----- Original Message -----
From: "Michiel van Baak" <[EMAIL PROTECTED]>
To: <asterisk-users@lists.digium.com>
Sent: Wednesday, April 02, 2008 10:51 AM
Subject: Re: [asterisk-users] Howto connect to Cirpack softswitch
withAsterisk ?
> On 10:11, Wed 02 Apr 08, Robert Rozman wrote:
>> Hi,
>>
>> has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any
>> howto
>> or more info about needed Asterisk SW and setup ?
>
> Yes, it works fine.
> Where do you get stuck ?
> It's basically a normal sip connection setup.
>
Hi,
thanks for response....
I have it registered and receiveing incoming calls, but outgoing calls don't
work. I'm attaching sip log below, the basic problem is that some sort of
authentication is desired on outgoing calls...
Cirpack says: SIP/2.0 407 authentication required
and then
Cirpack says: SIP/2.0 403 Wrong login or password
I'm attaching full log below.. I'd kindly ask if someone can shed some
light, where to specify outgoing authentication (I use freepbx also) ?
Can incoming calls be proceeded to ring local extensions without actually
taking call (so ISP won't charge for just ringing) ?
Thanks in advance,
regards,
Rob.
SIP full log :
Really destroying SIP dialog '[EMAIL PROTECTED]'
Method: REGISTER
-- Executing [EMAIL PROTECTED]:1] Macro("SIP/202-b654e668",
"dialout-trunk|2|041461620||") in new stack
-- Executing [EMAIL PROTECTED]:1] Set("SIP/202-b654e668",
"DIAL_TRUNK=2") in new stack
-- Executing [EMAIL PROTECTED]:2] Set("SIP/202-b654e668",
"DIAL_NUMBER=041461620") in new stack
-- Executing [EMAIL PROTECTED]:3] Set("SIP/202-b654e668",
"ROUTE_PASSWD=") in new stack
-- Executing [EMAIL PROTECTED]:4] GotoIf("SIP/202-b654e668",
"1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing [EMAIL PROTECTED]:6] GotoIf("SIP/202-b654e668",
"0?disabletrunk|1") in new stack
-- Executing [EMAIL PROTECTED]:7] Set("SIP/202-b654e668",
"_NODEST=") in new stack
-- Executing [EMAIL PROTECTED]:8] Set("SIP/202-b654e668",
"DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [EMAIL PROTECTED]:9] Set("SIP/202-b654e668",
"GROUP()=OUT_2") in new stack
-- Executing [EMAIL PROTECTED]:10] Macro("SIP/202-b654e668",
"user-callerid|SKIPTTL") in new stack
-- Executing [EMAIL PROTECTED]:1] NoOp("SIP/202-b654e668",
"user-callerid: device 202") in new stack
-- Executing [EMAIL PROTECTED]:2] Set("SIP/202-b654e668",
"AMPUSER=202") in new stack
-- Executing [EMAIL PROTECTED]:3] GotoIf("SIP/202-b654e668",
"0?report") in new stack
-- Executing [EMAIL PROTECTED]:4] GotoIf("SIP/202-b654e668",
"0?start") in new stack
-- Executing [EMAIL PROTECTED]:5] Set("SIP/202-b654e668",
"REALCALLERIDNUM=202") in new stack
-- Executing [EMAIL PROTECTED]:6] NoOp("SIP/202-b654e668",
"REALCALLERIDNUM is 202") in new stack
-- Executing [EMAIL PROTECTED]:7] Set("SIP/202-b654e668",
"AMPUSER=202") in new stack
-- Executing [EMAIL PROTECTED]:8] Set("SIP/202-b654e668",
"AMPUSERCIDNAME=pl_51") in new stack
-- Executing [EMAIL PROTECTED]:9] GotoIf("SIP/202-b654e668",
"0?report") in new stack
-- Executing [EMAIL PROTECTED]:10] Set("SIP/202-b654e668",
"AMPUSERCID=202") in new stack
-- Executing [EMAIL PROTECTED]:11] Set("SIP/202-b654e668",
"CALLERID(all)="pl_51" <202>") in new stack
-- Executing [EMAIL PROTECTED]:12] Set("SIP/202-b654e668",
"REALCALLERIDNUM=202") in new stack
-- Executing [EMAIL PROTECTED]:13] NoOp("SIP/202-b654e668", "TTL:
ARG1: SKIPTTL") in new stack
-- Executing [EMAIL PROTECTED]:14] GotoIf("SIP/202-b654e668",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [EMAIL PROTECTED]:23] NoOp("SIP/202-b654e668", "Using
CallerID "pl_51" <202>") in new stack
-- Executing [EMAIL PROTECTED]:11] Macro("SIP/202-b654e668",
"record-enable|202|OUT") in new stack
-- Executing [EMAIL PROTECTED]:1] GotoIf("SIP/202-b654e668",
"0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [EMAIL PROTECTED]:4] AGI("SIP/202-b654e668",
"recordingcheck|20080402-143454|1207139694.24") in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
