Congested/Busy After Hanging Up a Call

Anyone ever get this message after hanging up an outgoing/incoming call?

asterisk dial_exec: Unable to create channel of type ‘SIP’ Everyone is busy/congested at this time

Then the message plays “I’m sorry that is not a valid extension”

Any idea how to fix is?

A lot of times I get this message in my logs

Apr 2 11:21:45 NOTICE[1063]: chan_sip.c:6721 handle_response: Peer ‘PHONENUMBER’ is now TOO LAGGED!
Apr 2 11:21:55 NOTICE[1063]: chan_sip.c:6715 handle_response: Peer ‘PHONENUMBER’ is now REACHABLE!

I’ve the problem too.

When a called peer hangs up the call. Correct call.

[code] – Executing [420000@office:1] Dial(“SIP/201-0000019d”, “SIP/ttk/420000,T”) in new stack
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.31.1.34:5060:
INVITE sip:420000@172.31.1.34 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK039025f0
Max-Forwards: 70
From: “Test2” sip:910000@172.30.1.206;tag=as16da7a9c
To: sip:420000@172.31.1.34
Contact: sip:910000@172.30.1.206:5060
Call-ID: 1445c0af175505fd7766a1ff26395496@172.30.1.206:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.8.0
Date: Mon, 09 Jan 2012 13:50:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1059445238 1059445238 IN IP4 172.30.1.206
s=Asterisk PBX 1.8.8.0
c=IN IP4 172.30.1.206
t=0 0
m=audio 10902 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Called SIP/ttk/420000

<— SIP read from UDP:172.31.1.34:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK039025f0
Call-ID: 1445c0af175505fd7766a1ff26395496@172.30.1.206:5060
From: "Test2"sip:910000@172.30.1.206;tag=as16da7a9c
To: sip:420000@172.31.1.34
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:172.31.1.34:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK039025f0
Call-ID: 1445c0af175505fd7766a1ff26395496@172.30.1.206:5060
From: "Test2"sip:910000@172.30.1.206;tag=as16da7a9c
To: sip:420000@172.31.1.34;tag=9559f665
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: sip:420000@172.31.1.34:5060;user=phone
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 9874980 9874980 IN IP4 172.31.1.34
s=Sip Call
c=IN IP4 172.31.1.34
t=0 0
m=audio 32274 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (10 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.31.1.34:32274
– SIP/ttk-0000019e is ringing
– SIP/ttk-0000019e is making progress passing it to SIP/201-0000019d

<— SIP read from UDP:172.31.1.34:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK039025f0
Call-ID: 1445c0af175505fd7766a1ff26395496@172.30.1.206:5060
From: "Test2"sip:910000@172.30.1.206;tag=as16da7a9c
To: sip:420000@172.31.1.34;tag=9559f665
CSeq: 102 INVITE
Contact: sip:420000@172.31.1.34:5060;user=phone
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 9874980 9874981 IN IP4 172.31.1.34
s=Sip Call
c=IN IP4 172.31.1.34
t=0 0
m=audio 32274 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (10 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.31.1.34:32274
list_route: hop: sip:420000@172.31.1.34:5060;user=phone
set_destination: Parsing sip:420000@172.31.1.34:5060;user=phone for address/port to send to
set_destination: set destination to 172.31.1.34:5060
Transmitting (no NAT) to 172.31.1.34:5060:
ACK sip:420000@172.31.1.34:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK7e244841
Max-Forwards: 70
From: “Test2” sip:910000@172.30.1.206;tag=as16da7a9c
To: sip:420000@172.31.1.34;tag=9559f665
Contact: sip:910000@172.30.1.206:5060
Call-ID: 1445c0af175505fd7766a1ff26395496@172.30.1.206:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.8.0
Content-Length: 0


-- SIP/ttk-0000019e answered SIP/201-0000019d

<— SIP read from UDP:172.31.1.34:5060 —>
OPTIONS sip:910000@172.30.1.206:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bK4e153a2c7e6ec67ce772bd91d
Call-ID: 1445c0af175505fd7766a1ff26395496@172.30.1.206:5060
From: sip:420000@172.31.1.34;tag=9559f665
To: "Test2"sip:910000@172.30.1.206;tag=as16da7a9c
CSeq: 1 OPTIONS
Accept: application/sdp
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —

<— Transmitting (no NAT) to 172.31.1.34:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bK4e153a2c7e6ec67ce772bd91d;received=172.31.1.34
From: sip:420000@172.31.1.34;tag=9559f665
To: "Test2"sip:910000@172.30.1.206;tag=as16da7a9c
Call-ID: 1445c0af175505fd7766a1ff26395496@172.30.1.206:5060
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:910000@172.30.1.206:5060
Accept: application/sdp
Content-Length: 0

<------------>

<— SIP read from UDP:172.31.1.34:5060 —>
BYE sip:910000@172.30.1.206:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bKf712fa4e75270d0254538e796
Call-ID: 1445c0af175505fd7766a1ff26395496@172.30.1.206:5060
From: sip:420000@172.31.1.34;tag=9559f665
To: "Test2"sip:910000@172.30.1.206;tag=as16da7a9c
CSeq: 2 BYE
Max-Forwards: 70
Reason: Q.850;cause=16;text="normal call clearing"
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Sending to 172.31.1.34:5060 (no NAT)
Scheduling destruction of SIP dialog ‘1445c0af175505fd7766a1ff26395496@172.30.1.206:5060’ in 32000 ms (Method: BYE)

<— Transmitting (no NAT) to 172.31.1.34:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bKf712fa4e75270d0254538e796;received=172.31.1.34
From: sip:420000@172.31.1.34;tag=9559f665
To: "Test2"sip:910000@172.30.1.206;tag=as16da7a9c
Call-ID: 1445c0af175505fd7766a1ff26395496@172.30.1.206:5060
CSeq: 2 BYE
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (office, 420000, 1) exited non-zero on ‘SIP/201-0000019d’
[/code]

When I (a calling peer) hang up the call. Incorrect call.

[code] – Executing [420000@office:1] Dial(“SIP/201-00000199”, “SIP/ttk/420000,T”) in new stack
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 172.31.1.34:5060:
INVITE sip:420000@172.31.1.34 SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK39c629e0
Max-Forwards: 70
From: “Test2” sip:910000@172.30.1.206;tag=as0492590f
To: sip:420000@172.31.1.34
Contact: sip:910000@172.30.1.206:5060
Call-ID: 7976cc6d244d16a46e20e07a656b8163@172.30.1.206:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.8.0
Date: Mon, 09 Jan 2012 13:47:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 1333822630 1333822630 IN IP4 172.30.1.206
s=Asterisk PBX 1.8.8.0
c=IN IP4 172.30.1.206
t=0 0
m=audio 19164 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv


-- Called SIP/ttk/420000

<— SIP read from UDP:172.31.1.34:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK39c629e0
Call-ID: 7976cc6d244d16a46e20e07a656b8163@172.30.1.206:5060
From: "Test2"sip:910000@172.30.1.206;tag=as0492590f
To: sip:420000@172.31.1.34
CSeq: 102 INVITE
Content-Length: 0

<------------->
— (7 headers 0 lines) —

<— SIP read from UDP:172.31.1.34:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK39c629e0
Call-ID: 7976cc6d244d16a46e20e07a656b8163@172.30.1.206:5060
From: "Test2"sip:910000@172.30.1.206;tag=as0492590f
To: sip:420000@172.31.1.34;tag=a0a610ac
CSeq: 102 INVITE
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,PRACK,SUBSCRIBE,NOTIFY,UPDATE,MESSAGE,REFER
Contact: sip:420000@172.31.1.34:5060;user=phone
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 9874975 9874975 IN IP4 172.31.1.34
s=Sip Call
c=IN IP4 172.31.1.34
t=0 0
m=audio 32256 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (10 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.31.1.34:32256
– SIP/ttk-0000019a is ringing
– SIP/ttk-0000019a is making progress passing it to SIP/201-00000199

