One Asterisk not understand error code from another Asterisk

Asterisk Asterisk Support
1

Hello,

i have one problem. If i calling CiscoIP->Ast1->Ast2->Provider, and provider send to Ast2 that Client not unavailable in this time, Ast1 easy continue fake ringing to Cisco.

If i call from CiscoIP direct to ->Ast2->Prov, that all its good, in the phone i hear that, client unavailable in this time, please call back letter.

Please help my, why ast1 not accept right answer from ast2.

My configuration its:
CiscoIP -> Asterisk 1.8.32.3 -> Asterisk 13.3.0 -> Providers

ast1 - sip.conf
;CiscoIP in ast 1 and last the same, only context another name.
[100]
type=friend
host=dynamic
username=100
secret=1234
nat=yes
canreinvite=no
context=trans
directmedia=yes
disallow=all
allow=alaw

[to-ast2]
type=friend
host=xx.xx.xx.xx
disallow=all
allow=alaw
insecure=no
nat=no
qualify=yes
directmedia=no
dtmfmode=RFC2833

ast1 - extensions.conf
[trans]
exten => _X.,1,Dial(SIP/${EXTEN}@to-ast2)

ast2 - sip.conf

[fromast1]
host=x.x.x.x
context=voipprov
type=friend
disallow=all
allow=alaw
insecure=port,invite
nat=no
qualify=yes
directmedia=yes
dtmfmode=inband
dtmf=RFC2833
rfc2833compensate=yes

[voip_prov]
type=friend
;type=peer
host=xx.xxx.xx.xxx
disallow=all
allow=alaw
insecure=no
nat=no
qualify=yes
directmedia=no
dtmfmode=auto

ast2 - extensions.conf

[voipprov]
exten => _X.,1,Dial(SIP/${EXTEN}@ voip_prov)

thank you for help.

2

More likely there isn’t an error code, but rather just early media. You have to call Progress() for early media to be passed through.

3

== Using SIP RTP CoS mark 5
– Executing [420XXXXXX@AST2:1] Dial(“SIP/AST1-0000bf8b”, “OOH323/420XXXXXX@ ISP1”) in new stack

-- Called OOH323/420XXXXXX@ ISP1

== Everyone is busy/congested at this time (1:0/1/0)
– Executing [420XXXXXX@AST2:2] Dial(“SIP/AST1-0000bf8b”, “SIP/420XXXXXX@ ISP2”) in new stack

== Using SIP RTP CoS mark 5
– Called SIP/420XXXXXX@ ISP2
– SIP/tc150119-0000bf8d is ringing
– SIP/tc150119-0000bf8d is making progress passing it to SIP/AST1-0000bf8b

4

It is not clear what this log is supposed to be showing, but the first call has an error code and the second one has early media.

5

yes, but my questions its, that asterisk 1 send to the phone normal ring tone, if then you can see, in the log going error code. If i call direct touch asterisk 2, that waiting for answer and, if will be error, than going to hangup phone. Im not understanding, why asterisk 1 not waiting from asterisk 2 answer, and send immedlity fake ring.

6

I am having difficult understanding your English, there are a lot of clearly wrong words, but I can’t work out what all of them should be.

Also, you haven’t provided enough logging to be really clear what is happening. You need “sip set debug on” on Asterisk 2.

However, the log you have provided shows two calls. The first call is via ISP1. The second call is via ISP2.

ISP1 has responded with an error final status, with on intermediate status, except for 100 Trying.

The log shows no final status from ISP2, but they have sent:

180 Ringing

followed by:

183 Progress

Typically in such a case, the 183 Progress is followed by early media, either consisting of ringback tone, or an error indication, which may be a tone, or a voice announcement. These tones will be forwarded to the caller, but not interpreted by Asterisk. The call will stay in an early media state until either ISP2 sends a final status, or the caller cancels the call.

7
8

so, i finding who its my problem. its resolved.
in sip.conf => progressinband = yes.

this parameter help me.

thank you for all. For david55 two much thank you !