Question Concerning Setup.

[Kerry C]

Also to your remark of this:

Your register string and username/pwd/authtype may not be necessary if you have provided the VOIP carrier with your IP address. In these cases authentication is handled already and no longer necessary, in which case those items may be entirely omitted.

VOIP said this:

Kerry, the registration string is only for trunks that require a registration. We have configured you for IP authentication so that is not needed.

[williamconley]

this was taking off the 8's you had, the 9 is in the campaigns, and 1 was set in the load leads setup.

Guess what. Put the 8s back.

Executing Dial("Local/8600051@default-5b40,1", "Zap/g2/12033338500||To")

Spawn extension (default, 912033338500, 3)

Tells me that your "dial 9" scenario is not working so well. that's why i said to use dial 8.

[Kerry C]

ok I added 8 back in the area you said in the carriers admin:

heres the cli when i try to connect.

== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/201-0820c670 was answered.
-- Executing MeetMe("SIP/201-0820c670", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-a86b,2", "8600051|F") in new stack
> Channel Local/8600051@default-a86b,1 was answered.
-- Executing AGI("Local/8600051@default-a86b,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-a86b,1", "Zap/g2/12033345949||To") in new stack
May 30 13:58:30 NOTICE[29501]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("Local/8600051@default-a86b,1", "") in new stack
== Spawn extension (default, 912033345949, 3) exited non-zero on 'Local/8600051@default-a86b,1'
-- Executing DeadAGI("Local/8600051@default-a86b,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-a86b,2'
-- Executing DeadAGI("Local/8600051@default-a86b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMeAdmin("Local/55558600051@default-b6ea,2", "8600051|K") in new stack
-- Executing Hangup("Local/55558600051@default-b6ea,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-b6ea,2'
-- Executing DeadAGI("Local/55558600051@default-b6ea,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- Playing 'conf-kicked' (language 'en')
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Hungup 'Zap/pseudo-348817745'
-- Executing DeadAGI("SIP/201-0820c670", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

[Kerry C]

make note:

the leads do not have a 1 in front of them, we are only dialing from the U.S in the U.S all I need is a 1 to go in front of the number, and we do not need 9, if i take the 9 out in the campaigns area and put x it says no extension in agents area

[williamconley]

(default, 912033345949, 3)

Your leads must have a "dial code" of 1 when loaded into the list.

You must change the "Campaign" to dial "8" instead of "9".

This will result in "812033345949" instead of "91203345949".

Your campaign Dialing 81NXXNXXXXXX will go through the dial plan for THIS carrier you have constructed.

Your campaign Dialing 91NXXNXXXXXX will go through the "sample" configuration hard coded into Vicibox which attempts to dial through "Zap" which does not exist in your system. (which is why it fails):

"Zap/g2/12033345949||To")

[Kerry C]

changed campaign numer to 8 instead of 9

logged back into agent, also in the lead load, i set the specified area you said to 1, which was done way before, I have always had it at 1.

heres the cli

== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-dbd9,2", "8600051|F") in new stack
> Channel Local/8600051@default-dbd9,1 was answered.
-- Executing AGI("Local/8600051@default-dbd9,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-dbd9,1", "SIP/bestvoipusa/12033483191||To") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMeAdmin("Local/55558600051@default-93fa,2", "8600051|K") in new stack
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/201-0820c670'
-- Executing Hangup("Local/55558600051@default-93fa,2", "") in new stack
-- Executing DeadAGI("SIP/201-0820c670", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-93fa,2'
-- Executing DeadAGI("Local/55558600051@default-93fa,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- Playing 'conf-kicked' (language 'en')
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Hungup 'Zap/pseudo-806334468'
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

[Kerry C]

waited, made sure the campign was at 8 in prefix, leads code on 1, heres another cli, it still gets no ring


> Channel SIP/201-0820c670 was answered.
-- Executing MeetMe("SIP/201-0820c670", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-395d,2", "8600051|F") in new stack
> Channel Local/8600051@default-395d,1 was answered.
-- Executing AGI("Local/8600051@default-395d,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-395d,1", "SIP/bestvoipusa/12033665674||To") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-395d,2'
-- Executing DeadAGI("Local/8600051@default-395d,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMeAdmin("Local/55558600051@default-67f2,2", "8600051|K") in new stack
-- Hungup 'Zap/pseudo-1372361019'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/201-0820c670'
== Parsing '/etc/asterisk/meetme.conf': -- Executing DeadAGI("SIP/201-0820c670", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
Foundox*CLI>
May 30 14:23:23 NOTICE[1673]: app_meetme.c:2210 admin_exec: Conference Number not found
-- Executing Hangup("Local/55558600051@default-67f2,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-67f2,2'
-- Executing DeadAGI("Local/55558600051@default-67f2,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
May 30 14:23:43 WARNING[1639]: chan_sip.c:2018 create_addr: No such host: bestvoipusa
May 30 14:23:43 NOTICE[1639]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("Local/8600051@default-395d,1", "") in new stack
== Spawn extension (default, 812033665674, 3) exited non-zero on 'Local/8600051@default-395d,1'
-- Executing DeadAGI("Local/8600051@default-395d,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Refreshing DNS lookups.
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
> Channel SIP/201-0820c670 was answered.
-- Executing MeetMe("SIP/201-0820c670", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-2f9f,2", "8600051|F") in new stack
> Channel Local/8600051@default-2f9f,1 was answered.
-- Executing AGI("Local/8600051@default-2f9f,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-2f9f,1", "SIP/bestvoipusa/12033684049||To") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMeAdmin("Local/55558600051@default-6e8a,2", "8600051|K") in new stack
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/201-0820c670'
-- Executing Hangup("Local/55558600051@default-6e8a,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-6e8a,2'
-- Executing DeadAGI("Local/55558600051@default-6e8a,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- Executing DeadAGI("SIP/201-0820c670", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Playing 'conf-kicked' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
-- Hungup 'Zap/pseudo-1571828091'
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

[Kerry C]

here is a couple of more things, NAT is set to yes in the sip.conf, I dont have NAT through the router.

the way I have this system set up is t1>router(server is setup from router)then I have a ethernet from router to switch where the computers are setup. I get internet access with no prob. I have dhcp enabled on the router, for the switch and computers.


and this dialer needs to dial the number 1 then number, not 9 then 1 then number, not 8 then 1 then number, just 1 then number.

