vicidial.org

by **dajaz27** » Fri Aug 19, 2011 5:52 am

Go Autodial CE 2.1 ISO install

[Aug 19 06:49:18] -- Executing [918135627155@default:1] AGI("SIP/8001-00000002", "agi://127.0.0.1:4577/call_log") in new stack
[Aug 19 06:49:18] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Aug 19 06:49:18] -- Executing [918135627155@default:2] Dial("SIP/8001-00000002", "SIP/18135627155@goautodial||tTo") in new stack
[Aug 19 06:49:18] WARNING[4301]: chan_sip.c:3225 create_addr: No such host: goautodial
[Aug 19 06:49:18] WARNING[4301]: app_dial.c:1310 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Aug 19 06:49:18] == Everyone is busy/congested at this time (1:0/0/1)
[Aug 19 06:49:18] -- Executing [918135627155@default:3] Hangup("SIP/8001-00000002", "") in new stack
[Aug 19 06:49:18] == Spawn extension (default, 918135627155, 3) exited non-zero on 'SIP/8001-00000002'
[Aug 19 06:49:18] -- Executing [h@default:1] DeadAGI("SIP/8001-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Aug 19 06:49:18] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Aug 19 06:50:01] == Parsing '/etc/asterisk/manager.conf': [Aug 19 06:50:01] Found
[Aug 19 06:50:01] == Manager 'sendcron' logged on from 127.0.0.1

by **striker** » Sat Aug 20, 2011 9:30 am

post your carrier setup and dialplan for outgoing calls

also go to asterisk cli and check whether the trunk is registered

sip show peers
sip show registry

by **williamconley** » Sat Aug 20, 2011 9:56 am

1) You'll need a carrier. Calls aren't free.

2) when you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

this IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation

by **dajaz27** » Sat Aug 20, 2011 11:43 am

I read every sticky there is both related and unrelated to my issue. Sorry about not posting iso information. Anyway here is carrier setup:

[GODOMESTIC1-NYC-OUT]
type=peer
context=GODOMESTIC-INBOUND
host=nyc01-01.fs.broadvox.net
canreinvite=no
[GODOMESTIC1-DFW-OUT]
type=peer
context=GODOMESTIC-INBOUND
host=dfw01-01.fs.broadvox.net
canreinvite=no
[GODOMESTIC1-LAX-OUT]
type=peer
context=GODOMESTIC-INBOUND
host=lax01-01.fs.broadvox.net
canreinvite=no

Carrier Dial plan
[GODOMESTIC1-INBOUND]
include => ROUTES
; This tells Asterisk to route calls based on the TO header field in the
; SIP INVITE instead of the Request-URI. This is needed when using
; registration in order to properly route calls based on the dialed number
; instead of the BTN. Fusion trunks support sending the dialed number in
; either the Request-URI or the To-URI for registered trunks. In either case,
; the dialed number will always be present in the To-URI, so we recommend you
; use this method of routing your calls.
[ROUTES]
exten=>s,1,Goto(DIDS,${SIP_HEADER(TO):5:10},1)
exten=>s,n,Goto(AutoAttendant,s,1)
[DIDS]
exten => 5555398198,1,Dial(SIP/${EXTEN}@SIPURA1-OUT)
exten => 5555398199,1,Dial(SIP/${EXTEN}@SIPURA2-OUT)
; This is where you would define your auto-attendant
[AutoAttendant]
;exten=>s,1,Answer
;exten=>s,n,Playback(aa-greeting)
; Any calls hitting sipura-inbound are dialed from our phone, so they are
; actually outbound calls. Here we tell how to send the calls out to
; Broadvox, including whether to use GO!Local or GO!Domestic.
[SIPURA-INBOUND]
; We are coming in from a phone, so include our extensions list
include => extensions
; GO!Local should come first to catch toll-free and local calling rules
include => GOANYWHERE1-OUT-PLAN
include => GODOMESTIC1-OUT-PLAN

here is my dial plan:
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@goautodial,,tTo)
exten => _91XXXXXXXXXX,3,Hangup

Here is my trunk configuration:

[broadvox]
disallow=all
allow=all
type=peer
host=208.93.224.230
dtmfmode=rfc2833
context=trunkinbound
nat=yes

by **striker** » Sat Aug 20, 2011 12:21 pm

little bit tuff to understand what u have posted
but as of now do the below changes

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@goautodial,,tTo)
exten => _91XXXXXXXXXX,3,Hangup

to

exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@broadvox,,tTo)
exten => _91XXXXXXXXXX,3,Hangup

make sure the broadvox carrier is registered properly
go to asteriskcli cmd: asterisk -vvvvr
go>sip show peers --it should show ok for the broadvocie

by **dajaz27** » Sat Aug 20, 2011 12:25 pm

It's IP to IP no registration. Thanks for the help I will try it now.

by **dajaz27** » Sat Aug 20, 2011 12:31 pm

gonna try to do some call captures post the output

by **williamconley** » Sat Aug 20, 2011 1:16 pm

your dial string is off.

PROTOCOL/CONTEXT/NUMBER

or

PROTOCOL/NUMBER@CONTEXT

any of these items can be a variable or the real deal.

goautodial and broadvox are not sip contexts, so neither will have "meaning" when the system tries to execute.

try GODOMESTIC1-NYC-OUT which is an actual sip context.

we usually recommend using a variable instead, which would contain both protocol and context (to allow changing to a new provider merely by altering the variable name).

by **dajaz27** » Mon Aug 22, 2011 8:53 am

[GODOMESTIC1]
disallow=all
allow=ulaw
allow=g729
type=peer
dtmfmode=rfc2833
context=trunkinbound
host=208.93.227.214
canreinvite=no
nat=yes

Global String: GODOMESTIC1=SIP/godomestic1

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${GODOMESTIC1:5060}/${EXTEN:2},60,tTor)
exten => _91NXXNXXXXXX,3,Hangup

by **williamconley** » Mon Aug 22, 2011 7:49 pm

Now you're just confused.

in the globals string: TRUNKVARIABLE=SIP/carriername

in the account entry: [carriername]

in the dial string: ${TRUNKVARIABLE}

You seem to have the carriername as the name of the variable which leads to a nonexistent sip context. And you broke your variable by trying to specify the port number ... which is best left alone since it is the default port, and would be better done with a "port=5060" in the SIP context if you really wanted to do it anyway.

