running Asterisk 13.17.0. My sip trunk through IPComms is working for incoming calls, but outgoing calls are giving “all circuits busy now”. There are way too many lines in the logs to post here, so some advise on just what I should be looking for would be helpful. I have the same username/password on the trunk for incoming and outgoing, and they both show registered.
I have several log segments similar to this:
[2017-08-05 12:36:00] DEBUG[64857] manager.c: Examining AMI event:
Event: Newexten
Privilege: call,all
Channel: SIP/1001-00000024
ChannelState: 4
ChannelStateDesc: Ring
CallerIDNum: 8566811126
CallerIDName:
ConnectedLineNum: 19188500508
ConnectedLineName: CID:8566811126
Language: en
AccountCode:
Context: macro-dialout-trunk
Exten: s
Priority: 30
Uniqueid: 1501954560.36
Linkedid: 1501954560.36
Extension: s
Application: Dial
AppData: SIP/IPComms_Out/19188500508,300,T
Eventually, I see the IPComms channel show up:
[2017-08-05 12:36:00] DEBUG[64857] manager.c: Examining AMI event:
Event: Newchannel
Privilege: call,all
Channel: SIP/IPComms_Out-00000025
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
ConnectedLineNum:
ConnectedLineName:
Language: en
AccountCode:
Context: from-trunk
Exten: s
Priority: 1
Uniqueid: 1501954560.37
Linkedid: 1501954560.36
Finally it shows:
[2017-08-05 12:36:00] DEBUG[64857] manager.c: Examining AMI event:
Event: DialBegin
Privilege: call,all
Channel: SIP/1001-00000024
ChannelState: 4
ChannelStateDesc: Ring
CallerIDNum: 8566811126
CallerIDName:
ConnectedLineNum: 19188500508
ConnectedLineName: CID:8566811126
Language: en
AccountCode:
Context: macro-dialout-trunk
Exten: s
Priority: 30
Uniqueid: 1501954560.36
Linkedid: 1501954560.36
DestChannel: SIP/IPComms_Out-00000025
DestChannelState: 0
DestChannelStateDesc: Down
DestCallerIDNum: 919188500508
DestCallerIDName: CID:8566811126
DestConnectedLineNum: 8566811126
DestConnectedLineName:
DestLanguage: en
DestAccountCode:
DestContext: from-trunk
DestExten: 919188500508
DestPriority: 1
DestUniqueid: 1501954560.37
DestLinkedid: 1501954560.36
DialString: IPComms_Out/19188500508
This error shows up next:
[2017-08-05 12:36:01] ERROR[1339] tcptls.c: Unable to connect SIP socket to 64.154.41.158:5060: Connection refused
Further down:
[2017-08-05 12:36:32] WARNING[2581] chan_sip.c: Retransmission timeout reached on transmission [email protected]:5060 for seqno 102 (Critical Request) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32006ms with no response
[2017-08-05 12:36:32] WARNING[2581] chan_sip.c: Hanging up call [email protected]:5060 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
[2017-08-05 12:36:32] VERBOSE[1327][C-00000019] pbx.c: Executing [s@macro-dialout-trunk:31] NoOp(“SIP/1001-00000024”, “Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 18”) in new stack
[2017-08-05 12:36:32] VERBOSE[1327][C-00000019] pbx.c: Executing [s@macro-dialout-trunk:32] GotoIf(“SIP/1001-00000024”, “1?continue,1:s-CHANUNAVAIL,1”) in new stack
[2017-08-05 12:36:32] DEBUG[2521] devicestate.c: No provider found, checking channel drivers for SIP - IPComms_Out
The PEER details for the outgoing part of the trunk:
username=XXXXXXXXX
type=friend
secret=XXXXXXXX
host=64.154.41.158
context=from-trunk
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite
nat=yes
directmedia=no
trustrpid=yes
These are the EXACT same as I have set on the “incoming”, which seems to be working.
sip show peers shows them as:
Name/username Host Dyn Forcerport Comedia ACL Port Status
IPComms_In/XXXXXXXXX 64.154.41.158 Yes Yes 5060 Unmonitored
IPComms_Out/XXXXXXXXXX 64.154.41.158 Yes Yes 5060 Unmonitored