Incoming Calls Dropping

Hi All,
Would anybody be able to tell me why the incoming calls are dropping? Here is a the info from my CDR.
61379541|81945714| mycontext| “6137954171”| <6137954171> |SIP/148374-000000da |Hangup| 04/12/2012 20:42|04/12/2012 20:42|04/12/2012 20:42|1| 1| ANSWERED|DOCUMENTATION| 1354653724

Because the dialplan called the Hnagup application! Note that you appear to be missing the Data field, from after the Application field.

If you think it shouldn’t be calling the Hangup application, you are going to have to provide at least a verbose level 3 CLI trace, and maybe more. CDRs are not designed as a debugging aid.

[quote=“david55”]Because the dialplan called the Hnagup application! Note that you appear to be missing the Data field, from after the Application field.

If you think it shouldn’t be calling the Hangup application, you are going to have to provide at least a verbose level 3 CLI trace, and maybe more. CDRs are not designed as a debugging aid.[/quote]

Hi David,
It seems like I get an error 401 when placing an inbound call
Here is a part of the debug file let me know if you would need more.

<------------->
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 0 [ 46]: ACK sip:@206.248.x.y:5060 SIP/2.0
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 1 [ 64]: Via: SIP/2.0/UDP 67.205.x.y:5060;branch=z9hG4bK54e0e365;rport
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 2 [ 64]: From: “” <sip:@67.205.x.y>;tag=as57517af4
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 3 [ 55]: To: <sip:@206.248.x.y:5060>;tag=as666a1048
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 4 [ 39]: Contact: <sip:@67.205.x.y>
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 5 [ 55]: Call-ID: 67eb8c6d5d901f72558032fc77f7f393@67.205.x.y
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 6 [ 13]: CSeq: 102 ACK
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 7 [ 25]: User-Agent: VoIPMS/SERAST
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 8 [ 16]: Max-Forwards: 70
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 9 [ 82]: Remote-Party-ID: “” <sip:@67.205.x.y>;privacy=off;screen=no
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 10 [ 17]: Content-Length: 0
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Header 11 [ 0]:
[Dec 4 21:38:44] VERBOSE[26208] chan_sip.c: — (11 headers 0 lines) —
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15041
[Dec 4 21:38:44] DEBUG[26208] chan_sip.c: Stopping retransmission on '67eb8c6d5d901f72558032fc77f7f393@67.205.x.y’ of Response 102: Match Found
[Dec 4 21:38:44] VERBOSE[26208] chan_sip.c:
<— SIP read from UDP:67.205.x.y:5060 —>
INVITE sip:@206.248.x.y:5060 SIP/2.0^M
Via: SIP/2.0/UDP 67.205.x.y:5060;branch=z9hG4bK04be5e63;rport^M
From: “” <sip:@67.205.x.y>;tag=as57517af4^M
To: <sip:@206.248.x.y:5060>^M
Contact: <sip:@67.205.x.y>^M
Call-ID: 67eb8c6d5d901f72558032fc77f7f393@67.205.x.y^M
CSeq: 103 INVITE^M
User-Agent: VoIPMS/SERAST^M
Max-Forwards: 70^M
Remote-Party-ID: “” <sip:@67.205.x.y>;privacy=off;screen=no^M
Authorization: Digest username=“myUserName”, realm=“NAS”, algorithm=MD5, uri=“sip:@206.248.x.y:5060”, nonce=“08734a0b”, response=“d47194427d08caf9aaa1b2e9cdf3afea”^M
Date: Wed, 05 Dec 2012 02:35:15 GMT^M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO^M
Supported: replaces^M
Content-Type: application/sdp^M
Content-Length: 287^M
^M
v=0^M
o=root 6552 6553 IN IP4 67.205.x.y^M
s=session^M
c=IN IP4 67.205.x.y^M
t=0 0^M
m=audio 15130 RTP/AVP 0 18 101^M
a=rtpmap:0 PCMU/8000^M
a=rtpmap:18 G729/8000^M
a=fmtp:18 annexb=no^M
a=rtpmap:101 telephone-event/8000^M
a=fmtp:101 0-16^M
a=silenceSupp:off - - - -^M
a=ptime:20^M
a=sendrecv^M

