[Freeswitch-users] Dropped calls on transfer

Mon Feb 20 11:31:02 MSK 2012

SPA504G: 7.4.8a
SPA942: 6.1.5(a)
SPA962: 6.1.5(a)

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004
Sent: 17 February 2012 19:18
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropped calls on transfer

What's the phones firmware anyways?
On 2012-02-17 8:23 AM, "Brian Foster" <bdfoster at endigotech.com<mailto:bdfoster at endigotech.com>> wrote:
I have an SPA921, and i could not get blind transfer/attended transfer/conference to work either.

2012-02-17 11:19:01.258020 [WARNING] sofia.c:6720 Cannot transfer channels that are not in a bridge.

...that's the error message I get. This is probably a bug or the phone isn't actually set up right. I'm going to do some research.

-BDF
On Fri, Feb 17, 2012 at 3:38 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
What commands do you want me to run under the CLI?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Anthony Minessale
Sent: 16 February 2012 17:04

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropped calls on transfer

Its ok if its large.  The point is to get the maximum debug while the issue happens to determine the problem.
the little snippet you showed is not enough to go on.
If you can single out a section of log that shows the whole attempt to perform the transfer with the sip trace and debug enabled I can try to tell you what your problem is (its not a common problem so its specific to something you have , [phone type, config, dialplan])

On Thu, Feb 16, 2012 at 3:00 AM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
Sorry Anthony, I didn’t see it – it was getting late last night.

I’ve read it now.

How large are these logs going to get?  We’re running on ISDN with calls coming in practically every minute.  At the moment the “freeswitch.log” per day is over 20MB (it rotates the log every 10MB).

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Anthony Minessale
Sent: 15 February 2012 22:49

To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropped calls on transfer

Did you not see my last email?
On Wed, Feb 15, 2012 at 4:36 PM, Daniel Knaggs <Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>> wrote:
Thanks for a reply, Brian.

That’s what I told them they should be doing but they won’t listen.  I cringe when I overhear someone saying “call straight through” (ie, “I’ve pressed xfer twice”)  ;)

Looking at the logs when it happens it would appear that a hangup request came from the phone and not the PBX, does that sound right?

Maybe this will help from my debugging last week during a transfer: -

2012-02-08 14:28:26.614963 [DEBUG] switch_ivr_bridge.c:1613 (FreeTDM/1:1/000) State Change CS_EXECUTE -> CS_HIBERNATE
2012-02-08 14:28:26.614963 [DEBUG] switch_core_session.c:1180 Send signal FreeTDM/1:1/000 [BREAK]
2012-02-08 14:28:26.614963 [DEBUG] switch_ivr_bridge.c:1615 (sofia/internal/sip:216 at X.X.X.104:5060) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE
2012-02-08 14:28:26.614963 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:216 at X.X.X.104:5060 [BREAK]
2012-02-08 14:28:26.614963 [DEBUG] switch_core_session.c:791 Send signal sofia/internal/sip:216 at X.X.X.104:5060 [BREAK]
2012-02-08 14:28:26.614963 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:216 at X.X.X.104:5060) Running State Change CS_HIBERNATE
2012-02-08 14:28:26.614963 [DEBUG] switch_core_session.c:791 Send signal FreeTDM/1:1/000 [BREAK]
2012-02-08 14:28:26.614963 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL]
2012-02-08 14:28:26.614963 [DEBUG] switch_core_state_machine.c:432 (sofia/internal/sip:216 at X.X.X.104:5060) State HIBERNATE
2012-02-08 14:28:26.614963 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [ATTENDED_TRANSFER]
2012-02-08 14:28:26.614963 [DEBUG] mod_sofia.c:222 sofia/internal/sip:216 at X.X.X.104:5060 SOFIA HIBERNATE
2012-02-08 14:28:26.614963 [DEBUG] switch_ivr_bridge.c:749 (sofia/internal/sip:216 at X.X.X.104:5060) State Change CS_HIBERNATE -> CS_RESET
2012-02-08 14:28:26.614963 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 601 [ATTENDED_TRANSFER]
2012-02-08 14:28:26.614963 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/sip:216 at X.X.X.104:5060 [BREAK]
2012-02-08 14:28:26.614963 [DEBUG] switch_core_state_machine.c:432 (sofia/internal/sip:216 at X.X.X.104:5060) State HIBERNATE going to sleep
2012-02-08 14:28:26.614963 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:216 at X.X.X.104:5060) Running State Change CS_RESET
2012-02-08 14:28:26.614963 [INFO] mod_dptools.c:2900 Originate Failed.  Cause: ATTENDED_TRANSFER

