486 - Busy Here!

Asterisk Asterisk Support
1

Hello,

I have a weird problem and was hoping someone from the asterisk community would know what i’m talking about!

I have an asterisk box that is hooked to a PRI. Multiple DID are configured. Everything works perfectly, almost!

You can call DIDs withtout any problems. But sometimes, for no apparent reasons, when you call a DID you get a busy signal and shouldn’t (The person is not on the phone, the PRI is not full, channels are not blocked…).

In CLI, you see;
– Executing Dial(“Zap/42-1”, “SIP/405|||”) in new stack

– Called 405
– Got SIP response 486 “Busy Here” back from 172.16.2.30
– SIP/405-b266 is busy
== Everyone is busy/congested at this time (1:1/0/0)

That 486 Busy Here message is weird. I’ve talk to several persons about this issue, went into asterisk meetings, discussion forums, etc… No one is able to tell why I have this error message.

I have several differents sites, and it’s doing the same thing for all of them!

Any help would be appreciated… if you ever saw this kind or problem, would be nice to know! (I feel pretty alone!)

2

Sorry I have not gotten into the PRI line w/ DID just yet… I am about to do just such a setup 1Q 2007… I am assuming here based on your post that you are using SIP phones dialed from incoming lines. Can you do a packet capture on the SIP conversation to see if it is the phone sending the 486 message? It seems from what you have posted that the phone itself is rejecting the call and that it has nothing to do with Asterisk per-se. Can you recreate the situation with internal extension to extension calls?

Just some random thoughs… -Cheers, Peter.

3

[quote=“pbrunnen”]Sorry I have not gotten into the PRI line w/ DID just yet… I am about to do just such a setup 1Q 2007… I am assuming here based on your post that you are using SIP phones dialed from incoming lines. Can you do a packet capture on the SIP conversation to see if it is the phone sending the 486 message? It seems from what you have posted that the phone itself is rejecting the call and that it has nothing to do with Asterisk per-se. Can you recreate the situation with internal extension to extension calls?

Just some random thoughs… -Cheers, Peter.[/quote]

Hi Peter!

I’m using analog phones that connects to a Gateway (mediatrix 1124). All my gateways are connected to a switch that is connected to my Asterisk server.

It’s really difficult to ‘recreate’ the problem, because it’s on and off… Most of the time it’s ok… What I did in order to get that debug, I started CLI and logged everything… Every half-an-hour I looked at at log, when I saw a 486 error message, than I tried to dial this number. If the problem was still ongoing (486-Busy Here), than I Sip Debug…

Here’s a Sip Debug of the problem:

INVITE sip:405@172.16.2.30:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.2.4:5060;branch=z9hG4bK4c326231;rport From: “4509011020” sip:4509011020@172.16.2.4;tag=as73c4d99b To: sip:405@172.16.2.30:5060 Contact: sip:4509011020@172.16.2.4 Call-ID: 5d4e54285cd6671864a10aed4fa73890@172.16.2.4 CSeq: 102 INVITE User-Agent: Connectit Networks Max-Forwards: 70 Remote-Party-ID: “4509011020” sip:4509011020@172.16.2.4;privacy=off;screen=yes Date: Thu, 02 Nov 2006 16:27:41 GMT Alert-Info: http://127.0.0.1/Bellcore-dr1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 212 v=0 o=root 31058 31058 IN IP4 172.16.2.4 s=session c=IN IP4 172.16.2.4 t=0 0 m=audio 17088 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off
--------- Called 405

localhost*CLI> <-- SIP read from 172.16.2.30:5060:
SIP/2.0 100 Trying Call-ID: 5d4e54285cd6671864a10aed4fa73890@172.16.2.4 CSeq: 102 INVITE From: “4509011020” sip:4509011020@172.16.2.4;tag=as73c4d99b To: sip:405@172.16.2.30:5060;tag=3c916a08e386798 Via: SIP/2.0/UDP 172.16.2.4:5060;branch=z9hG4bK4c326231;rport Content-Length: 0 User-Agent: MxSipApp/5.0.14.83 MxSF/v3.2.1.1

