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Anyone know of a company that makes a pass through ATA? By pass through I mean have an Ethernet switch built into the ATA, like most desktop phones have. All of the dual ethernet ATA's I have seen have WAN/LAN ports, not two LAN ports. I fooled around with DMZ etc...but it just doesn't work as well. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, On our Polycom phones we can not activate the Buddy Watch feature. When you add or edit a contact, the list ends at "Auto Divert".....I know it is the end of the list b/c the down arrow on the right side of the screen disappears when I get to Auto Divert. When I add <bw>1</bw> manually to the speed dial file it doesn't change anything. The buttons work well for

Anyone know of a SIP phone that supports group listen? Group listen allow you use the handset but what the far end says comes out the speaker...it is F802 on a Norstar. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080313/a219002e/attachment.htm

We are still using 1.4 and were going to start testing with 1.6.0, but then 1.6.1 was released and now 1.6.2 is already in beta 2. That seems like a lot of independent releases to maintain. I read about all the regressions ans hurried dot releases, makes us nervous. How is everyone doing their testing? -Matt -------------- next part -------------- An HTML attachment was scrubbed... URL:

We have a customer on a wireless connection that has very bad jitter. They can hear people fine, but people have a very hard time hearing them. They are connected via a SPA-2102. It is a SIP client going to a SIP trunk. Something like this in sip.conf [general] would be in effect for all SIP clients: jbenable = yes jbmaxsize = 150 jbresyncthreshold = 1000 jbimpl = fixed jblog = yes I only want

I have an Asterisk box, 1.4.30 with a PRI. A Mitel 3300 is connected to the Asterisk box via SIP trunking. When a user calls from the Mitel through the Asterisk box the user can speak but can not hear the far end. But - when I route the Mitel user to echo() it works, send and receive. The Mitel user also can record and playback greetings. One thing I have noticed is that when the Mitel user

I am trying to figure out which one of our agents is answering the calls. According to http://www.voip-info.org/wiki/view/Asterisk+log+queue_log the only time the queue_log puts the channel (agent) is during logoff & logon. There is the connect & completeagent message, but it doesn't show which channel (agent) answered the phone. I can't even figure it our cross referencing the

We have an issue where Polycom's lose BLF functionality after a reboot. The only way to fix it is to reboot the Polycoms. Anyone else have this issue? We are using 1.4.18. If I run 'sip show subscriptions' all the subscriptions come back after the restart but the lights on the phones do not work. Any help would be appreciated. -Thermal -------------- next part -------------- An

We have Valet Park working well with 1.4.25. We have programmed the Polycom softkeys to include a park button that does a blind transfer to the park extension. Has anyone gotten the a button to activate when a particular park orbit is in use? It would be great if you could press the button to retrieve the parked call. -- -Thermal -------------- next part -------------- An HTML attachment was

I am running Asterisk V 1.4.22 Twice during the last two days the Context Switches on our box has gone from about 7K to 80K in 2.5 hours. The load average would spike to 17, drop to 0.35 then spike again. When connecting to the console 'core show channels' will list the channels but not total calls. 'restart now' had no effect, the only way to stop Asterisk is to kill the

I would like to be able to place a call on hold via the manager interface and be able to retrieve it. The user can click a button in the Order entry form to put the caller on hold when they are looking up information. It saves them from having their hands leave the keyboard and press hold on the phone. I don't see 'hold' & 'retrieve' commands for the manager interface.

I have a virtual server with godaddy but can not compile DAHDI as it complains that I do not have the correct kernel source. The package installed is - kernel-devel-2.6.18-164.11.1.el5.i686: Package kernel-devel-2.6.18-164.11.1.el5.i686 already installed and latest version Nothing to do uname -a returns: Linux ip-XXX-XXX-XXX-XXX.ip.secureserver.net 2.6.18-028stab064.7 #1 SMP Wed Aug 26 13:11:07

Does anyone know if you can have multiple TE110P cards in one chassis? -Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061011/adbceeb5/attachment.htm

Hello, I have been tearing my hair out on this issue for 2 days, any help would be appreciated. We have a normal network and a Cisco SGE2010P switch - a 48 port PoE switch There are two VLANs, 1(data) & 50(VoIP). When Polycoms are connected to the switch with VLAN 50 hard coded in the config they grab a DHCP address from VLAN 1, the PVID for the switch port. The ports have membership in

Hello, I have a bunch of Polycom 601's and Asterisk 1.4.13. The problem is that the MWI indicators will never go off (The blinking red light and envelope in the LCD). I have tried to upgrade to 1.4.14 and all different SIP versions on the Polycoms. I am now at 1.6.7 Here is the SIP Message that turns on the lights: Scheduling destruction of SIP dialog '

I would like to hire someone to help us tweak our asterisk system for Snom phones. We would like to enable things like: One touch recording One touch park orbits Presence Please contact off-list if you will be able to help. Thermal -------------- next part -------------- An HTML attachment was scrubbed... URL: