[asterisk-users] Howto connect to Cirpack softswitch withAsterisk ?
Robert Rozman
robert.rozman at comutel.si
Wed Apr 2 07:45:48 CDT 2008
----- Original Message -----
From: "Michiel van Baak" <michiel at vanbaak.info>
To: <asterisk-users at 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 '4c1c47a81b68c8ee5810ed4e359aa498 at 192.168.0.1'
Method: REGISTER
-- Executing [041461620 at from-internal:1] Macro("SIP/202-b654e668",
"dialout-trunk|2|041461620||") in new stack
-- Executing [s at macro-dialout-trunk:1] Set("SIP/202-b654e668",
"DIAL_TRUNK=2") in new stack
-- Executing [s at macro-dialout-trunk:2] Set("SIP/202-b654e668",
"DIAL_NUMBER=041461620") in new stack
-- Executing [s at macro-dialout-trunk:3] Set("SIP/202-b654e668",
"ROUTE_PASSWD=") in new stack
-- Executing [s at macro-dialout-trunk:4] GotoIf("SIP/202-b654e668",
"1?noauth") in new stack
-- Goto (macro-dialout-trunk,s,6)
-- Executing [s at macro-dialout-trunk:6] GotoIf("SIP/202-b654e668",
"0?disabletrunk|1") in new stack
-- Executing [s at macro-dialout-trunk:7] Set("SIP/202-b654e668",
"_NODEST=") in new stack
-- Executing [s at macro-dialout-trunk:8] Set("SIP/202-b654e668",
"DIAL_TRUNK_OPTIONS=tr") in new stack
-- Executing [s at macro-dialout-trunk:9] Set("SIP/202-b654e668",
"GROUP()=OUT_2") in new stack
-- Executing [s at macro-dialout-trunk:10] Macro("SIP/202-b654e668",
"user-callerid|SKIPTTL") in new stack
-- Executing [s at macro-user-callerid:1] NoOp("SIP/202-b654e668",
"user-callerid: device 202") in new stack
-- Executing [s at macro-user-callerid:2] Set("SIP/202-b654e668",
"AMPUSER=202") in new stack
-- Executing [s at macro-user-callerid:3] GotoIf("SIP/202-b654e668",
"0?report") in new stack
-- Executing [s at macro-user-callerid:4] GotoIf("SIP/202-b654e668",
"0?start") in new stack
-- Executing [s at macro-user-callerid:5] Set("SIP/202-b654e668",
"REALCALLERIDNUM=202") in new stack
-- Executing [s at macro-user-callerid:6] NoOp("SIP/202-b654e668",
"REALCALLERIDNUM is 202") in new stack
-- Executing [s at macro-user-callerid:7] Set("SIP/202-b654e668",
"AMPUSER=202") in new stack
-- Executing [s at macro-user-callerid:8] Set("SIP/202-b654e668",
"AMPUSERCIDNAME=pl_51") in new stack
-- Executing [s at macro-user-callerid:9] GotoIf("SIP/202-b654e668",
"0?report") in new stack
-- Executing [s at macro-user-callerid:10] Set("SIP/202-b654e668",
"AMPUSERCID=202") in new stack
-- Executing [s at macro-user-callerid:11] Set("SIP/202-b654e668",
"CALLERID(all)="pl_51" <202>") in new stack
-- Executing [s at macro-user-callerid:12] Set("SIP/202-b654e668",
"REALCALLERIDNUM=202") in new stack
-- Executing [s at macro-user-callerid:13] NoOp("SIP/202-b654e668", "TTL:
ARG1: SKIPTTL") in new stack
-- Executing [s at macro-user-callerid:14] GotoIf("SIP/202-b654e668",
"1?continue") in new stack
-- Goto (macro-user-callerid,s,23)
-- Executing [s at macro-user-callerid:23] NoOp("SIP/202-b654e668", "Using
CallerID "pl_51" <202>") in new stack
-- Executing [s at macro-dialout-trunk:11] Macro("SIP/202-b654e668",
"record-enable|202|OUT") in new stack
-- Executing [s at macro-record-enable:1] GotoIf("SIP/202-b654e668",
"0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing [s at macro-record-enable: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 [s at macro-record-enable:5] NoOp("SIP/202-b654e668", "No
recording needed") in new stack
-- Executing [s at macro-dialout-trunk:12] GotoIf("SIP/202-b654e668",
