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@from-sip:1]
Answer("SIP/9097406868-09e110f8",
"") in new stack
-- Executing [9097406868@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@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@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@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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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@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@from-sip:1] Answer("SIP/9097406868-09e110f8", > "") in new stack > > -- Executing [9097406868@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@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@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@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!!! > > > > > > > > > > > > _______________________________________________ > --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 > >-- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070604/527d92bb/attachment.htm
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@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@from-sip:1] Answer("SIP/9097406868-09e110f8", > "") in new stack > > -- Executing [9097406868@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@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@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@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!!! > > > > > > > > > > > > _______________________________________________ > --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 > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070605/12de631d/attachment.htm
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 <compnetbobby@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@from-sip:1] Answer("SIP/9097406868-09e110f8", > "") in new stack > > -- Executing [9097406868@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@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@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@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!!! > > > > > > > > > > > > _______________________________________________ > --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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070605/f02e8a56/attachment-0001.htm