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@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@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@X.X.X.104:5060 [BREAK] 2012-02-08 14:28:26.614963 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:216@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@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@X.X.X.104:5060 SOFIA HIBERNATE 2012-02-08 14:28:26.614963 [DEBUG] switch_ivr_bridge.c:749 (sofia/internal/sip:216@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@X.X.X.104:5060 [BREAK] 2012-02-08 14:28:26.614963 [DEBUG] switch_core_state_machine.c:432 (sofia/internal/sip:216@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@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@X.X.X.2) Callstate Change EARLY -> HANGUP 2012-02-15 13:05:09.651368 [NOTICE] switch_channel.c:3276 Hangup sofia/internal/212@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@X.X.X.2 [KILL] 2012-02-15 13:05:09.651368 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/212@X.X.X.2 [BREAK] 2012-02-15 13:05:09.651368 [DEBUG] switch_ivr_originate.c:3222 sofia/internal/212@X.X.X.2 Media Establishment Failed. 2012-02-15 13:05:09.651368 [DEBUG] switch_channel.c:2848 (sofia/internal/sip:211@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@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? [cid:image8ca977.PNG@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@realitysolutions.co.uk<mailto:Daniel.Knaggs@realitysolutions.co.uk> www.realitysolutions.co.uk<http://www.realitysolutions.co.uk/> ________________________________ Sage Accredited Business Partner serving businesses in Yorkshire & Lincolnshire [cid:image6a28e2.PNG@68fbc2d4.4c8c5174] From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@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 Fax: +1 (918) 420-9267 brian@freeswitch.org<mailto:brian@freeswitch.org> http://www.freeswitch.org
