Re: [asterisk-users] PAP2T provisioning via SRV record?
On 06/16/2011 07:58 AM, Mike Diehl wrote: Well, I ran a simple test by trying to configure the second port to use the DNS SRV record, as described below. Here is what I have: (sanitized) == Proxy_2_ diehlnet.com/Proxy_2_ Outbound_Proxy_2_ fqdn/Outbound_Proxy_2_ Display_Name_2_ ua=nausername/Display_Name_2_ User_ID_2_ ua=nausername/User_ID_2_ Password_2_ ua=napassword/Password_2_ Use_Auth_ID_2_ ua=naYes/Use_Auth_ID_2_ Auth_ID_2_ ua=nausername/Auth_ID_2_ Use_DNS_SRV_2_yes/Use_DNS_SRV_2_ == With this configuration, the second port does NOT register. A sniffer trace on the inside interface of my router gives me some clues, though: 23:54:34.906089 IP 10.0.1.87.60198 208.67.222.222.53: 1+ A? diehlnet.com. (30) 23:54:35.102409 IP 208.67.222.222.53 10.0.1.87.60198: 1 1/0/0 A 173.10.242.193 (46) 23:54:35.104484 IP 10.0.1.87.5061 173.10.242.193.5060: UDP, length: 527 23:54:35.104553 IP 173.10.242.193 10.0.1.87: icmp 556: 173.10.242.193 udp port 5060 unreachable It seems that the device is still looking for an A record for diehlnet.com, which does exist. It should be looking for the SRV record. What am I missing? Make sure you also have set: DNS_SRV_Auto_Prefix_2_ ua=naYes/DNS_SRV_Auto_Prefix_2_ From the manual: Enables the phone to automatically prepend the proxy or outbound proxy name with _sip._udp when performing a DNS SRV lookup on that name. Defaults to no. Best regards, Marius -- _ -- 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] PAP2T provisioning via SRV record?
On 6/14/2011 5:08 AM, Paul Hayes wrote: On 13/06/11 19:44, Mike Diehl wrote: Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: Proxy_1_ _sip._udp.example.com/Proxy_1_ However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine. What else do I need to do to get the PAP to work this way? TIA, There's a setting in the Line 1 and Line 2 page called Use DNS SRV which is set to No by default for some reason. Set this to yes and set the proxy to example.com. So something like: Use_DNS_SRV_1_yes/Use_DNS_SRV_1_ Proxy_1_example.com/Proxy_1_ In addition to this, you also need to set the DNS_SRV_Auto_Prefix to 'yes'. DNS_SRV_Auto_Prefix_1_yes/DNS_SRV_Auto_Prefix_1_ cheers, Paul. -- _ -- 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 -- Technical Support http://www.neuroredes.com -- _ -- 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] PAP2T provisioning via SRV record?
On Thursday 16 June 2011 8:10:38 am Marius Pedersen wrote: On 06/16/2011 07:58 AM, Mike Diehl wrote: Well, I ran a simple test by trying to configure the second port to use the DNS SRV record, as described below. Here is what I have: (sanitized) == Proxy_2_ diehlnet.com/Proxy_2_ Outbound_Proxy_2_ fqdn/Outbound_Proxy_2_ Display_Name_2_ ua=nausername/Display_Name_2_ User_ID_2_ ua=nausername/User_ID_2_ Password_2_ ua=napassword/Password_2_ Use_Auth_ID_2_ ua=naYes/Use_Auth_ID_2_ Auth_ID_2_ ua=nausername/Auth_ID_2_ Use_DNS_SRV_2_yes/Use_DNS_SRV_2_ == With this configuration, the second port does NOT register. A sniffer trace on the inside interface of my router gives me some clues, though: 23:54:34.906089 IP 10.0.1.87.60198 208.67.222.222.53: 1+ A? diehlnet.com. (30) 23:54:35.102409 IP 208.67.222.222.53 10.0.1.87.60198: 1 1/0/0 A 173.10.242.193 (46) 23:54:35.104484 IP 10.0.1.87.5061 173.10.242.193.5060: UDP, length: 527 23:54:35.104553 IP 173.10.242.193 10.0.1.87: icmp 556: 173.10.242.193 udp port 5060 unreachable It seems that the device is still looking for an A record for diehlnet.com, which does exist. It should be looking for the SRV record. What am I missing? Make sure you also have set: DNS_SRV_Auto_Prefix_2_ ua=naYes/DNS_SRV_Auto_Prefix_2_ From the manual: Enables the phone to automatically prepend the proxy or outbound proxy name with _sip._udp when performing a DNS SRV lookup on that name. Defaults to no. That was the final change that made it work. Wish I had the manual you have. Mine didn't say much. Thank you. -- Take care and have fun, Mike Diehl. -- _ -- 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] PAP2T provisioning via SRV record?
