[Asterisk-Users] Switch from NBX to Asterisk
We've been talking about moving from our NBX100 to an asterisk solution for some time now. Well, to make a long story short, the call processor on our NBX has went south. We are now forced to purchase a new call processor >$500 or move our solution to asterisk. (Which is up and running and have been tested with Grandstream Adapters.) My question: Is there anyone out there that have moved from NBX to Asterisk, and what are the differences for your users, hold, transferring calls, etc. Also, what features will we loose, if any. And do you recommend analog phones with sip adapters (Which we already have.) or making an investment in IP phones? Our setup: NBX had 4 analog lines coming into it with 14 phones. Asterisk is set up running 1.0.7 with a PRI and 100 DID numbers. We have 20 or so Handytone 286 Adapters as well. Any advice would be appreciated! Adam ___ 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] Survey: E1 prices
Not sure about E1's but we are paying $1200 per month for a PRI. Get this, for a PRI without Caller ID services its $950... Free incoming calls and free local calling. I think that is really high, but I don't think I have a whole lot of choices here in central Illinois with Consolidated Communications... Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leo Ann Boon Sent: Friday, May 27, 2005 8:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Survey: E1 prices In .SG, the basic monthly recurring cost is S$360 (roughly US$200) excluding outgoing per minute charges (at S$0.014 peak - that's one point four cents). The one time installation charge is S$2000. FYI. X - Filter wrote: > Hello List, > > I'd like to ask how much you guys pay for an E1 (30 voice lines) and > what company. You can email me personally and not the list. > > Best regards, > Eddie > > _ > FREE pop-up blocking with the new MSN Toolbar - get it now! > http://toolbar.msn.click-url.com/go/onm00200415ave/direct/01/ > > ___ > 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
[Asterisk-Users] CallerID when transferring calls.
If extension 101 calls 102 and user 102 hits # and then 103, the caller ID of 103’s phone says 102. I’ve been looking for a way to have 103’s Caller ID show the person that is being transferred not the person transferring. So if my receptionist answers the phone and transfers it to one of my techs, I want my techs phone to display the caller ID of the person who called the receptionist. Does anyone have a solution to this problem? Thanks! Adam ___ 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] CallerID when transferring calls.
I do not see the 'o' option. I'm using 1.0.7??? Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Hamill Sent: Saturday, May 28, 2005 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID when transferring calls. On Saturday 28 May 2005 23:54, Adam Vocks wrote: > So if my receptionist answers the phone and transfers it to one of my > techs, I want my techs phone to display the caller ID of the person who > called the receptionist. > > Does anyone have a solution to this problem? Hi :) > show application dial Have a look at the 'o' option - is this what you're after? gdh ___ 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] CallerID when transferring calls.
Installed the new version and, yep, that’s what I wanted! Thanks asterisk developers. Adam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Vocks Sent: Saturday, May 28, 2005 6:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] CallerID when transferring calls. I do not see the 'o' option. I'm using 1.0.7??? Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gavin Hamill Sent: Saturday, May 28, 2005 6:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] CallerID when transferring calls. On Saturday 28 May 2005 23:54, Adam Vocks wrote: > So if my receptionist answers the phone and transfers it to one of my > techs, I want my techs phone to display the caller ID of the person who > called the receptionist. > > Does anyone have a solution to this problem? Hi :) > show application dial Have a look at the 'o' option - is this what you're after? gdh ___ 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] Pass-through
In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didn’t know if this could be done with two T1 cards and asterisk… Here is a primitive sketch. If anyone has information, please share. Thank You Adam Vocks CTI ___ 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] Pass-through
