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.<br><br><div class="gmail_quote">On Thu, May 27, 2010 at 6:15 AM, Daniel Bareiro <span dir="ltr"><<a href="mailto:daniel-listas@gmx.net">daniel-listas@gmx.net</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;"><div class="im">-----BEGIN PGP SIGNED MESSAGE-----<br>
Hash: SHA1<br>
<br>
</div>On Fri, May 21, 2010 at 23:35:14 -0300, Tim Nelson wrote:<br>
<br>
> Greetings!<br>
<br>
Hi, Tim!<br>
<div class="im"><br>
>> I had the opportunity to test a Sangoma A200 card and I have some<br>
>> doubts that I would like to consult:<br>
>><br>
>> I tried to detect the card and I had no success using the wctdm<br>
>> module with DAHDI. I guess this is because electronics is different<br>
>> because the TDM400P and OpenVox A400P cards have separate modules for<br>
>> each channel, while the Sangoma A200 each module operates two<br>
>> channels. I had to compile Wanpipe for the card was detected. Is it<br>
>> the only way?<br>
<br>
> Wanpipe is what interfaces the hardware with Dahdi/Zaptel. Then,<br>
> Dahdi/Zaptel interfaces with Asterisk. This is normal.<br>
<br>
</div>Well, then wanpipe is necessary.<br>
<div class="im"><br>
>> Another thing I want to try is to connect Asterisk with Siemens PBX<br>
>> so that the extensions on Asterisk can communicate with the<br>
>> extensions on the Siemens PBX and vice versa. For this should I<br>
>> connect a FXO channel on Asterisk with a FXS channel of Siemens PBX?<br>
<br>
> Personally, if possible, I'd connect one of each(FXO/FXS) on Asterisk<br>
> to one of each(FXO/FXS) on the Siemens. This allows for proper dialing<br>
> between systems and passing your ${EXTEN} as expected.<br>
<br>
</div>I'm not sure if I understood well. Must I use two FXO/FXS connections? A<br>
FXO (Asterisk) / FXS (Siemens) connection and another FXO (Siemens) /<br>
FXS (Asterisk) connection? does not serve a single connection for<br>
incoming and outgoing calls like when we connect Asterisk to the PSTN?<br>
<div class="im"><br>
>> I noticed that, unlike OpenVox A400P card, RJ connectors on the<br>
>> Sangoma A200 card are smaller. Apparently, the OpenVox use standard<br>
>> telephone connectors.<br>
<br>
> Sangoma's cards come with a half-height PCI bracket for smaller<br>
> systems. To ensure the card stays small, they use smaller jacks, RJ14<br>
> or 'handset' jacks IIRC. Again, this is something specific to Sangoma<br>
> and normal.<br>
<br>
</div>Today I was doing tests connecting FXO channel on Sangoma card to a<br>
extension of Siemens PBX. Previously, connecting a phone, I made sure in<br>
that socket I had a dial tone.<br>
<br>
I tried calling the extension 509 on Siemens PBX, but I get a busy tone<br>
with the following message in the CLI:<br>
<br>
- -------------------------------------------------------------------------<br>
dynatac*CLI><br>
-- Executing [9509@from-internal:1] Dial("SIP/200-00000004",<br>
"DAHDI/3/509") in new stack<br>
[May 26 14:47:59] WARNING[3031]: app_dial.c:1298 dial_exec_full: Unable<br>
to create channel of type 'DAHDI' (cause 0 - Unknown)<br>
== Everyone is busy/congested at this time (1:0/0/1)<br>
-- Executing [9509@from-internal:2] Hangup("SIP/200-00000004", "")<br>
in new stack<br>
== Spawn extension (from-internal, 9509, 2) exited non-zero on<br>
'SIP/200-00000004'<br>
-- Executing [9509@from-internal:1] Dial("SIP/200-00000005",<br>
"DAHDI/3/509") in new stack<br>
[May 26 14:48:32] WARNING[3032]: app_dial.c:1298 dial_exec_full: Unable<br>
to create channel of type 'DAHDI' (cause 0 - Unknown)<br>
== Everyone is busy/congested at this time (1:0/0/1)<br>
-- Executing [9509@from-internal:2] Hangup("SIP/200-00000005", "")<br>
in new stack<br>
== Spawn extension (from-internal, 9509, 2) exited non-zero on<br>
'SIP/200-00000005'<br>
- -------------------------------------------------------------------------<br>
<br>
This is the configuration I'm using in chan_dahdi.conf:<br>
<br>
- -------------------------------------------------------------------------<br>
[trunkgroups]<br>
<br>
[channels]<br>
language=es<br>
defaultzone=es<br>
usecallerid=yes<br>
hidecallerid=no<br>
callwaiting=no<br>
threewaycalling=yes<br>
transfer=yes<br>
echocancel=yes<br>
echotraining=yes<br>
inmediate=no<br>
<br>
; DGB - 20100322<br>
busydetect=yes<br>
busycount=3<br>
<br>
<br>
;Sangoma AFT-A200 [slot:8 bus:1 span:1] <wanpipe1><br>
context=from-internal<br>
mailbox=300@voicemail<br>
callerid="Jane Doe" <300><br>
group=1<br>
echocancel=yes<br>
signalling = fxo_ls<br>
channel => 1<br>
<br>
context=from-internal<br>
group=2<br>
echocancel=yes<br>
signalling = fxo_ks<br>
