RE: [Asterisk-Users] Dial via sip gateway?
The mediatrix does have unique username/passwd for each port. At least the 1104 FXS does. Each port can be registered separately with *. I assume other way round should work as well then. regards, Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Greg Hill Sent: Sunday, February 01, 2004 10:01 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Dial via sip gateway? On Sun, 1 Feb 2004, Rich Adamson wrote: I don't believe the above will work. There is only one IP address for the box, and no way that I've found to send a sip packet to the box with additional information that would suggest using port 1 vs port 2. From what others have hinted at (and it seems the majority of us are limited either to what's printed or experimentation), the 1204 has an internal function that kind of resembles a trunk group. It decides which port to use. As mentioned previously, the sip register function in the box is inop in both directions, therefore there does not seem to be a way to address the ports through contexts or anything else. Mediatrix has provided the mib variables where one can enter a different password for each port, but that has no value either since the register function doesn't work. What happens if you don't use a register = line in sip.conf, but do include a section like: [mediatrixport1] username= password= host= Just to check my theory, I did some testing via fwd. I discovered that if I include a register = line with my fwd info, then when I call my fwd number (outbound through iaxtel) it rings in. But I can't call out via fwd. So then I put in my [fwd] service definition, removed the register line, and waited for the old registration to expire. Then I tried calling my fwd number (again through iaxtel). This time I got the message about the user being offline. But now I can call out via fwd, even though calls wouldn't come in. This demonstrates that the [fwd] section is used by Dial() when I try to place a call out through that service, and that the register line isn't needed for the outbound call. Somebody mentioned that the mediatrix lets you set a unique username/password for each of its ports. It seems that you could set up four service definitions, each using a different user/pwd pair. Then * will use a different user/pwd pair to log in to the mediatrix, depending upon which service definition was called for by the Dial() statement. Or does the mediatrix not really have a distinct user/pwd pair for accessing each port? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dial via sip gateway?
That makes a lot of sense. It would appear the Mediatrix marketing target was for the 1104 (FXS) and 1204 (FXO) to be used in pairs as a toll bypass mechanism across the Internet (mostly in a standalone form without a sip proxy). That is exactly how their extensive documentation is written as well. Looking at it from that perspective, the originating end (1104 fxs) is where we'd place the 'register' function if we were designing the product, and the 1204-fxo is just considered a bunch of pstn CO lines that ordinarily would not need the register function (that it doesn't have it now). Given the software authors probably shared common libraries across the two products, it also suggests why the 1204 has the snmp mib variables for entering the username:password on a per-port basis even though they do nothing with them today. If they can get the 1204 enhanced a little more and drop the retail price by a little, looks like it would make a good 4-port pstn box that really isn't addressed very well in the market today. Rich The mediatrix does have unique username/passwd for each port. At least the 1104 FXS does. Each port can be registered separately with *. I assume other way round should work as well then. regards, Dave -Original Message- On Sun, 1 Feb 2004, Rich Adamson wrote: I don't believe the above will work. There is only one IP address for the box, and no way that I've found to send a sip packet to the box with additional information that would suggest using port 1 vs port 2. From what others have hinted at (and it seems the majority of us are limited either to what's printed or experimentation), the 1204 has an internal function that kind of resembles a trunk group. It decides which port to use. As mentioned previously, the sip register function in the box is inop in both directions, therefore there does not seem to be a way to address the ports through contexts or anything else. Mediatrix has provided the mib variables where one can enter a different password for each port, but that has no value either since the register function doesn't work. What happens if you don't use a register = line in sip.conf, but do include a section like: [mediatrixport1] username= password= host= Just to check my theory, I did some testing via fwd. I discovered that if I include a register = line with my fwd info, then when I call my fwd number (outbound through iaxtel) it rings in. But I can't call out via fwd. So then I put in my [fwd] service definition, removed the register line, and waited for the old registration to expire. Then I tried calling my fwd number (again through iaxtel). This time I got the message about the user being offline. But now I can call out via fwd, even though calls wouldn't come in. This demonstrates that the [fwd] section is used by Dial() when I try to place a call out through that service, and that the register line isn't needed for the outbound call. Somebody mentioned that the mediatrix lets you set a unique username/password for each of its ports. It seems that you could set up four service definitions, each using a different user/pwd pair. Then * will use a different user/pwd pair to log in to the mediatrix, depending upon which service definition was called for by the Dial() statement. Or does the mediatrix not really have a distinct user/pwd pair for accessing each port? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
Mike,
I'm hoping one can specify a particular mediatrix port in the Dial Sip
command, but haven't found any Dial syntax that would allow passing a
userid/password to the gateway. Since the 1204 provides a AuthUsrPwd on
a per port basis, my guess would be that we either have to pass the Alias
defined for that port or the AuthUsrPwd in the Dial command.
