have to click Answer to call the number. it seems it is two step to calling the number. If I type the number direct to my client softphone, it calls directly the number without show me to choose Answer to calling. First call connect with client and then come into my screen and showing me to choose Answer and Decline.I'm not able to listen ringing sound because call is connecting first with client and then connect with my softphone. That's the normal way things work. If you use Asterisk to set up a call, either through AMI or by means of a call file, then you have to lift your receiver to set the other end ringing. (You can prove this easily enough.) Even if the phone on your end is a softphone, you still have to lift the receiver by pressing Answer. What you really want, is to start your softphone and make it dial the number. How exactly to do this depends on your setup. -- AJS Answers come *after* questions. -- _ -- 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] two steps when calling from web!
Dear All. When I calling a number from web, my softphone show me Answer and Decline bottoms, and then I have to click Answer to call the number. it seems it is two step to calling the number. If I type the number direct to my client softphone, it calls directly the number without show me to choose Answer to calling. First call connect with client and then come into my screen and showing me to choose Answer and Decline.I'm not able to listen ringing sound because call is connecting first with client and then connect with my softphone. My source code is in AMI socket open to make call from web. how can I call direct to the number? -- _ -- 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] two steps when calling from web!
Dear All. When I calling a number from web, my softphone show me Answer and Decline bottoms, and then I have to click Answer to call the number. it seems it is two step to calling the number. If I type the number direct to my client softphone, it calls directly the number without show me to choose Answer to calling. First call connect with client and then come into my screen and showing me to choose Answer and Decline.I'm not able to listen ringing sound because call is connecting first with client and then connect with my softphone. My source code is in AMI socket open to make call from web. how can I call direct to the number? -- _ -- 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] Register Sip extension with out Sip phone
Dear all, I have two system Sys A and Sys X. Sys A is normal PC. Sys X have installed asterisk 1.6 and i want register(or reserved) sip extension(like 4001,4002,4003..) through Sys A(Sys A have some ip address) but i don't use any soft-phone means i want to write Perl or php(any language) script to register sip extension. Suppose to 4001 is reserved with Sys A. 4002 is reserved with Sys B. 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] Problem with call transfer from one server to another server
Server A ( which contain pri line) *chan_dahdi.conf* [channels] group=1 context=outbound usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes faxdetect=both callprogress=no progzone=in pulsedial=yes ;busydetect=yes callreturn=yes echocancel=no echocancelwhenbridged=no rxgain=0.5 txgain=0.5 relaxdtmf=yes callgroup=1 pickupgroup=1 pritimer = t309,6000 immediate=no switchtype=euroisdn context=outgoing signalling=pri_cpe pridialplan=unknown group=1 channel = 1-15,17-31 context=outgoing signalling=pri_cpe pridialplan=unknown group=2 channel = 32-46,48-62 context=outgoing signalling=pri_cpe pridialplan=unknown group=3 channel = 63-77,79-93 context=outgoing signalling=pri_cpe pridialplan=unknown group=4 channel = 94-108,110-124 *sip.conf* [general] context=default port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=all nat=yes callerid = LITE externip= externhost= autocreatepeer=yes autodomain=yes localnet=192.168.53.197/255.255.255.0 canreinvite=yes language=En allowtransfer=yes realm=telunet domain=192.168.53.197 maxexpiry=3600 defaultexpiry=200 useragent=LITE PBX usereqphone = yes dtmfmode = rfc2833 alwaysauthreject = no regcontext=sipregistrations rtptimeout=60 rtpholdtimeout=300 rtcachefriends=yes ;--- SIP DEBUGGING --- sipdebug = yes registertimeout=60 registerattempts=5 callgroup=1 pickupgroup=1 callevents=yes ;register = username:password:username@Sip Proxy IP or domain name [authentication] [4001] type=friend context=outbound defaultuser=4001 secret=4001 callerid=EXT1 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4002] type=friend context=outbound defaultuser=4002 secret=4002 callerid=EXT2 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4003] type=friend context=outbound defaultuser=4003 secret=4003 callerid=EXT3 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4004] type=friend context=outbound defaultuser=4004 secret=4004 callerid=EXT4 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all On Sun, Oct 20, 2013 at 11:14 AM, Mitul Limbani mi...@enterux.in wrote: Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd link here. Mitul On Oct 20, 2013 11:07 AM, akhilesh chand omakhileshch...