[godomestic1]
disallow=all
allow=ulaw
allow=g729
type=peer
dtmfmode=rfc2833
context=trunkinbound
host=208.93.227.214
canreinvite=no
nat=yes

Global String: DOMESTICTRUNK1=SIP/godomestic1

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${DOMESTICTRUNK1}/${EXTEN:2},60,tTor)
exten => _91NXXNXXXXXX,3,Hangup

When dialed, the trunk's global variable will now be replaced with the value, and you'll end up with SIP/godomestic1/${EXTEN:2} where exten:2 will be replaced by the extension with the "91" removed. Then the call will be executed by initiating the SIP protocol, looking for the godomestic1 sip context (defined above) and then calling that provider with a request to dial the extension supplied. Let's see how that works. 8-)

by **dajaz27** » Tue Aug 23, 2011 9:23 am

I can not make calls. Calls come into the system but are not routed properly (so no phone rings). Which leads me to believe there is a problem with either my trunk and or my dial plan. So how do I fix either my trunk or dial plan?

by **striker** » Tue Aug 23, 2011 10:27 am

hi

post the asterisk cli results while the calls lands into your system, while will helpful to figure out the issue

by **dajaz27** » Tue Aug 23, 2011 2:37 pm

User-Agent: Broadvox Fusion
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 283
P-Asserted-Identity: "8135627155" <sip:8135627155@208.93.227.214>

v=0
o=Sonus_UAC 22372 4413579646750769704 IN IP4 10.128.34.100
s=SIP Media Capabilities
c=IN IP4 64.152.60.71
t=0 0
m=audio 21334 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
[Aug 23 15:31:15] --- (17 headers 12 lines) ---
[Aug 23 15:31:15] Ignoring this INVITE request
[Aug 23 15:31:15] NOTICE[3018]: chan_sip.c:15793 handle_request_invite: Unable to create/find SIP channel for this INVITE
[Aug 23 15:31:15]
<--- Transmitting (NAT) to 208.93.227.214:5060 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 208.93.227.214;branch=z9hG4bKmjc6XaBgmFD5m;received=208.93.227.214;rport=5060
From: "8135627155" <sip:8135627155@208.93.227.214>;tag=4UHZvB83U28tN
To: <sip:8775189339@71.180.173.37:5060>;tag=as46688799
Call-ID: 5d0ccdee-4861-122f-66b6-f04da23d7069
CSeq: 16727885 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
ontent-Length: 0

<------------>
[Aug 23 15:31:15] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7c44510 (len 459) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 23 15:31:15] Scheduling destruction of SIP dialog '5d0ccdee-4861-122f-66b6-f04da23d7069' in 6400 ms (Method: INVITE)
[Aug 23 15:31:19] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 23 15:31:19] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK50b21691;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as490261d7
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 139a56e31aae4698456ff8432d5bb795@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 23 Aug 2011 19:31:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 23 15:31:19] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7d2076c (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 23 15:31:23] Really destroying SIP dialog '139a56e31aae4698456ff8432d5bb795@127.0.0.1' Method: OPTIONS
[Aug 23 15:31:33] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 23 15:31:33] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK7f029b8d;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as488b6814
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 1f2c2ba47c8481230b36f06d3582969c@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 23 Aug 2011 19:31:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 23 15:31:33] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7d2076c (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 23 15:31:33] Reliably Transmitting (NAT) to 192.168.1.142:19138:
OPTIONS sip:8001@192.168.1.142:19138;rinstance=6c54c55b48685446;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK340b2477;rport
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as18cac523
To: <sip:8001@192.168.1.142:19138;rinstance=6c54c55b48685446;cpd=on>
Contact: <sip:asterisk@192.168.1.2>
Call-ID: 266f176b0a4c2536097ca64066f391f3@192.168.1.2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 23 Aug 2011 19:31:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 23 15:31:33]
<--- SIP read from 192.168.1.142:19138 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK340b2477;rport=5060
Contact: <sip:192.168.1.142:19138>
To: <sip:8001@192.168.1.142:19138;rinstance=6c54c55b48685446;cpd=on>;tag=cd3f6e1d
From: "asterisk"<sip:asterisk@192.168.1.2>;tag=as18cac523
Call-ID: 266f176b0a4c2536097ca64066f391f3@192.168.1.2
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

<------------->
[Aug 23 15:31:33] --- (13 headers 0 lines) ---
[Aug 23 15:31:33] Really destroying SIP dialog '266f176b0a4c2536097ca64066f391f3@192.168.1.2' Method: OPTIONS
[Aug 23 15:31:33]
<--- SIP read from 192.168.1.142:19138 --->

<------------->
[Aug 23 15:31:37] Really destroying SIP dialog '1f2c2ba47c8481230b36f06d3582969c@127.0.0.1' Method: OPTIONS
[Aug 23 15:31:47] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 23 15:31:47] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK67b1a276;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as31da3cac
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 1fd69b1c484d0d1e7b6218a83be2149c@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 23 Aug 2011 19:31:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 23 15:31:47] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7d2076c (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 23 15:31:51] Really destroying SIP dialog '1fd69b1c484d0d1e7b6218a83be2149c@127.0.0.1' Method: OPTIONS
[Aug 23 15:32:01] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 23 15:32:01] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK235a1ea8;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as62a5ffcc
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 07da445a403caa80774e22fc650a37b0@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 23 Aug 2011 19:32:01 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 23 15:32:01] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7d2076c (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 23 15:32:01] == Parsing '/etc/asterisk/manager.conf': [Aug 23 15:32:01] Found
[Aug 23 15:32:01] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 23 15:32:01] == Parsing '/etc/asterisk/manager.conf': [Aug 23 15:32:01] Found
[Aug 23 15:32:01] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 23 15:32:01] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 23 15:32:03] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 23 15:32:03]
<--- SIP read from 192.168.1.142:19138 --->

