setting up Vicidial for 1 agent

[boybawang]

yeah could be good fo 1 to 5 agents

[noworldorder]

so everything was sort of working... I was able to call but it was VERY broken up. I rebooted the server and then... I could longer communicate with it. I reboot again as well as the router but no luck. I pinged the server with no response.

lI decided to try again with Vicibox (last time a vicidial miracle worker helped me do it from scratch). I am not sure if I input the IP address, mask, and gateway correctly. But presently there is no communication possible with the server.

Is there a way to check the IP, mask, and gateway from the server now that Ubuntu is on it? How do I trouble shoot this?

Thanks a heap for everyone's assistance!
Chris

[williamconley]

log into the interface locally and type

Code: Select all

ifconfig

also

Code: Select all

ip route show

to check name server:

Code: Select all

nano /etc/resolv.conf

and then enjoy

Code: Select all

nano /etc/network/interfaces

to test it if you think it may be online but you have no access to it for some other reason:

Code: Select all

ping www.google.com -c 2

[noworldorder]

thanks william - I tried everything and couldn't get it to work. I tried VicidialNow (instead of Vicibocx) and it is working.

I will go with what is working for now.

I am trying to get add a new carrier and am having some trouble. Wondering if I could get one of you kind computer persons to help me. I need to know exactrly what to add in the "add a new carrier" screen.

Her are the asterisk details:

BASIC Asterisk CONFIGURATION FOR VOIPGO

1. Edit file sip.conf:

*Add/change [general] section to indicate the following parameters:
[general]
dtmfmode = rfc2833
context=from-voipgo
srvlookup=yes
register => SIPUsername:SIPPassword@66.49.255.51/SIPUsername

*Add the following section to handle calls to/from voipgo:
[voipgo]
type=peer
context=from-voipgo
host=66.49.255.51
username=SIPUsername
secret=SIPPassword
fromuser=SIPUsername
fromdomain=66.49.255.51
insecure=very

*Add a section to handle calls to/from your SIP phone. This is just a sample. Refer to Asterisk documentation and your SIP phone documentation for details. 123 is the extension of your phone.
[123]
context=to-voipgo
type=friend
username=123
secret=PHONESECRET
host=dynamic


2. Edit the file extensions.conf:

*Add the following section to route calls FROM voipgo TO your SIP phone with extension 123:
[from-voipgo]
exten => s,1,Dial(SIP/123)

*Add the following section to route calls FROM your SIP phone TO voipgo:
[to-voipgo]
exten => _X.,1,Dial(SIP/${EXTEN}@voipgo)


3. Verify Asterisk operations

*Connect to asterisk console by running:
# asterisk -r
*Verify that Asterisk is registered to voipgo with console command 'sip show registry'
*CLI> sip show registry

Host Username Refresh State
66.49.255.51:5060 (number) 17 Registered

*Verify that your SIP phone is registered to Asterisk with console command 'sip show peers'
pbx*CLI> sip show peers

Name/username 123/123
Host 10.11.22.33
Dyn Nat ACL D
Mask 255.255.255.255
Port 5060
Status Unmonitored

If you see Host as "(Unspecified)" and Port as "0", then your SIP phone is not configured correctly.

*Disconnect from Asterisk by typing "exit".

4. Placing Test Calls

You can make a test call to a traditional landline or mobile phone by dialing either: 1 + the area code and number for calls to the US/Canada
OR
011 + the country code, area code, and number for calls worldwide.

[williamconley]

the add a new carrier screen has sample entries already.

take the sip sample. get a sip VOIP provider.

ordinarily you will then change the "registration string" to one provided by your carrier and you will change the "host" "username" and "secret" in the "account entry" panel

you will then change (hey! I didn't set this up!) 91999 to 81NXX three times in the bottom panel named "Dial Plan".