We’re still using Lucent PM3’s Adam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, June 01, 2005 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Pass-through This may or may not work due to timings slips that you may experiance with the Digium Cards. Your are correct in assuming this scenaro. I did the same (pre-asterisk) with an Adtran Atlas. It is rock solid and works great. What modem access bank are you using, there has been some talk about using the PM3 as an IAX gateway. (highly vaporware at this point), The Acend unit support SIP, the Cisco suport..., etc. etc, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Vocks Sent: Wednesday, June 01, 2005 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Pass-through In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didn’t know if this could be done with two T1 cards and asterisk… Here is a primitive sketch. If anyone has information, please share. Thank You Adam Vocks CTI ___ 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] Pass-through
Would something as simple as this work? [InFromZap1] ;Context for incoming telco calls exten => 1234567890, 1, Dial(Zap/g2) ;g2 would be the second digium card connected to our Lucent PM3 with a crossover cable. Thanks Adam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Vocks Sent: Wednesday, June 01, 2005 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Pass-through We’re still using Lucent PM3’s Adam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alexander Lopez Sent: Wednesday, June 01, 2005 10:24 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Pass-through This may or may not work due to timings slips that you may experiance with the Digium Cards. Your are correct in assuming this scenaro. I did the same (pre-asterisk) with an Adtran Atlas. It is rock solid and works great. What modem access bank are you using, there has been some talk about using the PM3 as an IAX gateway. (highly vaporware at this point), The Acend unit support SIP, the Cisco suport..., etc. etc, From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Adam Vocks Sent: Wednesday, June 01, 2005 11:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Pass-through In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didn’t know if this could be done with two T1 cards and asterisk… Here is a primitive sketch. If anyone has information, please share. Thank You Adam Vocks CTI ___ 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] Pass-through
Our modem banks are currently using E&M wink. So that would not be a problem. Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of BJ Weschke Sent: Wednesday, June 01, 2005 1:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Pass-through It is likely possible. It's going to depend on getting * and your modem bank to play nice together. If your modem bank is collecting ANI or any kind of other carrier signaling info for normal operation, you might have an easier time doing E&M wink between * and the modem bank if your modem bank support thats. The setup you're looking to put together here doesn't appear to be much different than folks who have hooked up an * device to an Avaya Definity or other PBX via a PRI tie line. On 6/1/05, Adam Vocks <[EMAIL PROTECTED]> wrote: > > > > In an order to save money, I would like to use a PRI that we have going to > one of our dial-up modem banks (We are an ISP.) During business hours these > channels are idle and during our peak internet times, we are closed. Sounds > too good to be true, but I thought I would throw it out there. These are > modem calls that if they would call our modem bank number, they would be > bridged to the outbound zap channels??? And of course, if they dial our > business number we would send them to the appropriate sip channels. I > didn't know if this could be done with two T1 cards and asterisk... > > > > Here is a primitive sketch. > > > > If anyone has information, please share. > > > > Thank You > > > > Adam Vocks > > CTI > > > ___ > 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] a simple call to my girlfriend
I would be interested in a simple call to a woman. :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Goodyear Sent: Thursday, June 02, 2005 2:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] a simple call to my girlfriend "a simple call to my girlfriend" Unfortunately, the technology does not currently exist to make that possible. -Sorry, couldn't resist. ;-) ___ 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] login/logout of call queue
