Several wierd ones


(Dan S) #1

I have one extension that gets a sip 488 on every call Not sure why but can call inbound to that

pjsip debug info
INVITE sip:32897@10.123.245.18;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.119.101.228:5060;branch=z9hG4bK-524287-1—72ed8fb0ef15d299;rport
Max-Forwards: 70
Contact: <sip:32481@10.119.101.228:5060;transport=UDP>
To: <sip:32897@10.123.245.18>
From: <sip:32481@10.123.245.18;transport=UDP>;tag=6005011c
Call-ID: bXaGzv6zi6rdrCiq5Yhz5g…
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.4.6 rv2.10.10.2
Allow-Events: presence, kpml, talk
Content-Length: 227v=0
o=Z 1600112593446 1 IN IP4 10.119.101.228
s=Z
c=IN IP4 10.119.101.228
t=0 0
m=audio 20000 RTP/AVP 18 101 0 8
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv<— Transmitting SIP response (512 bytes) to UDP:10.119.101.228:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.119.101.228:5060;rport=5060;received=10.119.101.228;branch=z9hG4bK-524287-1—72ed8fb0ef15d299
Call-ID: bXaGzv6zi6rdrCiq5Yhz5g…
From: <sip:32481@10.123.245.18>;tag=6005011c
To: <sip:32897@10.123.245.18>;tag=z9hG4bK-524287-1—72ed8fb0ef15d299
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1600112593/c8ae2fc76df0137b0f927842f56bf0c4”,opaque=“7c4f221a0c4ae2ab”,algorithm=md5,qop=“auth”
Server: FPBX-15.0.16.73(16.12.0)
Content-Length: 0<— Received SIP request (356 bytes) from UDP:10.119.101.228:5060 —>
ACK sip:32897@10.123.245.18;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.119.101.228:5060;branch=z9hG4bK-524287-1—72ed8fb0ef15d299;rport
Max-Forwards: 70
To: <sip:32897@10.123.245.18>;tag=z9hG4bK-524287-1—72ed8fb0ef15d299
From: <sip:32481@10.123.245.18;transport=UDP>;tag=6005011c
Call-ID: bXaGzv6zi6rdrCiq5Yhz5g…
CSeq: 1 ACK
Content-Length: 0<— Received SIP request (1090 bytes) from UDP:10.119.101.228:5060 —>
INVITE sip:32897@10.123.245.18;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.119.101.228:5060;branch=z9hG4bK-524287-1—113634197e59905a;rport
Max-Forwards: 70
Contact: <sip:32481@10.119.101.228:5060;transport=UDP>
To: <sip:32897@10.123.245.18>
From: <sip:32481@10.123.245.18;transport=UDP>;tag=6005011c
Call-ID: bXaGzv6zi6rdrCiq5Yhz5g…
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.4.6 rv2.10.10.2
Authorization: Digest username=“32481”,realm=“asterisk”,nonce=“1600112593/c8ae2fc76df0137b0f927842f56bf0c4”,uri=“sip:32897@10.123.245.18;transport=UDP”,response=“e0032f69f1723c85eccadf7bb9834397”,cnonce=“611e5946972284a608bc3728932daf57”,nc=00000001,qop=auth,algorithm=md5,opaque=“7c4f221a0c4ae2ab”
Allow-Events: presence, kpml, talk
Content-Length: 227v=0
o=Z 1600112593446 1 IN IP4 10.119.101.228
s=Z
c=IN IP4 10.119.101.228
t=0 0
m=audio 20000 RTP/AVP 18 101 0 8
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv == Setting global variable ‘SIPDOMAIN’ to ‘10.123.245.18’
<— Transmitting SIP response (320 bytes) to UDP:10.119.101.228:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.119.101.228:5060;rport=5060;received=10.119.101.228;branch=z9hG4bK-524287-1—113634197e59905a
Call-ID: bXaGzv6zi6rdrCiq5Yhz5g…
From: <sip:32481@10.123.245.18>;tag=6005011c
To: <sip:32897@10.123.245.18>
CSeq: 2 INVITE
Server: FPBX-15.0.16.73(16.12.0)
Content-Length: 0<— Transmitting SIP response (374 bytes) to UDP:10.119.101.228:5060 —>
SIP/2.0 488 Not Acceptable Here

I have 2 extensions that can call out but don’t receive calls. In the console I see the pjsip ringing that extensions. I haven’t’ got a debug captures.

I have one other that on some number of calls, they get a busy signal to there DID> Not all the time but like 40% of the time.

Any help would be great. I’m underwater with conversion to freepbx from bare asterisk.


#2

488 indicates either you’ve got codecs that don’t match up (unlikely since you are offering PCMU and PCMA which are generally accepted everywhere) or you are offering a call without encryption but the other side requires it.