recordingcheck|20080402-143454|1207139694.24: Outbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing [EMAIL PROTECTED]:5] NoOp("SIP/202-b654e668", "No
recording needed") in new stack
-- Executing [EMAIL PROTECTED]:12] GotoIf("SIP/202-b654e668",
"0?skipoutcid") in new stack
-- Executing [EMAIL PROTECTED]:13] Set("SIP/202-b654e668",
"DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [EMAIL PROTECTED]:14] Macro("SIP/202-b654e668",
"outbound-callerid|2") in new stack
-- Executing [EMAIL PROTECTED]:1] GotoIf("SIP/202-b654e668",
"1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing [EMAIL PROTECTED]:3] NoOp("SIP/202-b654e668",
"REALCALLERIDNUM is 202") in new stack
-- Executing [EMAIL PROTECTED]:4] GotoIf("SIP/202-b654e668",
"1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing [EMAIL PROTECTED]:9] Set("SIP/202-b654e668",
"USEROUTCID="pl_51" <202>") in new stack
-- Executing [EMAIL PROTECTED]:10] Set("SIP/202-b654e668",
"EMERGENCYCID=") in new stack
-- Executing [EMAIL PROTECTED]:11] Set("SIP/202-b654e668",
"TRUNKOUTCID="Robert Rozman" <0038659972778>") in new stack
-- Executing [EMAIL PROTECTED]:12] GotoIf("SIP/202-b654e668",
"1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing [EMAIL PROTECTED]:16] GotoIf("SIP/202-b654e668",
"0?usercid") in new stack
-- Executing [EMAIL PROTECTED]:17] Set("SIP/202-b654e668",
"CALLERID(all)=Robert Rozman <0038659972778>") in new stack
-- Executing [EMAIL PROTECTED]:18] GotoIf("SIP/202-b654e668",
"0?report") in new stack
-- Executing [EMAIL PROTECTED]:19] Set("SIP/202-b654e668",
"CALLERID(all)=pl_51 <202>") in new stack
-- Executing [EMAIL PROTECTED]:20] GotoIf("SIP/202-b654e668",
"1?report:hidecid") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing [EMAIL PROTECTED]:22] NoOp("SIP/202-b654e668",
"CallerID set to "pl_51" <202>") in new stack
-- Executing [EMAIL PROTECTED]:15] GotoIf("SIP/202-b654e668",
"0?nomax") in new stack
-- Executing [EMAIL PROTECTED]:16] GotoIf("SIP/202-b654e668",
"0?chanfull") in new stack
-- Executing [EMAIL PROTECTED]:17] AGI("SIP/202-b654e668",
"fixlocalprefix") in new stack
-- Launched AGI Script /usr/share/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix completed, returning 0
-- Executing [EMAIL PROTECTED]:18] Set("SIP/202-b654e668",
"OUTNUM=041461620") in new stack
-- Executing [EMAIL PROTECTED]:19] Set("SIP/202-b654e668",
"custom=SIP/SIOL") in new stack
-- Executing [EMAIL PROTECTED]:20] GotoIf("SIP/202-b654e668",
"1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,24)
-- Executing [EMAIL PROTECTED]:24] GotoIf("SIP/202-b654e668",
"0?customtrunk") in new stack
-- Executing [EMAIL PROTECTED]:25] Dial("SIP/202-b654e668",
"SIP/SIOL/041461620|300|") in new stack
Audio is at 10.135.125.59 port 10150
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.253.1.31:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.135.125.59:5060;branch=z9hG4bK552e1b7d
From: "pl_51" <sip:[EMAIL PROTECTED]>;tag=as1c51c8ff
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 02 Apr 2008 12:34:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 502
v=0
o=root 29289 29289 IN IP4 10.135.125.59
s=session
c=IN IP4 10.135.125.59
t=0 0
m=audio 10150 RTP/AVP 0 8 3 112 5 10 7 110 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called SIOL/041461620
dcerouter*CLI>
<--- SIP read from 10.253.1.31:5060 --->
SIP/2.0 407 authentication required
Allow: UPDATE,REFER,INFO
Call-ID: [EMAIL PROTECTED]
Contact: <sip:[EMAIL PROTECTED]:5060;user=phone>
CSeq: 102 INVITE
From: "pl_51" <sip:[EMAIL PROTECTED]>;tag=as1c51c8ff
Proxy-Authenticate: Digest
realm="voip.siol",nonce="17a6b2744e3c103608b7e42770da4b90",opaque="17a4975f0046e08",stale=false,algorithm=MD5
Server: Cirpack/v4.41f (gw_sip)
To: <sip:[EMAIL PROTECTED]>;tag=00-08009-17a6b4b1-4c4c7b8b3
Via: SIP/2.0/UDP
10.135.125.59:5060;received=10.135.125.59;branch=z9hG4bK552e1b7d
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 10.253.1.31:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.135.125.59:5060;branch=z9hG4bK552e1b7d
