PIAF - Asterisk, Google Voice, and a PSTN line - Tandem transfer function
DocM:
Sry for the delay. For some odd reason, my OBI110 is no longer acting like an extension. I'm in the process of repairing it. Could you tell me how to read the syslog?
Also, for clarification, to test out R and *98, I would replace the ENTERINFO in the context found below, correct?
[custom-tandem-transfer]
exten => _X.,1,Playback(followme/pls-hold-while-try)
exten => _X.,n,senddtmf(ENTERINFO)
exten => _X.,n,senddtmf(${EXTEN})
exten => _X.,n,Hangup()
QBZappy:
DocM,
Quote from: DocM on February 27, 2012, 10:47:57 am
On the other end, the obi110 needs to recognize the flash sent by asterisk or the extension code and perform the flash on the PSTN line.
Finally, my last newb question that I can't figure out is how to generate a debug log (or syslog) through OBI110.
1) Replace the "ENTERINFO" with "R". See what gives. I'm not certain it will work as I don't have an Asterisk to test it on.
2) *98 was an idea of incorporating it into the dial plan.
3) Syslog found here:
System Management->Device Admin->Syslog->
Congifure these settings.
Server,Port, Level
I'm assumeing you have a syslog server available. If not OBihai has a DOS based one somewhere on their web site, which is pretty good. It can be used to output a txt file on your hard disk. A little tricky to use since it is not GUI based. Otherwise Google for one. There are several free ones out there.
4) I was reading there is also the possibility of sending hook flash in Asterisk using another method which involves setting up the tones in the exten. I don't remember which site I saw it on. For such questions a web site dedicated to Asterisk is problably the best place to ask. Although I' sure there are some Asterisk experts who frequent this forum.
DocM:
1) R didn't work. It simply exits as soon as it hits the senddtmf(R) line.
2) Will attempt this route but currently unsure how to implement and whether other user will hear dtmf tones
3) Heads up to anyone who wants to utilize syslog: http://www.obitalk.com/forum/index.php?topic=707.0
4) I didn't understand whether the hook flash was to be performed by asterisk or obi110. If it can be performed by both, I'll also inquire from the asterisk forum about my situation.
While I was further experimenting with the obi110 and asterisk setup, I realized that I can't make calls from my obi110 phone line extension. I want my calls to go to the asterisk server and then be routed to whatever location I want. In asterisk, I have extension 703 setup for obi110. I can send calls to obi110 and answer them using my phone port. However, when I call out from obi110 via the phone port, it states the number I have dialed is not in service.
Looking at asterisk's log, obi110 doesn't seem to be telling asterisk what extension it is.
--- SIP read from UDP:192.168.0.10:5061 --->
INVITE sip:13213214321@192.168.0.12:5060 SIP/2.0
Call-ID: d66935c1@192.168.0.10
Content-Length: 312
CSeq: 8002 INVITE
From: <sip:OBITRUNK1@192.168.0.12>;tag=SP231de4161132a0aa5
Max-Forwards: 70
To: <sip:13213214321@192.168.0.12>
Via: SIP/2.0/UDP 192.168.0.10:5061;branch=z9hG4bK-65f3ebcf;rport
Authorization: DIGEST algorithm=MD5,nonce="08712552",realm="asterisk",response="9b2d40930fc95b3d239abc360b8b3772",uri="sip:13213214321@192.168.0.12:5060",username="OBITRUNK1"
User-Agent: OBIHAI/OBi110-1.3.0.2675
Contact: <sip:OBITRUNK1@192.168.0.10:5061>
Expires: 60
Supported: replaces
Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER
Remote-Party-ID: <sip:OBITRUNK1@192.168.0.12>;party=calling;privacy=off
Content-Type: application/sdp
v=0
o=- 536806 1 IN IP4 192.168.0.10
s=-
c=IN IP4 192.168.0.10
t=0 0
m=audio 16820 RTP/AVP 0 8 18 104 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:104 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=xg726bitorder:big-endian
<------------->
Currently, the only change I made in Physical Interface>PHONE Port is setting PrimarilyLine:SP2 Service. My SP2 Service acts like the trunk for Google Voice and Line. I think I need to alter OutboundCallRoute to state that extension 703 is sending invite to asterisk server, not OBITRUNK1. However, I have no idea what I should change it to.
DocM:
It seems that by altering the OutboundCallRoute, I can make it go, pretty much, where ever I want in my PBX (i.e. to other extensions, IVR). However, I don't know how to make it follow the Outbound Route on the PBX.
QBZappy:
DocM,
Several OBi installation instructions with Asterisk can be found on this forum. This first link by Ad_Hominem is quite detailed.
http://www.obitalk.com/forum/index.php?topic=1157.msg7261#msg7261
http://www.obitalk.com/forum/index.php?topic=140.msg461#msg461
http://michigantelephone.wordpress.com/2011/02/06/how-to-use-the-obihai-obi100-or-obi110-voip-device-as-a-gateway-between-asteriskfreepbx-and-google-voice-andor-the-obitalk-network/
http://www.obitalk.com/forum/index.php?topic=68.msg125#msg125
http://www.obitalk.com/forum/index.php?topic=57.msg3964#msg3964
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