Cisco 7911 call transfer

hey all community
I recently instruct and asterisk 13 as my PBX for an organization where have about 25 client
some on soft-phone (zoiper) and others on cisco 7911
I changed the firmware of cisco sccp to sip , and provide them with xml file to have their configuration
all of them have registered and work properly but the thing is that when I transfer one call from one of cisco phones to another cisco phone I get hold music from transferee phone but no ringing on destination phone . ODD!
I can provide any log or tcpdump file if needed.

Initially, please provide the chan_sip or chan_pjsip, SIP protocol captures for the INVITE, NOTIFY and REFER transactions. I assume you are doing SIP transfers, rather than features.conf ones.

OPTIONS and REGISTER transactions are not relevant and make the logs difficult to read.

I have both logs in asterisk verbosity level 4 and tcpdump file.
which one I need to give.

and plz guide me on how to send it to you its pcap file
as I am new to this forum

The SIP trace files created by Asterisk are plain text in /var/log/asterisk.

Uncomment the full log file in logger.conf.

I have only queue log file
and 3 folder named:
cdr-csv cdr-custom cel-custom

Your logger.conf isn’t as distributed. Change it to enable the full log.

(People also screen scrape the SIP trace from the console, but that loses many of the time stamps and is limited by the amount of back history that is displayed.)

The standard distribution also enables a /var/log/messages file.

yes I have these now:
full messages queue_log
which one is needed?

full

I think the SIP trace requires a non-zero verbosity, so you may also have to set that. It should be obvious if you have successfully obtained a SIP trace, but if you want an example, there are thousands on the forum.