@gmail.com wrote: Dear All, I have pri with E1 facility that have 30 line and 100 pri number which is provided by service provider.Number started like 23568561,23568562,23568563 and so on. Service provider provide last four digit number for did mapping like 4561,4562,4563. exten = 8561,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8561,n,hangup() exten = 8562,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8562,n,hangup() Call comes into first server successful.But problem with second server when call came into second server i got following error: * chan_sip.c:20063 handle_request_invite: Call from '' to extension '4001' rejected because extension not found.* In one more scenario: when i create one extension and call forwarding with this extension that time I'm able to transfer call successful the code is given below: exten = 5001,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 5001,n,hangup() 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] Problem with call transfer from one server to another server
Server B(child server) *chan_dahdi.conf* [trunkgroups] [channels] group=1 context=outbound usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes faxdetect=both callprogress=no progzone=in pulsedial=yes ;busydetect=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.5 txgain=0.5 callgroup=1 pickupgroup=1 pritimer = t309,6000 immediate=no switchtype=euroisdn context=outgoing signalling=pri_cpe pridialplan=unknown group=1 channel = 1-15,17-31 context=outgoing signalling=pri_cpe pridialplan=unknown group=2 channel = 32-46,48-62 context=outgoing signalling=pri_cpe pridialplan=unknown group=3 channel = 63-77,79-93 context=outgoing signalling=pri_cpe pridialplan=unknown group=4 channel = 94-108,110-124 *Sip.conf* [general] pear=type context=hunt_incoming port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=all nat=yes callerid = LITE externip= externhost= autocreatepeer=yes autodomain=yes localnet=192.168.14.112/255.255.255.0 canreinvite=yes language=En allowtransfer=yes realm=telunet domain=192.168.14.112 maxexpiry=3600 defaultexpiry=200 useragent=LITE PBX usereqphone = yes dtmfmode = rfc2833 alwaysauthreject = no regcontext=sipregistrations rtptimeout=3600 rtpholdtimeout=300 rtcachefriends=yes ;--- SIP DEBUGGING --- sipdebug = yes registertimeout=60 registerattempts=5 callgroup=1 pickupgroup=1 callevents=yes Disallow=all Allow=all ;Allow=ulaw ;Allow=gsm Canreinvite=no ;register = username:password:username@Sip Proxy IP or domain name [authentication] [4001] type=friend context=outbound defaultuser=4001 secret=4001 callerid=EXT1 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all [4002] type=friend context=outbound defaultuser=4002 secret=4002 callerid=EXT2 host=dynamic nat=no dtfmode=rfc2833 disallow=all subscribecontext=outbound canreinvite=no allow=all On Sun, Oct 20, 2013 at 11:14 AM, Mitul Limbani mi...@enterux.in wrote: Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd link here. Mitul On Oct 20, 2013 11:07 AM, akhilesh chand omakhileshch...@gmail.com wrote: Dear All, I have pri with E1 facility that have 30 line and 100 pri number which is provided by service provider.Number started like 23568561,23568562,23568563 and so on. Service provider provide last four digit number for did mapping like 4561,4562,4563. exten = 8561,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8561,n,hangup() exten = 8562,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8562,n,hangup() Call comes into first server successful.But problem with second server when call came into second server i got following error: * chan_sip.c:20063 handle_request_invite: Call from '' to extension '4001' rejected because extension not found.* In one more scenario: when i create one extension and call forwarding with this extension that time I'm able to transfer call successful the code is given below: exten = 5001,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 5001,n,hangup() 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
[asterisk-users] Problem with call transfer from one server to another server
Dear All, I have pri with E1 facility that have 30 line and 100 pri number which is provided by service provider.Number started like 23568561,23568562,23568563 and so on. Service provider provide last four digit number for did mapping like 4561,4562,4563. exten = 8561,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8561,n,hangup() exten = 8562,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 8562,n,hangup() Call comes into first server successful.But problem with second server when call came into second server i got following error: * chan_sip.c:20063 handle_request_invite: Call from '' to extension '4001' rejected because extension not found.* In one more scenario: when i create one extension and call forwarding with this extension that time I'm able to transfer call successful the code is given below: exten = 5001,1,Dial(SIP/4001@192.168.14.110,120,tT) exten = 5001,n,hangup() 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
[asterisk-users] How to disable Internal call ?