[williamconley]

The dialer needs to SEND 1+number to the provider. This is accomplished by :1 after EXTEN (which strips off the 8 ).

Executing Dial("Local/8600051@default-395d,1", "SIP/bestvoipusa/12033665674||To") in new stack

Note that it is dialing 12033665674 NOT 8, because it is stripped off before being sent to the carrier. The 8 is there only to allow asterisk to identify WHICH carrier to send it to, then the 8 is discarded, having accomplished this task.

Before asterisk was trying to use "Zap". Now it's trying to use "bestvoipusa". Progress.

Now you should turn on sip debug and find out what is happening.

Also:

May 30 14:23:43 WARNING[1639]: chan_sip.c:2018 create_addr: No such host: bestvoipusa
May 30 14:23:43 NOTICE[1639]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)

Tells me something was unusual in your system during this call sequence, perhaps you were still in the middle of a reload or something.

So, it may have been about to work and you got impatient ... or something may be wrong requiring a reboot.

Try again, and then try with sip debug on and find out what happens to the call. It may be a format issue or several other possibilities.

[Kerry C]

here is the call sequence again through CLI, Now I am stating to understand what its doing in CLI:)


== Manager 'sendcron' logged off from 127.0.0.1
== Refreshing DNS lookups.
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/201-0820c670 was answered.
-- Executing MeetMe("SIP/201-0820c670", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-034a,2", "8600051|F") in new stack
> Channel Local/8600051@default-034a,1 was answered.
-- Executing AGI("Local/8600051@default-034a,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-034a,1", "SIP/bestvoipusa/12033744845||To") in new stack
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
May 30 16:09:29 WARNING[19667]: chan_sip.c:2018 create_addr: No such host: bestvoipusa
May 30 16:09:29 NOTICE[19667]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing Hangup("Local/8600051@default-034a,1", "") in new stack
== Spawn extension (default, 812033744845, 3) exited non-zero on 'Local/8600051@default-034a,1'
-- Executing DeadAGI("Local/8600051@default-034a,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-034a,2'
-- Executing DeadAGI("Local/8600051@default-034a,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMeAdmin("Local/55558600051@default-d96b,2", "8600051|K") in new stack
-- Hungup 'Zap/pseudo-765774001'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/201-0820c670'
-- Executing DeadAGI("SIP/201-0820c670", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
May 30 16:09:38 NOTICE[19802]: app_meetme.c:2210 admin_exec: Conference Number not found
-- Executing Hangup("Local/55558600051@default-d96b,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-d96b,2'
-- Executing DeadAGI("Local/55558600051@default-d96b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

[Kerry C]

this was letting it wait on ring for 30 seconds, BTW how do I setup sip debug?

[Kerry C]

enabled sip debug in cli, ran call got this from cli:




-- Executing Dial("Local/8600051@default-7826,1", "SIP/bestvoipusa/12034071956||To") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMeAdmin("Local/55558600051@default-b30c,2", "8600051|K") in new stack
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/201-081e4cb0'
-- Executing DeadAGI("SIP/201-081e4cb0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- Executing Hangup("Local/55558600051@default-b30c,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-b30c,2'
-- Executing DeadAGI("Local/55558600051@default-b30c,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- Playing 'conf-kicked' (language 'en')
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Scheduling destruction of call '233e709e2616bb5c43b83094726ab090@192.168.1.86' in 32000 ms
set_destination: Parsing <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a> for address/port to send to
set_destination: set destination to 192.168.1.115, port 17190
Reliably Transmitting (NAT) to 192.168.1.115:17190:
BYE sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK59a1d2b7;rport
From: "S0905301615528600051" <sip:0000000000@192.168.1.86>;tag=as4b9b10d6
To: <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a>;tag=ec4f0564
Call-ID: 233e709e2616bb5c43b83094726ab090@192.168.1.86
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S0905301615528600051" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Content-Length: 0


---
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Vicibox*CLI>
<-- SIP read from 192.168.1.115:17190:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK59a1d2b7;rport=5060
Contact: <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a>
To: <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a>;tag=ec4f0564
From: "S0905301615528600051"<sip:0000000000@192.168.1.86>;tag=as4b9b10d6
Call-ID: 233e709e2616bb5c43b83094726ab090@192.168.1.86
CSeq: 103 BYE
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0


--- (9 headers 0 lines) ---
Destroying call '233e709e2616bb5c43b83094726ab090@192.168.1.86'
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Hungup 'Zap/pseudo-2119577457'
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
12 headers, 0 lines
Reliably Transmitting (NAT) to xx.x.xxx.xx:5060:
OPTIONS sip:64.2.142.93 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK3f2477c2;rport
From: "asterisk" <sip:asterisk@192.168.1.86>;tag=as749b7a21
To: <sip:xx.x.xxx.xx>
Contact: <sip:asterisk@192.168.1.86>
Call-ID: 50e788fd7317cfa54ce189026204358e@192.168.1.86
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 30 May 2009 20:16:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Vicibox*CLI>
<-- SIP read from xx.x.xxx.xx:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK3f2477c2;received=xx.x.xxx.xx;rport=5060
To: <sip:xx.x.xxx.xx>;tag=as0a402d49
From: "asterisk" <sip:asterisk@192.168.1.86>;tag=as749b7a21
Call-ID: 50e788fd7317cfa54ce189026204358e@192.168.1.86
CSeq: 102 OPTIONS
Content-Length: 0
User-Agent: Packetrino
Supported: replaces
Accept: application/sdp
Record-Route: <sip:xx.x.xxx.xx:5060;lr>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY


--- (12 headers 0 lines) ---
Destroying call '50e788fd7317cfa54ce189026204358e@192.168.1.86'
-- Timeout on Local/8600051@default-7826,2
== CDR updated on Local/8600051@default-7826,2
-- Executing Goto("Local/8600051@default-7826,2", "#|1") in new stack
-- Goto (default,#,1)
-- Executing Playback("Local/8600051@default-7826,2", "invalid") in new stack
-- Playing 'invalid' (language 'en')
-- Executing Hangup("Local/8600051@default-7826,2", "") in new stack
== Spawn extension (default, #, 2) exited non-zero on 'Local/8600051@default-7826,2'
-- Executing DeadAGI("Local/8600051@default-7826,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Vicibox*CLI>
<-- SIP read from 192.168.1.115:17190:



