Can't make outbound calls via cisco spa 8800

Asterisk Asterisk Support
1

Hello!
I try to make an outbound calls to PSTN via CISCO SPA 8800

CISCO 8800:
10.100.1.197
User id: pstn2
FXO Line PORT: 5060
Make Call Without Reg: YES
Ans Call Without Reg: YES

sip.conf:
[pstn2]
type=friend
host=10.100.1.197 ;IP address of the SPA8800
port=5060 ;5161 is the default SIP port for line 2 on the SPA8800
dtmfmode=rfc2833
context=pstn2
insecure=very

extensions.conf:
exten => _44.,1,Dial(SIP/${EXTEN:1}@pstn2,60,r)

sip show peer pstn2

* Name : pstn2 Secret : <Not set> MD5Secret : <Not set> Remote Secret: <Not set> Context : pstn2 Subscr.Cont. : <Not set> Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : MOH Suggest : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 10 Max forwards : 0 Dynamic : No Callerid : "" <> MaxCallBR : 384 kbps Expire : -1 Insecure : no Force rport : Yes ACL : No DirectMedACL : No T.38 support : Yes T.38 EC mode : FEC T.38 MaxDtgrm: 400 DirectMedia : Yes PromiscRedir : No User=Phone : No Video Support: Yes Text Support : No Ign SDP ver : No Trust RPID : No Send RPID : No Subscriptions: Yes Overlap dial : No DTMFmode : rfc2833 Timer T1 : 500 Timer B : 32000 ToHost : 10.100.1.197 Addr->IP : 10.100.1.197:5060 Defaddr->IP : (null) Prim.Transp. : UDP Allowed.Trsp : UDP Def. Username: SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (alaw:20,ulaw:20) Auto-Framing : No Status : Unmonitored Useragent : Reg. Contact : Qualify Freq : 60000 ms Sess-Timers : Accept Sess-Refresh : uas Sess-Expires : 1800 secs Min-Sess : 90 secs RTP Engine : asterisk Parkinglot : Use Reason : No Encryption : No

But I Have problem:
CLI :

– Executing [4456@mgmn-users:1] Dial(“SIP/1117-00036a32”, “SIP/456@pstn2,60,r”) in new stack
== Using SIP RTP CoS mark 5
– Called SIP/456@pstn2
– Got SIP response 503 “Service Unavailable” back from 10.100.1.197:5060
– SIP/pstn2-00036a33 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Auto fallthrough, channel ‘SIP/1117-00036a32’ status is ‘CONGESTION’

What is wrong?
Thank you!

2

503 is a generic rejection code, so you need to debug this from Cisco side. Does the gateway provide logging?

Note that your insecure=very is being ignored, presumably because you are using a version of Asterisk that no longer recognizes that value for the parameter (probably because it was being misused). Also there is no reason why you would want insecure=invite, as you don’t have a password. I would suggest checking the logs from when you started Asterisk, as there may be other parameters that it is rejecting!

Also type=peer is almost always better than type=friend in sip.conf.

Neither of these will affect outgoing calls.