Dear All, I want to disable internal call facility.Means agent(4002) does not make call to agent(4003) or other extensions. 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
[asterisk-users] Read Telnet Packet
Dear All, I want to read telnet packet continuously whenever a new call is originated and store into a variable after that pass into window server. I have written a Perl script to read telnet packet but problem is that whenever I executed Perl script then got a telnet packet( mean Only when i execute Perl script) here I want to put scheduler,event or other technique whenever a new call will come Perl script automatically run. 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
[asterisk-users] utils.c: fwrite() returned error: Broken pipe how to solve it ???
Dear all,
I want to make call through socket i have set code given below:
#!/usr/bin/perl -w
use IO::Socket::INET;
sub asterisk_command ()
{
# my $command=$_[0];
my
$ami=IO::Socket::INET-new(PeerAddr='127.0.0.1',PeerPort=5038,Proto='tcp')
or die failed to connect to AMI!;
print $ami Action: Login\r\nUsername: lite\r\nSecret:
4003\r\n\r\nAction: Logoff\r\n\r\n;
}
asterisk_command(Channel: DAHDI/27/7702009896\r\nExten: s\r\nContext:
outbound\r\nCallerID: 20048645\r\nPriority: 1\r\nMaxRetries: 2\r\n);
Whenever i execute that code i'm get following error
[Oct 10 15:13:23] ERROR[856]: utils.c:1175 ast_careful_fwrite: fwrite()
returned error: Broken pipe
[Oct 10 15:13:23] ERROR[856]: utils.c:1175 ast_careful_fwrite: fwrite()
returned error: Broken pipe
[Oct 10 15:13:23] ERROR[856]: utils.c:1175 ast_careful_fwrite: fwrite()
returned error: Broken pipe
[Oct 10 15:13:23] ERROR[856]: utils.c:1175 ast_careful_fwrite: fwrite()
returned error: Broken pipe
asterisk verison :- 1.6.2.7
CentOS release 5.3
kernel version :- 2.6.18-128.el5
--
_
-- 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] utils.c: fwrite() returned error: Broken pipe how to solve it ???
thanks a lot Tony
On Thu, Oct 10, 2013 at 4:31 PM, Tony Mountifield t...@softins.co.ukwrote:
In article
cae6_ne+dxtsgadtg0mp-9jumngxguwo4exadm_hrwc8opuo...@mail.gmail.com,
akhilesh chand omakhileshch...@gmail.com wrote:
I want to make call through socket i have set code given below:
#!/usr/bin/perl -w
use IO::Socket::INET;
sub asterisk_command ()
{
# my $command=$_[0];
my
$ami=IO::Socket::INET-new(PeerAddr='127.0.0.1',PeerPort=5038,Proto='tcp')
or die failed to connect to AMI!;
print $ami Action: Login\r\nUsername: lite\r\nSecret:
4003\r\n\r\nAction: Logoff\r\n\r\n;
}
asterisk_command(Channel: DAHDI/27/7702009896\r\nExten: s\r\nContext:
outbound\r\nCallerID: 20048645\r\nPriority: 1\r\nMaxRetries: 2\r\n);
Whenever i execute that code i'm get following error
[Oct 10 15:13:23] ERROR[856]: utils.c:1175 ast_careful_fwrite: fwrite()
returned error: Broken pipe
[Oct 10 15:13:23] ERROR[856]: utils.c:1175 ast_careful_fwrite: fwrite()
returned error: Broken pipe
[Oct 10 15:13:23] ERROR[856]: utils.c:1175 ast_careful_fwrite: fwrite()
returned error: Broken pipe
[Oct 10 15:13:23] ERROR[856]: utils.c:1175 ast_careful_fwrite: fwrite()
returned error: Broken pipe
asterisk verison :- 1.6.2.7
CentOS release 5.3
kernel version :- 2.6.18-128.el5
AMI is a *two-way* protocol. You mustn't just fire in a bunch of commands
and close the socket!
