Got a strange problem.
Trying to call from a sip client to a normal phone or exetension.
This results always in a one way audio connenction.
I use the odbc database, and can't really find the problem.
Can anybody help me in the right direction.
There seems to be no errors at all.
[general]
context=public
allowguest=no
allowoverlap=no
udpbindaddr=0.0.0.0:15060
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
language=ja
externaddr=x.x.x.x
localnet=x.x.x.x/255.255.240.0
nat=force_rport,comedia
rtcachefriends=yes
disallow=all
allow=ulaw
allow=alaw
allow=gsm
/var/log/asterisk/messages
[Apr 12 10:44:36] VERBOSE[23055][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] pbx.c: Executing [52431824#context_tok:1] NoOp("SIP/inbound_1_1-00000003", "inbound") in new stack
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] pbx.c: Executing [52431824#context_tok:2] Dial("SIP/inbound_1_1-00000003", "SIP/1_1_1_1/1_1_1_1&SIP/1_1_1_2/1_1_1_2&SIP/1_1_1_3/1_1_1_3&SIP/1_1_1_4/1_1_1_4") in new stack
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] WARNING[25771][C-00000001] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] netsock2.c: Using SIP RTP CoS mark 5
[Apr 12 10:44:36] WARNING[25771][C-00000001] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: Called SIP/1_1_1_1/1_1_1_1
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: Called SIP/1_1_1_3/1_1_1_3
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_1-00000004 is ringing
[Apr 12 10:44:36] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_3-00000005 is ringing
[Apr 12 10:44:44] VERBOSE[25771][C-00000001] app_dial.c: SIP/1_1_1_3-00000005 answered SIP/inbound_1_1-00000003
[Apr 12 10:44:44] VERBOSE[25846][C-00000001] bridge_channel.c: Channel SIP/1_1_1_3-00000005 joined 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:44] VERBOSE[25771][C-00000001] bridge_channel.c: Channel SIP/inbound_1_1-00000003 joined 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25846][C-00000001] bridge_channel.c: Channel SIP/1_1_1_3-00000005 left 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25771][C-00000001] bridge_channel.c: Channel SIP/inbound_1_1-00000003 left 'simple_bridge' basic-bridge <16f760ce-43f9-4f36-8aa3-865c4f2e8151>
[Apr 12 10:44:52] VERBOSE[25771][C-00000001] pbx.c: Spawn extension (context_tok, 52431824, 2) exited non-zero on 'SIP/inbound_1_1-00000003'
Have tried several things, and searched on the net, coudn't find the correct solution.
To help people in the future with the same problem.
In the odbc database we had the standard data on null.
Setting qualify to yes for the users in the database fixed the problem.
Related
I am having an issue/trying to make something work within Asterisk. I have a trunk to an Ascom Nurse call system and there is a basic function to dial from a handheld device into a patient room. I am able to establish the call from Asterisk to the nurse call server. The nurse call server sends a Refer message
to a different address on the same subnet but Asterisk cannot find that device. If I manually type a dialplan to route calls to that device it does connect but the system could have many addresses and it would be impossible to tell what address and dial number would be in the Refer message
Asterik 10.2.87.201
Ascom 10.2.87.1
Refer Message refer-to: sip:V1003B0G65605773L0#10.2.87.11
I need to be able to have Asterisk transfer to that ext and domain dynamically.