  == Using SIP RTP CoS mark 5
       > 0x7fc900023760 -- Strict RTP learning after remote address set to: 192.168.3.4:10106
    -- Executing [77516652@incomming:1] Answer("SIP/GW-0000001a", "") in new stack
       > 0x7fc900023760 -- Strict RTP switching to RTP target address 192.168.3.4:10106 as source
    -- Executing [77516652@incomming:2] GotoIf("SIP/GW-0000001a", "0?blocked,1") in new stack
    -- Executing [77516652@incomming:3] GotoIfTime("SIP/GW-0000001a", "06:30-23:59,sat,*,*?daygreeting") in new stack
    -- Executing [77516652@incomming:4] GotoIfTime("SIP/GW-0000001a", "06:30-20:30,sun,*,*?daygreeting") in new stack
    -- Goto (incomming,77516652,11)
    -- Executing [77516652@incomming:11] BackGround("SIP/GW-0000001a", "/var/lib/asterisk/sounds/pr-medical/my/daygreeting") in new stack
    -- <SIP/GW-0000001a> Playing '/var/lib/asterisk/sounds/pr-medical/my/daygreeting.slin' (language 'en')
    -- Remote UNIX connection disconnected
  == Spawn extension (incomming, 77516652, 11) exited non-zero on 'SIP/GW-0000001a'
[Oct 28 16:17:45] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77516652@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
       > Saved useragent "Cisco-CP7911G/8.4.0" for peer 6014
  == Using SIP RTP CoS mark 5
       > 0x7fc900023760 -- Strict RTP learning after remote address set to: 192.168.3.4:10150
    -- Executing [77516652@incomming:1] Answer("SIP/GW-0000001b", "") in new stack
       > 0x7fc900023760 -- Strict RTP switching to RTP target address 192.168.3.4:10150 as source
    -- Executing [77516652@incomming:2] GotoIf("SIP/GW-0000001b", "0?blocked,1") in new stack
    -- Executing [77516652@incomming:3] GotoIfTime("SIP/GW-0000001b", "06:30-23:59,sat,*,*?daygreeting") in new stack
    -- Executing [77516652@incomming:4] GotoIfTime("SIP/GW-0000001b", "06:30-20:30,sun,*,*?daygreeting") in new stack
    -- Goto (incomming,77516652,11)
    -- Executing [77516652@incomming:11] BackGround("SIP/GW-0000001b", "/var/lib/asterisk/sounds/pr-medical/my/daygreeting") in new stack
    -- <SIP/GW-0000001b> Playing '/var/lib/asterisk/sounds/pr-medical/my/daygreeting.slin' (language 'en')
  == Spawn extension (incomming, 77516652, 11) exited non-zero on 'SIP/GW-0000001b'
[Oct 28 16:17:59] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77516652@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
  == Using SIP RTP CoS mark 5
       > 0x7fc900023760 -- Strict RTP learning after remote address set to: 192.168.3.18:24608
    -- Executing [106@LocalSets:1] Answer("SIP/6017-0000001c", "") in new stack
       > 0x7fc900023760 -- Strict RTP switching to RTP target address 192.168.3.18:24608 as source
    -- Executing [106@LocalSets:2] Verbose("SIP/6017-0000001c", "2,Call to exten 106") in new stack
  == Call to exten 106
    -- Executing [106@LocalSets:3] Gosub("SIP/6017-0000001c", "SubVoicemail,start,1(106,SIP/6009)") in new stack
    -- Executing [start@SubVoicemail:1] Verbose("SIP/6017-0000001c", "2,Call to ext 106") in new stack
  == Call to ext 106
    -- Executing [start@SubVoicemail:2] GotoIf("SIP/6017-0000001c", "0?blocked,1") in new stack
    -- Executing [start@SubVoicemail:3] Dial("SIP/6017-0000001c", "SIP/6009,40,tk") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/6009
    -- SIP/6009-0000001d is ringing
[Oct 28 16:18:11] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77616163@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
       > 0x7fc95c007a40 -- Strict RTP learning after remote address set to: 192.168.3.11:26816
    -- SIP/6009-0000001d answered SIP/6017-0000001c
    -- Channel SIP/6009-0000001d joined 'simple_bridge' basic-bridge <7912e5b9-88f3-434d-9807-82879b11d959>
    -- Channel SIP/6017-0000001c joined 'simple_bridge' basic-bridge <7912e5b9-88f3-434d-9807-82879b11d959>
       > 0x7fc95c007a40 -- Strict RTP switching to RTP target address 192.168.3.11:26816 as source
