RE: [Asterisk-Users] Bill seconds
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of C F > Sent: Sunday, 19 June 2005 2:19 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Bill seconds > > > > next month and tell hoe much cell phone costs you. > > > > The point I was making is that the charges are NOT on _My_ > cell phone > > bill, > > Why is it that if you pay 10 times as much to call a cell > phone you consider it NOT part of your cell phone bill? Who says I do? Where did you pull that "10 times" stuff? I don't have to pay anything more to call a Cell phone that I do to call a land line. In fact for the 5 mobiles that I own, (my family members) the calls between them and my land lines are free. Again, as the originator of the call I get to choose the amount I spend. > Don't > you see how they succeeded in making you believe that your > cell phone is cheaper? I told you that none Amercians might > not understand this. :) Yeah, I see how _some_ americans don't get it. > > > when I don't originate the call, however in .us if you get > called, you > > pay, that can easily cost you a heap of money that you can only > > control by switching the phone off, and where is the point in that? > > Really?? cost you a heap of money? only by swithcing the phone off? > what ever happened to not picking up? Ok, there is that, so long as you take time to determine whether you recognise the number etc It does however make rec'ving calls on the Cell phone much less attractive. > what about unlimited > nights and weekends completely free that most providers give > you here. What about the fact that even when you do pay for > the incoming it costs around > $.05 a minute? How about just not having to pay for incoming calls at all, that sounds much better. It makes being in touch easier and cheaper. > I think I said enough. how does one respond to that? > > > > > So if I rec'v 500 calls a week on my cell phone, it still > costs me nothing. > > Wrong, because your provider succeeded in convincing your > freind to make the same calculation, so when you have to call > your friend you then pay 10 times as much than to a regular phone. Pure and unadulterated crapola, did you know that when people pluck numbers out of the air like that it belittles their entire point? > > > And in some cases if I have the Cell and the Landline from the same > > telco (in .au), calls between them are free too, regardless > of where I > > happen to be in australia at the time. > > So this we will take out of the argument since most American > providers don't charge in network either. They do for out of zone calls, however with the telco I am using and the account arrangements I have, it doesn't matter where the cell phone is, even 4000km away is still a free call to my home land line. > > > > > Oh, and cucumber seem to be doing you no favours either > > > > I can place a call to the US using my Cell phone for 1-2c/minute, > > Caviat Emptor? > > Actualy you are right about this one, didn't realize they > changed the rates to au, it used to be $.039 a minute. Thanks > for pointing this out. In any case I know that Australia has > now very good rates to call UK and the states, but that is > only as far as LD goes. I have VoIP for calls to the .us and .uk I also can route my call via my home * box and then over VoIP to many other places to make the calls ** so with a call to .us for instance, I can use my cell to call one of my home land lines ** and then via * connect to the us using one of the IP Telcos *<1c/min>* , or to my office in Houston to the * box there ** Further: In the .US there is a groundswell of people that are angry with telemarketers calling them on their cell phones, Why is this? A: because the cost of the call is shifted to the called party, just like spam. The .au model of "caller pays" has pretty much ensured that telemarketers wont be a problem on _my_ cell phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Bill seconds
> > next month and tell hoe much cell phone costs you. > > The point I was making is that the charges are NOT on _My_ cell phone bill, Why is it that if you pay 10 times as much to call a cell phone you consider it NOT part of your cell phone bill? Don't you see how they succeeded in making you believe that your cell phone is cheaper? I told you that none Amercians might not understand this. :) > when I don't originate the call, however in .us if you get called, you pay, > that can easily cost you a heap of money that you can only control by > switching the phone off, and where is the point in that? Really?? cost you a heap of money? only by swithcing the phone off? what ever happened to not picking up? what about unlimited nights and weekends completely free that most providers give you here. What about the fact that even when you do pay for the incoming it costs around $.05 a minute? I think I said enough. > > So if I rec'v 500 calls a week on my cell phone, it still costs me nothing. Wrong, because your provider succeeded in convincing your freind to make the same calculation, so when you have to call your friend you then pay 10 times as much than to a regular phone. > And in some cases if I have the Cell and the Landline from the same telco > (in .au), calls between them are free too, regardless of where I happen to > be in australia at the time. So this we will take out of the argument since most American providers don't charge in network either. > > Oh, and cucumber seem to be doing you no favours either > > I can place a call to the US using my Cell phone for 1-2c/minute, > Caviat Emptor? Actualy you are right about this one, didn't realize they changed the rates to au, it used to be $.039 a minute. Thanks for pointing this out. In any case I know that Australia has now very good rates to call UK and the states, but that is only as far as LD goes. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] channel.c:1884 set_format: Unable to find a path from g729 to gsm
Did you purchase/install the g729 codec?
Kumara Jayaweera wrote:
Hi All,
I have this codec problem, I use "gsm" in my iax.conf file and in teliax
settings also, but the error is still appearing as below. please help me.
Kumara
Starting simple switch on 'Zap/1-1'
-- Executing Dial("Zap/1-1","IAX2/[EMAIL PROTECTED]/01194777070239|30|tr") in
new stack
-- Called [EMAIL PROTECTED]/01194777070239
-- Call accepted by 208.139.204.228 (format g729)
-- Format for call is g729
Jun 18 19:17:28 NOTICE[8554]: channel.c:1884 set_format: Unable to find a
path from g729 to gsm
Jun 18 19:17:28 NOTICE[8554]: channel.c:1884 set_format: Unable to find a
path from g729 to gsm
Jun 18 19:17:28 NOTICE[8554]: channel.c:1884 set_format: Unable to find a
path from g729 to gsm
-- IAX2/teliax-2 is ringing
-- Nobody picked up in 3 ms
-- Hungup 'IAX2/teliax-2'
== Auto fallthrough, channel 'Zap/1-1' status is
'NOANSWER'
-- Hungup 'Zap/1-1'
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[Asterisk-Users] channel.c:1884 set_format: Unable to find a path from g729 to gsm
Hi All,
I have this codec problem, I use "gsm" in my iax.conf file and in teliax
settings also, but the error is still appearing as below. please help me.