This is a working example with a dialplan telling the system to manually send to the 10.2.87.11 address
exten => _VX.,n,Dial(SIP/${EXTEN}#10.2.87.11)
-- Executing [201*65609848#default:2] Dial("SIP/1341-000001cd", "SIP/T6/201*65609848") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/T6/201*65609848
> 0x7fba38016560 -- Strict RTP learning after remote address set to: 10.2.87.1:8766
-- SIP/T6-000001ce answered SIP/1341-000001cd
-- Channel SIP/T6-000001ce joined 'simple_bridge' basic-bridge <07b9ccbf-138d-423b-87cf-ad6c41336591>
-- Channel SIP/1341-000001cd joined 'simple_bridge' basic-bridge <07b9ccbf-138d-423b-87cf-ad6c41336591>
> Bridge 07b9ccbf-138d-423b-87cf-ad6c41336591: switching from simple_bridge technology to native_rtp
> Locally RTP bridged 'SIP/1341-000001cd' and 'SIP/T6-000001ce' in stack
> 0x7fb9c017aa00 -- Strict RTP learning after remote address set to: 192.168.21.82:16734
> Locally RTP bridged 'SIP/1341-000001cd' and 'SIP/T6-000001ce' in stack
> 0x7fb9c017aa00 -- Strict RTP switching to RTP target address 192.168.21.82:16734 as source
> Locally RTP bridged 'SIP/1341-000001cd' and 'SIP/T6-000001ce' in stack
-- Channel SIP/T6-000001ce left 'native_rtp' basic-bridge <07b9ccbf-138d-423b-87cf-ad6c41336591>
> Bridge 07b9ccbf-138d-423b-87cf-ad6c41336591: switching from native_rtp technology to simple_bridge
-- Channel SIP/1341-000001cd left 'simple_bridge' basic-bridge <07b9ccbf-138d-423b-87cf-ad6c41336591>
-- Executing [V1003B0G65609848L0#default:1] NoOp("SIP/1341-000001cd", "V1003B0G65609848L0") in new stack
-- Executing [V1003B0G65609848L0#default:2] Dial("SIP/1341-000001cd", "SIP/V1003B0G65609848L0#10.2.87.11") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/V1003B0G65609848L0#10.2.87.11
> 0x7fb9c017aa00 -- Strict RTP learning complete - Locking on source address 192.168.21.82:16734
> 0x7fba3802cb80 -- Strict RTP learning after remote address set to: 10.2.87.11:5012
-- SIP/10.2.87.11-000001cf answered SIP/1341-000001cd
-- Channel SIP/10.2.87.11-000001cf joined 'simple_bridge' basic-bridge <149e09d8-7d19-46fb-9135-43ffa33862e1>
-- Channel SIP/1341-000001cd joined 'simple_bridge' basic-bridge <149e09d8-7d19-46fb-9135-43ffa33862e1>
> Bridge 149e09d8-7d19-46fb-9135-43ffa33862e1: switching from simple_bridge technology to native_rtp
> Locally RTP bridged 'SIP/1341-000001cd' and 'SIP/10.2.87.11-000001cf' in stack
> 0x7fba3802cb80 -- Strict RTP switching to RTP target address 10.2.87.11:5012 as source
> 0x7fba3802cb80 -- Strict RTP learning complete - Locking on source address 10.2.87.11:5012
-- Channel SIP/1341-000001cd left 'native_rtp' basic-bridge <149e09d8-7d19-46fb-9135-43ffa33862e1>
-- Channel SIP/10.2.87.11-000001cf left 'native_rtp' basic-bridge <149e09d8-7d19-46fb-9135-43ffa33862e1>
== Spawn extension (default, V1003B0G65609848L0, 2) exited non-zero on 'SIP/1341-000001cd'
Not working when using a Transfer dialplan
> 0x7fb9c017aa00 -- Strict RTP learning after remote address set to: 192.168.21.82:16766
-- Executing [201*65609851#default:1] NoOp("SIP/1341-000001de", "201*65609851") in new stack
-- Executing [201*65609851#default:2] Dial("SIP/1341-000001de", "SIP/T6/201*65609851") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/T6/201*65609851
> 0x7fba4001ebf0 -- Strict RTP learning after remote address set to: 10.2.87.1:8766
-- SIP/T6-000001df answered SIP/1341-000001de
-- Channel SIP/T6-000001df joined 'simple_bridge' basic-bridge <9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab>