The reason Asterisk reports the fwrite() error is because you have closed
the socket before it had a chance to send you the responses.
What you need to do is this:
1. Connect to the AMI port.
2. Read the one-line greeting message that Asterisk sends you. It will tell
you the version of the protocol (which might be of interest if you
wanted
to be compatible with different versions of Asterisk).
3. Send the Login action with username, secret and terminating blank line.
4. Read the response lines from Asterisk until it gives you a blank line.
5. Send whatever command you want it to do, and go back to step 4.
6. When you have done the commands you want, send the Logoff action.
7. *** READ THE RESPONSE TO THE LOGOFF
8. Close the socket.
If it helps. Here is similar piece of code I wrote to query pri spans.
Note carefully the setting of $/ in two places, and the inclusion of
Events: off to avoid responses getting confused by asynchronous events.
==
#!/usr/bin/perl
use IO::Socket;
my $numspans = 4;
my $host = 'localhost';
my $login = Action: login\r\nUsername: \r\nSecret: \r\nEvents:
off\r\n\r\n;
$/ = \r\n;# reads a single line for signon banner
my $s = IO::Socket::INET-new($host:5038) or die can't connect to
$host: $!\n;
my $banner = $s; # read the banner
#my $line = ('-' x 78).\n;
#print $banner,$line;
$/ = \r\n\r\n;# reads a complete response ending in a blank
line
print $s $login;
my $resp = $s;
#print $resp,$line;
my @spans;
foreach $span (1..$numspans) {
print $s Action: Command\r\nCommand: pri show span $span\r\n\r\n;
$resp = $s;
#print $resp,$line;
if ($resp =~ /Status: (.*)\n/) {
$status = $1;
} else {
$status = 'Unknown';
}
$spans[$span-1] = Span $span status = $status\n;
}
print $s Action: Logoff\r\n\r\n;
$resp = $s;
#print $resp,$line;
close $s;
# go on to display the results from @spans
==
Cheers
Tony
--
Tony Mountifield
Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to disable call transfers?
Dear All, I want to disable call transfers internally.Means agent(4002) does not transfer call to agent(4003) or other extensions. But i want to create two extensions as supervisor who are able to take a internal call.Suppose to agent(4001) able transfer call agent(5001) or agent(5002). 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
[asterisk-users] Queue Management
Dear All, I have six different campaign and 5 different agent have login on that campaign.*Same thing i have done using agi and database,i never use queue management on this scenario. Agent** can also shuffling one campaign to anther campaign. * Now i want to do some work with queue.I want to use single queue to managing this. Eg: campaign Agent Login A a_1,a_3 (In campaign A 2 agents are login) B a_2,a_1 (In campaign B 2 agents are login) C a_3,a_1,a_4 (In campaign C 3 agents are login) D a_4,a_5,a_3 (In campaign D 3 agents are login) E a_1,a_3,1_2 (In campaign E 3 agents are login) F a_5,a_4(In campaign F 2 agents are login) When a call come to campaign A that call goes to agent a_1 or a_3 not goes to other campaigns agents. 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
[asterisk-users] Managing Abandoned Call
Dear All, I have a query ,basically i use three server for own call center. The server A and B i have configure the 60-60 channel each server. Server A and B(or call transfers into server X) calls hitting into server X.Both the server have contain same CLI mean anybody call 8032(mean server A an B) call goes to Server X. In the case of Server A 8032 mapped with toll-free,it is configured with Server A, Anybody dial toll-free call goes to server X via Server A. In the case of Server B Soppose to any anybody dial directly 8032 call goes to server X via Server B. Supposed to two call originate at same time one call come via toll-free and another one call come via 8032 (dial directly pilot number) and both the channels dial into same extension(4002) due to this reason one of the call is abandoned and another one is pick by the agent. ** 8032 is pilot number Please help me. 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] Remove Abandoned call
In server_X, 60 agnet are login and call comes from server_A and server_B which is connected with pri(total 90 channel), Supposed to two call originate 02:25 PM from A B and hit into server_X goes to agent 2002(agnet extension) one of call is abandoned and another one pick up agent 2002 .Both the calls hit into 2002 extension(X_server).I want to set the priority. akhilesh On Thu, Feb 21, 2013 at 1:50 PM, Leandro Dardini ldard...@gmail.com wrote: 2013/2/21 akhilesh chand omakhileshch...@gmail.com hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call transfer into server_X server_B is connected with pri and configure with 30 channel for call transfer into server_X my query is that some time two call originate same time from two different server_A and server_B and hit into server_X and one call is abandoned and another one have taken by the agent But i don't want to abandoned the call, I want to set the priority, supposed to server_A and server_B call originate same time server_X take the call from server_A first and then take the call server_B after 1 sec please guide me Regards Akhilesh I am sorry if I haven't completely understood your question, but english is not my native language. If calls from server_A and server_B are put in the same queue in server_X, how can one of them being abandoned? Calls will be processed in the same order as they arrive. Leandro -- _ -- 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] Remove Abandoned call