"0?skipoutcid") in new stack
-- Executing [s at macro-dialout-trunk:13] Set("SIP/202-b654e668",
"DIAL_TRUNK_OPTIONS=") in new stack
-- Executing [s at macro-dialout-trunk:14] Macro("SIP/202-b654e668",
"outbound-callerid|2") in new stack
-- Executing [s at macro-outbound-callerid:1] GotoIf("SIP/202-b654e668",
"1?start") in new stack
-- Goto (macro-outbound-callerid,s,3)
-- Executing [s at macro-outbound-callerid:3] NoOp("SIP/202-b654e668",
"REALCALLERIDNUM is 202") in new stack
-- Executing [s at macro-outbound-callerid:4] GotoIf("SIP/202-b654e668",
"1?normcid") in new stack
-- Goto (macro-outbound-callerid,s,9)
-- Executing [s at macro-outbound-callerid:9] Set("SIP/202-b654e668",
"USEROUTCID="pl_51" <202>") in new stack
-- Executing [s at macro-outbound-callerid:10] Set("SIP/202-b654e668",
"EMERGENCYCID=") in new stack
-- Executing [s at macro-outbound-callerid:11] Set("SIP/202-b654e668",
"TRUNKOUTCID="Robert Rozman" <0038659972778>") in new stack
-- Executing [s at macro-outbound-callerid:12] GotoIf("SIP/202-b654e668",
"1?trunkcid") in new stack
-- Goto (macro-outbound-callerid,s,16)
-- Executing [s at macro-outbound-callerid:16] GotoIf("SIP/202-b654e668",
"0?usercid") in new stack
-- Executing [s at macro-outbound-callerid:17] Set("SIP/202-b654e668",
"CALLERID(all)=Robert Rozman <0038659972778>") in new stack
-- Executing [s at macro-outbound-callerid:18] GotoIf("SIP/202-b654e668",
"0?report") in new stack
-- Executing [s at macro-outbound-callerid:19] Set("SIP/202-b654e668",
"CALLERID(all)=pl_51 <202>") in new stack
-- Executing [s at macro-outbound-callerid:20] GotoIf("SIP/202-b654e668",
"1?report:hidecid") in new stack
-- Goto (macro-outbound-callerid,s,22)
-- Executing [s at macro-outbound-callerid:22] NoOp("SIP/202-b654e668",
"CallerID set to "pl_51" <202>") in new stack
-- Executing [s at macro-dialout-trunk:15] GotoIf("SIP/202-b654e668",
"0?nomax") in new stack
-- Executing [s at macro-dialout-trunk:16] GotoIf("SIP/202-b654e668",
"0?chanfull") in new stack
-- Executing [s at macro-dialout-trunk: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 [s at macro-dialout-trunk:18] Set("SIP/202-b654e668",
"OUTNUM=041461620") in new stack
-- Executing [s at macro-dialout-trunk:19] Set("SIP/202-b654e668",
"custom=SIP/SIOL") in new stack
-- Executing [s at macro-dialout-trunk:20] GotoIf("SIP/202-b654e668",
"1?gocall") in new stack
-- Goto (macro-dialout-trunk,s,24)
-- Executing [s at macro-dialout-trunk:24] GotoIf("SIP/202-b654e668",
"0?customtrunk") in new stack
-- Executing [s at macro-dialout-trunk: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:041461620 at 10.253.1.31 SIP/2.0
Via: SIP/2.0/UDP 10.135.125.59:5060;branch=z9hG4bK552e1b7d
From: "pl_51" <sip:202 at 10.135.125.59>;tag=as1c51c8ff
To: <sip:041461620 at 10.253.1.31>
Contact: <sip:202 at 10.135.125.59>
Call-ID: 07e9465d3d1fa713277ae3fc34fda2d9 at 10.135.125.59
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: 07e9465d3d1fa713277ae3fc34fda2d9 at 10.135.125.59
Contact: <sip:041461620 at 10.253.1.31:5060;user=phone>
CSeq: 102 INVITE
From: "pl_51" <sip:202 at 10.135.125.59>;tag=as1c51c8ff
Proxy-Authenticate: Digest
realm="voip.siol",nonce="17a6b2744e3c103608b7e42770da4b90",opaque="17a4975f0046e08",stale=false,algorithm=MD5
Server: Cirpack/v4.41f (gw_sip)
To: <sip:041461620 at 10.253.1.31>;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:041461620 at 10.253.1.31 SIP/2.0