[Oct 28 16:18:14] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77516652@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
       > 0x7fc900023760 -- Strict RTP learning complete - Locking on source address 192.168.3.18:24608
    -- Started music on hold, class 'default', on channel 'SIP/6017-0000001c'
  == Using SIP RTP CoS mark 5
       > 0x7fc9000320f0 -- Strict RTP learning after remote address set to: 192.168.3.11:21962
    -- Executing [115@LocalSets:1] Answer("SIP/6009-0000001e", "") in new stack
       > 0x7fc9000320f0 -- Strict RTP switching to RTP target address 192.168.3.11:21962 as source
    -- Executing [115@LocalSets:2] Verbose("SIP/6009-0000001e", "2,Call to exten 115") in new stack
  == Call to exten 115
    -- Executing [115@LocalSets:3] Gosub("SIP/6009-0000001e", "SubVoicemail,start,1(115,SIP/6021)") in new stack
    -- Executing [start@SubVoicemail:1] Verbose("SIP/6009-0000001e", "2,Call to ext 115") in new stack
  == Call to ext 115
    -- Executing [start@SubVoicemail:2] GotoIf("SIP/6009-0000001e", "0?blocked,1") in new stack
    -- Executing [start@SubVoicemail:3] Dial("SIP/6009-0000001e", "SIP/6021,40,tk") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/6021
    -- SIP/6021-0000001f is ringing
       > 0x7fc9000320f0 -- Strict RTP learning complete - Locking on source address 192.168.3.11:21962
       > 0x7fc94400e740 -- Strict RTP learning after remote address set to: 192.168.3.19:18618
    -- SIP/6021-0000001f answered SIP/6009-0000001e
    -- Channel SIP/6021-0000001f joined 'simple_bridge' basic-bridge <edf750ee-157b-4e01-acea-176f068c763e>
    -- Channel SIP/6009-0000001e joined 'simple_bridge' basic-bridge <edf750ee-157b-4e01-acea-176f068c763e>
       > 0x7fc94400e740 -- Strict RTP switching to RTP target address 192.168.3.19:18618 as source
[Oct 28 16:18:27] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77686929@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
[Oct 28 16:18:28] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77605666@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
[Oct 28 16:18:29] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77516651@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
[Oct 28 16:18:29] ERROR[6130]: chan_sip.c:17790 register_verify: Peer 'GW' is trying to register, but not configured as host=dynamic
[Oct 28 16:18:29] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:GW@192.168.3.2>' failed for '192.168.3.4:5060' - Peer is not supposed to register
[Oct 28 16:18:29] ERROR[6130]: chan_sip.c:17790 register_verify: Peer 'GW' is trying to register, but not configured as host=dynamic
[Oct 28 16:18:29] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:GW@192.168.3.2>' failed for '192.168.3.4:5060' - Peer is not supposed to register
       > 0x7fc94400e740 -- Strict RTP learning complete - Locking on source address 192.168.3.19:18618
    -- Started music on hold, class 'default', on channel 'SIP/6009-0000001e'
    -- Stopped music on hold on SIP/6009-0000001e
    -- Channel SIP/6009-0000001e left 'simple_bridge' basic-bridge <edf750ee-157b-4e01-acea-176f068c763e>
    -- Channel SIP/6021-0000001f left 'simple_bridge' basic-bridge <edf750ee-157b-4e01-acea-176f068c763e>
  == Spawn extension (SubVoicemail, start, 3) exited non-zero on 'SIP/6009-0000001e'
  == Using SIP RTP CoS mark 5
       > 0x7fc900042390 -- Strict RTP learning after remote address set to: 192.168.3.19:22176
    -- Executing [106@LocalSets:1] Answer("SIP/6021-00000020", "") in new stack
       > 0x7fc900042390 -- Strict RTP switching to RTP target address 192.168.3.19:22176 as source
    -- Executing [106@LocalSets:2] Verbose("SIP/6021-00000020", "2,Call to exten 106") in new stack
  == Call to exten 106
    -- Executing [106@LocalSets:3] Gosub("SIP/6021-00000020", "SubVoicemail,start,1(106,SIP/6009)") in new stack