--- (0 headers 1 lines) ---
May 30 16:16:27 WARNING[20949]: chan_sip.c:2018 create_addr: No such host: bestvoipusa
Destroying call '517ca99b04bb26135daabbad5dc22493@192.168.1.86'
May 30 16:16:27 NOTICE[20949]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing DeadAGI("Local/8600051@default-7826,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Vicibox*CLI>
Vicibox*CLI>
Vicibox*CLI>
Vicibox*CLI>
Vicibox*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
We're at 192.168.1.86 port 19744
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 11 lines
Reliably Transmitting (NAT) to 192.168.1.115:17190:
INVITE sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK00f312f0;rport
From: "S0905301616448600051" <sip:0000000000@192.168.1.86>;tag=as6102381c
To: <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a>
Contact: <sip:0000000000@192.168.1.86>
Call-ID: 68e40bb82f7d212e1e539538045ab860@192.168.1.86
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S0905301616448600051" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Date: Sat, 30 May 2009 20:16:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1392 1392 IN IP4 192.168.1.86
s=session
c=IN IP4 192.168.1.86
t=0 0
m=audio 19744 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Vicibox*CLI>
<-- SIP read from 192.168.1.115:17190:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK00f312f0;rport=5060
To: <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a>
From: "S0905301616448600051" <sip:0000000000@192.168.1.86>;tag=as6102381c
Call-ID: 68e40bb82f7d212e1e539538045ab860@192.168.1.86
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines) ---
Vicibox*CLI>
<-- SIP read from 192.168.1.115:17190:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK00f312f0;rport=5060
Contact: <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a>
To: <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a>;tag=bf39c840
From: "S0905301616448600051"<sip:0000000000@192.168.1.86>;tag=as6102381c
Call-ID: 68e40bb82f7d212e1e539538045ab860@192.168.1.86
CSeq: 102 INVITE
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0


--- (9 headers 0 lines) ---
Vicibox*CLI>
<-- SIP read from 192.168.1.115:17190:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK00f312f0;rport=5060
Contact: <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a>
To: <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a>;tag=bf39c840
From: "S0905301616448600051"<sip:0000000000@192.168.1.86>;tag=as6102381c
Call-ID: 68e40bb82f7d212e1e539538045ab860@192.168.1.86
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 187

v=0
o=- 1 2 IN IP4 192.168.1.115
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.115
t=0 0
m=audio 40530 RTP/AVP 0 3 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.115:40530
Found description format telephone-event
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a>
set_destination: Parsing <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a> for address/port to send to
set_destination: set destination to 192.168.1.115, port 17190
Transmitting (NAT) to 192.168.1.115:17190:
ACK sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK0477d955;rport
From: "S0905301616448600051" <sip:0000000000@192.168.1.86>;tag=as6102381c
To: <sip:201@192.168.1.115:17190;rinstance=c3eb78ead04f641a>;tag=bf39c840
Contact: <sip:0000000000@192.168.1.86>
Call-ID: 68e40bb82f7d212e1e539538045ab860@192.168.1.86
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S0905301616448600051" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Content-Length: 0


---
> Channel SIP/201-0820c670 was answered.
-- Executing MeetMe("SIP/201-0820c670", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-0183,2", "8600051|F") in new stack
> Channel Local/8600051@default-0183,1 was answered.
-- Executing AGI("Local/8600051@default-0183,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-0183,1", "SIP/bestvoipusa/12034666550||To") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-0183,2'
-- Executing DeadAGI("Local/8600051@default-0183,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Vicibox*CLI>
<-- SIP read from 192.168.1.115:17190:



--- (0 headers 1 lines) ---
Vicibox*CLI>

[Kerry C]

Here is another cli after restarting asterisk from a restart now command then logging into agent and trying to make call after showing sip debug.


<-- SIP read from xx.x.xxx.xx:5060:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK197e9dd2;received=66.253.57.198;rport=5060
To: <sip:64.2.142.93>;tag=as41998fe6
From: "asterisk" <sip:asterisk@192.168.1.86>;tag=as5152eef8
Call-ID: 7e3119b63556f6cf52c32468080c19e6@192.168.1.86
CSeq: 102 OPTIONS
Content-Length: 0
User-Agent: Packetrino
Supported: replaces
Accept: application/sdp
Record-Route: <sip:xx.x.xxx.xx:5060;lr>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY


--- (12 headers 0 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Destroying call '7e3119b63556f6cf52c32468080c19e6@192.168.1.86'
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
We're at 192.168.1.86 port 10700
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 11 lines
Reliably Transmitting (NAT) to 192.168.1.115:12978:
INVITE sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK2b5e5c7a;rport
From: "S0905301711208600051" <sip:0000000000@192.168.1.86>;tag=as38b210b0
To: <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce>
Contact: <sip:0000000000@192.168.1.86>
Call-ID: 0db32c1a313d3d7e101b6ed2262c9b63@192.168.1.86
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S0905301711208600051" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Date: Sat, 30 May 2009 21:11:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1392 1392 IN IP4 192.168.1.86
s=session
c=IN IP4 192.168.1.86
t=0 0
m=audio 10700 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Vicibox*CLI>
<-- SIP read from 192.168.1.115:12978:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK2b5e5c7a;rport=5060
To: <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce>
From: "S0905301711208600051" <sip:0000000000@192.168.1.86>;tag=as38b210b0
Call-ID: 0db32c1a313d3d7e101b6ed2262c9b63@192.168.1.86
CSeq: 102 INVITE
Content-Length: 0


--- (7 headers 0 lines) ---
Vicibox*CLI>
<-- SIP read from 192.168.1.115:12978:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK2b5e5c7a;rport=5060
Contact: <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce>
To: <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce>;tag=4046fd0b
From: "S0905301711208600051"<sip:0000000000@192.168.1.86>;tag=as38b210b0
Call-ID: 0db32c1a313d3d7e101b6ed2262c9b63@192.168.1.86
CSeq: 102 INVITE
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0


