Fresh GoAutoDial - inbound/outbound dialplan problems

[iranitehrani]

Hi Folks,
- GoAutoDial 3.0 CE (all updates as of yesterday) - CentOS 5.9
- Dell CS24-SC Dual Xeon Quad-Core L5420 2.5GHz


1- Sip registry is OK
2- softphone is working internally


Inbound call from my Flowroute DID(7025532595):

NOTICE[9094]: chan_sip.c:15495 handle_request_invite: Failed to authenticate user <sip:+16198954365@flowroute>;tag=3587332128-542414


seems like my carrier config is wrong.

I'm using Flowroute for my DID and SIP Termination

I've experimented quite a bit but this is my latest:

Account Entry:
[flowroute]
disallow=all
allow=ulaw
type=friend
username=xxxx
secret=xxxx
host=sip.flowroute
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=port,invite (very flowroute suggestion, was 'insecure=very' previously)
nat=yes

following ports open
port 5060, 10K-25K


If someone has a few minutes to help directly I will pay for your time via PayPal

[iranitehrani]

Globals String: TRUNK9=SIP/flowroute

Dialplan Entry:
exten =>_1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten =>_1NXXNXXXXXX,2,Dial(${TRUNK9}/${EXTEN:1}@flowroute,,tTor)
exten =>_1NXXNXXXXXX,3,Hangup

[iranitehrani]

Sorry for the multiple posts, but the anti-SPAM triggers are pretty tight...



Outbound call:
[Sep 4 18:34:28] NOTICE[9094]: chan_sip.c:13822 handle_response_peerpoke: Peer '8001' is now Reachable. (5ms / 2000ms)
[Sep 4 18:34:38] -- Executing [16197102637@default:1] Dial("SIP/8001-00000031", "SIP/16197102637@flowroute") in new stack
[Sep 4 18:34:38] WARNING[11297]: chan_sip.c:3225 create_addr: No such host: flowroute
[Sep 4 18:34:38] WARNING[11297]: app_dial.c:1310 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
[Sep 4 18:34:38] == Everyone is busy/congested at this time (1:0/0/1)
[Sep 4 18:34:38] == Auto fallthrough, channel 'SIP/8001-00000031' status is 'CHANUNAVAIL'
[Sep 4 18:34:38] -- Executing [h@default:1] DeadAGI("SIP/8001-00000031", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----20-----CHANUNAVAIL----------") in new stack

[williamconley]

1) Welcome to the Party!

2) As you are obviously new here, I have some suggestions to help us all help you:

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.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Remember that Goautodial is not software, it's an installer for Vicidial. Essentially you gave us the make and model of the Rig that drove your car into town and asked us to troubleshoot a problem with your car with that information. LOL Always post your Vicidial version (as requested above) when asking about the dialer software itself (which this question is). That being said ...

4) You have problems with both Inbound and Outbound calling. As a rule, you should authenticate inbound by IP of the provider and your provider will tell you how to authenticate your outbound calls to them. While often related, they should be handled one at a time to avoid confusion.

5) Your globals string looks good, and it used properly in the "dialplan entry", but the dialplan entry is malformed. Note that the "flowroute" name is already included in the variable "TRUNK9" (because that variable contains SIP/flowroute). But you also included flowroute directly in the dialplan entry. So when the system replaces the variable TRUNK9 with SIP/flowroute, you will start with:

Code: Select all

Dial(${TRUNK9}/${EXTEN:1}@flowroute,,tTor)

And end up with:

Code: Select all

Dial(SIP/flowroute/${EXTEN:1}@flowroute,,tTor)

Which cannot be good. LOL. I've never tried it, mind you, but it's likely to cause problems. (if you didn't quite get that, there are two "flowroute" instances inside one "dial" command. Not good!)

Instead try:

Code: Select all

Dial(${TRUNK9}/${EXTEN},,tTor)

Note that I also removed the :1 from ${EXTEN} as this will strip the "1" (first digit) from all your numbers. Most carriers want that one to stay ther.

Next: Be sure your account entry contains this as the first line:

Code: Select all

[flowroute]

This is what SIP/flowroute points to. Without it, there is no "flowroute" for the SIP protocol to look up and dial through. When SIP is invoked, it looks for the name its given in sip-vicidial.conf inside [brackets] known as a context. If you don't have a [flowroute] in there, it will fail and you'll get "No such host: flowroute".

Happy Hunting!

[striker]

Hi try fromuser and fromdomain option in sip setting to receive the incomming call

[flowroute]
disallow=all
allow=ulaw
type=friend
username=xxxx
fromuser=XXXX
secret=xxxx
host=sip.flowroute
fromdomain=sip.flowroute
dtmfmode=rfc2833
context=trunkinbound
qualify=yes
insecure=port,invite (very flowroute suggestion, was 'insecure=very' previously)
nat=yes


For outgoing , as william said changethe dialplan
exten =>_1NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten =>_1NXXNXXXXXX,2,Dial(${TRUNK9}/${EXTEN:1},,tTor)
exten =>_1NXXNXXXXXX,3,Hangup