<------------->
[Aug 23 15:32:05] Really destroying SIP dialog '07da445a403caa80774e22fc650a37b0@127.0.0.1' Method: OPTIONS
[Aug 23 15:32:06] == Parsing '/etc/asterisk/manager.conf': [Aug 23 15:32:06] Found
[Aug 23 15:32:06] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 23 15:32:06] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 23 15:32:15] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 23 15:32:15] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK3e614b34;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as22558c91
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 194c215055ff397d7b5b74326c2661a2@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 23 Aug 2011 19:32:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 23 15:32:15] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7d2076c (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 23 15:32:19] Really destroying SIP dialog '194c215055ff397d7b5b74326c2661a2@127.0.0.1' Method: OPTIONS
go*CLI>

by **williamconley** » Tue Aug 23, 2011 10:00 pm

dajaz27 wrote:I can not make calls. Calls come into the system but are not routed properly (so no phone rings). Which leads me to believe there is a problem with either my trunk and or my dial plan. So how do I fix either my trunk or dial plan?

We were working before on your outbound. Did that get workin'?

chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7d2076c (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable

I think you have a problem with your network that may be completely unrelated to configuration. Are you behind a firewall? Are you using unusual networking? (Multicast? IPv6?)

by **dajaz27** » Wed Aug 24, 2011 6:20 am

Nothing is blocking traffic.

[Aug 24 07:15:34]
<--- SIP read from 192.168.1.142:19138 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK62becff4;rport=5060
Contact: <sip:192.168.1.142:19138>
To: <sip:8001@192.168.1.142:19138;rinstance=6c54c55b48685446;cpd=on>;tag=cb019113
From: "asterisk"<sip:asterisk@192.168.1.2>;tag=as07daf077
Call-ID: 509af0801e212a2a6ee335f7259b0322@192.168.1.2
CSeq: 102 OPTIONS
Accept: application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

<------------->
[Aug 24 07:15:34] --- (13 headers 0 lines) ---
[Aug 24 07:15:34] Really destroying SIP dialog '509af0801e212a2a6ee335f7259b0322@192.168.1.2' Method: OPTIONS
[Aug 24 07:15:34]
<--- SIP read from 192.168.1.142:19138 --->

<------------->
[Aug 24 07:15:35] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 24 07:15:35] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK35efe328;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as2cfce397
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 7fef11941c7d944e21aefc7d58023f0f@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 24 Aug 2011 11:15:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 24 07:15:35] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0x8ba4dac (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 24 07:15:37]
<--- SIP read from 192.168.1.142:19138 --->
SUBSCRIBE sip:asterisk@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.142:19138;branch=z9hG4bK-d8754z-f51a051e5863dd0e-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8001@192.168.1.142:19138>
To: "8001"<sip:8001@192.168.1.2>;tag=as4e360786
From: "8001"<sip:8001@192.168.1.2>;tag=79f4bd8c
Call-ID: N2M3ZDI3ZjZmOThjMTZhNDBmNjY0ZjU4NmFkZDg1OTQ.
CSeq: 1229 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="8001",realm="asterisk",nonce="1d161e6d",uri="sip:asterisk@192.168.1.2",response="e18e0a97a286b0f50ff7ed39e99eb771",algorithm=MD5
Event: message-summary
Content-Length: 0

<------------->
[Aug 24 07:15:37] --- (14 headers 0 lines) ---
[Aug 24 07:15:37] Found peer '8001'
[Aug 24 07:15:37] NOTICE[3018]: chan_sip.c:9330 check_auth: Correct auth, but based on stale nonce received from '"8001"<sip:8001@192.168.1.2>;tag=79f4bd8c'
[Aug 24 07:15:37]
<--- Transmitting (NAT) to 192.168.1.142:19138 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.142:19138;branch=z9hG4bK-d8754z-f51a051e5863dd0e-1---d8754z-;received=192.168.1.142;rport=19138
From: "8001"<sip:8001@192.168.1.2>;tag=79f4bd8c
To: "8001"<sip:8001@192.168.1.2>;tag=as4e360786
Call-ID: N2M3ZDI3ZjZmOThjMTZhNDBmNjY0ZjU4NmFkZDg1OTQ.
CSeq: 1229 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5523ef1a", stale=true
Content-Length: 0

<------------>
[Aug 24 07:15:37] Scheduling destruction of SIP dialog 'N2M3ZDI3ZjZmOThjMTZhNDBmNjY0ZjU4NmFkZDg1OTQ.' in 6400 ms (Method: SUBSCRIBE)
[Aug 24 07:15:38]
<--- SIP read from 192.168.1.142:19138 --->
SUBSCRIBE sip:asterisk@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.142:19138;branch=z9hG4bK-d8754z-d48891b04732ce21-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8001@192.168.1.142:19138>
To: "8001"<sip:8001@192.168.1.2>;tag=as4e360786
From: "8001"<sip:8001@192.168.1.2>;tag=79f4bd8c
Call-ID: N2M3ZDI3ZjZmOThjMTZhNDBmNjY0ZjU4NmFkZDg1OTQ.
CSeq: 1230 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="8001",realm="asterisk",nonce="5523ef1a",uri="sip:asterisk@192.168.1.2",response="fa4f396d4021de0c4c602153473f50dc",algorithm=MD5
Event: message-summary
Content-Length: 0