    -- Executing [start@SubVoicemail:1] Verbose("SIP/6021-00000020", "2,Call to ext 106") in new stack
  == Call to ext 106
    -- Executing [start@SubVoicemail:2] GotoIf("SIP/6021-00000020", "0?blocked,1") in new stack
    -- Executing [start@SubVoicemail:3] Dial("SIP/6021-00000020", "SIP/6009,40,tk") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/6009
    -- SIP/6009-00000021 is ringing
       > 0x7fc95400b8b0 -- Strict RTP learning after remote address set to: 192.168.3.11:19518
    -- SIP/6009-00000021 answered SIP/6021-00000020
    -- Channel SIP/6009-00000021 joined 'simple_bridge' basic-bridge <cd7438ad-327c-41ae-855e-0021ed1f7a5e>
    -- Channel SIP/6021-00000020 joined 'simple_bridge' basic-bridge <cd7438ad-327c-41ae-855e-0021ed1f7a5e>
       > 0x7fc95400b8b0 -- Strict RTP switching to RTP target address 192.168.3.11:19518 as source
[Oct 28 16:18:44] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77516652@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
       > 0x7fc900042390 -- Strict RTP learning complete - Locking on source address 192.168.3.19:22176
  == Using SIP RTP CoS mark 5
       > 0x7fc9000320f0 -- Strict RTP learning after remote address set to: 192.168.3.4:10130
    -- Executing [77516651@incomming:1] Answer("SIP/GW-00000022", "") in new stack
       > 0x7fc9000320f0 -- Strict RTP switching to RTP target address 192.168.3.4:10130 as source
    -- Executing [77516651@incomming:2] GotoIf("SIP/GW-00000022", "0?blocked,1") in new stack
    -- Executing [77516651@incomming:3] GotoIfTime("SIP/GW-00000022", "06:30-23:59,sat,*,*?daygreeting") in new stack
    -- Executing [77516651@incomming:4] GotoIfTime("SIP/GW-00000022", "06:30-20:30,sun,*,*?daygreeting") in new stack
    -- Goto (incomming,77516651,11)
    -- Executing [77516651@incomming:11] BackGround("SIP/GW-00000022", "/var/lib/asterisk/sounds/pr-medical/my/daygreeting") in new stack
    -- <SIP/GW-00000022> Playing '/var/lib/asterisk/sounds/pr-medical/my/daygreeting.slin' (language 'en')
       > 0x7fc95400b8b0 -- Strict RTP learning complete - Locking on source address 192.168.3.11:19518
    -- Stopped music on hold on SIP/6017-0000001c
    -- Channel SIP/6017-0000001c left 'simple_bridge' basic-bridge <7912e5b9-88f3-434d-9807-82879b11d959>
    -- Channel SIP/6009-0000001d left 'simple_bridge' basic-bridge <7912e5b9-88f3-434d-9807-82879b11d959>
  == Spawn extension (SubVoicemail, start, 3) exited non-zero on 'SIP/6017-0000001c'
    -- Executing [77516651@incomming:12] Goto("SIP/GW-00000022", "menuprompt") in new stack
    -- Goto (incomming,77516651,13)
    -- Executing [77516651@incomming:13] BackGround("SIP/GW-00000022", "/var/lib/asterisk/sounds/pr-medical/my/main-menu1") in new stack
    -- <SIP/GW-00000022> Playing '/var/lib/asterisk/sounds/pr-medical/my/main-menu1.slin' (language 'en')
       > 0x7fc9000320f0 -- Strict RTP learning complete - Locking on source address 192.168.3.4:10130
  == Using SIP RTP CoS mark 5
       > 0x7fc90004c3b0 -- Strict RTP learning after remote address set to: 192.168.3.18:22836
    -- Executing [115@LocalSets:1] Answer("SIP/6017-00000023", "") in new stack
       > 0x7fc90004c3b0 -- Strict RTP switching to RTP target address 192.168.3.18:22836 as source
    -- Executing [115@LocalSets:2] Verbose("SIP/6017-00000023", "2,Call to exten 115") in new stack
  == Call to exten 115
    -- Executing [115@LocalSets:3] Gosub("SIP/6017-00000023", "SubVoicemail,start,1(115,SIP/6021)") in new stack
    -- Executing [start@SubVoicemail:1] Verbose("SIP/6017-00000023", "2,Call to ext 115") in new stack
  == Call to ext 115
    -- Executing [start@SubVoicemail:2] GotoIf("SIP/6017-00000023", "0?blocked,1") in new stack
    -- Executing [start@SubVoicemail:3] Dial("SIP/6017-00000023", "SIP/6021,40,tk") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/6021
    -- SIP/6021-00000024 is ringing
