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arahming

asked on

Voip and Broadvox

Per reqeust I am posting the information provided by broadvox

I was wondering what I would need to do set this up I have done this before and now for some reason can not

using Asterisk with FreePBX

Allow The Media IPs below and all UDP Ports Up To 65535
209.249.3.58       209.249.3.60

Trunk Number
1001342

Turn-up Ticket
347055

Trunk Type
GoLocal

BTN
6032854107

Source IP
69.30.78.72

Password
--NA--

DNS A Record
dfwnx01ga1.pa.broadvox.net

DNS SRV Record
dfwnx01ga1.psrv.broadvox.net

IP Addr 1
209.249.3.59

DIDs
6032854107  BTN
7039144158


Avatar of Kamran Arshad
Kamran Arshad
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Hi,

What is the issue? A problem statement can help better in answering your query.
Avatar of arahming
arahming

ASKER

I had rebuilt an elastix box with asterisk and freepbx and now I can't get it to connect at all had it so I could dial out not in so I rebuilt it again now nothing (guess I'm SOS lose more as I redo)

I know I have to set up the trunk, sip_custom.conf and sip_nat.conf just need to know with what

may internal network is 192.168.0.0/255.255.255.0

no firewall the the box is connected to the LAN through a switch and for eth0 and comcast directly through eth1 all port open will set up firewall later
Avatar of arahming
arahming

ASKER

I was on the phone with broadvox and realized that my softphone could dial out from the pbx but not in. however the outside calls will route to the asterisk pbx but not into the LAN(tested by setting up voice mail which picked up). I assume its routing caused by the configuration of my network I have two network cards in the PBX the WAN interface is connected directly to the NIC via 172.12.23.143 the internal LAN is connected via 192.168.2.100. I assume I need to create a static route telling the box any VOIP calls from 172.12.23.143 need to be routed to 192.168.2.0/255.255.255.0
ASKER CERTIFIED SOLUTION
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koszegi
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Avatar of arahming
arahming

ASKER

I have done all this the problem I am facing is trying to recreate a network one of the other experts told me he/she had. I have one box with asterisk on it with two nic cards. one is connected to the LAN where the sip phone resides. The other is connected to the ISP with its static IP on the card. I can call outside no problem but when I call in I get a fast busy signal and an error 603 allong with the text below. I believe its a routing issue becuase the call is hitting the box and if programed to will go to voice mail any help would be appreciated as I am tired and have been rebilding asterisks over and over again

--- (10 headers 0 lines) ---
173-11-27-133-oregon*CLI>
<--- SIP read from UDP://209.249.3.59:5060 --->
INVITE sip:9713454107@172.23.217.145:5060;transport=udp SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Supported: timer, 100rel
To: <sip:9713454107@209.249.3.56:5060>
From: "My Name  " <sip:5037234567@209.249.3.59>;tag=3453053515-752135
P-Asserted-Identity:"My Name  "<sip:5037234567@64.152.60.74:5060>
Call-ID: 5108895-3453053515-752092@NXT02.broadvox.net
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 209.249.3.59:5060;branch=z9hG4bKeb469130d6618a50676a899df6534177
Contact: <sip:5037234567@209.249.3.59:5060>
Call-Info: <sip:209.249.3.59>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 249

v=0
o=NXT02 6744 21046 IN IP4 209.249.3.59
s=sip call
c=IN IP4 209.249.3.60
t=0 0
m=audio 18764 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
--- (15 headers 12 lines) ---
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
Sending to 209.249.3.59 : 5060 (NAT)
Using INVITE request as basis request - 5108895-3453053515-752092@NXT02.broadvox.net
Found peer 'Broadvox' for '5037234567' from 209.249.3.59:5060
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 209.249.3.60:18764
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x104 (ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 209.249.3.60:18764
Looking for 9713454107 in from-trunk (domain 172.23.217.145)
list_route: hop: <sip:5037234567@209.249.3.59:5060>

<--- Transmitting (NAT) to 209.249.3.59:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 209.249.3.59:5060;branch=z9hG4bKeb469130d6618a50676a899df6534177;received=209.249.3.59
From: "My Name  " <sip:5037234567@209.249.3.59>;tag=3453053515-752135
To: <sip:9713454107@209.249.3.56:5060>
Call-ID: 5108895-3453053515-752092@NXT02.broadvox.net
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:9713454107@172.23.217.145>
Content-Length: 0


