Re: [asterisk-users] Asterisk Queue's
Date: Tue, 02 Sep 2008 18:08:52 +1200 From: Paul Crane [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Queue's To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Philipp Kempgen wrote: Tobias Ahlander schrieb: From: Mark Michelson [EMAIL PROTECTED] Tobias Ahlander wrote: Yes, I have autofill set in queues.conf. I suspect that this behaviour is because the Polycom phones I use have 2 lines. Has anyone used this function with polycom phones before? Also, my agents are Dynamic, perhaps this works better with Static agents? Here's my queues.conf (with commented lines deleted for easier reading): [general] autofill = yes monitor-type = MixMonitor [sales] strategy = rrmemory wrapuptime=15 Depending on which Asterisk version you are using, there was a bug in the queue application for some 1.4 releases where the autofill option would only be set properly if it were placed inside a queue. In other words, you may want to try putting autofill=yes inside the [sales] queue in your configuration. Also, if you're using a version of Asterisk 1.2, autofill is not a valid option and you'll be stuck with the behavior you're seeing. Unfortunately this didn't help at all... Anyone else has any tips? Is there a way to limit the polycom phones to only take one call from the Queue at the same time? Asterisk version running is 1.4.13 Maybe the phones have call-waiting enabled? Does it work if you remove the second line? Philipp Kempgen Try setting the call-limit to 1 in sip.conf as well as limitonpeer to yes. - -- Paul Crane Technical Support Officer VentureVoIP Ltd John Wickliffe House 265 Princes Street Dunedin Paul, This option doesn't help me that much. When I have it enabled, I can't put a call on hold and transfer it since Asterisk rejects usage limit to 1. Philipp, I'm using Polycom phones. When I set the Calls Per Line (which I'm told is Call Waiting) I seem to be able to transfer calls etc, but I'm still noticing the same behaviour with the queues as before. Any more tricks I can try? Thanks, Best regards, Tobias ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Newbie Polycom: ACD AgentLogin display on phone
I have been coding my own IVR for ACD (aka queue) using Polycom phones using AEL2. In particular, I have coded my own AgentCallbackLogin because a) cmd AgentCallbackLogin() is buggy and will not be supported by dev anymore b) I can put in features like hotdesking and additional validation like prohibiting repeated logins and current phone already logged on by other agent and so forth. Having said that, that still leaves one feature not available which is a visible display on the Polycom phone that an agent has already logged on to the phone. I searched the mailing list up and low and there were some sketchy notes about bweschke had developed a patch which could understand the acd-login-logout of Polycom phones. However, I hope someone can answer the following questions for me. a) Is bweschke's patch available in the current version or do we have to download and install it separately? b) Does bweschke's patch only interface with the AgentLogin() command? In other words, after we enabled the acd-login-logout parameters in the Polycom config files and we pressed the key on the phone, will the phone then basically initiate an AgentLogin() command to the Asterisk server? And does the light beside the key shows red to signify that an agent has logged on successfully. c) I have coded my own Agent Login and Logout extension and it would be great if the softkey could call my own agent login and logout extension (this bit is easy) and then showing the red light if it is a successful login (hard?). Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial timeout to cell phones
It was challenging to figure this out, since a lot of the online
examples seem to work differently, depending on older versions of Asterisk.
I wanted to ring my cellphone (via SIP provider) and deskphone (via Zap)
simultaneously, but didn't want the call to end up with the cellphone
voicemail, so press 1 on my cellphone if I want to accept the call
there. I even see the original caller ID of the inbound caller on my
cellphone, since I'm out-dialing via a SIP provider.
[inbound]
exten = 211212,1,Playtones(ring) ; play fake ring so caller doesn't
wonder
exten = 211212,n,Dial(Zap/g10local/[EMAIL PROTECTED],,) ; ring FXS and
cell
; http://www.voip-info.org/wiki/index.php?page=Asterisk+Local+channels
;
[internals]
exten = 101,1,Dial(${MARKCELL},30,tM(screen)) ; play message before
connecting
; http://www.voip-info.org/wiki/view/Asterisk+tips+findme
; play message to cellphone before connecting inbound call
; http://lists.digium.com/pipermail/asterisk-dev/2005-June/013598.html
;
[macro-screen]
exten = s,1,Wait(0.5)
exten = s,n,Read(ACCEPT,inbound,1,,1,20)
exten = s,n,GotoIf($[${ACCEPT} = 1]?yes:no)
exten = s,n(yes),Background(connecting)
exten = s,n,Goto(end) ; Continue on in dialplan to bridge the call
exten = s,n(no),Set(MACRO_RESULT=CONTINUE) ; Hangup the called party and
continue on in the dialplan
exten = s,n(end),NoOp
Thanks for the detailed response, Mark. Looks like a clever trick!
I'll try this out and post results soon.
Steve
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[asterisk-users] multiple passwords for one meetme!
Hi: Can one conference room have multiple passwords for example 10 passwords for one meetme room ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie Polycom: ACD AgentLogin display on phone
I played with the Polycom login/logout function about a year ago, and it looked brilliant. I could never get it to work, but at the time I had both Polycom and Digium agree that it would be worth getting running. I ran out of time on that project, and have never re-visited it. But it would be a great feature to get working! PaulH Lee, John (Sydney) wrote: I have been coding my own IVR for ACD (aka queue) using Polycom phones using AEL2. In particular, I have coded my own AgentCallbackLogin because a) cmd AgentCallbackLogin() is buggy and will not be supported by dev anymore b) I can put in features like hotdesking and additional validation like prohibiting repeated logins and current phone already logged on by other agent and so forth. Having said that, that still leaves one feature not available which is a visible display on the Polycom phone that an agent has already logged on to the phone. I searched the mailing list up and low and there were some sketchy notes about bweschke had developed a patch which could understand the acd-login-logout of Polycom phones. However, I hope someone can answer the following questions for me. a) Is bweschke's patch available in the current version or do we have to download and install it separately? b) Does bweschke's patch only interface with the AgentLogin() command? In other words, after we enabled the acd-login-logout parameters in the Polycom config files and we pressed the key on the phone, will the phone then basically initiate an AgentLogin() command to the Asterisk server? And does the light beside the key shows red to signify that an agent has logged on successfully. c) I have coded my own Agent Login and Logout extension and it would be great if the softkey could call my own agent login and logout extension (this bit is easy) and then showing the red light if it is a successful login (hard?). Any thoughts? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallbackLogin AddQueueMember
Just out of curiosity, where do you get this AddQueueMember syntax from? Here: http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf page: 367 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MixMonitor-Saving Recorded file with AgentId.
Hi, I am using asterisk 1.4.18. I am using Queues and recording all the calls to agents by using MixMonitor. There are 4 agents. I want to save recorded files with AgentId so that I can access recorded files of specific agent. e.g Agent Id.gsm please give hint abt it. thanks Syed Nasruddin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallbackLogin AddQueueMember
Just out of curiosity, where do you get this AddQueueMember syntax from? Here: http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.c om /books/9780596510480.pdf page: 367 Oh so the VOIP Wiki is out of date! Now, where should we go to for reliable Asterisk info then? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue's
Tobias Ahlander wrote: Date: Tue, 02 Sep 2008 18:08:52 +1200 From: Paul Crane [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Queue's To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Philipp Kempgen wrote: Tobias Ahlander schrieb: From: Mark Michelson [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Tobias Ahlander wrote: Yes, I have autofill set in queues.conf. I suspect that this behaviour is because the Polycom phones I use have 2 lines. Has anyone used this function with polycom phones before? Also, my agents are Dynamic, perhaps this works better with Static agents? Here's my queues.conf (with commented lines deleted for easier reading): [general] autofill = yes monitor-type = MixMonitor [sales] strategy = rrmemory wrapuptime=15 Depending on which Asterisk version you are using, there was a bug in the queue application for some 1.4 releases where the autofill option would only be set properly if it were placed inside a queue. In other words, you may want to try putting autofill=yes inside the [sales] queue in your configuration. Also, if you're using a version of Asterisk 1.2, autofill is not a valid option and you'll be stuck with the behavior you're seeing. Unfortunately this didn't help at all... Anyone else has any tips? Is there a way to limit the polycom phones to only take one call from the Queue at the same time? Asterisk version running is 1.4.13 Maybe the phones have call-waiting enabled? Does it work if you remove the second line? Philipp Kempgen Try setting the call-limit to 1 in sip.conf as well as limitonpeer to yes. - -- Paul Crane Technical Support Officer VentureVoIP Ltd John Wickliffe House 265 Princes Street Dunedin Paul, This option doesn't help me that much. When I have it enabled, I can't put a call on hold and transfer it since Asterisk rejects usage limit to 1. Philipp, I'm using Polycom phones. When I set the Calls Per Line (which I'm told is Call Waiting) I seem to be able to transfer calls etc, but I'm still noticing the same behaviour with the queues as before. Any more tricks I can try? Have you tried ringinuse=no in the queue definition in queues.conf and call-limit=2 in sip.conf? Alex ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] multiple passwords for one meetme!