When I attempt
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
I get an immediate 407 Proxy Authentication Required back. However, with
a packet sniffer running, * isn't even sending a packet to the mediatrix.
I'd have to guess and assume * is doing this because the mediatrix isn't
'registered' with *, but the mediatrix was not designed to register anyway.
I'm stuck in the Dial syntax, and can't seem to find any google reference
as to how to pass the needed parameters.
Rich
Bob, I have a question into mediatrix for this, but maybe you have
figured it out. I am trying to map a SIP user to a specific PSTN line. I
have my extensions.conf file as you show below, but on the 1204, it just
grabs whatever line is available, whereas I want extension 101 to always
be port1 on 1204, and extension 102 to be port 2 and so on. I noticed a
NetToPstnSourceFilter MIB per port, and their docs hint at using this,
but the example in the docs describes their FXS to FXO, so I am not sure
what I would put in that MIB. CallerID info? * calling sip extension
number? Have you been able to make this work?
On Sat, 2004-01-31 at 20:22, Bob Knight wrote:
Rich Adamson wrote:
I'm having a brain fart
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
;**
[trunk-local]
;**
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9NXX,2,Congestion
[trunk-toll]
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Congestion
--
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---End of Original Message-
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Re: [Asterisk-Users] Dial via sip gateway?
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
;**
[trunk-local]
;**
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
The above does not seem to work either. Since the mediatrix has four pstn
ports, there must be a way to construct a Dial command that would embed
a userid:password, port alias name, or something like that. Just can't find
any reference to what that syntax would look like. (The gateway is properly
handling incoming pstn calls, just not the outgoing pstn attempts.)
Really need to the sip dial command to include...
- the string of digits to be called
- either a userid:password, or, port alias name (or both)
- ip address of the gateway
Anybody have a clue what that dial sip syntax would look like
Rich
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Re: [Asterisk-Users] Dial via sip gateway?
Rich Adamson wrote:
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
;**
[trunk-local]
;**
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
The above does not seem to work either. Since the mediatrix has four pstn
ports, there must be a way to construct a Dial command that would embed
a userid:password, port alias name, or something like that. Just can't find
any reference to what that syntax would look like. (The gateway is properly
handling incoming pstn calls, just not the outgoing pstn attempts.)
Really need to the sip dial command to include...
- the string of digits to be called
- either a userid:password, or, port alias name (or both)
- ip address of the gateway
Anybody have a clue what that dial sip syntax would look like
Yes, it's
SIP/[EMAIL PROTECTED]
There's no 'sub-extension'.
So SIP/[EMAIL PROTECTED] is the proper way to go, where extension is
the string of digits to be called. If the box acts as a SIP proxy, you
might need to register with a register= in sip.conf beforehand.
This is like calling any FWD extension. First, register, then place
a call with
DIAL(SIP/[EMAIL PROTECTED])
Any pointer to the manual?
/O
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Re: [Asterisk-Users] Dial via sip gateway?
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
;**
[trunk-local]
;**
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
The above does not seem to work either. Since the mediatrix has four pstn
ports, there must be a way to construct a Dial command that would embed
a userid:password, port alias name, or something like that. Just can't find
any reference to what that syntax would look like. (The gateway is properly
handling incoming pstn calls, just not the outgoing pstn attempts.)
Really need to the sip dial command to include...
- the string of digits to be called
- either a userid:password, or, port alias name (or both)
- ip address of the gateway
Anybody have a clue what that dial sip syntax would look like
Yes, it's
SIP/[EMAIL PROTECTED]
There's no 'sub-extension'.
So SIP/[EMAIL PROTECTED] is the proper way to go, where extension is
the string of digits to be called. If the box acts as a SIP proxy, you
might need to register with a register= in sip.conf beforehand.
This is like calling any FWD extension. First, register, then place
a call with
DIAL(SIP/[EMAIL PROTECTED])
Any pointer to the manual?
No, the manual is very verbose but no * examples at all. The box sells as
either a 323 or sip, with different images (sort of like C7960's) and
different manuals.
The box does not support the register function in either direction. I just
tried the * sip register, and got a 501 Not Implemented with sniffer.