-- Channel SIP/1341-000001de joined 'simple_bridge' basic-bridge <9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab>
> Bridge 9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab: switching from simple_bridge technology to native_rtp
> Locally RTP bridged 'SIP/1341-000001de' and 'SIP/T6-000001df' in stack
> 0x7fb9c017aa00 -- Strict RTP learning after remote address set to: 192.168.21.82:16766
> Locally RTP bridged 'SIP/1341-000001de' and 'SIP/T6-000001df' in stack
> 0x7fb9c017aa00 -- Strict RTP switching to RTP target address 192.168.21.82:16766 as source
> Locally RTP bridged 'SIP/1341-000001de' and 'SIP/T6-000001df' in stack
-- Channel SIP/T6-000001df left 'native_rtp' basic-bridge <9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab>
> Bridge 9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab: switching from native_rtp technology to simple_bridge
-- Channel SIP/1341-000001de left 'simple_bridge' basic-bridge <9b4ac0b8-3bcf-4ef3-862a-46b36a7cddab>
-- Executing [V1003B0G65609851L0#default:1] NoOp("SIP/1341-000001de", "V1003B0G65609851L0") in new stack
-- Executing [V1003B0G65609851L0#default:2] Transfer("SIP/1341-000001de", "SIP/V1003B0G65609851L0") in new stack
== Using SIP RTP CoS mark 5
> 0x7fb9c0174940 -- Strict RTP learning after remote address set to: 192.168.21.82:16770
-- Executing [V1003B0G65609851L0#default:1] NoOp("SIP/1341-000001e0", "V1003B0G65609851L0") in new stack
-- Executing [V1003B0G65609851L0#default:2] Transfer("SIP/1341-000001e0", "SIP/V1003B0G65609851L0") in new stack
-- Auto fallthrough, channel 'SIP/1341-000001e0' status is 'UNKNOWN'
== Using SIP RTP CoS mark 5
> 0x7fb9c0174940 -- Strict RTP learning after remote address set to: 192.168.21.82:16770
-- Executing [V1003B0G65609851L0#default:1] NoOp("SIP/1341-000001e1", "V1003B0G65609851L0") in new stack
-- Executing [V1003B0G65609851L0#default:2] Transfer("SIP/1341-000001e1", "SIP/V1003B0G65609851L0") in new stack
-- Auto fallthrough, channel 'SIP/1341-000001e1' status is 'UNKNOWN'
== Using SIP RTP CoS mark 5
> 0x7fb9c0174940 -- Strict RTP learning after remote address set to: 192.168.21.82:16770
-- Executing [V1003B0G65609851L0#default:1] NoOp("SIP/1341-000001e2", "V1003B0G65609851L0") in new stack
-- Executing [V1003B0G65609851L0#default:2] Transfer("SIP/1341-000001e2", "SIP/V1003B0G65609851L0") in new stack
-- Auto fallthrough, channel 'SIP/1341-000001e2' status is 'UNKNOWN'
> 0x7fb9c0174940 -- Strict RTP learning after remote address set to: 192.168.21.82:16770
-- Auto fallthrough, channel 'SIP/1341-000001de' status is 'ANSWER'
In short I need the last example to dynamically send to the domain in the refer message.
Thank you for you help
I am not sure if this is correct but I got this to work by creating users for all the devices I needed to communicate to and adjusting my extensions.conf
exten => _VX.,n,Dial(SIP/${EXTEN}#${SIPDOMAIN})
fullname = 10.2.87.10
secret =
hasvoicemail = no
host = 10.2.87.10
userqphone = yes
qualify = no
hassip = yes
hasiax = no
callwaiting = yes
context = default
Using Asterisk 16 and a local SIP trunk provider, we can make outgoing calls, but incoming calls do not work.
If I call from my mobile, I see the call Invite on the server, and I see the call being answered. On my mobile, however, the call remains unanswered, and is then rejected.
These exact settings worked, when working on a different server (hosted locally instead of at the data center).
EXTENSIONS.CONF
[IncomingContext]
exten=>[NumberRemoved],1,NoOp(Incoming)
same=>n,Answer()
same=>n,Playback(tt-monkeys)
same=>n,HangUp()
PJSIP.CONF
[transport-udp-main]
type=transport
protocol=udp
bind=0.0.0.0:5060
;DEVICES
;TEMPLATES
[gEndpoint](!)
type=endpoint
transport=transport-udp-main
context=from-internal
disallow=all
allow=alaw
;allow=opus
;allow=alaw
[gAors](!)
type=aor
max_contacts=2
[gAuth](!)
type=auth
auth_type=userpass
[LOCAL1040](gEndpoint)
aors=LOCAL1040
auth=LOCAL1040
[LOCAL1040](gAors)
[LOCAL1040](gAuth)
username=LOCAL1040
password=#1040*
;--------TRUNK PROVIDER
[siptrunk]
type=auth
auth_type=userpass
username=USERNAME
password=PASSWORD
[siptrunk]
type=aor
contact=sip:sip.PROVIDER.co.za:5060
[siptrunk]
type=registration
outbound_auth=siptrunk
server_uri=sip:USERNAME#sip.PROVIDER.co.za:5060
client_uri=sip:USERNAME#sip.PROVIDER.co.za
contact_user=9999
[siptrunk]
type=identify
endpoint=siptrunk
match=sip.PROVIDER.co.za
[siptrunk]
type=endpoint
transport=transport-udp-main
context=IncomingContext
direct_media=yes
disallow=all
allow=ulaw,iLBC,opus,GSM
outbound_auth=siptrunk
aors=siptrunk
Logs:
[Oct 7 14:02:41] Asterisk 16.5.1 built by root # asterisk on a x86_64 running Linux on 2019-09-20 09:48:51 UTC
[Oct 7 14:02:41] NOTICE[1498] loader.c: 304 modules will be loaded.
[Oct 7 14:02:47] NOTICE[1498] cdr.c: CDR logging disabled.
[Oct 7 14:02:49] NOTICE[1498] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Oct 7 14:02:49] WARNING[1498] res_phoneprov.c: Unable to find a valid server address or name.
[Oct 7 14:02:49] NOTICE[1498] chan_skinny.c: Configuring skinny from skinny.conf
[Oct 7 14:02:49] ERROR[1498] ari/config.c: No configured users for ARI
[Oct 7 14:02:49] NOTICE[1498] confbridge/conf_config_parser.c: Adding default_menu menu to app_confbridge
[Oct 7 14:02:49] WARNING[1498] config.c: parse error: No category context for line 33 of /etc/asterisk/cdr_adaptive_odbc.conf
[Oct 7 14:02:49] WARNING[1498] cdr_adaptive_odbc.c: Unable to load cdr_adaptive_odbc.conf. No adaptive ODBC CDRs.
[Oct 7 14:02:49] NOTICE[1498] cel_custom.c: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Oct 7 14:02:50] WARNING[1498] res_hep_pjsip.c: res_hep is disabled; declining module load
[Oct 7 14:02:50] WARNING[1498] loader.c: Some non-required modules failed to load.
[Oct 7 14:02:50] ERROR[1498] loader.c: Error loading module 'cdr_addon_odbc.so': /usr/lib/asterisk/modules/cdr_addon_odbc.so: cannot open shared object file: No such file or directory
[Oct 7 14:02:50] ERROR[1498] loader.c: cel_sqlite3_custom declined to load.
[Oct 7 14:02:50] ERROR[1498] loader.c: cdr_sqlite3_custom declined to load.
[Oct 7 14:02:50] ERROR[1498] loader.c: res_hep_pjsip declined to load.
[Oct 7 14:02:50] VERBOSE[1498] asterisk.c: Asterisk Ready.
