[asterisk-users] Calls being dropped

[Compnet Bobby]

Where do I get oej's patch, and how do I install it?

 

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Andreas
Brodmann
Sent: Tuesday, June 05, 2007 2:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calls being dropped

 

We have a similar problem at our place, since a few months.

oej, mentioned a patch he has made after the release of asterisk-1.4.4. So
we're
all desperately waiting for asterisk-1.4.5 to be released; unless you want
to install 
from svn.




2007/6/4, Compnet Bobby <compnetbobby at hotmail.com>:

 

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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