spa 3102 setting with elastix pbx

syedtafazull

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#1
I have connected spa3102 voip getway as ATA for my elastix PBX, i have done all the setting descried in elastix with out tears guide, but when i make call i can here the ringing but at the destination there is no response (no ring noise is there ). what i shoud do for that to solve the problum, please help me soon, THANK U. :S
 

danardf

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#2
Hi syedtafazull and welcome to our Elastix Forum.

Could you discribe exactly your config? (side Elastix and side SPA) :huh:
 

syedtafazull

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#3
this is the setting i have done
Line 1 tab
Proxy: 192.168.1.201
Register Expires: 60
Display Name: stc
User ID: 10
Password: 10
Silence Threshold: medium
DTMF Tx Method: INFO
Hook Flash Tx Method: INFO

PSTN Line tab
Proxy: 192.168.1.201
Register: no
Make Call Without Reg: yes
Ans Call Without Reg: yes
Display Name: No name
User ID: PSTN
Password: password
Silence Supp Enable: no
Echo Canc Enable: no
Echo Canc Adapt Enable: no
Echo Supp Enable: no
FAX CED Detect Enable: yes
FAX CNG Detect Enable: yes
FAX Passthru Codec: G711u
FAX Codec Symmetric: no
FAX Passthru Method: None
DTMF Tx Method: INFO
FAX Process NSE: no
Dial Plan 1: (S0<:myphone number>)
VoIP Caller Default DP: none
PSTN Ring Thru Line 1: no
PSTN CID For VoIP CID: yes
Elastix Without Tears Page 94 of 299
PSTN Answer Delay: 2
PSTN Ring Thru Delay: 3
PSTN Ring Timeout: 4
PSTN Hook Flash Len: .1
Disconnect Tone: 425@-30,425@-30;1(.375/.375/1+2)
FXO Port Impedance: 220+820||120nF
On-Hook Speed: 26ms


Outbound Caller ID: <my phone number> (
Maximum Channels: 1
Dial Rules: 0+NXXXXXXXX (for example)
0011+ZXXXXXXXXXX.
Trunk Name: stc
Peer Details:
canreinvite=no
context=from-pstn
host=192.168.1.200 (spa IP)
insecure=very
nat=no
port=5061 (for example)
qualify=yes
type=peer
username=PSTN
User Context: stc-incoming
User Details:
canreinvite=no
context=from-pstn
host=192.168.1.200
insecure=very
nat=no
port=5061 for example
type=user
username=PSTN
 

danardf

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#4
Side SPA, it seems right.
Side Elastix, in my case, this trunk is made like that:

Code:
canreinvite=no
context=from-trunk
dtmfmode=rfc2833
host=ip_SPA
insecure=very
nat=no
port=5061 
qualify=yes
type=peer
username=pstn
disallow=all
allow=alaw
fromdomain=ip_elastix
But nothing into user context and details.
It works well for me.

About SIP port, Line and PSTN should be different. For example:
5060 and 5061, 5061 and 5062....etc

You could apply also RFC2833 instead of INFO about dtmfmode.

There's some years, i made a little document, but it's in French. However, there's some screen captures. ;)

Look at this HERE, and let me know.



Regards
 

syedtafazull

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#5
i done setting according to elastix with out tear guide as well as u described, but when i make call the destination is ringing but at the destination there is no answer, but for sum destination after one ring it is show in connected but there is no reply from destination.
 

danardf

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#6
Hi.

What's the result of: sip show peers ? (only your trunk line).
 

syedtafazull

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#7
this is the result of (sip show peer PSTN) command

* Name : PSTN
Secret : <Set>
MD5Secret : <Not set>
Remote Secret: <Not set>
Context : from-trunk
Subscr.Cont. : <Not set>
Language :
AMA flags : Unknown
Transfer mode: open
CallingPres : Presentation Allowed, Not Screened
Callgroup :
Pickupgroup :
Mailbox :
VM Extension : *97
LastMsgsSent : 32767/65535
Call limit : 0
Dynamic : No
Callerid : "" <>
MaxCallBR : 384 kbps
Expire : -1
Insecure : no
Nat : No
ACL : No
T.38 support : No
T.38 EC mode : Unknown
T.38 MaxDtgrm: -1
DirectMedia : No
PromiscRedir : No
User=Phone : No
Video Support: No
Text Support : No
Ign SDP ver : No
Trust RPID : No
Send RPID : No
Subscriptions: Yes
Overlap dial : Yes
DTMFmode : rfc2833
Timer T1 : 500
Timer B : 32000
ToHost : 192.168.1.200
Addr->IP : 192.168.1.200 Port 5061
Defaddr->IP : 0.0.0.0 Port 5060
Prim.Transp. : UDP
Allowed.Trsp : UDP
Def. Username: PSTN
SIP Options : (none)
Codecs : 0x4 (ulaw)
Codec Order : (ulaw:20)
Auto-Framing : No
100 on REG : Yes
Status : OK (5 ms)
Useragent :
Reg. Contact :
Qualify Freq : 60000 ms
Sess-Timers : Accept
Sess-Refresh : uas
Sess-Expires : 1800 secs
Min-Sess : 90 secs
Parkinglot :
 

danardf

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#8
Ok.

