Dear all,
i have device d-link DVG-6004S,
i need to make outgoing phone call throw it but when i make report i can see the numbers i call it,
my configuration is,
DVG-6004S i register it as a sip client in my asterisk with extension 510
when i dial 510 i have tone so i can make phone call but in this case when i make report the number i have call will not paper ,
i wish that i explain my problem will .
please i need urgent help kindly help me.
thanks
Insufficient information.
kindly let me know what kind of information you need
Definitely the verbose CLI log output for the failed call.
Possibly, sip.conf, and extensions.conf (assuming a non-GUI install - GUI ones really need you to find the relevant parts and these may be in included files or users.conf).
Possibly sip set debug on type SIP tracing.
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;
[general]
; These files will all be included in the [general] context
;
#include sip_general_additional.conf
;sip_general_custom.conf is the proper file location for placing any sip general
;options that you might need set. For example: enable and force the sip jitterbuffer.
;If these settings are desired they should be set the sip_general_custom.conf file.
;
; jbenable=yes
; jbforce=yes
;
;It is also the proper place to add the lines needed for sip nat’ing when going
;through a firewall. For nat’ing you’d need to add the following lines:
; nat=yes , externip= , localhost= , and optionally fromdomain= .
;
#include sip_general_custom.conf
;sip_nat.conf is here for legacy support reasons and for those that upgrade
;from previous versions. If you have this file with lines in it please make
;sure they are not duplicated in sip_general_custom.conf, if so remove them
;from sip_nat.conf as sip_general_custom.conf will have precedence.
#include sip_nat.conf
;sip_registrations_custom.conf is for any customizations you might need to do to
;the automatically generated registrations that FreePBX makes.
;
#include sip_registrations_custom.conf
#include sip_registrations.conf
; These files should all be expected to come after the [general] context
;
#include sip_custom.conf
#include sip_additional.conf
;sip_custom_post.conf If you have extra parameters that are needed for a
;extension to work to for example, those go here. So you have extension
;1000 defined in your system you start by creating a line 1000 in this
;file. Then on the next line add the extra parameter that is needed.
;When the sip.conf is loaded it will append your additions to the end of
;that extension.
;
#include sip_custom_post.conf
;--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. All modifications to ;
; this file must be done via the web gui. There are alternative files to make ;
; custom modifications, details at: freepbx.org/configuration_files ;
;--------------------------------------------------------------------------------;
;
[4010]
deny=0.0.0.0/0.0.0.0
type=friend
secret=401011
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=4010@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/4010
context=from-internal
canreinvite=no
callgroup=
callerid=device <4010>
accountcode=
call-limit=50
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
– Executing [00xxxxxxxx@from-internal:1] Macro(“SIP/420-b677baa0”, “user-callerid,SKIPTTL,”) in new stack
– Executing [s@macro-user-callerid:1] Set(“SIP/420-b677baa0”, “AMPUSER=420”) in new stack
– Executing [s@macro-user-callerid:2] GotoIf(“SIP/420-b677baa0”, “0?report”) in new stack
– Executing [s@macro-user-callerid:3] ExecIf(“SIP/420-b677baa0”, “1?Set(REALCALLERIDNUM=420)”) in new stack
– Executing [s@macro-user-callerid:4] Set(“SIP/420-b677baa0”, “AMPUSER=420”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/420-b677baa0”, “AMPUSERCIDNAME=massa”) in new stack
– Executing [s@macro-user-callerid:6] GotoIf(“SIP/420-b677baa0”, “0?report”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/420-b677baa0”, “AMPUSERCID=420”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/420-b677baa0”, “CALLERID(all)=“massa” <420>”) in new stack