From what I can tell (box is about 48 hrs old for me), it seems to be a
rather incomplete or just-bare-sip-minimum functionality. It also appears
as though all four ports are treated as a group-of-lines, and one doesn't
have any choice (from a sip perspective) on which port to use for outgoing
calls. Since this one is set up with 1:home, 2:business, 3:outgoing calls
I really need to be able to select which port * is going to use, particularly
since outgoing 'home' long distance calls must use a different port then for
outgoing 'business' calls.
The entire box (4 ports) has only a single IP, so if the dial sip command
doesn't have any additional parameters/strings to destinguish selected ports,
guess I'll return it to the reseller. There appears to be a way to set certain
types of filters on a per port basis in the box, but I can't see how that
could be used to differentiate home vs business calls, etc.
Since I don't know anything about 323, does that control protocol allow some
sort of sub-selection where each port would be addressable? If not, it certainly
seems as though Mediatrix needs to go back to work on their code or something.
Can you think of any other way that * might interact with this thing via sip?
Rich
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Re: [Asterisk-Users] Dial via sip gateway?
On Sun, 1 Feb 2004, Rich Adamson wrote: The above does not seem to work either. Since the mediatrix has four pstn ports, there must be a way to construct a Dial command that would embed a userid:password, port alias name, or something like that. Just can't find any reference to what that syntax would look like. (The gateway is properly handling incoming pstn calls, just not the outgoing pstn attempts.) Really need to the sip dial command to include... - the string of digits to be called - either a userid:password, or, port alias name (or both) - ip address of the gateway Anybody have a clue what that dial sip syntax would look like I have only recently begun actually playing with *, but I'll venture a guess.. You (or somebody else) mentioned that you can force a call to go out a particular port on the Mediatrix by using the username/password pair which corresponds to that port, and this guess is based on that assumption. (I hope it's a valid assumption!) At http://www.voip-info.org/wiki-Asterisk+SIP+channels, under Using a SIP channel in extensions.conf, we read that the dial string format is either SIP/exten@peer or SIP/peer/exten. peer may be a hostname of a SIP proxy server, a domain where * should look for a SRV record, or a service defined in sip.conf. So try something like this in extensions.conf: exten = 101,1,Dial(SIP/number@mediatrixport1) exten = 102,1,Dial(SIP/number@mediatrixport2) exten = 103,1,Dial(SIP/number@mediatrixport3) exten = 104,1,Dial(SIP/number@mediatrixport4) and then define those services in sip.conf: [mediatrixport1] username=username for access to port1 password= host=mediatrix IP/name [mediatrixport2] username=username for access to port2 password= host=same mediatrix IP/name and so on for ports 3 and 4. I think a setup like this will allow you to use distinct username/password pairs for connections to the same SIP proxy. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
On Sun, 1 Feb 2004, Greg Hill wrote:
snip
So try something like this in extensions.conf:
exten = 101,1,Dial(SIP/number@mediatrixport1)
exten = 102,1,Dial(SIP/number@mediatrixport2)
exten = 103,1,Dial(SIP/number@mediatrixport3)
exten = 104,1,Dial(SIP/number@mediatrixport4)
Oops, maybe I should have written these extensions to be more like this:
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _8NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _7NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _6NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
so that you can choose which port you'll dial out on by prefixing your
number with 9/8/7/6.
Greg
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Re: [Asterisk-Users] Dial via sip gateway?
Greg,
So try something like this in extensions.conf:
exten = 101,1,Dial(SIP/number@mediatrixport1)
exten = 102,1,Dial(SIP/number@mediatrixport2)
exten = 103,1,Dial(SIP/number@mediatrixport3)
exten = 104,1,Dial(SIP/number@mediatrixport4)
Oops, maybe I should have written these extensions to be more like this:
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _8NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _7NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _6NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
so that you can choose which port you'll dial out on by prefixing your
number with 9/8/7/6.
I don't believe the above will work. There is only one IP address for the box,
and no way that I've found to send a sip packet to the box with additional
information that would suggest using port 1 vs port 2. From what others have
hinted at (and it seems the majority of us are limited either to what's printed
or experimentation), the 1204 has an internal function that kind of resembles
a trunk group. It decides which port to use.
As mentioned previously, the sip register function in the box is inop in
both directions, therefore there does not seem to be a way to address the
ports through contexts or anything else. Mediatrix has provided the mib
variables where one can enter a different password for each port, but that
has no value either since the register function doesn't work.
You've sort of touched on a method that might work, by prefixing called numbers
with a digit, then strip it in the 1204, etc. However, when you think that
process through for anything other than the simpliest of cases, it creates
a fairly major dialplan management issue. For the price of the box, think
I'll delay the purchase for now.