8 because there is a "hard coded" "9" in extensions.conf that will be in the way if you try to use "9" instead.

you will PROBABLY be required to change "EXTEN:2" to "EXTEN:1" because the number after the : determines the number of digits to be "discarded" at the beginning of the dial string. Your provider is NOT expecting the "9" or "8", that is only to CHOOSE a carrier withing vicidial, and must be discarded. But your carrier IS probably expecting the "1" before the 10 digit phone number ... so if you discard two digits (81) the 1 will be missing and you'll never make a call. so change it to EXTEN:1

you do not need to change "testcarrier" to your provider name, but if you do you must change it in BOTH the account entry "[testcarrier]" and the globals string "SIP/testcarrier" (leave the "SIP/" though!). Those must match for the mapping of the call to work.

neither do you need to change TESTSIPTRUNK to 'DIAL8TRUNK', but if you do, you must change it BOTH in the globals string AND in the "Dial Plan" second line inside the "dial(" command. those must also match to map the call.

that's not a bad idea, but not at all necessary for "getting it going". good idea to come back and do later AFTER you have a functional dialer.

make sure your server ip is correct, no reason why it shouldn't be but it never hurts to check, and be sure it is "ACTIVE" at the bottom

be sure you wait at least a minute before you freak out that it didn't work (vicidial may take up to a minute to write the configuration file to disk and make it active in asterisk).

then remember that your provider may or may not allow instant on, some wait 5 or 10 minutes to activate you after you register before allowing you to call out.

didja get all that?

[noworldorder]

Who are you Willliam? And why are you so helpful? Thanks! (you should write vicidial for dumimes)

well I tried to do what you said and I am timing out on all mycalls. IT said waiting for answer... and then times out.

This is what I input for the carrier (I have XXXX over my details) and maybe you can tell me where I went wrong:

register => XXXXX:XXXXXX/reg2

[voipgo]
type=peer
context=from-voipgo
host=66.49.255.51
username=XXXXXXXX
secret=XXXXXXX
fromuser=XXXXXXX
fromdomain=66.49.255.51
insecure=very


voipgo= SIP/voipgo


exten => _81NXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _81NXX,2,Dial(${voipgo}/${EXTEN:1},,tTor)
exten => _81NXX,3,Hangup

[gmcust3]

Who are you Willliam? And why are you so helpful? Thanks! (you should write vicidial for dumimes)

No doubts abt it !!!

Best Mentor one can have !!!

[williamconley]

i would not recommend using voipgo for the SIP name and the variable name

i would use DIAL8SERVICE in the variable since it is 8 you must dial to get here. on the off chance you later decide to use dial8 for another carrier ... the fact that you dial 8 to get to it is useful, the actual name of the provider is irrelevant.

i also note that you have _81NXX for your exten ... that's not likely to work .. you need _81NXXNXXXXXX on all three lines instead.

the "_" at the front designates it as a "pattern" instead of an "extension" in the system. This pattern is "8" + "1" + "N" (represents any number from 2 through 9) + "XX" (represents any number) + "N" + "XXXXXX". Shortened: 81+10digit number. the 8 is discarded (used to bring the call to this dial8service) and the 1+10digit is sent to the carrier for processing your call.

for your next post, you will likely need to provide a few lines of "asterisk CLI" output if your calls are not successful. Getting close!

[williamconley]

Who are you Willliam? And why are you so helpful? Thanks! (you should write vicidial for dumimes)

No doubts abt it !!!

Best Mentor one can have !!!

but i DO have a price: if you learn from me you MUST contribute to this project. I don't care if you jump into the SVN head or trunk and help code, the bug tracker and help resolve deeper issues (or lighter ones for that matter), or if you come on here and help newbies ... or if you sponsor some "upgrades" at The Vicidial Group ... but do SOMETHING.

and you MUST buy the manual (HIGHLY recommended the minute your system is live if not before). Saves HOURS of headache.

and the MINUTE you get your first system up, begin building your second (with SVN or 2.2.0 or both). so you can have the experience of getting it working, wiping it clean, and getting it working again.

remember: you want to be able to make money with the software for a LOOOONG time. get good at it. and then help newbies do it, too!