— (8 headers 0 lines)—

localhost*CLI> <-- SIP read from 172.16.2.30:5060:
SIP/2.0 486 Busy Here Call-ID: 5d4e54285cd6671864a10aed4fa73890@172.16.2.4 CSeq: 102 INVITE From: “4509011020” sip:4509011020@172.16.2.4;tag=as73c4d99b To: sip:405@172.16.2.30:5060;tag=3c916a08e386798 Via: SIP/2.0/UDP 172.16.2.4:5060;branch=z9hG4bK4c326231;rport Content-Length: 0 User-Agent: MxSipApp/5.0.14.83 MxSF/v3.2.1.1

— (8 headers 0 lines)—

localhost*CLI>
– Got SIP response 486 “Busy Here” back from 172.16.2.30
Transmitting (no NAT) to 172.16.2.30:5060:
ACK sip:405@172.16.2.30:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.2.4:5060;branch=z9hG4bK4c326231;rport From: “4509011020” sip:4509011020@172.16.2.4;tag=as73c4d99b To: sip:405@172.16.2.30:5060;tag=3c916a08e386798 Contact: sip:4509011020@172.16.2.4 Call-ID: 5d4e54285cd6671864a10aed4fa73890@172.16.2.4 CSeq: 102 ACK User-Agent: Connectit Networks Max-Forwards: 70 Remote-Party-ID: “4509011020” sip:4509011020@172.16.2.4;privacy=off;screen=yes Content-Length: 0
----- SIP/405-98b9 is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Executing Goto(“Zap/35-1”, “s-BUSY|1”) in new stack
– Goto (macro-CAMBRIDGE_INCOMING,s-BUSY,1)
– Executing DBget(“Zap/35-1”, “CFB_STATUS=CFB/405”) in new stack
– DBget: varname=CFB_STATUS, family=CFB, key=405
– DBget: Value not found in database.
– Executing GotoIf(“Zap/35-1”, “0?s-CFB|1:3”) in new stack
– Goto (macro-CAMBRIDGE_INCOMING,s-BUSY,3)
– Executing DBget(“Zap/35-1”, “VM_STATUS=VM/405”) in new stack
– DBget: varname=VM_STATUS, family=VM, key=405
– DBget: Value not found in database.
– Executing GotoIf(“Zap/35-1”, “0?6:5”) in new stack
– Goto (macro-CAMBRIDGE_INCOMING,s-BUSY,5)
– Executing Busy(“Zap/35-1”, “”) in new stack

… It’s weird. It can happen to any phones anytime. I’m starting to think that my gateways might be the cause… But how can I be sure?

4

Don,
Ah… ok… so asterisk is sending the call to the analog station gateway via SIP trunking… got it. Yea… I would concur that it seems to be with your gateway. In your SIP conversation, you see asterisk sending the SIP INVITE to your gateway. The gateway responds first with a TRYING and then a BUSY.

My first recommendation would be to go and look if there is a bug like this reported at the mediatrix manufacurer’s knowledge base.

Second would be to check if there is a firmware upgrade. It might be a software glitch in the gateway.

Last, I would suggest attempting to see if it is your analog phone lines. I am not sure how close your phones are to the gateway or what type of cabling you are using… But next time you can catch a 486 Busy reply, try to unplug (or short out) the analog phone and immediately retry the call. Going back to old school telco stuff here, I wonder if your lines are resetting after the loop opens (e.g. on-hook after a phone call).

Another thought that just poped into my head… check your line signalling on your gateway. This is something that I find gets messed up by softswitch eqipment pretty often. A station should be using a loop start signalling type. In asterisk with digium cards I found that you have to use kewl start rather than loop start, which does not make much sense to me… so this could be something similar.

Hope that helps. -Cheers, Peter.

5

[quote=“pbrunnen”]Don,
Ah… ok… so asterisk is sending the call to the analog station gateway via SIP trunking… got it. Yea… I would concur that it seems to be with your gateway. In your SIP conversation, you see asterisk sending the SIP INVITE to your gateway. The gateway responds first with a TRYING and then a BUSY.

My first recommendation would be to go and look if there is a bug like this reported at the mediatrix manufacurer’s knowledge base.

Second would be to check if there is a firmware upgrade. It might be a software glitch in the gateway.

Last, I would suggest attempting to see if it is your analog phone lines. I am not sure how close your phones are to the gateway or what type of cabling you are using… But next time you can catch a 486 Busy reply, try to unplug (or short out) the analog phone and immediately retry the call. Going back to old school telco stuff here, I wonder if your lines are resetting after the loop opens (e.g. on-hook after a phone call).