fateme fatah schrieb: Can one conference room have multiple passwords for example 10 passwords for one meetme room ? Not natively in Asterisk but you can do that in an AGI script in the dialplan before you go to Meetme(). - Read in a password. - Call your AGI script which is free to do whatever is necessary (check with a database etc.). Set a channel variable as the return value. - Depending upon that, go to Meetme() or do something else. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inefficient Codec Translation
Brent/Steve, Thanks for the answer. Point here is that asterisk already knows about first leg and the codec so shouldn't it select the best codec for second leg to match first leg. Instead asterisk is selecting first codec in order. To illustrate, if the first leg was ilbc and second leg supports both g729/ilbc, I will assume that asterisk will select ilbc but that does not seems to be the case. Jim On 8/23/08, Brent Davidson [EMAIL PROTECTED] wrote: Steve Totaro wrote: On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin [EMAIL PROTECTED] wrote: We run asterisk to handle incoming DIDs and we have observed inefficient Codec Translation. Here is the scenario [DID Vendor] --- [Asterisk ] External GW [G729] | |--- External GW [iLBC] Our DID vendor and asterisk box supports both ilbc g729. However, our external gateway termination supports either ilbc or g729 (and not both) and depending on users location, we terminate it on either gateway. Since DID and asterisk box supports both the codecs, we assumed that asterisk will appropriately select codecs depending on where we terminate the call so that no codec translation happens. However, this seems to be an incorrect assumption and we see that different codecs get selected on two legs which leads to quality drop and extra CPU cycles. May be we are doing something wrong. Pls suggest what we are doing wrong. Below is asterisk configuration. [did] type=friend host=xxx canreinvite=yes disallow=all allow=g729 allow=ilbc [gw1] type=friend host=xxx canreinvite=yes disallow=all allow=g729 [gw2] type=friend host=xxx canreinvite=yes disallow=all allow=ilbc Thanks Jim Why don't you allow=g729 only on all entries. Maybe I have misread your email but I interpret what you wrote to mean that all endpoints support g729 I may be wrong but I understood the situation as the DID supplier supports either g.729 or ilibc, but the user has 2 locations that calls are routed to. One location supports iLibc only, the other supports g.729 only. What they seem to be trying to accomplish is to get the DID - Asterisk leg to use the same codec as the Asterisk - Remote Location leg. I think the problem is going to be that the call has to be established to the Asterisk box before a destination can be selected. The DID and Asterisk Box are going to negotiate the first available common codec before doing anything else, including setting a destination. Since you can't change a codec once a call has been established you're always going to end up with calls to one of the 2 remote locations being transcoded. The only solution I could think of would be if there was some way to identify which incoming calls were going to be routed to which location and set the codec accordingly. To do that, you'd either have to have 2 different DID's or some other massively more complicated mechanism. Forcing a reinvite (Is that even possible?) would be the only other long-shot I could think of. Good luck, Brent ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue's
Hi,
I'm using pauseQueueMember and UnPauseQueueMember to resolve this
issue. Here's part of my macro:
[macro-disca]
exten=s, 1,set(CDR(userfield)=${CALLERID(num)}_${CLIENTE})
exten=s, n,set(AGNT=${CALLERID(num)})
exten=s, n,set(_TAM=${LEN(${AGNT})})
exten=s, n,set(__TAMDST=${LEN(${MACRO_EXTEN})})
exten=s, n,GotoIf($[${TAMDST} = 4]?cfimCk) ;
exten=s, n,GotoIf($[${TAM} = 3]?pauseOn) ;is agent
exten=s, n,GotoIf($[${GRAVAR} = y]?rec:disk) ;record call
exten=s, n(pauseOn),PauseQueueMember(|Agent/${AGNT}) ;pause agent
exten=s, n(rec),Macro(gravacao,saintes,${MACRO_EXTEN}) ;record calls
exten=s, n,Dial(${ARG1},60,tTg) ;g-to execute line before
exten=s, n,GotoIf($[${TAM} = 3]?pauseOff:status)
exten=s, n(pauseOff),UnPauseQueueMember(|Agent/${AGNT})
exten=s, n(status),goto(s-${DIALSTATUS},1)
exten=s, n(end),HangUp()
exten=h,1,GotoIf($[${TAM} = 3]?2:3)
exten=h,2,UnPauseQueueMember(|Agent/${AGNT})
exten=h,3,HangUp()
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Re: [asterisk-users] Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte Randal Schwartz
On Wed, Sep 3, 2008 at 1:20 AM, Michael Graves [EMAIL PROTECTED] wrote: Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte Randal Schwartz This posted a few days ago. It's pretty general but Mark is in great form. http://twit.tv/floss38 Definitely worth listening to! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AgentCallbackLogin AddQueueMember
On Wed, Sep 3, 2008 at 11:09 AM, Lee, John (Sydney) [EMAIL PROTECTED] wrote: Just out of curiosity, where do you get this AddQueueMember syntax from? Here: http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.c om /books/9780596510480.pdf page: 367 Oh so the VOIP Wiki is out of date! It's wiki, anyone can update it. Now, where should we go to for reliable Asterisk info then? asterisk-dev-mc*CLI show application AddQueueMember asterisk-dev-mc*CLI -= Info about application 'AddQueueMember' =- [Synopsis] Dynamically adds queue members [Description] AddQueueMember(queuename[|interface[|penalty[|options[|membername): Dynamically adds interface to an existing queue. If the interface is already in the queue and there exists an n+101 priority then it will then jump to this priority. Otherwise it will return an error The option string may contain zero or more of the following characters: 'j' -- jump to +101 priority when appropriate. This application sets the following channel variable upon completion: AQMSTATUSThe status of the attempt to add a queue member as a text string, one of ADDED | MEMBERALREADY | NOSUCHQUEUE Example: AddQueueMember(techsupport|SIP/3000) Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Queue's
On Wed, Sep 3, 2008 at 9:42 AM, Tobias Ahlander [EMAIL PROTECTED] wrote: Date: Tue, 02 Sep 2008 18:08:52 +1200 From: Paul Crane [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk Queue's To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1 -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Philipp Kempgen wrote: Tobias Ahlander schrieb: From: Mark Michelson [EMAIL PROTECTED] Tobias Ahlander wrote: Yes, I have autofill set in queues.conf. I suspect that this behaviour is because the Polycom phones I use have 2 lines. Has anyone used this function with polycom phones before? Also, my agents are Dynamic, perhaps this works better with Static agents? Here's my queues.conf (with commented lines deleted for easier reading): [general] autofill = yes monitor-type = MixMonitor [sales] strategy = rrmemory wrapuptime=15 Depending on which Asterisk version you are using, there was a bug in the queue application for some 1.4 releases where the autofill option would only be set properly if it were placed inside a queue. In other words, you may want to try putting autofill=yes inside the [sales] queue in your configuration. Also, if you're using a version of Asterisk 1.2, autofill is not a valid option and you'll be stuck with the behavior you're seeing. Unfortunately this didn't help at all... Anyone else has any tips? Is there a way to limit the polycom phones to only take one call from the Queue at the same time? Asterisk version running is 1.4.13 Maybe the phones have call-waiting enabled? Does it work if you remove the second line? Philipp Kempgen Try setting the call-limit to 1 in sip.conf as well as limitonpeer to yes. - -- Paul Crane Technical Support Officer VentureVoIP Ltd John Wickliffe House 265 Princes Street Dunedin Paul, This option doesn't help me that much. When I have it enabled, I can't put a call on hold and transfer it since Asterisk rejects usage limit to 1. You have to set it to any value, so that device state events are generated, so set it to 10 or 20 to have no actual limit. Regards, Atis -- Atis Lezdins, VoIP Project Manager / Developer, [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Cell Phone: +1 800 7300689 Work phone: +1 800 7502835 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Live operator as a service?
Hi All, My employer is using a hosted PBX to connect several offices home office across the US. We have a simple IVR to route callers to the right person. However, we'd like to add the option to get to a live person who would be able to route the call based upon more complex criteria. Can anyone suggest such a live operator service? Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Live operator as a service?
Are you looking to pay per call or per hour? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Wednesday, 3 September 2008 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Live operator as a service? Hi All, My employer is using a hosted PBX to connect several offices home office across the US. We have a simple IVR to route callers to the right person. However, we'd like to add the option to get to a live person who would be able to route the call based upon more complex criteria. Can anyone suggest such a live operator service? Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte Randal Schwartz
I agree! Awesome interview, well worth the time to listen to. Thanks Michael for the post! On Wed, Sep 3, 2008 at 7:28 AM, randulo [EMAIL PROTECTED] wrote: On Wed, Sep 3, 2008 at 1:20 AM, Michael Graves [EMAIL PROTECTED] wrote: Mark Spencer on TWiT's FLOSS Weekly with Leo LaPorte Randal Schwartz This posted a few days ago. It's pretty general but Mark is in great form. http://twit.tv/floss38 Definitely worth listening to! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Selectively disable echo cancellation?
When the cards hears the fax tone it should auto disable the ec.
On Sep 2, 2008, at 9:42 PM, Octavio Ruiz wrote:
On Tue, Sep 2, 2008 at 6:16 PM, Ken D'Ambrosio [EMAIL PROTECTED] wrote:
Hi, all. I have a Sangoma A104D (on-board, DSP-based echo can); I'm
currently passing through some of my in-bound calls to a legacy PBX
(which
I hope to eventually replace). That being said, until I do, I'd
like to
kill echo cancellation for the passed-through calls -- I don't want
to
mess with their fax reception.
Any idea how to do this?
Is
echocancelwhenbridged=no
inside zapata.conf what are you looking for?
If not, what I figured out is if you run
System(wan_ec_client wanpipe1 disable ${VALUE}) ;
in your dialplan logic [perhaps inside a macro called with the M()
option for Dial()] would do the trick.
Don't forget that you obtain Zap/${VALUE}-1 from ${CHANNEL} (using
some variable stripping) and to run
System(wan_ec_client wanpipe1 enable ${VALUE}) ;
at Hangup.
Regards,
--
Octavio H. Ruiz Cervera
Tel.: (+52 55) 8590-9000 Ext. 7016
Mobile: (+52 1 55) 14-087790
Mobile: (+52 1 55) 41-351242
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[asterisk-users] Asterisk Crash
Hello, folks - Just upgraded to 1.4.21.2 on FC3. Now I've got random crashing of the 'asterisk' process. I thought it was due to mpg123 and music on hold - so I disabled all MOH classes in musiconhold.conf - but still random crashing! Here's a transcript from the console. Right at the Disconnected message, the asterisk process had crashed. I've got a watchdog that automatically restarts the process, but that still means all calls were lost. Any advice on how to troubleshoot or diagnose?? Thanks! -josiah asterisk*CLI set verbose 99 Verbosity was 1 and is now 99 The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core set verbose' instead. -- Music class default requested but no musiconhold loaded. -- Executing [EMAIL PROTECTED]:1] Macro(SIP/op-1-0902f218, stdexten|213|SIP/213) in new stack -- Executing [EMAIL PROTECTED]:1] GotoIf(SIP/op-1-0902f218, 1?999|1) in new stack -- Goto (macro-stdexten,999,1) -- Executing [EMAIL PROTECTED]:1] Set(SIP/op-1-0902f218, opt=m) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/op-1-0902f218, transfer) in new stack -- SIP/op-1-0902f218 Playing 'transfer' (language 'en') -- Executing [EMAIL PROTECTED]:3] Goto(SIP/op-1-0902f218, s|dial) in new stack -- Goto (macro-stdexten,s,3) -- Executing [EMAIL PROTECTED]:3] Dial(SIP/op-1-0902f218, SIP/213|20|m) in new stack -- Called 213 -- Music class default requested but no musiconhold loaded. -- AGI Script Executing Application: (Dial) Options: (SIP/201|30) -- SIP/213-090126f8 is ringing asterisk*CLI Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0). -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] res_cepstral.so
Has anyone got res_cepstral.so to work with Asterisk 1.4.21.1? It appears to crash Asterisk on my box (kernel 2.6.26 gcc 4.1.2). Tech supports doesn't seem to have any ideas. Norman Franke Answering Service for Directors, Inc. www.myasd.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crash
What type of hardware are you using? When is the last time you upgraded Fedora? core set verbose 6 should get you anything you need. Have a look at the dmesg output. On Wed, Sep 3, 2008 at 10:14 AM, Josiah Bryan [EMAIL PROTECTED] wrote: Hello, folks - Just upgraded to 1.4.21.2 on FC3. Now I've got random crashing of the 'asterisk' process. I thought it was due to mpg123 and music on hold - so I disabled all MOH classes in musiconhold.conf - but still random crashing! Here's a transcript from the console. Right at the Disconnected message, the asterisk process had crashed. I've got a watchdog that automatically restarts the process, but that still means all calls were lost. Any advice on how to troubleshoot or diagnose?? Thanks! -josiah asterisk*CLI set verbose 99 Verbosity was 1 and is now 99 The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core set verbose' instead. -- Music class default requested but no musiconhold loaded. -- Executing [EMAIL PROTECTED]:1] Macro(SIP/op-1-0902f218, stdexten|213|SIP/213) in new stack -- Executing [EMAIL PROTECTED]:1] GotoIf(SIP/op-1-0902f218, 1?999|1) in new stack -- Goto (macro-stdexten,999,1) -- Executing [EMAIL PROTECTED]:1] Set(SIP/op-1-0902f218, opt=m) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/op-1-0902f218, transfer) in new stack -- SIP/op-1-0902f218 Playing 'transfer' (language 'en') -- Executing [EMAIL PROTECTED]:3] Goto(SIP/op-1-0902f218, s|dial) in new stack -- Goto (macro-stdexten,s,3) -- Executing [EMAIL PROTECTED]:3] Dial(SIP/op-1-0902f218, SIP/213|20|m) in new stack -- Called 213 -- Music class default requested but no musiconhold loaded. -- AGI Script Executing Application: (Dial) Options: (SIP/201|30) -- SIP/213-090126f8 is ringing asterisk*CLI Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0). -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Andrew lathama Latham Principal TuxTone Inc. http://TuxTone.com [EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crash
Hardware wise, I've got a 1.5 GHz processor with 256 MB RAM running FC3 (kernel 2.6.9-1.667). (System output of uname -a and more is below the closing.) I've got two wctdm PCI cards running 4 FXO modules each: pci::02:08.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F pci::02:0a.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F As far as FC3 - I believe last yum update was ran on 6/01 of this year - - good suggestion, I'll re-run it right now as I type this...okay, yum update running. The only dmesg output that even looks odd is: post_create: setxattr failed, rc=28 (dev=dm-0 ino=451747) post_create: setxattr failed, rc=28 (dev=dm-0 ino=451748) Other than that, the only other dmesg output since reboot (this morning 8am or so) is some selinux deny messages related to snmpd and httpd. Suggestions? Thank you for taking the time to look at all of this. Regards, -josiah Here's uname, free, and /proc/cpuinfo: [EMAIL PROTECTED] asterisk]# uname -a Linux asterisk.productiveconcepts.com 2.6.9-1.667 #1 Tue Nov 2 14:41:25 EST 2004 i686 i686 i386 GNU/Linux [EMAIL PROTECTED] asterisk]# free total used free sharedbuffers cached Mem:255652 253416 2236 0 1380 81220 -/+ buffers/cache: 170816 84836 Swap: 524280 8340 515940 [EMAIL PROTECTED] asterisk]# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 1 model name : Intel(R) Pentium(R) 4 CPU 1.50GHz stepping: 2 cpu MHz : 1483.674 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm bogomips: 2924.54 Andrew Latham wrote: What type of hardware are you using? When is the last time you upgraded Fedora? core set verbose 6 should get you anything you need. Have a look at the dmesg output. On Wed, Sep 3, 2008 at 10:14 AM, Josiah Bryan [EMAIL PROTECTED] wrote: Hello, folks - Just upgraded to 1.4.21.2 on FC3. Now I've got random crashing of the 'asterisk' process. I thought it was due to mpg123 and music on hold - so I disabled all MOH classes in musiconhold.conf - but still random crashing! Here's a transcript from the console. Right at the Disconnected message, the asterisk process had crashed. I've got a watchdog that automatically restarts the process, but that still means all calls were lost. Any advice on how to troubleshoot or diagnose?? Thanks! -josiah asterisk*CLI set verbose 99 Verbosity was 1 and is now 99 The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core set verbose' instead. -- Music class default requested but no musiconhold loaded. -- Executing [EMAIL PROTECTED]:1] Macro(SIP/op-1-0902f218, stdexten|213|SIP/213) in new stack -- Executing [EMAIL PROTECTED]:1] GotoIf(SIP/op-1-0902f218, 1?999|1) in new stack -- Goto (macro-stdexten,999,1) -- Executing [EMAIL PROTECTED]:1] Set(SIP/op-1-0902f218, opt=m) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/op-1-0902f218, transfer) in new stack -- SIP/op-1-0902f218 Playing 'transfer' (language 'en') -- Executing [EMAIL PROTECTED]:3] Goto(SIP/op-1-0902f218, s|dial) in new stack -- Goto (macro-stdexten,s,3) -- Executing [EMAIL PROTECTED]:3] Dial(SIP/op-1-0902f218, SIP/213|20|m) in new stack -- Called 213 -- Music class default requested but no musiconhold loaded. -- AGI Script Executing Application: (Dial) Options: (SIP/201|30) -- SIP/213-090126f8 is ringing asterisk*CLI Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0). -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with Call Forward
Nhadie escribió:
Your extensions.conf looks familiar, are you using trixbox?