Kumara
Starting simple switch on 'Zap/1-1'
-- Executing Dial("Zap/1-1","IAX2/[EMAIL PROTECTED]/01194777070239|30|tr") in
new stack
-- Called [EMAIL PROTECTED]/01194777070239
-- Call accepted by 208.139.204.228 (format g729)
-- Format for call is g729
Jun 18 19:17:28 NOTICE[8554]: channel.c:1884 set_format: Unable to find a
path from g729 to gsm
Jun 18 19:17:28 NOTICE[8554]: channel.c:1884 set_format: Unable to find a
path from g729 to gsm
Jun 18 19:17:28 NOTICE[8554]: channel.c:1884 set_format: Unable to find a
path from g729 to gsm
-- IAX2/teliax-2 is ringing
-- Nobody picked up in 3 ms
-- Hungup 'IAX2/teliax-2'
== Auto fallthrough, channel 'Zap/1-1' status is
'NOANSWER'
-- Hungup 'Zap/1-1'
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[Asterisk-Users] How to setup two Asterisk boxes - keeping the registration
I have now two asterisk boxes running, one on the IP *18 and one on IP *20 Both are working, use the same dialplan and realtime. I did not find out how to set it up that if a phone is registered in *18 can reach a phone registered in *20. For the wake up call I also see some troubles, since both machines point to the same NFS space. It could be that both machines start the wakeup call. Has anybody solved that? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] FATAL: Error running install command for wctdm
Jason Becker wrote: Ronald Wiplinger wrote: FATAL: Error inserting wctdm (/lib/modules/2.6.8-24.11-default/extra/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm Freed a Wildcard module wctdm unsupported by SUSE/Novell, tainting kernel. wctdm: disagrees about version of symbol zt_receive Check out: http://lists.digium.com/pipermail/asterisk-users/2005-March/096532.html Regards, It seems the new CVS acts different. Before all modules have been come into /lib/modules/kernelversion/misc, while old modules from Suse have been in the /extra/ New compiling of asterisk put it into extra I cleaned the misc directory with the modules from April and made a depmod I get now another error: vpbx:/etc/asterisk # asterisk-restart Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) ERROR: Module wctdm does not exist in /proc/modules However, Astrisk started and with show version I see now: show version Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a x86_64 running Linux on 2005-06-18 14:53:44 Why this error come up? How can I avoid it? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Voip-info.org mirror/translation
http://sites.gizoogle.com/?url=http://www.voip-info.org -Matt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Three way calling with Cisco 12SP+
How do I setup three way calling with Asterisk and a Cisco 12SP+ telephone? I would like to be able to three way Voipjet numbers as well as IAX calls. Thanks BlakeOPS ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ASTCC Rate Calculation
Rusty Shackleford wrote:
On Sat, June 18, 2005 2:34 pm, Darren Wiebe said:
Okay, I'll post both pieces of code. What I was seeing is that calls
where being billed more than I thought they should be. Lets use an
example with the following info:
Call Length: 147 Seconds
Increments: 6 Seconds
Connect Charge: 100
Included Seconds: 30
Cost per minute: 100
1. Present Code:
eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment)
* $increment };
#adjtime = 152
This might be where your error is creeping in. $adjtime SHOULD equal 150.
Remember, the int() function removes the value to the right of the decimal
point - so int(($answerdtime + $increment -1) / $increment) = 25 and not
25.3~, as your example appears to show. This makes $adjtime
actually 150, not 152.
You are right, I missed the one bracket.
eval { $cost = int($adjcost * $adjtime / 60) };
#cost = 253
Corrected, this would be 250.
Viewed another way, using a 6 second increment, 147 seconds represents 25
such increments (actually 24.5, but we get all of the last increment, so
it's 25).
25 * 10 (the cost of one 6-second increment) = 250.
Yes, but we need to allow for 30,6 6,1 60,30 billing. I think the
easiest/best way to handle this is the connect charges as ASTCC
presently supports them.
$cost += $adjconn;
#Total Cost = 353
2. My Proposed Code:
$total_seconds = ($answeredtime - $numdata->{includedseconds})/$increment;
#Total_Seconds(This variable is not very well named) = 19.5
$bill_increments = ceil($total_seconds);
#We need to bill for 20 6 second increments.
$billseconds = $bill_increments * $increment;
#This translates to 120 seconds.
Which cheats us out of 27 seconds worth of revenue (actually 30 seconds,
since that 27 seconds represents five 6-second increments).
I don't think it does. The 30 seconds come out because that is the
amount included in the connect charge. The connect charge is added back
in here:
$cost = ($billseconds / 60) * $adjcost + $adjconn;
Darren Wiebe
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Re: [Asterisk-Users] ASTCC Rate Calculation
On Sat, June 18, 2005 2:34 pm, Darren Wiebe said:
> Okay, I'll post both pieces of code. What I was seeing is that calls
> where being billed more than I thought they should be. Lets use an
> example with the following info:
>
> Call Length: 147 Seconds
> Increments: 6 Seconds
> Connect Charge: 100
> Included Seconds: 30
> Cost per minute: 100
>
>
> 1. Present Code:
> eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment)
> * $increment };
> #adjtime = 152
This might be where your error is creeping in. $adjtime SHOULD equal 150.
Remember, the int() function removes the value to the right of the decimal
point - so int(($answerdtime + $increment -1) / $increment) = 25 and not
25.3~, as your example appears to show. This makes $adjtime
actually 150, not 152.
> eval { $cost = int($adjcost * $adjtime / 60) };
> #cost = 253
Corrected, this would be 250.
Viewed another way, using a 6 second increment, 147 seconds represents 25
such increments (actually 24.5, but we get all of the last increment, so
it's 25).
25 * 10 (the cost of one 6-second increment) = 250.
> $cost += $adjconn;
> #Total Cost = 353
>
> 2. My Proposed Code:
> $total_seconds = ($answeredtime - $numdata->{includedseconds})/$increment;
> #Total_Seconds(This variable is not very well named) = 19.5
> $bill_increments = ceil($total_seconds);
> #We need to bill for 20 6 second increments.
> $billseconds = $bill_increments * $increment;
> #This translates to 120 seconds.
Which cheats us out of 27 seconds worth of revenue (actually 30 seconds,
since that 27 seconds represents five 6-second increments).
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RE: [Asterisk-Users] [partialy - solved] IAX with shaw cable not going through
You are right, I'll take it back, it is not Motorola SB5100 related problem. When I talked to Shaw again I was told that they are cashing the MAC address for about 4-hours (some kind of security reason) but they wouldn't explain why. Apparently that cashing is only implemented with their Extreme Speed connection, not with the standard modem. So powering down the Cable Modem will not solve the problem. -- #Joseph > Actually, the SB5100's are one of the best cable modems on the market. > The question here is, how does Shaw configure their network? When you > originally signed up with them, did you have to give them the MAC > address off your network card? Or just the MAC off the modem? If it is > the second, then power down the modem for a minute or two, then with the > new firewall in place, power the SB5100 back up, then power up the new > firewall. No, you will probably not retain the same IP address. That is > just life in the cable HIS industry. > > It is not the SB5100 causing the issue. If anything, it is Shaw and > their DHCP policies. I have a SB5100, and by power cyclcing, I can > change firewalls all day long with no issues. > > Robert > > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] [partialy - solved] IAX with shaw cable not going through
