* You are viewing Posts Tagged ‘inbound calls’

I Can’t Make Outbound Calls (status Is ‘CHANUNAVAIL’)

Hello:

I have this situation: I can make calls internally, I can make inbound calls but I can’t make outbound calls.

Thanks in advance.



These are my devices:
* asterisk 11.8.1 = 192.168.1.22
* sipphone grandstream gxp2160 = 192.168.1.5
* gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4
port 1 (FXS) connected to an analog phone
port 3 (FXO) connected to the PSTN

These are my sip.conf and extensions.conf files:

sip.conf
——

Host = Dynamic In A Register Free Setup

Hello Everyone.

Our environment is a register free setup, and our phones are set as host=dynamic. The problem we are experiencing is for inbound calls:

Name/username Host Dyn Forcerport ACL Port Status Realtime
222/222 (Unspecified) D N A 0
Unmonitored Cached RT

So when we DIAL 222 we get:

WARNING[23103]: app_dial.c:2198 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 – Unknown)

My question is how can we get Asterisk to fill in the gaps (ie, ipaddr, port) for a dynamic peer in a register free environment.

Kind Regards,

Nick.

Weird Issue With Set(CALLERID(name)=string);

Hi folks,

We’ve been having a weird issue… It is happening more often in the last few months…

Most inbound calls, we have in our dialplan before Queue():

Set(CALLERID(name)=${PARTNER}:0:${CALLERID(num)}:${UNIQUEID}:${CHANNEL});

So when the call rings a member, softphone will show this string ….

The issue is that sometimes the string showing in the softphone is not the same. Its a string from a past call, in the latest case I’ve seen, from about 40 days ago!!

User took a screenshot, I’ve searched for that uniqueid showing in softphone in cdr, and that string was valid for a different call 40 days ago!!


I searched full log, and Set() sets the correct string… I can’t figure why softphone shows a string from a past call !!

:(

Any hints ?

Asterisk AMI – PHP Or Node.js?

I would like to develop a Call Center Dialer (outbound and inbound calls)
and it would use AMI method to communicate with Asterisk Server.

A daemon would need to run in the background, would you recommend coding in PHP or Node.js? which would be much faster and stable.

Thanks

Go To Context From Server 1 To Server 2

hello list



i have create i trunk Sip between 2 servers in the same network



when i call a number (inbound calls) in the first server i can forward this number to my sip 222 in the second server



exten => 0522xxxxxx,1,Dial(SIP/222@trunk_created,30)



my question if there is any possibility to GOTO a context in the second server after like below



exten => 0522xxxxxx,1,Dial(SIP/222@ trunk_created,30)

same => 0522xxxxxx,n,GoTo (context in the second server)



thanks and regards

Transfer Only, No Outbound Calling

OK, it’s been a while since I drank from the pool of wisdom hear on the list.

After cracking my head against the wall for a few days trying to figure this out, I have decided to swallow my pride and take the drink.

So, on to my question:

I have some agents/operators setup in sip.conf which point to a context where I have just about disabled outbound calls (only specific numbers can be dialed).

The purpose of this is to allow the inbound calls to come in, then if the operator has a need, they transfer the call to a pre-defined extension which lives in the limited context defined in sip.conf.

This has worked for some time to restrict outbound calling and where calls can be transferred to.

Now I would like to open up the numbers the inbound calls can be transferred to. So, easy enough I thought and I went on my merry way adding the regular patterns to the context such as NXXNXXXXXX and so on.

Hooray, now the operators can transfer anywhere.

New Problem, now operators can pick up the previous inbound only line and dial out to anything that matches the patterns I have defined in the context for their extension in sip.conf.

What I really need to make work here is Attended-Transfer since that is what is desired by those using the system.

It seems that any variables I try to set on the way in don’t carry through too well during an attended transfer.

Basically, I need the ability to know for sure at the point the call ends up in the outbound context (defined in sip.conf) if the call is actually a transfer from an inbound call or if it’s a direct dial outbound call with no incoming call attached. If I can figure out how to know this for sure, I
can just do a GoToIf type of thing in the outbound context that just kills the call if there is no proof that it’s a transfer.

I hope this makes sense, please let me know if more info is needed.

Running Asterisk 1.8.8.0.

A huge thanks in advance to the list for any help with this, it’s driving me batty.

Regards, Todd R.