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Hi all, Some of my asterisk users have used their maximum call limit for incoming calls (peers). There incoming call limit should automatically reset to zero after hangup but its not happening and they no longer can recieve any calls as their allowed limit is already full. So is there any way to reset the call limit on peers by commands or do i have to restart my asterisk server? -- Best Regards

I have a Toshiba DK424 connected via T1 E&M to a TE110P card. Intermittently when a user dials a number I am getting 'getdtmf' errors on the Ast server and the calls do not go through. If they dial the number once or twice more, it works fine and I receive no DTMF problems. On another note, end users are complaining about intermittent disconnects. T1 is ESF/B8ZS - 24 chan. Other

Hello All, I apologize for the slightly off-topic question, but I'm sure that the people best acquainted with the issue would be hanging around here. We recently deployed several Linksys POE switches for some smaller customers (10-24 station) and appear to be suffering from intermittent lock-ups of the SNOM phones attached. Obviously we are running Asterisk for the gateway, but I was

I am attempting to deploy a TE210P with one port configured (all 24 channels) as Cisco HDLC, currently running 2.6.11 FC4 with HDLC module compiled. HDLC (RAW) appears to work fine, however, using Cisco HDLC (via sethdlc & sethdlc-new) results in no traffic being encapsulated or transmitted via the hdlc0 interface. Is there anybody here currently running a similar configuration on a

I have an odd problem between a Nortel BCM and Asterisk (1.2.4) Zaptel ( 1.2.3). Calls all appear to be working fine, however, when someone originates a call from the PBX and the Switch returns a 486 (Busy) the PRI channel hangs in a busy state indefinitely. Subsequent to this, the PBX side shows the channels as IDLE and apparently attempts to route calls to the siezed/BUSY channels. Normal

I have been trying to track this down for a while to no avail. I have a variety of different SNOM phones (the entire 3XX series) and have also tried on a variety of different Asterisk versions (pretty much the whole 1.2 and 1.4 train) When I "Page()" phones in Asterisk, I only intermittently get an audio indication on the recipient phone. Audio Indicator is turned on, and I'm

I have been trying to track this down for a while to no avail. I have a variety of different SNOM phones (the entire 3XX series) and have also tried on a variety of different Asterisk versions (pretty much the whole 1.2 and 1.4 train) When I "Page()" phones in Asterisk, I only intermittently get an audio indication on the recipient phone. Audio Indicator is turned on, and I'm

I have a client who is using a Shoretel PBX. This PBX apparently does not send DTMF information OOB, but instead sends this inband via the B-channel. This is traversing an Asterisk box via a PRI. The user calls the IVR (1-800-CALL-DHL), receives audio, but is not able to present DTMF to engage the IVR. With some light research it appears that the DSP is not activating until the call is