Rich
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Re: [Asterisk-Users] Dial via sip gateway?
On Sun, 1 Feb 2004, Rich Adamson wrote: I don't believe the above will work. There is only one IP address for the box, and no way that I've found to send a sip packet to the box with additional information that would suggest using port 1 vs port 2. From what others have hinted at (and it seems the majority of us are limited either to what's printed or experimentation), the 1204 has an internal function that kind of resembles a trunk group. It decides which port to use. As mentioned previously, the sip register function in the box is inop in both directions, therefore there does not seem to be a way to address the ports through contexts or anything else. Mediatrix has provided the mib variables where one can enter a different password for each port, but that has no value either since the register function doesn't work. What happens if you don't use a register = line in sip.conf, but do include a section like: [mediatrixport1] username= password= host= Just to check my theory, I did some testing via fwd. I discovered that if I include a register = line with my fwd info, then when I call my fwd number (outbound through iaxtel) it rings in. But I can't call out via fwd. So then I put in my [fwd] service definition, removed the register line, and waited for the old registration to expire. Then I tried calling my fwd number (again through iaxtel). This time I got the message about the user being offline. But now I can call out via fwd, even though calls wouldn't come in. This demonstrates that the [fwd] section is used by Dial() when I try to place a call out through that service, and that the register line isn't needed for the outbound call. Somebody mentioned that the mediatrix lets you set a unique username/password for each of its ports. It seems that you could set up four service definitions, each using a different user/pwd pair. Then * will use a different user/pwd pair to log in to the mediatrix, depending upon which service definition was called for by the Dial() statement. Or does the mediatrix not really have a distinct user/pwd pair for accessing each port? Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
On Sun, 1 Feb 2004 08:21:55 -0600, Rich Adamson wrote [long snip] No, the manual is very verbose but no * examples at all. The box sells as either a 323 or sip, with different images (sort of like C7960's) and different manuals. The box does not support the register function in either direction. I just tried the * sip register, and got a 501 Not Implemented with sniffer. From what I can tell (box is about 48 hrs old for me), it seems to be a rather incomplete or just-bare-sip-minimum functionality. It also appears as though all four ports are treated as a group-of-lines, and one doesn't have any choice (from a sip perspective) on which port to use for outgoing calls. Since this one is set up with 1:home, 2:business, 3:outgoing calls I really need to be able to select which port * is going to use, particularly since outgoing 'home' long distance calls must use a different port then for outgoing 'business' calls. I have an idea of a crude hack that just might work - e.g. if you need to dial a number on line 3, first make two outgoing calls to a bogus number (just to keep the lines busy for a second) and then place the 3rd call to the destination you want - if I understand the situation correctly, the 1204 should dial on the 3rd line then and the first two calls should drop quickly (no such number). Of course, in that case you need to keep the line state e.g. in the DB so that, say, line 1 in use doesn't mess things up. Yes, I know it's ugly. If it's also bound not to work, I'm all ears as to *why* :) The entire box (4 ports) has only a single IP, so if the dial sip command doesn't have any additional parameters/strings to destinguish selected ports, guess I'll return it to the reseller. There appears to be a way to set certain types of filters on a per port basis in the box, but I can't see how that could be used to differentiate home vs business calls, etc. Since I don't know anything about 323, does that control protocol allow some sort of sub-selection where each port would be addressable? If not, it certainly seems as though Mediatrix needs to go back to work on their code or something. Can you think of any other way that * might interact with this thing via sip? Rich Regards, Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
I don't believe the above will work. There is only one IP address for the box, and no way that I've found to send a sip packet to the box with additional information that would suggest using port 1 vs port 2. From what others have hinted at (and it seems the majority of us are limited either to what's printed or experimentation), the 1204 has an internal function that kind of resembles a trunk group. It decides which port to use. As mentioned previously, the sip register function in the box is inop in both directions, therefore there does not seem to be a way to address the ports through contexts or anything else. Mediatrix has provided the mib variables where one can enter a different password for each port, but that has no value either since the register function doesn't work. What happens if you don't use a register = line in sip.conf, but do include a section like: [mediatrixport1] username= password= host= The above is basically what I did, however since the 1204 never attempts to register, the username and password have no value. The host= is the only statement above that has value, and its the only thing that can be used to associate a context with the gateway. Attempts to use a register statement within * (and watching packets with