Digium Wildcard TE110P T1/E1 Card

Asterisk Asterisk Support
1

Hi!

We have Asterisk 1.6, Panasonic KX-TDA 200. Devices communicate with each other over E1.

In our Asterisk we have Digium Wildcard TE110P T1/E1 card.

All accounts in Asterisk call: Asterisk-> E1 -> TDA - > PSTN.

Now we want make two groups and every groups must have its channel in E1, for example:

one group must have - channel 21,22
other group must have - channel 23,4

In my DAHDI conf file now we have only this:

; Span 2: WCT1/0 “Digium Wildcard TE110P T1/E1 Card 0” HDB3/CCS/CRC4
context=internal
group=12
switchtype = qsig
signalling = pri_cpe
channel => 25-39,41-55

How i must describe every channel in dahdi?

2

You have:

; Span 2: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4
context=internal
group=12
switchtype = qsig
signalling = pri_cpe
channel => 25-39,41-55

To put specific channels in specific groups, do something like:

; Span 2: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" HDB3/CCS/CRC4
context=internal
group=12
switchtype = qsig
signalling = pri_cpe
channel => 25
group=13
channel => 26
group=14
channel => 27
...
etc...
3

Thanks very much!
Will trying.

4

Hi everybody.

I made all that says in top post, in dahdi_channels.conf i have this:

; Span 2: WCT1/0 “Digium Wildcard TE110P T1/E1 Card 0” HDB3/CCS/CRC4
context=internal
group=12
switchtype = qsig
signalling = pri_cpe
channel => 25
;-39,41-55
group=13
channel => 26

In extensions.conf i have this:
[to-tda]
exten => _1XX,1,Dial(DAHDI/g13/${EXTEN},2400,tT)
exten => _1XX,n,Hangup()

exten => _2XX,1,Dial(DAHDI/g13/${EXTEN},2400,tT)
exten => _2XX,n,Hangup()

exten => _3XX,1,Dial(DAHDI/g13/${EXTEN},2400,tT)
exten => _3XX,n,Hangup()

I change the name group on g13 (was g12)

Try to call in 303 internal and have this error:

Verbosity is at least 3
== Using SIP RTP CoS mark 5
– Executing [303@sip-internal:1] Dial(“SIP/451-00000181”, “DAHDI/g13/303,2400,tT”) in new stack
[May 10 14:37:22] WARNING[23350]: app_dial.c:2039 dial_exec_full: Unable to create channel of type ‘DAHDI’ (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing [303@sip-internal:2] Hangup(“SIP/451-00000181”, “”) in new stack
== Spawn extension (sip-internal, 303, 2) exited non-zero on ‘SIP/451-00000181’

If i change name group on g12 - it’s ok? look at this:

Reloading SIP
== Using SIP RTP CoS mark 5
– Executing [303@sip-internal:1] Dial(“SIP/451-00000182”, “DAHDI/g12/303,2400,tT”) in new stack
– Requested transfer capability: 0x00 - SPEECH
– Called g12/303
– DAHDI/i2/303-1ba is proceeding passing it to SIP/451-00000182
– DAHDI/i2/303-1ba is ringing
– Hungup ‘DAHDI/i2/303-1ba’
== Spawn extension (sip-internal, 303, 1) exited non-zero on 'SIP/451-00000182

where is the problem?

5

the problem is resolved:)

i forget about dahdi restart and asterisk restart

6

Hi there. I would appreciate your help.

The TDA 200 (Panasonic) and Elastix box are linked together through an E1. The E1 trunk is sincronized but there are no completed calls from each one of the PBX.

When I call from an extension of the TDA200 to an extension of the Elastix box I get an empty sound and after about 10 seconds I get busy or reorder tone. In the asterisk CLI, I see the following:

New MFC/R2 call detected on chan 19 but nothing more.