[williamconley]

...
[voipgo]
...
host=66.49.255.51
...
fromdomain=66.49.255.51
...

i do not believe that those should be the same.

http://www.asterisk.org/doxygen/trunk/Config_sip.html

; fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
; non-peers, use your primary domain "identity"
; for From: headers instead of just your IP
; address. This is to be polite and
; it may be a mandatory requirement for some
; destinations which do not have a prior
; account relationship with your server.

however, i defer to your provider instructions on that as it CAN be used differently by different providers (and their interpretation or system build)

this can easily be resolved by registering and making a call. if the make the call for you ... the setting is right!

[noworldorder]

Hi William - I will definitely give back to extent that I can (have already made a little $ donation)

for your next post, you will likely need to provide a few lines of "asterisk CLI" output if your calls are not successful. Getting close!

I still have the same problem so how do I go about getting his output?

[williamconley]

log in via PuTTY and enter "asterisk -R"

then enter "set verbose 20"

then you can hold down the enter to "clear the screen" just as you make a call ... then type "exit" to freeze the screen so you can copy and paste.

when you "highlight" an area of the screen in putty, it is NOT necessary to "ctrl c" to copy. the copy is done automatically for you when you highlight.

paste the RELEVANT information here (not 300 lines of code, only a few are generated for each call). assuming SOMETHING is generated for each call! LOL

[williamconley]

http://www.chiark.greenend.org.uk/~sgta ... nload.html

"putty.exe"

you will need the IP address of your server and your password (the user will be "root")

[noworldorder]

Hi William - cool! Some of this text occured before the call. It was being generated as soon as I entered asterisk -R.

It looks like matrix code to me...


Mar 19 17:44:04 WARNING[2641]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0xb7700de 8 (len 399) to 66.49.255.51:5060 returned -1: Network is unreachable
Mar 19 17:44:05 WARNING[2641]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0xb7700de 8 (len 399) to 66.49.255.51:5060 returned -1: Network is unreachable
== 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
Mar 19 17:44:07 WARNING[2641]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0xb7700de 8 (len 399) to 66.49.255.51:5060 returned -1: Network is unreachable
Mar 19 17:44:11 WARNING[2641]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0xb7700de 8 (len 399) to 66.49.255.51:5060 returned -1: Network is unreachable
== Manager 'sendcron' logged off from 127.0.0.1
Mar 19 17:44:15 WARNING[2641]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0xb7700de 8 (len 399) to 66.49.255.51:5060 returned -1: Network is unreachable
Mar 19 17:44:19 WARNING[2641]: chan_sip.c:1082 __sip_xmit: sip_xmit of 0xb7700de8 (len 399) to 66.49.255.51:5060 returned -1: Network is unreachable
Mar 19 17:44:23 NOTICE[2641]: chan_sip.c:5529 sip_reg_timeout: -- Registration for 'reg2502209872@66.49.255.51' timed out, trying again (Attempt #268)
Mar 19 17:44:23 WARNING[2641]: acl.c:313 ast_ouraddrfor: Cannot connect

[williamconley]

test:

Code: Select all

ping www.google.com -c 2

try:

Code: Select all

ping 66.49.255.51 -c 2

if you cannot ping google, you have a network issue

if you can ping google but not 66.49.255.51, then there is a problem with 66.49.255.51 (or they don't respond to pings).

if you can ping both, then they are ignoring your sip call attempt for some reason
try:

Code: Select all

asterisk -rx "sip show registry"

or

Code: Select all

asterisk -R

then (at the asterisk CLI):

Code: Select all

sip show registry

[noworldorder]

so this would indicate a network problem:

[root@vici ~]# ping www.google.com -c 2
ping: unknown host www.google.com
[root@vici ~]# ping www.google.com
ping: unknown host www.google.com
[root@vici ~]#

[williamconley]

yup

Code: Select all

cat /etc/resolv.conf

[noworldorder]

[root@vici ~]# cat /resolv.conf
cat: /resolv.conf: No such file or directory
[root@vici ~]# cat.resolv.conf
-bash: cat.resolv.conf: command not found
[root@vici ~]#

[williamconley]

you're not very good at copy and paste, are you?

look again.

