Call Forwarding with 1 X100P Card - Possible?

Asterisk Asterisk Support
1

Hi all,

I am trying to use * feature where a caller calls my asterisk server via x100p-PSTN line and inputs a particular extensions for the person he/she is trying to reach. I am not getting this feature to work right in the manner I would like. For instance, because i have it connected with openser/asterisk or because I only have one X100P, when the caller calls and enters the extension of the party he/she is trying to reach, (i.e. 31005) they are then transferred to voicemail. All of the extensions that i use have openser as front-end and asterisk as backend. Below is a snippet of my extensions.conf file and my zapata.conf file has a context=default.

extensions.conf file

[incoming]
exten => s,1,Answer()
exten => s,2,Background(enter-ext-of-person)
exten => s,3,WaitExten(10)
exten => 31005,1,Dial(Zap/1,10)
exten => 31005,2,Playback(vm-nobodyavail)
exten => 31005,3,Hangup
exten => 1,1,Playback(digits/1)
exten => 1,2,Goto(incoming,s,1)
exten => 2,1,Playback(digits/2)
exten => 2,2,Goto(incoming,s,2)
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(default,s,1)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()

[default]
exten => 31005,1,Dial(SIP/31005@${OPENSERADDRESS},10,r)
exten => 31005,2,Voicemail(31005@default)
exten => 31005,3,Hangup

How can I accomplish callers calling into the x100p phone line getting transferred to their extensions after entering the desired extension without destroying the current plan of openser’s callers leaving voicemails when users are unavailable/busy. I notice that the call can be transfer to voicemail, so in essence could not it also somehow be transferred to their extensions even with one x100p?

if so, please advise me on how to do so.

TIA

2

why have context=default in zapata.conf when there is no ‘s’ extension in that context ? surely you want it to be context=incoming

and i would think that the IVR option for 31005 should go to default,s,1 too.

3

Hi,
Actually, I had initially had context=incoming for zapata.conf but had numerous troubles getting openser calls relaying with this particular context setting. I suppose because many items in my mysql database for openser/asterisk combination were setup with context=default - to say the least, (just an assumption), so once i changed zapata.conf to default everything worked where I can make outside calls from any sip phone no matter what network a phone may be on, whether nat’d or not, that is, of course, for subscribers to my sip router.

Interesting that you should imply that i should be able to accomplish what I want, because after posing this question to this forum, I have since read different docuwikis that have lead me to believe that i must have at least two x100p’s to transfer a call to either another extension or the PSTN. (1 channel = 1 line).

Do you stand to differ?

4

i have 1 POTS line here, and a half dozen SIP/IAX2 trunks. if a call comes in on the POTS channel, and yes, i can transfer it to other UAs on the network, and transfer it to a destination at the end of a SIP/IAX2 trunk.

if you want to transfer to a destination that can only be reached via a PSTN trunk, then yes, you’ll need a second card if your telco doesn’t support an appropriate feature and you can get the flash to work.

5

Hi all,

I can see how it works in your setup Baconbuttie. But it is not so in my case, my users actually register via the sip router and only use asteisk for vm and vm_main (as a B2BUA), so when i have a call from the pstn line coming in for one of them via asterisk, asterisk transfers accordingly after a caller inputs the “enter-ext-of-person” to their vm as this is what is available to these users in asterisk, and anything else would have to use other channels. Maybe if i had a setup where my users registered with asterisk, it would be different.

So i do see that it also depends upon one’s setup.

6

Hi all,

Just in case anyone else may try something like this, it can definitely be done, I DID IT. You only need the one x100p card and calls coming in thru it can be transferred to any extension you designate, though the setup may be a lil different than with asterisk running without openser.

However, if you need to transfer to a PSTN line, then you need more than one x100p. I just could not believe it could not be done.