    -- Channel SIP/6003-00000014 left 'simple_bridge' basic-bridge <da206ac1-e5bd-4f09-90c5-bc2d3eb01d2d>
    -- Channel SIP/192.168.3.4:5060-00000015 left 'simple_bridge' basic-bridge <da206ac1-e5bd-4f09-90c5-bc2d3eb01d2d>
  == Spawn extension (LocalSets, 09121861003, 1) exited non-zero on 'SIP/6003-00000014'
       > Saved useragent "Cisco-CP7911G/8.4.0" for peer 6017
       > 0x7fc90004c3b0 -- Strict RTP learning complete - Locking on source address 192.168.3.18:22836
    -- Executing [1@incomming:1] Verbose("SIP/GW-00000022", "1, Caller "" <09122931680> has entered the sales queue") in new stack
  Caller "" <09122931680> has entered the sales queue
    -- Executing [1@incomming:2] Goto("SIP/GW-00000022", "Queues,8001,1") in new stack
    -- Goto (Queues,8001,1)
    -- Executing [8001@Queues:1] Verbose("SIP/GW-00000022", "2,"" <09122931680> entering the paziresh queue") in new stack
  == "" <09122931680> entering the paziresh queue
    -- Executing [8001@Queues:2] Queue("SIP/GW-00000022", "paziresh,tk") in new stack
[Oct 28 16:19:11] WARNING[6408][C-00000015]: app_queue.c:8035 queue_exec: Unable to join queue 'paziresh'
    -- Executing [8001@Queues:3] Hangup("SIP/GW-00000022", "") in new stack
  == Spawn extension (Queues, 8001, 3) exited non-zero on 'SIP/GW-00000022'
[Oct 28 16:19:11] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77616163@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
    -- Channel SIP/6021-00000020 left 'simple_bridge' basic-bridge <cd7438ad-327c-41ae-855e-0021ed1f7a5e>
    -- Channel SIP/6009-00000021 left 'simple_bridge' basic-bridge <cd7438ad-327c-41ae-855e-0021ed1f7a5e>
  == Spawn extension (SubVoicemail, start, 3) exited non-zero on 'SIP/6021-00000020'
  == Spawn extension (SubVoicemail, start, 3) exited non-zero on 'SIP/6017-00000023'
[Oct 28 16:19:27] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77686929@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
[Oct 28 16:19:28] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77605666@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
[Oct 28 16:19:29] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77516651@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
[Oct 28 16:19:29] ERROR[6130]: chan_sip.c:17790 register_verify: Peer 'GW' is trying to register, but not configured as host=dynamic
[Oct 28 16:19:29] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:GW@192.168.3.2>' failed for '192.168.3.4:5060' - Peer is not supposed to register
[Oct 28 16:19:29] ERROR[6130]: chan_sip.c:17790 register_verify: Peer 'GW' is trying to register, but not configured as host=dynamic
[Oct 28 16:19:29] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:GW@192.168.3.2>' failed for '192.168.3.4:5060' - Peer is not supposed to register
[Oct 28 16:19:29] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77516652@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
  == Using SIP RTP CoS mark 5
       > 0x7fc90000b8e0 -- Strict RTP learning after remote address set to: 192.168.3.18:30664
    -- Executing [106@LocalSets:1] Answer("SIP/6017-00000025", "") in new stack
       > 0x7fc90000b8e0 -- Strict RTP switching to RTP target address 192.168.3.18:30664 as source
    -- Executing [106@LocalSets:2] Verbose("SIP/6017-00000025", "2,Call to exten 106") in new stack
  == Call to exten 106
    -- Executing [106@LocalSets:3] Gosub("SIP/6017-00000025", "SubVoicemail,start,1(106,SIP/6009)") in new stack
    -- Executing [start@SubVoicemail:1] Verbose("SIP/6017-00000025", "2,Call to ext 106") in new stack
  == Call to ext 106
    -- Executing [start@SubVoicemail:2] GotoIf("SIP/6017-00000025", "0?blocked,1") in new stack
    -- Executing [start@SubVoicemail:3] Dial("SIP/6017-00000025", "SIP/6009,40,tk") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/6009
    -- SIP/6009-00000026 is ringing