--- (9 headers 0 lines) ---
Vicibox*CLI>
<-- SIP read from 192.168.1.115:12978:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK2b5e5c7a;rport=5060
Contact: <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce>
To: <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce>;tag=4046fd0b
From: "S0905301711208600051"<sip:0000000000@192.168.1.86>;tag=as38b210b0
Call-ID: 0db32c1a313d3d7e101b6ed2262c9b63@192.168.1.86
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 187

v=0
o=- 6 2 IN IP4 192.168.1.115
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.115
t=0 0
m=audio 44582 RTP/AVP 0 3 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

--- (11 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.115:44582
Found description format telephone-event
Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x6 (gsm|ulaw)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce>
set_destination: Parsing <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce> for address/port to send to
set_destination: set destination to 192.168.1.115, port 12978
Transmitting (NAT) to 192.168.1.115:12978:
ACK sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK7c9ff511;rport
From: "S0905301711208600051" <sip:0000000000@192.168.1.86>;tag=as38b210b0
To: <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce>;tag=4046fd0b
Contact: <sip:0000000000@192.168.1.86>
Call-ID: 0db32c1a313d3d7e101b6ed2262c9b63@192.168.1.86
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S0905301711208600051" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Content-Length: 0


---
> Channel SIP/201-08173cd0 was answered.
-- Executing MeetMe("SIP/201-08173cd0", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-24dc,2", "8600051|F") in new stack
> Channel Local/8600051@default-24dc,1 was answered.
-- Executing AGI("Local/8600051@default-24dc,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-24dc,1", "SIP/bestvoipusa/12034689589||To") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
Vicibox*CLI>
<-- SIP read from 192.168.1.115:12978:



--- (0 headers 1 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMeAdmin("Local/55558600051@default-9354,2", "8600051|K") in new stack
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/201-08173cd0'
-- Executing DeadAGI("SIP/201-08173cd0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- Executing Hangup("Local/55558600051@default-9354,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-9354,2'
-- Executing DeadAGI("Local/55558600051@default-9354,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Playing 'conf-kicked' (language 'en')
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Scheduling destruction of call '0db32c1a313d3d7e101b6ed2262c9b63@192.168.1.86' in 32000 ms
set_destination: Parsing <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce> for address/port to send to
set_destination: set destination to 192.168.1.115, port 12978
Reliably Transmitting (NAT) to 192.168.1.115:12978:
BYE sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK7af2966b;rport
From: "S0905301711208600051" <sip:0000000000@192.168.1.86>;tag=as38b210b0
To: <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce>;tag=4046fd0b
Call-ID: 0db32c1a313d3d7e101b6ed2262c9b63@192.168.1.86
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "S0905301711208600051" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Content-Length: 0


---
Vicibox*CLI>
<-- SIP read from 192.168.1.115:12978:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK7af2966b;rport=5060
Contact: <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce>
To: <sip:201@192.168.1.115:12978;rinstance=492ea85a2d4a52ce>;tag=4046fd0b
From: "S0905301711208600051"<sip:0000000000@192.168.1.86>;tag=as38b210b0
Call-ID: 0db32c1a313d3d7e101b6ed2262c9b63@192.168.1.86
CSeq: 103 BYE
User-Agent: X-Lite release 1100l stamp 47546
Content-Length: 0


--- (9 headers 0 lines) ---
Destroying call '0db32c1a313d3d7e101b6ed2262c9b63@192.168.1.86'
== Manager 'sendcron' logged off from 127.0.0.1
-- Hungup 'Zap/pseudo-1818046168'
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- Timeout on Local/8600051@default-24dc,2
== CDR updated on Local/8600051@default-24dc,2
-- Executing Goto("Local/8600051@default-24dc,2", "#|1") in new stack
-- Goto (default,#,1)
-- Executing Playback("Local/8600051@default-24dc,2", "invalid") in new stack
-- Playing 'invalid' (language 'en')
-- Executing Hangup("Local/8600051@default-24dc,2", "") in new stack
== Spawn extension (default, #, 2) exited non-zero on 'Local/8600051@default-24dc,2'
-- Executing DeadAGI("Local/8600051@default-24dc,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
May 30 17:11:57 WARNING[30384]: chan_sip.c:2018 create_addr: No such host: bestvoipusa
Destroying call '31c38ea56171102b7d11929131240893@192.168.1.86'
May 30 17:11:57 NOTICE[30384]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing DeadAGI("Local/8600051@default-24dc,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----3-----CHANUNAVAIL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Vicibox*CLI>

[williamconley]

How was that call generated? On the screen as a manual dial by the agent or as a lead using ratio dialing after the agent resumed?

[Kerry C]

i logged into agent portal via web, in campaign created, with basic setup from the tutorials in the manual.

what I am doing is logging in via submit, hearing voice " you are currently the only one logged into this conference" then click dial next number, wait 10 to 15 second, as it says waiting on ring, then i hang up and click no answer then logout.

I copy and past the cli to you here.

[Kerry C]

Any Ideas?

May 30 17:11:57 WARNING[30384]: chan_sip.c:2018 create_addr: No such host: bestvoipusa
Destroying call '31c38ea56171102b7d11929131240893@192.168.1.86'
May 30 17:11:57 NOTICE[30384]: app_dial.c:1076 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)

[Kerry C]

When you gave me the sip.conf entry for the admin in vicidial it was this:

Account Entry:
[bestvoiopusa]
disallow=all
allow=all
type=friend
username=(contact your provider for this)
secret=(contact your provider for this)
auth=(contact your provider for this)
host=xx.x.xxx.xx
dtmfmode=rfc2833
context=trunkinbound
insecure=very
canreinvite=no
qualify=500

bestvoip was spelled incorrectly so i changed this:

[bestvoiopusa]

to this

[bestvoipusa]


now on the cli I get this when connecting manual in agent login:( please read over this I did get a live call via what the agent web interface said but i heard no ring, and no one on the other line.


