[Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem
I have the follwoing setup: Asterisk SVN-tag-1.2.1-r7367 6 Polycom 500 Sip version 1.5.x 4 Sipura SPA3000 (not sure what build) (FXO port) All on flat single network, no NAT, and no gateways to reach each other. Sometimes (happens around 3 times a day, but sometimes far more often), while on the phone to an outside caller (on the PSTN using the FXO on the spa3k), the call dissconects from the polycom and goes thru the incoming extension for the sipura. In other words, astrisk at least as far as I can see from what gets executed in the DP (and maybe spa3k) sees this as if the follwoing has happened: 1. The polycom user hungup, 2. A new call came in on the spa3k. The follwoing is part of the log that I think might help: Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from channel: SIP/201-8ba1 Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels SIP/201-8ba1 and SIP/804-fd83 SIP/201 is the Polycom, while SIP/804 is the spa3k. If I'm losing a frame, is there a way to configure asterisk not to drop the channel? Or is this something the Polycom/Sipura are doing? FYI, asterisk is running on a VIA/MPIA platform. Thank You ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem
I have the follwoing setup: Asterisk SVN-tag-1.2.1-r7367 6 Polycom 500 Sip version 1.5.x 4 Sipura SPA3000 (not sure what build) (FXO port) All on flat single network, no NAT, and no gateways to reach each other. Sometimes (happens around 3 times a day, but sometimes far more often), while on the phone to an outside caller (on the PSTN using the FXO on the spa3k), the call dissconects from the polycom and goes thru the incoming extension for the sipura. In other words, astrisk at least as far as I can see from what gets executed in the DP (and maybe spa3k) sees this as if the follwoing has happened: 1. The polycom user hungup, 2. A new call came in on the spa3k. The follwoing is part of the log that I think might help: Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from channel: SIP/201-8ba1 Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels SIP/201-8ba1 and SIP/804-fd83 SIP/201 is the Polycom, while SIP/804 is the spa3k. If I'm losing a frame, is there a way to configure asterisk not to drop the channel? Or is this something the Polycom/Sipura are doing? FYI, asterisk is running on a VIA/MPIA platform. Pure guess is that something happened (unknown what) and the error messages posted above are the result of that, and not the root cause. Finding the root cause may require you to implement the syslog server and debug server options in the spa3k, and compare those log entries to what * records for log messages during a failure. Implementing the log functions on the spa3k does require a reboot. Their log messages are rather cryptic, but looking at keywords and timestamps might identify which box(es) are involved with the dropped calls. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem
For somereason I think it's the polycom, which means I need logging for the Polycom and not the spa. On 12/28/05, Rich Adamson [EMAIL PROTECTED] wrote: I have the follwoing setup: Asterisk SVN-tag-1.2.1-r7367 6 Polycom 500 Sip version 1.5.x 4 Sipura SPA3000 (not sure what build) (FXO port) All on flat single network, no NAT, and no gateways to reach each other. Sometimes (happens around 3 times a day, but sometimes far more often), while on the phone to an outside caller (on the PSTN using the FXO on the spa3k), the call dissconects from the polycom and goes thru the incoming extension for the sipura. In other words, astrisk at least as far as I can see from what gets executed in the DP (and maybe spa3k) sees this as if the follwoing has happened: 1. The polycom user hungup, 2. A new call came in on the spa3k. The follwoing is part of the log that I think might help: Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from channel: SIP/201-8ba1 Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels SIP/201-8ba1 and SIP/804-fd83 SIP/201 is the Polycom, while SIP/804 is the spa3k. If I'm losing a frame, is there a way to configure asterisk not to drop the channel? Or is this something the Polycom/Sipura are doing? FYI, asterisk is running on a VIA/MPIA platform. Pure guess is that something happened (unknown what) and the error messages posted above are the result of that, and not the root cause. Finding the root cause may require you to implement the syslog server and debug server options in the spa3k, and compare those log entries to what * records for log messages during a failure. Implementing the log functions on the spa3k does require a reboot. Their log messages are rather cryptic, but looking at keywords and timestamps might identify which box(es) are involved with the dropped calls. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Asterisk 1.2.1 and Sipura SPA3000 strange problem
In any case I'm trying to figure out if maybe someone else has seen this problem. Or if they know what it might be. On 12/28/05, C F [EMAIL PROTECTED] wrote: For somereason I think it's the polycom, which means I need logging for the Polycom and not the spa. On 12/28/05, Rich Adamson [EMAIL PROTECTED] wrote: I have the follwoing setup: Asterisk SVN-tag-1.2.1-r7367 6 Polycom 500 Sip version 1.5.x 4 Sipura SPA3000 (not sure what build) (FXO port) All on flat single network, no NAT, and no gateways to reach each other. Sometimes (happens around 3 times a day, but sometimes far more often), while on the phone to an outside caller (on the PSTN using the FXO on the spa3k), the call dissconects from the polycom and goes thru the incoming extension for the sipura. In other words, astrisk at least as far as I can see from what gets executed in the DP (and maybe spa3k) sees this as if the follwoing has happened: 1. The polycom user hungup, 2. A new call came in on the spa3k. The follwoing is part of the log that I think might help: Dec 28 01:28:24 DEBUG[3368] channel.c: Didn't get a frame from channel: SIP/201-8ba1 Dec 28 01:28:24 DEBUG[3368] channel.c: Bridge stops bridging channels SIP/201-8ba1 and SIP/804-fd83 SIP/201 is the Polycom, while SIP/804 is the spa3k. If I'm losing a frame, is there a way to configure asterisk not to drop the channel? Or is this something the Polycom/Sipura are doing? FYI, asterisk is running on a VIA/MPIA platform. Pure guess is that something happened (unknown what) and the error messages posted above are the result of that, and not the root cause. Finding the root cause may require you to implement the syslog server and debug server options in the spa3k, and compare those log entries to what * records for log messages during a failure. Implementing the log functions on the spa3k does require a reboot. Their log messages are rather cryptic, but looking at keywords and timestamps might identify which box(es) are involved with the dropped calls. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
