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jonforrest.beck at gma...
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 Posted: Thu May 08, 2008 4:22 pm Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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I have a client that is using the Sangoma A200DE with two phone lines
attached.
The problem is:
They use their phone (Grandstream GXP2020) to dial out of the system.
Instead of getting ringing, there is someone on the other end of the
line that happened to dial in at the exact same moment.
So now they are stuck talking with this person, instead of the one the
originally called.
The ZAP channels are in a dial plan context that instructs it to just
dial the office phones.
[zap1]
exten => s,1,Dial(SIP/1001&SIP/1002&SIP/1003)
exten => s,n,Voicemail(1000 at vm)
Anyone know how to get around this?
Thanks! |
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cwallace at lodgingcom...
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| Posted: Thu May 08, 2008 4:44 pm Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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At 5:22 PM on 08 May 2008, Forrest Beck wrote:
| Quote: | I have a client that is using the Sangoma A200DE with two phone
lines attached.
The problem is:
They use their phone (Grandstream GXP2020) to dial out of the system.
Instead of getting ringing, there is someone on the other end of the
line that happened to dial in at the exact same moment.
So now they are stuck talking with this person, instead of the one
the originally called.
The ZAP channels are in a dial plan context that instructs it to
just dial the office phones.
[zap1]
exten => s,1,Dial(SIP/1001&SIP/1002&SIP/1003)
exten => s,n,Voicemail(1000 at vm)
Anyone know how to get around this?
|
This is known in the telephony world as "glare", and there's not much
you can do about it, especially if you only have one line.
If you have multiple lines on an over-ring (or hunt group or whatever
you call it), the best thing to do is find out which way the telco
assigns calls to those lines wrt how they are assigned to the Asterisk
box. And then allocate outgoing calls in the other direction.
On our installation, the calls are allocated from the first FXO port
(Zap/25) up. So we set Asterisk to dial out starting from the last FXO
port in the group by calling Dial(Zap/G2) (capital G means dial down
from last, lowercase g means dial up from first). That minimizes glare.
But, as I said before, if you only have one line, you can't do that...
--
C. Chad Wallace, B.Sc.
The Lodging Company
http://www.skihills.com/
OpenPGP Public Key ID: 0x262208A0
Debian Hint #19: If you're interested in building packages from source,
you should consider installing the apt-src package. |
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bwentdg at pipeline.com
Guest
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| Posted: Thu May 08, 2008 4:57 pm Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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I know that everyone has gaps in their knowledge, but I am just
staggered that
systems are being sold/deployed with such fundamental TELCO workings not
being
understood. Frightening.
C. Chad Wallace wrote:
| Quote: | At 5:22 PM on 08 May 2008, Forrest Beck wrote:
| Quote: | I have a client that is using the Sangoma A200DE with two phone
lines attached.
The problem is:
They use their phone (Grandstream GXP2020) to dial out of the system.
Instead of getting ringing, there is someone on the other end of the
line that happened to dial in at the exact same moment.
So now they are stuck talking with this person, instead of the one
the originally called.
The ZAP channels are in a dial plan context that instructs it to
just dial the office phones.
[zap1]
exten => s,1,Dial(SIP/1001&SIP/1002&SIP/1003)
exten => s,n,Voicemail(1000 at vm)
Anyone know how to get around this?
|
This is known in the telephony world as "glare", and there's not much
you can do about it, especially if you only have one line.
If you have multiple lines on an over-ring (or hunt group or whatever
you call it), the best thing to do is find out which way the telco
assigns calls to those lines wrt how they are assigned to the Asterisk
box. And then allocate outgoing calls in the other direction.
On our installation, the calls are allocated from the first FXO port
(Zap/25) up. So we set Asterisk to dial out starting from the last FXO
port in the group by calling Dial(Zap/G2) (capital G means dial down
from last, lowercase g means dial up from first). That minimizes glare.
But, as I said before, if you only have one line, you can't do that...
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lists at darl.com
Guest
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| Posted: Fri May 09, 2008 7:58 am Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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Al Baker wrote:
| Quote: | I know that everyone has gaps in their knowledge, but I am just
staggered that
systems are being sold/deployed with such fundamental TELCO workings not
being
understood. Frightening.
|
Yep, unbelievable.
This is the reason most PBXs are ground start, is there Zap hardware
that does ground start? I really never looked. At least the outbound
call won't go out on a line with a incoming call if they had ground start.
-Ron |
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eric at fnords.org
Guest
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| Posted: Fri May 09, 2008 8:01 am Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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RE Kushner List Account wrote:
| Quote: | Al Baker wrote:
| Quote: | I know that everyone has gaps in their knowledge, but I am just
staggered that
systems are being sold/deployed with such fundamental TELCO workings not
being
understood. Frightening.
|
Yep, unbelievable.