This is what we use. Adam exten => *450,1,AddQueueMember(CTITechnicalSupport) exten => *450,2,PlayBack(agent-loginok) exten => *450,3,Hangup exten => #450,1,RemoveQueueMember(CTITechnicalSupport) exten => #450,2,PlayBack(agent-loggedoff) exten => #450,3,Hangup From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Scott Sent: Friday, June 03, 2005 4:57 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] login/logout of call queue We are using Grandstream GXP-2000 phones with Asterisk and was wondering if anyone had a procedure or way of logging out the phone from the network or taking the ext out of a queue? The reason we want to do this is if someone was on a break or away they could remove themselves from their queue. I was thinking of some type of * keypad sequence (like *99 etc) that could be used and then programmed into the speed dial buttons. Any help would be appreciated. Thanks, Scott. ___ 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] 911 and ISDN PRI
I have used 911 with PRI with nothing else configured. Telco had to make changes to their router for DID numbers to call through. Adam From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Pukepail Sent: Tuesday, February 07, 2006 12:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 911 and ISDN PRI Does asterisk support this? I have a location that I planned to only put a PRI line, but testing 911 (I called them first), I just get a hangup. Does 911 normally work over a PRI line? Anything special I have to setup in asterisk? ___ --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] 1-800 number
Just call tech support for a large company. Your always on hold longer than 10 minutes! Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christoph Eicke Sent: Thursday, August 18, 2005 8:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 1-800 number I'm trying to solve my Nikotel problem (see previous post) where the problem is that I get a hangup after 2 minutes, therefore I need some number that doesn't cost anything and gives me some audio for a long time... On Thursday 18 August 2005 15:08, Jonathan k. Creasy wrote: > What problem are you trying to solve with this? Just stepping out on a > limb but it sounds like you are trying to swat a fly with an F-16. > > -Jonathan > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Christoph > Eicke > Sent: Wednesday, August 17, 2005 4:34 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] 1-800 number > > Hi! > > I'm searching for a 1-800 number that simply plays music for a long time > > (>3mins) and no one picks up. I've bothered the AT&T lines so far when > trying > out my SIP->PSTN connection but then always someone answered :-) > Anyone have a number? > > Christoph > ___ > 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 ___ 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 Upgrade to 1.2
Hello all,
After upgrading to asterisk 1.2 or the current CVS HEAD, asterisk doesn’t
receive all of the DID number that is calling in. I have a T100P and a freshly
installed version of Fedora.
zapata.conf and zaptel are default except for enableing e&m wink.
Any ideas? If I install version 1.0 it works fine.
Thanks!
Adam
Here is my debug info.
-- Starting simple switch on 'Zap/3-1'
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 2 on
Zap/3-1
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:1402 zt_enable_ec: Enabled echo
cancellation on channel 3
-- Executing BackGround("Zap/3-1", "demo-moreinfo")
in new stack
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:2706 zt_answer: Took Zap/3-1 off
hook
Jan 2 17:20:42 DEBUG[3674]: channel.c:774 channel_find_locked: Avoiding
initial deadlock for 'Zap/3-1'
Jan 2 17:20:42 DEBUG[3852]: channel.c:1710 ast_settimeout: Scheduling
timer at 160 sample intervals
-- Playing 'demo-moreinfo' (language 'en')
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 1 on
Zap/3-1
Jan 2 17:20:42 DEBUG[3852]: channel.c:1710 ast_settimeout: Scheduling
timer at 0 sample intervals
Jan 2 17:20:42 DEBUG[3852]: pbx.c:2252 __ast_pbx_run: Oooh, got
something to jump out with ('1')!
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 7 on
Zap/3-1
-- Invalid extension '17' in context 'default' on Zap/3-1
== CDR updated on Zap/3-1
-- Executing Playback("Zap/3-1", "invalid") in
new stack
Jan 2 17:20:42 DEBUG[3852]: channel.c:1710 ast_settimeout: Scheduling
timer at 160 sample intervals
-- Playing 'invalid' (language 'en')
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 8 on
Zap/3-1
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 2 on
Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 4 on
Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 6 on
Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 3 on
Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 9 on
Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 8 on
Zap/3-1
Jan 2 17:20:46 DEBUG[3852]: channel.c:1710 ast_settimeout: Scheduling
timer at 0 sample intervals
Jan 2 17:20:46 DEBUG[3852]: channel.c:1710 ast_settimeout: Scheduling
timer at 0 sample intervals
== Auto fallthrough, channel 'Zap/3-1' status is 'UNKNOWN'
___
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RE: [Asterisk-Users] Asterisk Upgrade to 1.2
With 1.0 I get the following output.
