RE: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
> > Not knowing about the switch side of things, but how many interfaces > would a PRI card be able to handle in that switch? I'm betting for $35k > it is quite a few. It may be something to sit down with your copy of the > local tariffs and decide how many circuits over how many months would > pay off that card and make the business pitch to get it. > The card outputs 30 D channels. You still use 24 channels out of your IMT trunks, mapping channel 24 to one of the 30 D channels. Eventually I am sure they will get the card. They were hoping we could make things work in the short term some other way. Might just have to bite the bullet. > Your only other potential solution would probably involve SS7, and that > isn't supported under asterisk now. SS7 on asterisk would be nice for many reasons. Thanks for all your answers. ___ 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] Strange T1 Problem - FIXED plus new question
On Fri, 2004-04-16 at 11:05, Mike Machado wrote: > My situation is a little different. The carrier switch is about 100 feet > away from my asterisk box. The company I am working for is a CLEC and > they have their own switch. The switch I am connected to does not have a > very expensive PRI signaling card ($35k), so they can only do CAS. > > If I was a customer and getting service from a carrier, I would > definitely have gotten a PRI. Thanks for your perspective on this. Not knowing about the switch side of things, but how many interfaces would a PRI card be able to handle in that switch? I'm betting for $35k it is quite a few. It may be something to sit down with your copy of the local tariffs and decide how many circuits over how many months would pay off that card and make the business pitch to get it. Your only other potential solution would probably involve SS7, and that isn't supported under asterisk now. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] Strange T1 Problem - FIXED plus new question
On Fri, 2004-04-16 at 08:45, mattf wrote: > Sadly it's a limited to PRIs(at least that's what I've been told) You just > can't send that much call data with a good old non-PRI T1. The up side is > you get one extra voice channel to use as compared to a PRI. It's strange > that the carrier doesn't have PRI cabability, I've run into many more > carriers that can't do non-PRI. I've even had one that couldn't send me > ANI(CallerID) on non-PRI lines. > > Once you get into the world of ISDN and PRI, you start to be able to do a > lot more with the signaling of calls, and you even have the ability to do > faster call switching as compared to a non-PRI. > > My advice is to get a different carrier, there are hundreds out there. And > if you absolutely have to have dynamic CallerID transmission you should > verify that your new carrier will let you do that before you sign a > contract(Make sure you verify it by talking to an actual switch tech from > the carrier, sales people will lie through their teeth to get you to sign > that contract). > > Hope this helps, > > MATT--- > My situation is a little different. The carrier switch is about 100 feet away from my asterisk box. The company I am working for is a CLEC and they have their own switch. The switch I am connected to does not have a very expensive PRI signaling card ($35k), so they can only do CAS. If I was a customer and getting service from a carrier, I would definitely have gotten a PRI. Thanks for your perspective on this. ___ 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] Strange T1 Problem - FIXED plus new question
On Fri, 2004-04-16 at 10:46, Ryan Thrash wrote: > First, check zapata.conf to see what is in there. Next, I've not heard > of any luck with the name portion on T1s, but the number can be changed > for us. The name portion is no problem if you are on a PRI. I personally have had fun spoofing the callerid during some prank calls to my friends. In a channelized T1 like is mentioned below though, you don't have enough signaling to specify it. FATIK, you can oply specify the outgoing digits via some prearranged length and protocol similar to callerid and dnis on inbound calls. I'm not sure if asterisk supports this on channelized T1. > On Apr 16, 2004, at 10:30 AM, Mike Machado wrote: > > > > >>> > >> You can usually get CLI on an E&M robbed bit T1 by configuring it > >> right. > >> Instead of just sending you the DNIS as a string of DTMF they usually > >> send ***. The DNIS and CLI may be swapped, and there may be > >> less than 3 *s in the string - wonderful consistency, eh? :-\ > > > > I am getting CallerID and DNIS on the inbound calls. What I really need > > is to be able to set callerID on outbound calls. I am trying to set the > > callerid using SetCIDNum just before using Dial on a zap channel, but > > it > > looks like the switch guys have it set to always stamp the same > > callerID > > on the my outbound calls no matter what I put in SetCIDNum or what > > channel on the T1 I use. Is this a misconfiguration of the switch or a > > limitation of the signaling protocol? If its the switch, can you give > > me > > any pointers as to what I could ask them to look for, or if its the > > protocol, do you know any other signaling protocol that lets me set > > outbound callerID (besides PRI)? > -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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] Strange T1 Problem - FIXED plus new question