– Executing [s@macro-user-callerid:9] Set(“SIP/420-b677baa0”, “REALCALLERIDNUM=420”) in new stack
– Executing [s@macro-user-callerid:10] ExecIf(“SIP/420-b677baa0”, “0?Set(CHANNEL(language)=)”) in new stack
– Executing [s@macro-user-callerid:11] GotoIf(“SIP/420-b677baa0”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,20)
– Executing [s@macro-user-callerid:20] NoOp(“SIP/420-b677baa0”, “Using CallerID “massa” <420>”) in new stack
– Executing [00xxxxxxxx@from-internal:2] Set(“SIP/420-b677baa0”, “_NODEST=”) in new stack
– Executing [00xxxxxxxx@from-internal:3] Macro(“SIP/420-b677baa0”, “record-enable,420,OUT,”) in new stack
– Executing [s@macro-record-enable:1] GotoIf(“SIP/420-b677baa0”, “1?check”) in new stack
– Goto (macro-record-enable,s,4)
– Executing [s@macro-record-enable:4] AGI(“SIP/420-b677baa0”, “recordingcheck,20110919-214237,1316457757.2111”) in new stack
– Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck,20110919-214237,1316457757.2111: Outbound recording not enabled
– <SIP/420-b677baa0>AGI Script recordingcheck completed, returning 0
– Executing [s@macro-record-enable:5] MacroExit(“SIP/420-b677baa0”, “”) in new stack
– Executing [00xxxxxxxx@from-internal:4] Macro(“SIP/420-b677baa0”, “dialout-trunk,5,00xxxxxxxx,”) in new stack
– Executing [s@macro-dialout-trunk:1] Set(“SIP/420-b677baa0”, “DIAL_TRUNK=5”) in new stack
– Executing [s@macro-dialout-trunk:2] GosubIf(“SIP/420-b677baa0”, “0?sub-pincheck,s,1”) in new stack
– Executing [s@macro-dialout-trunk:3] GotoIf(“SIP/420-b677baa0”, “0?disabletrunk,1”) in new stack
– Executing [s@macro-dialout-trunk:4] Set(“SIP/420-b677baa0”, “DIAL_NUMBER=00xxxxxxxx”) in new stack
– Executing [s@macro-dialout-trunk:5] Set(“SIP/420-b677baa0”, “DIAL_TRUNK_OPTIONS=tr”) in new stack
– Executing [s@macro-dialout-trunk:6] Set(“SIP/420-b677baa0”, “OUTBOUND_GROUP=OUT_5”) in new stack
– Executing [s@macro-dialout-trunk:7] GotoIf(“SIP/420-b677baa0”, “1?nomax”) in new stack
– Goto (macro-dialout-trunk,s,9)
– Executing [s@macro-dialout-trunk:9] GotoIf(“SIP/420-b677baa0”, “0?skipoutcid”) in new stack
– Executing [s@macro-dialout-trunk:10] Set(“SIP/420-b677baa0”, “DIAL_TRUNK_OPTIONS=”) in new stack
– Executing [s@macro-dialout-trunk:11] Macro(“SIP/420-b677baa0”, “outbound-callerid,5”) in new stack
– Executing [s@macro-outbound-callerid:1] ExecIf(“SIP/420-b677baa0”, “0?Set(CALLERPRES()=)”) in new stack
– Executing [s@macro-outbound-callerid:2] ExecIf(“SIP/420-b677baa0”, “0?Set(REALCALLERIDNUM=420)”) in new stack
– Executing [s@macro-outbound-callerid:3] GotoIf(“SIP/420-b677baa0”, “1?normcid”) in new stack
– Goto (macro-outbound-callerid,s,6)
– Executing [s@macro-outbound-callerid:6] Set(“SIP/420-b677baa0”, “USEROUTCID=”) in new stack
– Executing [s@macro-outbound-callerid:7] Set(“SIP/420-b677baa0”, “EMERGENCYCID=”) in new stack
– Executing [s@macro-outbound-callerid:8] Set(“SIP/420-b677baa0”, “TRUNKOUTCID=410”) in new stack
– Executing [s@macro-outbound-callerid:9] GotoIf(“SIP/420-b677baa0”, “1?trunkcid”) in new stack
– Goto (macro-outbound-callerid,s,12)
– Executing [s@macro-outbound-callerid:12] ExecIf(“SIP/420-b677baa0”, “1?Set(CALLERID(all)=410)”) in new stack
– Executing [s@macro-outbound-callerid:13] ExecIf(“SIP/420-b677baa0”, “0?Set(CALLERID(all)=)”) in new stack
– Executing [s@macro-outbound-callerid:14] ExecIf(“SIP/420-b677baa0”, “0?Set(CALLERPRES()=prohib_passed_screen)”) in new stack
– Executing [s@macro-dialout-trunk:12] ExecIf(“SIP/420-b677baa0”, “0?AGI(fixlocalprefix)”) in new stack
– Executing [s@macro-dialout-trunk:13] Set(“SIP/420-b677baa0”, “OUTNUM=00xxxxxxxx”) in new stack
– Executing [s@macro-dialout-trunk:14] Set(“SIP/420-b677baa0”, “custom=SIP/410”) in new stack
– Executing [s@macro-dialout-trunk:15] ExecIf(“SIP/420-b677baa0”, “0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^))”) in new stack