(etc)

[noworldorder]

[root@vici ~]# cat /etc/resolv.conf
nameserver 208.67.222.222
[root@vici ~]#

[noworldorder]

and what does this tell me other than I have graduated from cut and paste 100

[williamconley]

can you ping that ip address? cuz it's theoretically your nameserver ... if you can't ping it you cannot use DNS .. without dns you really don't have "net". that's a start.

you can always use google's dns instead if that one is not working for you ...

with the ping command always add "-c 2" so you only ping twice (instead of forever, which is rude).

[williamconley]

this is a problem that does not ordinarily occur. before you begin you should be aware of your server's basic network settings so you can enter them during "setup" and immediately verify that you have set everything up correctly at the end by using a few "hey, did it work?" commands to see if your install was (likely) successful:

Code: Select all

ping www.google.com -c 2

SHOULD result in a few lines telling how it was 0% losss (You'll have to read a bit, but it's in there and fairly obvious)

Code: Select all

asterisk -R

which will tell you your asterisk version and give you an asterisk command line (exit to get back out) and

Code: Select all

screen -list

which should show you a list of 8 or 9 "screens" running in detached mode (this is actually Vicidial, running)

Now: During the installation of VicidialNOW, there is no actual network required (all the code resides on the .iso). which is COOL, until you find out you do not have net. On ViciBox, you MUST have net during installation because it will download updates several times during several stages of the installation. If you have no net, several stages will fail and you'll have to do it over.

So you are in the happy land of VicidialNOW's all-inclusive code, but the not-so-happy land of still needing to know your proper network settings to complete the install and FUNCTION.

You must have:

The IP Address of your intended server
*which should NOT interfere with any other computer on your network
*but SHOULD be on your local subnet so your other computers can communicate with it.

The Subnet Mask of your local subnet
*must be paired with the IP address
*ordinarily it is 255.255.255.0 for most local subnets

The DNS server
*often we just use the local address of the router
*one can also use a public one, or even Google's (http://code.google.com/speed/public-dns/)

The local subnet
*generally just the same as the IP address, but ends in 0
*sample: 192.168.2.0

Broadcast address
*generally just the same as the IP address, but ends in 255
*sample: 192.168.2.255

Gateway address
*The IP Address of your "Router" or "Gateway" as it is also called
*Many routers come pre-configured with an address like "192.168.0.1"

On your "PC" you can type "ipconfig /all" and it will rat out your router for you. Look for "Default Gateway" in the mess that comes out. You will also likely find your "Subnet Mask"

[noworldorder]

above and beyond the call as usual...

Perhaps the best thing would be to start over with Vicibox and the new information about my server that I will soon uncover. What do you think?

And I tried "ipconfig /all" like you said but it didn't work (I am sure I did something wrong) so I did "ifconfig". Is that useful? This is what I got:

[root@vici ~]# ipconfig /all
-bash: ipconfig: command not found
[root@vici ~]# ifconfig
eth0 Link encap:Ethernet HWaddr 00:1E:33:36:70:CD
inet addr:192.168.1.2 Bcast:192.168.1.255 Mask:255.255.255.0
inet6 addr: fe80::21e:33ff:fe36:70cd/64 Scope:Link
UP BROADCAST RUNNING MULTICAST MTU:1500 Metric:1
RX packets:290941 errors:0 dropped:0 overruns:0 frame:0
TX packets:82657 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:1000
RX bytes:58911878 (56.1 MiB) TX bytes:19670085 (18.7 MiB)
Interrupt:50 Base address:0xc000

lo Link encap:Local Loopback
inet addr:127.0.0.1 Mask:255.0.0.0
inet6 addr: ::1/128 Scope:Host
UP LOOPBACK RUNNING MTU:16436 Metric:1
RX packets:1693603 errors:0 dropped:0 overruns:0 frame:0
TX packets:1693603 errors:0 dropped:0 overruns:0 carrier:0
collisions:0 txqueuelen:0
RX bytes:132951508 (126.7 MiB) TX bytes:132951508 (126.7 MiB)

[root@vici ~]#

[williamconley]

ipconfig /all on your PC will give you the WORKING parameters that your PC is successfully using to get to the internet. this can be helpful when attempting to get the ones for your DIALER. that's where i was going with that.

probly explained too much.

and yes, i'd recommend re-installing SEVERAL TIMES with both vicidialnow and vicibox.

do NOT be afraid to do it again! seriously. i CAN be more useful with vicibox (memorized) than with vicidialNOW (memorized ... but forgotten where everything is! have to remember every time).

where did you get the value for the DNS server that was in the resolv.conf file? (the server didn't "guess", did it? YOU gave it that address ...)

on the other hand, if you find out what DNS server your PC is using, build it again with that. it obviously works for your PC, right?