a sniffer), the register attempt is greated with 501 Not Implemented from the 1204. Just to check my theory, I did some testing via fwd. I discovered that if I include a register = line with my fwd info, then when I call my fwd number (outbound through iaxtel) it rings in. But I can't call out via fwd. So then I put in my [fwd] service definition, removed the register line, and waited for the old registration to expire. Then I tried calling my fwd number (again through iaxtel). This time I got the message about the user being offline. But now I can call out via fwd, even though calls wouldn't come in. This demonstrates that the [fwd] section is used by Dial() when I try to place a call out through that service, and that the register line isn't needed for the outbound call. Sure, but fwd and your asterisk both understand the register function. The 1204 does not. Somebody mentioned that the mediatrix lets you set a unique username/password for each of its ports. That was me that said it in an earlier email attempting to find out if it was me or the 1204 that didn't understand what was going on. Turned out to be the 1204. It seems that you could set up four service definitions, each using a different user/pwd pair. Then * will use a different user/pwd pair to log in to the mediatrix, depending upon which service definition was called for by the Dial() statement. which, again, all depends on the register function working. Or does the mediatrix not really have a distinct user/pwd pair for accessing each port? It has the mib variables and one can set them, the 1204 just doesn't do anything with them. The bottom line really is 501 Not Implemented, period. Until that's implemented there really isn't anyway to address individual ports in any form that is reasonable. For what its worth, it would appear from the Mediatrix web site (takes a little digging) the group behind writing the sip code must have had some financial problems. They received some funding in November along with apparently some senior management changes. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial via sip gateway?
From what I can tell (box is about 48 hrs old for me), it seems to be a rather incomplete or just-bare-sip-minimum functionality. It also appears as though all four ports are treated as a group-of-lines, and one doesn't have any choice (from a sip perspective) on which port to use for outgoing calls. Since this one is set up with 1:home, 2:business, 3:outgoing calls I really need to be able to select which port * is going to use, particularly since outgoing 'home' long distance calls must use a different port then for outgoing 'business' calls. I have an idea of a crude hack that just might work - e.g. if you need to dial a number on line 3, first make two outgoing calls to a bogus number (just to keep the lines busy for a second) and then place the 3rd call to the destination you want - if I understand the situation correctly, the 1204 should dial on the 3rd line then and the first two calls should drop quickly (no such number). Of course, in that case you need to keep the line state e.g. in the DB so that, say, line 1 in use doesn't mess things up. Yes, I know it's ugly. If it's also bound not to work, I'm all ears as to *why* :) Yup, that's a very ugly one. Given this is an eval box with an option to buy, I'd rather send it back. Other then the register function, the box appears to be a very nice one. Maybe a little pricey, but would bet it fits into a very large number of businesses/homes very nicely. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dial via sip gateway?
I'm having a brain fart
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
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Re: [Asterisk-Users] Dial via sip gateway?
On Sat, 31 Jan 2004, Rich Adamson wrote:
I'm having a brain fart
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
should you say ${EXTEN:1} rather than ${EXTEN-1} to drop that 6 off the
front of the extension?
Greg
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Re: [Asterisk-Users] Dial via sip gateway?
Rich Adamson wrote:
I'm having a brain fart
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
;**
[trunk-local]
;**
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9NXX,2,Congestion
[trunk-toll]
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Congestion
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
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Re: [Asterisk-Users] Dial via sip gateway?
Bob, I have a question into mediatrix for this, but maybe you have
figured it out. I am trying to map a SIP user to a specific PSTN line. I
have my extensions.conf file as you show below, but on the 1204, it just
grabs whatever line is available, whereas I want extension 101 to always
be port1 on 1204, and extension 102 to be port 2 and so on. I noticed a
NetToPstnSourceFilter MIB per port, and their docs hint at using this,
but the example in the docs describes their FXS to FXO, so I am not sure
what I would put in that MIB. CallerID info? * calling sip extension
number? Have you been able to make this work?
On Sat, 2004-01-31 at 20:22, Bob Knight wrote:
Rich Adamson wrote:
I'm having a brain fart
What's the proper syntax for dialing out via a sip g/w (Mediatrix)?
Been trying stuff similar to:
exten = _6X.,1,Dial(SIP/[EMAIL PROTECTED]/${EXTEN-1})
where 3091 is alias for the port on the Mediatrix. Sniffer indicates * did
even try the IP.
Rich
from my extensions.conf:
;**
[trunk-local]
;**
exten = _9NXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9NXX,2,Congestion
[trunk-toll]
exten = _91NXXNXX,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _91NXXNXX,2,Congestion
--
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