First, check zapata.conf to see what is in there. Next, I've not heard of any luck with the name portion on T1s, but the number can be changed for us. HTH, Ryan On Apr 16, 2004, at 10:30 AM, Mike Machado wrote: You can usually get CLI on an E&M robbed bit T1 by configuring it right. Instead of just sending you the DNIS as a string of DTMF they usually send ***. The DNIS and CLI may be swapped, and there may be less than 3 *s in the string - wonderful consistency, eh? :-\ I am getting CallerID and DNIS on the inbound calls. What I really need is to be able to set callerID on outbound calls. I am trying to set the callerid using SetCIDNum just before using Dial on a zap channel, but it looks like the switch guys have it set to always stamp the same callerID on the my outbound calls no matter what I put in SetCIDNum or what channel on the T1 I use. Is this a misconfiguration of the switch or a limitation of the signaling protocol? If its the switch, can you give me any pointers as to what I could ask them to look for, or if its the protocol, do you know any other signaling protocol that lets me set outbound callerID (besides PRI)? ___ 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] Strange T1 Problem - FIXED plus new question
Sadly it's a limited to PRIs(at least that's what I've been told) You just can't send that much call data with a good old non-PRI T1. The up side is you get one extra voice channel to use as compared to a PRI. It's strange that the carrier doesn't have PRI cabability, I've run into many more carriers that can't do non-PRI. I've even had one that couldn't send me ANI(CallerID) on non-PRI lines. Once you get into the world of ISDN and PRI, you start to be able to do a lot more with the signaling of calls, and you even have the ability to do faster call switching as compared to a non-PRI. My advice is to get a different carrier, there are hundreds out there. And if you absolutely have to have dynamic CallerID transmission you should verify that your new carrier will let you do that before you sign a contract(Make sure you verify it by talking to an actual switch tech from the carrier, sales people will lie through their teeth to get you to sign that contract). Hope this helps, MATT--- -Original Message- From: Mike Machado [mailto:[EMAIL PROTECTED] Sent: Friday, April 16, 2004 11:30 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question > > > You can usually get CLI on an E&M robbed bit T1 by configuring it right. > Instead of just sending you the DNIS as a string of DTMF they usually > send ***. The DNIS and CLI may be swapped, and there may be > less than 3 *s in the string - wonderful consistency, eh? :-\ I am getting CallerID and DNIS on the inbound calls. What I really need is to be able to set callerID on outbound calls. I am trying to set the callerid using SetCIDNum just before using Dial on a zap channel, but it looks like the switch guys have it set to always stamp the same callerID on the my outbound calls no matter what I put in SetCIDNum or what channel on the T1 I use. Is this a misconfiguration of the switch or a limitation of the signaling protocol? If its the switch, can you give me any pointers as to what I could ask them to look for, or if its the protocol, do you know any other signaling protocol that lets me set outbound callerID (besides PRI)? ___ 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] Strange T1 Problem - FIXED plus new question
> > > You can usually get CLI on an E&M robbed bit T1 by configuring it right. > Instead of just sending you the DNIS as a string of DTMF they usually > send ***. The DNIS and CLI may be swapped, and there may be > less than 3 *s in the string - wonderful consistency, eh? :-\ I am getting CallerID and DNIS on the inbound calls. What I really need is to be able to set callerID on outbound calls. I am trying to set the callerid using SetCIDNum just before using Dial on a zap channel, but it looks like the switch guys have it set to always stamp the same callerID on the my outbound calls no matter what I put in SetCIDNum or what channel on the T1 I use. Is this a misconfiguration of the switch or a limitation of the signaling protocol? If its the switch, can you give me any pointers as to what I could ask them to look for, or if its the protocol, do you know any other signaling protocol that lets me set outbound callerID (besides PRI)? ___ 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] Strange T1 Problem
Hello, From: "Joe Dennick" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Strange T1 Problem Date: Fri, 16 Apr 2004 07:44:17 -0500 Can one use a pipe '|' for the Dial application the same way that one would use a comma ','? I know this one works, but what I don't know is if it will also work using pipes in place of the commas. Joe Yes, You can use '|' for the dial application. In fact even if you use comma(,) in your extensions.conf, Asterisk replaces it with '|' when it builds the dial plan. See the following entry in extensions.conf: [test] exten => 1234,1,Dial(SIP/1234,20,r) exten => 1234,2,Voicemail(u1234) and the dialplan for this is: * CLI> show dialplan test [ Context 'test' created by 'pbx_config' ] '1234' => 1. Dial(SIP/1234|20|r) [pbx_config] 2. Voicemail(u1234) [pbx_config] Regards, Girish _ Easiest Money Transfer to India. Send Money To 6000 Indian Towns. http://go.msnserver.com/IN/42198.asp Easiest Way To Send Money Home! ___ 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] Strange T1 Problem
Can one use a pipe '|' for the Dial application the same way that one
would use a comma ','?