> Channel SIP/201-081f7dc0 was answered.
-- Executing MeetMe("SIP/201-081f7dc0", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-7819,2", "8600051|F") in new stack
> Channel Local/8600051@default-7819,1 was answered.
-- Executing AGI("Local/8600051@default-7819,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-7819,1", "SIP/bestvoipusa/12036699249||To") in new stack
-- Called bestvoipusa/12036699249
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/bestvoipusa-08223940 is making progress passing it to Local/8600051@default-7819,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 812036699249, 2) exited non-zero on 'Local/8600051@default-7819,1'
-- Executing DeadAGI("Local/8600051@default-7819,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-7819,2'
-- Executing DeadAGI("Local/8600051@default-7819,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMeAdmin("Local/55558600051@default-adc2,2", "8600051|K") in new stack
-- Hungup 'Zap/pseudo-900813218'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/201-081f7dc0'
== Parsing '/etc/asterisk/meetme.conf': Found
-- Executing DeadAGI("SIP/201-081f7dc0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
May 31 11:29:29 NOTICE[4601]: app_meetme.c:2210 admin_exec: Conference Number not found
-- Executing Hangup("Local/55558600051@default-adc2,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-adc2,2'
-- Executing DeadAGI("Local/55558600051@default-adc2,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
Vicibox*CLI>




HERE IS THE SIP DEBUG ENTRY FOR NEXT TRY:

From: "asterisk" <sip:asterisk@192.168.1.86>;tag=as4496ac6d
Call-ID: 090d19311c56da9b7dc424681b175179@192.168.1.86
CSeq: 102 OPTIONS
Content-Length: 0
User-Agent: Packetrino
Supported: replaces
Accept: application/sdp
Record-Route: <sip:xx.x.xxx.xx:5060;lr>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY


--- (12 headers 0 lines) ---
Destroying call '090d19311c56da9b7dc424681b175179@192.168.1.86'
Vicibox*CLI>
<-- SIP read from 192.168.1.101:46448:



--- (0 headers 1 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Scheduling destruction of call '1624070b32e1ce6f402b743337ac4edd@192.168.1.86' in 32000 ms
Reliably Transmitting (NAT) to xx.x.xxx.xx:5060:
CANCEL sip:12037291135@xx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK4fb02bac;rport
From: "M0531114433000060040" <sip:0000000000@192.168.1.86>;tag=as584190c9
To: <sip:12037291135@xx.x.xxx.xx>
Call-ID: 1624070b32e1ce6f402b743337ac4edd@192.168.1.86
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M0531114433000060040" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Content-Length: 0


---
== Spawn extension (default, 812037291135, 2) exited non-zero on 'Local/8600051@default-a57a,1'
-- Executing DeadAGI("Local/8600051@default-a57a,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-a57a,2'
-- Executing DeadAGI("Local/8600051@default-a57a,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Vicibox*CLI>
<-- SIP read from xx.x.xxx.xx:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK4fb02bac;received=xx.xxx.xx.xxx;rport=5060
To: <sip:12037291135@xx.x.xxx.xx>;tag=as50fa0c77
From: "M0531114433000060040" <sip:0000000000@192.168.1.86>;tag=as584190c9
Call-ID: 1624070b32e1ce6f402b743337ac4edd@192.168.1.86
CSeq: 102 INVITE
Content-Length: 0
User-Agent: Packetrino
Supported: replaces
Record-Route: <sip:xx.x.xxx.xx:5060;lr>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY


--- (11 headers 0 lines) ---
Transmitting (NAT) to xx.x.xxx.xx:5060:
ACK sip:12037291135@xx.x.xxx.xx SIP/2.0







**************************************************************************************

this established a live call in the agnet area but I heard no ring, nor heard anyone on other end.






a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- (13 headers 15 lines) ---
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 64.2.142.51:13402
Found description format GSM
Found description format PCMU
Found description format G729
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x106 (gsm|ulaw|g729)/video=0x0 (nothing), combined - 0x106 (gsm|ulaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
-- SIP/bestvoipusa-08234ff8 is making progress passing it to Local/8600051@default-0e7c,1
== Manager 'sendcron' logged off from 127.0.0.1
-- Timeout on Local/8600051@default-1905,2
== CDR updated on Local/8600051@default-1905,2
-- Executing Goto("Local/8600051@default-1905,2", "#|1") in new stack
-- Goto (default,#,1)
-- Executing Playback("Local/8600051@default-1905,2", "invalid") in new stack
-- Playing 'invalid' (language 'en')
-- Executing Hangup("Local/8600051@default-1905,2", "") in new stack
== Spawn extension (default, #, 2) exited non-zero on 'Local/8600051@default-1905,2'
-- Executing DeadAGI("Local/8600051@default-1905,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Scheduling destruction of call '11c4e5763286bee85aca4ed374b10bd0@192.168.1.86' in 32000 ms
Reliably Transmitting (NAT) to xx.x.xxx.xx:5060:
CANCEL sip:12037294213@xx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK64e8475c;rport
From: "M0531114512000060041" <sip:0000000000@192.168.1.86>;tag=as34296c44
To: <sip:12037294213@xx.x.xxx.xx>
Call-ID: 11c4e5763286bee85aca4ed374b10bd0@192.168.1.86
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M0531114512000060041" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Content-Length: 0


---
== Spawn extension (default, 812037294213, 2) exited non-zero on 'Local/8600051@default-1905,1'
-- Executing DeadAGI("Local/8600051@default-1905,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Vicibox*CLI>
<-- SIP read from xx.x.xxx.xx:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK64e8475c;received=xx.x.xxx.xx;rport=5060
To: <sip:12037294213@xx.x.xxx.xx>;tag=as389fa7d5
From: "M0531114512000060041" <sip:0000000000@192.168.1.86>;tag=as34296c44
Call-ID: 11c4e5763286bee85aca4ed374b10bd0@192.168.1.86
CSeq: 102 INVITE
Content-Length: 0
User-Agent: Packetrino
Supported: replaces
Record-Route: <sip:xx.x.xxx.xx:5060;lr>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY


--- (11 headers 0 lines) ---
Transmitting (NAT) to xx.x.xxx.xx:5060:
ACK sip:12037294213@xx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK64e8475c;rport
From: "M0531114512000060041" <sip:0000000000@192.168.1.86>;tag=as34296c44
To: <sip:12037294213@xx.x.xxx.xx>;tag=as389fa7d5
Contact: <sip:0000000000@192.168.1.86>
Call-ID: 11c4e5763286bee85aca4ed374b10bd0@192.168.1.86
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M0531114512000060041" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Content-Length: 0


