Hi, i´m trying to create a SIP Trunk between Asterisk and a CUCM 8.6, in the sip.conf this is the configuration for the call manager:
[callmanageroth]
type=friend
context=Fromtrunk
host=10.109.5.10
disallow=all
allow=ulaw
allow=alaw
canreinvite=yes
qualify=yes
In the CUCM i have created the SIP Trunk following the instructions on this page: voip-info.org/wiki/view/Aste … ntegration
In extensions.conf i have the followign contexts:
[code][trunk]
exten => _2XXX,1,Dial(SIP/${EXTEN}@callmanageroth)
exten => _7XXX,1,Dial(SIP/${EXTEN}@callmanageroth)
[Fromtrunk]
exten => #87,1,Dial(SIP/softphone1) ;softphone installed on PC[/code]
Now, when i try to make a call to an extesion of the Callmanager, for example 2953 i get the following error:
== Using SIP RTP CoS mark 5
-- Executing [2953@LocalSets:1] Dial("SIP/softphone1-00000017", "SIP/2953@callmanageroth") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/2953@callmanageroth
-- Got SIP response 503 "Service Unavailable" back from 10.109.5.10:5060
-- SIP/callmanageroth-00000018 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/softphone1-00000017' status is 'CONGESTION'
When i enable SIP Debug this is what i get:
<------------>
Audio is at 17578
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.109.5.10:5060:
INVITE sip:2953@10.109.5.10 SIP/2.0
Via: SIP/2.0/UDP 10.109.20.62:5060;branch=z9hG4bK3c15ea50
Max-Forwards: 70
From: "Carlos Garcia" <sip:8211@10.109.20.62>;tag=as7f84f26b
To: <sip:2953@10.109.5.10>
Contact: <sip:8211@10.109.20.62:5060>
Call-ID: 06a36d7d2045e68b5c53c0f76a6e25ff@10.109.20.62:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-11-r406722
Date: Thu, 13 Feb 2014 23:28:30 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 299
v=0
o=root 123368445 123368445 IN IP4 10.109.20.62
s=Asterisk PBX SVN-branch-11-r406722
c=IN IP4 10.109.20.62
t=0 0
m=audio 17578 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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:10.109.5.10:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.109.20.62:5060;branch=z9hG4bK3c15ea50
From: "Carlos Garcia" <sip:8211@10.109.20.62>;tag=as7f84f26b
To: <sip:2953@10.109.5.10>
Date: Thu, 13 Feb 2014 23:28:25 GMT
Call-ID: 06a36d7d2045e68b5c53c0f76a6e25ff@10.109.20.62:5060
CSeq: 102 INVITE
Allow-Events: presence
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:10.109.5.10:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.109.20.62:5060;branch=z9hG4bK3c15ea50
From: "Carlos Garcia" <sip:8211@10.109.20.62>;tag=as7f84f26b
To: <sip:2953@10.109.5.10>;tag=1921275670
Date: Thu, 13 Feb 2014 23:28:25 GMT
Call-ID: 06a36d7d2045e68b5c53c0f76a6e25ff@10.109.20.62:5060
CSeq: 102 INVITE
Allow-Events: presence
Warning: 399 CM-GVTA-P "Unable to find a device handler for the request received on port 5060 from 10.109.20.62"
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (no NAT) to 10.109.5.10:5060:
ACK sip:2953@10.109.5.10 SIP/2.0
Via: SIP/2.0/UDP 10.109.20.62:5060;branch=z9hG4bK3c15ea50
Max-Forwards: 70
From: "Carlos Garcia" <sip:8211@10.109.20.62>;tag=as7f84f26b
To: <sip:2953@10.109.5.10>;tag=1921275670
Contact: <sip:8211@10.109.20.62:5060>
Call-ID: 06a36d7d2045e68b5c53c0f76a6e25ff@10.109.20.62:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX SVN-branch-11-r406722
Content-Length: 0
---
<--- Reliably Transmitting (NAT) to 10.109.20.25:17078 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 10.109.20.25:17078;branch=z9hG4bK-d8754z-39ad1100c43a5c5d-1---d8754z-;received=10.109.20.25;rport=17078
From: "Carlos Garcia"<sip:softphone1@10.109.20.62>;tag=c8dd6f35
To: <sip:2953@10.109.20.62>;tag=as7ada35c9
Call-ID: YTk1MGU4NDJiNzY2MDJhZmY3OGU0MjIyMWM4N2M1YWE
CSeq: 2 INVITE
Server: Asterisk PBX SVN-branch-11-r406722
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
I suspect the problem is related to the configuration on the CUCM, but i haven´t been able to find an accurate configuration guide for the SIP Trunk.
Any help would be apreciated.