Via: SIP/2.0/UDP 10.135.125.59:5060;branch=z9hG4bK552e1b7d
From: "pl_51" <sip:202 at 10.135.125.59>;tag=as1c51c8ff
To: <sip:041461620 at 10.253.1.31>;tag=00-08009-17a6b4b1-4c4c7b8b3
Contact: <sip:202 at 10.135.125.59>
Call-ID: 07e9465d3d1fa713277ae3fc34fda2d9 at 10.135.125.59
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:041461620 at 10.253.1.31 SIP/2.0
Via: SIP/2.0/UDP 10.135.125.59:5060;branch=z9hG4bK0969839a
From: "pl_51" <sip:202 at 10.135.125.59>;tag=as1c51c8ff
To: <sip:041461620 at 10.253.1.31>
Contact: <sip:202 at 10.135.125.59>
Call-ID: 07e9465d3d1fa713277ae3fc34fda2d9 at 10.135.125.59
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="59972778", realm="voip.siol",
algorithm=MD5, uri="sip:041461620 at 10.253.1.31",
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: 07e9465d3d1fa713277ae3fc34fda2d9 at 10.135.125.59
Contact: <sip:10.253.1.31:5060>
CSeq: 103 INVITE
From: "pl_51" <sip:202 at 10.135.125.59>;tag=as1c51c8ff
Server: Cirpack/v4.41f (gw_sip)
To: <sip:041461620 at 10.253.1.31>
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: 07e9465d3d1fa713277ae3fc34fda2d9 at 10.135.125.59
Contact: <sip:10.253.1.31:5060>
CSeq: 103 INVITE
From: "pl_51" <sip:202 at 10.135.125.59>;tag=as1c51c8ff
Reason: q.850;cause=21
Server: Cirpack/v4.41f (gw_sip)
To: <sip:041461620 at 10.253.1.31>;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:041461620 at 10.253.1.31 SIP/2.0
Via: SIP/2.0/UDP 10.135.125.59:5060;branch=z9hG4bK0969839a
From: "pl_51" <sip:202 at 10.135.125.59>;tag=as1c51c8ff
To: <sip:041461620 at 10.253.1.31>;tag=00-08116-17a6b4c5-184bf8a93
Contact: <sip:202 at 10.135.125.59>
Call-ID: 07e9465d3d1fa713277ae3fc34fda2d9 at 10.135.125.59
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 [s at macro-dialout-trunk:26] Goto("SIP/202-b654e668",
"s-CONGESTION|1") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,1)
-- Executing [s-CONGESTION at macro-dialout-trunk:1]
GotoIf("SIP/202-b654e668", "1?noreport") in new stack
-- Goto (macro-dialout-trunk,s-CONGESTION,3)
-- Executing [s-CONGESTION at macro-dialout-trunk:3]
NoOp("SIP/202-b654e668", "TRUNK Dial failed due to CONGESTION - failing
through to other trunks") in new stack
-- Executing [041461620 at from-internal:2] Macro("SIP/202-b654e668",
"outisbusy|") in new stack
-- Executing [s at macro-outisbusy: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
'07e9465d3d1fa713277ae3fc34fda2d9 at 10.135.125.59' Method: INVITE
-- Executing [s at macro-outisbusy:2] Playback("SIP/202-b654e668",
"pls-try-call-later|noanswer") in new stack
-- <SIP/202-b654e668> Playing 'pls-try-call-later' (language 'en')
-- Executing [s at macro-outisbusy:3] Macro("SIP/202-b654e668",
"hangupcall") in new stack
-- Executing [s at macro-hangupcall:1] ResetCDR("SIP/202-b654e668", "w") in
new stack
-- Executing [s at macro-hangupcall:2] NoCDR("SIP/202-b654e668", "") in new
stack
-- Executing [s at macro-hangupcall:3] GotoIf("SIP/202-b654e668",
"1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s at macro-hangupcall:6] GotoIf("SIP/202-b654e668",
"1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s at macro-hangupcall:9] GotoIf("SIP/202-b654e668",
"1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s at macro-hangupcall: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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