[asterisk-users] About Sangoma cards and Asterisk integration with other PBX
Daniel Bareiro
daniel-listas at gmx.net
Wed May 26 19:45:07 CDT 2010
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
On Fri, May 21, 2010 at 23:35:14 -0300, Tim Nelson wrote:
> Greetings!
Hi, Tim!
>> I had the opportunity to test a Sangoma A200 card and I have some
>> doubts that I would like to consult:
>>
>> I tried to detect the card and I had no success using the wctdm
>> module with DAHDI. I guess this is because electronics is different
>> because the TDM400P and OpenVox A400P cards have separate modules for
>> each channel, while the Sangoma A200 each module operates two
>> channels. I had to compile Wanpipe for the card was detected. Is it
>> the only way?
> Wanpipe is what interfaces the hardware with Dahdi/Zaptel. Then,
> Dahdi/Zaptel interfaces with Asterisk. This is normal.
Well, then wanpipe is necessary.
>> Another thing I want to try is to connect Asterisk with Siemens PBX
>> so that the extensions on Asterisk can communicate with the
>> extensions on the Siemens PBX and vice versa. For this should I
>> connect a FXO channel on Asterisk with a FXS channel of Siemens PBX?
> Personally, if possible, I'd connect one of each(FXO/FXS) on Asterisk
> to one of each(FXO/FXS) on the Siemens. This allows for proper dialing
> between systems and passing your ${EXTEN} as expected.
I'm not sure if I understood well. Must I use two FXO/FXS connections? A
FXO (Asterisk) / FXS (Siemens) connection and another FXO (Siemens) /
FXS (Asterisk) connection? does not serve a single connection for
incoming and outgoing calls like when we connect Asterisk to the PSTN?
>> I noticed that, unlike OpenVox A400P card, RJ connectors on the
>> Sangoma A200 card are smaller. Apparently, the OpenVox use standard
>> telephone connectors.
> Sangoma's cards come with a half-height PCI bracket for smaller
> systems. To ensure the card stays small, they use smaller jacks, RJ14
> or 'handset' jacks IIRC. Again, this is something specific to Sangoma
> and normal.
Today I was doing tests connecting FXO channel on Sangoma card to a
extension of Siemens PBX. Previously, connecting a phone, I made sure in
that socket I had a dial tone.
I tried calling the extension 509 on Siemens PBX, but I get a busy tone
with the following message in the CLI:
- -------------------------------------------------------------------------
dynatac*CLI>
-- Executing [9509 at from-internal:1] Dial("SIP/200-00000004",
"DAHDI/3/509") in new stack
[May 26 14:47:59] WARNING[3031]: app_dial.c:1298 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 [9509 at from-internal:2] Hangup("SIP/200-00000004", "")
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-00000004'
-- Executing [9509 at from-internal:1] Dial("SIP/200-00000005",
"DAHDI/3/509") in new stack
[May 26 14:48:32] WARNING[3032]: app_dial.c:1298 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 [9509 at from-internal:2] Hangup("SIP/200-00000005", "")
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-00000005'
- -------------------------------------------------------------------------
This is the configuration I'm using in chan_dahdi.conf:
- -------------------------------------------------------------------------
[trunkgroups]
[channels]
language=es
defaultzone=es
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
inmediate=no
; DGB - 20100322
busydetect=yes
busycount=3
;Sangoma AFT-A200 [slot:8 bus:1 span:1] <wanpipe1>
context=from-internal
mailbox=300 at voicemail
callerid="Jane Doe" <300>
group=1
echocancel=yes
signalling = fxo_ls
channel => 1
context=from-internal
group=2
echocancel=yes
signalling = fxo_ks
channel => 2
context=from-zaptel
group=3
echocancel=yes
signalling = fxs_ks
channel => 3
context=from-zaptel
group=4
echocancel=yes
signalling = fxs_ks
channel => 4
- -------------------------------------------------------------------------
And the extensions.conf file is the following:
- -------------------------------------------------------------------------
; DGB - 20100511
[general]
autofallthrough=no
[macro-dial]
exten => s,1,Dial(${ARG1},15)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@voicemail,u)
exten => s-NOANSWER,n,Hangup
exten => s-BUSY,1,Voicemail(${MACRO_EXTEN}@voicemail,b)
exten => s-BUSY,n,Hangup
exten => s-CHANUNAVAIL,1,Playback(pbx-invalid)
[from-internal]
; SIP extensions
exten => _2xx,1,Macro(dial,SIP/${EXTEN})
exten => _2xx,n,Hangup
; analog extension
exten => 300,1,Macro(dial,DAHDI/1)
exten => 300,n,Hangup
; Outgoing calls
exten => _9.,1,Dial(DAHDI/3/${EXTEN:1})
exten => _9.,n,Hangup
[from-zaptel]
include => from-internal
- -------------------------------------------------------------------------
On the other hand, I observed the following:
- -------------------------------------------------------------------------
dynatac:~# cat /proc/dahdi/*
Span 1: WRTDM/0 "wrtdm Board 1" (MASTER)
1 WRTDM/0/0 FXOLS (In use) (SWEC: MG2)
2 WRTDM/0/1 FXOKS (In use) (SWEC: MG2)
3 WRTDM/0/2 FXSKS (In use) RED(SWEC: MG2)
4 WRTDM/0/3 FXSKS (In use) RED(SWEC: MG2)
5 WRTDM/0/4
6 WRTDM/0/5
7 WRTDM/0/6
8 WRTDM/0/7
9 WRTDM/0/8
10 WRTDM/0/9
11 WRTDM/0/10
12 WRTDM/0/11
13 WRTDM/0/12
14 WRTDM/0/13
15 WRTDM/0/14
16 WRTDM/0/15
17 WRTDM/0/16
18 WRTDM/0/17
19 WRTDM/0/18
20 WRTDM/0/19
21 WRTDM/0/20
22 WRTDM/0/21
23 WRTDM/0/22
24 WRTDM/0/23
- -------------------------------------------------------------------------
I understand that if FXO channel is connected, then it would not have to
appear RED. Is it correct?
> A few last thoughts... While OpenVOX may be tempting due to price,
> you'll want to think long and hard about quality and support. Sangoma
> has hands down the best support out of any of the telephony interface
> card manufacturers. Also, the warranty is hard to beat. You will pay
> more for this, but it is worth it to me. In your situation this boils
> down to the importance of the system you're working with. For my
> personal Asterisk boxen at home, I use OpenVOX. They work as expected
> and if they die, I'm not concerned about the 'mission critical' nature
> of my test systems. On the other hand, when we ship telephony
> appliances to customers domestically and around the world and want to
> feel 'comfy and cozy' that things will 'just work', we install a
> Sangoma board.
>
> Please accept my apologies if I sound like I'm on a soapbox trying to
> hardsell Sangoma to you. Frankly, there are very few companies and
> products that impress me any more, and even less so in the IT and
> telephony space. Sangoma happens to be one of these few and I feel I
> must make you aware of it. :-)
Don't worry :-) I appreciate all the information you provided me.
Always is welcome everything is obtained like fruit of the experience,
especially when one is relatively new with Asterisk and VoIP.
Thanks for your reply.
Regards,
Daniel
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.9 (GNU/Linux)
iEYEARECAAYFAkv9wMkACgkQZpa/GxTmHTfe6ACfZtgEdJPqZnGT+DBtqYtpX7sk
XM8Anj+JqKkUGTp0/dIINI9Dobm9/8gA
=NQfT
-----END PGP SIGNATURE-----
More information about the asterisk-users
mailing list