Here is a different one I’ve seen today: -

2012-02-15 13:05:09.651368 [DEBUG] switch_channel.c:2848 (sofia/internal/212 at X.X.X.2<mailto:sofia/internal/212 at X.X.X.2>) Callstate Change EARLY -> HANGUP
2012-02-15 13:05:09.651368 [NOTICE] switch_channel.c:3276 Hangup sofia/internal/212 at X.X.X.2<mailto:sofia/internal/212 at X.X.X.2> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION]
2012-02-15 13:05:09.651368 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/212 at X.X.X.2<mailto:sofia/internal/212 at X.X.X.2> [KILL]
2012-02-15 13:05:09.651368 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/212 at X.X.X.2<mailto:sofia/internal/212 at X.X.X.2> [BREAK]
2012-02-15 13:05:09.651368 [DEBUG] switch_ivr_originate.c:3222 sofia/internal/212 at X.X.X.2<mailto:sofia/internal/212 at X.X.X.2> Media Establishment Failed.
2012-02-15 13:05:09.651368 [DEBUG] switch_channel.c:2848 (sofia/internal/sip:211 at X.X.X.105:5060) Callstate Change ACTIVE -> HANGUP
2012-02-15 13:05:09.651368 [NOTICE] switch_ivr_originate.c:3224 Hangup sofia/internal/sip:211 at X.X.X.105:5060 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION]

Also had one report of it dropping a call after taking a call off-hold but no more.

Jira?  Sorry what is that?

[Description: cid:image8ca977.PNG at ff333969.478fdf3a]

Daniel Knaggs

Software Developer

Reality Solutions Ltd, 1 Global Business Park, Hamburg Road, Kingston upon Hull, East Yorkshire, HU7 0AE
Tel: 01482 828000 / Fax: 01482 373100
Daniel.Knaggs at realitysolutions.co.uk<mailto:Daniel.Knaggs at realitysolutions.co.uk>
www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/>

________________________________

Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire

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From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Brian West
Sent: 15 February 2012 18:58
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Dropped calls on transfer

If your users want to do that they should be doing a blind transfer (Users are hard to train).   Collect data, use latest git, open jira.

/b

On Feb 15, 2012, at 8:37 AM, Daniel Knaggs wrote:

Hello,

We’ve recently switched from Asterisk to FreeSwitch and experiencing a minor problem.

Currently we’ve got twenty LinkSys/Cisco phones (12x SPA942, 6x SPA962, and 2x SPA504) and experiencing dropped calls when transferring calls.

This seems to happen more often when the caller has hit the xfer button for the second time before the callee has even answered the phone.  But sometimes drops no more than one call a day if the caller waits until the callee has answered.

•          All phones are running the latest firmware (v6) and have been factory reset
•          FreeSwitch is running on version “1.0.head (git-b9b7266 2012-02-10 12-23-58 -0600)”
•          RTP packet size has been changed to 0.020 as per wiki for all phones

Has anyone got any ideas?

--
Brian West
FreeSWITCH Solutions, LLC
Phone: +1 (918) 420-9266<tel:%2B1%20%28918%29%20420-9266>
Fax:   +1 (918) 420-9267<tel:%2B1%20%28918%29%20420-9267>
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http://www.freeswitch.org

_________________________________________________________________________
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Anthony Minessale II

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_________________________________________________________________________
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Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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Official FreeSWITCH Sites
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--
Brian D. Foster
Endigo Computer LLC
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