This is the reason most PBXs are ground start, is there Zap hardware
that does ground start? I really never looked. At least the outbound
call won't go out on a line with a incoming call if they had ground start.
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I don't think the analog cards support anything except FXOLS and FXOKS,
the newer 2400 and 800 analog cards might support this. I believe it is
a driver issue rather than a hardware issue.
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide. |
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drew at oanda.com
Guest
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| Posted: Fri May 09, 2008 8:18 am Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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I think the scary thing is that, for most people, basic knowledge of
telephony was almost impossible to come by outside the opaque and
secretive world of telco.
That is until Asterisk came along!
Perhaps there should be a regulatory requirement to read The Future of
Telephony, cover to cover, before installing any Asterisk system! 
http://www.asteriskdocs.org/
regards,
Drew
Al Baker wrote:
| Quote: | I know that everyone has gaps in their knowledge, but I am just
staggered that
systems are being sold/deployed with such fundamental TELCO workings not
being
understood. Frightening.
C. Chad Wallace wrote:
| Quote: | At 5:22 PM on 08 May 2008, Forrest Beck wrote:
| Quote: | I have a client that is using the Sangoma A200DE with two phone
lines attached.
The problem is:
They use their phone (Grandstream GXP2020) to dial out of the system.
Instead of getting ringing, there is someone on the other end of the
line that happened to dial in at the exact same moment.
So now they are stuck talking with this person, instead of the one
the originally called.
The ZAP channels are in a dial plan context that instructs it to
just dial the office phones.
[zap1]
exten => s,1,Dial(SIP/1001&SIP/1002&SIP/1003)
exten => s,n,Voicemail(1000 at vm)
Anyone know how to get around this?
|
This is known in the telephony world as "glare", and there's not much
you can do about it, especially if you only have one line.
If you have multiple lines on an over-ring (or hunt group or whatever
you call it), the best thing to do is find out which way the telco
assigns calls to those lines wrt how they are assigned to the Asterisk
box. And then allocate outgoing calls in the other direction.
On our installation, the calls are allocated from the first FXO port
(Zap/25) up. So we set Asterisk to dial out starting from the last FXO
port in the group by calling Dial(Zap/G2) (capital G means dial down
from last, lowercase g means dial up from first). That minimizes glare.
But, as I said before, if you only have one line, you can't do that...
|
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
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--
Drew Gibson
Systems Administrator
OANDA Corporation
www.oanda.com |
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eric at fnords.org
Guest
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| Posted: Fri May 09, 2008 8:24 am Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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Drew Gibson wrote:
| Quote: | I think the scary thing is that, for most people, basic knowledge of
telephony was almost impossible to come by outside the opaque and
secretive world of telco.
That is until Asterisk came along!
Perhaps there should be a regulatory requirement to read The Future of
Telephony, cover to cover, before installing any Asterisk system!
http://www.asteriskdocs.org/
|
People that try to "wing it" and install Asterisk when they don't know
telecom just gives people a bad impression of Asterisk and VoIP in
general. This helps nobody except the pocketbook of the consultant.
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide. |
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drew at oanda.com
Guest
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| Posted: Fri May 09, 2008 8:49 am Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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Eric Wieling wrote:
| Quote: | Drew Gibson wrote:
| Quote: | I think the scary thing is that, for most people, basic knowledge of
telephony was almost impossible to come by outside the opaque and
secretive world of telco.
That is until Asterisk came along!
Perhaps there should be a regulatory requirement to read The Future of
Telephony, cover to cover, before installing any Asterisk system!
http://www.asteriskdocs.org/
|
People that try to "wing it" and install Asterisk when they don't know
telecom just gives people a bad impression of Asterisk and VoIP in
general. This helps nobody except the pocketbook of the consultant.
|
but how else do they learn?
--
Drew Gibson
Systems Administrator
OANDA Corporation
www.oanda.com |
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eric at fnords.org
Guest
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| Posted: Fri May 09, 2008 9:57 am Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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Drew Gibson wrote:
| Quote: | Eric Wieling wrote:
| Quote: | Drew Gibson wrote:
| Quote: | I think the scary thing is that, for most people, basic knowledge of
telephony was almost impossible to come by outside the opaque and
secretive world of telco.
That is until Asterisk came along!