and are you using the web interface to configure trunk and call forwards?
if you do please post the config of your outbound route and call forward.
ron
Yes, im using trixbox. I use the features codes to configure the call
forward. I mean, pressing *72 in the phone, etc...
The outbound routes and trunks works well. I dont mentioned it, but the
extensions can make/recieve external calls without any problem.
Here is the outbound routes
[outrt-001-Xtra]
include = outrt-001-Xtra-custom
exten = _00XXX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _00XXX,n,Macro(outisbusy,)
exten = _00,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _00,n,Macro(outisbusy,)
exten = _00X,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _00X,n,Macro(outisbusy,)
exten = _00XX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _00XX,n,Macro(outisbusy,)
exten = _XXX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _XXX,n,Macro(outisbusy,)
exten = _,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _,n,Macro(outisbusy,)
exten = _X,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _X,n,Macro(outisbusy,)
exten = _X,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _X,n,Macro(outisbusy,)
I configured it from the gui interface.
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Re: [asterisk-users] Asterisk Crash
I have had issues with 2.6.9 in the past but it sounds like that is not you issue. You upgraded from ___?___ to 1.4.21.2 and it crashes. If you upgraded from 1.2 did you check your dialplan to see if the commands are depreciated and you also understand that a lot has change on zaptel which is now DAHDI On Wed, Sep 3, 2008 at 11:43 AM, Josiah Bryan [EMAIL PROTECTED] wrote: Hardware wise, I've got a 1.5 GHz processor with 256 MB RAM running FC3 (kernel 2.6.9-1.667). (System output of uname -a and more is below the closing.) I've got two wctdm PCI cards running 4 FXO modules each: pci::02:08.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F pci::02:0a.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F As far as FC3 - I believe last yum update was ran on 6/01 of this year - - good suggestion, I'll re-run it right now as I type this...okay, yum update running. The only dmesg output that even looks odd is: post_create: setxattr failed, rc=28 (dev=dm-0 ino=451747) post_create: setxattr failed, rc=28 (dev=dm-0 ino=451748) Other than that, the only other dmesg output since reboot (this morning 8am or so) is some selinux deny messages related to snmpd and httpd. Suggestions? Thank you for taking the time to look at all of this. Regards, -josiah Here's uname, free, and /proc/cpuinfo: [EMAIL PROTECTED] asterisk]# uname -a Linux asterisk.productiveconcepts.com 2.6.9-1.667 #1 Tue Nov 2 14:41:25 EST 2004 i686 i686 i386 GNU/Linux [EMAIL PROTECTED] asterisk]# free total used free sharedbuffers cached Mem:255652 253416 2236 0 1380 81220 -/+ buffers/cache: 170816 84836 Swap: 524280 8340 515940 [EMAIL PROTECTED] asterisk]# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 1 model name : Intel(R) Pentium(R) 4 CPU 1.50GHz stepping: 2 cpu MHz : 1483.674 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm bogomips: 2924.54 Andrew Latham wrote: What type of hardware are you using? When is the last time you upgraded Fedora? core set verbose 6 should get you anything you need. Have a look at the dmesg output. On Wed, Sep 3, 2008 at 10:14 AM, Josiah Bryan [EMAIL PROTECTED] wrote: Hello, folks - Just upgraded to 1.4.21.2 on FC3. Now I've got random crashing of the 'asterisk' process. I thought it was due to mpg123 and music on hold - so I disabled all MOH classes in musiconhold.conf - but still random crashing! Here's a transcript from the console. Right at the Disconnected message, the asterisk process had crashed. I've got a watchdog that automatically restarts the process, but that still means all calls were lost. Any advice on how to troubleshoot or diagnose?? Thanks! -josiah asterisk*CLI set verbose 99 Verbosity was 1 and is now 99 The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core set verbose' instead. -- Music class default requested but no musiconhold loaded. -- Executing [EMAIL PROTECTED]:1] Macro(SIP/op-1-0902f218, stdexten|213|SIP/213) in new stack -- Executing [EMAIL PROTECTED]:1] GotoIf(SIP/op-1-0902f218, 1?999|1) in new stack -- Goto (macro-stdexten,999,1) -- Executing [EMAIL PROTECTED]:1] Set(SIP/op-1-0902f218, opt=m) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/op-1-0902f218, transfer) in new stack -- SIP/op-1-0902f218 Playing 'transfer' (language 'en') -- Executing [EMAIL PROTECTED]:3] Goto(SIP/op-1-0902f218, s|dial) in new stack -- Goto (macro-stdexten,s,3) -- Executing [EMAIL PROTECTED]:3] Dial(SIP/op-1-0902f218, SIP/213|20|m) in new stack -- Called 213 -- Music class default requested but no musiconhold loaded. -- AGI Script Executing Application: (Dial) Options: (SIP/201|30) -- SIP/213-090126f8 is ringing asterisk*CLI Disconnected from Asterisk server Executing last minute cleanups Asterisk cleanly ending (0). -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224
Re: [asterisk-users] Inefficient Codec Translation
2008/9/3 Jim Boykin [EMAIL PROTECTED]: Brent/Steve, Thanks for the answer. Point here is that asterisk already knows about first leg and the codec so shouldn't it select the best codec for second leg to match first leg. Instead asterisk is selecting first codec in order. To illustrate, if the first leg was ilbc and second leg supports both g729/ilbc, I will assume that asterisk will select ilbc but that does not seems to be the case. Jim Perhaps Asterisk needs a notranscode pseudo-codec, which tries to use any existing codec that it has a handle on before then continuing through the preferred list. This would be ignored if * could not identify any existing codec, or if the existing codec was not at least listed as valid later in the list. It certainly has no such feature at present, and generally IIRC you will get the first matchable codec as listed on the caller's end. In fact the current behaviour makes perfect sense as it results in last-minute transcoding, and utilises the fact that Asterisk may know better about the capabilities of an onward link than the originator's phone! By specifying g729/ilbc, you are saying that g729 is better on that link, so Asterisk tries to be helpful - If ilbc is better, specify ilbc/g729 instead :) Just my 2p. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Live operator as a service?
On Wed, 3 Sep 2008 09:39:39 -0400, Dean Collins wrote: Are you looking to pay per call or per hour? We likely per call, with some monthly minimum, as we expect a very light call volume. Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Graves Sent: Wednesday, 3 September 2008 9:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Live operator as a service? Hi All, My employer is using a hosted PBX to connect several offices home office across the US. We have a simple IVR to route callers to the right person. However, we'd like to add the option to get to a live person who would be able to route the call based upon more complex criteria. Can anyone suggest such a live operator service? Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP TLS / Nokia E51
Hi, did anybody get SIP TLS working with E51? If I enable security in the phone's SIP config, the E51 attempts a REGISTER via 5060 UDP with method TLS, digest. My asterisk (latest SVN) just answers 401 UNAUTHORIZED. Is there some comprehensive howto for configuring SIP TLS? Terve, Stefan -- Last words of a stormchaser: Where is that rotation on the radar?! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crash
From was a CVS-HEAD version from way back pre 1.2 days, sometime in the 1.0 days (I think.) I've reviewed my dialplan based on the and UPGRADE.txt notes (and UPGRADE-1.2 notes) and I've made all the necessary changes (e.g. SetVar, etc.) - really not much was affected in the dialplan. I'm just doing the basic calls come in to receptionist, she transfers to users extensions paradigm. yum update is still doing header downloads for the upgrade transaction, and I havn't seen a kernel update come through yet - I'll keep an eye out. As far as DAHDI - didn't know that - googling turned up the digium blog on the topic, but the linked page (http://www.asterisk.org/zaptel-to-dahdi) was empty content-wise. I did use a fresh build of zaptel-1.4 (svn r4506) from http://svn.digium.com/svn/zaptel/branches/1.4 for the asterisk 1.4 upgrade. My watchdog process still is reporting frequent crashes of asterisk (most recent at 12:35 EST - they are on average an hour or less apart - some 5 or 10 minutes apart.) Suggestions for further debugging? /var/log/asterisk shows a bunch of log files - event_log is blank, messages is just warnings from the console - but NOTHING in /var/log/asterisk/messages from around the crash times (e.g. at 12:35 EST in messages there is nothing, last msg was at 12:01 and next msg was at 12:36 indicating a restart of asterisk with cdr.c: CDR simple logging enabled message.). Any way to get asterisk to tell me *why* or what app is causing the crash or termination? Thanks for your help with this mess. Cheers! -josiah Andrew Latham wrote: I have had issues with 2.6.9 in the past but it sounds like that is not you issue. You upgraded from ___?___ to 1.4.21.2 and it crashes. If you upgraded from 1.2 did you check your dialplan to see if the commands are depreciated and you also understand that a lot has change on zaptel which is now DAHDI On Wed, Sep 3, 2008 at 11:43 AM, Josiah Bryan [EMAIL PROTECTED] wrote: Hardware wise, I've got a 1.5 GHz processor with 256 MB RAM running FC3 (kernel 2.6.9-1.667). (System output of uname -a and more is below the closing.) I've got two wctdm PCI cards running 4 FXO modules each: pci::02:08.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F pci::02:0a.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F As far as FC3 - I believe last yum update was ran on 6/01 of this year - - good suggestion, I'll re-run it right now as I type this...okay, yum update running. The only dmesg output that even looks odd is: post_create: setxattr failed, rc=28 (dev=dm-0 ino=451747) post_create: setxattr failed, rc=28 (dev=dm-0 ino=451748) Other than that, the only other dmesg output since reboot (this morning 8am or so) is some selinux deny messages related to snmpd and httpd. Suggestions? Thank you for taking the time to look at all of this. Regards, -josiah Here's uname, free, and /proc/cpuinfo: [EMAIL PROTECTED] asterisk]# uname -a Linux asterisk.productiveconcepts.com 2.6.9-1.667 #1 Tue Nov 2 14:41:25 EST 2004 i686 i686 i386 GNU/Linux [EMAIL PROTECTED] asterisk]# free total used free sharedbuffers cached Mem:255652 253416 2236 0 1380 81220 -/+ buffers/cache: 170816 84836 Swap: 524280 8340 515940 [EMAIL PROTECTED] asterisk]# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 1 model name : Intel(R) Pentium(R) 4 CPU 1.50GHz stepping: 2 cpu MHz : 1483.674 cache size : 256 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 2 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm bogomips: 2924.54 Andrew Latham wrote: What type of hardware are you using? When is the last time you upgraded Fedora? core set verbose 6 should get you anything you need. Have a look at the dmesg output. On Wed, Sep 3, 2008 at 10:14 AM, Josiah Bryan [EMAIL PROTECTED] wrote: Hello, folks - Just upgraded to 1.4.21.2 on FC3. Now I've got random crashing of the 'asterisk' process. I thought it was due to mpg123 and music on hold - so I disabled all MOH classes in musiconhold.conf - but still random crashing! Here's a transcript from the console. Right at the Disconnected message, the asterisk process had crashed. I've got a watchdog that automatically restarts the process, but that still means all calls were lost. Any advice on how to troubleshoot or diagnose?? Thanks! -josiah asterisk*CLI set verbose 99 Verbosity was 1 and is now 99 The 'set verbose' command is deprecated and will be removed in a future release. Please use 'core set verbose' instead. -- Music class default
Re: [asterisk-users] Problem with Call Forward
Hi,
I'm not sure if the list would allow me to discuss it here or should
this be on the trixbox mailing list. anyway, i will still try to help
i thought you were using their followme setup on the gui, i am not
familiar with *72, but you can verify what *72 is looking by searching
for exten = *72 on any extensions*.conf file.
also check on the console while you are dialing what are the errors. Try
setting verbosity higher so you can see what's happening.
core set verbosity 10 or higher if not much information.
regards,
nhadie
Dpto. Datos Television Costa Blanca wrote:
Nhadie escribió:
Your extensions.conf looks familiar, are you using trixbox?
and are you using the web interface to configure trunk and call forwards?
if you do please post the config of your outbound route and call forward.
ron
Yes, im using trixbox. I use the features codes to configure the call
forward. I mean, pressing *72 in the phone, etc...
The outbound routes and trunks works well. I dont mentioned it, but the
extensions can make/recieve external calls without any problem.
Here is the outbound routes
[outrt-001-Xtra]
include = outrt-001-Xtra-custom
exten = _00XXX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _00XXX,n,Macro(outisbusy,)
exten = _00,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _00,n,Macro(outisbusy,)
exten = _00X,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _00X,n,Macro(outisbusy,)
exten = _00XX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _00XX,n,Macro(outisbusy,)
exten = _XXX,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _XXX,n,Macro(outisbusy,)
exten = _,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _,n,Macro(outisbusy,)
exten = _X,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _X,n,Macro(outisbusy,)
exten = _X,1,Macro(dialout-trunk,2,${EXTEN},,)
exten = _X,n,Macro(outisbusy,)
I configured it from the gui interface.
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Re: [asterisk-users] Problem with Call Forward
Dpto. Datos Television Costa Blanca schrieb: Nhadie escribió: Your extensions.conf looks familiar, are you using trixbox? and are you using the web interface to configure trunk and call forwards? Yes, im using trixbox. Why didn't you mention that in the first place? That makes it a Trixbox problem, not an Asterisk problem. Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Congestion in Outgoing call through PRI
On Wed, Sep 3, 2008 at 10:33 AM, Richard Lyman [EMAIL PROTECTED] wrote: Octavio Ruiz wrote: On Sat, Aug 30, 2008 at 12:17 PM, Shariq Khan [EMAIL PROTECTED] wrote: The output of a CLI pri intese debug at Asterisk CLI before make a test call would be very useful, libPRI 1.4.7 is just fine. I am amazed no one else have suggested trying a different phone type like an IAX2 softphone. (if i am right, this will work) For me is complete clear that -- Zap/1-1 is circuit-busy -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/1/0) the Zap channel is the one which returns the congestion status, not the other leg (whatever the technology is). Anyway, if he try both options nobody is going to be hurt. I forgot completely mention (and carefully read their zaptel.conf configuration and see dchan=16 declared rather than hardhdlc=16 ) that probably their issue is already solved and documented just right here: http://wiki.sangoma.com/Asterisk-FAQ#hardhdlc Shariq, can you tell us your wanrouter + zaptel version? -- Octavio H. Ruiz Cervera Tel.: (+52 55) 8590-9000 Ext. 7016 Mobile: (+52 1 55) 14-087790 Mobile: (+52 1 55) 41-351242 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip to sip unplanned conference! help!!
first of all my topology is as such:Softphones-- asterisk --
sipurasoftphone with peer number 100, calls another softphone with peer number
as 200. (both has asterisk as gateway)relevant extensions.conf:
exten = _1XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten = _1XX,2,VoiceMail([EMAIL PROTECTED]) ; direct 2 voicemail box if line
is busy or unavailable
exten = _1XX,3,HangUp()
exten = _2XX,1,Dial(SIP/${EXTEN},20) ;each ring equals to 5 seconds so it will
ring 3 times
exten = _2XX,2,VoiceMail([EMAIL PROTECTED]) ; directs to voicemail box if line
is busy or unavailable
exten = _2XX,3,HangUp()
relevant sip.conf:
[200]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
[200]
type=friend
host=dynamic
secret=1234
context=spa
[EMAIL PROTECTED]
in the meantime, an incoming call comes through Sipura which is directed to:
[incoming-samer]
exten = 201,1,Answer() ; Answer inbound calls
exten = 201,2,Playback(silence/1)
exten = 201,3,Background(joyce) ; input an extension
exten = 201,4,WaitExten(8)
exten = 201,5,Dial(SIP/220,15)
exten = 201,4,Wait(8)
include = spa
exten = 201,n,Hangup()
suddenly, the first conversation between 100 and 200, hears the attendant audio
message joyce welcoming the caller(the one calling sipura in a completely
different call) and listens to the entire conversation that the incoming caller
is having..
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Re: [asterisk-users] Inefficient Codec Translation
Steve Davies wrote: In fact the current behaviour makes perfect sense as it results in last-minute transcoding, and utilises the fact that Asterisk may know better about the capabilities of an onward link than the originator's phone! By specifying g729/ilbc, you are saying that g729 is better on that link, so Asterisk tries to be helpful - If ilbc is better, specify ilbc/g729 instead :) As it is supposed to work today, when creating the outbound channels, if the format being used by the incoming channel is available on the outbound channel, that format is supposed to be brought to the top of the list (made first priority), so that we can minimize the need to transcode. If this is not happening, then please open a bug report on bugs.digium.com, with the details that are needed (Asterisk version, console log trace, 'sip debug' or packet capture, etc.). -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G722 and Asterisk 1.6
I have a Grandstream GXP1200 and eager to try this codec. I've heard good things about the quality. Anyone tried it with asterisk? I can't until 1.6 is released. Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Faxing through Zap cards
On Tue, 2 Sep 2008 11:38:17 -0500, James Sneeringer [EMAIL PROTECTED] said: On Sun, Aug 31, 2008 at 1:45 AM, C F [EMAIL PROTECTED] wrote: No, in the beginning you asked because you don't have the experience so folks like myself that do have the experience answered. It might work for you, no one knows and you THINK it will work, it's a hit and miss, stability is huge issue, thats where experience comes in. If you want something that I or the other people here just think works, then just get an ATA. If you want something we have experienced and know that it works, then get a channel bank. I'd like to draw on your experience. At one point you mentioned that the fax stability goes from perfect to anybody's guess when the call leaves the PRI card. I think I understand the underlying architecture well enough to know why this is the case. Here's the question: In an installation where there are only Analog Telco drops, can pri/channel bank reliability be achieved on analog cards by keeping fax traffic *within* a single Digium TDM card *because* of the fact that card would not be subject to the limitations of the PCI/PCX interface bus and/or underlying OS? For example 4 analog fax lines into (and out of) a single TDM800--4 telco lines to 4 FXO, 4 fax machines from 4 FXS). Do you have any practical or theoretical knowledge as to whether similar reliability to the PRI/Channel-bank setup can be achieved PROVIDED that traffic is never allowed to leave the internals of the card. Depending on how ZAP services the card, there may be exactly ZERO difference between the aforementioned setup and one involving multiple SEPARATE cards. If traffic stays within the card, where (if anywhere) does the process becomes compromised? Certainly it would be trivial to design a card that could handle fax pass-through, so the logical conclusion seems to be that NOT having done so was done to achieve a GREATER good in a mutually exclusive design trade-off. I'm sure that I (and others) would be very interested to gain a better understanding of this if you (or anyone) can speak intelligently to it. Thanks -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
On Thu, 4 Sep 2008 00:25:12 +0530, Steve Repo wrote: I have a Grandstream GXP1200 and eager to try this codec. I've heard good things about the quality. Anyone tried it with asterisk? I can't until 1.6 is released. Steve I can't speak to Asterisk but I really like that codec. It's in my Polycom IP650s. I hope yours works better than it did on the BudgeTones. There, while the codec is supported, the hardware limits the call quality. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
On Wed, 3 Sep 2008 06:53:11 Olivier [EMAIL PROTECTED] said: and power it ... Maybe, an external USB port could be used to power the board but the enclosure question remains ... You're right. At this time there's no Rhino enclosure for the Single Port Failover card, but it definitely designed to be powered by the External USB port if you mount it externally. If I ever need to mount it externally, I'll get a $5 project box at radio shack if there's not better available by then. ...in response to: 2008/9/1 Karl Fife [EMAIL PROTECTED] So this card has interesting price position, the main drawback being, IMHO, it's eating a slot, which can be a rare resource in rackable servers. You raise a very important point. This device uses a BRACKET, but not a motherboard SLOT. In other words, it hangs free in one of the chassis slots that do not have a corresponding slot on the motherboard. If you do not have a bracket slot, you could mount it externally, but you'd have to engineer a way to hold it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk voicemail message order
Hello, Anyone know if there is a way to reverse the message order for saved voicemail messages in asterisk (1.2.x)? For example, when I listen to a new message and it moves to the Old folder, the next time I retrieve messages from Old, start with the most recent message rather than having to press 6 lots of times to plough through 20 messages to get to the most recent message? (Or, an option to skip to the last message in a particular folder?) Regards, Rob ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Splitter
We recently got a fsv-4fps failover switch from failsafevoip and it seems to be pretty solid and affordable for the ability to switch 4 trunks. It switches on power failure or death of asterisk. The only drawback being that its an external enclosure, if you're short of space on the rack it might be hard finding a spot for it. Karl Fife wrote: On Wed, 3 Sep 2008 06:53:11 Olivier [EMAIL PROTECTED] said: and power it ... Maybe, an external USB port could be used to power the board but the enclosure question remains ... You're right. At this time there's no Rhino enclosure for the Single Port Failover card, but it definitely designed to be powered by the External USB port if you mount it externally. If I ever need to mount it externally, I'll get a $5 project box at radio shack if there's not better available by then. ...in response to: 2008/9/1 Karl Fife [EMAIL PROTECTED] So this card has interesting price position, the main drawback being, IMHO, it's eating a slot, which can be a rare resource in rackable servers. You raise a very important point. This device uses a BRACKET, but not a motherboard SLOT. In other words, it hangs free in one of the chassis slots that do not have a corresponding slot on the motherboard. If you do not have a bracket slot, you could mount it externally, but you'd have to engineer a way to hold it. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Congestion in Outgoing call through PRI
Octavio Ruiz wrote: On Wed, Sep 3, 2008 at 10:33 AM, Richard Lyman [EMAIL PROTECTED] wrote: Octavio Ruiz wrote: On Sat, Aug 30, 2008 at 12:17 PM, Shariq Khan [EMAIL PROTECTED] wrote: The output of a CLI pri intese debug at Asterisk CLI before make a test call would be very useful, libPRI 1.4.7 is just fine. I am amazed no one else have suggested trying a different phone type like an IAX2 softphone. (if i am right, this will work) For me is complete clear that -- Zap/1-1 is circuit-busy -- Hungup 'Zap/1-1' == Everyone is busy/congested at this time (1:0/1/0) the Zap channel is the one which returns the congestion status, not the other leg (whatever the technology is). Anyway, if he try both options nobody is going to be hurt. I forgot completely mention (and carefully read their zaptel.conf configuration and see dchan=16 declared rather than hardhdlc=16 ) that probably their issue is already solved and documented just right here: http://wiki.sangoma.com/Asterisk-FAQ#hardhdlc Shariq, can you tell us your wanrouter + zaptel version? If i remember correctly you also had a Zap/1 PROGRESS and PROCEEDING message just above this where it said 'passing to SIP/xxx'. So, that means it wasn't the Zap side that caused the drop. Please, just do the test with an IAX2 softphone. It is *only* a test! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crash