<------------->
[Aug 24 07:15:38] --- (14 headers 0 lines) ---
[Aug 24 07:15:38] Found peer '8001'
[Aug 24 07:15:38] Scheduling destruction of SIP dialog 'N2M3ZDI3ZjZmOThjMTZhNDBmNjY0ZjU4NmFkZDg1OTQ.' in 310000 ms (Method: SUBSCRIBE)
[Aug 24 07:15:38]
<--- Transmitting (NAT) to 192.168.1.142:19138 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.142:19138;branch=z9hG4bK-d8754z-d48891b04732ce21-1---d8754z-;received=192.168.1.142;rport=19138
From: "8001"<sip:8001@192.168.1.2>;tag=79f4bd8c
To: "8001"<sip:8001@192.168.1.2>;tag=as4e360786
Call-ID: N2M3ZDI3ZjZmOThjMTZhNDBmNjY0ZjU4NmFkZDg1OTQ.
CSeq: 1230 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 300
Contact: <sip:asterisk@192.168.1.2>;expires=300
Content-Length: 0

<------------>
[Aug 24 07:15:38] Reliably Transmitting (NAT) to 192.168.1.142:19138:
NOTIFY sip:8001@192.168.1.142:19138;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3bcdb2d2;rport
Route: <sip:8001@192.168.1.142:19138>
From: "asterisk" <sip:asterisk@192.168.1.2>;tag=as4e360786
To: <sip:8001@192.168.1.142:19138;cpd=on>;tag=79f4bd8c
Contact: <sip:asterisk@192.168.1.2>
Call-ID: N2M3ZDI3ZjZmOThjMTZhNDBmNjY0ZjU4NmFkZDg1OTQ.
CSeq: 720 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 91

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.1.2
Voice-Message: 0/0 (0/0)

---
[Aug 24 07:15:38]
<--- SIP read from 192.168.1.142:19138 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.2:5060;branch=z9hG4bK3bcdb2d2;rport=5060
Contact: <sip:8001@192.168.1.142:19138>
To: <sip:8001@192.168.1.142:19138;cpd=on>;tag=79f4bd8c
From: "asterisk"<sip:asterisk@192.168.1.2>;tag=as4e360786
Call-ID: N2M3ZDI3ZjZmOThjMTZhNDBmNjY0ZjU4NmFkZDg1OTQ.
CSeq: 720 NOTIFY
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 0

<------------->
[Aug 24 07:15:38] --- (9 headers 0 lines) ---
[Aug 24 07:15:39] Really destroying SIP dialog '7fef11941c7d944e21aefc7d58023f0f@127.0.0.1' Method: OPTIONS
[Aug 24 07:15:40] Really destroying SIP dialog 'NzIwMjMxZDIzZDlhZDMzZjE2ZjVmZjFhYmI0NDA5YTM.' Method: ACK
[Aug 24 07:15:49] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 24 07:15:49] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK361c58ac;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as22aef97a
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 7c0e2f6c58425cfe5219542b3320aade@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 24 Aug 2011 11:15:49 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 24 07:15:49] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0x8ba4dac (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 24 07:15:53] Really destroying SIP dialog '7c0e2f6c58425cfe5219542b3320aade@127.0.0.1' Method: OPTIONS
[Aug 24 07:16:01] == Parsing '/etc/asterisk/manager.conf': [Aug 24 07:16:01] Found
[Aug 24 07:16:01] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 24 07:16:01] == Parsing '/etc/asterisk/manager.conf': [Aug 24 07:16:01] Found
[Aug 24 07:16:01] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 24 07:16:01] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 24 07:16:03] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 24 07:16:03] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK1b90186c;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as649522ef
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 3c83db8b295d5ac402762c5077046518@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 24 Aug 2011 11:16:03 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 24 07:16:03] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0x8ba4dac (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 24 07:16:03] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 24 07:16:04]
<--- SIP read from 192.168.1.142:19138 --->

<------------->
[Aug 24 07:16:06] == Parsing '/etc/asterisk/manager.conf': [Aug 24 07:16:06] Found
[Aug 24 07:16:06] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 24 07:16:06] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 24 07:16:07] Really destroying SIP dialog '3c83db8b295d5ac402762c5077046518@127.0.0.1' Method: OPTIONS
[Aug 24 07:16:17] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 24 07:16:17] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK47fa0cd2;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as31069b23
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 2a38dd5b54300abf098aaf5d3d3bc31b@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 24 Aug 2011 11:16:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 24 07:16:17] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0x8ba4dac (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
go*CLI>

by **dajaz27** » Wed Aug 24, 2011 6:30 am

m=audio 22976 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
[Aug 24 07:26:00] --- (17 headers 12 lines) ---
[Aug 24 07:26:00] Ignoring this INVITE request
[Aug 24 07:26:01] == Parsing '/etc/asterisk/manager.conf': [Aug 24 07:26:01] Found
[Aug 24 07:26:01] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 24 07:26:01] == Parsing '/etc/asterisk/manager.conf': [Aug 24 07:26:01] Found
[Aug 24 07:26:01] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 24 07:26:01] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 24 07:26:02]
<--- SIP read from 208.93.227.214:5060 --->
INVITE sip:8775189339@71.180.173.37:5060 SIP/2.0
Via: SIP/2.0/UDP 208.93.227.214;rport;branch=z9hG4bKX1H7NagB4BaKN
Max-Forwards: 69
From: "8134215841" <sip:8134215841@208.93.227.214>;tag=gyD2tHact6Qre
To: <sip:8775189339@71.180.173.37:5060>
Call-ID: bf4f3f8e-48e6-122f-66b6-f04da23d7069
CSeq: 16756529 INVITE
Contact: <sip:8134215841@208.93.227.214:5060>
User-Agent: Broadvox Fusion
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 268
P-Asserted-Identity: "8134215841" <sip:8134215841@208.93.227.214>