<------------>
    -- Executing [9713454107@from-trunk:1] Set("SIP/Broadvox-09daedc0", "__FROM_DID=9713454107") in new stack
    -- Executing [9713454107@from-trunk:2] Gosub("SIP/Broadvox-09daedc0", "app-blacklist-check,s,1") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("SIP/Broadvox-09daedc0", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Return("SIP/Broadvox-09daedc0", "") in new stack
    -- Executing [9713454107@from-trunk:3] ExecIf("SIP/Broadvox-09daedc0", "0 ?Set(CALLERID(name)=5037234567)") in new stack
    -- Executing [9713454107@from-trunk:4] Set("SIP/Broadvox-09daedc0", "FAX_RX=disabled") in new stack
    -- Executing [9713454107@from-trunk:5] Set("SIP/Broadvox-09daedc0", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [9713454107@from-trunk:6] Set("SIP/Broadvox-09daedc0", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [9713454107@from-trunk:7] Goto("SIP/Broadvox-09daedc0", "from-did-direct,2000,1") in new stack
    -- Goto (from-did-direct,2000,1)
    -- Executing [2000@from-did-direct:1] Set("SIP/Broadvox-09daedc0", "__RINGTIMER=5") in new stack
    -- Executing [2000@from-did-direct:2] Macro("SIP/Broadvox-09daedc0", "exten-vm,2000,2000") in new stack
    -- Executing [s@macro-exten-vm:1] Macro("SIP/Broadvox-09daedc0", "user-callerid") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/Broadvox-09daedc0", "AMPUSER=5037234567") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/Broadvox-09daedc0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/Broadvox-09daedc0", "1?Set(REALCALLERIDNUM=5037234567)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/Broadvox-09daedc0", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/Broadvox-09daedc0", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/Broadvox-09daedc0", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/Broadvox-09daedc0", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("SIP/Broadvox-09daedc0", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("SIP/Broadvox-09daedc0", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/Broadvox-09daedc0", "Using CallerID "My Name  " <5037234567>") in new stack
    -- Executing [s@macro-exten-vm:2] Set("SIP/Broadvox-09daedc0", "RingGroupMethod=none") in new stack
    -- Executing [s@macro-exten-vm:3] Set("SIP/Broadvox-09daedc0", "VMBOX=2000") in new stack
    -- Executing [s@macro-exten-vm:4] Set("SIP/Broadvox-09daedc0", "EXTTOCALL=2000") in new stack
    -- Executing [s@macro-exten-vm:5] Set("SIP/Broadvox-09daedc0", "CFUEXT=") in new stack
    -- Executing [s@macro-exten-vm:6] Set("SIP/Broadvox-09daedc0", "CFBEXT=") in new stack
    -- Executing [s@macro-exten-vm:7] Set("SIP/Broadvox-09daedc0", "RT=5") in new stack
    -- Executing [s@macro-exten-vm:8] Macro("SIP/Broadvox-09daedc0", "record-enable,2000,IN") in new stack
    -- Executing [s@macro-record-enable:1] GotoIf("SIP/Broadvox-09daedc0", "1?check") in new stack
    -- Goto (macro-record-enable,s,4)
    -- Executing [s@macro-record-enable:4] AGI("SIP/Broadvox-09daedc0", "recordingcheck,20090603-143348,1244064828.32") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
 recordingcheck,20090603-143348,1244064828.32: Failed to execute '/var/lib/asterisk/agi-bin/recordingcheck': Permission denied
    -- Executing [s@macro-record-enable:5] MacroExit("SIP/Broadvox-09daedc0", "") in new stack
    -- Executing [s@macro-exten-vm:9] Macro("SIP/Broadvox-09daedc0", "dial,5,tr,2000") in new stack
    -- Executing [s@macro-dial:1] GotoIf("SIP/Broadvox-09daedc0", "1?dial") in new stack
    -- Goto (macro-dial,s,3)
    -- Executing [s@macro-dial:3] AGI("SIP/Broadvox-09daedc0", "dialparties.agi") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