Alright, praise diety, I think I've got an idea on *what* its crashing on- I've tested the change below and Asterisk no longer crashes at that point. I'm crosing my fingers hoping that it doesn't crash anywhere else. Bottom line: Asterisk is crashing when AGI tells it to 'Dial' for an active call that the operator just transfered. Details: I've got an AGI script that routes the call to one of three receptionists based on call load for that SIP device (uses manager to show channels to get a count of how many calls each operator is handling at that moment. Got that so far? Okay. Here's the basic flow: The AGI script figures out which SIP device to send the call to. It sends an agi exec command to Asterisk to Dial $device|45. Operator answers call, does her script. Operator then presses transfer button on her phone to transfer to whomever the call is destined for. Somewhere after the time she presses transfer, asterisk seems to go back to executing the AGI script. Now, the next command after the Dial $device|45 is another 'Dial' to a backup operator (normally myself or another person in the IT department) - that way we can be sure the call gets to a person even if the operators don't answer. Anyway, as soon as the second Dial is executed to the backup SIP device (whoever is on duty at that time), asterisk crashes and burns. What it seems is that even though the call went thru AGI, AGI dialed the operator, operator xfered to dest exten - as soon as the operator let go of the call (by xfering), asterisk let the AGI continue...almost as if the operator didn't answer. Normally, in my pre-1.4 environment, the second Dial in the AGI was rarely reached - it was only reached if the first operator didn't answer - or if the call was hungup, before SIGHUP was received. But now, post-1.4, it seems to fall through so to speak as soon as the operator transfers the call. As a work around, I've just disabled the second Dial command. All the calls on my server just cleared out and I just tested it - and yes, that was indeed the problem. Any ideas why Dial seems to fall thru after the operator xfers the call or if I can even do anything about that? Thanks for your help and your time with all of this. Cheers! -josiah Josiah Bryan wrote: From was a CVS-HEAD version from way back pre 1.2 days, sometime in the 1.0 days (I think.) I've reviewed my dialplan based on the and UPGRADE.txt notes (and UPGRADE-1.2 notes) and I've made all the necessary changes (e.g. SetVar, etc.) - really not much was affected in the dialplan. I'm just doing the basic calls come in to receptionist, she transfers to users extensions paradigm. yum update is still doing header downloads for the upgrade transaction, and I havn't seen a kernel update come through yet - I'll keep an eye out. As far as DAHDI - didn't know that - googling turned up the digium blog on the topic, but the linked page (http://www.asterisk.org/zaptel-to-dahdi) was empty content-wise. I did use a fresh build of zaptel-1.4 (svn r4506) from http://svn.digium.com/svn/zaptel/branches/1.4 for the asterisk 1.4 upgrade. My watchdog process still is reporting frequent crashes of asterisk (most recent at 12:35 EST - they are on average an hour or less apart - some 5 or 10 minutes apart.) Suggestions for further debugging? /var/log/asterisk shows a bunch of log files - event_log is blank, messages is just warnings from the console - but NOTHING in /var/log/asterisk/messages from around the crash times (e.g. at 12:35 EST in messages there is nothing, last msg was at 12:01 and next msg was at 12:36 indicating a restart of asterisk with cdr.c: CDR simple logging enabled message.). Any way to get asterisk to tell me *why* or what app is causing the crash or termination? Thanks for your help with this mess. Cheers! -josiah Andrew Latham wrote: I have had issues with 2.6.9 in the past but it sounds like that is not you issue. You upgraded from ___?___ to 1.4.21.2 and it crashes. If you upgraded from 1.2 did you check your dialplan to see if the commands are depreciated and you also understand that a lot has change on zaptel which is now DAHDI On Wed, Sep 3, 2008 at 11:43 AM, Josiah Bryan [EMAIL PROTECTED] wrote: Hardware wise, I've got a 1.5 GHz processor with 256 MB RAM running FC3 (kernel 2.6.9-1.667). (System output of uname -a and more is below the closing.) I've got two wctdm PCI cards running 4 FXO modules each: pci::02:08.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F pci::02:0a.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F As far as FC3 - I believe last yum update was ran on 6/01 of this year - - good suggestion, I'll re-run it right now as I type this...okay, yum update running. The only dmesg output that even looks odd is: post_create: setxattr failed, rc=28 (dev=dm-0 ino=451747) post_create: setxattr failed, rc=28 (dev=dm-0
Re: [asterisk-users] Faxing through Zap cards
I agree with Karl, We have working an Digium TDM card and grand stream ata. On ata device we connect an old fax machine. All work fine, you can send or receive fax form old fax machine using an zaptel device. Regards, Luis Morales On Thu, Sep 4, 2008 at 2:41 PM, Karl Fife [EMAIL PROTECTED] wrote: On Tue, 2 Sep 2008 11:38:17 -0500, James Sneeringer [EMAIL PROTECTED] said: On Sun, Aug 31, 2008 at 1:45 AM, C F [EMAIL PROTECTED] wrote: No, in the beginning you asked because you don't have the experience so folks like myself that do have the experience answered. It might work for you, no one knows and you THINK it will work, it's a hit and miss, stability is huge issue, thats where experience comes in. If you want something that I or the other people here just think works, then just get an ATA. If you want something we have experienced and know that it works, then get a channel bank. I'd like to draw on your experience. At one point you mentioned that the fax stability goes from perfect to anybody's guess when the call leaves the PRI card. I think I understand the underlying architecture well enough to know why this is the case. Here's the question: In an installation where there are only Analog Telco drops, can pri/channel bank reliability be achieved on analog cards by keeping fax traffic *within* a single Digium TDM card *because* of the fact that card would not be subject to the limitations of the PCI/PCX interface bus and/or underlying OS? For example 4 analog fax lines into (and out of) a single TDM800--4 telco lines to 4 FXO, 4 fax machines from 4 FXS). Do you have any practical or theoretical knowledge as to whether similar reliability to the PRI/Channel-bank setup can be achieved PROVIDED that traffic is never allowed to leave the internals of the card. Depending on how ZAP services the card, there may be exactly ZERO difference between the aforementioned setup and one involving multiple SEPARATE cards. If traffic stays within the card, where (if anywhere) does the process becomes compromised? Certainly it would be trivial to design a card that could handle fax pass-through, so the logical conclusion seems to be that NOT having done so was done to achieve a GREATER good in a mutually exclusive design trade-off. I'm sure that I (and others) would be very interested to gain a better understanding of this if you (or anyone) can speak intelligently to it. Thanks -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Ringing on Console after a page
Hello, all - Alright, after my fun with Asterisk crashing, I'm onto my next item in my checklist of stuff-to-fix-after-upgrading. I've noticed a very troubling problem when paging over Console/dsp. (I'm not sure if this has anything to do with the Dial oddities that I experienced with the Crashing problem in my other thread or not...) The problem is that after the user dials the extension, connects, speaks their page, hangsup, ringing is heard over the paging system (as in, the tone heard when you dial a person and you hear the phone ringing - that ringing tone - I don't know the proper term for it, but you get the drift.) I've gone through the source code, trying to figure out what it could be doing - however, since this is the first time I've really looked at the source for asterisk, I really didn't know what to look for. Here's the relevant context (which is included in a general context for all users): [paging] exten = 249,1,Goto(paging,s,1) exten = s,1,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Playback(vm-goodbye) exten = s,n,Hangup Here's the console output when I dial extension 249 to page. (I dial, paging answers, I say whatever (or even just hangup immediately) - then, right after the call termination, I hear the ringing over the paging system. I have to *manually* issue then hangup command seen below to stop it from ringing - however, the oddest thing is asterisk tells me that there is no call to hangup. Its not like the console got transfered to any extension - literally no channels active while the ringing is taking place (core show channels reports 0 active channels even while the ringing is heard.) asterisk*CLI set verbose 99 Verbosity is at least 99 -- Zap/1-1 answered SIP/236-09f0ea20 asterisk*CLI set debug 99 Core debug was and is now 99 asterisk*CLI -- Executing [EMAIL PROTECTED]:1] Goto(SIP/josiah2-09f0ea20, paging|s|1) in new stack -- Goto (paging,s,1) -- Executing [EMAIL PROTECTED]:1] Playback(SIP/josiah2-09f0ea20, beep) in new stack -- SIP/josiah2-09f0ea20 Playing 'beep' (language 'en') -- Executing [EMAIL PROTECTED]:2] Dial(SIP/josiah2-09f0ea20, Console/dsp) in new stack Call placed to 'dsp' on console Auto-answered -- Called dsp -- ALSA/default answered SIP/josiah2-09f0ea20 Hangup on console == Spawn extension (paging, s, 2) exited non-zero on 'SIP/josiah2-09f0ea20' Really destroying SIP dialog '[EMAIL PROTECTED]' Method: BYE asterisk*CLI hangup No call to hangup up I'm open to any and all suggestions. Thanks for your time and patience! -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DID number
Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me some tips about how to accomplish this task? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
Michael Graves wrote: On Thu, 4 Sep 2008 00:25:12 +0530, Steve Repo wrote: I have a Grandstream GXP1200 and eager to try this codec. I've heard good things about the quality. Anyone tried it with asterisk? I can't until 1.6 is released. Steve I can't speak to Asterisk but I really like that codec. It's in my Polycom IP650s. I hope yours works better than it did on the BudgeTones. There, while the codec is supported, the hardware limits the call quality. Michael -- I can't speak for the GXP1200, but with the GXP2000s I have, you cannot tell the difference between G.711 and G.722. The only Polycom I have is an IP501 (that I bought to test, found to be solid, reliable, feature barren and awkward). Sadly this is very limited wrt its codec support. (supports even fewer than cheap chinese handsets). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crash
I know many are thinking this but why don't you use a queue with fewestcalls for the strategy? On Wed, Sep 3, 2008 at 4:04 PM, Josiah Bryan [EMAIL PROTECTED] wrote: Alright, praise diety, I think I've got an idea on *what* its crashing on- I've tested the change below and Asterisk no longer crashes at that point. I'm crosing my fingers hoping that it doesn't crash anywhere else. Bottom line: Asterisk is crashing when AGI tells it to 'Dial' for an active call that the operator just transfered. Details: I've got an AGI script that routes the call to one of three receptionists based on call load for that SIP device (uses manager to show channels to get a count of how many calls each operator is handling at that moment. Got that so far? Okay. Here's the basic flow: The AGI script figures out which SIP device to send the call to. It sends an agi exec command to Asterisk to Dial $device|45. Operator answers call, does her script. Operator then presses transfer button on her phone to transfer to whomever the call is destined for. Somewhere after the time she presses transfer, asterisk seems to go back to executing the AGI script. Now, the next command after the Dial $device|45 is another 'Dial' to a backup operator (normally myself or another person in the IT department) - that way we can be sure the call gets to a person even if the operators don't answer. Anyway, as soon as the second Dial is executed to the backup SIP device (whoever is on duty at that time), asterisk crashes and burns. What it seems is that even though the call went thru AGI, AGI dialed the operator, operator xfered to dest exten - as soon as the operator let go of the call (by xfering), asterisk let the AGI continue...almost as if the operator didn't answer. Normally, in my pre-1.4 environment, the second Dial in the AGI was rarely reached - it was only reached if the first operator didn't answer - or if the call was hungup, before SIGHUP was received. But now, post-1.4, it seems to fall through so to speak as soon as the operator transfers the call. As a work around, I've just disabled the second Dial command. All the calls on my server just cleared out and I just tested it - and yes, that was indeed the problem. Any ideas why Dial seems to fall thru after the operator xfers the call or if I can even do anything about that? Thanks for your help and your time with all of this. Cheers! -josiah Josiah Bryan wrote: From was a CVS-HEAD version from way back pre 1.2 days, sometime in the 1.0 days (I think.) I've reviewed my dialplan based on the and UPGRADE.txt notes (and UPGRADE-1.2 notes) and I've made all the necessary changes (e.g. SetVar, etc.) - really not much was affected in the dialplan. I'm just doing the basic calls come in to receptionist, she transfers to users extensions paradigm. yum update is still doing header downloads for the upgrade transaction, and I havn't seen a kernel update come through yet - I'll keep an eye out. As far as DAHDI - didn't know that - googling turned up the digium blog on the topic, but the linked page (http://www.asterisk.org/zaptel-to-dahdi) was empty content-wise. I did use a fresh build of zaptel-1.4 (svn r4506) from http://svn.digium.com/svn/zaptel/branches/1.4 for the asterisk 1.4 upgrade. My watchdog process still is reporting frequent crashes of asterisk (most recent at 12:35 EST - they are on average an hour or less apart - some 5 or 10 minutes apart.) Suggestions for further debugging? /var/log/asterisk shows a bunch of log files - event_log is blank, messages is just warnings from the console - but NOTHING in /var/log/asterisk/messages from around the crash times (e.g. at 12:35 EST in messages there is nothing, last msg was at 12:01 and next msg was at 12:36 indicating a restart of asterisk with cdr.c: CDR simple logging enabled message.). Any way to get asterisk to tell me *why* or what app is causing the crash or termination? Thanks for your help with this mess. Cheers! -josiah Andrew Latham wrote: I have had issues with 2.6.9 in the past but it sounds like that is not you issue. You upgraded from ___?___ to 1.4.21.2 and it crashes. If you upgraded from 1.2 did you check your dialplan to see if the commands are depreciated and you also understand that a lot has change on zaptel which is now DAHDI On Wed, Sep 3, 2008 at 11:43 AM, Josiah Bryan [EMAIL PROTECTED] wrote: Hardware wise, I've got a 1.5 GHz processor with 256 MB RAM running FC3 (kernel 2.6.9-1.667). (System output of uname -a and more is below the closing.) I've got two wctdm PCI cards running 4 FXO modules each: pci::02:08.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F pci::02:0a.0 wctdm+ e159:0001 Wildcard TDM400P REV E/F As far as FC3 - I believe last yum update was ran on 6/01 of this year - - good suggestion, I'll re-run it right now as I type this...okay, yum update running. The
