Gunnar Þ. Gestsson
2004-Nov-02 08:06 UTC
[Asterisk-Users] Allied Telesyn Residential Gateway 613
Hello. Has anyone successfully connected Allied Telesyn Residential Gateway 613 via SIP to Asterisk ? If so, would you mind sharing the config of both ends ? Regards, Gunnar Gestsson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041102/edf11045/attachment.htm
Kanuri, Seshu (Company IT)
2004-Nov-02 08:11 UTC
[Asterisk-Users] Allied Telesyn Residential Gateway 613
I spoke to them a couple of months ago to buy their SIP Boards so that I can make my own Dialup SIP Phone and I was told that their residential gateway products are now out of life. Their SIP Implementation has not worked well and hence they decided to close that business. I dont know their current status though Seshu Kanuri _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gunnar ?. Gestsson Sent: Tuesday, November 02, 2004 10:07 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Allied Telesyn Residential Gateway 613 Hello. Has anyone successfully connected Allied Telesyn Residential Gateway 613 via SIP to Asterisk ? If so, would you mind sharing the config of both ends ? Regards, Gunnar Gestsson -------------------------------------------------------- NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041102/5dabbfc2/attachment.htm
Gunnar Þ. Gestsson
2004-Nov-02 08:27 UTC
[Asterisk-Users] Allied Telesyn Residential Gateway 613
There is not a word about that on the website: http://www.alliedtelesyn.co.uk/en-gb/products/cat/family.asp?cid=16&fid=146 Gunnar Gestsson ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Kanuri, Seshu (Company IT) Sent: 2. n?vember 2004 15:11 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Allied Telesyn Residential Gateway 613 I spoke to them a couple of months ago to buy their SIP Boards so that I can make my own Dialup SIP Phone and I was told that their residential gateway products are now out of life. Their SIP Implementation has not worked well and hence they decided to close that business. I dont know their current status though Seshu Kanuri ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Gunnar ?. Gestsson Sent: Tuesday, November 02, 2004 10:07 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Allied Telesyn Residential Gateway 613 Hello. Has anyone successfully connected Allied Telesyn Residential Gateway 613 via SIP to Asterisk ? If so, would you mind sharing the config of both ends ? Regards, Gunnar Gestsson ________________________________ NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041102/c30c9025/attachment.htm
>Hello.>Has anyone successfully connected Allied Telesyn Residential Gateway >613 via SIP to Asterisk ? >If so, would you mind sharing the config of both ends ? >Regards, >Gunnar Gestsson Hi, Asterisk sip.conf example for extension 1000 password 1000: [8567] type=friend username=1000 secret=1000 host=dynamic mailbox=1000 context=default canreinvite=no dtmfmode=inband RG613 config with asterisk running at 192.168.1.10 as an example with g711a codec in Norway: (your digitmap may have to be changed) ip set interface ip0 dhcp enabled dhcpclient update voip sip protocol enable voip sip protocol set netinterface ip0 voip sip proxyserver create asterisk-1 contact 192.168.1.10:5060/udp voip sip locationserver create asterisk-1 contact 192.168.1.10:5060/udp voip ep analogue create tel1 type al-fxs-del physical-port tel1 voip ep analogue set tel1 lec 32 voip ep analog set tel1 codecs g711a voip ep analog set tel1 country norway voip ep analog set tel1 digitmap [1-8]xxx|9x|000x.T|011x|00[1-9]xxx|01[2-9]x.T|0[2-9]xxxxxxx voip ep analog set tel1 vad off voip ep analogue set tel1 clip BELL voip sip user create 1000 address 1000 authentication 1000:1000 domain 192.168.1.10 transport udp voip sip user add 1000 port tel1 voip ep analog set tel1 cfwd enable all-calls on-prefix *21* on-suffix # off-prefix #21# voip ep analog set tel1 cfwd enable on-busy on-prefix *67* on-suffix # off-prefix #67# voip ep analog set tel1 cfwd enable on-no-answer on-prefix *61* on-suffix # off-prefix #61# voip ep analog set tel1 cfwd on-no-answer timeout 20 system config create 1000.cfg system config set 1000.cfg Regards, Tor