 dialparties.agi: Failed to execute '/var/lib/asterisk/agi-bin/dialparties.agi': Permission denied
    -- Executing [s@macro-dial:4] NoOp("SIP/Broadvox-09daedc0", "Returned from dialparties with no extensions to call and DIALSTATUS: ") in new stack
    -- Executing [s@macro-exten-vm:10] GotoIf("SIP/Broadvox-09daedc0", "0?exit,return") in new stack
    -- Executing [s@macro-exten-vm:11] Set("SIP/Broadvox-09daedc0", "SV_DIALSTATUS=") in new stack
    -- Executing [s@macro-exten-vm:12] GosubIf("SIP/Broadvox-09daedc0", "0?docfu,1") in new stack
    -- Executing [s@macro-exten-vm:13] GosubIf("SIP/Broadvox-09daedc0", "0?docfb,1") in new stack
    -- Executing [s@macro-exten-vm:14] Set("SIP/Broadvox-09daedc0", "DIALSTATUS=") in new stack
    -- Executing [s@macro-exten-vm:15] NoOp("SIP/Broadvox-09daedc0", "Voicemail is '2000'") in new stack
    -- Executing [s@macro-exten-vm:16] GotoIf("SIP/Broadvox-09daedc0", "0?s-,1") in new stack
    -- Executing [s@macro-exten-vm:17] NoOp("SIP/Broadvox-09daedc0", "Sending to Voicemail box 2000") in new stack
    -- Executing [s@macro-exten-vm:18] Macro("SIP/Broadvox-09daedc0", "vm,2000,,") in new stack
    -- Executing [s@macro-vm:1] Macro("SIP/Broadvox-09daedc0", "user-callerid,SKIPTTL") in new stack
    -- Executing [s@macro-user-callerid:1] Set("SIP/Broadvox-09daedc0", "AMPUSER=5037234567") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("SIP/Broadvox-09daedc0", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("SIP/Broadvox-09daedc0", "0?Set(REALCALLERIDNUM=5037234567)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("SIP/Broadvox-09daedc0", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("SIP/Broadvox-09daedc0", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("SIP/Broadvox-09daedc0", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
    -- Executing [s@macro-user-callerid:10] GotoIf("SIP/Broadvox-09daedc0", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("SIP/Broadvox-09daedc0", "Using CallerID "My Name  " <5037234567>") in new stack
    -- Executing [s@macro-vm:2] Set("SIP/Broadvox-09daedc0", "VMGAIN=""") in new stack
    -- Executing [s@macro-vm:3] GotoIf("SIP/Broadvox-09daedc0", "1?vmx,1") in new stack
    -- Goto (macro-vm,vmx,1)
    -- Executing [vmx@macro-vm:1] GotoIf("SIP/Broadvox-09daedc0", "0?s-,1") in new stack
    -- Executing [vmx@macro-vm:2] Set("SIP/Broadvox-09daedc0", "MODE=unavail") in new stack
    -- Executing [vmx@macro-vm:3] GotoIf("SIP/Broadvox-09daedc0", "1?notdirect") in new stack
    -- Goto (macro-vm,vmx,5)
    -- Executing [vmx@macro-vm:5] NoOp("SIP/Broadvox-09daedc0", "Checking if ext 2000 is enabled: ") in new stack
    -- Executing [vmx@macro-vm:6] GotoIf("SIP/Broadvox-09daedc0", "1?s-,1") in new stack
    -- Goto (macro-vm,s-,1)
    -- Executing [2000@from-did-direct:3] Goto("SIP/Broadvox-09daedc0", "vmret,1") in new stack
    -- Goto (from-did-direct,vmret,1)
    -- Executing [vmret@from-did-direct:1] GotoIf("SIP/Broadvox-09daedc0", "0?playret") in new stack
    -- Executing [vmret@from-did-direct:2] Hangup("SIP/Broadvox-09daedc0", "") in new stack
  == Spawn extension (from-did-direct, vmret, 2) exited non-zero on 'SIP/Broadvox-09daedc0'
Scheduling destruction of SIP dialog '5108895-3453053515-752092@NXT02.broadvox.net' in 6400 ms (Method: INVITE)
173-11-27-133-oregon*CLI>
<--- Reliably Transmitting (NAT) to 209.249.3.59:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 209.249.3.59:5060;branch=z9hG4bKeb469130d6618a50676a899df6534177;received=209.249.3.59
From: "My Name  " <sip:5037234567@209.249.3.59>;tag=3453053515-752135
To: <sip:9713454107@209.249.3.56:5060>;tag=as7b278e03
Call-ID: 5108895-3453053515-752092@NXT02.broadvox.net
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


