Re: [asterisk-users] international numbers...
This is the required dial plan: 0+61|XXX. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Withnall Sent: Friday, June 22, 2007 5:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] international numbers... Using trixbox (or a custom dialplan if needed) has anyone been able to convert a number dialled like +61242110 to something like 02422110 ie (remove the +61 and replace with 0) i just dont know how to set it up, there seems to be no dialplan wildcard i can use to match +. I was thinking of something like .61XX but that still seems wrong to me. it could match other numbers. anyone had to do this in the past ? thanks. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Transit problem
Hi! Hope someone can help me. I'm trying to pass SIP traffic from one asterisk to another through a third server. Here is the desired scenario: ServerA -- SIP -- ServerB -- SIP -- ServerC When a call is placed on a ServerA local, I can see that ServerB receives the call and dials ServerC. But ServerC says: Jun 8 09:38:32 NOTICE[3269] chan_sip.c: Failed to authenticate user asterisk sip:[EMAIL PROTECTED];tag=as15c8b5e0 However, when I change the configuration between ServerA and ServerB such that: ServerA -- IAX/2 -- ServerB -- SIP -- ServerC This works just fine. If I understand correctly, ServerA only needs to authenticate to ServerB. The fact that ServerB dials ServerC when both legs are SIP seems to indicate that there is no AUTH problem between A and B. And with the 2nd scenario, it proves that there is no auth issue between B and C. Am I missing something? Has anybody got a recipe for this? I'd appreciate any info. Thanks Jug Mensenares ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] J1/INS1500 and the Redirect Number
Hello Andrew!
Thanks for taking the time to reply.
Sorry but no, it doesnt seem to show up.
Here's my dial plan:
exten = _X.,s,Answer()
exten = _X.,n,Set(Ani=${CALLERID(ani)})
exten = _X.,n,Set(Dnis=${CALLERID(dnis)})
exten = _X.,n,Set(Rdnis=${CALLERID(rdnis)})
exten = _X.,n,NoOp(ANI: ${Ani})
exten = _X.,n,NoOp(DNIS: ${dnis})
exten = _X.,n,NoOp(RDNIS: ${rdnis})
exten = _X.,n,SayDigits(${Ani})
exten = _X., Hangup()
This is what I get:
[Jan 19 14:20:14] DEBUG[24089]: chan_zap.c:7649 pri_find_empty_chan: Found
empty available channel 0/23
-- Accepting call from '090' to '4590' on channel 0/23, span 2
[Jan 19 14:20:14] DEBUG[24089]: chan_zap.c:1458 zt_enable_ec: Enabled echo
cancellation on channel 47
-- Executing [EMAIL PROTECTED]:1] Answer(Zap/47-1, ) in new stack
[Jan 19 14:20:14] DEBUG[24083]: channel.c:895 channel_find_locked: Avoiding
initial deadlock for channel '0x8b86720'
[Jan 19 14:20:14] DEBUG[7387]: pbx.c:1621 pbx_substitute_variables_helper_full:
Function result is '090'
-- Executing [EMAIL PROTECTED]:2] Set(Zap/47-1, Ani=090) in new
stack
[Jan 19 14:20:14] ERROR[7387]: func_callerid.c:91 callerid_read: Unknown
callerid data type 'dnis'.
[Jan 19 14:20:14] DEBUG[7387]: pbx.c:1621 pbx_substitute_variables_helper_full:
Function result is ''
-- Executing [EMAIL PROTECTED]:3] Set(Zap/47-1, Dnis=) in new stack
[Jan 19 14:20:14] DEBUG[7387]: pbx.c:1621 pbx_substitute_variables_helper_full:
Function result is ''
-- Executing [EMAIL PROTECTED]:4] Set(Zap/47-1, Rdnis=) in new stack
-- Executing [EMAIL PROTECTED]:5] NoOp(Zap/47-1, ANI: 090)
in new stack
-- Executing [EMAIL PROTECTED]:6] NoOp(Zap/47-1, DNIS: ) in new stack
-- Executing [EMAIL PROTECTED]:7] NoOp(Zap/47-1, RDNIS: ) in new
stack
-- Executing [EMAIL PROTECTED]:8] SayDigits(Zap/47-1, 090) in
new stack
However, we are quite positive that the redirect number is in there because on
mobile terminals, both the redirect and the ani are identified.