On 06/16/2011 04:49 PM, Mike Diehl wrote: On Thursday 16 June 2011 8:10:38 am Marius Pedersen wrote: On 06/16/2011 07:58 AM, Mike Diehl wrote: Well, I ran a simple test by trying to configure the second port to use the DNS SRV record, as described below. Here is what I have: (sanitized) == Proxy_2_ diehlnet.com/Proxy_2_ Outbound_Proxy_2_ fqdn/Outbound_Proxy_2_ Display_Name_2_ ua=nausername/Display_Name_2_ User_ID_2_ ua=nausername/User_ID_2_ Password_2_ ua=napassword/Password_2_ Use_Auth_ID_2_ ua=naYes/Use_Auth_ID_2_ Auth_ID_2_ ua=nausername/Auth_ID_2_ Use_DNS_SRV_2_yes/Use_DNS_SRV_2_ == With this configuration, the second port does NOT register. A sniffer trace on the inside interface of my router gives me some clues, though: 23:54:34.906089 IP 10.0.1.87.60198 208.67.222.222.53: 1+ A? diehlnet.com. (30) 23:54:35.102409 IP 208.67.222.222.53 10.0.1.87.60198: 1 1/0/0 A 173.10.242.193 (46) 23:54:35.104484 IP 10.0.1.87.5061 173.10.242.193.5060: UDP, length: 527 23:54:35.104553 IP 173.10.242.193 10.0.1.87: icmp 556: 173.10.242.193 udp port 5060 unreachable It seems that the device is still looking for an A record for diehlnet.com, which does exist. It should be looking for the SRV record. What am I missing? Make sure you also have set: DNS_SRV_Auto_Prefix_2_ ua=naYes/DNS_SRV_Auto_Prefix_2_ From the manual: Enables the phone to automatically prepend the proxy or outbound proxy name with _sip._udp when performing a DNS SRV lookup on that name. Defaults to no. That was the final change that made it work. Wish I had the manual you have. Mine didn't say much. If you take a look at the Cisco Small Business SPA300 Series, SPA500 Series, and WIP310 IP Phone Administration Guide you can find information on most of the settings there as the PAP2T and SPA-series configuration options are quite similar. Best regards, Marius -- _ -- 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] PAP2T provisioning via SRV record?
Well, I ran a simple test by trying to configure the second port to use the DNS SRV record, as described below. Here is what I have: (sanitized) == Proxy_2_ diehlnet.com /Proxy_2_ Outbound_Proxy_2_ fqdn /Outbound_Proxy_2_ Display_Name_2_ ua=nausername/Display_Name_2_ User_ID_2_ ua=nausername/User_ID_2_ Password_2_ ua=napassword/Password_2_ Use_Auth_ID_2_ ua=naYes/Use_Auth_ID_2_ Auth_ID_2_ ua=nausername/Auth_ID_2_ Use_DNS_SRV_2_yes/Use_DNS_SRV_2_ == With this configuration, the second port does NOT register. A sniffer trace on the inside interface of my router gives me some clues, though: 23:54:34.906089 IP 10.0.1.87.60198 208.67.222.222.53: 1+ A? diehlnet.com. (30) 23:54:35.102409 IP 208.67.222.222.53 10.0.1.87.60198: 1 1/0/0 A 173.10.242.193 (46) 23:54:35.104484 IP 10.0.1.87.5061 173.10.242.193.5060: UDP, length: 527 23:54:35.104553 IP 173.10.242.193 10.0.1.87: icmp 556: 173.10.242.193 udp port 5060 unreachable It seems that the device is still looking for an A record for diehlnet.com, which does exist. It should be looking for the SRV record. What am I missing? Mike. On Tuesday 14 June 2011 3:08:33 am Paul Hayes wrote: On 13/06/11 19:44, Mike Diehl wrote: Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: Proxy_1_ _sip._udp.example.com/Proxy_1_ However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine. What else do I need to do to get the PAP to work this way? TIA, There's a setting in the Line 1 and Line 2 page called Use DNS SRV which is set to No by default for some reason. Set this to yes and set the proxy to example.com. So something like: Use_DNS_SRV_1_yes/Use_DNS_SRV_1_ Proxy_1_example.com/Proxy_1_ cheers, Paul. -- _ -- 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 -- Take care and have fun, Mike Diehl. -- _ -- 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] PAP2T provisioning via SRV record?