> -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Joseph > Sent: Saturday, June 18, 2005 6:09 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] [partialy - solved] IAX with > shaw cable not going through > > > Long Story: Shaw has those new Cable Modem - Motorola > SURFboard SB5100 > > that once configured to an IP address with one firewall it > will retain > > that MAC address of that first firewall for about 4-hours. When I > > first experimented that Cable Modem I've connected my > backup firewall > > and the Modem retained that MAC address. > > So in order to connect the second firewall and get the same IP > > address, I need to spoof the MAC address of the first > firewall or wait 4-hours. > > So I went with the second solution but I don't see how that > could make > > a difference, the only way to tell is to wait 4-hours to remove the > > spoof MAC address from the firewall. > > It seems to me spoofing MAC address is causing the problem. > I've connected the original firewall that I tested (without > spoof MAC address assigned to firewall) and every connection > is working FWD, VoipJet. > > It seems it me that new Shaw Cable - Motorola SURFboard > SB5100 is a piece or crap. > Actually, the SB5100's are one of the best cable modems on the market. The question here is, how does Shaw configure their network? When you originally signed up with them, did you have to give them the MAC address off your network card? Or just the MAC off the modem? If it is the second, then power down the modem for a minute or two, then with the new firewall in place, power the SB5100 back up, then power up the new firewall. No, you will probably not retain the same IP address. That is just life in the cable HIS industry. It is not the SB5100 causing the issue. If anything, it is Shaw and their DHCP policies. I have a SB5100, and by power cyclcing, I can change firewalls all day long with no issues. Robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] IAX with shaw cable not going through
Rogers does the same thing all you need to do is a DHCP release (or the equivalent in your FW). I had similar issues (not asterisk related) since I have a pix fw and it has no option to do a dhcp release. John -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Saturday, June 18, 2005 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX with shaw cable not going through On Sat, 2005-06-18 at 14:16 -0600, Joseph wrote: > I just changed the from DSL to Shaw Cable (static IP) configure the > firewall but now asterisk I can not register with FWD nor VoipJet calls > going out. > > I am using IAX with FWD > Did I missed to change a setting? I don't think there is any though. > > I am on shaw extreme connection; I just talked shaw tech. and they are > not blocking any port - I was told. > So why IAX will not register with FWD and calls to VoipJet are not > getting connected. I've boot my asterisk backup server to ADSL and everything is working FWD, VoipJet. Short story: The only thing I've done differently is I've spoofed MAC address on the firewall on an external port - eth0 to get the same IP address from Shaw, but I don't see how that could make a difference. Long Story: Shaw has those new Cable Modem - Motorola SURFboard SB5100 that once configured to an IP address with one firewall it will retain that MAC address of that first firewall for about 4-hours. When I first experimented that Cable Modem I've connected my backup firewall and the Modem retained that MAC address. So in order to connect the second firewall and get the same IP address, I need to spoof the MAC address of the first firewall or wait 4-hours. So I went with the second solution but I don't see how that could make a difference, the only way to tell is to wait 4-hours to remove the spoof MAC address from the firewall. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] [partialy - solved] IAX with shaw cable not going through
> Long Story: Shaw has those new Cable Modem - Motorola SURFboard SB5100 > that once configured to an IP address with one firewall it will retain > that MAC address of that first firewall for about 4-hours. When I first > experimented that Cable Modem I've connected my backup firewall and the > Modem retained that MAC address. > So in order to connect the second firewall and get the same IP address, > I need to spoof the MAC address of the first firewall or wait 4-hours. > So I went with the second solution but I don't see how that could make a > difference, the only way to tell is to wait 4-hours to remove the spoof > MAC address from the firewall. It seems to me spoofing MAC address is causing the problem. I've connected the original firewall that I tested (without spoof MAC address assigned to firewall) and every connection is working FWD, VoipJet. It seems it me that new Shaw Cable - Motorola SURFboard SB5100 is a piece or crap. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ASTCC Rate Calculation
Okay, I'll post both pieces of code. What I was seeing is that calls
where being billed more than I thought they should be. Lets use an
example with the following info:
Call Length: 147 Seconds
Increments: 6 Seconds
Connect Charge: 100
Included Seconds: 30
Cost per minute: 100
1. Present Code:
eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment)
* $increment };
#adjtime = 152
eval { $cost = int($adjcost * $adjtime / 60) };
#cost = 253
$cost += $adjconn;
#Total Cost = 353
2. My Proposed Code:
$total_seconds = ($answeredtime - $numdata->{includedseconds})/$increment;
#Total_Seconds(This variable is not very well named) = 19.5
$bill_increments = ceil($total_seconds);
#We need to bill for 20 6 second increments.
$billseconds = $bill_increments * $increment;
#This translates to 120 seconds.
$cost = ($billseconds / 60) * $adjcost + $adjconn;
Therefore the cost = 300
3. Proposed Correction to original Code
The difference I see is that the first one is double billing for the
included seconds. That would be easier fixed as follows:
eval { my $adjtime = int((($answeredtime - $numdata->{includedseconds})
+ $increment - 1) / $increment) * $increment };
Doing the math this way the cost on the call would come out @ 303
Which example is correct? Which code is easier to follow? :-)
Darren Wiebe
[EMAIL PROTECTED]
Rusty Shackleford wrote:
On Fri, June 17, 2005 5:19 pm, Darren Wiebe said:
Good Day
Has anybody here looked closely at the call cost calculation in ASTCC?
Can you duplicate the way the cost of a call is calculated? I believe
that there is an error in the code. I have fixed it, I think and
submitted a patch but we need user comments. I would appreciate if
anybody involved would slip over to chech out this link on the
bugtracker and provide feedback. http://bugs.digium.com/view.php?id=4480
I may well be wrong but I believe the issue needs visiting. Somebody
was asking me how it calculates costs as they thought they knew what a
call should cost. I said "I'll show you". Mistake, I could not come up
with an answer that made sense.
Darren,
I took a quick look at the patch. I'm not certain, but it appears that
you've taken out the formula that factors in the billing increment. This
forumla, inything other than a 1 second incement, will always "add" time
to the call for any number of seconds not equally divisible by the billing
increment integer, resulting in a slightly higher cost than might be
expected at first glance. This is the way it is supposed to work.
As I said, I only glanced at it briefly. Could you describe your changes
and the error you were seeing?
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Re: [Asterisk-Users] IAX with shaw cable not going through