-- Starting simple switch on 'Zap/15-1'
Jan 2 17:57:51 DEBUG[3645]:
chan_zap.c:4059 zt_read: DTMF digit: 2 on Zap/15-1
Jan 2 17:57:51 DEBUG[3645]:
chan_zap.c:4059 zt_read: DTMF digit: 1 on Zap/15-1
Jan 2 17:57:51 DEBUG[3645]:
chan_zap.c:4059 zt_read: DTMF digit: 7 on Zap/15-1
Jan 2 17:57:51 DEBUG[3645]:
chan_zap.c:4059 zt_read: DTMF digit: 8 on Zap/15-1
Jan 2 17:57:52 DEBUG[3645]:
chan_zap.c:4059 zt_read: DTMF digit: 2 on Zap/15-1
Jan 2 17:57:52 DEBUG[3645]:
chan_zap.c:4059 zt_read: DTMF digit: 4 on Zap/15-1
Jan 2 17:57:52 DEBUG[3645]:
chan_zap.c:4059 zt_read: DTMF digit: 6 on Zap/15-1
Jan 2 17:57:52 DEBUG[3645]:
chan_zap.c:4059 zt_read: DTMF digit: 3 on Zap/15-1
Jan 2 17:57:52 DEBUG[3645]:
chan_zap.c:4059 zt_read: DTMF digit: 9 on Zap/15-1
Jan 2 17:57:52 DEBUG[3645]:
chan_zap.c:4059 zt_read: DTMF digit: 8 on Zap/15-1
Jan 2 17:57:52 DEBUG[3645]:
chan_zap.c:1231 zt_enable_ec: Enabled echo cancellation on channel 15
-- Executing AGI("Zap/15-1",
"openclose.agi") in new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/openclose.agi
-- AGI Script openclose.agi completed,
returning 0
Jan 2 17:57:53 DEBUG[3645]: pbx.c:1195
pbx_substitute_variables_helper: _expression_ is '0'
-- Executing
GotoIf("Zap/15-1", "0?11") in new stack
Jan 2 17:57:53 DEBUG[3645]: pbx.c:4826
pbx_builtin_gotoif: Not taking any branch
-- Executing NoOp("Zap/15-1",
"") in new stack
Jan 2 17:57:53 DEBUG[3645]: pbx.c:1195
pbx_substitute_variables_helper: _expression_ is '1'
-- Executing GotoIf("Zap/15-1",
"1?11") in new stack
-- Goto (InFromZap,2178246398,11)
-- Executing Goto("Zap/15-1",
"CTIClosed|s|1") in new stack
-- Goto (CTIClosed,s,1)
-- Executing
BackGround("Zap/15-1", "CTI/CTIClosed") in new stack
Jan 2 17:57:53 DEBUG[3645]: chan_zap.c:2301
zt_answer: Took Zap/15-1 off hook
Jan 2 17:57:53 DEBUG[3645]: channel.c:1128
ast_settimeout: Scheduling timer at 160 sample intervals
Playing
'CTI/CTIClosed' (language 'en')
Thanks for any help!
Adam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Vocks
Sent: Monday, January 02, 2006
5:24 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk
Upgrade to 1.2
Hello all,
After upgrading to asterisk 1.2 or the current CVS HEAD, asterisk
doesn’t receive all of the DID number that is calling in. I have a
T100P and a freshly installed version of Fedora.
zapata.conf and zaptel are default except for enableing e&m wink.
Any ideas? If I install version 1.0 it works fine.
Thanks!
Adam
Here is my debug info.
-- Starting simple switch on 'Zap/3-1'
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 2
on Zap/3-1
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:1402 zt_enable_ec: Enabled
echo cancellation on channel 3
-- Executing BackGround("Zap/3-1",
"demo-moreinfo") in new stack
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:2706 zt_answer: Took
Zap/3-1 off hook
Jan 2 17:20:42 DEBUG[3674]: channel.c:774 channel_find_locked:
Avoiding initial deadlock for 'Zap/3-1'
Jan 2 17:20:42 DEBUG[3852]: channel.c:1710 ast_settimeout:
Scheduling timer at 160 sample intervals
-- Playing 'demo-moreinfo' (language 'en')
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 1
on Zap/3-1
Jan 2 17:20:42 DEBUG[3852]: channel.c:1710 ast_settimeout:
Scheduling timer at 0 sample intervals
Jan 2 17:20:42 DEBUG[3852]: pbx.c:2252 __ast_pbx_run: Oooh, got
something to jump out with ('1')!