On 13/06/11 19:44, Mike Diehl wrote: Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: Proxy_1_ _sip._udp.example.com/Proxy_1_ However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine. What else do I need to do to get the PAP to work this way? TIA, There's a setting in the Line 1 and Line 2 page called Use DNS SRV which is set to No by default for some reason. Set this to yes and set the proxy to example.com. So something like: Use_DNS_SRV_1_yes/Use_DNS_SRV_1_ Proxy_1_example.com/Proxy_1_ cheers, Paul. -- _ -- 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] PAP2T provisioning via SRV record?
Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: Proxy_1_ _sip._udp.example.com /Proxy_1_ However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine. What else do I need to do to get the PAP to work this way? TIA, -- Take care and have fun, Mike Diehl. -- _ -- 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] PAP2T provisioning
Tom Moore schrieb: I'm not sure if this trick will work with this device, but I was able to pull down a spa8000's config by connecting to: http://ipaddress/admin/spacfg.xml Tom Hello, The spacfg.xml link doesnt work on a Pap2T but you could use this link to get the xml config from an Spa9xx with Firmware greater thatn 5.x. which is the same for pap2. I´ve attached you an complete XML file of an pap2 i´ve found. best regards Steve Smith -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // s...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - flat-profile Restricted_Access_Domains ua=na/Restricted_Access_Domains Enable_Web_Server ua=naYes/Enable_Web_Server Web_Server_Port ua=na80/Web_Server_Port Enable_Web_Admin_Access ua=naYes/Enable_Web_Admin_Access Admin_Passwd ua=naPASSWORD/Admin_Passwd User_Password ua=rwPASSWORD/User_Password DHCP ua=rwNo/DHCP Static_IP ua=rw192.168.1.9/Static_IP NetMask ua=rw255.255.255.0/NetMask Gateway ua=rw192.168.1.1/Gateway HostName ua=rwMyPAP2/HostName Domain ua=rwvuckUfonage.net/Domain Primary_DNS ua=rw192.168.1.11/Primary_DNS Secondary_DNS ua=rw192.168.1.1/Secondary_DNS DNS_Server_Order ua=naManual,DHCP/DNS_Server_Order DNS_Query_Mode ua=naSequential/DNS_Query_Mode Syslog_Server ua=na/Syslog_Server Debug_Server ua=na/Debug_Server Debug_Level ua=na99/Debug_Level Primary_NTP_Server ua=natime.vonage.net/Primary_NTP_Server Secondary_NTP_Server ua=nantp1-nyc.vonage.net/Secondary_NTP_Server Provision_Enable ua=naNo/Provision_Enable Resync_On_Reset ua=naYes/Resync_On_Reset Resync_Random_Delay ua=na2/Resync_Random_Delay Resync_Periodic ua=na1800/Resync_Periodic Resync_Error_Retry_Delay ua=na1800/Resync_Error_Retry_Delay Forced_Resync_Delay ua=na43200/Forced_Resync_Delay Resync_From_SIP ua=naYes/Resync_From_SIP Resync_After_Upgrade_Attempt ua=naYes/Resync_After_Upgrade_Attempt Resync_Trigger_1 ua=na$D ne $E/Resync_Trigger_1 Resync_Fails_On_FNFNo/Resync_Fails_On_FNF Profile_Rule ua=na ($O eq C)? (GPP_P = 2400; GPP_N = $O; GPP_O = end;)! | ($O eq B)? (GPP_P = 21; GPP_N = $O; GPP_O = C;)! |(GPP_P = 69; GPP_N = $O; GPP_O = B;)! /Profile_Rule Profile_Rule_B ua=na ($H eq upgactive and $UPGST ne 2 and $SWVER ge 2.0.10)? (GPP_H = ; GPP_G = $F;)! /Profile_Rule_B Profile_Rule_C ua=na (GPP_E = $D;)! [--key $K] tftp://192.168.1.11:$P/spa0014BFFC39C9.xml /Profile_Rule_C Profile_Rule_D ua=na(GPP_O = $N;)!