On Sat, 2005-06-18 at 14:16 -0600, Joseph wrote: > I just changed the from DSL to Shaw Cable (static IP) configure the > firewall but now asterisk I can not register with FWD nor VoipJet calls > going out. > > I am using IAX with FWD > Did I missed to change a setting? I don't think there is any though. > > I am on shaw extreme connection; I just talked shaw tech. and they are > not blocking any port - I was told. > So why IAX will not register with FWD and calls to VoipJet are not > getting connected. I've boot my asterisk backup server to ADSL and everything is working FWD, VoipJet. Short story: The only thing I've done differently is I've spoofed MAC address on the firewall on an external port - eth0 to get the same IP address from Shaw, but I don't see how that could make a difference. Long Story: Shaw has those new Cable Modem - Motorola SURFboard SB5100 that once configured to an IP address with one firewall it will retain that MAC address of that first firewall for about 4-hours. When I first experimented that Cable Modem I've connected my backup firewall and the Modem retained that MAC address. So in order to connect the second firewall and get the same IP address, I need to spoof the MAC address of the first firewall or wait 4-hours. So I went with the second solution but I don't see how that could make a difference, the only way to tell is to wait 4-hours to remove the spoof MAC address from the firewall. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] H323 implementations
I am about to add h323 to my system and although I have found information on the Wiki, comparing the asterisk implementation to oh323, I have not found anything about the new ooh323, which is included in the addons. Can anyone please compare this to the other two? Thanks, Richard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - fax - spandsp
Hi Steve We have been struggling for some time to get spandsp working properly on our asterisk system. We are using a TE410P card with a single E1 from our PTT (Zaptel primary clock source) and another E1 to a NEC PBX. The NEC PBX in turn has an E1 connection to a Tenovis I55 PBX, which then has Basic Rate connections to a Cycos MRS Fax server. We never experience problems with the Cycos Fax server (which uses an Eicon Diva card). We also do not experience problems on the 7 or 8 fax machines connected to our NEC and Tenovis PBXs (nor do we hear any ‘clicks’ on voice calls). Therefore we conclude that we do not have any ‘timing slips’ or suchlike with our setup. However, receiving faxes on the Asterisk with spandsp presents us with a problem – if we send a fax from a particular fax machine (old MITA machine), spandsp invariably does not receive the whole fax, cutting it at some unpredictable point. This particular fax machine is in daily use and works fine to any other fax machine. My knowledge of fax principles being quite limited, it seems to me that spandsp communicates perfectly with the transmitting fax machine up to a point, whereupon it fails to react to the transmitting machine’s instructions to advance to the next line scanned. This results in subsequent lines being superimposed upon each other, producing a dark horizontal line somewhere within the page (usually less than halfway). I’m inclined to believe that this must be a small problem which can be catered for in spandsp’s code, and make myself available to send you test faxes if you would like to verify this problem. Best regards Les Caroto Logitel Telecom Johannesburg South Africa -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.7/20 - Release Date: 2005/06/16 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Fwd: [Asterisk-Users] IAX with shaw cable not going through]
On Sat, 2005-06-18 at 13:17 -0700, Matthew Asham wrote: > For what it's worth I use a residential Shaw connection and have no > problems with IAX registration nor SIP to our Asterisk PBX elsewhere on > the 'net. When I had a dynapic IP connection with shaw, there was not problem either. I just changed to static and IAX can not connect. How can I test it? iax2 show registry Host UsernamePerceived Refresh State 65.39.205.121:4569491581 60 Timeout -- #Joseph > -Forwarded Message- > From: Joseph <[EMAIL PROTECTED]> > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] IAX with shaw cable not going through > Date: Sat, 18 Jun 2005 14:16:02 -0600 > > I just changed the from DSL to Shaw Cable (static IP) configure the > firewall but now asterisk I can not register with FWD nor VoipJet calls > going out. > > I am using IAX with FWD > Did I missed to change a setting? I don't think there is any though. > > I am on shaw extreme connection; I just talked shaw tech. and they are > not blocking any port - I was told. > So why IAX will not register with FWD and calls to VoipJet are not > getting connected. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX with shaw cable not going through
I just changed the from DSL to Shaw Cable (static IP) configure the firewall but now asterisk I can not register with FWD nor VoipJet calls going out. I am using IAX with FWD Did I missed to change a setting? I don't think there is any though. I am on shaw extreme connection; I just talked shaw tech. and they are not blocking any port - I was told. So why IAX will not register with FWD and calls to VoipJet are not getting connected. -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Want to test my * behind firewall
Can someone leave a message at x 200 on my * server. External IP two one six . nine . zero . three four Connect as x 202 password zxc123 using IAX2 thanks, -B __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to make outbound calls
Hi All, I am a new bee to *. I just installed [EMAIL PROTECTED] on FC3. I hv a FXO card. I hv configured two extensions one x-lite and other iaxComm. I configured * using AMP. The following setup works - x-lite (x 200) to iaxComm (x 201) - PSTN to x-lite - PSTN to iaxComm Voice mail, weather etc work fine. When i try to make an external call i am getting message "All routes are busy". In the asterisk console i am seeing "Everyone is busy/congested at this time". In AMP - Outbound dialing i hv configured a route which i call 'local'. The dial pattern is 1NXXNXX NXXNXX NXX and using trunk ZAP/g1. Any idea why i am unable to make outbound call. thanks for your help. -B __ Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 11, Issue 68
Hello All i have big problem for unicall. my system work successful with sangoma card, E1 and CAS signalling (vietnam). when at the some time. i have trouble then my system is half (CPU instructions = 100) i tested for some case as belows: - When i dial, then my system became answer, the caller hangup. system error message show (loop without condition and half machine) Jun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handlerJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handle rJun 11 12:15:45 WARNING[1496]: channel.c:1447 ast_read: Exception flag set on 'UniCall/6-1', butno exception handler the next case: when i choose voicemail, then received mail. for some times, caller hangup phone, system error show as above, then system half. I used unicall-0.0.3pre3 Please help me Thanks___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] 2nd Dialtone after answer
Thanks all for replying. Yes, DISA is what I needed. Thanks Oswaldo -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Henry Coleman Sent: Friday, June 17, 2005 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 2nd Dialtone after answer Hi all, DID is Direct Inward Dial where the last 3 or 4 digits of the dialed number are passed through and are used/translated to call a specific extension. (See Centrex) DISA is Direct System Access where incoming line(s)are auto-answered and receive internal dial tone, the caller then has access to the facilities of the system.(including calling an extension.) I hope this clears things up TTFN Henry Chris Coulthurst wrote: > Check out DISA. > > Chris Coulthurst > [EMAIL PROTECTED] > > > > |-Original Message- > |From: [EMAIL PROTECTED] > |[mailto:[EMAIL PROTECTED] On Behalf Of Oswaldo > |Arratia > |Sent: Friday, June 17, 2005 7:51 AM > |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > |Subject: [Asterisk-Users] 2nd Dialtone after answer > | > | > |Hi > |I am trying to achive this for a specific need of a customer. > | > |He has a DID pointed to an Asterisk server, I need to provide him > |dialtone when the calls hits the server. How can I achieve this? > | > |Let's say something like this: > | > |Exten => s,1,Answer > |Exten => s,2, "Provide Dial tone" > |Exten => s,3, "Dial the number the person will enter after receiving > |the dial tone" Exten => s,4,Hangup > | > |Any ideas? > | > |Thanks very much > | > |Oswaldo > | > | > |___ > |Asterisk-Users mailing list > |Asterisk-Users@lists.digium.com > |http://lists.digium.com/mailman/listinfo/asteri|sk-users > |To > |UNSUBSCRIBE or update options visit: > | http://lists.digium.com/mailman/listinfo/asterisk-users > | > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] FATAL: Error running install command for wctdm