My dialplan looks like this:
exten => 1234,1,Dial(SIP/1234,20,r)
exten => 1234,2,Voicemail(u1234)
etc
I know this one works, but what I don't know is if it will also work
using pipes in place of the commas.
Joe
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, April 16, 2004 7:30 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Strange T1 Problem
Lookie here:
This is what you have
> exten => 1234567890,2,Dial(SIP/user1|r)
But, perhaps, here's what it shouls be:
exten => 1234567890,2,Dial(SIP/user1||r)
The second argument is *timeout*.
Normally you'd have something like
Dial(Channel,time,options)
exten => 1234567890,2,Dial(SIP/user1|60|r)
But the empty time works as well. It will just ring
forever.
Cheers,
Willy
- Original Message Follows -
>
> On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
>
> > Explicitly answer the line. If that doesn't handle
> > inband audio, there is a r flag to dial. This was
> discussed very recently.
>
> This must be a different problem, because neither of those solutions
> worked.
>
>
>
> zapata.conf sends call to fixup context:
>
>
> [fixup]
>
> ; Receive call as **
> exten => _.,1,Answer
> exten => _.,2,Cut(CALLING=EXTEN,*,2)
> exten => _.,3,SetCIDNum(${CALLING})
> exten => _.,4,Cut(CALLED=EXTEN,*,3)
> exten => _.,5,Goto(default|${CALLED}|1)
>
>
> [default]
>
> exten => 1234567890,1,Answer
> exten => 1234567890,2,Dial(SIP/user1|r)
>
>
> user1's phone rings, but no ring from PSTN caller. user1 picks up,
> both can talk ok.
>
>
> I have been using cvs stable branch. I will try HEAD and
> see if that fixes it as suggested by Eric.
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
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Willy Wouters
ypOne Publishing
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Re: [Asterisk-Users] Strange T1 Problem
Lookie here:
This is what you have
> exten => 1234567890,2,Dial(SIP/user1|r)
But, perhaps, here's what it shouls be:
exten => 1234567890,2,Dial(SIP/user1||r)
The second argument is *timeout*.
Normally you'd have something like
Dial(Channel,time,options)
exten => 1234567890,2,Dial(SIP/user1|60|r)
But the empty time works as well. It will just ring
forever.
Cheers,
Willy
- Original Message Follows -
>
> On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
>
> > Explicitly answer the line. If that doesn't handle
> > inband audio, there is a r flag to dial. This was
> discussed very recently.
>
> This must be a different problem, because neither of those
> solutions worked.
>
>
>
> zapata.conf sends call to fixup context:
>
>
> [fixup]
>
> ; Receive call as **
> exten => _.,1,Answer
> exten => _.,2,Cut(CALLING=EXTEN,*,2)
> exten => _.,3,SetCIDNum(${CALLING})
> exten => _.,4,Cut(CALLED=EXTEN,*,3)
> exten => _.,5,Goto(default|${CALLED}|1)
>
>
> [default]
>
> exten => 1234567890,1,Answer
> exten => 1234567890,2,Dial(SIP/user1|r)
>
>
> user1's phone rings, but no ring from PSTN caller. user1
> picks up, both can talk ok.
>
>
> I have been using cvs stable branch. I will try HEAD and
> see if that fixes it as suggested by Eric.
