Re: [asterisk-users] How to get Asterisk acting like a Multi-thread application?

2014-01-06 Thread s...@yahoo.com
Hello, I am really appreciate for your help.Thanks alot.   Regard Sami From: Tzafrir Cohen To: asterisk-users@lists.digium.com Sent: Sunday, 5 January 2014, 17:13:43 Subject: Re: [asterisk-users] How to get Asterisk acting like a Multi-thread application?

[asterisk-users] Dropped call on new CISCO router for no reason!

2014-01-06 Thread Nick Cameo
Hello Everyone, Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT setup. After 2 seconds on a successfully established call we reach retrans max, and asterisk disconnects the call. We

Re: [asterisk-users] Dropped call on new CISCO router for no reason!

2014-01-06 Thread Eric Wieling
This is a classic symptom of having reinvites and/or direct media enabled on Asterisk or SIP ALG enabled on the router. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Cameo Sent: Monday, January 06, 2014

Re: [asterisk-users] Dropped call on new CISCO router for no reason!

2014-01-06 Thread Nick Cameo
Hello Eric, I knew this problem all so well however, never knew CISCO sip alg was enabled by default. The following settings got us up and going shortly after the email: no ip nat service sip udp port 5060 ip nat inside source static udp 192.168.2.5 5060 interface Dialer0 5060 access-list 130 pe

Re: [asterisk-users] Dropped call on new CISCO router for no reason!

2014-01-06 Thread Paul Belanger
On 14-01-06 09:27 AM, Nick Cameo wrote: Hello Everyone, Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT setup. After 2 seconds on a successfully established call we reach retrans

[asterisk-users] Cisco 7940 SIP 8.12 no audio when using Outbound Proxy

2014-01-06 Thread Jr Richardson
Hi All, Simple scenario: 7940 SIP>http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/aster