Ronald Wiplinger wrote: FATAL: Error inserting wctdm (/lib/modules/2.6.8-24.11-default/extra/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm Freed a Wildcard module wctdm unsupported by SUSE/Novell, tainting kernel. wctdm: disagrees about version of symbol zt_receive Check out: http://lists.digium.com/pipermail/asterisk-users/2005-March/096532.html Regards, -- Jason Becker Director & CEO Coalescent Systems Inc. Enabling Open Source Telephony 403.244.8089 www.coalescentsystems.ca ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] ASTCC Rate Calculation
On Fri, June 17, 2005 5:19 pm, Darren Wiebe said: > Good Day > > Has anybody here looked closely at the call cost calculation in ASTCC? > Can you duplicate the way the cost of a call is calculated? I believe > that there is an error in the code. I have fixed it, I think and > submitted a patch but we need user comments. I would appreciate if > anybody involved would slip over to chech out this link on the > bugtracker and provide feedback. http://bugs.digium.com/view.php?id=4480 > I may well be wrong but I believe the issue needs visiting. Somebody > was asking me how it calculates costs as they thought they knew what a > call should cost. I said "I'll show you". Mistake, I could not come up > with an answer that made sense. > Darren, I took a quick look at the patch. I'm not certain, but it appears that you've taken out the formula that factors in the billing increment. This forumla, inything other than a 1 second incement, will always "add" time to the call for any number of seconds not equally divisible by the billing increment integer, resulting in a slightly higher cost than might be expected at first glance. This is the way it is supposed to work. As I said, I only glanced at it briefly. Could you describe your changes and the error you were seeing? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] FATAL: Error running install command for wctdm
Richard Lyman wrote: Ronald Wiplinger wrote: app_addon_sql_mysql.so app_intercom.so app_saycountpl.so cdr_addon_mysql.so format_mp3.so res_config_mysql.so WARNING WARNING WARNING I cannot remember that I have seen that before. you must have checkout'd asterisk-addons and compiled it at some point. so, you should update those also, and recompile them. I did I even recompiled all the thing again, My road map is: cd /usr/src/linux make cloneconfig make dep mkdir /usr/src/asterisk cd asterisk/ export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot cvs login Logging in to :pserver:[EMAIL PROTECTED]:2401/usr/cvsroot CVS password: anoncvs cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds iaxyprov astcc vi /usr/src/asterisk/asterisk/apps/app_voicemail.c edit this line and change 100 to 999: #define MAXMSG 100 In each directory: make clean; make update; make install ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] PIX Firewall Ports and Access-Lists
OK, I found the command: access-list acl-prod permit udp any host EXTERNAL_*_IP_HERE range 1 2 Unfortunately that doesn't help my intermittent audio issue as I had hoped. When I view the access list, that rule doesn't report any hits so I'm not sure it is being used. Any suggestions as to port forwarding the correct ports through our PIX and if that has an effect on my audio issues? We have old firmaware that has a bug in the SIP Fixup so it has been turned off. Could that be the issue? -Original Message- From: Geoff Manning To: Asterisk Users (E-mail) Sent: 6/17/05 1:29 PM Subject: [Asterisk-Users] PIX Firewall Ports and Access-Lists Hello, I am not too familiar with the settings in our PIX (learning though). Here is the only access-list setting that we have in place for Asterisk: access-list acl-prod permit udp any host EXTERNAL_*_IP_HERE eq 5060 In rtp.conf we are allowing ports 1 - 2. We are not using SIP Fixup in our PIX due to firmware version. How do I go about adding the ability for udp ports 1 - 2 to forward to our Asterisk server? We have intermittent audio issues on calls and I have narrowed it down (hopefully) to the PIX. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] FATAL: Error running install command for wctdm
Ronald Wiplinger wrote: app_addon_sql_mysql.so app_intercom.so app_saycountpl.so cdr_addon_mysql.so format_mp3.so res_config_mysql.so WARNING WARNING WARNING I cannot remember that I have seen that before. you must have checkout'd asterisk-addons and compiled it at some point. so, you should update those also, and recompile them. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] FATAL: Error running install command for wctdm
I got a new motherboard and upgraded Asterisk, ... However, I am not lucky enough to get it running again. To compile went ok, besides this remark at the end of asterisk: WARNING WARNING WARNING Your Asterisk modules directory, located at /usr/lib/asterisk/modules contains modules that were not installed by this version of Asterisk. Please ensure that these modules are compatible with this version before attempting to run Asterisk. app_addon_sql_mysql.so app_intercom.so app_saycountpl.so cdr_addon_mysql.so format_mp3.so res_config_mysql.so WARNING WARNING WARNING I cannot remember that I have seen that before. asterisk-restart Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) FATAL: Error inserting wctdm (/lib/modules/2.6.8-24.11-default/extra/wctdm.ko): Unknown symbol in module, or unknown parameter (see dmesg) FATAL: Error running install command for wctdm vpbx:/usr/local/src/asterisk-sounds # Asterisk ended with exit status 1 Asterisk died with code 1. Automatically restarting Asterisk. Freed a Wildcard module wctdm unsupported by SUSE/Novell, tainting kernel. wctdm: disagrees about version of symbol zt_receive wctdm: Unknown symbol zt_receive wctdm: disagrees about version of symbol zt_qevent_lock wctdm: Unknown symbol zt_qevent_lock wctdm: disagrees about version of symbol zt_ec_chunk wctdm: Unknown symbol zt_ec_chunk wctdm: disagrees about version of symbol zt_transmit wctdm: Unknown symbol zt_transmit wctdm: disagrees about version of symbol zt_unregister wctdm: Unknown symbol zt_unregister wctdm: disagrees about version of symbol zt_hooksig wctdm: Unknown symbol zt_hooksig wctdm: disagrees about version of symbol zt_register wctdm: Unknown symbol zt_register /lib/modules/2.6.8-24.11-default # ls -l misc total 3610 drwxr-xr-x 2 root root464 Apr 17 06:36 . drwxr-xr-x 5 root root568 Jun 18 22:23 .. -rw-r--r-- 1 root root 250382 Apr 17 06:36 pciradio.ko -rw-r--r-- 1 root root 242487 Apr 17 06:36 tor2.ko -rw-r--r-- 1 root root 177722 Apr 17 06:36 torisa.ko -rw-r--r-- 1 root root 158258 Apr 17 06:36 wcfxo.ko -rw-r--r-- 1 root root 211775 Apr 17 06:36 wct1xxp.ko -rw-r--r-- 1 root root 722472 Apr 17 06:36 wct4xxp.ko -rw-r--r-- 1 root root 383373 Apr 17 06:36 wctdm.ko -rw-r--r-- 1 root root 285014 Apr 17 06:36 wcte11xp.ko -rw-r--r-- 1 root root 217011 Apr 17 06:36 wcusb.ko -rw-r--r-- 1 root root 534695 Apr 17 06:36 zaptel.ko -rw-r--r-- 1 root root 140851 Apr 17 06:36 ztd-eth.ko -rw-r--r-- 1 root root 123269 Apr 17 06:36 ztd-loc.ko -rw-r--r-- 1 root root 91057 Apr 17 06:36 ztdummy.ko -rw-r--r-- 1 root root 123237 Apr 17 06:36 ztdynamic.ko /lib/modules/2.6.8-24.11-default # ls -l extra total 10622 drwxr-xr-x 2 root root2072 Jun 18 22:23 . drwxr-xr-x 5 root root 568 Jun 18 22:23 .. -rw-r--r-- 1 root root 254027 Jan 15 00:13 acx_pci.ko -rw-r--r-- 1 root root 78181 Jan 15 00:13 adm8211.ko -rw-r--r-- 1 root root 10837 Jan 15 00:13 at76_usbdfu.ko -rw-r--r-- 1 root root 10750 Jan 15 00:13 at76c503-i3861.ko -rw-r--r-- 1 root root9104 Jan 15 00:13 at76c503-i3863.ko -rw-r--r-- 1 root root9110 Jan 15 00:13 at76c503-rfmd-acc.ko -rw-r--r-- 1 root root 11160 Jan 15 00:13 at76c503-rfmd.ko -rw-r--r-- 1 root root 143088 Jan 15 00:13 at76c503.ko -rw-r--r-- 1 root root8936 Jan 15 