Now, you could try to make a trace from your call.
Stay on CLI mode, and make a call.
You could see step by step the way to do. For example, if your call take the trunk, the called number ..Etc
 

syedtafazull

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#9
How 2 do call through cli mode plese help me
 

danardf

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#10
If you know to do sip show peer, you know how make a trace.
Simply:
# asterisk -rvvvvvvvvvvvvvvvv
CLI>

Make your call.
Look at this

CLI>[ctrl] + [C] or exit

The result will be something like that.
Code:
-- Executing [s@macro-dialout-trunk:29] Goto("SIP/39014-0000000f", "s-CHANUNAVAIL,1") in new stack
    -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
    -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/39014-0000000f", "RC=1") in new stack
....
    -- Executing [continue@macro-dialout-trunk:4] Set("SIP/39014-0000000f", "CALLERID(number)=39014") in new stack
    -- Executing [35000@from-internal:7] Macro("SIP/39014-0000000f", "outisbusy,") in new stack
    -- Executing [s@macro-outisbusy:1] Progress("SIP/39014-0000000f", "") in new stack
    -- Executing [s@macro-outisbusy:2] GotoIf("SIP/39014-0000000f", "0?emergency,1") in new stack
    -- Executing [s@macro-outisbusy:3] GotoIf("SIP/39014-0000000f", "0?intracompany,1") in new stack
    -- Executing [s@macro-outisbusy:4] Playback("SIP/39014-0000000f", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
    -- <SIP/39014-0000000f> Playing 'all-circuits-busy-now.gsm' (language 'fr')
  == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/39014-0000000f' in macro 'outisbusy'
  == Spawn extension (from-internal, 35000, 7) exited non-zero on 'SIP/39014-0000000f'
    -- Executing [h@from-internal:1] Macro("SIP/39014-0000000f", "hangupcall") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("SIP/39014-0000000f", "1?noautomon") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] NoOp("SIP/39014-0000000f", "TOUCH_MONITOR_OUTPUT=") in new stack
    -- Executing [s@macro-hangupcall:4] GotoIf("SIP/39014-0000000f", "1?noautomon2") in new stack
    -- Goto (macro-hangupcall,s,6)
    -- Executing [s@macro-hangupcall:6] NoOp("SIP/39014-0000000f", "MONITOR_FILENAME=") in new stack
    -- Executing [s@macro-hangupcall:7] GotoIf("SIP/39014-0000000f", "1?skiprg") in new stack
    -- Goto (macro-hangupcall,s,10)
    -- Executing [s@macro-hangupcall:10] GotoIf("SIP/39014-0000000f", "1?skipblkvm") in new stack
    -- Goto (macro-hangupcall,s,13)
    -- Executing [s@macro-hangupcall:13] GotoIf("SIP/39014-0000000f", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,15)
    -- Executing [s@macro-hangupcall:15] Hangup("SIP/39014-0000000f", "") in new stack
  == Spawn extension (macro-hangupcall, s, 15) exited non-zero on 'SIP/39014-0000000f' in macro 'hangupcall'
  == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/39014-0000000f'
 

danardf

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#11
A way, you must find your called number with the good format.
Remember that you have a dial rules:
0+NXXXXXXXX
0011+ZXXXXXXXXXX

This way could modify your called number, so .... a wrong number sent.

You could try without these dial rules, just for test.
 

syedtafazull

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#12

syedtafazull

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#13
yes it is showing that , it is making use of trunk, and the dile pattern that i defined in that trunk.but still not working. is there some other technique to do this.
 

danardf

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#14
Ok.
From Elastix is sending: 054757...etc
So at first step, you must have a dial pattern from outgoing call route.
05xxxxxxxx (for example).
Next, point your PSTN trunk on with this route.

Leave blank dial rules in your trunk.

Make your call. 054757...

Next, you'll see taking up the PSTN line , and more, in the SPA web page, the last called number.
If you could see the last called numbre on this page, i guess that's working.
 

syedtafazull

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#15
it is not showing last called number on spa Graphical web interface, bu on cli of elastix it show , savin useragent "Linksys/SPA3102-3.3.6(GW)"for peer 10.
 

danardf

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#16
By default, the dtmfmode is RFC2833. Try to harmonised this mode.
Include this method in your config trunk.
 

syedtafazull

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#17
ya i have included dtmfmode is RFC2833 in trunk, but thing is that in my spa dtmf mode i set to AUTO,because there is no option of rfc2833
 

danardf

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#18
Right, you could select AUTO.
 

syedtafazull

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#19
after setting auto also it is not working, ia there any another method to do this.
 

syedtafazull

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#20
after setting auto also it is not working, is there any another method to do this.
 

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