hello all, i have two asterisk server for call transfer and one more asterisk server for agent login(server_X) where agent take the call. server_A and server_B server_A is connected with pri and configure with 60 channel for call transfer into server_X server_B is connected with pri and configure with 30 channel for call transfer into server_X my query is that some time two call originate same time from two different server_A and server_B and hit into server_X and one call is abandoned and another one have taken by the agent But i don't want to abandoned the call, I want to set the priority, supposed to server_A and server_B call originate same time server_X take the call from server_A first and then take the call server_B after 1 sec please guide me 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
[asterisk-users] Web based Click to Call Application
Dear All, I want to develop click to call(C2C) web based application.Is there any study material. I will really appreciate your help, thank you. 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] Web based Click to Call Application
On Fri, Nov 9, 2012 at 4:11 PM, OCEANET - Cédric BASSAGET ced...@oceanet.com wrote: Or use a php socket and the AMI. Cédric Le 09/11/2012 11:39, A J Stiles a écrit : On Friday 09 November 2012, akhilesh chand wrote: Dear All, I want to develop click to call(C2C) web based application.Is there any study material. I will really appreciate your help, thank you. Look into call files. Basically, you inject a file into the folder /var/spool/asterisk/outgoing/ and this sets up a call for you. And search the archives; because I remember posting a simple click-to-call example script on this list, sometime back this Summer just gone. -- OCEANET --**--**--- [AGENCE DU MANS] 7, rue des Frênes ZAC de la Pointe 72190 SARGE LES LE MANS [t] +33 (0)2.43.50.26.50 [f] +33 (0)2.43.72.21.14 [AGENCE D'ANGERS] 5, rue Fleming Angers Technopole 49066 ANGERS [t] +33 (0)2.41.19.28.65 [f] +33 (0)2.52.19.22.00 http://www.oceanet.com http://www.oceanet-telecom.com I'm basically use Asterisk::Manager package ,php and perl. 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-**usershttp://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] Installation Problem with Asterisk 1.6
Hello, I am working with CentOS 5.3, asterisk 1.6.2.24 ,Whenever i executemake command, i got the following error when installing asterisk: . make[2]: *** No rule to make target `anaFilter.o', needed by `libilbc.a'. Stop. make[1]: *** [ilbc/libilbc.a] Error 2 make: *** [codecs] Error 2 i will really appreciate your help, thank you. 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
[asterisk-users] Installation Problem with asterisk 1.6
Dear All, I'm installing the asterisk-1.6.2.24 in Centos 5.3, whenever i'm running following command ./configure I got below error: configure: *** XML documentation will not be available because the 'libxml2' development package is missing. configure: *** Please run the 'configure' script with the '--disable-xmldoc' parameter option configure: *** or install the 'libxml2' development package. I have installed already libxml2 in current os Please help me. 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] Installation Problem with asterisk 1.6
Hello, I am working with CentOS 5.3, asterisk 1.6.2.24 and i have downloaded the ilbc codec (all the .h and .c required) but i think the Makefile is not appropriate (it is not even complete as the one of the lpc10). so i got the following error when installing asterisk: . make[2]: *** No rule to make target `anaFilter.o', needed by `libilbc.a'. Stop. make[1]: *** [ilbc/libilbc.a] Error 2 make: *** [codecs] Error 2 i will really appreciate your help, thank you. -- _ -- 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] (no subject)
Hi, I'm not able to configure 8 port card whenever I configure it is showing fatal: error inserting wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown symbol in module, or unknown parameter Please help. 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] (no subject)
Thanks ajs On Monday, July 30, 2012, A J Stiles wrote: On Monday 30 July 2012, akhilesh chand wrote: Hi, I'm not able to configure 8 port card whenever I configure it is showing fatal: error inserting wct4xxp(/lib/modules/2.6.18-128.el5/dahdi/wct4xxp/wct4xxp.ko):unknown symbol in module, or unknown parameter It sounds as though you need to recompile DAHDI-Linux. (Did you compile it before you acquired this card?) Just download the latest DAHDI package Source Code, and compile and install it. If you didn't compile your own kernel from Source Code, then you will also need the package kernel-devel (on Fedora / CentOS) or linux-headers (on Ubuntu). -- AJS Price Engines Ltd. DDI: 01283 707058. -- AJS Answers come *after* questions. -- _ -- 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] Regrading Speech Recognition.