[Oct 7 14:03:38] WARNING[1708] res_pjsip_pubsub.c: No registered publish handler for event presence
[Oct 7 14:04:48] VERBOSE[2253] pbx_variables.c: Setting global variable 'SIPDOMAIN' to '192.168.26.11'
[Oct 7 14:04:48] VERBOSE[2258][C-00000001] pbx.c: Executing [[Number Removed]#IncomingContext:1] NoOp("PJSIP/siptrunk-00000000", "Incoming [Number Removed]") in new stack
[Oct 7 14:04:48] VERBOSE[2258][C-00000001] pbx.c: Executing [[Number Removed]#IncomingContext:2] Answer("PJSIP/siptrunk-00000000", "") in new stack
[Oct 7 14:04:48] VERBOSE[2253] res_rtp_asterisk.c: 0x7f17400122e0 -- Strict RTP learning after remote address set to: XXX.XXX.XXX.XXX:40570
[Oct 7 14:04:49] VERBOSE[2258][C-00000001] pbx.c: Executing [[Number Removed]#IncomingContext:3] Playback("PJSIP/siptrunk-00000000", "tt-monkeys") in new stack
[Oct 7 14:04:49] VERBOSE[2258][C-00000001] file.c: <PJSIP/siptrunk-00000000> Playing 'tt-monkeys.gsm' (language 'en')
[Oct 7 14:05:05] VERBOSE[2258][C-00000001] pbx.c: Executing [[Number Removed]#IncomingContext:4] Hangup("PJSIP/siptrunk-00000000", "") in new stack
[Oct 7 14:05:05] VERBOSE[2258][C-00000001] pbx.c: Spawn extension (IncomingContext, [Number Removed], 4) exited non-zero on 'PJSIP/siptrunk-00000000'
May be your server behind the nat. In pjsip configuration analogue net=yes is below.
Add this three line into siptrunk endpoint.
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
Note: Don't forget reload Asterisk.
Check that your firewall not block rtp ports
Default ports are 10000-20000. For more info see rtp.conf
I am trying to make a outgoing from an asterisk pbx using .call file but every time .call file is moved in outgoing folder my cli shows
[Jun 16 15:38:12] NOTICE[30435]: pbx_spool.c:372 attempt_thread: Call failed to go through, reason (1) Hangup
[Jun 16 15:38:12] NOTICE[30435]: pbx_spool.c:375 attempt_thread: Queued call to DAHDI/g0/09716927126 expired without completion after 0 attempts
-- Span 1: Channel 0/1 got hangup request, cause 16
-- Hungup 'DAHDI/i1/09711590094-103a'
[Jun 16 15:38:17] NOTICE[30434]: pbx_spool.c:372 attempt_thread: Call failed to go through, reason (1) Hangup
[Jun 16 15:38:17] NOTICE[30434]: pbx_spool.c:375 attempt_thread: Queued call to DAHDI/g0/09711590094 expired without completion after 0 attempts
-- Attempting call on DAHDI/g0/09711590094 for 4759509#outgoing1:1 (Retry 1)
-- Attempting call on DAHDI/g0/09716927126 for 4759509#outgoing1:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Requested transfer capability: 0x00 - SPEECH
-- Span 1: Channel 0/2 got hangup request, cause 31
-- Hungup 'DAHDI/i1/09716927126-103d'
my .call file
Channel: DAHDI/g0/09711590094
MaxRetries: 1
RetryTime: 600
WaitTime: 30
Context: outgoing1
Extension: 10
Priority: 1
The call could not be connected.Anybody knows what would be the possible reason for that?
Thanks in advance
This error mean you can't call as requested via dahdi/g0
Very likly you have configure correctly your dahdi card.