v=0
o=Sonus_UAC 23604 4697 IN IP4 10.128.34.100
s=SIP Media Capabilities
c=IN IP4 64.152.60.71
t=0 0
m=audio 27706 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
[Aug 24 07:26:02] --- (17 headers 12 lines) ---
[Aug 24 07:26:02] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 24 07:26:02] Sending to 208.93.227.214 : 5060 (NAT)
[Aug 24 07:26:02] Using INVITE request as basis request - bf4f3f8e-48e6-122f-66b6-f04da23d7069
[Aug 24 07:26:02] Found peer 'godomestic1'
[Aug 24 07:26:02] Found RTP audio format 0
[Aug 24 07:26:02] Found RTP audio format 18
[Aug 24 07:26:02] Found RTP audio format 101
[Aug 24 07:26:02] Found audio description format PCMU for ID 0
[Aug 24 07:26:02] Found audio description format G729 for ID 18
[Aug 24 07:26:02] Found audio description format telephone-event for ID 101
[Aug 24 07:26:02] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
[Aug 24 07:26:02] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Aug 24 07:26:02] Peer audio RTP is at port 64.152.60.71:27706
[Aug 24 07:26:02] Looking for 8775189339 in trunkinbound (domain 71.180.173.37)
[Aug 24 07:26:02] list_route: hop: <sip:8134215841@208.93.227.214:5060>
[Aug 24 07:26:02]
<--- Transmitting (NAT) to 208.93.227.214:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 208.93.227.214;branch=z9hG4bKX1H7NagB4BaKN;received=208.93.227.214;rport=5060
From: "8134215841" <sip:8134215841@208.93.227.214>;tag=gyD2tHact6Qre
To: <sip:8775189339@71.180.173.37:5060>
Call-ID: bf4f3f8e-48e6-122f-66b6-f04da23d7069
CSeq: 16756529 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:8775189339@0.0.0.0>
Content-Length: 0

<------------>
[Aug 24 07:26:02] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7c444e0 (len 474) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 24 07:26:02] -- Executing [8775189339@trunkinbound:1] AGI("SIP/godomestic1-00000019", "agi-DID_route.agi") in new stack
[Aug 24 07:26:02] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-DID_route.agi
[Aug 24 07:26:02] -- AGI Script Executing Application: (Monitor) Options: (wav|/var/spool/asterisk/monitor/MIX/20110824072602_8775189339_8134215841)
[Aug 24 07:26:02] ERROR[30911]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Aug 24 07:26:02] ERROR[30911]: utils.c:967 ast_carefulwrite: write() returned error: Broken pipe
[Aug 24 07:26:02] -- AGI Script agi-DID_route.agi completed, returning 0
[Aug 24 07:26:02] -- Executing [99909*3***DID@default:1] Answer("SIP/godomestic1-00000019", "") in new stack
[Aug 24 07:26:02] Audio is at 0.0.0.0 port 16794
[Aug 24 07:26:02] Adding codec 0x4 (ulaw) to SDP
[Aug 24 07:26:02] Adding codec 0x100 (g729) to SDP
[Aug 24 07:26:02] Adding non-codec 0x1 (telephone-event) to SDP
[Aug 24 07:26:02]
<--- Reliably Transmitting (NAT) to 208.93.227.214:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 208.93.227.214;branch=z9hG4bKX1H7NagB4BaKN;received=208.93.227.214;rport=5060
From: "8134215841" <sip:8134215841@208.93.227.214>;tag=gyD2tHact6Qre
To: <sip:8775189339@71.180.173.37:5060>;tag=as10531efc
Call-ID: bf4f3f8e-48e6-122f-66b6-f04da23d7069
CSeq: 16756529 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
ontact: <sip:8775189339@0.0.0.0>
Content-Type: application/sdp
Content-Length: 248

v=0
o=root 2920 2920 IN IP4 0.0.0.0
s=session
c=IN IP4 0.0.0.0
t=0 0
m=audio 16794 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Aug 24 07:26:02] WARNING[30911]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0x8ba7404 (len 766) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 24 07:26:02] == Spawn extension (default, 99909*3***DID, 1) exited non-zero on 'SIP/godomestic1-00000019'
[Aug 24 07:26:02] -- Executing [h@default:1] DeadAGI("SIP/godomestic1-00000019", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Aug 24 07:26:02] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Aug 24 07:26:02] Scheduling destruction of SIP dialog 'bf4f3f8e-48e6-122f-66b6-f04da23d7069' in 6400 ms (Method: INVITE)
[Aug 24 07:26:03]
<--- SIP read from 208.93.227.214:5060 --->
INVITE sip:8775189339@71.180.173.37:5060 SIP/2.0
Via: SIP/2.0/UDP 208.93.227.214;rport;branch=z9hG4bKX1H7NagB4BaKN
Max-Forwards: 69
From: "8134215841" <sip:8134215841@208.93.227.214>;tag=gyD2tHact6Qre
To: <sip:8775189339@71.180.173.37:5060>
Call-ID: bf4f3f8e-48e6-122f-66b6-f04da23d7069
CSeq: 16756529 INVITE
Contact: <sip:8134215841@208.93.227.214:5060>
User-Agent: Broadvox Fusion
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, refer
Privacy: none
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 268
P-Asserted-Identity: "8134215841" <sip:8134215841@208.93.227.214>