Hi, I want to develop a IVR application that repond to speech input from the caller in asterisk. For example, imagine a caller who wants to speak with Ram Kumar. On a traditional IVR/auto attendant, the caller may be entering “76484” to spell “Kumar” and the system may respond with: “Press 1 for Radhe Kumar, 2 for Shyam Kumar, 3 for Krishan Kumar, ..., 5 for Ram Kumar.” The caller can simply say “Ram Kumar” and conversation can be established much more quickly. Is there any article or link regrading the same please guide me. Regrads Thanks 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] Regrading Speech Recognition.
ok,how can i develop with short vocab like sales,support,etc. I have read many article but I'm not able to pick the right point, how can i develop or configure speech reorganization with asterisk. Is there any article or link please share and guide me. Regards Akhilesh On Thu, Jul 5, 2012 at 12:29 PM, Mitul Limbani mi...@enterux.in wrote: Things that look simple r quite complex to build :-) Indian Accent ASR on proper names is herculean task. No speech recognition known to mankind as of date can handle so many dialects being spoken in India, so in short what you want is nice to have, but nearly impossible to develop. Better try with short vocab on generic words (sales, support, etc.) Mitul On Jul 5, 2012 12:23 PM, akhilesh chand omakhileshch...@gmail.com wrote: Hi, I want to develop a IVR application that repond to speech input from the caller in asterisk. For example, imagine a caller who wants to speak with Ram Kumar. On a traditional IVR/auto attendant, the caller may be entering “76484” to spell “Kumar” and the system may respond with: “Press 1 for Radhe Kumar, 2 for Shyam Kumar, 3 for Krishan Kumar, ..., 5 for Ram Kumar.” The caller can simply say “Ram Kumar” and conversation can be established much more quickly. Is there any article or link regrading the same please guide me. Regrads Thanks 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
[asterisk-users] How to play different different hold music.
Dear All, I have two server 'A' and 'B' . In Server 'A', five different ivr (Sevices) is playing and call is *forwarding *into Server 'B'. Server 'B' basically use for agent login(Extension). I want to play different hold music(Server 'B') bases on the corresponding services which is running into server 'A'. A single agent takes the call from different different services but hold music is play astrisk own by default. Is there any way to play different hold music bases on services which run into server A. I have some changes into musiconhold.conf (server B) but problem is no solve. please help me. 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] How to play different different hold music.
hi,
Server A
extentsion.conf
exten = N,n,Set(Service_name=Test)
exten = N,n,Dial(IAX2/
server2:server2@192.168.14.112/${result},${Service_name})
but Server B doesn't identify service_name.