– Executing [s@macro-dialout-trunk:16] Macro(“SIP/420-b677baa0”, “dialout-trunk-predial-hook,”) in new stack
– Executing [s@macro-dialout-trunk-predial-hook:1] MacroExit(“SIP/420-b677baa0”, “”) in new stack
– Executing [s@macro-dialout-trunk:17] GotoIf(“SIP/420-b677baa0”, “0?bypass,1”) in new stack
– Executing [s@macro-dialout-trunk:18] GotoIf(“SIP/420-b677baa0”, “0?customtrunk”) in new stack
– Executing [s@macro-dialout-trunk:19] Dial(“SIP/420-b677baa0”, “SIP/410/00xxxxxxxx,300,”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using SIP VRTP TOS bits 136
== Using SIP VRTP CoS mark 6
– Called 410/00xxxxxxxx
– Got SIP response 482 “Loop Detected” back from 10.40.0.115
– Now forwarding SIP/420-b677baa0 to ‘Local/00xxxxxxxx@from-sip-external’ (thanks to SIP/410-b6a7af10)
– Executing [00xxxxxxxx@from-sip-external:1] NoOp(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “Received incoming SIP connection from unknown peer to 00xxxxxxxx”) in new stack
– Executing [00xxxxxxxx@from-sip-external:2] Set(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “DID=00xxxxxxxx”) in new stack
– Executing [00xxxxxxxx@from-sip-external:3] Goto(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “0?from-trunk,00xxxxxxxx,1”) in new stack
– Executing [s@from-sip-external:2] Set(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “TIMEOUT(absolute)=15”) in new stack
Channel will hangup at 2011-09-19 21:42:53.000 EEST.
– Executing [s@from-sip-external:3] Answer(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “”) in new stack
– Local/00xxxxxxxx@from-sip-external-5a59;1 answered SIP/420-b677baa0
– Executing [s@from-sip-external:4] Wait(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “2”) in new stack
– Executing [s@from-sip-external:5] Playback(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “ss-noservice”) in new stack
– Executing [s@from-sip-external:6] PlayTones(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “congestion”) in new stack
== Spawn extension (from-sip-external, s, 6) exited non-zero on ‘Local/00xxxxxxxx@from-sip-external-5a59;2’
– Executing [h@from-sip-external:1] NoOp(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “Hangup”) in new stack
– Executing [h@from-sip-external:2] Set(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “DID=s”) in new stack
– Executing [h@from-sip-external:3] Goto(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “s,1”) in new stack
– Goto (from-sip-external,s,1)
– Executing [s@from-sip-external:1] GotoIf(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “0?from-trunk,s,1”) in new stack
– Executing [s@from-sip-external:2] Set(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “TIMEOUT(absolute)=15”) in new stack
Channel will hangup at 2011-09-19 21:42:55.000 EEST.
– Executing [s@from-sip-external:3] Answer(“Local/00xxxxxxxx@from-sip-external-5a59;2”, “”) in new stack
== Spawn extension (from-sip-external, s, 3) exited non-zero on ‘Local/00xxxxxxxx@from-sip-external-5a59;2’
– Executing [h@macro-dialout-trunk:1] Macro(“SIP/420-b677baa0”, “hangupcall,”) in new stack
– Executing [s@macro-hangupcall:1] ResetCDR(“SIP/420-b677baa0”, “vw”) in new stack
– Executing [s@macro-hangupcall:2] NoCDR(“SIP/420-b677baa0”, “”) in new stack
– Executing [s@macro-hangupcall:3] GotoIf(“SIP/420-b677baa0”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,6)
– Executing [s@macro-hangupcall:6] GotoIf(“SIP/420-b677baa0”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] GotoIf(“SIP/420-b677baa0”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,11)
– Executing [s@macro-hangupcall:11] Hangup(“SIP/420-b677baa0”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/420-b677baa0’ in macro ‘hangupcall’
== Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on ‘SIP/420-b677baa0’
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on ‘SIP/420-b677baa0’ in macro ‘dialout-trunk’
== Spawn extension (from-internal, 00xxxxxxxx, 4) exited non-zero on ‘SIP/420-b677baa0’
You should not post the passwords for your SIP accounts!