[noworldorder]

Hi Willaim,

I adapted the network settings per Demian Biscocho's guidance:

Run: "system-config-network" on the VicidialNOW server. Save and quit. Run: "service network restart" afterwards.

I knew what to enter based on your posts.

It worked as I get a response when ping google.com. The dialer, however, is still waiting for a ring. The results of the call are as follows:

vici*CLI>
== 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
> Channel SIP/cc100-09c167e8 was answered.
-- Executing MeetMe("SIP/cc100-09c167e8", "8600051|F") 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
-- Executing MeetMe("Local/8600051@default-c65f,2", "8600051|F") in new stack
> Channel Local/8600051@default-c65f,1 was answered.
== Starting Local/8600051@default-c65f,1 at default,912503802844,1 failed so falling back to exten 's'
== Starting Local/8600051@default-c65f,1 at default,s,1 still failed so falling back to context 'default'
-- Sent into invalid extension 's' in context 'default' on Local/8600051@default-c65f,1
-- Executing Playback("Local/8600051@default-c65f,1", "invalid") in new stack
-- Playing 'invalid' (language 'en')
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-09c12fd8'
-- Executing DeadAGI("SIP/cc100-09c12fd8", "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
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-ed88,2'
-- Executing DeadAGI("Local/8600051@default-ed88,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- Executing Hangup("Local/55558600051@default-998d,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-998d,2'
-- Executing DeadAGI("Local/55558600051@default-998d,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- Playing 'conf-onlyperson' (language 'en')
Mar 20 03:31:00 WARNING[11234]: file.c:1045 ast_waitstream: Unexpected control subclass '-1'
-- 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
== 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
== Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-c65f,2'
-- Executing DeadAGI("Local/8600051@default-c65f,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
-- Executing DeadAGI("Local/8600051@default-c65f,1", "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
== Manager 'sendcron' logged off from 127.0.0.1
== Refreshing DNS lookups.
== Manager 'sendcron' logged off from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
vici*CLI> exit
[root@vici ~]#
[root@vici ~]#

[noworldorder]

I followed your other instructions:

ping 66.49.255.51 -c 2


[root@vici ~]# ping 66.49.255.51 -c 2
PING 66.49.255.51 (66.49.255.51) 56(84) bytes of data.
64 bytes from 66.49.255.51: icmp_seq=1 ttl=56 time=98.1 ms
64 bytes from 66.49.255.51: icmp_seq=2 ttl=56 time=96.0 ms


try Code asterisk -rx "sip show registry"


--- 66.49.255.51 ping statistics ---
2 packets transmitted, 2 received, 0% packet loss, time 1000ms
rtt min/avg/max/mdev = 96.082/97.093/98.105/1.058 ms




[root@vici ~]# asterisk -rx "sip show registry"
Host Username Refresh State
66.49.255.51:5060 reg250220987 105 Registered
Verbosity is at least 20
[root@vici ~]#
[root@vici ~]#

[williamconley]

912503802844

i believe your dial service is set to "dial 8" not dial 9.

try

812503802844

campaign prefix 8 if you were using auto-dial or another campaign

[noworldorder]

soooooo close! It is calling now! I have it set to my home phone and it works however there is no sound. I have tested the viop line and it is find when not going through the dialer.

So I am just about there!