Once I hang up I see:

MFC/R2 call disconnected on channel 19
MFC/R2 call end on channel 19

When I call from an extension of the Elastix box to an extension of the TDA 200, I get an empty sound and after a few seconds I get the recorded voice of all circuits are busy. In the asterisk CLI, I see the following:

– Called g1/106
(after a few seconds appears the following)
MFC/R2 call disconnected on channel 17
– DAHDI/17-1 is circuit-busy
– Hungup ‘DAHDI/17-1’
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [s@macro-dialout-trunk:20] NoOp(“SIP/300-00000006”, “Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 0”) in new stack
– Executing [s@macro-dialout-trunk:21] Goto(“SIP/300-00000006”, “s-CONGESTION,1”) in new stack
– Goto (macro-dialout-trunk,s-CONGESTION,1)
– Executing [s-CONGESTION@macro-dialout-trunk:1] Set(“SIP/300-00000006”, “RC=0”) in new stack
– Executing [s-CONGESTION@macro-dialout-trunk:2] Goto(“SIP/300-00000006”, “0,1”) in new stack
– Goto (macro-dialout-trunk,0,1)
– Executing [0@macro-dialout-trunk:1] Goto(“SIP/300-00000006”, “continue,1”) in new stack
– Goto (macro-dialout-trunk,continue,1)
– Executing [continue@macro-dialout-trunk:1] GotoIf(“SIP/300-00000006”, “1?noreport”) in new stack
– Goto (macro-dialout-trunk,continue,3)
– Executing [continue@macro-dialout-trunk:3] NoOp(“SIP/300-00000006”, “TRUNK Dial failed due to CONGESTION HANGUPCAUSE: 0 - failing through to other trunks”) in new stack
– Executing [continue@macro-dialout-trunk:4] Set(“SIP/300-00000006”, “CALLERID(number)=300”) in new stack
– Executing [106@from-internal:5] Macro(“SIP/300-00000006”, “outisbusy,”) in new stack
– Executing [s@macro-outisbusy:1] Progress(“SIP/300-00000006”, “”) in new stack
– Executing [s@macro-outisbusy:2] GotoIf(“SIP/300-00000006”, “0?emergency,1”) in new stack
– Executing [s@macro-outisbusy:3] GotoIf(“SIP/300-00000006”, “0?intracompany,1”) in new stack
– Executing [s@macro-outisbusy:4] Playback(“SIP/300-00000006”, “all-circuits-busy-now&pls-try-call-later, noanswer”) in new stack
– <SIP/300-00000006> Playing ‘all-circuits-busy-now.gsm’ (language ‘en’)
MFC/R2 call end on channel 17
– <SIP/300-00000006> Playing ‘pls-try-call-later.gsm’ (language ‘en’)
== Spawn extension (macro-outisbusy, s, 4) exited non-zero on ‘SIP/300-00000006’ in macro ‘outisbusy’
== Spawn extension (from-internal, 106, 5) exited non-zero on ‘SIP/300-00000006’
– Executing [h@from-internal:1] Macro(“SIP/300-00000006”, “hangupcall”) in new stack
– Executing [s@macro-hangupcall:1] GotoIf(“SIP/300-00000006”, “1?skiprg”) in new stack
– Goto (macro-hangupcall,s,4)
– Executing [s@macro-hangupcall:4] GotoIf(“SIP/300-00000006”, “1?skipblkvm”) in new stack
– Goto (macro-hangupcall,s,7)
– Executing [s@macro-hangupcall:7] GotoIf(“SIP/300-00000006”, “1?theend”) in new stack
– Goto (macro-hangupcall,s,9)
– Executing [s@macro-hangupcall:9] Hangup(“SIP/300-00000006”, “”) in new stack
== Spawn extension (macro-hangupcall, s, 9) exited non-zero on ‘SIP/300-00000006’ in macro ‘hangupcall’
== Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/300-00000006’

I think there is some information missing on both PBX´s. This info is related to the E1 connection.

Waiting for your comments

David Medina

7

Hi there again

Can you please me send some info related to the asterisk configuration and the TDA200 configuration that you have that made succesfully your interconnection?

Waiting for your comments

David Medina