Re: [Asterisk-Users] Asterisk with HT 488 FXO
Pasqualotto Enrico wrote: Is possible that the Inbound routing routed only "from-pstn"? My FXO (300) is in a from-internal! Yes, is possible! -- Pasqualotto Enrico email: [EMAIL PROTECTED] web: http://www.pasqualotto.org -BEGIN GEEK CODE BLOCK- Version: 3.12 GIT d? s: a-- C+++ UL P L++ E--- W++ N++ o K- w--- O-- M V-- PS+ PE+ Y PGP- t--- 5 X R tv-- b+ DI- D+ G e h++ r+ y+ --END GEEK CODE BLOCK-- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with HT 488 FXO
Soner Tari wrote:
> Hi Pasqualotto,
>
> Actually, I've seen your post on Asterisk-Users list yesterday, but I
could
> not understand back then. Now, I've checked your sip configuration
again, I
> think you make a mistake in "type" of sip account. I use "friend" not
> "peer". I am not sure though.
Ok, thanks, now with my new "type" the call from FXO (300) are correctly
forwarded to my extension (204) after n second.
Now I have another problem: I want that the calls from 300 to 204 are
redirected to my ring-group.
With [EMAIL PROTECTED] & Inbound routing I have add these lines in
extension.conf:
-- cut ---
[ext-did]
include => ext-did-custom
exten => s/204,1,SetVar(FROM_DID=s/204)
exten => s/204,2,Goto(ext-group,1,1)
exten => _X./204,1,Goto(s/204)
[ext-group]
include => ext-group-custom
exten => 1,1,Macro(rg-group,ringall,60,,201-202-203-204-205-206)
exten => 1,2,Goto(ext-group,1,1); jump
-- cut -
The calls from context "from-pstn" (SIP account) is also redirected to
ring-group and these work.
I found this in Asterisk CLI:
-- Executing Macro("SIP/300-3bb9", "exten-vm|novm|204") in new stack
-- Executing Macro("SIP/300-3bb9", "user-callerid") in new stack
-- Executing DBget("SIP/300-3bb9", "AMPUSER=DEVICE/300/user") in
new stack
-- DBget: varname=AMPUSER, family=DEVICE, key=300/user
-- DBget: set variable AMPUSER to 300
-- Executing DBget("SIP/300-3bb9",
"AMPUSERCIDNAME=AMPUSER/300/cidname") in new stack
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=300/cidname
-- DBget: set variable AMPUSERCIDNAME to ht488
-- Executing GotoIf("SIP/300-3bb9", "0?5") in new stack
-- Executing SetCallerID("SIP/300-3bb9", ""ht488" <300>") in new stack
-- Executing NoOp("SIP/300-3bb9", "Using CallerID "ht488" <300>")
in new stack
-- Executing SetVar("SIP/300-3bb9", "FROMCONTEXT=exten-vm") in new
stack
-- Executing Macro("SIP/300-3bb9", "record-enable|204|IN") in new stack
-- Executing GotoIf("SIP/300-3bb9", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("SIP/300-3bb9",
"recordingcheck|20060228-133504|1141151704.8") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060228-133504|1141151704.8: Inbound recording not
enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("SIP/300-3bb9", "No recording needed") in new stack
-- Executing Macro("SIP/300-3bb9", "dial|15|tr|204") in new stack
-- Executing GotoIf("SIP/300-3bb9", "0?4:2") in new stack
-- Goto (macro-dial,s,2)
-- Executing GotoIf("SIP/300-3bb9", "0?5:4") in new stack
-- Goto (macro-dial,s,4)
-- Executing AGI("SIP/300-3bb9", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
-- dialparties.agi: priority = 4
-- dialparties.agi: callingani2 = 0
-- dialparties.agi: accountcode =
-- dialparties.agi: channel = SIP/300-3bb9
-- dialparties.agi: callerid = 300
-- dialparties.agi: context = macro-dial
-- dialparties.agi: callington = 0
-- dialparties.agi: dnid = 204
-- dialparties.agi: request = dialparties.agi
-- dialparties.agi: calleridname = ht488
-- dialparties.agi: extension = s
-- dialparties.agi: language = en
-- dialparties.agi: uniqueid = 1141151704.8
-- dialparties.agi: callingpres = 0
-- dialparties.agi: type = SIP
-- dialparties.agi: rdnis = unknown
-- dialparties.agi: callingtns = 0
-- dialparties.agi: enhanced = 0.0
dialparties.agi: Caller ID name and number are '300'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 204 to extension map
-- dialparties.agi: Extension 204 cf is disabled
-- dialparties.agi: Extension 204 do not disturb is disabled
-- dialparties.agi: Checking CW and CFB status for extension 204
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
-- dialparties.agi: Correct AMPMGRUSER and AMPMGRPASS
== Manager 'admin' logged off from 127.0.0.1
dialparties.agi: Extension 204 is available...skipping checks
-- dialparties.agi: DbSet CALLTRACE/204 to 300
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("SIP/300-3bb9", "SIP/204|15|tr") in new stack
-- Called 204
-- SIP/204-1d0a is ringing
-- SIP/204-1d0a answered SIP/300-3bb9
-- Attempting native bridge of SIP/300-3bb9 and SIP/204-1d0a
-- Started music on hold, class 'default', on channel 'SIP/300-3bb9'
-- Stopped music on hold on SIP/300-3bb9
== Spawn extension (macro-dial, s, 10) exited non-zero on
'SIP/300-3bb9' in macro 'dial'
== Spawn extension (macro-exten-vm, s, 4) exited non-zero on
'SIP/300-3bb9' in macro 'exten-vm'
Re: [Asterisk-Users] Asterisk with HT 488 FXO
Hi Pasqualotto,
Actually, I've seen your post on Asterisk-Users list yesterday, but I could
not understand back then. Now, I've checked your sip configuration again, I
think you make a mistake in "type" of sip account. I use "friend" not
"peer". I am not sure though.