[noworldorder]

so - as I said - it calls but there is no sound. here are the results of the call:

thanks...


vici*CLI>
== 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
> Channel SIP/cc100-09c1b640 was answered.
-- Executing MeetMe("SIP/cc100-09c1b640", "8600051|F") 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
-- Executing MeetMe("Local/8600051@default-4ab0,2", "8600051|F") in new stack
> Channel Local/8600051@default-4ab0,1 was answered.
-- Executing AGI("Local/8600051@default-4ab0,1", "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-4ab0,1", "SIP/12503802844@Voipgo-outbound||o") in new stack
-- Called 12503802844@Voipgo-outbound
-- SIP/Voipgo-outbound-09b35288 is ringing
-- Hungup 'Zap/pseudo-1697360583'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-09c0a660'
-- Executing DeadAGI("SIP/cc100-09c0a660", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
Mar 20 15:45:18 NOTICE[3352]: app_meetme.c:2210 admin_exec: Conference Number not found
-- Executing Hangup("Local/55558600051@default-7e83,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-7e83,2'
-- Executing DeadAGI("Local/55558600051@default-7e83,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (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
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/Voipgo-outbound-09b35288 answered Local/8600051@default-4ab0,1
== 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
== Spawn extension (default, 912503802844, 2) exited non-zero on 'Local/8600051@default-4ab0,1'
-- Executing DeadAGI("Local/8600051@default-4ab0,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----21-----13") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -21-----13 completed, returning 0
== 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
== 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-afff,2", "8600051|K") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
vici*CLI> exit
[root@vici ~]#

[williamconley]

Executing Dial("Local/8600051@default-4ab0,1", "SIP/12503802844@Voipgo-outbound||o")

what do you have in your "dial plan" for this carrier?

||o is not correct

[noworldorder]

I tried the dial plan in the VicidialNOW manual:

[voipgo-outbound]
type=friend
dtmfmode=auto
host=66.49.255.51
context=outbound
username=**********
fromuser=**********
trustrpid=yes
sendrpid=yes
secret=**********
allow=all
nat=yes

exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(SIP/${EXTEN:1}@Voipgo-outbound,,o)
exten => _91NXXNXXXXXX,3,Hangup

[williamconley]

Called 12503802844@Voipgo-outbound
-- SIP/Voipgo-outbound-09b35288 is ringing
-- Hungup 'Zap/pseudo-1697360583'
== Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/cc100-09c0a660'
-- Executing DeadAGI("SIP/cc100-09c0a660", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
Mar 20 15:45:18 NOTICE[3352]: app_meetme.c:2210 admin_exec: Conference Number not found
-- Executing Hangup("Local/55558600051@default-7e83,2", "") in new stack
== Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-7e83,2'
-- Executing DeadAGI("Local/55558600051@default-7e83,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '8600051'
-- Playing 'conf-onlyperson' (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
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/Voipgo-outbound-09b35288 answered Local/8600051@default-4ab0,1

so you're in the conference room. you manually dial a number. as soon as it dials out, you are disconnected from the conference room and then log back in before the call you just made finishes ringing and answers. that seems a little odd to me.

i also note that you are not using the global variable at all (which is ok, a lot of people see it as a formality and skip ALL SORTS of steps). which brings up a point ... what other steps did you skip? i don't generally do the vicidialNOW setup (or follow that manual). in my manual it says to use Dial(${TESTSIPTRUNK}/${EXTEN:1},,tTor) (the globals variable and ,,tTor, instead of the SIP trunk direct and ,,o).

None of that would explain why your call disconnects after you dial .. but then again i suppose you could have gotten impatient and hung up while it was connecting, too.

can you do that again and explain to me what happened when each line executes, if you can look while it is happening a couple times ...?

and DO try to decipher the CLI entries, they don't seem to make much sense until you can observe them while making calls/attempts and see what happens at what stage. there will come a time you will read it just like the matrix.

[williamconley]

and our NEXT trip will likely be "sip debug" followed by "sip no debug" to exit debug mode.

[noworldorder]

and DO try to decipher the CLI entries, they don't seem to make much sense until you can observe them while making calls/attempts and see what happens at what stage. there will come a time you will read it just like the matrix.

Yes Morpheus...

I am not intentionally trying to skip steps. I want this to work! Neither in the manager's manual nor the VicidialNOW manual is there an example of what to place in the global string blank.

I tried altering the dial pllan but any changes I make stop the calls from ringing.