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 7
on Zap/3-1
-- Invalid extension '17' in context 'default' on
Zap/3-1
== CDR updated on Zap/3-1
-- Executing Playback("Zap/3-1",
"invalid") in new stack
Jan 2 17:20:42 DEBUG[3852]: channel.c:1710 ast_settimeout:
Scheduling timer at 160 sample intervals
-- Playing 'invalid' (language 'en')
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 8
on Zap/3-1
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 2
on Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 4
on Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 6
on Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 3
on Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 9
on Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 8
on Zap/3-1
Jan 2 17:20:46 DEBUG[3852]: channel.c:1710 ast_settimeout:
Scheduling timer at 0 sample intervals
Jan 2 17:20:46 DEBUG[3852]: chan
RE: [Asterisk-Users] Asterisk Upgrade to 1.2
Is it possible to adjust the timings of
how long asterisk listens before it trys to match or is there something else
that I should be looking for? It acts as though asterisk is seeing a match in
the dial plan for the very first dtmf tone.
Thanks
Adam
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Vocks
Sent: Monday, January 02, 2006
5:24 PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk
Upgrade to 1.2
Hello all,
After upgrading to asterisk 1.2 or the current CVS HEAD, asterisk
doesn’t receive all of the DID number that is calling in. I have a
T100P and a freshly installed version of Fedora.
zapata.conf and zaptel are default except for enableing e&m wink.
Any ideas? If I install version 1.0 it works fine.
Thanks!
Adam
Here is my debug info.
-- Starting simple switch on 'Zap/3-1'
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 2
on Zap/3-1
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:1402 zt_enable_ec: Enabled
echo cancellation on channel 3
-- Executing BackGround("Zap/3-1",
"demo-moreinfo") in new stack
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:2706 zt_answer: Took
Zap/3-1 off hook
Jan 2 17:20:42 DEBUG[3674]: channel.c:774 channel_find_locked:
Avoiding initial deadlock for 'Zap/3-1'
Jan 2 17:20:42 DEBUG[3852]: channel.c:1710 ast_settimeout:
Scheduling timer at 160 sample intervals
-- Playing 'demo-moreinfo' (language 'en')
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 1
on Zap/3-1
Jan 2 17:20:42 DEBUG[3852]: channel.c:1710 ast_settimeout:
Scheduling timer at 0 sample intervals
Jan 2 17:20:42 DEBUG[3852]: pbx.c:2252 __ast_pbx_run: Oooh, got
something to jump out with ('1')!
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 7
on Zap/3-1
-- Invalid extension '17' in context 'default' on
Zap/3-1
== CDR updated on Zap/3-1
-- Executing Playback("Zap/3-1",
"invalid") in new stack
Jan 2 17:20:42 DEBUG[3852]: channel.c:1710 ast_settimeout:
Scheduling timer at 160 sample intervals
-- Playing 'invalid' (language 'en')
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 8
on Zap/3-1
Jan 2 17:20:42 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 2
on Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 4
on Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 6
on Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 3
on Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 9
on Zap/3-1
Jan 2 17:20:43 DEBUG[3852]: chan_zap.c:4650 zt_read: DTMF digit: 8
on Zap/3-1
Jan 2 17:20:46 DEBUG[3852]: channel.c:1710 ast_settimeout:
Scheduling timer at 0 sample intervals
Jan 2 17:20:46 DEBUG[3852]: channel.c:1710 ast_settimeout:
Scheduling timer at 0 sample intervals
== Auto fallthrough, channel 'Zap/3-1' status is 'UNKNOWN'
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[Asterisk-Users] Predictive Dialer
Hello all, I have a client interested in GnuDialer. My question is: Is there anyone on this list who has been using GnuDialer and I was wondering if you would be willing to share your experiences with it. Thank You Adam ___ --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
[Asterisk-Users] RE: Predictive Dialer
OK, so apparently no one is using GnuDialer, is anyone out there using any other predictive dialers on asterisk? Thank you, Adam Vocks From: Adam Vocks Sent: Thursday, March 09, 2006 12:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Predictive Dialer Hello all, I have a client interested in GnuDialer. My question is: Is there anyone on this list who has been using GnuDialer and I was wondering if you would be willing to share your experiences with it. Thank You Adam ___ --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] RE: Predictive Dialer
