[asterisk-users] configure my voip provider

2012-03-04 Thread Baha @ SH
Hello all, I am trying to configure my voip provider on asterisk box, I cannot get the right configuration after trying many possibilities, always getting circuits busy ,message, please let me know the meaning of the debug message. And by the way, when I use a normal sip phone, I can dial normall

Re: [asterisk-users] No compatible codecs, not accepting this offer

2012-03-04 Thread SamyGo
HI, Instead of taking traces on asterisk console, try capturing traffic in pcap using tcpdump command and later analyze it with wireshark. #tcpdump -i eth*N *-s 0 -w sip-capture.pcap -v Download this file to any machine with wireshark installed and apply "sip" as filter there. Regards. Sammy On F

Re: [asterisk-users] Force sip peers to re register

2012-03-04 Thread Guy Gold
I second Eric's opinion, Polycom are like the Borg, (I hope I spelled it right). The fact that you have 100's of units deployed should only help you out,when it comes to Polycom. On Sun,Mar 04 03:24:PM, Eric Wieling wrote: > You should be able to configure the Polycom phones to failover/failback

Re: [asterisk-users] sip proxy

2012-03-04 Thread Guy Gold
On Sat,Mar 03 09:17:PM, K bharathan wrote: >thru DSL connection the callers can hear only > one way;asterisk pbx is behind NAT; Greetings, If the calls are esteblished,then SIP did its work. The system may RTP problems, NAT may or may not be the casue for the issue. Defaults RTP ports for aster

Re: [asterisk-users] Line noise/hiss on Openvox A400P card on FXO

2012-03-04 Thread Sebastian Arcus
Hi, I've tried both an AMD and an Intel motherboard - with identical results. Sebastian On 04/03/12 15:32, Carlos Rojas wrote: Hello Are you using a amd server? Sometimes openvox doesn't work fine with amd processor Regards On Mar 1, 2012 2:07 PM, "Dave Platt" mailto:dpl...@radagast.org>>

Re: [asterisk-users] Force sip peers to re register

2012-03-04 Thread Eric Wieling
You should be able to configure the Polycom phones to failover/failback more quickly. Check the Admin Guide. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Sunday, March 04, 2012 2:52 PM

Re: [asterisk-users] Force sip peers to re register

2012-03-04 Thread Alex Balashov
No, only endpoints decide when to retry registration attempts. If the registration info is your only means of knowing how to reach those peers from Asterisk, and that information is still valid at a given time, it wouldn't make much sense to force them to reregister, would it? :-) And if the in

[asterisk-users] Force sip peers to re register

2012-03-04 Thread research
I have hundreds of sip endpoints (mostly polycom) which i would like to immediate request them to reregister when we failover/fallback to the standby server. However it takes so long and i would like to know if there is a command to force all sip peers to attempt registration. I have tried both '

Re: [asterisk-users] Line noise/hiss on Openvox A400P card on FXO

2012-03-04 Thread Carlos Rojas
Hello Are you using a amd server? Sometimes openvox doesn't work fine with amd processor Regards On Mar 1, 2012 2:07 PM, "Dave Platt" wrote: > > 5. Placing ferrite cores on the phone cables. > > Do either of the phone lines in question have DSL on them? > > If so, a ferrite core (which will bl