Re: [asterisk-users] G722 and Asterisk 1.6
On Wed, 03 Sep 2008 21:19:40 +0100, Thomas Kenyon wrote: I can't speak for the GXP1200, but with the GXP2000s I have, you cannot tell the difference between G.711 and G.722. The only Polycom I have is an IP501 (that I bought to test, found to be solid, reliable, feature barren and awkward). Sadly this is very limited wrt its codec support. (supports even fewer than cheap chinese handsets). Hardware phones usually support fewer codecs than soft phones, almost all support at least G.711u/a and G.729 families. The nice thing about the IP550/650 is that they give you visual confirmation that they have negotiated a G.722 connection. The show a little HD animation on the soft button label corresponding to that line. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
michel freiha wrote: Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me some tips about how to accomplish this task? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, I have never used that provider but usually either the provider knows your switch's ip and routes the did traffic to it or you have asterisk register with the provider so that it knows where to route the calls. Once thats done you can do something like exten = XX,1,Answer exten = XX,n,Playback(file) Where the x's are the number that you see coming in from your provider. If you're routed all your dids from what looks like one number(callcentric does this) then you might need to use the sip header to route your did to the particular extension you want. You shouldn't have to bother with this if you only have one did. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk with E1 interface vs IP PBX
People, I have an Asterisk SIP server and I have to connect my voip users to the PSTN via a E1 trunk. I have two ways to do this: 1) Connect a E1 interface to my Asterisk (is it possible ???) and then connect my Asterisk to the PSTN 2) Buy a commercial IP PBX with a E1 interface, generating the SIP users again, and connect the IP PBX to the PSTN What do you recommend to me based on your experience ??? Thank you Alejandro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Ringing on Console after a page
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Josiah Bryan wrote: [paging] exten = 249,1,Goto(paging,s,1) exten = s,1,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Playback(vm-goodbye) exten = s,n,Hangup If the caller has hung up, to whom are you playing the vm-goodbye message? Also, why the Goto? [paging] exten = 249,1,Playback(beep) exten = 249,n,Dial(Console/dsp) Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFIvvVQCFu3bIiwtTARAp9XAJ0Ra8LLo2COS89loyBFgWutV5SxcgCbB6Md 54ve7snza6SLYZ1ufR4BVJY= =Y8MF -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Crash
Well, at the time I wrote the AGI, fewestcalls wasn't an option (or at least, I couldn't find it through googling or on the voip-info wiki). Since then, the script has been in production use for 3+ years and I havn't bothered to go back rework the dialplan. Sorry for the trouble though. However, it still begs the question, why does Dial seem to fall through like that after the operator transfers the call? Is that expected/designed behavior? If yes, Has that changed since the 1.0 days of asterisk? If yes, Is there a switch that can turn that off? Thanks for your patience with all these questions. Regards, -josiah Andrew Latham wrote: I know many are thinking this but why don't you use a queue with fewestcalls for the strategy? On Wed, Sep 3, 2008 at 4:04 PM, Josiah Bryan [EMAIL PROTECTED] wrote: Alright, praise diety, I think I've got an idea on *what* its crashing on- I've tested the change below and Asterisk no longer crashes at that point. I'm crosing my fingers hoping that it doesn't crash anywhere else. Bottom line: Asterisk is crashing when AGI tells it to 'Dial' for an active call that the operator just transfered. Details: I've got an AGI script that routes the call to one of three receptionists based on call load for that SIP device (uses manager to show channels to get a count of how many calls each operator is handling at that moment. Got that so far? Okay. Here's the basic flow: The AGI script figures out which SIP device to send the call to. It sends an agi exec command to Asterisk to Dial $device|45. Operator answers call, does her script. Operator then presses transfer button on her phone to transfer to whomever the call is destined for. Somewhere after the time she presses transfer, asterisk seems to go back to executing the AGI script. Now, the next command after the Dial $device|45 is another 'Dial' to a backup operator (normally myself or another person in the IT department) - that way we can be sure the call gets to a person even if the operators don't answer. Anyway, as soon as the second Dial is executed to the backup SIP device (whoever is on duty at that time), asterisk crashes and burns. What it seems is that even though the call went thru AGI, AGI dialed the operator, operator xfered to dest exten - as soon as the operator let go of the call (by xfering), asterisk let the AGI continue...almost as if the operator didn't answer. Normally, in my pre-1.4 environment, the second Dial in the AGI was rarely reached - it was only reached if the first operator didn't answer - or if the call was hungup, before SIGHUP was received. But now, post-1.4, it seems to fall through so to speak as soon as the operator transfers the call. As a work around, I've just disabled the second Dial command. All the calls on my server just cleared out and I just tested it - and yes, that was indeed the problem. Any ideas why Dial seems to fall thru after the operator xfers the call or if I can even do anything about that? Thanks for your help and your time with all of this. Cheers! -josiah Josiah Bryan wrote: From was a CVS-HEAD version from way back pre 1.2 days, sometime in the 1.0 days (I think.) I've reviewed my dialplan based on the and UPGRADE.txt notes (and UPGRADE-1.2 notes) and I've made all the necessary changes (e.g. SetVar, etc.) - really not much was affected in the dialplan. I'm just doing the basic calls come in to receptionist, she transfers to users extensions paradigm. yum update is still doing header downloads for the upgrade transaction, and I havn't seen a kernel update come through yet - I'll keep an eye out. As far as DAHDI - didn't know that - googling turned up the digium blog on the topic, but the linked page (http://www.asterisk.org/zaptel-to-dahdi) was empty content-wise. I did use a fresh build of zaptel-1.4 (svn r4506) from http://svn.digium.com/svn/zaptel/branches/1.4 for the asterisk 1.4 upgrade. My watchdog process still is reporting frequent crashes of asterisk (most recent at 12:35 EST - they are on average an hour or less apart - some 5 or 10 minutes apart.) Suggestions for further debugging? /var/log/asterisk shows a bunch of log files - event_log is blank, messages is just warnings from the console - but NOTHING in /var/log/asterisk/messages from around the crash times (e.g. at 12:35 EST in messages there is nothing, last msg was at 12:01 and next msg was at 12:36 indicating a restart of asterisk with cdr.c: CDR simple logging enabled message.). Any way to get asterisk to tell me *why* or what app is causing the crash or termination? Thanks for your help with this mess. Cheers! -josiah Andrew Latham wrote: I have had issues with 2.6.9 in the past but it sounds like that is not you issue. You upgraded from ___?___ to 1.4.21.2 and it crashes. If you upgraded from 1.2 did you check your dialplan to see if the commands are depreciated and you also understand
Re: [asterisk-users] Ringing on Console after a page
Good questions - the only answer to the Goto is that this was a legacy dialplan that I first wrote 3+ years ago when I first set up asterisk - and I havn't gone go back and re-work it after learning more about asterisk - it worked up till the upgrade to 1.4 and that was that. However, you're right - simpler is better anyway. I changed it to the 249,n,Dial(Console/dsp) format (as you described below) and it still plays the ringing indicator over the console after I hangup my phone. As an aside, In the 3+ years that the system has been online, users know that when they dialed 249 and heard Goodbye! right away, they weren't going to be able to page and Something was wrong. (Usually, someone had put 249 on hold or something like that.) Thats the primary reason I left the goodbye in there. Anyway, thoughts on how to debug? Thanks for your help and your suggestions. -josiah Barry L. Kline wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Josiah Bryan wrote: [paging] exten = 249,1,Goto(paging,s,1) exten = s,1,Playback(beep) exten = s,n,Dial(Console/dsp) exten = s,n,Playback(vm-goodbye) exten = s,n,Hangup If the caller has hung up, to whom are you playing the vm-goodbye message? Also, why the Goto? [paging] exten = 249,1,Playback(beep) exten = 249,n,Dial(Console/dsp) Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFIvvVQCFu3bIiwtTARAp9XAJ0Ra8LLo2COS89loyBFgWutV5SxcgCbB6Md 54ve7snza6SLYZ1ufR4BVJY= =Y8MF -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Josiah Bryan IT Manager Productive Concepts, Inc. [EMAIL PROTECTED] (765) 964-6009, ext. 224 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
Hello Air, I did what you asked for but I got the following error: extensions.conf: [stations] exten = 442033553,1,Answer exten = 442033553,n,Playback(demo-nogo) Error message: [Sep 3 20:43:02] NOTICE[14092]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '442033553' rejected because extension not found. Regards On Wed, Sep 3, 2008 at 11:36 PM, Igor Hernandez [EMAIL PROTECTED] wrote: michel freiha wrote: Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me some tips about how to accomplish this task? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, I have never used that provider but usually either the provider knows your switch's ip and routes the did traffic to it or you have asterisk register with the provider so that it knows where to route the calls. Once thats done you can do something like exten = XX,1,Answer exten = XX,n,Playback(file) Where the x's are the number that you see coming in from your provider. If you're routed all your dids from what looks like one number(callcentric does this) then you might need to use the sip header to route your did to the particular extension you want. You shouldn't have to bother with this if you only have one did. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
