Attended Transfer and Music On Hold

Hello…

We use Asterisk 1.4.21.2 with Digium B410P Wildcard (Quad BRI ISDN card)… Our telephones are Grandstream GXP280 and 2000…

We experience problems in our office with Attended transfers…

Client (A) is calling in our office and our represetattive (B) answers the call and starts talking… Then (B) wants to make an attended transfer to person ©… So (B) press Flash button and (A) is placed on hold hearing Music On Hold… (B) speaks with © to give info for the customer (A) and then press TRANSFER button to connect (A) and ©… But the connection is problematic, because (A) can hear © person’s voice, but © can hear only the Music On Hold from (A) and not his voice…

I checked the recordings and i saw that all the recordings in those situations are OK, that means both parties are recorded…

Does anyone have any idea?

Thanks in advance…

Hello…

I tracked down a bug fix for that problem (issues.asterisk.org/view.php?id=4418 - Digium Issue 4418) which seems to be solved back in 2005… But in my case is still happening… Below is a prtion of the Log file where cleraly we can see that after the Attended transfer the target phone is placed on Hold and Music on Hold starts playing…

[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: **** Received REFER (9) - Command in SIP REFER
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: Attended transfer: Will use Replace-Call-ID : 0055c66ccb3e4f2a@192.168.0.213 (No check of from/to tags)
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: Looking for callid 0055c66ccb3e4f2a@192.168.0.213 (fromtag bea7849c3a4e1657 totag as69afd50c)
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: Matched INCOMING call - their tag is bea7849c3a4e1657 Our tag is as69afd50c
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: Sip transfer:--------------------
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: – Transferer to PBX channel: SIP/304-0a109ff0 State Up
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: – Transferer to PBX second channel (target): SIP/304-b6115770 State Up
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: – Bridged call to transferee: Local/304@from-internal-db2c,2 State Up
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: – Bridged call to transfer target: SIP/203-0a1b2058 State Up
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: – END Sip transfer:--------------------
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: SIP transfer: Four channels to handle
[2009-05-27 10:08:56] VERBOSE[3958] logger.c: – Stopped music on hold on Local/304@from-internal-db2c,2
[2009-05-27 10:08:56] DEBUG[3958] channel.c: Set channel Local/304@from-internal-db2c,2 to write format alaw
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: SIP transfer: trying to masquerade Local/304@from-internal-db2c,2 into SIP/304-b6115770
[2009-05-27 10:08:56] DEBUG[3958] channel.c: Planning to masquerade channel Local/304@from-internal-db2c,2 into the structure of SIP/304-b6115770
[2009-05-27 10:08:56] DEBUG[3958] channel.c: Done planning to masquerade channel Local/304@from-internal-db2c,2 into the structure of SIP/304-b6115770
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: SIP transfer: Succeeded to masquerade channels.
[2009-05-27 10:08:56] DEBUG[3958] chan_sip.c: SIP attended transfer: Unlocking channel SIP/304-b6115770

[2009-05-27 10:08:56] DEBUG[8523] res_musiconhold.c: SIP/203-0a1b2058 Opened file 0 ‘/var/lib/asterisk/mohmp3//wav_1846_suppe_-poet_and_peasant-_overture’