channel => 2<br>
<br>
context=from-zaptel<br>
group=3<br>
echocancel=yes<br>
signalling = fxs_ks<br>
channel => 3<br>
<br>
context=from-zaptel<br>
group=4<br>
echocancel=yes<br>
signalling = fxs_ks<br>
channel => 4<br>
- -------------------------------------------------------------------------<br>
<br>
And the extensions.conf file is the following:<br>
<br>
- -------------------------------------------------------------------------<br>
; DGB - 20100511<br>
<br>
[general]<br>
autofallthrough=no<br>
<br>
[macro-dial]<br>
exten => s,1,Dial(${ARG1},15)<br>
exten => s,n,Goto(s-${DIALSTATUS},1)<br>
exten => s-NOANSWER,1,Voicemail(${MACRO_EXTEN}@voicemail,u)<br>
exten => s-NOANSWER,n,Hangup<br>
exten => s-BUSY,1,Voicemail(${MACRO_EXTEN}@voicemail,b)<br>
exten => s-BUSY,n,Hangup<br>
exten => s-CHANUNAVAIL,1,Playback(pbx-invalid)<br>
<br>
[from-internal]<br>
<br>
; SIP extensions<br>
exten => _2xx,1,Macro(dial,SIP/${EXTEN})<br>
exten => _2xx,n,Hangup<br>
<br>
; analog extension<br>
exten => 300,1,Macro(dial,DAHDI/1)<br>
exten => 300,n,Hangup<br>
<br>
; Outgoing calls<br>
exten => _9.,1,Dial(DAHDI/3/${EXTEN:1})<br>
exten => _9.,n,Hangup<br>
<br>
[from-zaptel]<br>
<br>
include => from-internal<br>
- -------------------------------------------------------------------------<br>
<br>
On the other hand, I observed the following:<br>
<br>
- -------------------------------------------------------------------------<br>
dynatac:~# cat /proc/dahdi/*<br>
Span 1: WRTDM/0 "wrtdm Board 1" (MASTER)<br>
<br>
1 WRTDM/0/0 FXOLS (In use) (SWEC: MG2)<br>
2 WRTDM/0/1 FXOKS (In use) (SWEC: MG2)<br>
3 WRTDM/0/2 FXSKS (In use) RED(SWEC: MG2)<br>
4 WRTDM/0/3 FXSKS (In use) RED(SWEC: MG2)<br>
5 WRTDM/0/4<br>
6 WRTDM/0/5<br>
7 WRTDM/0/6<br>
8 WRTDM/0/7<br>
9 WRTDM/0/8<br>
10 WRTDM/0/9<br>
11 WRTDM/0/10<br>
12 WRTDM/0/11<br>
13 WRTDM/0/12<br>
14 WRTDM/0/13<br>
15 WRTDM/0/14<br>
16 WRTDM/0/15<br>
17 WRTDM/0/16<br>
18 WRTDM/0/17<br>
19 WRTDM/0/18<br>
20 WRTDM/0/19<br>
21 WRTDM/0/20<br>
22 WRTDM/0/21<br>
23 WRTDM/0/22<br>
24 WRTDM/0/23<br>
- -------------------------------------------------------------------------<br>
<br>
I understand that if FXO channel is connected, then it would not have to<br>
appear RED. Is it correct?<br>
<div class="im"><br>
> A few last thoughts... While OpenVOX may be tempting due to price,<br>
> you'll want to think long and hard about quality and support. Sangoma<br>
> has hands down the best support out of any of the telephony interface<br>
> card manufacturers. Also, the warranty is hard to beat. You will pay<br>
> more for this, but it is worth it to me. In your situation this boils<br>
> down to the importance of the system you're working with. For my<br>
> personal Asterisk boxen at home, I use OpenVOX. They work as expected<br>
> and if they die, I'm not concerned about the 'mission critical' nature<br>
> of my test systems. On the other hand, when we ship telephony<br>
> appliances to customers domestically and around the world and want to<br>
> feel 'comfy and cozy' that things will 'just work', we install a<br>
> Sangoma board.<br>
><br>
> Please accept my apologies if I sound like I'm on a soapbox trying to<br>
> hardsell Sangoma to you. Frankly, there are very few companies and<br>
> products that impress me any more, and even less so in the IT and<br>
> telephony space. Sangoma happens to be one of these few and I feel I<br>
> must make you aware of it. :-)<br>
<br>
</div>Don't worry :-) I appreciate all the information you provided me.<br>
Always is welcome everything is obtained like fruit of the experience,<br>
especially when one is relatively new with Asterisk and VoIP.<br>
<br>
<br>
Thanks for your reply.<br>
<div class="im"><br>
Regards,<br>
Daniel<br>
<br>
-----BEGIN PGP SIGNATURE-----<br>
Version: GnuPG v1.4.9 (GNU/Linux)<br>
<br>
</div>iEYEARECAAYFAkv9wMkACgkQZpa/GxTmHTfe6ACfZtgEdJPqZnGT+DBtqYtpX7sk<br>
XM8Anj+JqKkUGTp0/dIINI9Dobm9/8gA<br>
=NQfT<br>
-----END PGP SIGNATURE-----<br>
<div><div></div><div class="h5"><br>
<br>
--<br>
_____________________________________________________________________<br>
-- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
New to Asterisk? Join us for a live introductory webinar every Thurs:<br>
<a href="http://www.asterisk.org/hello" target="_blank">http://www.asterisk.org/hello</a><br>
<br>
asterisk-users mailing list<br>
To UNSUBSCRIBE or update options visit:<br>
<a href="http://lists.digium.com/mailman/listinfo/asterisk-users" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>
</div></div></blockquote></div><br><br clear="all"><br>-- <br>Thank you with regards,<br>Gopalakrishnan A.N,<br><br><br>