Server B
iax.conf
[general]
register = server1:server1@192.168.14.110
[server2]
type=friend
user=server2
secret=server2
host=dynamic
context=outgoing
auth=md5
trunk=yes
extentsion.conf
[outgoing]
exten = _X.,1,Set(_CALLTIME=${STRFTIME(,Asia/Calcutta,%d-%b-%y-%H-%M-%S)})
exten = _X.,n,Set(CHANNEL(musicclass)=${Service_name})
exten =
_X.,n,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})
exten = _X.,n,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})
exten = _X.,n,Set(FILENAME=${EXTEN}_${CALLERID(num)}_${CALLTIME}.WAV)
exten = _X.,n,Set(RECORDFILENAME=${RECSUBDIR}/${FILENAME})
exten = _X.,n,MixMonitor(${RECORDING_PATH}${RECORDFILENAME})
exten = _X.,n,Dial(SIP/${EXTEN},120);EXTEN=4004,4005,4006
exten = _X.,n,Hangup()
sip.conf
[4004]
type=friend
context=outbound
defaultuser=4004
secret=4004
callerid=EXT4
host=dynamic
nat=no
dtfmode=rfc2833
subscribecontext=outbound
canreinvite=no
[4005]
type=friend
context=outbound
defaultuser=4005
secret=4005
callerid=EXT5
host=dynamic
nat=no
dtfmode=rfc2833
subscribecontext=outbound
canreinvite=no
[4006]
type=friend
context=outbound
defaultuser=4006
secret=4006
callerid=EXT6
host=dynamic
nat=no
dtfmode=rfc2833
subscribecontext=outbound
canreinvite=no
[ccm100]
type = friend
context = outgoing
host = 192.168.14.91
disallow = all
allow = ulaw
allow = alaw
nat=yes
canreinvite = yes
qualify = yes
On Tue, Jul 3, 2012 at 7:46 PM, Danny Nicholas da...@debsinc.com wrote:
Since you’re using IAX2 to contact Server B, you can use channel
variables to control the moh class. There was a good thread in June on
this. An “easier” way however would be to have each service dial a
different IAX number, then have each IAX number on server B select it’s MOH
Class.
Server A
[service1]
Exten = N,1,Set(Service_name=service1)
Exten = N,n,Dial(IAX2,server2:1234)
[service2]
Exten = N,1,Set(Service_name=service2)
Exten = N,n,Dial(IAX2,server2:3456)
** **
Server B
[default]
Exten = N,1,Verbose(start)
Exten = N,1234,answer()
Exten = N,n,Set(MOHClass=1)
Exten = N,3456,answer()
Exten = N,n,Set(MOHClass=2)
** **
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *akhilesh chand
*Sent:* Tuesday, July 03, 2012 9:11 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] How to play different different hold
music.
** **
hi,
Server A extentsion.conf
exten = N,n,Set(Service_name=Test)
exten = N,n,Dial(IAX2/
server2:server2@192.168.14.112/${result},${Service_name}http://server2:server2@192.168.14.112/$%7bresult%7d,$%7bService_name%7d
)
but Server B doesn't identify service_name.
extentsion.conf
[outgoing]
exten = _X.,1,Set(_CALLTIME=${STRFTIME(,Asia/Calcutta,%d-%b-%y-%H-%M-%S)})
exten = _X.,1,Set(CHANNEL(musicclass)=${Service_name})
exten =
_X.,n,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})*
***
exten = _X.,n,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})**
**
exten = _X.,n,Set(FILENAME=${EXTEN}_${CALLERID(num)}_${CALLTIME}.WAV)
exten = _X.,n,Set(RECORDFILENAME=${RECSUBDIR}/${FILENAME})
exten = _X.,n,MixMonitor(${RECORDING_PATH}${RECORDFILENAME})
exten = _X.,n,Dial(SIP/${EXTEN},120)
exten = _X.,n,Hangup()
Regards
Akhilesh
** **
On Tue, Jul 3, 2012 at 6:00 PM, akhilesh chand omakhileshch...@gmail.com
wrote:
Dear All,
I have two server 'A' and 'B' . In Server 'A', five different
ivr (Sevices) is playing and call is *forwarding *into Server 'B'. Server
'B' basically use for agent login(Extension).
I want to play different hold music(Server 'B') bases on the corresponding
services which is running into server 'A'.
A single agent takes the call from different different services but hold
music is play astrisk own by default.
Is there any way to play different hold music bases on services which
run into server A.
I have some changes into musiconhold.conf (server B) but problem is no
solve.
please help me.
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