[quote]-- Got SIP response 482 “Loop Detected” back from 10.40.0.115
– Now forwarding SIP/420-b677baa0 to ‘Local/00xxxxxxxx@from-sip-external’ (thanks to SIP/410-b6a7af10)[/quote]
What is 10.40.0.115?
As Asterisk is a back to back user agent, you should not be getting incoming 482’s even if it is misconfigured. There was a bug in which some versions of Asterisk generated this outbound in response to redirects, but you are not sourcing a redirect.
Have you possibly routed it to yourself (are you 10.40.0.115)?
Incidentally, I did say you needed to be selective in what you copied from GUIs.
Thanks for your advice,
10.40.0.115 this is my asterisk server
In that case, you are routing the call directly to yourself.
kindly i need to connect to my IP gateway from trunk ,
please can you help me to make that.
“exten = s,1,Dial(SIP/701,20,gtTwWmD(ww${ARG1:1}))“.
701 appears nowhere in your trace.
4010 does not appear in your extract of sip.conf.
At the moment, it looks like 4010 has been misconfigured to point back at your own machine.
If that doesn’t allow you to work out what you have done wrong, you will need to use sip set debug on and find out exactly what is happening at the SIP level.
As a general point, it is important to say that you are using a GUI (FreePBX is a GUI) at the start, as the normal approach to debugging problems on this forum is to assume that you write the dialplan yourself. GUIs are difficult to debug because the dialplans are complex and relatively few people know how they are supposed to work, and what can and can’t be configured.
701 is extension for IP-gateway
Hello, did you manage to move forward with this toppic?
I’m actually facing something similar as it looks to me. Topology:
100 is a sip phone, IP 192.168.1.107
Asterisk 1.8.8.0 running on a DD-WRT box with two ip interfaces: LAN 192.168.1.254 and WAN 172.16.1.100. Further on I am connected to a DSL router with public IP 92.81.131.67.
My provider is voip.clicknet.ro, the siptrunk with this provider is called romtelecom_voip.
When I originate any outgoing call using this SIP provider I get the error “SIP/2.0 482 Loop Detected” from my Asterisk.
In sip.conf I have the following configuration wih regards to NAT:
nat=yes
externaddr = ********.dyndns.org
localnet = 192.168.1.100/255.255.255.0
localnet = 172.16.1.100/255.255.255.0
qualify= yes
<— SIP read from UDP:192.168.1.107:5060 —>
INVITE sip:0754333333@192.168.1.254:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK35174378a4b5d2928
Max-Forwards: 70
From: Main Number sip:100@192.168.1.254;tag=ff699c2c47;epid=SC09e274
To: sip:0754333333@192.168.1.254:5060;transport=udp
Call-ID: 8d1ff4449bb75e25
CSeq: 1 INVITE
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,UPDATE
Allow-Events: talk, hold, conference
Contact: Main Number sip:100@192.168.1.107:5060;transport=udp
Min-SE: 90
Supported: timer, replaces, 100rel
User-Agent: optiPoint 410 Standard/V7 V7 R6.3.0
X-Siemens-Call-Type: ST-insecure
Content-Type: application/sdp
Content-Length: 371
v=0
o=MxSIP 0 1198510064 IN IP4 192.168.1.107
s=SIP Call
c=IN IP4 192.168.1.107
t=0 0
m=audio 5004 RTP/AVP 8 0 9 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
<------------->
— (16 headers 17 lines) —
Sending to 192.168.1.107:5060 (NAT)
Using INVITE request as basis request - 8d1ff4449bb75e25
Found peer ‘100’ for ‘100’ from 192.168.1.107:5060
<— Reliably Transmitting (NAT) to 192.168.1.107:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK35174378a4b5d2928;received=192.168.1.107;rport=5060
From: Main Number sip:100@192.168.1.254;tag=ff699c2c47;epid=SC09e274
To: sip:0754333333@192.168.1.254:5060;transport=udp;tag=as44a4d29e
Call-ID: 8d1ff4449bb75e25
