[asterisk-users] Calls being dropped

[Mike Lynchfield]

that becasue the reinvite is using a private ip probably..

sip debug

pastebin the results..

look in the re-invite part..



On 6/4/07, Compnet Bobby <compnetbobby at hotmail.com> 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 [9097406868 at from-sip:1] Answer("SIP/9097406868-09e110f8",
> "") in new stack
>
>     -- Executing [9097406868 at from-sip: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 [103 at from-sip: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
> LAXMGC0120070529230718052251 at 209.244.63.13 for seqno 1 (Critical Response)
>
> [May 29 09:05:22] WARNING[2678]: chan_sip.c:1916 retrans_pkt: Hanging up
> call LAXMGC0120070529230718052251 at 209.244.63.13 - 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!!!
>
>
>
>
>
>
>
>
>
>
>
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>


-- 
Mike
Sales Manager
http://www.voicemeup.com
Making it happen
1.877.807.VOIP (8647)
1.514.312.7030
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