       > 0x7fc968008200 -- Strict RTP learning after remote address set to: 192.168.3.11:19162
    -- SIP/6009-00000026 answered SIP/6017-00000025
    -- Channel SIP/6009-00000026 joined 'simple_bridge' basic-bridge <a10c1ecd-f9a3-4a1a-acda-5b6defc72b11>
    -- Channel SIP/6017-00000025 joined 'simple_bridge' basic-bridge <a10c1ecd-f9a3-4a1a-acda-5b6defc72b11>
       > 0x7fc968008200 -- Strict RTP switching to RTP target address 192.168.3.11:19162 as source
       > 0x7fc90000b8e0 -- Strict RTP learning complete - Locking on source address 192.168.3.18:30664
       > 0x7fc968008200 -- Strict RTP learning complete - Locking on source address 192.168.3.11:19162
  == Using SIP RTP CoS mark 5
       > 0x7fc900023760 -- Strict RTP learning after remote address set to: 192.168.1.68:8000
    -- Executing [112@LocalSets:1] Answer("SIP/6001-00000027", "") in new stack
       > 0x7fc900023760 -- Strict RTP switching to RTP target address 192.168.1.68:8000 as source
    -- Executing [112@LocalSets:2] Verbose("SIP/6001-00000027", "2,Call to exten 112") in new stack
  == Call to exten 112
    -- Executing [112@LocalSets:3] Gosub("SIP/6001-00000027", "SubVoicemail,start,1(112,SIP/6011)") in new stack
    -- Executing [start@SubVoicemail:1] Verbose("SIP/6001-00000027", "2,Call to ext 112") in new stack
  == Call to ext 112
    -- Executing [start@SubVoicemail:2] GotoIf("SIP/6001-00000027", "0?blocked,1") in new stack
    -- Executing [start@SubVoicemail:3] Dial("SIP/6001-00000027", "SIP/6011,40,tk") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/6011
    -- SIP/6011-00000028 is ringing
    -- Started music on hold, class 'default', on channel 'SIP/6017-00000025'
  == Using SIP RTP CoS mark 5
       > 0x7fc90002d1d0 -- Strict RTP learning after remote address set to: 192.168.3.11:20650
    -- Executing [107@LocalSets:1] Answer("SIP/6009-00000029", "") in new stack
       > 0x7fc90002d1d0 -- Strict RTP switching to RTP target address 192.168.3.11:20650 as source
    -- Executing [107@LocalSets:2] Verbose("SIP/6009-00000029", "2,Call to exten 107") in new stack
  == Call to exten 107
    -- Executing [107@LocalSets:3] Gosub("SIP/6009-00000029", "SubVoicemail,start,1(107,SIP/6016)") in new stack
    -- Executing [start@SubVoicemail:1] Verbose("SIP/6009-00000029", "2,Call to ext 107") in new stack
  == Call to ext 107
    -- Executing [start@SubVoicemail:2] GotoIf("SIP/6009-00000029", "0?blocked,1") in new stack
    -- Executing [start@SubVoicemail:3] Dial("SIP/6009-00000029", "SIP/6016,40,tk") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/6016
    -- SIP/6016-0000002a is ringing
       > 0x7fc900023760 -- Strict RTP learning complete - Locking on source address 192.168.1.68:8000
  == Spawn extension (SubVoicemail, start, 3) exited non-zero on 'SIP/6009-00000029'
    -- Started music on hold, class 'default', on channel 'SIP/192.168.3.4:5060-00000018'
       > 0x7fc9580073c0 -- Strict RTP learning after remote address set to: 192.168.1.85:8002
    -- SIP/6011-00000028 answered SIP/6001-00000027
    -- Channel SIP/6011-00000028 joined 'simple_bridge' basic-bridge <67ea6fdc-0e1b-469a-98bf-e26b5d9e43e8>
    -- Channel SIP/6001-00000027 joined 'simple_bridge' basic-bridge <67ea6fdc-0e1b-469a-98bf-e26b5d9e43e8>
       > 0x7fc9580073c0 -- Strict RTP switching to RTP target address 192.168.1.85:8002 as source
    -- Stopped music on hold on SIP/192.168.3.4:5060-00000018
    -- Channel SIP/192.168.3.4:5060-00000018 left 'simple_bridge' basic-bridge <9e103730-9878-4044-9fc5-d8f925ab7171>
    -- Channel SIP/6011-00000017 left 'simple_bridge' basic-bridge <9e103730-9878-4044-9fc5-d8f925ab7171>
  == Spawn extension (LocalSets, 33801344, 1) exited non-zero on 'SIP/6011-00000017'
       > 0x7fc9580073c0 -- Strict RTP learning complete - Locking on source address 192.168.1.85:8002
    -- Stopped music on hold on SIP/6017-00000025
