Rudolf Ladyzhenskii
2005-Feb-25 23:08 UTC
[Asterisk-Users] Seting up for afirst time -- can not call
Hi, all I am setting up Asterisk for the first time and have some problems. Setup is very simple -- Astersik box and two Polycom SP300 phones. I will add bells and whistles as I go, at the moment things are very simple. No TFTP servers, so phones run with their default configuration. I set up IP addresses, netmask and gateway IPs manually on the phones. Now, I have read of problems with polycom phones. Here is my sip.conf file: ; SIP configuration file [general] port=5060 bindaddr=0.0.0.0 context=default [polycom_sp300_ext101] type=user host=192.168.1.101 secret=101 context=default [polycom_sp300_ext101] type=peer secret=101 host=192.168.1.101 context=default callerid="Ext 101" [polycom_sp300_ext102] type=user host=192.168.1.102 secret=101 context=default [polycom_sp300_ext102] type=peer secret=102 host=192.168.1.102 context=default callerid="Ext 102" First question is about the secret. Should I set up something on teh phone? Is it phone password (default 456)? Now, I am trying to have some extensions. So I have edited the extensions.conf file and changed the [default] section: [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; ;include => demo exten => 101,1,Dial,(SIP/polycom_sp300_ext101) exten => 102,1,Dial,(SIP/polycom_sp300_ext102) The rest of the file is "as is" as it came with Asterisk. Now I run 'reload' command as CLI. Is ist all I have to do to be able to call between those two phones? If I try to call from one phone to another, after I enter first two digits '10', I get "connecting" on phone screen and instant busy tone. Any help is greatly appreciated. Thanks, Rudolf
Race Vanderdecken
2005-Feb-26 13:16 UTC
[Asterisk-Users] Seting up for afirst time -- can not call
Okay, About the secret, comment out the line. You do have to set the secret in the phone. So when the INVITE is exchanged Asterisk will ask the phone for the secret, no secret, no connection. I don't have a polycom phone so that is about all I can help with. Oh yeah, you need a context [from-sip] [from-sip] exten => 101,1,Dial,(SIP/polycom_sp300_ext101) exten => 102,1,Dial,(SIP/polycom_sp300_ext102) As far as I know when the calls come into asterisk via SIP asterisk checks the [from-sip context] be default. Remember that Asterisk is first a PBX, then a VoIP/SIP Server. SIP is sort of step-child status in Asterisk. Race "The Tyrant" Vanderdecken. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rudolf Ladyzhenskii Sent: Saturday, February 26, 2005 1:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Seting up for afirst time -- can not call Hi, all I am setting up Asterisk for the first time and have some problems. Setup is very simple -- Astersik box and two Polycom SP300 phones. I will add bells and whistles as I go, at the moment things are very simple. No TFTP servers, so phones run with their default configuration. I set up IP addresses, netmask and gateway IPs manually on the phones. Now, I have read of problems with polycom phones. Here is my sip.conf file: ; SIP configuration file [general] port=5060 bindaddr=0.0.0.0 context=default [polycom_sp300_ext101] type=user host=192.168.1.101 secret=101 context=default [polycom_sp300_ext101] type=peer secret=101 host=192.168.1.101 context=default callerid="Ext 101" [polycom_sp300_ext102] type=user host=192.168.1.102 secret=101 context=default [polycom_sp300_ext102] type=peer secret=102 host=192.168.1.102 context=default callerid="Ext 102" First question is about the secret. Should I set up something on teh phone? Is it phone password (default 456)? Now, I am trying to have some extensions. So I have edited the extensions.conf file and changed the [default] section: [default] ; ; By default we include the demo. In a production system, you ; probably don't want to have the demo there. ; ;include => demo exten => 101,1,Dial,(SIP/polycom_sp300_ext101) exten => 102,1,Dial,(SIP/polycom_sp300_ext102) The rest of the file is "as is" as it came with Asterisk. Now I run 'reload' command as CLI. Is ist all I have to do to be able to call between those two phones? If I try to call from one phone to another, after I enter first two digits '10', I get "connecting" on phone screen and instant busy tone. Any help is greatly appreciated. Thanks, Rudolf _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users