[2009-05-27 10:08:56] DEBUG[8523] channel.c: Actually Masquerading Local/304@from-internal-db2c,2(6) into the structure of SIP/304-b6115770(6)
[2009-05-27 10:08:56] DEBUG[8523] channel.c: Got clone lock for masquerade on ‘Local/304@from-internal-db2c,2’ at 0xa13c398
[2009-05-27 10:08:56] DEBUG[8523] chan_sip.c: SIP Fixup: New owner for dialogue 0055c66ccb3e4f2a@192.168.0.213: Local/304@from-internal-db2c,2 (Old parent: Local/304@from-internal-db2c,2)
[2009-05-27 10:08:56] DEBUG[8523] chan_sip.c: Hangup call Local/304@from-internal-db2c,2, SIP callid 0055c66ccb3e4f2a@192.168.0.213)
[2009-05-27 10:08:56] DEBUG[8523] chan_sip.c: Updating call counter for incoming call
[2009-05-27 10:08:56] DEBUG[8523] devicestate.c: Notification of state change to be queued on device/channel SIP/304
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: No provider found, checking channel drivers for SIP - 304
[2009-05-27 10:08:56] DEBUG[3921] chan_sip.c: Checking device state for peer 304
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: Changing state for SIP/304 - state 8 (On Hold)
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: No provider found, checking channel drivers for SIP - 304
[2009-05-27 10:08:56] DEBUG[3921] chan_sip.c: Checking device state for peer 304
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: Checking if I can find provider for “Custom” - number: DND304
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: Checking provider SLA with Custom
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: Checking provider Meetme with Custom
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: Checking provider Custom with Custom
[2009-05-27 10:08:56] DEBUG[3921] db.c: Unable to find key ‘DND304’ in family ‘CustomDevstate’
[2009-05-27 10:08:56] DEBUG[3939] app_queue.c: Device ‘SIP/304’ changed to state ‘8’ (On Hold) but we don’t care because they’re not a member of any queue.
[2009-05-27 10:08:56] DEBUG[8523] chan_sip.c: Call from peer ‘304’ removed from call limit 50
[2009-05-27 10:08:56] DEBUG[8523] devicestate.c: Notification of state change to be queued on device/channel SIP/304
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: No provider found, checking channel drivers for SIP - 304
[2009-05-27 10:08:56] DEBUG[3921] chan_sip.c: Checking device state for peer 304
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: Changing state for SIP/304 - state 8 (On Hold)
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: No provider found, checking channel drivers for SIP - 304
[2009-05-27 10:08:56] DEBUG[3921] chan_sip.c: Checking device state for peer 304
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: Checking if I can find provider for “Custom” - number: DND304
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: Checking provider SLA with Custom
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: Checking provider Meetme with Custom
[2009-05-27 10:08:56] DEBUG[3921] devicestate.c: Checking provider Custom with Custom
[2009-05-27 10:08:56] DEBUG[3921] db.c: Unable to find key ‘DND304’ in family ‘CustomDevstate’
[2009-05-27 10:08:56] DEBUG[3939] app_queue.c: Device ‘SIP/304’ changed to state ‘8’ (On Hold) but we don’t care because they’re not a member of any queue.
[2009-05-27 10:08:56] DEBUG[8523] channel.c: Putting channel Local/304@from-internal-db2c,2 in 8/8 formats
[2009-05-27 10:08:56] DEBUG[8523] channel.c: Released clone lock on ‘SIP/304-b6115770’
[2009-05-27 10:08:56] DEBUG[8523] channel.c: Done Masquerading Local/304@from-internal-db2c,2 (6)
[2009-05-27 10:08:56] DEBUG[8525] audiohook.c: Read factory 0xa135908 was pretty quick last time, waiting for them.
[2009-05-27 10:08:56] DEBUG[8447] channel.c: Didn’t get a frame from channel: SIP/304-b6115770
[2009-05-27 10:08:56] DEBUG[8447] channel.c: Bridge stops bridging channels SIP/304-b6115770 and SIP/304-0a109ff0
[2009-05-27 10:08:56] DEBUG[8447] channel.c: Hanging up channel ‘SIP/304-0a109ff0’
[2009-05-27 10:08:56] DEBUG[8447] chan_sip.c: Updating call counter for outgoing call
[2009-05-27 10:08:56] DEBUG[8447] devicestate.c: Notification of state change to be queued on device/channel SIP/304
[2009-05-27 10:08:56] DEBUG[8447] chan_sip.c: Call to peer ‘304’ removed from call limit 50
[2009-05-27 10:08:56] DEBUG[8447] devicestate.c: Notification of state change to be queued on device/channel SIP/304
[2009-05-27 10:08:56] DEBUG[8447] chan_sip.c: SIP Transfer: Not hanging up right now… Rescheduling hangup for 6cdc8cf97233da7f5db9b8e639cc9eba@192.168.0.200.
[2009-05-27 10:08:56] DEBUG[8447] devicestate.c: Notification of state change to be queued on device/channel SIP/304-0a109ff0
[2009-05-27 10:08:56] DEBUG[8447] devicestate.c: Notification of state change to be queued on device/channel SIP/304
[2009-05-27 10:08:56] DEBUG[8447] rtp.c: Channel ‘’ has no RTP, not doing anything
[2009-05-27 10:08:56] DEBUG[8447] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[2009-05-27 10:08:56] DEBUG[8447] app_macro.c: Spawn extension (macro-dial,s,7) exited non-zero on ‘SIP/304-b6115770’ in macro ‘dial’
[2009-05-27 10:08:56] DEBUG[8447] app_macro.c: Spawn extension (macro-dial,s,7) exited non-zero on ‘SIP/304-b6115770’ in macro ‘exten-vm’
[2009-05-27 10:08:56] DEBUG[8447] pbx.c: Spawn extension (macro-dial,s,7) exited non-zero on ‘SIP/304-b6115770’
[2009-05-27 10:08:56] VERBOSE[8447] logger.c: == Spawn extension (macro-dial, s, 7) exited non-zero on ‘SIP/304-b6115770’
[2009-05-27 10:08:56] DEBUG[8447] channel.c: Soft-Hanging up channel ‘SIP/304-b6115770’

I can’t see a reason why it happens or the causes that make it happen…

We use Grandstream GXP280 and 2000… And Asterisk 1.4.21.2 through PiaF distro…

Thank you for your help in advance…