00:13 at76c505-rfmd.ko -rw-r--r-- 1 root root9684 Jan 15 00:13 at76c505-rfmd2958.ko -rw-r--r-- 1 root root9112 Jan 15 00:13 at76c505a-rfmd2958.ko -rw-r--r-- 1 root root 202617 Jan 15 00:13 ath_hal.ko -rw-r--r-- 1 root root 86456 Jan 15 00:13 ath_pci.ko -rw-r--r-- 1 root root 15618 Jan 15 00:13 ath_rate_onoe.ko -rw-r--r-- 1 root root6416 Jan 15 00:13 av5100.ko -rw-r--r-- 1 root root 20183 Jan 15 00:13 cloop.ko -rw-r--r-- 1 root root 232828 Jan 15 00:13 drbd.ko -rw-r--r-- 1 root root 159960 Jan 15 00:13 hostap.ko -rw-r--r-- 1 root root 19838 Jan 15 00:13 hostap_crypt_ccmp.ko -rw-r--r-- 1 root root 16392 Jan 15 00:13 hostap_crypt_tkip.ko -rw-r--r-- 1 root root9597 Jan 15 00:13 hostap_crypt_wep.ko -rw-r--r-- 1 root root 83500 Jan 15 00:13 hostap_cs.ko -rw-r--r-- 1 root root 77600 Jan 15 00:13 hostap_pci.ko -rw-r--r-- 1 root root 79617 Jan 15 00:13 hostap_plx.ko -rw-r--r-- 1 root root 648077 Jan 15 00:13 ieee80211.ko -rw-r--r-- 1 root root 131109 Jan 15 00:13 ieee80211_crypt.ko -rw-r--r-- 1 root root 139271 Jan 15 00:13 ieee80211_crypt_ccmp.ko -rw-r--r-- 1 root root 154982 Jan 15 00:13 ieee80211_crypt_tkip.ko -rw-r--r-- 1 root root 124535 Jan 15 00:13 ieee80211_crypt_wep.ko -rw-r--r-- 1 root root 996760 Jan 15 00:13 ipw2100.ko -rw-r--r-- 1 root root 793716 Jan 15 00:13 ipw2200.ko -rw-r--r-- 1 root root 289300 Jan 15 00:13 iscsi.ko -rw-r--r-- 1 root root 787123 Jan 15 00:12 libafs.ko -rw-r--r-- 1 root root 186642 Jan 15 00:13 megaide.ko -rw-r--r-- 1 root root 51402 Jan 15 00:13 p80211.ko -rw-r--r-- 1 root root6592 Jan 15 00:13 pbe5.ko -rw-r--r-- 1 root root
Re: [Asterisk-Users] ASTCC Rate Calculation
Ronald Wiplinger wrote: Darren Wiebe wrote: Good Day Has anybody here looked closely at the call cost calculation in ASTCC? Can you duplicate the way the cost of a call is calculated? I believe that there is an error in the code. I have fixed it, I think and submitted a patch but we need user comments. I would appreciate if anybody involved would slip over to chech out this link on the bugtracker and provide feedback. http://bugs.digium.com/view.php?id=4480 I may well be wrong but I believe the issue needs visiting. Somebody was asking me how it calculates costs as they thought they knew what a call should cost. I said "I'll show you". Mistake, I could not come up with an answer that made sense. Sorry, I posted the wrong link. http://bugs.digium.com/view.php?id=4479 the "update database" for users is removed as it is not needed and has been moved into an update datbase button on the main configure page. You should open a bug report for your tariff patch. Darren Wiebe [EMAIL PROTECTED] Darren, looking at the page you will find two patches. The description is very short. Which patch do you want us to instal? If one is wrong, would you please delete it. What should be the purpose of the patch? "Calc charges" does not give me a clear picture of it. Looking at the patch (picked one) I see that the entire "update database" is deleted. How should I than alter the tables? Another question regarding my tarriff patch, ... How to get this into CVS? bye Ronald Please let me know, Darren Wiebe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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] Unable to find a path from g729 to gsm
> Now, with some hard time and help from many genurous people's in the list, I > have come to this point with my TDM20B card & my teliax's IAX2 account. > > I hope someone may help me with this issue mentioned below. I have already > selected my codec as gms in my iax.conf as well as in teliax's "my account > page" but still i have the same error when I attempt to make a call. Based on the CLI stuff shown below, it would appear you have sip.conf entries for your phone defined as g729 and you haven't purchased/installed that codec. Either purchase it (www.digium.com) or change your phone definitions to something usable (g711u). > Second, my last digit is not allowed from teliax. that means I need one more > digit from teliax for dialing through them. This also sounds like a user configuration error. Without seeing the appropriate sections of sip.conf, extensions.conf, and iax.conf, its almost impossible to answer your questions with pure guessing. > Third, I have somewhat poor support from teliax since I have send them 3,4 > emails and so far i got no replies. As noted above, your issues seem to be asterisk configuration issues and teliax cannot be expected to resolve those for you particularily with no config data, etc, supplied. > The error I got > > Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a > path from g729 to gsm If teliax is truly using the gsm codec, then apparently your phone is configured to use g729 only. Change that to g711u and try again. > Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a > path from g729 to gsm > Jun 17 18:47:05 NOTICE[7396]: channel.c:1884 set_format: Unable to find a > path from g729 to gsm > -- IAX2/teliax-1 is ringing > -- IAX2/teliax-1 answered Zap/1-1 > Jun 17 18:47:18 WARNING[7396]: channel.c:2308 ast_channel_make_compatible: > No path to translate from Zap/1-1(68) to IAX2/teliax-1(256) > Jun 17 18:47:18 WARNING[7396]: app_dial.c:1324 dial_exec_full: Had to drop > call because I couldn't make Zap/1-1 compatible with IAX2/teliax-1 > -- Hungup 'IAX2/teliax-1' > == Spawn extension (outgoing, 19737228839, 1) exited non-zero on 'Zap/1-1' > -- Hungup 'Zap/1-1' The above does say you are sending the appropriate number of digits to teliax. Why are you thinking that you need to send "one more digit"? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] allowing outside dialing only for SIP users
Hi,
I looked at this article, and I cam up with this extension.conf
[common]
exten => 100,1,Voicemail(100)
exten => 101,1,Dial(Modem/ttyI2:**11)
exten => 102,1,Dial(Modem/ttyI2:**12)
exten => 111,1,Dial(SIP/prnet-win)
exten => 112,1,Dial(SIP/prnet-amilo)
exten => 8500,1,VoicemailMain
[default]
exten => s,1,Wait,1
exten => s,n,Answer
exten => s,n(restart),BackGround(demo-congrats)
exten => s,n(instruct),BackGround(demo-instruct)
exten => s,n,WaitExten
include => common
[callout]
include => common
exten => _X.,1,Dial(Modem/ttyI1:${EXTEN})
In sip.conf I added context: callout to a sip user.
If called in via phone, context default is called, and the items from
context common are included correctly. If called in via this SIP user,
it seems only respect the exten line and not the lines from context
common. Anyone knowing what could go wrong here ? If I add context:
default to a sip user, the exten lines from context common are respected.
Thanks in adavance,
Bye,
David Arendt
Wilson Pickett wrote:
allow this extension for outside
dialing only for SIP users ?
This is exactly what contexts are for. Put all sip users in one
context and everyone else in one or more contexts. Then put that line
(if you get it working) only in the sip users context and it will not
be available to anyone else.
If you have no idea of what was just said above, you should read the
available documents at
http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html
http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html
http://www.asteriskdocs.org
http://www.voip-info.org/wiki-Asterisk+Dialplan+Introduction
The last URL was obtained by using google for asterisk context dialplan
hth
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回复： [Asterisk-Users] How to view all call detail in asterisk server
You can use cdr analyser. U can find it at the http://areski.net/asterisk-stat-v2/about.phpSUBHASH RAVADA <[EMAIL PROTECTED]> 写道： Hi,I would like to see call details for both incoming and outgoing calls. What commands do i have to issue to view or export to any file.after i logged into asterisk i am at *CLI>The module /var/log/asterisk/cdr-csv was loaded when i checked.Could someone tell me how to view and import all the call details into any file.Thanks you,SubRav___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users Best Regards, Gary Li__赶快注册雅虎超大容量免费邮箱?http://cn.mail.yahoo.com___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re: Analog modems behind an Asterisk server?