<------------>
173-11-27-133-oregon*CLI>
<--- SIP read from UDP://209.249.3.59:5060 --->
ACK sip:9713454107@172.23.217.145:5060;transport=udp SIP/2.0
Max-Forwards: 70
To: <sip:9713454107@209.249.3.56:5060>;tag=as7b278e03
From: "My Name  " <sip:5037234567@209.249.3.59>;tag=3453053515-752135
Call-ID: 5108895-3453053515-752092@NXT02.broadvox.net
CSeq: 1 ACK
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Via: SIP/2.0/UDP 209.249.3.59:5060;branch=z9hG4bKeb469130d6618a50676a899df6534177
Contact: <sip:5037234567@209.249.3.59:5060>
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '5108890-3453053514-777640@NXT02.broadvox.net' Method: ACK
Really destroying SIP dialog '5108892-3453053514-995261@NXT02.broadvox.net' Method: ACK
Really destroying SIP dialog '5108893-3453053515-232559@NXT02.broadvox.net' Method: ACK
Really destroying SIP dialog '5108895-3453053515-752092@NXT02.broadvox.net' Method: ACK
Reliably Transmitting (NAT) to 209.249.3.59:5060:
OPTIONS sip:209.249.3.59 SIP/2.0
Via: SIP/2.0/UDP 172.23.217.145:5060;branch=z9hG4bK546a9b67;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.23.217.145>;tag=as46ac15a0
To: <sip:209.249.3.59>
Contact: <sip:Unknown@172.23.217.145>
Call-ID: 363e13ba05cfcdce23feabbc267c9df9@172.23.217.145
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.1.0
Date: Wed, 03 Jun 2009 21:33:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
173-11-27-133-oregon*CLI>
<--- SIP read from UDP://209.249.3.59:5060 --->
SIP/2.0 200 OK
Session-Expires: 3600
Require: timer
Via: SIP/2.0/UDP 172.23.217.145:5060;branch=z9hG4bK546a9b67;rport
To: <sip:209.249.3.59>;tag=3453053526-53403
From: "Unknown" <sip:Unknown@172.23.217.145>;tag=as46ac15a0
Call-ID: 363e13ba05cfcdce23feabbc267c9df9@172.23.217.145
CSeq: 102 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:209.249.3.59:5060>
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '363e13ba05cfcdce23feabbc267c9df9@172.23.217.145' Method: OPTIONS
Reliably Transmitting (NAT) to 209.249.3.59:5060:
OPTIONS sip:209.249.3.59 SIP/2.0
Via: SIP/2.0/UDP 172.23.217.145:5060;branch=z9hG4bK27de4b98;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@172.23.217.145>;tag=as070193bd
To: <sip:209.249.3.59>
Contact: <sip:Unknown@172.23.217.145>
Call-ID: 2b121cbb61d5b8bb1a2bc7491efa93db@172.23.217.145
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.1.0
Date: Wed, 03 Jun 2009 21:33:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0


---
173-11-27-133-oregon*CLI>
<--- SIP read from UDP://209.249.3.59:5060 --->
SIP/2.0 200 OK
Session-Expires: 3600
Require: timer
Via: SIP/2.0/UDP 172.23.217.145:5060;branch=z9hG4bK27de4b98;rport
To: <sip:209.249.3.59>;tag=3453053526-350827
From: "Unknown" <sip:Unknown@172.23.217.145>;tag=as070193bd
Call-ID: 2b121cbb61d5b8bb1a2bc7491efa93db@172.23.217.145
CSeq: 102 OPTIONS
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE
Contact: <sip:209.249.3.59:5060>
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '2b121cbb61d5b8bb1a2bc7491efa93db@172.23.217.145' Method: OPTIONS
173-11-27-133-oregon*CLI>
Avatar of koszegi
koszegi
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Verify what context registration of the SIP peer.  Paste your sip.conf here.  Are you using asterisk or trixbox?