/Profile_Rule_D Resync_Trigger_2 ua=na$PRVST eq 2 and $O ne end/Resync_Trigger_2 GPP_G ua=na/ GPP_H ua=na/ GPP_P ua=na/ GPP_A ua=na/GPP_A GPP_B ua=na/GPP_B GPP_C ua=na/GPP_C GPP_D ua=navbYdVfzzQZ/GPP_D GPP_E ua=na/ GPP_F ua=na PAP2-bin-03-01-06-LS.bin /GPP_F GPP_I ua=na/GPP_I GPP_J ua=na/GPP_J GPP_K ua=naMYFACTORYFRESHKEY/GPP_K GPP_L ua=na/GPP_L GPP_M ua=na/GPP_M GPP_SA ua=na/GPP_SA GPP_SB ua=na/GPP_SB GPP_SC ua=na/GPP_SC GPP_SD ua=na/GPP_SD Log_Resync_Request_Msg ua=na$PN $MAC -- Requesting resync $SCHEME://$SERVIP:$PORT$PATH/Log_Resync_Request_Msg Log_Resync_Success_Msg ua=na$PN $MAC -- Successful resync $SCHEME://$SERVIP:$PORT$PATH/Log_Resync_Success_Msg Log_Resync_Failure_Msg ua=na$PN $MAC -- Resync failed: $ERR/Log_Resync_Failure_Msg Report_Rule ua=na/Report_Rule Upgrade_Enable ua=naYes/Upgrade_Enable Upgrade_Error_Retry_Delay ua=na7200/Upgrade_Error_Retry_Delay Downgrade_Rev_Limit ua=na/Downgrade_Rev_Limit Upgrade_Rule ua=na (!3.1.22)?tftp://192.168.1.104/PAP2-3-1-22-LS.bin /Upgrade_Rule Log_Upgrade_Request_Msg ua=na$PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH/Log_Upgrade_Request_Msg Log_Upgrade_Success_Msg ua=na$PN $MAC -- Successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR/Log_Upgrade_Success_Msg Log_Upgrade_Failure_Msg ua=na$PN $MAC -- Upgrade failed: $ERR/Log_Upgrade_Failure_Msg Max_Forward ua=na70/Max_Forward Max_Redirection ua=na5/Max_Redirection Max_Auth ua=na2/Max_Auth SIP_User_Agent_Name ua=na$MAU $VERSION/SIP_User_Agent_Name SIP_Server_Name ua=na$MAU $VERSION/SIP_Server_Name SIP_Accept_Language ua=na/SIP_Accept_Language DTMF_Relay_MIME_Type ua=naapplication/dtmf-relay/DTMF_Relay_MIME_Type Hook_Flash_MIME_Type ua=naapplication/hook-flash/Hook_Flash_MIME_Type Remove_Last_Reg ua=naNo/Remove_Last_Reg Use_Compact_Header ua=naNo/Use_Compact_Header SIP_T1 ua=na2/SIP_T1 SIP_T2 ua=na32/SIP_T2 SIP_T4 ua=na5/SIP_T4 SIP_Timer_B ua=na6/SIP_Timer_B SIP_Timer_F ua=na31/SIP_Timer_F SIP_Timer_H ua=na32/SIP_Timer_H SIP_Timer_D ua=na32/SIP_Timer_D SIP_Timer_J ua=na32/SIP_Timer_J INVITE_Expires ua=na240/INVITE_Expires ReINVITE_Expires ua=na30/ReINVITE_Expires Reg_Min_Expires ua=na1/Reg_Min_Expires Reg_Max_Expires ua=na7200/Reg_Max_Expires Reg_Retry_Intvl ua=na60/Reg_Retry_Intvl Reg_Retry_Long_Intvl
Re: [asterisk-users] PAP2T provisioning
On Wed, 21 Jan 2009, Stefan Schmidt wrote: Tom Moore schrieb: I'm not sure if this trick will work with this device, but I was able to pull down a spa8000's config by connecting to: http://ipaddress/admin/spacfg.xml Tom Hello, The spacfg.xml link doesnt work on a Pap2T but you could use this link to get the xml config from an Spa9xx with Firmware greater thatn 5.x. which is the same for pap2. I?ve attached you an complete XML file of an pap2 i?ve found. best regards Steve Smith Wow, that's perfect! Thanks! j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning
- Original Message - From: Jeff LaCoursiere j...@jeff.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 21, 2009 9:55 AM Subject: Re: [asterisk-users] PAP2T provisioning On Wed, 21 Jan 2009, Stefan Schmidt wrote: Tom Moore schrieb: I'm not sure if this trick will work with this device, but I was able to pull down a spa8000's config by connecting to: http://ipaddress/admin/spacfg.xml Tom Hello, The spacfg.xml link doesnt work on a Pap2T but you could use this link to get the xml config from an Spa9xx with Firmware greater thatn 5.x. which is the same for pap2. I?ve attached you an complete XML file of an pap2 i?ve found. best regards Steve Smith Wow, that's perfect! Thanks! j I know it's pretty much a given, but don't forget to edit/remove the provisioning info. I'd hate to see someone's open device locked. Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning
I know it's pretty much a given, but don't forget to edit/remove the provisioning info. I'd hate to see someone's open device locked. Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users the file should be clean, i´ve download it by myself from somewhere, but its very helpfull cause all possible parameter are in there. best regards steve -- Für weitere Fragen stehen wir gerne unter v...@sil.at oder 059944 - 2440 zur Verfügung. Mit freundlichen Grüssen -- Stefan Schmidt Sysadmin/VOIP // s...@sil.at // Tel 059944-2440// - SILVER SERVER GmbH // Lorenz-Mandl-Gasse 33/1 // A-1160 Wien // Fax 059944-9000 // www.sil.at // - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning
On Wed, 21 Jan 2009, Stefan Schmidt wrote: I know it's pretty much a given, but don't forget to edit/remove the provisioning info. I'd hate to see someone's open device locked. Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users the file should be clean, i?ve download it by myself from somewhere, but its very helpfull cause all possible parameter are in there. best regards steve I am planning to cut out just about everything and only apply the changes from the factory defaults that are required to register it with a basic Trixbox installation. I'll post the results when I am happy with it. Its actually part of a larger project to produce a set of Polycom and Linksys ATA XML files from a flat text file that describes extension numbers, names, device types, and MAC address assignments. Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning
Jeff LaCoursiere wrote:
On Wed, 21 Jan 2009, Stefan Schmidt wrote:
I know it's pretty much a given, but don't forget to edit/remove the
provisioning info. I'd hate to see someone's open device locked.
Tim
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
the file should be clean, i?ve download it by myself from somewhere, but
its very helpfull cause all possible parameter are in there.
best regards
steve
I am planning to cut out just about everything and only apply the changes
from the factory defaults that are required to register it with a basic
Trixbox installation. I'll post the results when I am happy with it. Its
actually part of a larger project to produce a set of Polycom and Linksys
ATA XML files from a flat text file that describes extension numbers,
names, device types, and MAC address assignments.
Why don't you just edit the Trixbox endpoint manager files. They
produce basic XML files for Linksys and Polycom phones. It is trivial
to add support for any Linksys ATA as well.
File is: /var/www/html/maint/modules/11_endpointcfg/endpoint_linksys.php
Just start by editing this line: $smarty-assign(phone_types,
array('SPA942' = 'SPA942', 'SPA941' = 'SPA941', 'SPA921' = 'SPA921',
'SPA922' = 'SPA922', 'SPA962' = 'SPA962','SPA2102' = 'SPA2102'));
...and add the devices you need. As you can see we already added the
SPA2102.
Andres
http://www.neuroredes.com
Cheers,
j
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning
On Wed, 21 Jan 2009, Andres wrote:
Why don't you just edit the Trixbox endpoint manager files. They
produce basic XML files for Linksys and Polycom phones. It is trivial
to add support for any Linksys ATA as well.
File is: /var/www/html/maint/modules/11_endpointcfg/endpoint_linksys.php
Just start by editing this line: $smarty-assign(phone_types,
array('SPA942' = 'SPA942', 'SPA941' = 'SPA941', 'SPA921' = 'SPA921',
'SPA922' = 'SPA922', 'SPA962' = 'SPA962','SPA2102' = 'SPA2102'));
...and add the devices you need. As you can see we already added the
SPA2102.
I looked at this briefly a while back and although it does give you a nice
GUI to manage your endpoints, I still prefer my text file method.
Consider it this way - to setup a new 50 station system I generally get a
spreadsheet from the client with names and extension numbers. I can
quickly turn this into a text file that I currently process with a csh
script that turns out the Polycom XML files and creates the extensions in
the Trixbox database.