Great! Thanks for the info. Sounds like vicidial has more "experience" at least on this list. I will inform my client. Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Friday, March 10, 2006 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: Predictive Dialer Hello, I have used GnuDialer in a test environment and it does work. There isn't much documentation out there on it but it is in production at several sites. You should go to the GnuDialer website and post on their forums for more information. http://www.gnudialer.org/ The other GPL predictive dialer for Asterisk is VICIDIAL(which I am the primary developer of) It is in production at over 100 companies around the world and installs on top of almost any existing Asterisk installation. Our company uses it for over 200 seats across 4 locations. The largest installation I know about is over 300 seats at a financial services company. There are also many installations in South and Central America and VICIDIAL is available fully translated in Spanish. http://astguiclient.sourceforge.net/vicidial.html MATT--- On 3/10/06, Vladimir Montealegre <[EMAIL PROTECTED]> wrote: > wath is the link of the vcidialer? > > Vladimir Montealegre Estailes > Bogota-Colombia > > Este Mensaje Esta Hecho 100% con Electrones Reciclados > - Original Message - > From: "Saul Diaz" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Friday, March 10, 2006 11:29 AM > Subject: Re: [Asterisk-Users] RE: Predictive Dialer > > > > Adam Vocks wrote: > > > >> OK, so apparently no one is using GnuDialer, is anyone out there using > >> any other predictive dialers on asterisk? > >> > >> > >> Thank you, > >> > >> > >> Adam Vocks > >> > >> > >> > >> > >> *From:* Adam Vocks > >> *Sent:* Thursday, March 09, 2006 12:41 PM > >> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > >> *Subject:* Predictive Dialer > >> > >> > >> Hello all, > >> > >> > >> I have a client interested in GnuDialer. My question is: Is there > >> anyone on this list who has been using GnuDialer and I was wondering if > >> you would be willing to share your experiences with it. > >> > >> > >> Thank You > >> > >> > >> Adam > >> > >> > >> > >>___ > >>--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 > >> > > I am using VCIDialer for testing purposes.. and work fine... 70 concurrent > > calls, a little heavy to install > > > > regards > > ___ > > --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 > > > > __ > Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. > Tutopia es Internet para todos. > ___ > --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 ___ --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] RE: Predictive Dialer
Matt, Without getting into a phone war... What phones or headsets or softphones do you use with your installation? Thanks Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Florell Sent: Friday, March 10, 2006 1:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] RE: Predictive Dialer Hello, I have used GnuDialer in a test environment and it does work. There isn't much documentation out there on it but it is in production at several sites. You should go to the GnuDialer website and post on their forums for more information. http://www.gnudialer.org/ The other GPL predictive dialer for Asterisk is VICIDIAL(which I am the primary developer of) It is in production at over 100 companies around the world and installs on top of almost any existing Asterisk installation. Our company uses it for over 200 seats across 4 locations. The largest installation I know about is over 300 seats at a financial services company. There are also many installations in South and Central America and VICIDIAL is available fully translated in Spanish. http://astguiclient.sourceforge.net/vicidial.html MATT--- On 3/10/06, Vladimir Montealegre <[EMAIL PROTECTED]> wrote: > wath is the link of the vcidialer? > > Vladimir Montealegre Estailes > Bogota-Colombia > > Este Mensaje Esta Hecho 100% con Electrones Reciclados > - Original Message - > From: "Saul Diaz" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > > Sent: Friday, March 10, 2006 11:29 AM > Subject: Re: [Asterisk-Users] RE: Predictive Dialer > > > > Adam Vocks wrote: > > > >> OK, so apparently no one is using GnuDialer, is anyone out there using > >> any other predictive dialers on asterisk? > >> > >> > >> Thank you, > >> > >> > >> Adam Vocks > >> > >> > >> > >> > >> *From:* Adam Vocks > >> *Sent:* Thursday, March 09, 2006 12:41 PM > >> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion' > >> *Subject:* Predictive Dialer > >> > >> > >> Hello all, > >> > >> > >> I have a client interested in GnuDialer. My question is: Is there > >> anyone on this list who has been using GnuDialer and I was wondering if > >> you would be willing to share your experiences with it. > >> > >> > >> Thank You > >> > >> > >> Adam > >> > >> > >> > >>___ > >>--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 > >> > > I am using VCIDialer for testing purposes.. and work fine... 70 concurrent > > calls, a little heavy to install > > > > regards > > ___ > > --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 > > > > __ > Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. > Tutopia es Internet para todos. > ___ > --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 ___ --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
[asterisk-users] e911 Signalling
Hi,
We've got a dedicated T1 with two trunks running into our ILECs
selective router for 911. Split out of the T1 into two MF CAMA trunks
on ILEC side.