---
Vicibox*CLI>
<-- SIP read fromxx.x.xxx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK64e8475c;received=66.253.57.198;rport=5060
To: <sip:12037294213@xx.x.xxx.xx>;tag=as389fa7d5
From: "M0531114512000060041" <sip:0000000000@192.168.1.86>;tag=as34296c44
Call-ID: 11c4e5763286bee85aca4ed374b10bd0@192.168.1.86
CSeq: 102 CANCEL
Contact: <sip:12037294213@xx.x.xxx.xx>
Content-Length: 0
Record-Route: <sip:xx.x.xxx.xx:5060>
User-Agent: Packetrino
Supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY


--- (12 headers 0 lines) ---
Destroying call '11c4e5763286bee85aca4ed374b10bd0@192.168.1.86'
Vicibox*CLI>
<-- SIP read from xx.x.xxx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK3d24ee98;received=xx.x.xxx.xx;rport=5060
To: <sip:12037296284@xx.x.xxx.xx>;tag=as09cc23bb
From: "M0531114614000060042" <sip:0000000000@192.168.1.86>;tag=as61fb3956
Call-ID: 7d59b4a21871061c0f33eb99105a049e@192.168.1.86
CSeq: 102 INVITE
Content-Type: application/sdp
Contact: <sip:12037296284@xx.x.xxx.xx>
Content-Length: 306
Record-Route: <sip:xx.x.xxx.xx:5060>
User-Agent: Packetrino
Supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

v=0
o=root 2632 2633 IN IP4 xx.x.xxx.xx
s=session
c=IN IP4 xx.x.xxx.xx
t=0 0
m=audio 13402 RTP/AVP 3 0 18 101
a=rtpmap:3 GSM/8000
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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- (13 headers 15 lines) ---
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port xx.x.xxx.xx:13402
Found description format GSM
Found description format PCMU
Found description format G729
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x106 (gsm|ulaw|g729)/video=0x0 (nothing), combined - 0x106 (gsm|ulaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:xx.x.xxx.xx:5060>
list_route: hop: <sip:12037296284@xx.x.xxx.xx>
set_destination: Parsing <sip:xx.x.xxx.xx:5060> for address/port to send to
set_destination: set destination to xx.x.xxx.xx, port 5060
Transmitting (NAT) toxx.x.xxx.xx:5060:
ACK sip:xx.x.xxx.xx:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK3693165b;rport
Route: <sip:12037296284@xx.x.xxx.xx>
From: "M0531114614000060042" <sip:0000000000@192.168.1.86>;tag=as61fb3956
To: <sip:12037296284@xx.x.xxx.xx>;tag=as09cc23bb
Contact: <sip:0000000000@192.168.1.86>
Call-ID: 7d59b4a21871061c0f33eb99105a049e@192.168.1.86
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M0531114614000060042" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Content-Length: 0


---
-- SIP/bestvoipusa-08234ff8 answered Local/8600051@default-0e7c,1
Vicibox*CLI>
<-- SIP read from 192.168.1.101:46448:



--- (0 headers 1 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Scheduling destruction of call '7d59b4a21871061c0f33eb99105a049e@192.168.1.86' in 32000 ms
set_destination: Parsing <sip:64.2.142.93:5060> for address/port to send to
set_destination: set destination to xx.x.xxx.xx, port 5060
Reliably Transmitting (NAT) to xx.x.xxx.xx:5060:
BYE sip:xx.x.xxx.xx:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK66dc9969;rport
Route: <sip:12037296284@xx.x.xxx.xx>
From: "M0531114614000060042" <sip:0000000000@192.168.1.86>;tag=as61fb3956
To: <sip:12037296284@xx.x.xxx.xx>;tag=as09cc23bb
Call-ID: 7d59b4a21871061c0f33eb99105a049e@192.168.1.86
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M0531114614000060042" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Content-Length: 0


---
== Spawn extension (default, 812037296284, 2) exited non-zero on 'Local/8600051@default-0e7c,1'
-- Executing DeadAGI("Local/8600051@default-0e7c,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----39-----19") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -39-----19 completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-0e7c,2'
-- Executing DeadAGI("Local/8600051@default-0e7c,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
Vicibox*CLI>
<-- SIP read from xx.x.xxx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK66dc9969;received=xx.x.xxx.xx;rport=5060
To: <sip:12037296284@64.2.142.93>;tag=as09cc23bb
From: "M0531114614000060042" <sip:0000000000@192.168.1.86>;tag=as61fb3956
Call-ID: 7d59b4a21871061c0f33eb99105a049e@192.168.1.86
CSeq: 103 BYE
Contact: <sip:12037296284@xx.x.xxx.xx>
Content-Length: 0
Record-Route: <sip:xx.x.xxx.xx:5060>
User-Agent: Packetrino
Supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY


--- (12 headers 0 lines) ---
Destroying call '7d59b4a21871061c0f33eb99105a049e@192.168.1.86'
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
Vicibox*CLI>

Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK4fb02bac;rport
From: "M0531114433000060040" <sip:0000000000@192.168.1.86>;tag=as584190c9
To: <sip:12037291135@xx.x.xxx.xx>;tag=as50fa0c77
Contact: <sip:0000000000@192.168.1.86>
Call-ID: 1624070b32e1ce6f402b743337ac4edd@192.168.1.86
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M0531114433000060040" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Content-Length: 0


---
Vicibox*CLI>
<-- SIP read from xx.x.xxx.xx:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK4fb02bac;received=xx.x.xxx.xx;rport=5060
To: <sip:12037291135@xx.x.xxx.xx>;tag=as50fa0c77
From: "M0531114433000060040" <sip:0000000000@192.168.1.86>;tag=as584190c9
Call-ID: 1624070b32e1ce6f402b743337ac4edd@192.168.1.86
CSeq: 102 CANCEL
Contact: <sip:12037291135@xx.x.xxx.xx>
Content-Length: 0
Record-Route: <sip:xx.x.xxx.xx:5060>
User-Agent: Packetrino
Supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY


