Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

VoIP Mailing List Archives
Mailing list archives for the VoIP community
 SearchSearch 

[asterisk-users] Calls drop + "Didn't get a frame from channel" log message


 
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users
View previous topic :: View next topic  
Author Message
gincantalupo at fgasof...
Guest





PostPosted: Tue Jun 24, 2008 10:37 am    Post subject: [asterisk-users] Calls drop + "Didn't get a frame from Reply with quote

Hi,

sometimes Asterisk drops calls and shows "Didn't get a frame from
channel" in its log file. Unfortunately Google gives no answers even if
a lot of people ask for help.
A fast look into the code shows Asterisk entering a loop where voice is
been transferred and every loop Asterisk waits for a frame, exiting the
loop if no frame has arrived. It seems to be a problem not depending on
the kind of channel...happens with ISDN and PRI lines.
What is stopping the frames, making Asterisk exiting that loop and
dropping the calls?

Thank you.

Giorgio.
Back to top
jlcurty at gmail.com
Guest





PostPosted: Tue Jun 24, 2008 5:05 pm    Post subject: [asterisk-users] Calls drop + "Didn't get a frame from Reply with quote

I have googled a lot to find solution to the same exact problem described in
your message but no real solution yet.

here is my config

1 physical network
25 pc windows
25 phones IP330 & IP550 SIP 2.1.2 no vlan CDP disabled some with dhcp some
with fixed ip to see if there is a diff

3 switchs connected to each others
1 cisco switch 35xx for pcs
2 linksys 24P P OE for phones

1 patton PRI gateway to isdn

1 asterisk server 1.12.18 talking sip to Patton , for each phone, type
friend can re-invite no, nat no

symptoms:

call drop randomly , can be after 10 s or 2000 seconds ! same log "didn't
get frame etc
fews drops per phones per day but very irritating for the customer Sad
tried to power phones with adapters to avoid power pbs from the switch ,
same result

if someone met this problem before get an idea to fix it , I wd appreciate
!!!!!

thanks
jl

On Tue, Jun 24, 2008 at 5:37 PM, gincantalupo <gincantalupo at fgasoftware.com>
wrote:

Quote:
Hi,

sometimes Asterisk drops calls and shows "Didn't get a frame from
channel" in its log file. Unfortunately Google gives no answers even if
a lot of people ask for help.
A fast look into the code shows Asterisk entering a loop where voice is
been transferred and every loop Asterisk waits for a frame, exiting the
loop if no frame has arrived. It seems to be a problem not depending on
the kind of channel...happens with ISDN and PRI lines.
What is stopping the frames, making Asterisk exiting that loop and
dropping the calls?

Thank you.

Giorgio.


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

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/20080625/7b8dcacb/attachment.htm
Back to top
gincantalupo at fgasof...
Guest





PostPosted: Wed Jun 25, 2008 2:56 am    Post subject: [asterisk-users] Calls drop + "Didn't get a frame from Reply with quote

Hi Jean,

I have an Asterisk 1.12.18 with about 30 pc each with a Doro SIP phone
on an unknown LAN.
I think google is useless in cases like this.
Many of the system we are working on are in production and we cannot
make tests with them so the only hope is to gather infos from people
experiencing the same problems and trying to understand something from
the code, maybe asking the developers for info...that's why I'd like to
understand what kind of frames Asterisk is waiting for in order to find
what is the cause of this problem (maybe this is not the right
mailing-list ?)
The strangest thing is problems like this are still happening and remain
unsolved.....but a PBX dropping calls (or not dropping at all, as
sometime happens) is like a car without seats.....yes, you can drive,
but it is not a comfortable experience!

Giorgio
Jean-Louis curty wrote:
Quote:
I have googled a lot to find solution to the same exact problem
described in your message but no real solution yet.

here is my config

1 physical network
25 pc windows
25 phones IP330 & IP550 SIP 2.1.2 no vlan CDP disabled some with dhcp
some with fixed ip to see if there is a diff

3 switchs connected to each others
1 cisco switch 35xx for pcs
2 linksys 24P P OE for phones

1 patton PRI gateway to isdn

1 asterisk server 1.12.18 talking sip to Patton , for each phone, type
friend can re-invite no, nat no

symptoms:

call drop randomly , can be after 10 s or 2000 seconds ! same log
"didn't get frame etc
fews drops per phones per day but very irritating for the customer Sad


tried to power phones with adapters to avoid power pbs from the switch
, same result

if someone met this problem before get an idea to fix it , I wd
appreciate !!!!!

thanks
jl

On Tue, Jun 24, 2008 at 5:37 PM, gincantalupo
<gincantalupo at fgasoftware.com <mailto:gincantalupo at fgasoftware.com>>
wrote:

Hi,

sometimes Asterisk drops calls and shows "Didn't get a frame from
channel" in its log file. Unfortunately Google gives no answers
even if
a lot of people ask for help.
A fast look into the code shows Asterisk entering a loop where
voice is
been transferred and every loop Asterisk waits for a frame,
exiting the
loop if no frame has arrived. It seems to be a problem not
depending on
the kind of channel...happens with ISDN and PRI lines.
What is stopping the frames, making Asterisk exiting that loop and
dropping the calls?

Thank you.

Giorgio.


_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


------------------------------------------------------------------------

_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users


--

_________________________________________________
Giorgio Incantalupo, mailto:gincantalupo at fgasoftware.com
FG&A srl - http://www.fgasoftware.com -
Voice at Work - The Agile PBX http://www.voiceatwork.eu
Tel: 02997663.14, Fax: 0291390172
Back to top
Display posts from previous:   
Post new topic   Reply to topic    VoIP Mailing List Archives Forum Index -> Asterisk Users All times are GMT - 5 Hours
Page 1 of 1

 
Jump to:  
You cannot post new topics in this forum
You cannot reply to topics in this forum
You cannot edit your posts in this forum
You cannot delete your posts in this forum
You cannot vote in polls in this forum


Powered by phpBB © 2001, 2005 phpBB Group

VoiceMeUp - Corporate & Wholesale VoIP Services