Otherwise I would be hand entering all the Trixbox extensions, then again
hand entering all the devices with MAC addresses to tie them to the
extensions.
A new project has 40 PAP2Ts involved, and I hadn't until yesterday ever
tried to centrally manage these things, because usually the project only
has a few involved. Unlike the Polycom interface the Linksys web
interface is actually fairly decent IMO, and I have always gotten by that
way.
If I need to make a change in the future, to say the dialplan, I make a
change to one XML file that is included by all the Polycom phones, and
reboot them. What would be the procedure through the endpoint manager?
For grins I poked around looking for the source of the Linksys template,
as I noticed it doesn't include the dialplan tag. I couldn't find it...
do you know where it is?
Cheers!
j
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning
Jeff LaCoursiere wrote:
On Wed, 21 Jan 2009, Andres wrote:
Why don't you just edit the Trixbox endpoint manager files. They
produce basic XML files for Linksys and Polycom phones. It is trivial
to add support for any Linksys ATA as well.
File is:
/var/www/html/maint/modules/11_endpointcfg/endpoint_linksys.php
Just start by editing this line: $smarty-assign(phone_types,
array('SPA942' = 'SPA942', 'SPA941' = 'SPA941', 'SPA921' = 'SPA921',
'SPA922' = 'SPA922', 'SPA962' = 'SPA962','SPA2102' = 'SPA2102'));
...and add the devices you need. As you can see we already added the
SPA2102.
I looked at this briefly a while back and although it does give you a
nice GUI to manage your endpoints, I still prefer my text file method.
Consider it this way - to setup a new 50 station system I generally
get a spreadsheet from the client with names and extension numbers. I
can quickly turn this into a text file that I currently process with a
csh script that turns out the Polycom XML files and creates the
extensions in the Trixbox database.
Otherwise I would be hand entering all the Trixbox extensions, then
again hand entering all the devices with MAC addresses to tie them to
the extensions.
A new project has 40 PAP2Ts involved, and I hadn't until yesterday
ever tried to centrally manage these things, because usually the
project only has a few involved. Unlike the Polycom interface the
Linksys web interface is actually fairly decent IMO, and I have always
gotten by that way.
If I need to make a change in the future, to say the dialplan, I make
a change to one XML file that is included by all the Polycom phones,
and reboot them. What would be the procedure through the endpoint
manager?
It does not contemplate that option as it stands. I don't think it
would requiere more than a few lines of code to implement that with the
Endpoint Manager though. I do agree that it would be very useful to
have that feature.
For grins I poked around looking for the source of the Linksys
template, as I noticed it doesn't include the dialplan tag. I
couldn't find it... do you know where it is?
You need to add it. Just look at where the flat-profile section is
and add it like this example we have:
flat-profile
!-- Subscriber Information --
Debug_Server ua=na192.168.1.49/Debug_Server
Debug_Level ua=na3/Debug_Level
Syslog_Server ua=na192.168.1.49/Syslog_Server
Text_Logo ua=naCustomer/Text_Logo
Select_Background_Picture ua=naBMP
Picture/Select_Background_Picture
BMP_Picture_Download_URL
ua=natftp://192.168.1.49/customer.bmp/BMP_Picture_Download_URL
Display_Name_1_ ua=na$UserID/Display_Name_1_
User_ID_1_ ua=na$UserID/User_ID_1_
Password_1_ ua=na$Password/Password_1_
Dial_Plan_1_ua=na([2-8][1-9]x|[1-6]xx|30x|9xxx|01[2-9]x|0[2-9].|*0xx|*800xxx)/Dial_Plan_1_
/flat-profile
This basically hard codes the dialplan. You can probably edit the html
file to actually enable you to input the values graphically, but that
was more involved than what we needed to do.
Andres
http://www.neuroredes.com
Cheers!