I have some screen shots of the CLI entries so you can see what happened when. http://www.mediafire.com/?oojxndqfd12

Thank you x 10000^2


vici*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
> Channel SIP/cc100-09c1b640 was answered.
-- Executing MeetMe("SIP/cc100-09c1b640", "8600051|F") 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
-- Executing MeetMe("Local/8600051@default-69a9,2", "8600051|F") in new stack
> Channel Local/8600051@default-69a9,1 was answered.
-- Executing AGI("Local/8600051@default-69a9,1", "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-69a9,1", "SIP/12503802844@voipgo-outbound||o") in new stack
-- Called 12503802844@voipgo-outbound
-- SIP/voipgo-outbound-09c134a0 is ringing
-- SIP/voipgo-outbound-09c134a0 answered Local/8600051@default-69a9,1
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Spawn extension (default, 912503802844, 2) exited non-zero on 'Local/8600051@default-69a9,1'
-- Executing DeadAGI("Local/8600051@default-69a9,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----35-----31") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -35-----31 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
== 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
== 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
== 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
== 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
== 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
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-b9c4,2", "8600051|F") in new stack
> Channel Local/8600051@default-b9c4,1 was answered.
-- Executing AGI("Local/8600051@default-b9c4,1", "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-b9c4,1", "SIP/12503802844@voipgo-outbound||o") in new stack
-- Called 12503802844@voipgo-outbound
-- SIP/voipgo-outbound-09c134a0 is ringing
-- SIP/voipgo-outbound-09c134a0 is ringing
-- Got SIP response 486 "Busy here" back from 66.49.255.51
-- SIP/voipgo-outbound-09c134a0 is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing Hangup("Local/8600051@default-b9c4,1", "") in new stack
== Spawn extension (default, 912503802844, 3) exited non-zero on 'Local/8600051@default-b9c4,1'
-- Executing DeadAGI("Local/8600051@default-b9c4,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----17-----BUSY----------") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
-- 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
== 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
== 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
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
Mar 20 20:23:28 NOTICE[2641]: chan_sip.c:11742 do_monitor: Disconnecting call 'SIP/cc100-09c1b640' for lack of RTP activity in 61 seconds
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing MeetMe("Local/8600051@default-2dab,2", "8600051|F") in new stack
> Channel Local/8600051@default-2dab,1 was answered.
-- Executing AGI("Local/8600051@default-2dab,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-2dab,1", "SIP/12503802844@voipgo-outbound||o") in new stack
-- Called 12503802844@voipgo-outbound
== Manager 'sendcron' logged off from 127.0.0.1

[/img]

[williamconley]

-- Executing Dial("Local/8600051@default-69a9,1", "SIP/12503802844@voipgo-outbound||o") in new stack
-- Called 12503802844@voipgo-outbound
-- SIP/voipgo-outbound-09c134a0 is ringing
-- SIP/voipgo-outbound-09c134a0 answered Local/8600051@default-69a9,1

what i didn't get was what was happening during all of this.

did this call connect? without sound? nothing? something?

remember, you're there, i'm here ... for all i know your speakers were off or something. it looks like it connected the call ...

[noworldorder]

the phone rang - I answered it - and there was nothing - no sound on the phone - no sound on my computer head phones.

[williamconley]

! oh! well that's easier.

Router settings

Use your router's Port Forwarding or Virtual Server or Application Settings (they use different terminology in every single freakin one of them!) to pass ports 5060 (UDP & TCP) & 4569(UDP & TCP) to your router, and the port RANGE of 10000-45000 (UDP only) as well.

As an alternative, for testing, you can switch to IAX (instead of SIP) and it will likely begin working. NOTE: (Only if your provider allows IAX, which will require setting up an IAX connection on the provider's system, you cannot use SIP settings for IAX!).

IAX generally has no issue with the Router (breaks right through). So if IAX works, and sip doesn't it's a router issue ... but who cares cuz it works now! LOL

There are several posts on this forum regarding the process and ports in detail.

[noworldorder]

thanks william - it was the router settings... it works now!

Now I will do as you suggested and start over with vicibox

[williamconley]

Cool beans. Do it several times with both vicibox and vicidialnow until you don't even have to read anything to do the install.

then upgrade to 2.2.0!