v=0
o=Sonus_UAC 23604 4697 IN IP4 10.128.34.100
s=SIP Media Capabilities
c=IN IP4 64.152.60.71
t=0 0
m=audio 27706 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
[Aug 24 07:26:03] --- (17 headers 12 lines) ---
[Aug 24 07:26:03] Ignoring this INVITE request
[Aug 24 07:26:03] NOTICE[3018]: chan_sip.c:15793 handle_request_invite: Unable to create/find SIP channel for this INVITE
[Aug 24 07:26:03]
<--- Transmitting (NAT) to 208.93.227.214:5060 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 208.93.227.214;branch=z9hG4bKX1H7NagB4BaKN;received=208.93.227.214;rport=5060
From: "8134215841" <sip:8134215841@208.93.227.214>;tag=gyD2tHact6Qre
To: <sip:8775189339@71.180.173.37:5060>;tag=as10531efc
Call-ID: bf4f3f8e-48e6-122f-66b6-f04da23d7069
CSeq: 16756529 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
[Aug 24 07:26:03] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7c44510 (len 459) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 24 07:26:03] Scheduling destruction of SIP dialog 'bf4f3f8e-48e6-122f-66b6-f04da23d7069' in 6400 ms (Method: INVITE)
[Aug 24 07:26:03] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 24 07:26:04]
<--- SIP read from 192.168.1.142:19138 --->

<------------->
[Aug 24 07:26:05] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 24 07:26:05] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK328104a6;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as4fbee3fd
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 719bc4052af92f0d044d91e3298b8f93@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 24 Aug 2011 11:26:05 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 24 07:26:05] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0x8b9c31c (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 24 07:26:06] == Parsing '/etc/asterisk/manager.conf': [Aug 24 07:26:06] Found
[Aug 24 07:26:06] == Manager 'sendcron' logged on from 127.0.0.1
[Aug 24 07:26:07] == Manager 'sendcron' logged off from 127.0.0.1
[Aug 24 07:26:09] Really destroying SIP dialog '719bc4052af92f0d044d91e3298b8f93@127.0.0.1' Method: OPTIONS
[Aug 24 07:26:19] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 24 07:26:19] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK2652bf74;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as662f17e0
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 1e07653f75725abb1110da7150cc312b@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 24 Aug 2011 11:26:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 24 07:26:19] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0x8b9b654 (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 24 07:26:23] Really destroying SIP dialog '1e07653f75725abb1110da7150cc312b@127.0.0.1' Method: OPTIONS
go*CLI>

by **williamconley** » Wed Aug 24, 2011 8:22 pm

williamconley wrote:
dajaz27 wrote:I can not make calls. Calls come into the system but are not routed properly (so no phone rings). Which leads me to believe there is a problem with either my trunk and or my dial plan. So how do I fix either my trunk or dial plan?
We were working before on your outbound. Did that get workin'?

by **dajaz27** » Wed Aug 24, 2011 8:37 pm

I created a trunk and a dial plan which I suspect was wrong. [/quote]

by **williamconley** » Wed Aug 24, 2011 9:04 pm

that still does not answer the question is outbound working?

by **dajaz27** » Wed Aug 24, 2011 10:18 pm

It never worked because the trunk/dialplan was not configured properly.

by **dajaz27** » Thu Aug 25, 2011 11:27 am

In the call capture...It says it can't route to exten 877****(toll free number) but I don't believe I setup a inbound ring group so it is not ringing to any extensions. Would that solve my inbound problem if the calls hit the server behind the firewall but not to individual phones?

by **callcrazy** » Thu Aug 25, 2011 11:55 am

Can you ping

208.93.227.214 from your server?

This warning is what everyone is talking about:

[Aug 24 07:26:03] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7c44510 (len 459) to 208.93.227.214:5060 returned -2: Network is unreachable

Network is unreachable. That would prevent calls from working.

by **williamconley** » Thu Aug 25, 2011 4:41 pm

dajaz27 wrote:It never worked because the trunk/dialplan was not configured properly.

When you're ready to get back to that, let me know.

by **dajaz27** » Thu Aug 25, 2011 10:50 pm

we are ready.

by **williamconley** » Fri Aug 26, 2011 9:10 am

Ok, show your present setup for the carrier for the outbound calls along with the asterisk CLI from a single attempt at an outbound call.

by **dajaz27** » Sun Aug 28, 2011 11:42 am

---
[Aug 28 12:37:30] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7d1d5bc (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 28 12:37:34] Really destroying SIP dialog '22d9c6527efb5ec3113794bc2344b50b@127.0.0.1' Method: OPTIONS
[Aug 28 12:37:44] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 28 12:37:44] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK5f0fb3f9;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as4e235dfe
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 362462376d2b37f232da68d67b3eae78@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 28 Aug 2011 16:37:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 28 12:37:44] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7d1d5bc (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
[Aug 28 12:37:48] Really destroying SIP dialog '362462376d2b37f232da68d67b3eae78@127.0.0.1' Method: OPTIONS
[Aug 28 12:37:53]
<--- SIP read from 192.168.1.142:61358 --->
INVITE sip:8134215841@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.142:61358;branch=z9hG4bK-d8754z-d9f4890d3f61cff5-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8001@192.168.1.142:61358>
To: <sip:8134215841@192.168.1.2>
From: "8001"<sip:8001@192.168.1.2>;tag=80e87a3f
Call-ID: YTQ2YThhZjg0N2MxN2VkOGQxMmIwYmQ5MWQ3MzJlOTc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 410

v=0
o=- 12959023081046875 1 IN IP4 192.168.1.142
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.1.142
t=0 0
a=ice-ufrag:f7af52
a=ice-pwd:4a3e18624780e9505106bf6a67d7a419
m=audio 57898 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.142 57898 typ host
a=candidate:1 2 UDP 659134 192.168.1.142 57899 typ host

