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...values, but nothing appears on screen. Also changed the pci slot where the board is. The zaptel drivers are load properly, I also have a TE110P on the same server as pri_cpe and has no issues. Has anyone of you suffered the same? how did you solve it? Thank you. -- Jose Flores Galicia <<FloJoSe at gmail.com>> BriefCode && Code Based Training -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080714/fa6a313d/attachment.htm

Hi all: when i make a call from linksys pap2t to an asterisk server a fake ring is heard some times ,but when sending calls between 2 asterisk servers through sip no fake ring is heard but real one. any suggestions please. _________________________________________________________________ Windows Live?: E-mail. Chat. Share. Get more ways to connect.

Hi, Does anyone know how to change the language for the user interface in a Polycom SoundPoint IP 300? Thanks -- Jose Flores Galicia <<FloJoSe at gmail.com>> BriefCode && Code Based Training -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080521/9ace1028/attachment.htm

I have 8 aastra phones that are loosing registration. On the phone gui it says 408 as the registation error after a minute or say they register. In the cli it eill say the phone is now unreachable then it will show it registering then available. At first they did it every hour all the phones. After messing with the experation it does it every 15 nin. Any ideas on how to troubleshoot this? I tried

I have a quick question. I am using Asterisk 1.4. I have a user that has changed phones (grandstream budge tone 200 to a polycom 330). I have changed the sip.conf and extensions.conf. I have also unplugged the old phone and plugged in the new phone. I get the ext showing on the phone, but when I do a sip show peer 5000 the old ip address and phone show up. I did a sip reload and a dialplan reload.

We are running a PBX box for our small company's phone system. There are two phones on our network that are not working though. When I reboot a phone that works correctly, like this (note 4907 is the phones extension): asterisk -rx "sip notify polycom-check-cfg 4907" It returns this: Sending NOTIFY of type 'polycom-check-cfg' to '4907' and the phone proceeds to

All, I am looking at a little support on this, as I haven't found it on google yet. I have had this work on Callweaver, but am moving to Asterisk for a variety of reasons. My dial plans, and everything else transferred perfectly, though I am not sure they are 'correct' for Asterisk 1.6.1, with simple things like SIP users outlined in the sip.conf file, not in the users file,

I'm trying to make an AMI call. I want to call a number, play an announcement when the call is answered, then call a second number and connect the two when the second call is answered. I an able to make a simple call to two numbers and connect them using the manager API but playing the announcement has me beat. Suggestions anyone? Bruce Ferrell

2015-02-09 14:36 GMT-06:00 jg <webaccounts173 at jgoettgens.de>: > Hi! > > Sometimes IAX peers are not reachable and with "iax2 set debug on" I get > something like this > > Tx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: PONG > Timestamp: 00014ms SCall: 00001 DCall: 01200 79.233.155.174:49153 > Rx-Frame Retry[ No] -- OSeqno: 001