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Hi I am new to goautodail, I configure all the setting with the help of this forum, as test basis I have installed Goautodail in virutal machine.

my system settings are below, can anyone assist me where I am making mistake.

Carrie name : PULSE INDIA

ACCOUNT ENTRY
[PULSEINDIA]
disallow=all
allow=g729
allow=gsm
allow=ulaw
type=friend
username=XXXXXXX
secret=XXXXXXXXXX
host=PULSE INDIA
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=very

GLOBAL STRING
SIP = SIP/PULSEINDIA

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

I tried as much i can but i am unable to fix it and this the error i am getting.

[Apr 11 22:26:19] -- Executing [917022242326@default:1] AGI("SIP/8001-00000004", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 11 22:26:19] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 11 22:26:19] -- Executing [917022242326@default:2] Dial("SIP/8001-00000004", "SIP/917022242326@PULSEINDIA||tTo") in new stack
[Apr 11 22:26:19] WARNING[25840]: app_dial.c:1310 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Apr 11 22:26:19] == Everyone is busy/congested at this time (1:0/0/1)
[Apr 11 22:26:19] -- Executing [917022242326@default:3] Hangup("SIP/8001-00000004", "") in new stack
[Apr 11 22:26:19] == Spawn extension (default, 917022242326, 3) exited non-zero on 'SIP/8001-00000004'
[Apr 11 22:26:19] -- Executing [h@default:1] DeadAGI("SIP/8001-00000004", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack
[Apr 11 22:26:19] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0


Thanks in advance
Yogesh

Executing [917022242326@default:2] Dial("SIP/8001-00000004", "SIP/917022242326@PULSEINDIA||tTo") in new stack

you are sending that number?

Unable to create channel of type 'SIP' (cause 20 - Unknown)

it could not create the channel. this is an indication that there is a problem with your SIP account.

I would expect you to use "sip show peers" at this point to see if the sip account is registered and/or has a valid IP listed and is "qualified" (ie: "OK" vs "UNREACHABLE"). But you did not post this information (likely cuz you're a newbie! no problem, we'll have a look at the sip account to see if there is anything obvious ...).

so we look in your sip account [PULSEINDIA] and find the "host" is 'PULSE INDIA' which is neither an IP address nor a Fully Qualified Domain Name.

If you changed this to protect the IP of the carrier, next time change it to "xxx.xxx.xxx.xxx" to show your hiding. If you did not, then that is your problem. It must be the IP or domain of the carrier. Otherwise, Asterisk has no way to know where to send these SIP packets. So your SIP account is not valid and the call fails.



Happy Hunting!

PS: Here are your newbie suggestions:

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

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

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

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

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