<— SIP read from UDP:172.31.1.34:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK39c629e0
Call-ID: 7976cc6d244d16a46e20e07a656b8163@172.30.1.206:5060
From: "Test2"sip:910000@172.30.1.206;tag=as0492590f
To: sip:420000@172.31.1.34;tag=a0a610ac
CSeq: 102 INVITE
Contact: sip:420000@172.31.1.34:5060;user=phone
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY,MESSAGE,REFER,PUBLISH
Content-Length: 203
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 9874975 9874976 IN IP4 172.31.1.34
s=Sip Call
c=IN IP4 172.31.1.34
t=0 0
m=audio 32256 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (10 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 172.31.1.34:32256
list_route: hop: sip:420000@172.31.1.34:5060;user=phone
set_destination: Parsing sip:420000@172.31.1.34:5060;user=phone for address/port to send to
set_destination: set destination to 172.31.1.34:5060
Transmitting (no NAT) to 172.31.1.34:5060:
ACK sip:420000@172.31.1.34:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK6a8b811b
Max-Forwards: 70
From: “Test2” sip:910000@172.30.1.206;tag=as0492590f
To: sip:420000@172.31.1.34;tag=a0a610ac
Contact: sip:910000@172.30.1.206:5060
Call-ID: 7976cc6d244d16a46e20e07a656b8163@172.30.1.206:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.8.0
Content-Length: 0


-- SIP/ttk-0000019a answered SIP/201-00000199

<— SIP read from UDP:172.31.1.34:5060 —>
OPTIONS sip:910000@172.30.1.206:5060 SIP/2.0
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bK2b18475285c9e93483d26f914
Call-ID: 7976cc6d244d16a46e20e07a656b8163@172.30.1.206:5060
From: sip:420000@172.31.1.34;tag=a0a610ac
To: "Test2"sip:910000@172.30.1.206;tag=as0492590f
CSeq: 1 OPTIONS
Accept: application/sdp
Max-Forwards: 70
Content-Length: 0

<------------->
— (9 headers 0 lines) —

<— Transmitting (no NAT) to 172.31.1.34:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.31.1.34:5060;branch=z9hG4bK2b18475285c9e93483d26f914;received=172.31.1.34
From: sip:420000@172.31.1.34;tag=a0a610ac
To: "Test2"sip:910000@172.30.1.206;tag=as0492590f
Call-ID: 7976cc6d244d16a46e20e07a656b8163@172.30.1.206:5060
CSeq: 1 OPTIONS
Server: Asterisk PBX 1.8.8.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:910000@172.30.1.206:5060
Accept: application/sdp
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘7976cc6d244d16a46e20e07a656b8163@172.30.1.206:5060’ in 32000 ms (Method: OPTIONS)
set_destination: Parsing sip:420000@172.31.1.34:5060;user=phone for address/port to send to
set_destination: set destination to 172.31.1.34:5060
Reliably Transmitting (no NAT) to 172.31.1.34:5060:
BYE sip:420000@172.31.1.34:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK0b02d2ed
Max-Forwards: 70
From: “Test2” sip:910000@172.30.1.206;tag=as0492590f
To: sip:420000@172.31.1.34;tag=a0a610ac
Call-ID: 7976cc6d244d16a46e20e07a656b8163@172.30.1.206:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.8.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


== Spawn extension (office, 420000, 1) exited non-zero on ‘SIP/201-00000199’

<— SIP read from UDP:172.31.1.34:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.1.206:5060;branch=z9hG4bK0b02d2ed
Call-ID: 7976cc6d244d16a46e20e07a656b8163@172.30.1.206:5060
From: "Test2"sip:910000@172.30.1.206;tag=as0492590f
To: sip:420000@172.31.1.34;tag=a0a610ac
CSeq: 103 BYE
Content-Length: 0

<------------->
— (7 headers 0 lines) —
[Jan 10 00:47:54] NOTICE[60372]: chan_sip.c:20249 handle_response_peerpoke: Peer ‘ttk’ is now Lagged. (387477ms / 0ms)
Really destroying SIP dialog ‘7976cc6d244d16a46e20e07a656b8163@172.30.1.206:5060’ Method: OPTIONS
[/code]

The problem has been fixed in 1.8.9.0-rc1.