>
> ___
> 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
Willy Wouters
ypOne Publishing
___
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Re: [Asterisk-Users] Strange T1 Problem - FIXED plus new question
Mike Machado wrote: cvs HEAD did infact fix the ringing problem. Thanks Eric! As I said, CVS STABLE also has the fix as of this afternoon. ___ 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] Strange T1 Problem - FIXED plus new question
Mike Machado wrote: cvs HEAD did infact fix the ringing problem. Thanks Eric! I have another question for all you T1 buffs out there. The T1 I am working with goes into our local phone switch (Excel switch). Currently we are using E & M Wink signaling. The problem is we cannot set callerid on the outbound side. My minimal understanding is that if we had a PRI, I could set the callerID. Unfortunately PRI is one signaling type they cannot do (not have expensive PRI card in switch). So, my question is what other signaling types CAN I set the callerID outbound? My local switch techs cannot seem to answer that question. They just always use E & M for everything. But if I can ask them to specifically try a certain signaling type (such as Feature Group D) or one of the others in the t100p supported list, I could probably get them to change the signaling type on my trunk. Do any signaling types other than PRI support passing outbound callerID? You can usually get CLI on an E&M robbed bit T1 by configuring it right. Instead of just sending you the DNIS as a string of DTMF they usually send ***. The DNIS and CLI may be swapped, and there may be less than 3 *s in the string - wonderful consistency, eh? :-\ Regards, Steve ___ 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] Strange T1 Problem - FIXED plus new question
cvs HEAD did infact fix the ringing problem. Thanks Eric! I have another question for all you T1 buffs out there. The T1 I am working with goes into our local phone switch (Excel switch). Currently we are using E & M Wink signaling. The problem is we cannot set callerid on the outbound side. My minimal understanding is that if we had a PRI, I could set the callerID. Unfortunately PRI is one signaling type they cannot do (not have expensive PRI card in switch). So, my question is what other signaling types CAN I set the callerID outbound? My local switch techs cannot seem to answer that question. They just always use E & M for everything. But if I can ask them to specifically try a certain signaling type (such as Feature Group D) or one of the others in the t100p supported list, I could probably get them to change the signaling type on my trunk. Do any signaling types other than PRI support passing outbound callerID? ___ 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] Strange T1 Problem
On Thu, 2004-04-15 at 15:26, Steven Critchfield wrote:
> Explicitly answer the line. If that doesn't handle inband audio, there
> is a r flag to dial. This was discussed very recently.
This must be a different problem, because neither of those solutions
worked.
zapata.conf sends call to fixup context:
[fixup]
; Receive call as **
exten => _.,1,Answer
exten => _.,2,Cut(CALLING=EXTEN,*,2)
exten => _.,3,SetCIDNum(${CALLING})
exten => _.,4,Cut(CALLED=EXTEN,*,3)
exten => _.,5,Goto(default|${CALLED}|1)
[default]
exten => 1234567890,1,Answer
exten => 1234567890,2,Dial(SIP/user1|r)
user1's phone rings, but no ring from PSTN caller. user1 picks up, both
can talk ok.
I have been using cvs stable branch. I will try HEAD and see if that
fixes it as suggested by Eric.
___
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Re: [Asterisk-Users] Strange T1 Problem
Fixed in CVS STABLE around 2pm CDT today. It's been fixed in CVS HEAD for a while. Mike Machado wrote: When people call into my * box over the T1 interface, they get no ring tone. It rings the SIP phone and when the SIP user picks up, both parties can hear each other ok, its just the PSTN user calling in hears no ring. What could be causing this? ___ 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] Strange T1 Problem
On Thu, 2004-04-15 at 17:18, Mike Machado wrote: > When people call into my * box over the T1 interface, they get no ring > tone. It rings the SIP phone and when the SIP user picks up, both > parties can hear each other ok, its just the PSTN user calling in hears > no ring. What could be causing this? > > I tried setting immediate to yes in zapata.conf, but that causes my DNIS > and CallerID to stop being available. > > T100P with E & M Wink start signaling, all 24 channels are inbound > channels (no channel bank or anything like that) to SIP ATAs. The ATA is > sending a 180 Ringing reply to the invite, but still no ring. Same > symptoms with different vendor ATA devices. Explicitly answer the line. If that doesn't handle inband audio, there is a r flag to dial. This was discussed very recently. -- Steven Critchfield <[EMAIL PROTECTED]> ___ 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