I'm trying to use asterisks e911 signaling, but I'm having trouble with
the dial command. (== Everyone is busy/congested at this time (1:1/0/0))
I'm missing something and I'm thinking it has to do with the hookstate
of the dahdi channel.
If anyone has a similar situation and want to provide some guidance, I'd
sure appreciate it.
Thanks!
Adam
Here's my config:
DAHDI version 2.8.0.1
[root@e911 dahdi]# dahdi_hardware
pci::02:01.0 wct1xxp+ e159:0001 Digium Wildcard T100P T1/PRI
or E100P E1/PRA Board
e911*CLI> core show version
Asterisk 11.7.0 built by root @ e911 on a x86_64 running Linux on
2014-01-28 15:50:19 UTC
/etc/dahdi/system.conf
span=1,1,0,esf,b8zs
e&m=1-2
/etc/asterisk/chan_dahdi.conf
[channels]
group=1
signalling=e911
channel=>1-2
/etc/asterisk/extensions.conf
[InFromSIP]
exten => s,1,dial(DAHDI/1/${CALLERID(num)})
e911*CLI> dahdi show status
Description Alarms IRQbpviol CRC
Fra Codi Options LBO
Digium Wildcard T100P T1/PRI Card 0 OK 0 0 0
ESF B8ZS 0 db (CSU)/0-133 feet (DSX-1)
e911*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret
BlockedState Description
pseudodefaultdefault
In Service
1public default
In Service
2public default
In Service
e911*CLI> dahdi show channel 1
Channel: 1
Description:
File Descriptor: 7
Span: 1
Extension:
Dialing: no
Context: public
Caller ID:
Calling TON: 0
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: E911 (MF)
Radio: 0
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
Echo Cancellation:
128 taps
currently OFF
Wait for dialtone: 0ms
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Hookstate (FXS only): Offhook
Here's a debug from a 911 call.
[Jan 31 11:29:53] DEBUG[9876][C-0005]: pbx.c:4890
pbx_extension_helper: Launching 'Dial'
-- Executing [s@InFromSIP:1] Dial("SIP/SIP-0005",
"DAHDI/1/212001") in new stack
[Jan 31 11:29:53] DEBUG[9876][C-0005]: sig_analog.c:820
analog_available: analog_available 1
[Jan 31 11:29:53] DEBUG[9876][C-0005]: sig_analog.c:845
analog_available: Channel 1 off hook, can't use
[Jan 31 11:29:53] WARNING[9876][C-0005]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'DAHDI' (cause 17 -
User busy)
== Everyone is busy/congested at this time (1:1/0/0)
[Jan 31 11:29:53] DEBUG[9876][C-0005]: app_dial.c:3100
dial_exec_full: Exiting with DIALSTATUS=BUSY.
