[asterisk-users] About Sangoma cards and Asterisk integration with other PBX
Gopalakrishnan A.N
saigop at gmail.com
Thu May 27 23:44:41 CDT 2010
I suspect the channel is not ceased correctly in Siemens PBX, since you get
dial tone from Siemens PBX the channel from Asterisk is rejected in your
Siemens PBX.
On Thu, May 27, 2010 at 6:15 AM, Daniel Bareiro <daniel-listas at gmx.net>wrote:
> -----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-----
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Thank you with regards,
Gopalakrishnan A.N,
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100528/61721448/attachment.htm
More information about the asterisk-users
mailing list