Sponsor: VoiceMeUp - Corporate & Wholesale VoIP Services

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

[asterisk-users] Dropped calls


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





PostPosted: Thu Jan 31, 2008 9:06 am    Post subject: [asterisk-users] Dropped calls Reply with quote

On Jan 31, 2008 6:45 AM, mccoy silva <mccoy.silva at gmail.com> wrote:
Quote:
I have a very serious problem with calls between PAP2-NA and a TDM2400 (8
FXO). Almost every call dropped after between 20 and 30 seconds with
conversation.
I disable the sound card, serial and other things on my server, but the
problem still continues. I've changed the RPT Packet Size to .20 on PAP2-NA,
but nothing.
Here a piece of my log:

[Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up channel 'Zap/17-1'
[Jan 31 07:10:43] DEBUG[3131] chan_zap.c: zt_hangup(Zap/17-1)
[Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Hangup: channel: 17 index = 0,
normal = 11, callwait = -1, thirdcall = -1
[Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/17-1
[Jan 31 07:10:43] DEBUG[3131] chan_zap.c: Updated conferencing on 17, with 0
conference users
[Jan 31 07:10:43] VERBOSE[3131] logger.c: -- Hungup 'Zap/17-1'
[Jan 31 07:10:43] DEBUG[3131] devicestate.c: Notification of state change
to be queued on device/channel Zap/17-1
[Jan 31 07:10:43] DEBUG[3131] app_dial.c: Exiting with DIALSTATUS=NOANSWER.
[Jan 31 07:10:43] DEBUG[2695] devicestate.c: No provider found, checking
channel drivers for Zap - 17
[Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for
channel '0x82042e8'
[Jan 31 07:10:43] VERBOSE[3131] logger.c: == Auto fallthrough, channel
'SIP/dep2_1154-08202968' status is 'NOANSWER'
[Jan 31 07:10:43] DEBUG[3131] channel.c: Soft-Hanging up channel
'SIP/dep2_1154-08202968'
[Jan 31 07:10:43] DEBUG[3131] channel.c: Hanging up channel
'SIP/dep2_1154-08202968'
[Jan 31 07:10:43] DEBUG[3131] chan_sip.c: Hangup call
SIP/dep2_1154-08202968, SIP callid f7bcd67d-dc20e8c1 at 192.168.4.205)
[Jan 31 07:10:43] DEBUG[3131] chan_sip.c: Hanging up channel in state Ring
(not UP)
[Jan 31 07:10:43] DEBUG[3131] devicestate.c: Notification of state change to
be queued on device/channel SIP/dep2_1154-08202968
[Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for
channel '0x82042e8'
[Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for
channel '0x82042e8'
[Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for
channel '0x82042e8'
[Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for
channel '0x82042e8'
[Jan 31 07:10:43] DEBUG[2695] channel.c: Avoiding initial deadlock for
channel '0x82042e8'
[Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID:
D1B9-141D-46684820168D9512F870-009 at SipHost Their Tag c136d668-768786 Our
tag: as0bc591fc
[Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID:
D1B9-141D-46684820F9EEEBF1F8F2-008 at SipHost Their Tag 2b4f6f33-768786 Our
tag: as496fd97d
[Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID:
D1B9-141D-4668482079ECFA697DF3-007 at SipHost Their Tag 73176828-768785 Our
tag: as1ab79f58
[Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID:
D1B9-141D-46684820D113C766B56C-006 at SipHost Their Tag eae1f94d-768783 Our
tag: as1b0024a8
[Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID:
D1B9-141D-46684820214365DAC91E-005 at SipHost Their Tag f0629993-768783 Our
tag: as3f520446
[Jan 31 07:10:43] DEBUG[2714] chan_sip.c: = No match Their Call ID:
D1B9-141D-466848205E42C7CB16A8-004 at SipHost Their Tag 728b9929-768782 Our
tag: as222bab2d

Regards,

McCoy

You need to Answer() the call in your dialplan, that is my guess
without seeing your dialplan.

Try adding EXTEN,1,Answer() before the rest of the stuff in your
dialplan in the context that handles your inbound calls.

Thanks,
Steve Totaro
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