On Wed, Sep 3, 2008 at 11:56 PM, michel freiha [EMAIL PROTECTED] wrote: Hello Air,Hi, I created an extension like ths: [442033553] user=442033553 type=pusers secret=1234 host=dynamic context=users nat=yes when calling the DID number from an extension registered on asterisk server everything looks fine...When dilaing the number fromPSTN number I'm still getting the the below erroe: [Sep 3 20:56:00] NOTICE[18440]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '442033553' rejected because extension not found. Do you think i should define a context to receive calls from outside the asterisk server?If yes do you have any context sample definition? Regards I did what you asked for but I got the following error: extensions.conf: [stations] exten = 442033553,1,Answer exten = 442033553,n,Playback(demo-nogo) Error message: [Sep 3 20:43:02] NOTICE[14092]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '442033553' rejected because extension not found. Regards On Wed, Sep 3, 2008 at 11:36 PM, Igor Hernandez [EMAIL PROTECTED]wrote: michel freiha wrote: Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me some tips about how to accomplish this task? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, I have never used that provider but usually either the provider knows your switch's ip and routes the did traffic to it or you have asterisk register with the provider so that it knows where to route the calls. Once thats done you can do something like exten = XX,1,Answer exten = XX,n,Playback(file) Where the x's are the number that you see coming in from your provider. If you're routed all your dids from what looks like one number(callcentric does this) then you might need to use the sip header to route your did to the particular extension you want. You shouldn't have to bother with this if you only have one did. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DID number
Hey, Did you reload asterisk after changing the extensions.conf? Also, if you try it with sip set debug on the console what do you see? michel freiha wrote: Hello Air, I did what you asked for but I got the following error: extensions.conf: [stations] exten = 442033553,1,Answer exten = 442033553,n,Playback(demo-nogo) Error message: [Sep 3 20:43:02] NOTICE[14092]: chan_sip.c:14035 handle_request_invite: Call from '' to extension '442033553' rejected because extension not found. Regards On Wed, Sep 3, 2008 at 11:36 PM, Igor Hernandez [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: michel freiha wrote: Hi All, I bought a DID number from VOxbone...this number could be dialed from any PSTN line and could be forwarded to any SIP server like asterisk server...Now I need to forward this number to my asterisk server so when a customer dial this number from his GSM or Land line PSTN number the call will be forwarde to my asterisk server and I need to play a wav file for example.. Can you please give me some tips about how to accomplish this task? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hello, I have never used that provider but usually either the provider knows your switch's ip and routes the did traffic to it or you have asterisk register with the provider so that it knows where to route the calls. Once thats done you can do something like exten = XX,1,Answer exten = XX,n,Playback(file) Where the x's are the number that you see coming in from your provider. If you're routed all your dids from what looks like one number(callcentric does this) then you might need to use the sip header to route your did to the particular extension you want. You shouldn't have to bother with this if you only have one did. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com http://www.escapetel.com/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
The Asterisk development team is pleased to announce new releases of Asterisk, Asterisk-Addons, Zaptel, and for the first time, DAHDI. This is a set of coordinated releases intended to begin the transition from Zaptel to DAHDI; for the reasons why this is being done, please see: http://blogs.digium.com/2008/05/19/zaptel-project-being-renamed-to-dahdi/ The list of packages released today includes: zaptel 1.2.27 zaptel 1.4.12 dahdi-linux 2.0.0-rc3 dahdi-tools 2.0.0-rc2 dahdi-linux-complete 2.0.0-rc3+2.0.0-rc2 asterisk 1.4.22-rc3 asterisk 1.6.0-rc4 asterisk-addons 1.6.0-rc1 All of these packages are available on http://downloads.digium.com in their respective directories. Detailed information about each package release is included below. === zaptel-1.2.27 === This will be the final release of Zaptel made from the 1.2 release branch. This release includes a number of bug fixes and other improvements, most notably compatibility with the 2.6.26 and 2.6.27 kernels and a fix for wctdm driver (for TDM400P cards) that corrects a problem introduced in 1.2.26 that caused FXO ports to not properly recognize incoming calls. The change log for this release is here: http://downloads.digium.com/pub/telephony/zaptel/releases/ChangeLog-1.2.27 === zaptel-1.4.12 === This will be the final release of Zaptel made from the 1.4 release branch. This release includes a number of bugs fixes and other improvements, including the changes that are listed above for the Zaptel 1.2.27 release. In addition, there are two major changes in this release: - Support for the Digium TC400B transcoder card has been completely rewritten, and the API used by Asterisk (or any other application) to the card has been changed significantly. The result of this work is that the transcoder interface is more reliable and stable, and it supports variable-sized G.729 frames (including G.729B silence detection frames). However, that means that any Asterisk user using a TC400B who wants to upgrade to this version of Zaptel must *also* upgrade their version of Asterisk to one of the releases included in this announcement; older versions of Asterisk will not be able to use the transcoder support in this release of Zaptel. - To help users make the transition to DAHDI (and especially if they need to move back to Zaptel for some reason), this version of Zaptel contains installation steps that will *uninstall* the important parts of DAHDI if they are present on the system during the 'make install' step. This will allow a user to 'switch back' to Zaptel from DAHDI without having to manually uninstall any portions of DAHDI. The change log for this release is here: http://downloads.digium.com/pub/telephony/zaptel/releases/ChangeLog-1.4.12 === dahdi-linux-complete-2.0.0-rc3+2.0.0-rc2 === This release combines dahdi-linux-2.0.0-rc3 and dahdi-tools-2.0.0-rc2 into a single download, one-package installation process, so that users who are installing DAHDI for the first time don't have to download and install the dahdi-linux and dahdi-tools packages separately. === dahdi-linux-2.0.0-rc3 === This is the first release candidate of the DAHDI Linux kernel modules package, which replaces the kernel modules components of Zaptel. It contains all the functionality of Zaptel 1.4 plus many improvements, but also has some old (generally unsupported) functionality from Zaptel removed, including (but not limited to): - Support for Linux 2.4.x kernels - Support for devfs dynamic device filesystems - The 'torisa' and 'wcusb' drivers Information on upgrading from Zaptel to DAHDI can be found in the included UPGRADE.txt file, which can also be read here: http://svn.digium.com/view/dahdi/linux/tags/2.0.0-rc3/UPGRADE.txt?view=co The change log for this release is here: http://downloads.digium.com/pub/telephony/dahdi-linux/releases/ChangeLog-2.0.0-rc3 === dahdi-tools-2.0.0-rc2 === This is the first release candidate of the DAHDI userspace tools package, which replaces the userspace components of Zaptel. It contains all the functionality of Zaptel 1.4 plus many improvements. Information on upgrading from Zaptel to DAHDI can be found in the included UPGRADE.txt file, which can also be read here: http://svn.digium.com/view/dahdi/tools/tags/2.0.0-rc2/UPGRADE.txt?view=co The change log for this release is here: http://downloads.digium.com/pub/telephony/dahdi-tools/releases/ChangeLog-2.0.0-rc2 === asterisk-1.4.22-rc3 === This release candidate includes a large number of bug fixes and also is the first release of Asterisk 1.4 that includes support for DAHDI, the package that is replacing Zaptel. This version of Asterisk can be built against *either* Zaptel or DAHDI, but since Zaptel 1.4.12 is the last release of Zaptel 1.4, users are encouraged to transition to DAHDI as soon as they can, so that they will be able to continue to receive bug fixes and other improvements. Information on how the transition from Zaptel to DAHDI affects this Asterisk release
Re: [asterisk-users] Faxing through Zap cards
On Wed, Sep 3, 2008 at 3:11 PM, Karl Fife [EMAIL PROTECTED] wrote: On Tue, 2 Sep 2008 11:38:17 -0500, James Sneeringer [EMAIL PROTECTED] said: On Sun, Aug 31, 2008 at 1:45 AM, C F [EMAIL PROTECTED] wrote: No, in the beginning you asked because you don't have the experience so folks like myself that do have the experience answered. It might work for you, no one knows and you THINK it will work, it's a hit and miss, stability is huge issue, thats where experience comes in. If you want something that I or the other people here just think works, then just get an ATA. If you want something we have experienced and know that it works, then get a channel bank. I'd like to draw on your experience. At one point you mentioned that the fax stability goes from perfect to anybody's guess when the call leaves the PRI card. I think I understand the underlying architecture well enough to know why this is the case. Here's the question: In an installation where there are only Analog Telco drops, can pri/channel bank reliability be achieved on analog cards by keeping fax traffic *within* a single Digium TDM card *because* of the fact that card would not be subject to the limitations of the PCI/PCX interface bus and/or underlying OS? For example 4 analog fax lines into (and out of) a single TDM800--4 telco lines to 4 FXO, 4 fax machines from 4 FXS). Do you have any practical or theoretical knowledge as to whether similar reliability to the PRI/Channel-bank setup can be achieved PROVIDED that traffic is never allowed to leave the internals of the card. Depending on how ZAP services the card, there may be exactly ZERO difference between the aforementioned setup and one involving multiple SEPARATE cards. If traffic stays within the card, where (if anywhere) does the process becomes compromised? I do this with a TDM2400 card and it works fine, but I only have it in one location like that, as I don't like it. I don't like the TDM2400 card (or any other analog 2 wire zap cards), and have since started using only channel banks, and I don't like using fax machines thru asterisk if the setup is only POTS. If a customer is running only POTS then they have a line dedicated as a fax line, in which case there is usually no point in having the fax line connected to the PBX. However, if the customer has a PRI then they are in most cases using a DID coming in over the PRI for faxing, in which case terminating that fax connection out of Asterisk on an FXS port is important. I have tried with Zap FXS cards (in a separate PCI slot than the PRI card) and faxing was not stable enough. The only time I was able to predict the stability of faxing was when using a multi port T1 Zap card where one is connected to a channel bank. Certainly it would be trivial to design a card that could handle fax pass-through, so the logical conclusion seems to be that NOT having done so was done to achieve a GREATER good in a mutually exclusive design trade-off. I'm sure that I (and others) would be very interested to gain a better understanding of this if you (or anyone) can speak intelligently to it. That greater good might have been that faxing is a technology of the past :) Thanks -Karl ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Combine sip audio and video from different sources
Hello list, Is it possibe to multiplex audio and video from 2 different sip phones and transmit both during one single call? I would like to setup something similar to the Cisco Video Advantage. Lets asume the following setup: Both parties have a sip hardphone and a sip video softphone. Both, the hardphone and the softphone register them self to asterisk. If one party calls the other party then the hardphone should ring. If the other side picks up the phone, then video should be transmitted between the soft-phones and audio between the hard-phones. (I think this can be achieved with some dailplan rules, setting up the softphone to auto-pickup and disabeling the audio codecs on the video phone (we just establish 2 seperate calls one for video and one for audio)). But the more difficult thing is what to do if one side has just a sip-video phone without a stand alone hardphone. Now audio and video from the hard and softphone must be multiplexed and transmitted to the single sip-video-softphone, so the other end can talk and watch using just the soft phone. Is something like this possible? Any ideas? Thanks, Artem ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
Michael Graves wrote: On Wed, 03 Sep 2008 21:19:40 +0100, Thomas Kenyon wrote: I can't speak for the GXP1200, but with the GXP2000s I have, you cannot tell the difference between G.711 and G.722. The only Polycom I have is an IP501 (that I bought to test, found to be solid, reliable, feature barren and awkward). Sadly this is very limited wrt its codec support. (supports even fewer than cheap chinese handsets). Hardware phones usually support fewer codecs than soft phones, almost all support at least G.711u/a and G.729 families. I don't know about that, the Grandstream phones support 8 different codecs, the old chinese PA168-based phones supported 6, as do the newer AR1688-based handsets, the linksys SPA922 has 4, the SPA962 has 5, the old Snom 300 has 7 (including G.711). The IP501 has 3. I know it's a considerably better handset than say the PA168 or AR1688-based handsets, but then they are $45 and the 501 is £125. I'd also be more sold on it if it had half the features of the GXP2000 (which is only a little over half the price). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk with E1 interface vs IP PBX