Perhaps there should be a regulatory requirement to read The Future of
Telephony, cover to cover, before installing any Asterisk system!
http://www.asteriskdocs.org/
|
People that try to "wing it" and install Asterisk when they don't know
telecom just gives people a bad impression of Asterisk and VoIP in
general. This helps nobody except the pocketbook of the consultant.
|
but how else do they learn?
|
Books are one of the best resources, the Wiki is not *too* bad when it
comes to general telecom stuff. You can also build prototype systems.
No, Asterisk did not suddenly unleash the gates of knowledge in telecom.
All that information was available before Asterisk. What was not
available was info on the specific inner workings of traditional PBXs.
Asterisk and Digium did reduce the hardware cost of building a PBX.
Traidional telecom is actually fairly simple if you compare it with IP
PSTN/IP PBXs. With an IP PBX like Asterisk you need to understand
telecom, IP networking (including routing, NAT, ports), Linux, as well
as Asterisk itself.
--
Consulting for Asterisk, Polycom, Sangoma, Digium, Cisco, LAN, WAN, QoS,
T-1, PRI, Frame Relay, Linux, and network design. Based near
Birmingham, AL. Now accepting clients worldwide. |
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drew at oanda.com
Guest
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| Posted: Fri May 09, 2008 10:41 am Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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Eric Wieling wrote:
| Quote: | Drew Gibson wrote:
| Quote: | Eric Wieling wrote:
| Quote: | Drew Gibson wrote:
| Quote: | I think the scary thing is that, for most people, basic knowledge of
telephony was almost impossible to come by outside the opaque and
secretive world of telco.
That is until Asterisk came along!
Perhaps there should be a regulatory requirement to read The Future of
Telephony, cover to cover, before installing any Asterisk system!
http://www.asteriskdocs.org/
|
People that try to "wing it" and install Asterisk when they don't know
telecom just gives people a bad impression of Asterisk and VoIP in
general. This helps nobody except the pocketbook of the consultant.
|
but how else do they learn?
|
Books are one of the best resources, the Wiki is not *too* bad when it
comes to general telecom stuff. You can also build prototype systems.
|
That's why http://www.asteriskdocs.org/
Is there a better place to start?
| Quote: | No, Asterisk did not suddenly unleash the gates of knowledge in telecom.
All that information was available before Asterisk. What was not
available was info on the specific inner workings of traditional PBXs.
Asterisk and Digium did reduce the hardware cost of building a PBX.
|
The knowledge may have been there but there was no "index". It was like
saying to a Unix newbie "just use the man pages, it's all in there". But
newbies don't know which command to look up!! (there is no index)
Asterisk reduced the cost of entry to the market by a several orders of
magnitude.
This, in turn, allowed more folks to become technically "telephony
literate".
Which, in turn again, made the knowledge exponentially easier to come by.
My start in telephony was thanks to 3Com and their efforts to promote
VoIP in the mid to late 90's but it is from the Asterisk community that
I've learned the most.
| Quote: | Traidional telecom is actually fairly simple if you compare it with IP
PSTN/IP PBXs. With an IP PBX like Asterisk you need to understand
telecom, IP networking (including routing, NAT, ports), Linux, as well
as Asterisk itself.
|
If it helps, I feel the same contempt for the telephony guys trying to
learn IP networking. But I try to remember that I have as much to learn
in their field as they do in mine. (Afraid I'm not much good at humility
either) <grin!>
regards,
Drew
--
Drew Gibson
Systems Administrator
OANDA Corporation
www.oanda.com |
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andersen at mwdental.com
Guest
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| Posted: Mon May 12, 2008 10:09 am Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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Eric Wieling wrote:
| Quote: | People that try to "wing it" and install Asterisk when they don't know
telecom just gives people a bad impression of Asterisk and VoIP in
general. This helps nobody except the pocketbook of the consultant.
|
I agree. But I think that comment is incredibly funny. I'd like to
re-write it for about 20 years ago... (and some even today)
"People that try to "wing it" and install Networks when they don't know
networking just gives people a bad impression of Servers and Computers in
general."
People = Telco Guys
Oh, yes. I saw an entire Cat 5 network on punch blocks one time!
Everybody needs to learn the "other side" before getting involved.
Bill |
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rob at hillis.dyndns.org
Guest
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| Posted: Mon May 12, 2008 5:53 pm Post subject: [asterisk-users] Zap Channels Collide (Incoming & Outgoi |
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There are krone blocks designed for CAT5, and I've seen them in use as well.
However, there's no way I'd be using them for today's networks.
/Especially/ having seen one of these krone blocks used to double-punch
two network ports together.
Bill Andersen wrote:
| Quote: | Oh, yes. I saw an entire Cat 5 network on punch blocks one time!
Everybody needs to learn the "other side" before getting involved.
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