I am Novice in asterisk I installed Asterisk but now when I calling with telephony call cannot
come into asterisk &
I config ed Outgoing call bat call cannot out asterisk when I write(asterisk -vvvvvr)
& I calling with outdoor display for me
-- Executing [09396464991#DLPN_Main:1] Macro("SIP/6001-00000000", "trunkdial -failover-0.3,DAHDI/g2/09396464991,DAHDI/g1/09396464991,trunk_1,trunk_1") in newstack
-- Executing [s#macro-trunkdial-failover-0.3:1] GotoIf("SIP/6001-00000000","0?1-fmsetcid,1") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:2] GotoIf("SIP/6001-00000000","0?1-setgbobname,1") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:3] Set("SIP/6001-00000000", "CALLERID(num)=6001") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:4] Set("SIP/6001-00000000", "CALLERID(all)=") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:5] GotoIf("SIP/6001-00000000","0?1-dial,1") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:6] Set("SIP/6001-00000000", "CALLERID(all)=") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:7] Set("SIP/6001-00000000", "CALLERID(all)=") in new stack
-- Executing [s#macro-trunkdial-failover-0.3:8] Goto("SIP/6001-00000000", "1-dial,1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-dial,1)
-- Executing [1-dial#macro-trunkdial-failover-0.3:1] Dial("SIP/6001-00000000:", "DAHDI/g2/09396464991") in new stack
[Mar 10 13:40:04] **WARNING[2106]: channel.c:5627 ast_request: No channel type registered for 'DAHDI'**
[Mar 10 13:40:04] WARNING[2106]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented)== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [1-dial#macro-trunkdial-failover-0.3:2] GotoIf("SIP/6001-00000000", "20 > 0?1-CHANUNAVAIL,1:1-out,1") in new stack
-- Goto (macro-trunkdial-failover-0.3,1-CHANUNAVAIL,1)
-- Executing [1-CHANUNAVAIL#macro-trunkdial-failover-0.3:1] Dial("SIP/6001-00000000", "DAHDI/g1/09396464991") in new stack [Mar 10 13:40:04] WARNING[2106]: channel.c:5627 ast_request: No channel type registered for 'DAHDI'
**[Mar 10 13:40:04] WARNING[2106]: app_dial.c:2274 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented)== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [1-CHANUNAVAIL#macro-trunkdial-failover-0.3:2] Hangup("SIP/6001-00000000", "") in new stack**== Spawn extension (macro-trunkdial-failover-0.3, 1-CHANUNAVAIL, 2) exited non-zero on 'SIP/6001-00000000' in macro 'trunkdial-failover-0.3'== Spawn extension (DLPN_Main, 09396464991, 1) exited non-zero on 'SIP/6001-00
1) First check your dahdi config is present and it is ok. Use following to check it
dahdi_cfg -vvvv
2) check that your asterisk have pri/dahdi support.
A user recently notified me that whenever they attempt to dial into a conference call at another company, the phone call would get dropped after 5 seconds or so. They also indicated that when the same number is called using a cell phone, there were no issues. I found the following entries in log file.
[May 4 11:58:20] VERBOSE[24063] app_dial.c: -- DAHDI/1-1 is ringing
[May 4 11:58:20] VERBOSE[24063] app_dial.c: -- DAHDI/1-1 answered SIP/145-00000005
[May 4 11:58:24] WARNING[24063] rtp.c: Don't know how to represent 'f'
[May 4 11:58:24] VERBOSE[24063] chan_dahdi.c: -- Redirecting DAHDI/1-1 to fax extension
[May 4 11:58:24] VERBOSE[24063] pbx.c: -- Executing [h#macro-dialout-trunk:1] Macro("SIP/145-00000005", "hangupcall,") in new stack
[May 4 11:58:24] VERBOSE[24063] pbx.c: -- Executing [s#macro-hangupcall:1] GotoIf("SIP/145-00000005", "1?theend") in new stack
I have not been able to determine a solution. Any insight or suggestions on solving this problem are appreciated.
(Using FreePBX v2.9; Asterisk v1.6.2.15.1; CentOS 5.5 (Final); Sangoma A102)
Try add into file
/etc/asterisk/sip_general_custom.conf
faxdetect=no
Also tried modiying chan_dahdi.conf, but that did not work.
Final solution was to modify these settings (changing from YES to NO) in /etc/wanrouter/wanpipe1.conf
TDMV_HW_DTMF = NO # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT = NO # YES: receive fax 1100hz events from hardware