Continual Busy/Congested

I’m trying to update a Centos 5 system running asterisk 1.8 to run on Centos 6. I had no success getting v 1.8 RPM’s on Centos 6 so I compiled them and Dahdi drivers using this guide

linuxmoz.com/asterisk-centos-6-install-guide/

The versions I use match the exact versions from the original Centos5 machine which works fine. The config files are almost exactly the same but for some reason I cannot make a connection.

Running asterisk -r on the sending machine gives me

== Using SIP RTP CoS mark 5 -- Executing [1025@default:1] Dial("SIP/127.0.0.1-00000521", "DAHDI/g1/1025") in new stack [Sep 8 10:24:22] WARNING[28227]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel 'SIP/127.0.0.1-00000521' status is 'CONGESTION'
No call information appears in asterisk on the receiving machine. ‘dahdi show channels’ shows every channel ‘In Service’ and the status shows:

dahdi show status Description Alarms IRQ bpviol CRC Fra Codi Options LBO T2XXP (PCI) Card 0 Span 1 OK 0 0 1 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T2XXP (PCI) Card 0 Span 2 OK 0 0 1 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1)
The sync lights on the card are green and the sending astersisk reports correctly if the connections are broken and remade.
sip.conf:

[code]#include sip_custom.conf

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes[/code]

With sip debug enabled on sender:

[code]CLI> sip set debug on
SIP Debugging enabled

<— SIP read from UDP:192.168.103.157:5061 —>
INVITE sip:1025@192.168.103.157:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-28672-1-0
From: sipp sip:sipp@127.0.0.1:5061;tag=28672SIPpTag001
To: sut sip:1025@192.168.103.157:5060
Call-ID: 1-28672@127.0.0.1
CSeq: 1 INVITE
Contact: sip:sipp@127.0.0.1:5061
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 129

v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
<------------->
— (11 headers 7 lines) —
Sending to 192.168.103.157:5061 (NAT)
Using INVITE request as basis request - 1-28672@127.0.0.1
No matching peer for ‘sipp’ from ‘192.168.103.157:5061’
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found audio description format PCMU for ID 0
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 127.0.0.1:6000
Looking for 1025 in default (domain 192.168.103.157)
list_route: hop: sip:sipp@127.0.0.1:5061

<— Transmitting (NAT) to 192.168.103.157:5061 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-28672-1-0;received=192.168.103.157;rport=5061
From: sipp sip:sipp@127.0.0.1:5061;tag=28672SIPpTag001
To: sut sip:1025@192.168.103.157:5060
Call-ID: 1-28672@127.0.0.1
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.26.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:1025@192.168.103.157:5060
Content-Length: 0

<------------>
– Executing [1025@default:1] Dial(“SIP/127.0.0.1-00000527”, “DAHDI/g1/1025”) in new stack
[Sep 8 11:17:41] WARNING[28673]: app_dial.c:2345 dial_exec_full: Unable to create channel of type ‘DAHDI’ (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/127.0.0.1-00000527’ status is ‘CONGESTION’

<— Reliably Transmitting (NAT) to 192.168.103.157:5061 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-28672-1-0;received=192.168.103.157;rport=5061
From: sipp sip:sipp@127.0.0.1:5061;tag=28672SIPpTag001
To: sut sip:1025@192.168.103.157:5060;tag=as6460538e
Call-ID: 1-28672@127.0.0.1
CSeq: 1 INVITE
Server: Asterisk PBX 1.8.26.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Circuit/channel congestion
X-Asterisk-HangupCauseCode: 34
Content-Length: 0

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

<— SIP read from UDP:192.168.103.157:5061 —>
ACK sip:1025@192.168.103.157:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-28672-1-0;received=192.168.103.157;rport=5061
From: sipp sip:sipp@127.0.0.1:5061;tag=28672SIPpTag001
To: sut sip:1025@192.168.103.157:5060;tag=as6460538e
Call-ID: 1-28672@127.0.0.1
CSeq: 1 ACK
Contact: sip:sipp@127.0.0.1:5061;transport=UDP
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Really destroying SIP dialog ‘1-28672@127.0.0.1’ Method: ACK

<— SIP read from UDP:192.168.103.157:5061 —>
INVITE sip:1025@192.168.103.157:5060 SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5061;branch=z9hG4bK-28672-2-0
From: sipp sip:sipp@127.0.0.1:5061;tag=28672SIPpTag002
To: sut sip:1025@192.168.103.157:5060
Call-ID: 2-28672@127.0.0.1
CSeq: 1 INVITE
Contact: sip:sipp@127.0.0.1:5061
Max-Forwards: 70
Subject: Performance Test
Content-Type: application/sdp
Content-Length: 129

v=0
o=user1 53655765 2353687637 IN IP4 127.0.0.1
s=-
c=IN IP4 127.0.0.1
t=0 0
m=audio 6000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
<------------->
[/code]

Not sure where to go from here.

In this case a full install of asterisk and Dahdi solved the problem

wiki.asterisk.org/wiki/display/ … rom+Source