(Dan S) #3

And it works for everyone else. so unlikely and the client is Zoiper 5 and i’m using an encrypted profile. so both of the commons ones are not on the list. And the profile is using Auto provisioning so all the users get the same zoiper config so no human way to screw that up on one device. I would screw it up on a few hundred but not one.


#4

Check your extension settings in FreePBX itself.


(Dan S) #5

Thats the first one. so now I have the 2 that dont’ get incoming calls.

Whats odd is i have pjsip set logger host on the IP of the phone and I see the registration but nothing for a phone call in


(Dan S) #6

<— Transmitting SIP request (648 bytes) to UDP:10.119.101.242:5060 —>
NOTIFY sip:32961@10.119.101.242:5060 SIP/2.0
Via: SIP/2.0/UDP 10.123.245.18:5060;rport;branch=z9hG4bKPjde38f00e-018b-45a6-a3e6-d3440f4b7a15
From: sip:32961@10.123.245.18;tag=046a0f81-4c36-4d1e-9fc7-daec4039658e
To: sip:32961@10.123.245.18;tag=802b4368
Contact: sip:10.123.245.18:5060
Call-ID: V8lR134sY4ExyrdcoEohIQ…
CSeq: 29151 NOTIFY
Event: message-summary
Subscription-State: active;expires=59
Allow-Events: message-summary, presence, dialog, refer
Max-Forwards: 70
User-Agent: FPBX-15.0.16.73(16.12.0)
Content-Type: application/simple-message-summary
Content-Length: 48

Messages-Waiting: no
Voice-Message: 0/0 (0/0)

<— Received SIP response (400 bytes) from UDP:10.119.101.242:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.123.245.18:5060;rport=5060;branch=z9hG4bKPjde38f00e-018b-45a6-a3e6-d3440f4b7a15
Contact: sip:32961@10.119.101.242:5060;transport=UDP
To: sip:32961@10.123.245.18;tag=802b4368
From: sip:32961@10.123.245.18;tag=046a0f81-4c36-4d1e-9fc7-daec4039658e
Call-ID: V8lR134sY4ExyrdcoEohIQ…
CSeq: 29151 NOTIFY
User-Agent: Z 5.4.6 rv2.10.10.2
Content-Length: 0

So the above keeps coming as it should but I’m not seeing the phone call traffic other than the normal stuff in the console.

Like I have this stuff
– Executing [s@macro-dial-one:52] ExecIf(“PJSIP/opmgw23-00003283”, “0?Set(CWRING=r(callwaiting)):Set(CWRING=)”) in new stack
– Executing [s@macro-dial-one:53] NoOp(“PJSIP/opmgw23-00003283”, “”) in new stack
– Executing [s@macro-dial-one:54] ExecIf(“PJSIP/opmgw23-00003283”, “0?Set(D_OPTIONS=HhtrIg)”) in new stack
– Executing [s@macro-dial-one:55] Dial(“PJSIP/opmgw23-00003283”, “pjsip/32961,15,HhtrIb(func-apply-sipheaders^s^1)”) in new stack
– PJSIP/32961-00003284 Internal Gosub(func-apply-sipheaders,s,1) start
– Executing [s@func-apply-sipheaders:1] NoOp(“PJSIP/32961-00003284”, “Applying SIP Headers to channel PJSIP/32961-00003284”) in new stack
– Executing [s@func-apply-sipheaders:2] Set(“PJSIP/32961-00003284”, “TECH=PJSIP”) in new stack
– Executing [s@func-apply-sipheaders:3] Set(“PJSIP/32961-00003284”, “SIPHEADERKEYS=”) in new stack
– Executing [s@func-apply-sipheaders:4] While(“PJSIP/32961-00003284”, “0”) in new stack
– Jumping to priority 10
– Executing [s@func-apply-sipheaders:11] Return(“PJSIP/32961-00003284”, “”) in new stack
== Spawn extension (from-internal, 32961, 1) exited non-zero on ‘PJSIP/32961-00003284’
– PJSIP/32961-00003284 Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
– Called pjsip/32961
== Using SIP RTP Audio TOS bits 184
== Using SIP RTP Audio CoS mark 5
– Connected line update to PJSIP/opmgw23-00003283 prevented.
– PJSIP/32961-00003284 is ringing
– PJSIP/32961-00003284 is ringing
== Spawn extension (macro-dial-one, s, 55) exited non-zero on ‘PJSIP/opmgw23-00003283’ in macro ‘dial-one’
== Spawn extension (macro-exten-vm, s, 14) exited non-zero on ‘PJSIP/opmgw23-00003283’ in macro ‘exten-vm’
== Spawn extension (ext-local, 32961, 3)

but I’m not seeing the pjsip logger stuff I was expecting to see.


(Dan S) #7

To address the last 2 I deleted the extensions and re added. So this one is closed.


(Tom Ray) #8

I’m pretty sure offering up g729 without it being licensed or active on FreePBX caused it to throw the 488 error, which is fully expected in that case.