j
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning
Dear Jeff, Please find attached the autoprovisionning file of the PAP2T...Kindly let me know if you need to know how to use it Regards On Wed, Jan 21, 2009 at 1:04 AM, Jeff LaCoursiere j...@jeff.net wrote: Anyone have an example XML file for the PAP2T? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users # *** # *** Sipura SPA Series Configuration Parameters # *** # *** System Configuration Restricted_Access_Domains ; Enable_Web_Server Yes ; Web_Server_Port 80 ; Enable_Web_Admin_Access Yes ; Admin_Passwd ; User_Password ! ; # *** Internet Connection Type DHCP! Yes ; Static_IP ! ; NetMask ! ; Gateway ! ; # *** Optional Network Configuration HostName! ; Domain ! ; Primary_DNS ! ; Secondary_DNS ! ; DNS_Server_Order Manual ; # options: Manual/Manual,DHCP/DHCP,Manual DNS_Query_ModeParallel ; # options: Parallel/Sequential Syslog_Server ; Debug_Server ; Debug_Level 0 ; # options: 0/1/2/3/0+H/1+H/2+H/3+H/99 Primary_NTP_Server ; Secondary_NTP_Server ; # *** Configuration Profile Provision_Enable Yes ; Resync_On_Reset Yes ; Resync_Random_Delay 2 ; Resync_Periodic 30 ; Resync_Error_Retry_Delay 3600 ; Forced_Resync_Delay 14400 ; Resync_From_SIP Yes ; Resync_After_Upgrade_Attempt Yes ; Resync_Trigger_1 ; Resync_Trigger_2 ; Resync_Fails_On_FNF No ; Profile_Rule http://domain.net/Config.cfg; ; Profile_Rule_B ; Profile_Rule_C ; Profile_Rule_D ; Log_Resync_Request_Msg$PN $MAC -- Requesting resync $SCHEME://$SERVIP:$PORT$PATH ; Log_Resync_Success_Msg$PN $MAC -- Successful resync $SCHEME://$SERVIP:$PORT$PATH ; Log_Resync_Failure_Msg$PN $MAC -- Resync failed: $ERR ; Report_Rule; # *** Firmware Upgrade Upgrade_EnableYes ; Upgrade_Error_Retry_Delay 3600 ; Downgrade_Rev_Limit; Upgrade_Rule ; Log_Upgrade_Request_Msg $PN $MAC -- Requesting upgrade $SCHEME://$SERVIP:$PORT$PATH ; Log_Upgrade_Success_Msg $PN $MAC -- Successful upgrade $SCHEME://$SERVIP:$PORT$PATH -- $ERR ; Log_Upgrade_Failure_Msg $PN $MAC -- Upgrade failed: $ERR ; # *** General Purpose Parameters GPP_A ; GPP_B ; GPP_C ; GPP_D ; GPP_E ; GPP_F ; GPP_G ; GPP_H ; GPP_I ; GPP_J ; GPP_K ; GPP_L ; GPP_M ; GPP_N ; GPP_O ; GPP_P ; GPP_SA ; GPP_SB ; GPP_SC ; GPP_SD ; # *** SIP Parameters Max_Forward 70 ; Max_Redirection 5 ; Max_Auth 2 ; SIP_User_Agent_Name $VERSION ; SIP_Server_Name $VERSION ; SIP_Accept_Language; DTMF_Relay_MIME_Type application/dtmf-relay ; Hook_Flash_MIME_Type application/hook-flash ; Remove_Last_Reg No ; Use_Compact_HeaderNo ; # *** SIP Timer Values (sec) SIP_T1.5 ; SIP_T24 ; SIP_T45 ; SIP_Timer_B 32 ; SIP_Timer_F 32 ; SIP_Timer_H 32 ; SIP_Timer_D 32 ; SIP_Timer_J 32 ; INVITE_Expires240 ; ReINVITE_Expires 30 ; Reg_Min_Expires 1 ; Reg_Max_Expires 7200 ; Reg_Retry_Intvl 30 ; Reg_Retry_Long_Intvl 1200 ; # *** Response Status Code Handling SIT1_RSC ; SIT2_RSC ; SIT3_RSC ; SIT4_RSC ;
Re: [asterisk-users] PAP2T provisioning
Hi, I think this kind of file requires Cisco's compiler... I have it working now with straight XML and am pretty happy with that. Thanks though! Cheers, j On Wed, 21 Jan 2009, michel freiha wrote: Dear Jeff, Please find attached the autoprovisionning file of the PAP2T...Kindly let me know if you need to know how to use it Regards On Wed, Jan 21, 2009 at 1:04 AM, Jeff LaCoursiere j...@jeff.net wrote: Anyone have an example XML file for the PAP2T? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PAP2T provisioning
Anyone have an example XML file for the PAP2T? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PAP2T provisioning
I'm not sure if this trick will work with this device, but I was able to pull down a spa8000's config by connecting to: http://ipaddress/admin/spacfg.xml Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, January 20, 2009 6:04 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] PAP2T provisioning Anyone have an example XML file for the PAP2T? Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