Another thought that just poped into my head… check your line signalling on your gateway. This is something that I find gets messed up by softswitch eqipment pretty often. A station should be using a loop start signalling type. In asterisk with digium cards I found that you have to use kewl start rather than loop start, which does not make much sense to me… so this could be something similar.

Hope that helps. -Cheers, Peter.[/quote]

Thanks pburnnen!
As per your message, I called Mediatrix in order to have more info on that matter… We’re doing some ongoing test as we speak…

But in the meantime, I configured a second server, so that some of the calls go through that second server and then to destination. So I had to register the second server as a peer. Everything worked fine. But then, all of a sudden, I was testing that server one more time, and I see that 486 Busy Here error message in CLI! … That’s strange, it means that every registered peers might give me this error… it also means that the problem doesn’t reside in my gateways :frowning:

… Here, I post my CLI result as well as my debug just in case someone would know…

CLI:
localhostCLI>
– Accepting call from ‘’ to ‘7821627’ on channel 0/2, span 2
localhostCLI>
– Executing NoCDR(“Zap/26-1”, “”) in new stack
localhostCLI>
Nov 15 18:07:35 WARNING[1135]: cdr.c:443 ast_cdr_free: CDR on channel ‘Zap/26-1’ not posted
ocalhostCLI>
– Executing Dial(“Zap/26-1”, “SIP/5147821627@172.16.2.5”) in new stack
localhostCLI>
– Called 5147821627@172.16.2.5
localhostCLI>
– Got SIP response 486 “Busy Here” back from 172.16.2.5
localhostCLI>
– SIP/172.16.2.5-22c2 is busy
localhostCLI>
== Everyone is busy/congested at this time (1:1/0/0)
localhostCLI>
== Spawn extension (macro-CAMBRIDGE_INCOMING, s, 64) exited non-zero on ‘Zap/30-1’ in macro 'CAMBRIDGE_INCOMING’
localhostCLI>
– Hungup ‘Zap/30-1’

SIP DEBUG:

localhostCLI> sip debug peer remote-site
localhostCLI> SIP Debugging Enabled for IP: 172.16.2.5:5060

localhostCLI> – Accepting call from ‘’ to ‘7821627’ on channel 0/21, span 2
localhostCLI> – Executing NoCDR(“Zap/45-1”, “”) in new stack
localhostCLI> Nov 15 17:29:40 WARNING[19428]: cdr.c:443 ast_cdr_free: CDR on channel ‘Zap/45-1’ not posted
localhostCLI> Nov 15 17:29:40 WARNING[19428]: cdr.c:445 ast_cdr_free: CDR on channel ‘Zap/45-1’ lacks end
localhostCLI> – Executing Dial(“Zap/45-1”, “SIP/5147821627@172.16.2.5”) in new stack
localhostCLI> We’re at 172.16.2.4 port 10744
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 10 lines
Reliably Transmitting (no NAT) to 172.16.2.5:5060:
INVITE sip:5147821627@172.16.2.5 SIP/2.0
Via: SIP/2.0/UDP 172.16.2.4:5060;branch=z9hG4bK6ba6da32;rport
From: “asterisk” sip:asterisk@172.16.2.4;tag=as3ffaaf2f
To: sip:5147821627@172.16.2.5
Contact: sip:asterisk@172.16.2.4
Call-ID: 050ecd7120295cca31a693623f97ad6e@172.16.2.4
CSeq: 102 INVITE
User-Agent: Connect
Max-Forwards: 70

Remote-Party-ID: “asterisk” sip:asterisk@172.16.2.4;privacy=full;screen=pass
Date: Wed, 15 Nov 2006 22:29:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 212

v=0
o=root 31058 31058 IN IP4 172.16.2.4
s=session
c=IN IP4 172.16.2.4
t=0 0
m=audio 10744 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

----- Called 5147821627@172.16.2.5
localhost*CLI>

<-- SIP read from 172.16.2.5:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.2.4:5060;branch=z9hG4bK6ba6da32;received=172.16.2.4;rport=5060
From: “asterisk” sip:asterisk@172.16.2.4;tag=as3ffaaf2f
To: sip:5147821627@172.16.2.5
Call-ID: 050ecd7120295cca31a693623f97ad6e@172.16.2.4
CSeq: 102 INVITE
User-Agent: Quintum/1.0.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:5147821627@172.16.2.5
Content-Length: 0