CSeq: 1 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="797691de"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘8d1ff4449bb75e25’ in 6400 ms (Method: INVITE)
<— SIP read from UDP:192.168.1.107:5060 —>
ACK sip:0754333333@192.168.1.254:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK35174378a4b5d2928
Max-Forwards: 70
From: Main Number sip:100@192.168.1.254;tag=ff699c2c47;epid=SC09e274
To: sip:0754333333@192.168.1.254:5060;transport=udp;tag=as44a4d29e
Call-ID: 8d1ff4449bb75e25
CSeq: 1 ACK
User-Agent: optiPoint 410 Standard/V7 V7 R6.3.0
Content-Length: 0
<------------->
— (9 headers 0 lines) —
<— SIP read from UDP:192.168.1.107:5060 —>
INVITE sip:0754333333@192.168.1.254:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK63c4aa06f1afb7c86
Max-Forwards: 70
From: Main Number sip:100@192.168.1.254;tag=ff699c2c47;epid=SC09e274
To: sip:0754333333@192.168.1.254:5060;transport=udp
Call-ID: 8d1ff4449bb75e25
CSeq: 2 INVITE
Allow: INVITE,ACK,CANCEL,BYE,REFER,NOTIFY,UPDATE
Allow-Events: talk, hold, conference
Authorization: Digest username=“100”,realm=“asterisk”,nonce=“797691de”,uri=“sip:0754333333@192.168.1.254:5060;transport=udp”,response=“8a2b876568ac274de43f394c5c5e498c”,algorithm=MD5
Contact: Main Number sip:100@192.168.1.107:5060;transport=udp
Min-SE: 90
Supported: timer, replaces, 100rel
User-Agent: optiPoint 410 Standard/V7 V7 R6.3.0
X-Siemens-Call-Type: ST-insecure
Content-Type: application/sdp
Content-Length: 371
v=0
o=MxSIP 0 1198510064 IN IP4 192.168.1.107
s=SIP Call
c=IN IP4 192.168.1.107
t=0 0
m=audio 5004 RTP/AVP 8 0 9 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:4 annexa=no
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=ptime:20
<------------->
— (17 headers 17 lines) —
Sending to 192.168.1.107:5060 (NAT)
Using INVITE request as basis request - 8d1ff4449bb75e25
Found peer ‘100’ for ‘100’ from 192.168.1.107:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 9
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G722 for ID 9
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format telephone-event for ID 101
Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x110d (g723|ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.107:5004
Looking for 0754333333 in outgoing (domain 192.168.1.254:5060)
list_route: hop: sip:100@192.168.1.107:5060;transport=udp
<— Transmitting (NAT) to 192.168.1.107:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK63c4aa06f1afb7c86;received=192.168.1.107;rport=5060
From: Main Number sip:100@192.168.1.254;tag=ff699c2c47;epid=SC09e274
To: sip:0754333333@192.168.1.254:5060;transport=udp
Call-ID: 8d1ff4449bb75e25
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:0754333333@192.168.1.254:5060
Content-Length: 0
<------------>
– Executing [0754333333@outgoing:1] Macro(“SIP/100-0000003c”, “trunkdial,SIP/romtelecom_voip/0754333333,0373301111”) in new stack
– Executing [s@macro-trunkdial:1] Set(“SIP/100-0000003c”, “CALLERID(all)=0373301111”) in new stack
– Executing [s@macro-trunkdial:2] Dial(“SIP/100-0000003c”, “SIP/romtelecom_voip/0754333333”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 5060
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 80.97.237.2:5060:
INVITE sip:0754333333@voip.clicknet.ro SIP/2.0
Via: SIP/2.0/UDP 172.16.1.100:5060;branch=z9hG4bK709cb7da;rport
Max-Forwards: 70
From: “0373301111” sip:77*******27@voip.clicknet.ro;tag=as64504dfe
To: sip:0754333333@voip.clicknet.ro
Contact: sip:77*******27@172.16.1.100:5060
Call-ID: 131df0286cd3ce622876f86615cc87cb@voip.clicknet.ro
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 27 Dec 2011 23:32:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 916402646 916402646 IN IP4 172.16.1.100