> >>Hello, we would like to hook up analog modems behind an Asterisk server, > >>and we're very interested in the experiences that others have made when > >>attempting that. We assume that there are no inherent problems with > >>modems in respect to the Asterisk software, but it appears that the > >>FXO/FXS hardware restricts this kind of a setup to lower data > >>transmission rates, is this correct? > >> > >>Currently, we only transmit at 1200bps, is this rate problematic with > >>Digium cards? Up to what data transmission rate are Digium cards known > >>to work reliable? We do not think we'll ever go beyond 9600bps, can we > >>do this with a let's say TDM400P? > >> > >>Will future Digium hardware improve the situation or will this stay the > >>same in the future? > >> > >>How is hardware from other vendors performing when using analog modems? > > > > > > The success rate of moving modem data through a digium analog TDM card > > (fxo & fxs ports) varies and appears to be somewhat related to the > > exact motherboard in use. The card very very frequently has an issue > > with missed data across the pci bus (card to motherboard). The missed > > data negatively impacts any modem call regardless of whether its a > > fax machine or pc modem. You are likely to have less then a 50% chance > > of making work correctly. > > Since we will assemble the Asterisk hosts ourselves we'll have full > control over the hardware components, including the motherboard. So, if > we choose carefully, can we expect stable results? I'm guessing the answer will likely be yes, however after messing with the TDM for about a year now and working with several others on this list, I've not found anyone as yet that can truly say they have a TDM card working with any motherboard that is not dropping data across the pci bus. Keep in mind also the TDM card has also gone through several design change iterations and it appears the latest is Rev I. Since there is absolutely no data available relative to what changes have been incorporated into each board revision and no hint of what those changes are impacting, its impossible to pick specific combinations of motherboards and TDM cards that actually work for anything other then voice. I think I've spent in access of 200 hours messing with it, rebuilding the system, different OS distros, OS patches, every combo of low latency changes, and have not found a combo that works as yet. Audio is no problem, modem data is a problem. > > Before a pile of people jump in to say "it works for me", keep in mind > > that various types of modems use different modulation schemes and some > > are more sensitive than others to distorted audio (missed data). In > > very general terms, the higher the modem speed the more likely it will > > be negatively impacted by the distortion (missed data). > > > > If you're not familiar with modem technology, I might add there are two > > primary items that are directly related to the modem's audio across > > analog lines. The "baud" rate of analog signal on the wire and the > > bit rate of encoded digital data. You might have a current modem that > > allows you to change the bit rate (digital side), but on most modems > > you have no control over the analog baud rate (or modulation scheme). > > So, changing the modem's bit rate won't impact how well the modem > > actually works through the TDM card. > > > > Some people have reported that point of sale and credit card authorization > > boxes have worked via the TDM card. However, the modem's used in that > > equipment typically are very slow speed modems that were intended to > > function in any business environment including those with noisy > > telephone lines. Those have a higher possibility of success, but > > should not be interpreted as being the same as a modem used with PC's, > > etc. > > Our modems are intended for use on noisy lines, that's also the reason > why we stick to such low data transmission rates (1200 baud, 1200bps), > we're really not planning to use anything that transmits more than > 9600bps. We've successfully bridged calls across various ISDN a/b > adapters and also across Voip/FXS-Gatway adapters (tiptel innovaphone 21). There are lots of * implementations that function correctly with PRI's and other digital interfaces. Seems the analog TDM card and/or drivers is _the_ issue. > > Bottom line... you will have far less then a 50% chance of making any > > PC modem work at acceptable speed through a TDM card. > > Hmm, I think I'll give it a try once I find stable hardware components. That's the only way that I know of. It would be helpful for others if you'd keep track of what works and what doesn't and post back to the list. > > The latest code for the Sipura boxes (spa3k) appear to have addressed > > modem signals (fxs to fxo). I just upgraded two spa3k's to that latest > > firmware, but have not attempted to use any modem through it. Might > > check to see if anyone
Re: [Asterisk-Users] ASTCC Rate Calculation
Darren Wiebe wrote: Good Day Has anybody here looked closely at the call cost calculation in ASTCC? Can you duplicate the way the cost of a call is calculated? I believe that there is an error in the code. I have fixed it, I think and submitted a patch but we need user comments. I would appreciate if anybody involved would slip over to chech out this link on the bugtracker and provide feedback. http://bugs.digium.com/view.php?id=4480 I may well be wrong but I believe the issue needs visiting. Somebody was asking me how it calculates costs as they thought they knew what a call should cost. I said "I'll show you". Mistake, I could not come up with an answer that made sense. Darren, looking at the page you will find two patches. The description is very short. Which patch do you want us to instal? If one is wrong, would you please delete it. What should be the purpose of the patch? "Calc charges" does not give me a clear picture of it. Looking at the patch (picked one) I see that the entire "update database" is deleted. How should I than alter the tables? Another question regarding my tarriff patch, ... How to get this into CVS? bye Ronald Please let me know, Darren Wiebe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com+886 (0) 939--77-55-16 or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK SMS Config problems
All,
I'm running CVS-HEAD as of 15thJune with an x100p and the x100p callerid
patch. I'm trying to use app_sms to recieve sms to my landline but get
the following response.
Any ideas.
-- Executing NoOp("Zap/1-1", "Testing without Answer 08005875290") in
new stack
-- Executing GotoIf("Zap/1-1", "0?s|5") in new stack
-- Executing GotoIf("Zap/1-1", "1?s|7") in new stack
-- Goto (pstn-inbound,s,7)
-- Executing Goto("Zap/1-1", "sms-in|s|1") in new stack
-- Goto (sms-in,s,1)
-- Executing Answer("Zap/1-1", "") in new stack
-- Executing Wait("Zap/1-1", "1") in new stack
-- Executing SMS("Zap/1-1", "default|a") in new stack
-- SMS TX 93 00 6D
-- SMS RX 93 00 6D
-- SMS TX 94 00 6C
-- SMS TX 92 01 FF 6E
-- SMS TX 92 01 FF 6E
== Spawn extension (sms-in, s, 3) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
And nothing in my /var/spool/asterisk.