Hi!!! my comments bellow: On Thu, Sep 4, 2008 at 4:07 PM, Alejandro Facultad [EMAIL PROTECTED] wrote: People, I have an Asterisk SIP server and I have to connect my voip users to the PSTN via a E1 trunk. I have two ways to do this: 1) Connect a E1 interface to my Asterisk (is it possible ???) and then connect my Asterisk to the PSTN Yes you can 2) Buy a commercial IP PBX with a E1 interface, generating the SIP users again, and connect the IP PBX to the PSTN Yes you can, take a look on asterisk appliance. What do you recommend to me based on your experience ??? Take the control your self. If you have an linux and asterisk experience take (1). Thank you Alejandro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 - Empieza por hacer lo necesario, luego lo que es posible... y de pronto estarás haciendo lo imposible Leonardo Da'Vinci - ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] All calls want to go out only on interface ZAP/g0
I have a legacy PBX that I want to slowly move off of. Below is a diagram of what I want my setup to look-like after testing. Old Mitel---24 Channels---Asterisk---PSTN | | | Ext. 3060 SIP. 2054 Cellular No matter my dial-plan logic; all calls seem to default to ZAP/g0. I can't seem to get any calls to go directly to ZAP/g2. NOTE: For testing 11# is added to the front of all calls coming from the PSTN. PSTN to Asterisk (g0) from-pstn Asterisk to LegacyPBX (g2) from-internal - -Deleted all Outbound routes. -Re-writing Zaptel to only include Port 1 Port 3 (No 'red alarms' in zttool) AMI, D4, E M and Wink - Master Timing on Port 3 (source from Port 1). -Added 'To_PSTN' on port g0. -Added 'To_LegacyPBX' on port g2. -Added New 'Catch all Route' to PSTN and to LegacyPBX (.) Test Performed: SIP to Cellular = Worked - Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7801d70, ZAP/g0/2085553870|300|) in new stack -- Called g0/2085553870 -- Zap/1-1 answered SIP/2054-b7801d70 Test Performed: SIP to 3060 = Failed SIP to 3060 seems to go out g0 then came back in from g0 -- Goto (macro-dialout-trunk,s,17) -- Executing [EMAIL PROTECTED]:17] Macro(SIP/2054-b7801d70, dialout-trunk-predial-hook|) in new stack -- Executing [EMAIL PROTECTED]:18] GotoIf(SIP/2054-b7801d70, 0?bypass|1) in new stack -- Executing [EMAIL PROTECTED]:19] GotoIf(SIP/2054-b7801d70, 0?customtrunk) in new stack -- Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7801d70, ZAP/g0/3060|300|) in new stack -- Called g0/3060 -- Starting simple switch on 'Zap/24-1' -- Zap/1-1 answered SIP/2054-b7801d70 == Unknown extension '11#3060' in context 'from-pstn' requested -- Zap/24-1 Playing 'ss-noservice' (language 'en') Added 11#3060 to both PSTN and LegacyPBX dialplan Test Performed: SIP to 3060 = Failed -Goes out g0 and comes back unknown. -- Executing [EMAIL PROTECTED]:13] Set(SIP/2054-b7802098, OUTNUM=3060) in new stack -- Executing [EMAIL PROTECTED]:14] Set(SIP/2054-b7802098, custom=ZAP/g0) in new stack -- Executing [EMAIL PROTECTED]:15] GotoIf(SIP/2054-b7802098, 1?gocall) in new stack -- Goto (macro-dialout-trunk,s,17) -- Executing [EMAIL PROTECTED]:17] Macro(SIP/2054-b7802098, dialout-trunk-predial-hook|) in new stack -- Executing [EMAIL PROTECTED]:18] GotoIf(SIP/2054-b7802098, 0?bypass|1) in new stack -- Executing [EMAIL PROTECTED]:19] GotoIf(SIP/2054-b7802098, 0?customtrunk) in new stack -- Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7802098, ZAP/g0/3060|300|) in new stack -- Called g0/3060 -- Starting simple switch on 'Zap/24-1' -- Zap/1-1 answered SIP/2054-b7802098 == Unknown extension '11#3060' in context 'from-pstn' requested -- Zap/24-1 Playing 'ss-noservice' (language 'en') -- Hungup 'Zap/1-1' NOTE: For testing 11# is added to the front of all calls comming from the PSTN. Trying a Misc. Destination Inbound route combination: Added Misc Destination 811#3060 Changed DialPLan on LegacyPBX . 11#3060 8|11#3060 8|11. 8|. 8|1NXXNXX 8|NXX Added 'inbound route' of 11#3060 - to go to 'Misc dest 811#3060' Test Performed: SIP to 3060 = Failed -- Zap/1-1 answered SIP/2054-b7801bf0 == Unknown extension '11#30603060' in context 'from-pstn' requested -- Zap/24-1 Playing 'ss-noservice' (language 'en') -- Hungup 'Zap/24-1' Added only 8|. to dial plan Test Performed: SIP to 3060 = Failed -Fast Busy -- Executing [EMAIL PROTECTED]:20] Dial(Zap/24-1, ZAP/g0/811#|300|) in new stack -- Called g0/811# What a mess! What else can I try? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All calls want to go out only on interface ZAP/g0
Slightly confused - this isn't to hard to do (I have done it quite a few times before ) The dialplan to do this should only be several lines long. Can you provide a copy of your dialplan? PaulH Mark Best wrote: I have a legacy PBX that I want to slowly move off of. Below is a diagram of what I want my setup to look-like after testing. Old Mitel---24 Channels---Asterisk---PSTN | | | Ext. 3060 SIP. 2054 Cellular No matter my dial-plan logic; all calls seem to default to ZAP/g0. I can't seem to get any calls to go directly to ZAP/g2. NOTE: For testing 11# is added to the front of all calls coming from the PSTN. PSTN to Asterisk (g0) from-pstn Asterisk to LegacyPBX (g2) from-internal - -Deleted all Outbound routes. -Re-writing Zaptel to only include Port 1 Port 3 (No 'red alarms' in zttool) AMI, D4, E M and Wink - Master Timing on Port 3 (source from Port 1). -Added 'To_PSTN' on port g0. -Added 'To_LegacyPBX' on port g2. -Added New 'Catch all Route' to PSTN and to LegacyPBX (.) *Test Performed: SIP to Cellular = Worked* - Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7801d70, ZAP/g0/2085553870|300|) in new stack -- Called g0/2085553870 -- Zap/1-1 answered SIP/2054-b7801d70 *Test Performed: SIP to 3060 = Failed* SIP to 3060 seems to go out g0 then came back in from g0 -- Goto (macro-dialout-trunk,s,17) -- Executing [EMAIL PROTECTED]:17] Macro(SIP/2054-b7801d70, dialout-trunk-predial-hook|) in new stack -- Executing [EMAIL PROTECTED]:18] GotoIf(SIP/2054-b7801d70, 0?bypass|1) in new stack -- Executing [EMAIL PROTECTED]:19] GotoIf(SIP/2054-b7801d70, 0?customtrunk) in new stack -- Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7801d70, ZAP/g0/3060|300|) in new stack -- Called g0/3060 -- Starting simple switch on 'Zap/24-1' -- Zap/1-1 answered SIP/2054-b7801d70 == Unknown extension '11#3060' in context 'from-pstn' requested -- Zap/24-1 Playing 'ss-noservice' (language 'en') Added 11#3060 to both PSTN and LegacyPBX dialplan *Test Performed: SIP to 3060 = Failed* -Goes out g0 and comes back unknown. -- Executing [EMAIL PROTECTED]:13] Set(SIP/2054-b7802098, OUTNUM=3060) in new stack -- Executing [EMAIL PROTECTED]:14] Set(SIP/2054-b7802098, custom=ZAP/g0) in new stack -- Executing [EMAIL PROTECTED]:15] GotoIf(SIP/2054-b7802098, 1?gocall) in new stack -- Goto (macro-dialout-trunk,s,17) -- Executing [EMAIL PROTECTED]:17] Macro(SIP/2054-b7802098, dialout-trunk-predial-hook|) in new stack -- Executing [EMAIL PROTECTED]:18] GotoIf(SIP/2054-b7802098, 0?bypass|1) in new stack -- Executing [EMAIL PROTECTED]:19] GotoIf(SIP/2054-b7802098, 0?customtrunk) in new stack -- Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7802098, ZAP/g0/3060|300|) in new stack -- Called g0/3060 -- Starting simple switch on 'Zap/24-1' -- Zap/1-1 answered SIP/2054-b7802098 == Unknown extension '11#3060' in context 'from-pstn' requested -- Zap/24-1 Playing 'ss-noservice' (language 'en') -- Hungup 'Zap/1-1' /NOTE: For testing 11# is added to the front of all calls comming from the PSTN./ *Trying a Misc. Destination Inbound route combination:* Added Misc Destination 811#3060 Changed DialPLan on LegacyPBX . 11#3060 8|11#3060 8|11. 8|. 8|1NXXNXX 8|NXX Added 'inbound route' of 11#3060 - to go to 'Misc dest 811#3060' *Test Performed: SIP to 3060 = Failed* -- Zap/1-1 answered SIP/2054-b7801bf0 == Unknown extension '11#30603060' in context 'from-pstn' requested -- Zap/24-1 Playing 'ss-noservice' (language 'en') -- Hungup 'Zap/24-1' Added only 8|. to dial plan *Test Performed: SIP to 3060 = Failed* -Fast Busy -- Executing [EMAIL PROTECTED]:20] Dial(Zap/24-1, ZAP/g0/811#|300|) in new stack -- Called g0/811# What a mess! What else can I try? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] All calls want to go out only on interface ZAP/g0
To provide a better example:
(this is untested hack work - as I usually provide to this list)
exten = _2XXX,1,Dial(SIP/${EXTEN})
exten = _3XXX,1,Dial(ZAP/G2/${EXTEN})
exten = _X.,1,Dial(ZAP/G0/{EXTEN})
Clean up and test as appropriate. :)
PaulH
Mark Best wrote:
I have a legacy PBX that I want to slowly move off of. Below is a
diagram of what I want my setup to look-like after testing.
Old Mitel---24 Channels---Asterisk---PSTN
| | |
Ext. 3060 SIP. 2054 Cellular
No matter my dial-plan logic; all calls seem to default to ZAP/g0. I
can't seem to get any calls to go directly to ZAP/g2.
NOTE: For testing 11# is added to the front of all calls coming from
the PSTN.
PSTN to Asterisk (g0) from-pstn
Asterisk to LegacyPBX (g2) from-internal
-
-Deleted all Outbound routes.
-Re-writing Zaptel to only include Port 1 Port 3 (No 'red alarms' in
zttool)
AMI, D4, E M and Wink - Master Timing on Port 3 (source from Port 1).
-Added 'To_PSTN' on port g0.
-Added 'To_LegacyPBX' on port g2.
-Added New 'Catch all Route' to PSTN and to LegacyPBX (.)
*Test Performed: SIP to Cellular = Worked*
- Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7801d70,
ZAP/g0/2085553870|300|) in new stack
-- Called g0/2085553870
-- Zap/1-1 answered SIP/2054-b7801d70
*Test Performed: SIP to 3060 = Failed*
SIP to 3060 seems to go out g0 then came back in from g0
-- Goto (macro-dialout-trunk,s,17)
-- Executing [EMAIL PROTECTED]:17] Macro(SIP/2054-b7801d70,
dialout-trunk-predial-hook|) in new stack
-- Executing [EMAIL PROTECTED]:18] GotoIf(SIP/2054-b7801d70, 0?bypass|1)
in new stack
-- Executing [EMAIL PROTECTED]:19] GotoIf(SIP/2054-b7801d70,
0?customtrunk) in new stack
-- Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7801d70,
ZAP/g0/3060|300|) in new stack
-- Called g0/3060
-- Starting simple switch on 'Zap/24-1'
-- Zap/1-1 answered SIP/2054-b7801d70
== Unknown extension '11#3060' in context 'from-pstn' requested
-- Zap/24-1 Playing 'ss-noservice' (language 'en')
Added 11#3060 to both PSTN and LegacyPBX dialplan
*Test Performed: SIP to 3060 = Failed*
-Goes out g0 and comes back unknown.
-- Executing [EMAIL PROTECTED]:13] Set(SIP/2054-b7802098, OUTNUM=3060) in
new stack
-- Executing [EMAIL PROTECTED]:14] Set(SIP/2054-b7802098, custom=ZAP/g0)
in new stack
-- Executing [EMAIL PROTECTED]:15] GotoIf(SIP/2054-b7802098, 1?gocall) in
new stack
-- Goto (macro-dialout-trunk,s,17)
-- Executing [EMAIL PROTECTED]:17] Macro(SIP/2054-b7802098,
dialout-trunk-predial-hook|) in new stack
-- Executing [EMAIL PROTECTED]:18] GotoIf(SIP/2054-b7802098, 0?bypass|1)
in new stack
-- Executing [EMAIL PROTECTED]:19] GotoIf(SIP/2054-b7802098,
0?customtrunk) in new stack
-- Executing [EMAIL PROTECTED]:20] Dial(SIP/2054-b7802098,
ZAP/g0/3060|300|) in new stack
-- Called g0/3060
-- Starting simple switch on 'Zap/24-1'
-- Zap/1-1 answered SIP/2054-b7802098
== Unknown extension '11#3060' in context 'from-pstn' requested
-- Zap/24-1 Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/1-1'
/NOTE: For testing 11# is added to the front of all calls comming from
the PSTN./
*Trying a Misc. Destination Inbound route combination:*
Added Misc Destination 811#3060
Changed DialPLan on LegacyPBX
.
11#3060
8|11#3060
8|11.
8|.
8|1NXXNXX
8|NXX
Added 'inbound route' of 11#3060 - to go to 'Misc dest 811#3060'
*Test Performed: SIP to 3060 = Failed*
-- Zap/1-1 answered SIP/2054-b7801bf0
== Unknown extension '11#30603060' in context 'from-pstn' requested
-- Zap/24-1 Playing 'ss-noservice' (language 'en')
-- Hungup 'Zap/24-1'
Added only 8|. to dial plan
*Test Performed: SIP to 3060 = Failed*
-Fast Busy
-- Executing [EMAIL PROTECTED]:20] Dial(Zap/24-1, ZAP/g0/811#|300|) in
new stack
-- Called g0/811#
What a mess! What else can I try?
___
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Re: [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
Great. But I'm still a little confused. Does zaptel 1.4.12 work with asterisk-1.6.0-rc4? It looks like we first upgrade to zaptel 1.4.12, and then to dahdi. We can go back to this release of zaptel if we have problems with dahdi. Or if we go back to zaptel, do we go back to 1.6.0-beta9 also? sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
On Sep 3, 2008, at 1:55 PM, Steve Repo wrote: I have a Grandstream GXP1200 and eager to try this codec. I've heard good things about the quality. Anyone tried it with asterisk? I can't until 1.6 is released. I have used G.722 with Asterisk many times. If you have more specific questions about it and Asterisk, I would be happy to try to answer them. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New Versions of Asterisk, Asterisk-addons, Zaptel, and DAHDI
On Sep 3, 2008, at 8:32 PM, sean darcy wrote: Great. But I'm still a little confused. