Re: [asterisk-users] About Sangoma cards and Asterisk integration with other PBX
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Wed, Jun 02, 2010 at 21:50:43 -0300, Daniel Bareiro wrote: 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? That might be one way, though I would think, depending on the Siemens hardware, a T1 connection might be more flexible and provide better integration. Lamentably, for the present, I do not believe that we buy a T1 card for Asterisk. As I said in another message of this thread, when trying to communicate with an extension of the Siemens PBX, I obtain busy/congested. When searching on the Internet if Asterisk requires some special configuration to interact with this type of PBX, I found that some Siemens models use proprietary protocols [1], although I'm not sure if the problem I'm having is because of it. Our PBX has two parts. I have understood that the smallest box (than it is on the other) is the DISA. If it serves as something, in the later part it has model 7655. An additional information that I got is that Siemens PBX is Hicom 150. Have you had (or someone on the list) the opportunity to integrate this type of PBX with Asterisk through a analog card? Thanks in advance for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkwH5ywACgkQZpa/GxTmHTer4QCeNdXMum9GU+mOAGCcFkFw5WaL WDMAn1UXK4AKyJKxzx5y4b/Em8tQLiZj =GDSS -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
Re: [asterisk-users] About Sangoma cards and Asterisk integration with other PBX
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, John. On Fri, May 21, 2010 at 23:35:41 -0300, John Novack wrote: 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? That might be one way, though I would think, depending on the Siemens hardware, a T1 connection might be more flexible and provide better integration. Lamentably, for the present, I do not believe that we buy a T1 card for Asterisk. As I said in another message of this thread, when trying to communicate with an extension of the Siemens PBX, I obtain busy/congested. When searching on the Internet if Asterisk requires some special configuration to interact with this type of PBX, I found that some Siemens models use proprietary protocols [1], although I'm not sure if the problem I'm having is because of it. Our PBX has two parts. I have understood that the smallest box (than it is on the other) is the DISA. If it serves as something, in the later part it has model 7655. I noticed that, unlike OpenVox A400P card, RJ connectors on the Sangoma A200 card are smaller. Correct. They are NOT RJ connectors, but 4 position 4 pin modular sockets, as used on US handsets. A better choice, IMO, as the 6 position 4 pin modular sockets can have the release tangs easily caught in the slot. A200 cards are provided when new, with adapter cords that have 4 position sockets on one end and 6 position on the other. Apparently, the OpenVox use standard telephone connectors. As do the Digium cards. NOTE: Using the RJ designation is not correct, though it is widely misused. RJ is an FCC designation for Registered Jack, and refers to the wiring scheme for various interconnections to the public switched network. there are 4 position, 6 position 8 position, and seldom seen 10 position modular plugs and sockets. The 4 position was only used, other than the Sangoma A200, for handsets on modular telephones, and never for PSTN connection, and never had an RJ designation. Misinformation available on the Internet shows various designations. Thanks for the explanation and clarification of nomenclature. And in what cases it would be correct to use the RJ designation? Thanks for your reply. Regards, Daniel [1] http://www.voip-info.org/wiki/view/Siemens+Hicom -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkwG/KgACgkQZpa/GxTmHTdPzgCfc7FtZPSd34tpOC9YNp64ITgw M6wAnRoE2i16KNtN0JUGyizW5eIuam4O =G2xi -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
Re: [asterisk-users] About Sangoma cards and Asterisk integration with other PBX
Daniel Bareiro wrote: SNIP SNIP Thanks for the explanation and clarification of nomenclature. And in what cases it would be correct to use the RJ designation? Thanks for your reply. Regards, Daniel RJ is short for Registered Jack. Strictly speaking it is only correct when it is an interface to the PSTN. An RJ11 is a Registered Jack interface for a single PSTN line, RJ14 for two lines, and RJ25 for three lines. RJ21 is a 50 pin 66 style block interface for up to 25 lines, often seen with orange covers with line numbers written ( or not ) on the inside. There are many others listed in earlier versions of the FCC part 68 regs that are obsolete. Part 68 began in 1978 when all equipment connected to the PSTN had to be either tested and certified and provided with a registration number, or was grandfathered because it was already connected. This program has fallen into disuse in recent times, as the telcos seldom contend that modern equipment can do harm to the public network. Handset cords, since they are internal to a registered piece of equipment, ( the telephone ) never had an RJ designation. Improper use of RJ designations is rampant though, due to lazy or ignorant folks who have no interest or education in the history of the industry. I also fight windmills in my spare time! John Novack -- Dog is my co-pilot -- _ -- 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
Re: [asterisk-users] About Sangoma cards and Asterisk integration with other PBX
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi, Gopalakrishnan. On Fri, May 28, 2010 at 01:44:41 -0300, Gopalakrishnan A.N wrote: 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. H... but this is something that should be reviewed on the side of Siemens PBX? Because I had thought it might be due to a configuration issue in Asterisk FXO channel. The strange thing is that when I connect a phone to that extension of the Siemens PBX, I get dial tone and I can even call to another extension of the Siemens PBX. In fact, callerid in the destination extension indicates that the call comes from the extension 568. Thanks for your reply. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkv/hT4ACgkQZpa/GxTmHTdAJQCgjz4urW3MW5Hcpcu6c0PGaLV0 DhkAn0lpaeYjym8mMrVw65g62EJ1O6O2 =i3xV -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
Re: [asterisk-users] About Sangoma cards and Asterisk integration with other PBX
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-lis...@gmx.netwrote:
-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 [9...@from-internal:1] Dial(SIP/200-0004,
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 [9...@from-internal:2] Hangup(SIP/200-0004, )
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-0004'
-- Executing [9...@from-internal:1] Dial(SIP/200-0005,
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 [9...@from-internal:2] Hangup(SIP/200-0005, )
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-0005'
- -
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=...@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_ext...@voicemail,u)
exten = s-NOANSWER,n,Hangup
exten = s-BUSY,1,Voicemail(${macro_ext...@voicemail,b)
exten = s-BUSY,n,Hangup
exten = s-CHANUNAVAIL,1,Playback(pbx-invalid)
[from-internal]
; SIP extensions
exten = _2xx,1,Macro(dial,SIP/${EXTEN})
exten = _2xx,n,Hangup
; analog extension
exten = 300,1,Macro(dial,DAHDI/1)
exten
Re: [asterisk-users] About Sangoma cards and Asterisk integration with other PBX
-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 [9...@from-internal:1] Dial(SIP/200-0004,
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 [9...@from-internal:2] Hangup(SIP/200-0004, )
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-0004'
-- Executing [9...@from-internal:1] Dial(SIP/200-0005,
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 [9...@from-internal:2] Hangup(SIP/200-0005, )
in new stack
== Spawn extension (from-internal, 9509, 2) exited non-zero on
'SIP/200-0005'
- -
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=...@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
-
[asterisk-users] About Sangoma cards and Asterisk integration with other PBX
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! 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? 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? I noticed that, unlike OpenVox A400P card, RJ connectors on the Sangoma A200 card are smaller. Apparently, the OpenVox use standard telephone connectors. Thanks in advance for your replies. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkv3NNIACgkQZpa/GxTmHTdwTQCfaVv5FZc3T33++JaiVAkgnITs vzYAnicGq+ItJH1tLYf0xMuX/peJjQxe =WVug -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
Re: [asterisk-users] About Sangoma cards and Asterisk integration with other PBX
Daniel Bareiro wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi all! 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. Did you not bother to follow the excellent installation instructions provided by Sangoma?? 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. No that is NOT the reason. It a completely different design. I had to compile Wanpipe for the card was detected. Is it the only way? YES Read the installation instructions provided on the Sangoma website! 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? That might be one way, though I would think, depending on the Siemens hardware, a T1 connection might be more flexible and provide better integration. I noticed that, unlike OpenVox A400P card, RJ connectors on the Sangoma A200 card are smaller. Correct. They are NOT RJ connectors, but 4 position 4 pin modular sockets, as used on US handsets. A better choice, IMO, as the 6 position 4 pin modular sockets can have the release tangs easily caught in the slot. A200 cards are provided when new, with adapter cords that have 4 position sockets on one end and 6 position on the other. Apparently, the OpenVox use standard telephone connectors. As do the Digium cards. NOTE: Using the RJ designation is not correct, though it is widely misused. RJ is an FCC designation for Registered Jack, and refers to the wiring scheme for various interconnections to the public switched network. there are 4 position, 6 position 8 position, and seldom seen 10 position modular plugs and sockets. The 4 position was only used, other than the Sangoma A200, for handsets on modular telephones, and never for PSTN connection, and never had an RJ designation. Misinformation available on the Internet shows various designations. John Novack Thanks in advance for your replies. Regards, Daniel -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.9 (GNU/Linux) iEYEARECAAYFAkv3NNIACgkQZpa/GxTmHTdwTQCfaVv5FZc3T33++JaiVAkgnITs vzYAnicGq+ItJH1tLYf0xMuX/peJjQxe =WVug -END PGP SIGNATURE- Checked by AVG - www.avg.com Version: 9.0.819 / Virus Database: 271.1.1/2886 - Release Date: 05/20/10 14:26:00 -- Dog is my co-pilot -- _ -- 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
Re: [asterisk-users] About Sangoma cards and Asterisk integration with other PBX
- Daniel Bareiro daniel-lis...@gmx.net wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
Greetings!
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.
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 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.
Thanks in advance for your replies.
Regards,
Daniel
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. :-)
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
--
_
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