extensions.conf
[sms-in]
exten => s,1,Answer
exten => s,2,Wait(1)
exten => s,3,SMS(default|a)
exten => s,4,Hangup
[pstn-inbound]
exten => s,1,Noop(Testing without Answer ${CALLERIDNUM})
exten => s,2,GotoIf($["${CALLERIDNUM}" = ""]?s|5)
exten => s,3,GotoIf($["${CALLERIDNUM}" = "08005875290"]?s,7)
exten => s,4,Dial(SIP/2204&SIP/2205,30,w)
exten => s,5,Voicemail(u2204)
exten => s,6,Hangup
exten => s,7,Goto(sms-in,s,1)
exten => fax,1,SetVar(FAXFILE=/tmp/faxfor-${EXTEN}-${TIMESTAMP}.tif)
exten => fax,2,rxfax(${FAXFILE})
exten => fax,3,Hangup
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[Asterisk-Users] E1 to E1 connection
Hi, I want to connect Asterisk to Avaya through E1 ports. Shall I use cross cable or patch cord (straight connection)? Regards, Cenk. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] error compiling asterisk
Dave Cotton wrote: On Sat, 2005-06-18 at 17:00 +0800, Ronald Wiplinger wrote: app_rxfax.c: In function `phase_e_handler': app_rxfax.c:93: error: structure has no member named `callerid' app_rxfax.c: At top level: app_rxfax.c:61: warning: `t30_flush' defined but not used make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/local/src/asterisk/apps' How to fix that??? change app_rxfax.c //#if (ASTERISK_VERSION_NUM <= 010300) // chan->callerid, //#else (chan->cid.cid_num) ? chan->cid.cid_num : "", //#endif Would that also mean that I could not receive fax anymore on my Zap line??? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] error compiling asterisk
On Sat, 2005-06-18 at 17:00 +0800, Ronald Wiplinger wrote: > app_rxfax.c: In function `phase_e_handler': > app_rxfax.c:93: error: structure has no member named `callerid' > app_rxfax.c: At top level: > app_rxfax.c:61: warning: `t30_flush' defined but not used > make[1]: *** [app_rxfax.o] Error 1 > make[1]: Leaving directory `/usr/local/src/asterisk/apps' > > > How to fix that??? change app_rxfax.c //#if (ASTERISK_VERSION_NUM <= 010300) // chan->callerid, //#else (chan->cid.cid_num) ? chan->cid.cid_num : "", //#endif -- Dave Cotton <[EMAIL PROTECTED]> ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] error compiling asterisk
app_rxfax.c: In function `phase_e_handler': app_rxfax.c:93: error: structure has no member named `callerid' app_rxfax.c: At top level: app_rxfax.c:61: warning: `t30_flush' defined but not used make[1]: *** [app_rxfax.o] Error 1 make[1]: Leaving directory `/usr/local/src/asterisk/apps' How to fix that??? bye Ronald ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TTS
aside from festival are there are other TTS engines out there that are free? I have written a simple script to snarf files from a foreign site with a really good TTS engine, but there is a lot of latency so I was looking to use something on my system, however festival is hard for me to understand (far too mechanical). Basically the script I have uses sitepal.com's TTS engine (after a fashion) by grabbing the SWF of a phrase, which includes (pretty much only) an MP3 which I SWFrip, convert and then play. I have implemented a caching system in there so its tolerable for static content, but live content is much too slow. I cant really complain since this is not an 'approved use' of sitepals system, and wanted something that was ... better on a variety of fronts. -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US +1 360 207 0479 or +1 516 687 5200 FreeWorldDialup: 635378 signature.asc Description: This is a digitally signed message part ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Re: Asterisk ael files
> I don't think so. It appears to be a front-end compiler that reads the > new AEL syntax and compiles it into the existing internal dialplan > representation (but I could be wrong, I haven't studied the fine detail). That sounds like what Mark said the goal of AEL was. It is just another way to declare the same intensions in a dialplan with no runtime penalty, but using a language that is a little more structured (meaning it looks like c). The resulting dialplan is stored exactly the same way. Quoting JerJer, "it's another way to write spaghetti coede that no one else will be able to understand." I think they're both right :) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Local numbers
> If I set up an * server will I still be able to use my local Anchorage > phone number through my * box? If you are asking if you can hook up asterisk to your POTS line, you only need an FXO interface and zaptel to do it. However, you should avoid trying to have asterisk in your bedroom connected to the phone line and family member or office co-workers using that same line with their phones. It won't work our well as the left can't see what the right is doing. Asterisk should be the "boss" of any POTS line it is connected to. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] allowing outside dialing only for SIP users
> allow this extension for outside > dialing only for SIP users ? This is exactly what contexts are for. Put all sip users in one context and everyone else in one or more contexts. Then put that line (if you get it working) only in the sip users context and it will not be available to anyone else. If you have no idea of what was just said above, you should read the available documents at http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html http://www.asteriskdocs.org http://www.voip-info.org/wiki-Asterisk+Dialplan+Introduction The last URL was obtained by using google for asterisk context dialplan hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Analog modems behind an Asterisk server?
Rich Adamson wrote: Hello, we would like to hook up analog modems behind an Asterisk server, and we're very interested in the experiences that others have made when attempting that. We assume that there are no inherent problems with modems in respect to the Asterisk software, but it appears that the FXO/FXS hardware restricts this kind of a setup to lower data transmission rates, is this correct? Currently, we only transmit at 1200bps, is this rate problematic with Digium cards? Up to what data transmission rate are Digium cards known to work reliable? We do not think we'll ever go beyond 9600bps, can we do this with a let's say TDM400P? Will future Digium hardware improve the situation or will this stay the same in the future? How is hardware from other vendors performing when using analog modems? The success rate of moving modem data through a digium analog TDM card (fxo & fxs ports) varies and appears to be somewhat related to the exact motherboard in use. The card very very frequently has an issue with missed data across the pci bus (card to motherboard). The missed data negatively impacts any modem call regardless of whether its a fax machine or pc modem. You are likely to have less then a 50% chance of making work correctly. Since we will assemble the Asterisk hosts ourselves we'll have full control over the hardware components, including the motherboard. So, if we choose carefully, can we expect stable results? Before a pile of people jump in to say "it works for me", keep in mind that various types of modems use different modulation schemes and some are more sensitive than others to distorted audio (missed data). In very general terms, the higher the modem speed the more likely it will be negatively impacted by the distortion (missed data). If you're not familiar with modem technology, I might add there are two primary items that are directly related to the modem's audio across analog lines. The "baud" rate of analog signal on the wire and the bit rate of encoded digital data. You might have a current modem that allows you to change the bit rate (digital side), but on most modems you have no control over the analog baud rate (or modulation scheme). So, changing the modem's bit rate won't impact how well the modem actually works through the TDM card. Some people have reported that point of sale and credit card authorization boxes have worked via the TDM card. However, the modem's used in that equipment typically are very slow speed modems that were intended to function in any business environment including those with noisy telephone lines. Those have a higher possibility of success, but should not be interpreted as being the same as a modem used with PC's, etc. Our modems are intended for use on noisy lines, that's also the reason why we stick to such low data transmission rates (1200 baud, 1200bps), we're really not planning to use anything that transmits more than 9600bps. We've successfully bridged calls across various ISDN a/b adapters and also across Voip/FXS-Gatway adapters (tiptel innovaphone 21). Bottom line... you will have far less then a 50% chance of making any PC modem work at acceptable speed through a TDM card. Hmm, I think I'll give it a try once I find stable hardware components. The latest code for the Sipura boxes (spa3k) appear to have addressed modem signals (fxs to fxo). I just upgraded two spa3k's to that latest firmware, but have not attempted to use any modem through it. Might check to see if anyone else on the list have tried it. The firmware was just released in the last day or two, so it might take a little while for folks to try it. Thank you very much for your insightful reply, Christian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk ael files
In article <[EMAIL PROTECTED]>, Shidan <[EMAIL PROTECTED]> wrote: > Hi noticing the cvs updates of late, I'm wondering if there is support > for fifo/shell commands in the extended dialplan language? can it > fully replace agi scripts? Looks really interesting... I don't think so. It appears to be a front-end compiler that reads the new AEL syntax and compiles it into the existing internal dialplan representation (but I could be wrong, I haven't studied the fine detail). I'm surprised no-one else has commented on the AEL stuff since it appeared! Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to view all call detail in asterisk server
Hi, I would like to see call details for both incoming and outgoing calls. What commands do i have to issue to view or export to any file.after i logged into asterisk i am at *CLI> The module /var/log/asterisk/cdr-csv was loaded when i checked. Could someone tell me how to view and import all the call details into any file. Thanks you, SubRav ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Analog modems behind an Asterisk server?
Florian Overkamp wrote: On a pure TDM path this should be fine. In fact I think you should not have any real limitation if you set everything correctly. By "pure TDM path", do you mean a) within a single TDM card or b) between two TDM cards within the same host? Thanks, Christian. Using VoIP in parts of the link does limit the connection, although we have seen 14k4 connections run stable for a long time. YMMV. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] allowing outside dialing only for SIP users
Hi,
I have one questions, which might be obvious but not for me as I am new
to asterisk.
How do I modify that line in order to allow this extension for outside
dialing only for SIP users ?
exten => _X.,1,Dial(Modem/ttyI1:${EXTEN})
Thanks in advance
Bye,
David Arendt
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