Here is what I have in the SIP.conf
[color=#0040FF][general]
register => 148zzz:Password@montreal.voip.ms:5060

[voipms]
canreinvite = no
context = mycontext
host = montreal.voip.ms
secret = Password
type = peer
username = 148ZZZ
disallow = all
allow = ulaw
fromuser = 148ZZZ
trustrpid = yes
sendrpid = yes
insecure = invite
nat = yes[/color]

Here is the extensions.conf

[color=#0000FF][mycontext]
include => voipms-inboud
include => voipms-outbound

[voipms-inbound]
;exten => DID,1,Answer()
exten = s,1,Goto(Default,6000,1)[/color]

I tried both of the configs in [voipms-inboud] and they both fail.

You don’t have anything in your Default context (this must be becaue you haven’t provided the complete extensions.conf). For safety you should treat context names as case sensitive, although I am not sure if they are.

You say you got a 401 status, but you didn’t start the trace until after it!

Getting a 401 status is normal, if you don’t have insecure=invite, so I suspect you are not matching the sip.conf entry.

On the other hand, you must have accepted the call for it to have been explicitly hung up.

You have cut off the part of the trace that shows which sip.conf entry was matched and you have not included the verbose 3 output.

Generally people will ignore requests for help in private messages.

If the log is very long, it probably means you are using a GUI, in which case you need the board for that GUI.

Long quotes can be done as code blocks, inline, or as attachments. Code blocks are easier for me.

[quote=“david55”]Generally people will ignore requests for help in private messages.

If the log is very long, it probably means you are using a GUI, in which case you need the board for that GUI.

Long quotes can be done as code blocks, inline, or as attachments. Code blocks are easier for me.[/quote]
This is all new to me. Sorry but I’m very green when it comes to Asterisk. I’ve done many configs with Cisco in the past. What would be the process that you mentioned above? Right now I putty into my NAS where Asterisk is installed and just drill down to where the MyDebugLog is sitting then just go into a regular vi to view it. I can do an output to a txt file from Putty.
My origianl debug was set to verbose 15 and debug 15 but I brought it down to 3 like you suggested. The file is still big (about 2000 lines) So what are you suggesting when you say code blocks?
Asterisk does have a GUI but I don’t like using it. I’m more of a CLI guy.

As for the “Default” context, you are right when you said there is no substance to it. But I also tried the exten => DID,1,Answer() (DID being my phone#) and still didn’t work.

Please advise on how you would like to proceed with viewing the debug log and also if possible to include steps since i’m so green in this process :blush:

[quote=“david55”]You don’t have anything in your Default context (this must be becaue you haven’t provided the complete extensions.conf). For safety you should treat context names as case sensitive, although I am not sure if they are.

You say you got a 401 status, but you didn’t start the trace until after it!

Getting a 401 status is normal, if you don’t have insecure=invite, so I suspect you are not matching the sip.conf entry.

On the other hand, you must have accepted the call for it to have been explicitly hung up.

You have cut off the part of the trace that shows which sip.conf entry was matched and you have not included the verbose 3 output.[/quote]
Here is the extensions.conf you asked.
[globals]
CONSOLE = Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO = guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK = DAHDI/G2 ; Trunk interface

TRUNKMSD = 1 ; MSD digits to strip (usually 1 or 0)
FEATURES =
DIALOPTIONS =
RINGTIME = 20
FOLLOWMEOPTIONS =
PAGING_HEADER = Intercom
PAGING_TIMEOUT = 60
123456 = SIP/123456
CID_6000 = 1234567891

[DID_123456]
include = DID_123456_default
include = DID_123456_timeinterval_24-7|${timeinterval_24-7}
include = DID_123456_default

[DID_123456_default]
exten=s,1,Answer

[CallingRule_Everything]
exten = _.,1,Macro(trunkdial-failover-0.3,${123456}/${EXTEN:0},123456,)

[DLPN_Everything]
include = CallingRule_Everything
include = default
include = voicemenus
include = voicemailgroups

[DID_123456_timeinterval_24-7]

[default]
exten = s,1,Goto(default,6000,1)
exten=s,1,Answer
exten=#6XXX,1,Set(MBOX=${EXTEN:1}@default)
exten=
#6XXX,n,VoiceMail(${MBOX})
exten=a,1,VoicemailMain(${MBOX})
exten=6100,1,VoiceMailMain(${CALLERID(num)}@default)

The CDR shows that the call is ending in “mycontext”. There is no mycontext in the file you provided. The CDR shows it is terminating by explicitly executing the Hangup application. There is no Hangup application call in the file you provided.