--- (12 headers 0 lines) ---
Destroying call '1624070b32e1ce6f402b743337ac4edd@192.168.1.86'
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-1905,2", "8600051|F") in new stack
> Channel Local/8600051@default-1905,1 was answered.
-- Executing AGI("Local/8600051@default-1905,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-1905,1", "SIP/bestvoipusa/12037294213||To") in new stack
We're at 192.168.1.86 port 18364
Adding codec 0x40 (slin) to SDP
Adding codec 0x1 (g723) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x100 (g729) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
14 headers, 22 lines
Reliably Transmitting (NAT) to xx.x.xxx.xx:5060:
INVITE sip:12037294213@xx.x.xxx.xx SIP/2.0
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK64e8475c;rport
From: "M0531114512000060041" <sip:0000000000@192.168.1.86>;tag=as34296c44
To: <sip:12037294213@xx.x.xxx.xx>
Contact: <sip:0000000000@192.168.1.86>
Call-ID: 11c4e5763286bee85aca4ed374b10bd0@192.168.1.86
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M0531114512000060041" <sip:0000000000@192.168.1.86>;privacy=off;screen=no
Date: Sun, 31 May 2009 15:45:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 510

v=0
o=root 5134 5134 IN IP4 192.168.1.86
s=session
c=IN IP4 192.168.1.86
t=0 0
m=audio 18364 RTP/AVP 10 4 3 0 8 111 5 7 18 110 97 101
a=rtpmap:10 L16/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:7 LPC/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called bestvoipusa/12037294213
Vicibox*CLI>
<-- SIP read from xx.x.xxx.xx:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK64e8475c;received=xx.x.xxx.xx;rport=5060
To: <sip:12037294213@xx.x.xxx.xx>
From: "M0531114512000060041" <sip:0000000000@192.168.1.86>;tag=as34296c44
Call-ID: 11c4e5763286bee85aca4ed374b10bd0@192.168.1.86
CSeq: 102 INVITE
Contact: <sip:12037294213@xx.x.xxx.xx>
Content-Length: 0
Record-Route: <sip:xx.x.xxx.xx:5060>
User-Agent: Packetrino
Supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY


--- (12 headers 0 lines) ---
Vicibox*CLI>
<-- SIP read from xx.x.xxx.xx:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.86:5060;branch=z9hG4bK64e8475c;received=xx.x.xxx.xx;rport=5060
To: <sip:12037294213@xx.x.xxx.xx>;tag=as389fa7d5
From: "M0531114512000060041" <sip:0000000000@192.168.1.86>;tag=as34296c44
Call-ID: 11c4e5763286bee85aca4ed374b10bd0@192.168.1.86
CSeq: 102 INVITE
Content-Type: application/sdp
Contact: <sip:12037294213@xx.x.xxx.xx>
Content-Length: 308
Record-Route: <sip:xx.x.xxx.xx:5060>
User-Agent: Packetrino
Supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

v=0
o=root 2411 2411 IN IP4 xx.x.xxx.xx
s=session
c=IN IP4 64.2.142.164
t=0 0
m=audio 18308 RTP/AVP 3 0 18 101
a=rtpmap:3 GSM/8000
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=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- (13 headers 15 lines) ---
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port xx.x.xxx.xx:18308
Found description format GSM
Found description format PCMU
Found description format G729
Found description format telephone-event
Capabilities: us - 0x1f07ff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|jpeg|png|h261|h263|h263p), peer - audio=0x106 (gsm|ulaw|g729)/video=0x0 (nothing), combined - 0x106 (gsm|ulaw|g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
-- SIP/bestvoipusa-08223940 is making progress passing it to Local/8600051@default-1905,1
== Manager 'sendcron' logged off from 127.0.0.1
Vicibox*CLI>
<-- SIP read from 192.168.1.101:46448:



--- (0 headers 1 lines) ---
Vicibox*CLI>

[Kerry C]

I set up a list with the numbers here in the office, i go into the agents campaign, manual dial, the phone rings in here, they can hear me fine, but i cannot hear them. I can talk to them, but they cant talk to me.


heres the cli:

== Manager 'sendcron' logged off from 127.0.0.1
May 31 12:47:21 NOTICE[5148]: chan_sip.c:11742 do_monitor: Disconnecting call 'SIP/bestvoipusa-0821b030' for lack of RTP activity in 62 seconds
== Spawn extension (default, 819047302570, 2) exited non-zero on 'Local/8600051@default-3c17,1'
-- Executing DeadAGI("Local/8600051@default-3c17,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----88-----62") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -88-----62 completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-3c17,2'
-- Executing DeadAGI("Local/8600051@default-3c17,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Refreshing DNS lookups.
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
> Channel SIP/201-0820dc00 was answered.
-- Executing MeetMe("SIP/201-0820dc00", "8600051|F") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-ea3f,2", "8600051|F") in new stack
May 31 12:53:28 WARNING[5140]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x8203b78', 9 retries!
> Channel Local/8600051@default-ea3f,1 was answered.
-- Executing AGI("Local/8600051@default-ea3f,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-ea3f,1", "SIP/bestvoipusa/19047302562||To") in new stack
-- Called bestvoipusa/19047302562
-- SIP/bestvoipusa-08204668 is making progress passing it to Local/8600051@default-ea3f,1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 819047302562, 2) exited non-zero on 'Local/8600051@default-ea3f,1'
-- Executing DeadAGI("Local/8600051@default-ea3f,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-ea3f,2'
-- Executing DeadAGI("Local/8600051@default-ea3f,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-d9f9,2", "8600051|F") in new stack
> Channel Local/8600051@default-d9f9,1 was answered.
-- Executing AGI("Local/8600051@default-d9f9,1", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/8600051@default-d9f9,1", "SIP/bestvoipusa/19043330531||To") in new stack
-- Called bestvoipusa/19043330531
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/bestvoipusa-08204668 is making progress passing it to Local/8600051@default-d9f9,1
-- SIP/bestvoipusa-08204668 answered Local/8600051@default-d9f9,1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 819043330531, 2) exited non-zero on 'Local/8600051@default-d9f9,1'
-- Executing DeadAGI("Local/8600051@default-d9f9,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----32-----16") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -32-----16 completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-d9f9,2'
-- Executing DeadAGI("Local/8600051@default-d9f9,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMeAdmin("Local/55558600051@default-0239,2", "8600051|K") in new stack
-- Hungup 'Zap/pseudo-1152166557'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/201-0820dc00'
-- Executing DeadAGI("SIP/201-0820dc00", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
May 31 12:54:14 NOTICE[20671]: app_meetme.c:2210 admin_exec: Conference Number not found
-- Executing Hangup("Local/55558600051@default-0239,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-0239,2'
-- Executing DeadAGI("Local/55558600051@default-0239,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0

[mflorell]

one way audio is usually a SIP/Firewall issue. Either try messing with your firewall settings or try IAX

[Kerry C]

whats the best way to set the router?