Anyone out there had any luck with the RDNIS before?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Andrew Joakimsen
Sent: Friday, January 19, 2007 9:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] J1/INS1500 and the Redirect Number
Is it not coming in as CallerID(RDIS)? The specifications for
the service don't seem too different from any other PRI.
On 1/16/07, Gary Mensenares [EMAIL PROTECTED] wrote:
Hi everyone!
I'm wondering if anyone on the list had the opportunity to
work with
an NTT INS1500 ISDN PRI service before.
You see, in Japan, if you receive a call that was just forwarded by
another number, the call presentation not only includes the caller
(ANI) and your number (DNIS), it will also usually include the
forwarding number (REDIRECT). Does anybody know how to
extract this field on Asterisk?
For reference, you can look at
http://www.ntt-east.co.jp/ISDN/tech/spec/espec/3-5/content_3.html.
This is the full specification of INS 1500 signalling. Any
assistance
would be very much appreciated.
Thanks again!
Sincerely,
Jug Mensenares
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[asterisk-users] J1/INS1500 and the Redirect Number
Hi everyone! I'm wondering if anyone on the list had the opportunity to work with an NTT INS1500 ISDN PRI service before. You see, in Japan, if you receive a call that was just forwarded by another number, the call presentation not only includes the caller (ANI) and your number (DNIS), it will also usually include the forwarding number (REDIRECT). Does anybody know how to extract this field on Asterisk? For reference, you can look at http://www.ntt-east.co.jp/ISDN/tech/spec/espec/3-5/content_3.html. This is the full specification of INS 1500 signalling. Any assistance would be very much appreciated. Thanks again! Sincerely, Jug Mensenares ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] HowTo Config Asterisk and SS7
Though I haven't really tried it out myself, one option I've seen would be to use the same set of A100-series cards (the same one also being used for T1/E1/J1) from Sangoma to handle the physical port and their commercial SS7 gateway software for the out-of-band signalling. However, the SS7 Gateway is not really cheap and has to be setup by Sangoma personnel themselves. The SS7 Gateway will set you back around US$5K. Anyone else have an alternative. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nitesh Divecha Sent: Tuesday, January 16, 2007 11:45 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] HowTo Config Asterisk and SS7 Hello Asterisk, Can anyone help or put some light on, how can I configure Asterisk to work with SS7? What do I need, in terms of Hardware and Software? Regards, Neel ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to detect long calls
show channels will display all calls including Duration and BridgeTo. You can check the BridgeTo column to determine if one call leg is still attached to the other. If that fails, you can also check the duration for hung calls. To automate, there are a number of approaches. I personally suggest looking into writing/deploying an Asterisk manager. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Savoy, Kevin - Williston, ND Sent: Wednesday, January 17, 2007 2:08 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] How to detect long calls We have been running an Asterisk box with 1.2.9.1 on it since August in a call center environment. We use the Asterisk box as an IVR and then pass the calls on to a Nortel Option 11C. Today we found in our long distance bill two calls that lasted a VERY long time. One was 58 hours and another was 38 DAYS!!! Nortel does not show this call being that long. Obviously the person that called in didn't hold the line for 58 days so somehow between Asterisk and MCI the call got stuck open and didn't hang up on the network. My question is two parts, part one, has anyone heard of anything like this where a call doesn't hang up properly and seems stuck in the system. Part two is there anyway to monitor in Asterisk the length of all active calls and then if a call lasts longer then, say one hour, we could send off a text message or warning. Any ideas or comments would be helpful Thanks _ Kevin Savoy Business Unit Telecom Analyst 2218 4th Ave W Williston, ND 58801 Ph: 701-774-4023 Fax: 701-774-2901 http://www.novo1.com/ http://www.novo1.com Novo 1 is a service mark of Novo 1, Inc image001.gif Description: GIF image ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