-- Auto fallthrough, channel 'SIP/SIP-0005' status is 'BUSY'
--
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Re: [asterisk-users] e911 Signalling
Well, good news, it was the telco side. They had their ports disabled.
I ended up having to use signalling=fgccamamf.
Adam
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Adam Vocks
Sent: Friday, January 31, 2014 11:43 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] e911 Signalling
Hi,
We've got a dedicated T1 with two trunks running into our ILECs
selective router for 911. Split out of the T1 into two MF CAMA trunks
on ILEC side.
I'm trying to use asterisks e911 signaling, but I'm having trouble with
the dial command. (== Everyone is busy/congested at this time (1:1/0/0))
I'm missing something and I'm thinking it has to do with the hookstate
of the dahdi channel.
If anyone has a similar situation and want to provide some guidance, I'd
sure appreciate it.
Thanks!
Adam
Here's my config:
DAHDI version 2.8.0.1
[root@e911 dahdi]# dahdi_hardware
pci::02:01.0 wct1xxp+ e159:0001 Digium Wildcard T100P T1/PRI
or E100P E1/PRA Board
e911*CLI> core show version
Asterisk 11.7.0 built by root @ e911 on a x86_64 running Linux on
2014-01-28 15:50:19 UTC
/etc/dahdi/system.conf
span=1,1,0,esf,b8zs
e&m=1-2
/etc/asterisk/chan_dahdi.conf
[channels]
group=1
signalling=e911
channel=>1-2
/etc/asterisk/extensions.conf
[InFromSIP]
exten => s,1,dial(DAHDI/1/${CALLERID(num)})
e911*CLI> dahdi show status
Description Alarms IRQbpviol CRC
Fra Codi Options LBO
Digium Wildcard T100P T1/PRI Card 0 OK 0 0 0
ESF B8ZS 0 db (CSU)/0-133 feet (DSX-1)
e911*CLI> dahdi show channels
Chan Extension Context Language MOH Interpret
BlockedState Description
pseudodefaultdefault
In Service
1public default
In Service
2public default
In Service
e911*CLI> dahdi show channel 1
Channel: 1
Description:
File Descriptor: 7
Span: 1
Extension:
Dialing: no
Context: public
Caller ID:
Calling TON: 0
Caller ID name:
Mailbox: none
Destroy: 0
InAlarm: 0
Signalling Type: E911 (MF)
Radio: 0
Owner:
Real:
Callwait:
Threeway:
Confno: -1
Propagated Conference: -1
Real in conference: 0
DSP: no
Busy Detection: no
TDD: no
Relax DTMF: no
Dialing/CallwaitCAS: 0/0
Default law: ulaw
Fax Handled: no
Pulse phone: no
Gains (RX/TX): 0.00/0.00
Dynamic Range Compression (RX/TX): 0.00/0.00
DND: no
Echo Cancellation:
128 taps
currently OFF
Wait for dialtone: 0ms
Actual Confinfo: Num/0, Mode/0x
Actual Confmute: No
Hookstate (FXS only): Offhook
Here's a debug from a 911 call.
[Jan 31 11:29:53] DEBUG[9876][C-0005]: pbx.c:4890
pbx_extension_helper: Launching 'Dial'
-- Executing [s@InFromSIP:1] Dial("SIP/SIP-0005",
"DAHDI/1/212001") in new stack
[Jan 31 11:29:53] DEBUG[9876][C-0005]: sig_analog.c:820
analog_available: analog_available 1
[Jan 31 11:29:53] DEBUG[9876][C-0005]: sig_analog.c:845
analog_available: Channel 1 off hook, can't use
[Jan 31 11:29:53] WARNING[9876][C-0005]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'DAHDI' (cause 17 -
User busy)
== Everyone is busy/congested at this time (1:1/0/0)
[Jan 31 11:29:53] DEBUG[9876][C-0005]: app_dial.c:3100
dial_exec_full: Exiting with DIALSTATUS=BUSY.
-- Auto fallthrough, channel 'SIP/SIP-0005' status is 'BUSY'
--
_
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