Following is what I had in my sip.conf file for the FXO port of HT488:
[41]
username=41
type=friend
secret=
host=dynamic
context=
callerid="Outside-line" <41>
dtmfmode=inband
group=1
callgroup=1
pickupgroup=1
Of course, you should configure HT488 FXO sip account accordingly too. You
should make sure that HT488 registers with Asterisk.
Also read again the following thread:
http://lists.digium.com/pipermail/asterisk-users/2005-August/123548.html
Now, when you call 41 from another phone, you should be able to hear the
dial tone. And if you configured HT488 to answer incomming calls to FXO and
where they should be directed to ("Forward to VoIP" box), then you should be
able to call in HT488 FXO and talk to Asterisk after a few rings. (HT488
configuration is also very important, I don't know what settings you have
there.)
I don't have a HT488 these days, so I cannot test your configurations,
sorry.
Soner
- Original Message -
From: "Pasqualotto Enrico" <[EMAIL PROTECTED]>
To:
Sent: Monday, February 27, 2006 9:54 PM
Subject: [Asterisk-Users] Asterisk with HT 488 FXO
Hi, i have a HT 488 and I want using this like an FXO for Asterisk.
I have find some configuration in the list archive & google but my HT with
these config not work.
my sip.conf
[HT-488]
username=400
type=peer
secret=wowowow
qualify=yes
port=5062
nat=no
host=192.168.1.157
fromuser=400
disallow=all
context=from-pstn
allow=g711u
allow=ulaw
allow=alaw
my sip debug:
--
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport
From: "Unknown" ;tag=as073738f8
To: ;tag=ebc4a8e2
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Grandstream HT488 1.0.2.16
Contact:
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Supported: replaces
Content-Length: 0
--- (11 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
asterisk1*CLI>
<-- SIP read from 192.168.1.157:5062:
SIP/2.0 481 No Such Call
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport
From: "Unknown" ;tag=as073738f8
To: ;tag=52242a6b
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
User-Agent: Grandstream HT488 1.0.2.16
Content-Length: 0
--- (8 headers 0 lines)---
Destroying call '[EMAIL PROTECTED]'
REGISTER 12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.1.157:5060:
REGISTER sip:192.168.1.157 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport
From: ;tag=as558874a4
To:
Call-ID: [EMAIL PROTECTED]
CSeq: 120 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact:
Event: registration
Content-Length: 0
---
Destroying call '[EMAIL PROTECTED]'
asterisk1*CLI>
<-- SIP read from 192.168.1.157:5060:
SIP/2.0 501 Not Implemented
Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport
From: ;tag=as558874a4
To: ;tag=3a733fa7
Call-ID: [EMAIL PROTECTED]
CSeq: 120 REGISTER
User-Agent: Grandstream HT488 1.0.2.16
Content-Length: 0
---
The register string ??
Can anyone help me??
Thanks
--
Pasqualotto Enrico
email: [EMAIL PROTECTED]
web: http://www.pasqualotto.org
-BEGIN GEEK CODE BLOCK-
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O-- M V-- PS+ PE+ Y PGP- t--- 5 X R tv-- b+ DI- D+
G e h++ r+ y+
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[Asterisk-Users] Asterisk with HT 488 FXO
Hi, i have a HT 488 and I want using this like an FXO for Asterisk. I have find some configuration in the list archive & google but my HT with these config not work. my sip.conf [HT-488] username=400 type=peer secret=wowowow qualify=yes port=5062 nat=no host=192.168.1.157 fromuser=400 disallow=all context=from-pstn allow=g711u allow=ulaw allow=alaw my sip debug: -- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport From: "Unknown" ;tag=as073738f8 To: ;tag=ebc4a8e2 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Grandstream HT488 1.0.2.16 Contact: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI> <-- SIP read from 192.168.1.157:5062: SIP/2.0 481 No Such Call Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK669516e2;rport From: "Unknown" ;tag=as073738f8 To: ;tag=52242a6b Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Grandstream HT488 1.0.2.16 Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' REGISTER 12 headers, 0 lines Reliably Transmitting (no NAT) to 192.168.1.157:5060: REGISTER sip:192.168.1.157 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport From: ;tag=as558874a4 To: Call-ID: [EMAIL PROTECTED] CSeq: 120 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: Event: registration Content-Length: 0 --- Destroying call '[EMAIL PROTECTED]' asterisk1*CLI> <-- SIP read from 192.168.1.157:5060: SIP/2.0 501 Not Implemented Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK42700737;rport From: ;tag=as558874a4 To: ;tag=3a733fa7 Call-ID: [EMAIL PROTECTED] CSeq: 120 REGISTER User-Agent: Grandstream HT488 1.0.2.16 Content-Length: 0 --- The register string ?? Can anyone help me?? Thanks -- Pasqualotto Enrico email: [EMAIL PROTECTED] web: http://www.pasqualotto.org -BEGIN GEEK CODE BLOCK- Version: 3.12 GIT d? s: a-- C+++ UL P L++ E--- W++ N++ o K- w--- O-- M V-- PS+ PE+ Y PGP- t--- 5 X R tv-- b+ DI- D+ G e h++ r+ y+ --END GEEK CODE BLOCK-- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
