I just solved a similar problem on my asterisk box. i just enabled nat=yes and removed the externip from the nat portion in sip.conf. Try it.
On 6/4/07, Compnet Bobby <[EMAIL PROTECTED]> wrote:
We have the latest version of asterisk, on a xeon dell server (2gb ram), with 6 snom320's(latest firmware) and 3 grandstream gxp 2000's (latest stable firmware) and are having a few problems. We have a basic menu that transfers calls to different extensions. The problems can be found on all extensions. We have 2 different incoming providers and the problem happens on both providers. I want your input on 2 problems, they are the following: 1. 60% of the time everything works fine and there are no problems, 40% of times when the calls are transferred to an extension, after a few seconds , the call drops. The log from the server is below(this is from pickup to hangup, the main area of concern is where it says warning). -- Executing [EMAIL PROTECTED]:1] Answer("SIP/9097406868-09e110f8", "") in new stack -- Executing [EMAIL PROTECTED]:2] BackGround("SIP/9097406868-09e110f8", "menus/welcome-to-exec") in new stack -- <SIP/9097406868-09e110f8> Playing 'menus/welcome-to-exec' (language 'en') == CDR updated on SIP/9097406868-09e110f8 -- Executing [EMAIL PROTECTED]:1] Dial("SIP/9097406868-09e110f8", "SIP/103|50|m") in new stack -- Called 103 -- Started music on hold, class 'default', on SIP/9097406868-09e110f8 -- SIP/103-09dedd68 is ringing [May 29 09:05:22] WARNING[2678]: chan_sip.c:1899 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 1 (Critical Response) [May 29 09:05:22] WARNING[2678]: chan_sip.c:1916 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. -- Stopped music on hold on SIP/9097406868-09e110f8 == Spawn extension (from-sip, 103, 1) exited non-zero on 'SIP/9097406868-09e110f8' 2. When a call comes in or is transferred(not on outgoing), there is a delay until the person on the incoming line can hear you. We can hear them, but they can't hear us. Sometimes there is no delay, sometimes for person calling in cant hear you for 6 seconds. Thanks for the help in advance!!! _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- Rizwan Hisham Software Engineer AXVOICE Inc.
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