— (10 headers 0 lines)—
localhost*CLI>

<-- SIP read from 172.16.2.5:5060:
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 172.16.2.4:5060;branch=z9hG4bK6ba6da32;received=172.16.2.4;rport=5060
From: “asterisk” sip:asterisk@172.16.2.4;tag=as3ffaaf2f
To: sip:5147821627@172.16.2.5;tag=as670846f4
Call-ID: 050ecd7120295cca31a693623f97ad6e@172.16.2.4
CSeq: 102 INVITE
User-Agent: Quintum/1.0.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:5147821627@172.16.2.5
Content-Length: 0
X-Asterisk-HangupCause: No route to destination

— (11 headers 0 lines)—
localhost*CLI>
– Got SIP response 486 “Busy Here” back from 172.16.2.5
Transmitting (no NAT) to 172.16.2.5:5060:
ACK sip:5147821627@172.16.2.5 SIP/2.0

Via: SIP/2.0/UDP 172.16.2.4:5060;branch=z9hG4bK6ba6da32;rport
From: “asterisk” sip:asterisk@172.16.2.4;tag=as3ffaaf2f
To: sip:5147821627@172.16.2.5;tag=as670846f4
Contact: sip:asterisk@172.16.2.4
Call-ID: 050ecd7120295cca31a693623f97ad6e@172.16.2.4
CSeq: 102 ACK
User-Agent: Connect
Max-Forwards: 70
Remote-Party-ID: “asterisk” sip:asterisk@172.16.2.4;privacy=full;screen=pass
Content-Length: 0

----- SIP/172.16.2.5-1c07 is busy
Destroying call ‘050ecd7120295cca31a693623f97ad6e@172.16.2.4’
== Everyone is busy/congested at this time (1:1/0/0)
localhostCLI>
– Channel 0/21, span 2 got hangup request
localhostCLI>
– Hungup ‘Zap/45-1’

It’s very strange… sometimes it’s all good, and some other times I have this error message. I asked to so many person about this problem… it seems like I’m the only person with that issue! … And what’s make it even strangier is that I have the same problem in all of my buildings!

Anyways, if someone has a clue/suggestion/hint… anyting! … I’m all ears…

Thanks alot boys!

6

Hi

Couple of ideas, check the rtp.conf and make sure its a big range and nothing else is using them

Make sure the localnet is setup correct in the sip.conf, also post a copy of the sip.conf.
Its odd that you have the problem at all your sites, This could point to a misconfig.
set the peers to qualify = yes

Ian

7

[quote=“ianplain”]Hi

Couple of ideas, check the rtp.conf and make sure its a big range and nothing else is using them

Make sure the localnet is setup correct in the sip.conf, also post a copy of the sip.conf.
Its odd that you have the problem at all your sites, This could point to a misconfig.
set the peers to qualify = yes

Ian[/quote]

Hi Ian! Thanks for you ideas!

Hummm… I’ve never modify that rtp.conf file, so it’s still the default in there. How can I make sure that nothing else is using the same RTP range? …

Here’s the default value (I don’t understand why in comment it says different values for the default since I never touch this file before!):

; RTP Configuration
;
[general]
;
; RTP start and RTP end configure start and end addresses
;
; Defaults are rtpstart=5000 and rtpend=31000
;
rtpstart=10000
rtpend=20000

Can someone tell me what’s their values in that rtp.conf file?

I’ve look in my sip.conf, I have no option that refer to Localnet. But isn’t that option required only if you’re behind a NAT? To my understanding, localnet is used to tell Asterisk which IP addresses are considered local so that the address in the SIP Header can be translated to that specified by externip… But if you think that could help, I will definitly put that config in my sip.conf and test it out…

Here’s my general section of my sip.conf

[general]
context=default
allowguest=no
realm=connect
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
tos=184
maxexpiry=3600
defaultexpiry=120
checkmwi=10
vmexten=*98
videosupport=no
recordhistory=no
disallow=all
allow=ulaw
musicclass=default
language=en
relaxdtmf=no
rtptimeout=60
rtpholdtimeout=300
trustrpid = no
sendrpid = yes
progressinband=never
useragent=Connect
promiscredir = no
usereqphone = yes
dtmfmode = rfc2833
compactheaders = no
sipdebug = no
subscribecontext = admin_dp
notifyringing = yes
;

More I think of it, more I agree with you Ian, it’s most probably a misconfig somewhere…