    -- Channel SIP/6017-00000025 left 'simple_bridge' basic-bridge <a10c1ecd-f9a3-4a1a-acda-5b6defc72b11>
    -- Channel SIP/6009-00000026 left 'simple_bridge' basic-bridge <a10c1ecd-f9a3-4a1a-acda-5b6defc72b11>
  == Spawn extension (SubVoicemail, start, 3) exited non-zero on 'SIP/6017-00000025'
[Oct 28 16:20:11] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77616163@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
[Oct 28 16:20:27] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77686929@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
[Oct 28 16:20:28] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77605666@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
[Oct 28 16:20:29] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77516651@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
[Oct 28 16:20:29] ERROR[6130]: chan_sip.c:17790 register_verify: Peer 'GW' is trying to register, but not configured as host=dynamic
[Oct 28 16:20:29] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:GW@192.168.3.2>' failed for '192.168.3.4:5060' - Peer is not supposed to register
[Oct 28 16:20:29] ERROR[6130]: chan_sip.c:17790 register_verify: Peer 'GW' is trying to register, but not configured as host=dynamic
[Oct 28 16:20:29] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:GW@192.168.3.2>' failed for '192.168.3.4:5060' - Peer is not supposed to register
[Oct 28 16:20:29] NOTICE[6130]: chan_sip.c:28691 handle_request_register: Registration from '<sip:77516652@192.168.3.2>' failed for '192.168.3.4:5060' - Wrong password
  == Using SIP RTP CoS mark 5
       > 0x7fc9000281a0 -- Strict RTP learning after remote address set to: 10.123.101.162:18018
    -- Executing [75197@incomming:1] Answer("SIP/75197-0000002b", "") in new stack
       > 0x7fc9000281a0 -- Strict RTP switching to RTP target address 10.123.101.162:18018 as source
    -- Executing [75197@incomming:2] GotoIf("SIP/75197-0000002b", "0?blocked,1") in new stack
    -- Executing [75197@incomming:3] GotoIfTime("SIP/75197-0000002b", "06:30-23:59,sat,*,*?daygreeting") in new stack
    -- Executing [75197@incomming:4] GotoIfTime("SIP/75197-0000002b", "06:30-20:30,sun,*,*?daygreeting") in new stack
    -- Goto (incomming,75197,11)
    -- Executing [75197@incomming:11] BackGround("SIP/75197-0000002b", "/var/lib/asterisk/sounds/pr-medical/my/daygreeting") in new stack
    -- <SIP/75197-0000002b> Playing '/var/lib/asterisk/sounds/pr-medical/my/daygreeting.slin' (language 'pr-medical')
       > 0x7fc9000281a0 -- Strict RTP learning complete - Locking on source address 10.123.101.162:18018
PBX*CLI>

the provided logs are from console verbosity as they didn’t logged out in the file named ‘full’ even when I increase the verbose level in 'core set verbose ’

and there is something else I figured out which is blind transfer is working in my cisco phones(but inter digit period time is very short which I dont know how to increase it)

You didn’t turn on the channel driver sip logging. As I can now see that you are using chan_sip, you need to do “sip set debug on” on the CLI.

Interdigit times suggests you are doing a features.conf transfer, which is technology independent, rather than a SIP one.

hi all
sorry for delay in answer , I did what you asked but as there are lots of out put in my log file its difficult to analyse the content,
I managed to run a SPAN on my cisco switch and capture traffic between two end device
(its good to filter the capture file for showing sip and rtp among other traffic)
my scenario :slight_smile:

6017(channel-in-asterisk)-----------> 6021--------------->6016
108(exten) ---------------------------- 115 -------------------- 107
192.168.3.18---------------------- 192.168.3.19---------------- 192.168.3.14

108 ext call to 115 and the 115 hit transfer button on phone and dial 107.
108 get hold music and 107 disconnect after one ring.
108 stuck in hold.

google-drive