<------------->
[Aug 28 12:37:53] --- (13 headers 14 lines) ---
[Aug 28 12:37:53] Sending to 192.168.1.142 : 61358 (NAT)
[Aug 28 12:37:53] Using INVITE request as basis request - YTQ2YThhZjg0N2MxN2VkOGQxMmIwYmQ5MWQ3MzJlOTc.
[Aug 28 12:37:53]
<--- Reliably Transmitting (NAT) to 192.168.1.142:61358 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.142:61358;branch=z9hG4bK-d8754z-d9f4890d3f61cff5-1---d8754z-;received=192.168.1.142;rport=61358
From: "8001"<sip:8001@192.168.1.2>;tag=80e87a3f
To: <sip:8134215841@192.168.1.2>;tag=as05534ebe
Call-ID: YTQ2YThhZjg0N2MxN2VkOGQxMmIwYmQ5MWQ3MzJlOTc.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="268286c4"
Content-Length: 0

<------------>
[Aug 28 12:37:53] Scheduling destruction of SIP dialog 'YTQ2YThhZjg0N2MxN2VkOGQxMmIwYmQ5MWQ3MzJlOTc.' in 32000 ms (Method: INVITE)
[Aug 28 12:37:53] Found user '8001'
[Aug 28 12:37:53]
<--- SIP read from 192.168.1.142:61358 --->
ACK sip:8134215841@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.142:61358;branch=z9hG4bK-d8754z-d9f4890d3f61cff5-1---d8754z-;rport
Max-Forwards: 70
To: <sip:8134215841@192.168.1.2>;tag=as05534ebe
From: "8001"<sip:8001@192.168.1.2>;tag=80e87a3f
Call-ID: YTQ2YThhZjg0N2MxN2VkOGQxMmIwYmQ5MWQ3MzJlOTc.
CSeq: 1 ACK
Content-Length: 0

<------------->
[Aug 28 12:37:53] --- (8 headers 0 lines) ---
[Aug 28 12:37:53]
<--- SIP read from 192.168.1.142:61358 --->
INVITE sip:8134215841@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.142:61358;branch=z9hG4bK-d8754z-b5d6a20ac95d1488-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:8001@192.168.1.142:61358>
To: <sip:8134215841@192.168.1.2>
From: "8001"<sip:8001@192.168.1.2>;tag=80e87a3f
Call-ID: YTQ2YThhZjg0N2MxN2VkOGQxMmIwYmQ5MWQ3MzJlOTc.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="8001",realm="asterisk",nonce="268286c4",uri="sip:8134215841@192.168.1.2",response="ef9d7ced4a36361fd6405bb79ee68a56",algorithm=MD5
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 410

v=0
o=- 12959023081046875 1 IN IP4 192.168.1.142
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.1.142
t=0 0
a=ice-ufrag:f7af52
a=ice-pwd:4a3e18624780e9505106bf6a67d7a419
m=audio 57898 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=candidate:1 1 UDP 659136 192.168.1.142 57898 typ host
a=candidate:1 2 UDP 659134 192.168.1.142 57899 typ host

<------------->
[Aug 28 12:37:53] --- (14 headers 14 lines) ---
[Aug 28 12:37:53] Sending to 192.168.1.142 : 61358 (NAT)
[Aug 28 12:37:53] Using INVITE request as basis request - YTQ2YThhZjg0N2MxN2VkOGQxMmIwYmQ5MWQ3MzJlOTc.
[Aug 28 12:37:53] Found user '8001'
[Aug 28 12:37:53] Found RTP audio format 107
[Aug 28 12:37:53] Found RTP audio format 0
[Aug 28 12:37:53] Found RTP audio format 8
[Aug 28 12:37:53] Found RTP audio format 101
[Aug 28 12:37:53] Found unknown media description format BV32 for ID 107
[Aug 28 12:37:53] Found audio description format telephone-event for ID 101
[Aug 28 12:37:53] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Aug 28 12:37:53] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Aug 28 12:37:53] Peer audio RTP is at port 192.168.1.142:57898
[Aug 28 12:37:53] Looking for 8134215841 in default (domain 192.168.1.2)
[Aug 28 12:37:53]
<--- Reliably Transmitting (NAT) to 192.168.1.142:61358 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.142:61358;branch=z9hG4bK-d8754z-b5d6a20ac95d1488-1---d8754z-;received=192.168.1.142;rport=61358
From: "8001"<sip:8001@192.168.1.2>;tag=80e87a3f
To: <sip:8134215841@192.168.1.2>;tag=as05534ebe
Call-ID: YTQ2YThhZjg0N2MxN2VkOGQxMmIwYmQ5MWQ3MzJlOTc.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
[Aug 28 12:37:53] NOTICE[3018]: chan_sip.c:15566 handle_request_invite: Call from '8001' to extension '8134215841' rejected because extension not found.
[Aug 28 12:37:53] Scheduling destruction of SIP dialog 'YTQ2YThhZjg0N2MxN2VkOGQxMmIwYmQ5MWQ3MzJlOTc.' in 32000 ms (Method: INVITE)
[Aug 28 12:37:53]
<--- SIP read from 192.168.1.142:61358 --->
ACK sip:8134215841@192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.142:61358;branch=z9hG4bK-d8754z-b5d6a20ac95d1488-1---d8754z-;rport
Max-Forwards: 70
To: <sip:8134215841@192.168.1.2>;tag=as05534ebe
From: "8001"<sip:8001@192.168.1.2>;tag=80e87a3f
Call-ID: YTQ2YThhZjg0N2MxN2VkOGQxMmIwYmQ5MWQ3MzJlOTc.
CSeq: 2 ACK
Content-Length: 0

<------------->
[Aug 28 12:37:53] --- (8 headers 0 lines) ---
[Aug 28 12:37:53]
<--- SIP read from 192.168.1.142:61358 --->

