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" <sip:Unknown@192.168.1.200>;tag=as073738f8 To: <sip:192.168.1.157:5062>;tag=ebc40000a8e20000 Call-ID: 4c2059f1770f97d80110fa427976d7e1@192.168.1.200 CSeq: 102 OPTIONS User-Agent: Grandstream HT488 1.0.2.16 Contact: <sip:400@192.168.1.157:5062;user=phone> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '4c2059f1770f97d80110fa427976d7e1@192.168.1.200' 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" <sip:Unknown@192.168.1.200>;tag=as073738f8 To: <sip:192.168.1.157:5062>;tag=522400002a6bffff Call-ID: 4c2059f1770f97d80110fa427976d7e1@192.168.1.200 CSeq: 102 OPTIONS User-Agent: Grandstream HT488 1.0.2.16 Content-Length: 0 --- (8 headers 0 lines)--- Destroying call '4c2059f1770f97d80110fa427976d7e1@192.168.1.200' 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: <sip:400@192.168.1.157>;tag=as558874a4 To: <sip:400@192.168.1.157> Call-ID: 5308aeca055e74a23ebe819d52d03d26@127.0.0.1 CSeq: 120 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Expires: 120 Contact: <sip:s@192.168.1.200> Event: registration Content-Length: 0 --- Destroying call '5308aeca055e74a23ebe819d52d03d26@127.0.0.1' 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: <sip:400@192.168.1.157>;tag=as558874a4 To: <sip:400@192.168.1.157>;tag=3a7300003fa70000 Call-ID: 5308aeca055e74a23ebe819d52d03d26@127.0.0.1 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: pasqu@linux.it 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------
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=<put your password here>
host=dynamic
context=<put your context here>
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" <pasqu@linux.it>
To: <asterisk-users@lists.digium.com>
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" <sip:Unknown@192.168.1.200>;tag=as073738f8
> To: <sip:192.168.1.157:5062>;tag=ebc40000a8e20000
> Call-ID: 4c2059f1770f97d80110fa427976d7e1@192.168.1.200
> CSeq: 102 OPTIONS
> User-Agent: Grandstream HT488 1.0.2.16
> Contact: <sip:400@192.168.1.157:5062;user=phone>
> Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
> Supported: replaces
> Content-Length: 0
>
>
> --- (11 headers 0 lines)---
> Destroying call '4c2059f1770f97d80110fa427976d7e1@192.168.1.200'
> 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" <sip:Unknown@192.168.1.200>;tag=as073738f8
> To: <sip:192.168.1.157:5062>;tag=522400002a6bffff
> Call-ID: 4c2059f1770f97d80110fa427976d7e1@192.168.1.200
> CSeq: 102 OPTIONS
> User-Agent: Grandstream HT488 1.0.2.16
> Content-Length: 0
>
>
> --- (8 headers 0 lines)---
> Destroying call '4c2059f1770f97d80110fa427976d7e1@192.168.1.200'
> 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: <sip:400@192.168.1.157>;tag=as558874a4
> To: <sip:400@192.168.1.157>
> Call-ID: 5308aeca055e74a23ebe819d52d03d26@127.0.0.1
> CSeq: 120 REGISTER
> User-Agent: Asterisk PBX
> Max-Forwards: 70
> Expires: 120
> Contact: <sip:s@192.168.1.200>
> Event: registration
> Content-Length: 0
>
>
> ---
> Destroying call '5308aeca055e74a23ebe819d52d03d26@127.0.0.1'
> 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: <sip:400@192.168.1.157>;tag=as558874a4
> To: <sip:400@192.168.1.157>;tag=3a7300003fa70000
> Call-ID: 5308aeca055e74a23ebe819d52d03d26@127.0.0.1
> 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: pasqu@linux.it
> 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------
>
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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 A@H & 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'
== Spawn extension (from-internal, 204, 1) exited non-zero on
'SIP/300-3bb9'
-- Executing Macro("SIP/300-3bb9", "hangupcall") in new
stack
-- Executing ResetCDR("SIP/300-3bb9", "w") in new stack
-- Executing NoCDR("SIP/300-3bb9", "") in new stack
-- Executing Wait("SIP/300-3bb9", "5") in new stack
-- Executing Hangup("SIP/300-3bb9", "") in new stack
== Spawn extension (macro-hangupcall, s, 4) exited non-zero on
'SIP/300-3bb9' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'SIP/300-3bb9'
asterisk1*CLI>
but the call is send only to extension 204.
Is possible that the Inbound routing routed only "from-pstn"? My FXO
(300) is in a from-internal!
Where is the problem?
Thanks
--
Pasqualotto Enrico
email: pasqu@linux.it
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------
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: pasqu@linux.it 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------