From: "pl_51" <sip:[EMAIL PROTECTED]>;tag=as1c51c8ff
To: <sip:[EMAIL PROTECTED]>;tag=00-08009-17a6b4b1-4c4c7b8b3
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Audio is at 10.135.125.59 port 10150
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.253.1.31:5060:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.135.125.59:5060;branch=z9hG4bK0969839a
From: "pl_51" <sip:[EMAIL PROTECTED]>;tag=as1c51c8ff
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="59972778", realm="voip.siol",
algorithm=MD5, uri="sip:[EMAIL PROTECTED]",
nonce="17a6b2744e3c103608b7e42770da4b90",
response="b4ac7ddbb27eee92b69be323cf89c335", opaque="17a4975f0046e08"
Date: Wed, 02 Apr 2008 12:34:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 502
v=0
o=root 29289 29290 IN IP4 10.135.125.59
s=session
c=IN IP4 10.135.125.59
t=0 0
m=audio 10150 RTP/AVP 0 8 3 112 5 10 7 110 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
dcerouter*CLI>
<--- SIP read from 10.253.1.31:5060 --->
SIP/2.0 100 Trying
Allow: UPDATE,REFER,INFO
Call-ID: [EMAIL PROTECTED]
Contact: <sip:10.253.1.31:5060>
CSeq: 103 INVITE
From: "pl_51" <sip:[EMAIL PROTECTED]>;tag=as1c51c8ff
Server: Cirpack/v4.41f (gw_sip)
To: <sip:[EMAIL PROTECTED]>
Via: SIP/2.0/UDP
10.135.125.59:5060;received=10.135.125.59;branch=z9hG4bK0969839a
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
dcerouter*CLI>
<--- SIP read from 10.253.1.31:5060 --->
SIP/2.0 403 Wrong login or password
Allow: UPDATE,REFER,INFO
Call-ID: [EMAIL PROTECTED]
Contact: <sip:10.253.1.31:5060>
CSeq: 103 INVITE
From: "pl_51" <sip:[EMAIL PROTECTED]>;tag=as1c51c8ff
Reason: q.850;cause=21
Server: Cirpack/v4.41f (gw_sip)
To: <sip:[EMAIL PROTECTED]>;tag=00-08116-17a6b4c5-184bf8a93
Via: SIP/2.0/UDP
10.135.125.59:5060;received=10.135.125.59;branch=z9hG4bK0969839a
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (no NAT) to 10.253.1.31:5060:
ACK sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 10.135.125.59:5060;branch=z9hG4bK0969839a
From: "pl_51" <sip:[EMAIL PROTECTED]>;tag=as1c51c8ff
To: <sip:[EMAIL PROTECTED]>;tag=00-08116-17a6b4c5-184bf8a93
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/SIOL-082053a8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing [EMAIL PROTECTED]:26] Goto("SIP/202-b654e668",
"s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [EMAIL PROTECTED]:1]
GotoIf("SIP/202-b654e668", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [EMAIL PROTECTED]:3]
NoOp("SIP/202-b654e668", "TRUNK Dial failed due to CONGESTION - failing
through to other trunks") in new stack
-- Executing [EMAIL PROTECTED]:2] Macro("SIP/202-b654e668",
"outisbusy|") in new stack
-- Executing [EMAIL PROTECTED]:1] Playback("SIP/202-b654e668",
"all-circuits-busy-now|noanswer") in new stack
-- <SIP/202-b654e668> Playing 'all-circuits-busy-now' (language 'en')
Really destroying SIP dialog
'[EMAIL PROTECTED]' Method: INVITE
-- Executing [EMAIL PROTECTED]:2] Playback("SIP/202-b654e668",
"pls-try-call-later|noanswer") in new stack
-- <SIP/202-b654e668> Playing 'pls-try-call-later' (language 'en')
-- Executing [EMAIL PROTECTED]:3] Macro("SIP/202-b654e668",
"hangupcall") in new stack
-- Executing [EMAIL PROTECTED]:1] ResetCDR("SIP/202-b654e668", "w") in
new stack
-- Executing [EMAIL PROTECTED]:2] NoCDR("SIP/202-b654e668", "") in new
stack
-- Executing [EMAIL PROTECTED]:3] GotoIf("SIP/202-b654e668",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [EMAIL PROTECTED]:6] GotoIf("SIP/202-b654e668",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [EMAIL PROTECTED]:9] GotoIf("SIP/202-b654e668",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [EMAIL PROTECTED]:11] Hangup("SIP/202-b654e668", "") in
new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/202-b654e668' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/202-b654e668' in macro 'outisbusy'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on
'SIP/202-b654e668'
dcerouter*CLI>
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