<------------->
[Aug 28 12:37:56]
<--- SIP read from 192.168.1.17:51414 --->

<------------->
[Aug 28 12:37:58] WARNING[3018]: acl.c:541 ast_ouraddrfor: Cannot connect
[Aug 28 12:37:58] Reliably Transmitting (NAT) to 208.93.227.214:5060:
OPTIONS sip:208.93.227.214;cpd=on SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK36b713a7;rport
From: "asterisk" <sip:asterisk@127.0.0.1>;tag=as1a7237c2
To: <sip:208.93.227.214;cpd=on>
Contact: <sip:asterisk@127.0.0.1>
Call-ID: 5343df942fc91d1b31ec79d81933a5d1@127.0.0.1
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 28 Aug 2011 16:37:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Aug 28 12:37:58] WARNING[3018]: chan_sip.c:1894 __sip_xmit: sip_xmit of 0xb7d0807c (len 500) to 208.93.227.214:5060 returned -2: Network is unreachable
go*CLI>

by **williamconley** » Sun Aug 28, 2011 1:39 pm

1) you have some obvious networking problems

2) you did not show your carrier setup

by **dajaz27** » Sun Aug 28, 2011 3:04 pm

carrier id: Broadvox
carrier name: GODOMESTIC1
global string DOMESTICTRUNK1=SIP/godomestic1

[godomestic1]
disallow=all
allow=ulaw
allow=g729
type=peer
dtmfmode=rfc2833
context=trunkinbound
host=208.93.227.214
canreinvite=no
nat=yes

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${DOMESTICTRUNK1}/${EXTEN:2},60,tTor)
exten => _91NXXNXXXXXX,3,Hangup

by **williamconley** » Sun Aug 28, 2011 5:41 pm

will need results from

Code: Select all: ping 208.93.227.214 -c 4

and dial 918134215841 instead of 8134215841 to match your dialplan (9 to choose the carrier, 1 for long distance, and then the 10 digit US phone number)

by **dajaz27** » Sun Aug 28, 2011 6:04 pm

[root@go ~]# ping 208.93.227.214 c-4
ping: unknown host c-4
You have new mail in /var/spool/mail/root
[root@go ~]#

by **williamconley** » Sun Aug 28, 2011 6:14 pm

you typed it instead of copy and paste. LOL

you missed a space after -c and before 4

by **dajaz27** » Sun Aug 28, 2011 6:14 pm

[root@go ~]# ifconfig
eth0 Link encap:Ethernet HWaddr F8:0F:41:1B:51:A8
inet addr:192.168.1.2 Bcast:192.168.1.255 Mask:255.255.255.0
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:1181880 errors:0 dropped:0 overruns:0 frame:0
TX packets:1312347 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:232087724 (221.3 MiB) TX bytes:903757521 (861.8 MiB)
Interrupt:233 Base address:0xc000

lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:25883554 errors:0 dropped:0 overruns:0 frame:0
TX packets:25883554 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:2620932805 (2.4 GiB) TX bytes:2620932805 (2.4 GiB)

[root@go ~]#

by **dajaz27** » Sun Aug 28, 2011 6:30 pm

l

[root@go ~]# ping 208.93.227.214 -c 4
connect: Network is unreachable
[root@go ~]#
[root@go ~]#
[root@go ~]# [root@go ~]# ping 208.93.227.214 -c 4
-bash: [root@go: command not found
[root@go ~]# connect: Network is unreachable
-bash: connect:: command not found
[root@go ~]# [root@go ~]#
-bash: [root@go: command not found
[root@go ~]#

by **williamconley** » Sun Aug 28, 2011 8:02 pm

if you cannot ping 208.93.227.214 ... you cannot send a call to 208.93.227.214. so go back to your getting started guide for GoAutoDial and find out what you did wrong with your network.

since your ip address is 192.168.1.2 ... what is the ip address of your router and what are your network settings?

by **dajaz27** » Mon Aug 29, 2011 11:36 am

I can ping the service provider from the router, other machines on the network. I can even ssh in to the goautodial.
No I really need to port forward,port trigger and static nat to get this to work?
Followed setup instructions.

Router 192.168.1.1
goautodial 192.168.1.2
255.255.255.0
default gateway 192.168.1.1
static route to 192.168.1.2 (5060 udp and tcp)
port fowarding 192.168.1.2 (5060 udp and tcp)
port trigger 192.168.1.2 (5060 udp and tcp)

by **williamconley** » Mon Aug 29, 2011 7:04 pm

try another router or return your router to "default" settings to clear any oddnees you have created. if your server cannot route to that IP address, you cannot make a call with that carrier.

can other computers on that local network ping the carrier?

by **dajaz27** » Tue Aug 30, 2011 10:00 am

Other computers can ping the carrier on the network. I can make ext. to ext calls. Just not out of the network.

by **striker** » Tue Aug 30, 2011 10:16 am

Are you able to ping your gateway ip

type ip route show
this will show your gateway which has been configured in you goautodial server.
make sure is it the correct gateway ip
and try to ping that ip.

by **dajaz27** » Tue Aug 30, 2011 10:31 am

I can ping the default gateway (verizon fios router) and the public IP address attached to the router.
In the ip route show i get this
192.168.1.0/24 dev eth0 proto kernel scope link src 192.168.1.2
168.254.0.0/16 dev eth0 scope link
default via 192.168.1.1 dev eth0

by **striker** » Tue Aug 30, 2011 10:54 am

so you able to ping 192.168.1.1

what is in the vi /etc/resolve.conf

vicidial.org

unable to send or receive calls

unable to send or receive calls

carrier setup

requested information

diaplan

will try

striker it didn't work

Is this the proper way to create the dial plan?

Let me start over

cli data

CLI logs

Re: Let me start over

Its now on a dmz and still the same problem. outbound call

Inbound call router not the issue. Calls make it to server.

The trunk/dial plan was never setup properly

it never worked

Here is a question may help me solve my outbound issue

Re: it never worked

i am ready

CLI logs

ping

This maybe the problem or related

ping

That didn't work either

yes

ping gateway

IP routing

ping gateway

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