s=Asterisk PBX 1.8.8.0
c=IN IP4 172.16.1.100
t=0 0
m=audio 10350 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
-- Called SIP/romtelecom_voip/0754333333
<— SIP read from UDP:92.81.131.67:5060 —>
INVITE sip:0754333333@voip.clicknet.ro SIP/2.0
Via: SIP/2.0/UDP 92.81.131.67:5060;branch=z9hG4bK709cb7da;rport
Max-Forwards: 70
From: “0373301111” sip:77*******27@voip.clicknet.ro;tag=as64504dfe
To: sip:0754333333@voip.clicknet.ro
Contact: sip:77*******27@92.81.131.67:5060
Call-ID: 131df0286cd3ce622876f86615cc87cb@voip.clicknet.ro
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 27 Dec 2011 23:32:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 916402646 916402646 IN IP4 92.81.131.67
s=Asterisk PBX 1.8.8.0
c=IN IP4 92.81.131.67
t=0 0
m=audio 14688 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
<------------->
— (14 headers 13 lines) —
<— Transmitting (NAT) to 92.81.131.67:5060 —>
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 92.81.131.67:5060;branch=z9hG4bK709cb7da;received=92.81.131.67;rport=5060
From: “0373301111” sip:77*******27@voip.clicknet.ro;tag=as64504dfe
To: sip:0754333333@voip.clicknet.ro;tag=as64504dfe
Call-ID: 131df0286cd3ce622876f86615cc87cb@voip.clicknet.ro
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '131df0286cd3ce622876f86615cc87cb@voip.clicknet.ro’ in 6400 ms (Method: INVITE)
<— SIP read from UDP:92.81.131.67:5060 —>
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 92.81.131.67:5060;branch=z9hG4bK709cb7da;received=92.81.131.67;rport=5060
From: “0373301111” sip:77*******27@voip.clicknet.ro;tag=as64504dfe
To: sip:0754333333@voip.clicknet.ro;tag=as64504dfe
Call-ID: 131df0286cd3ce622876f86615cc87cb@voip.clicknet.ro
CSeq: 102 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------->
— (10 headers 0 lines) —
– Got SIP response 482 “Loop Detected” back from 92.81.131.67:5060
Transmitting (NAT) to 92.81.131.67:5060:
ACK sip:0754333333@voip.clicknet.ro SIP/2.0
Via: SIP/2.0/UDP 172.16.1.100:5060;branch=z9hG4bK709cb7da;rport
Max-Forwards: 70
From: “0373301111” sip:77*******27@voip.clicknet.ro;tag=as64504dfe
To: sip:0754333333@voip.clicknet.ro;tag=as64504dfe
Contact: sip:77*******27@172.16.1.100:5060
Call-ID: 131df0286cd3ce622876f86615cc87cb@voip.clicknet.ro
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
<— SIP read from UDP:92.81.131.67:5060 —>
ACK sip:0754333333@voip.clicknet.ro SIP/2.0
Via: SIP/2.0/UDP 92.81.131.67:5060;branch=z9hG4bK709cb7da;rport
Max-Forwards: 70
From: “0373301111” sip:77*******27@voip.clicknet.ro;tag=as64504dfe
To: sip:0754333333@voip.clicknet.ro;tag=as64504dfe
Contact: sip:77*******27@92.81.131.67:5060
Call-ID: 131df0286cd3ce622876f86615cc87cb@voip.clicknet.ro
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
<------------->
— (10 headers 0 lines) —
– SIP/romtelecom_voip-0000003d is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [s@macro-trunkdial:3] Goto(“SIP/100-0000003c”, “s-CONGESTION,1”) in new stack
– Goto (macro-trunkdial,s-CONGESTION,1)
– Executing [s-CONGESTION@macro-trunkdial:1] NoOp(“SIP/100-0000003c”, “”) in new stack
– Auto fallthrough, channel ‘SIP/100-0000003c’ status is ‘CONGESTION’
<— Reliably Transmitting (NAT) to 192.168.1.107:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.107:5060;branch=z9hG4bK63c4aa06f1afb7c86;received=192.168.1.107;rport=5060
From: Main Number sip:100@192.168.1.254;tag=ff699c2c47;epid=SC09e274
To: sip:0754333333@192.168.1.254:5060;transport=udp;tag=as59d5d295
Call-ID: 8d1ff4449bb75e25
CSeq: 2 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Interworking, unspecified
X-Asterisk-HangupCauseCode: 127
Content-Length: 0
What could it be the cause? Please help.