[Kerry C]

Why is this here in the CLI: I still cannot hear the other person on the other end:

== Parsing '/etc/asterisk/meetme.conf': Found
Jun 1 10:47:08 NOTICE[13675]: app_meetme.c:2210 admin_ exec: Conference Number not found
-- Executing Hangup("Local/55558600051@default-2bac ,2", "") in new stack

[mflorell]

That specific log entry is for when the agent logs out, that is normal when logging out. Not sure why it says it can't find the conference, what it should say is that there are no participants in the conference.

[Kerry C]

I have this setup like this

T1 to router(Linksys wrt54g2)(vicidial server is coming out of router) then I have a switch hooked to the router ( dell power connect POE) to computers.

I have read up on issues about the linksys router causring problems like this read here.

let me know whats the best way to handle this>

The reason my brain centers on the dyndns is this: up until about a month ago I was running my Asterisk on a Linksys WRT54GS router, which passed traffic wonderfully because the Asterisk was actually on the public side of the router and not in a DMZ or behind forwarded ports. The WRT doesn't have enough horsepower to do switching and voice processing, so I couldn't set up either an IVR function or a Voice Mail function. That led me to shift the Asterisk to a dedicated Intel box.

I installed Asterisk@Home (because I couldn't get Asterisk CVS to install well over Fedora Core 3 on a Pentium 200 MMX), put the Asterisk server in my router's DMZ (the same WRT with Linksys firmware) and migrated my .conf files to the new box. The box roared to life, but would not pass audio to or from FWD or any of my off-prem extensions, although every other connection worked.

It wasn't until I set my externip to my dyndns FQDN that I could get audio on the troubled connections, although the audio would quit every once in a while. That's when I learned to reload my sip.conf after suffering a PPPoE IP address change. Everything has been smooth ever since, which is why I have been fixated on that part of the configuration.

Your question about ping response is simple. Unless you have forwarded the ping port (whose number escapes me) to your Asterisk box, it is your router that is responding to the ping. An easy way to confirm successful forwarding through your router is to forward port 80 (http) or 22 (ssh) through your router to the Asterisk box and see if you can reach AMP from a web browser or open a remote shell or SFTP session via an SSH client, since both of these server daemons are native to Asterisk@Home.

I wouldn't be so quick to change out your Asterisk version because of this problem. First of all, there's a directory /etc/asterisk/default that has clean copies of all of the .conf files should you feel you are irretrievably corrupt somewhere. Secondly, if you have other services and clients working you probably haven't screwed anything up.



*****************************************************

I have port 22 and 80 and 5060 and 10000 - 60000 forwarded to myservers ip.

any suggestions

[Kerry C]

I have tried everything, I have turned the firewall completely off, I have set my externip in sip.conf to my static ip, i have set the localip also, and still i cannot hear a voice, but they can hear me.

Any ideas anyone.

[Kerry C]

I have even set dmz to the servers ip and still nothing, please, any ideas or help would be great

[Kerry C]

can anyone verify this for me:

What I mean is I would use the 2nd NIC and place the IP directly on that card. I was talking with Bob, and this is an issue with asterisk machines. They need to have a public IP.

So, from what I can tell you can be up and running if you configure that live IP without a firewall to the NIC.
Ok 1st step is what you already have.

The 2nd step is to configure the other NIC on your server with with a live IP.

Now on to the networking.

Inbound to your building you have a T1.

Where the T1 is handed off to you there is s ethernet to your firewall / router.

This is what you need to do.

From the T1 place a small DMZ (regular unmanaged switch)
From the switch go to your 2nd NIC on the box with a live IP
If you still need to run the Firewall for other traffic go to that from the switch.

Your firewall will have an IP from the T1 and your server will have an IP from the T1.


would this work ? with my issue?

[gzpxyj]

I have tried everything, I have turned the firewall completely off, I have set my externip in sip.conf to my static ip, i have set the localip also, and still i cannot hear a voice, but they can hear me.

Any ideas anyone.

Please check you codec, test it with ulaw first. sometimes g729 codec will give one way audio if you did not install correctly or missing codec translation module for specific codec combination.
I suspect that it maybe the problem of port forwarding. I would suggest that you check your router port forwarding to your asterisk box. Following ports need to be forwarded to your asterisk box.
10000-20000 udp
16348-32768 udp
4800-5300 both udp & tcp
8000-8012 both
5060-5070 both
10000-11000 udp
4569 both
5004 udp

[Kerry C]

I can now hear the other person on the line, the way I accomplished this problem was this way:

I was running into the problem of not having any sound coming in once I made a call from the agents area, they could hear me but i could not hear them. Well basically I resolved that issue , by getting another static ip on my t1 and using it on the 2nd out port in which i made it a public ip for asterisk to make outbound calls on, I had to configure the interface are in etc/network, to recognize the 2nd nic on my server and add a line of code so the route gateway would not reset on reboot, once this was done and i could ping my 2nd nic ip irregardless if it rebooted or not i was good, I had the voip set the trunk for ip registration on there switch and now i can hear the other person on line and its making calls on manual and ratio predictive with 4 calls going out, now I am going to install the 729 codecs, and I need to know this, what and how do i get the answering machine detection setup, and also since i am only utilizing 7 agents for this server making 4 outbound calls on the predictive would the wrt linksys router handle this data, or does it matter since i am using the 2nd static on the eth 1 . I have learned a hell of alot in 12 days, now i am moving forward with beta testing, but as the questions come to me i will post them here, with results and things i have learned and figured out.

Kerry

- the guy who learned bash in 2 days.

[Kerry C]

also:

on the cli i am seeing this:


Jun 5 10:20:33 NOTICE[5198]: chan_iax2.c:5388 register_verify: No registration for peer 'TESTast' (from xx.xxx.xx.xx)
Jun 5 10:20:33 NOTICE[5198]: chan_iax2.c:7862 socket_read: Registration of 'TESTast' rejected: 'Registration Refused' from: 'xx.xxx.xx.xx'

[mflorell]

Is there an entry for that account in iax.conf?