The file you provided is syntactically invalid. Many of the “=” should be “=>”. “|” isn’t used as a delimiter for time dependent includes. I’d be surprised if variables were substituted for these, but, in any case, the variable appears to be undefined.

The file you provided is syntactically invalid. Many of the “=” should be “=>”. “|” isn’t used as a delimiter for time dependent includes. I’d be surprised if variables were substituted for these, but, in any case, the variable appears to be undefined.[/quote]

[quote=“david55”]The CDR shows that the call is ending in “mycontext”. There is no mycontext in the file you provided. The CDR shows it is terminating by explicitly executing the Hangup application. There is no Hangup application call in the file you provided.

The file you provided is syntactically invalid. Many of the “=” should be “=>”. “|” isn’t used as a delimiter for time dependent includes. I’d be surprised if variables were substituted for these, but, in any case, the variable appears to be undefined.[/quote]

I see there is a discrepancy between the actual config files that I edit in vi and the GUI. What I posted was different than what the actual config file in vi was saying under the CLI. The the config files all have the “=>” and also the “mycontext” is there when opening them in a vi editor. What I posted was from the GUI as it was easier. Sorry for the false information. but I just found out that the Synology NAS developpers published a pkg that had an incompatible GUI with 1.8.13 Asterisk.
I ended up blowing up the config yesterday night and restarting from scratch as to many bugs from the GUI. Here is the actual config for both extensions.conf and sip.conf that i’m using.
Before blowing away the configuration I used to get active channels for incoming calls going to the PBX but now I don’t get anything. I will refrain from making any changes until I hear from you. Thanks in advance. Does the extensions.conf over-wright what is in the users.conf?

sip.conf
[general]
register => “MyUserID”:“MyPassword”@montreal.voip.ms:5060
[voipms]
canreinvite=no
context=mycontext
host=montreal.voip.ms
secret=“MyPassword”;your password
type=peer
username=“MyUserID” ;your account
disallow=all
allow=ulaw
fromuser="MyUserID"
trustrpid=yes
sendrpid=yes
insecure=invite
nat=yes

extensions.conf
[general]
static = yes
writeprotect = no
clearglobalvars = yes

[globals]
CONSOLE = Console/dsp ; Console interface for demo
;CONSOLE=DAHDI/1
;CONSOLE=Phone/phone0
IAXINFO = guest ; IAXtel username/password
;IAXINFO=myuser:mypass
TRUNK = DAHDI/G2 ; Trunk interface
FEATURES =
DIALOPTIONS =
RINGTIME = 20
FOLLOWMEOPTIONS =
PAGING_HEADER = Intercom
PAGING_TIMEOUT = 60
12345 = SIP/12345
CID_6000 = 1234567891
timeinterval_24-7 = |mon-sun||*

[mycontext]
include => voipms-inbound
include => voipms-outbound

[voipms-outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@voipms)
exten => _1NXXNXXXXXX,n,Hangup()
exten => _NXXNXXXXXX,1,Dial(SIP/1${EXTEN}@voipms)
exten => _NXXNXXXXXX,n,Hangup()
exten => _011.,1,Dial(SIP/${EXTEN}@voipms)
exten => _011.,n,Hangup()
exten => _00.,1,Dial(SIP/${EXTEN}@voipms)
exten => _00.,n,Hangup()

[voipms-inbound]
exten => CID_6000,1,Answer()

more on this.
Since the GUI is tied to Asterisk, I have no choice to create Dialplans in the GUI and tie them with the User/extension. this is a bit frustrating to know that the GUI is the issue but I can’t bypass it.

Problems with GUI dialplans really need tobe directed to forums for those GUIs. Even the AsteriskNow forums here, don’t actually cover FreePBX issues that are not specific to AsteriskNow.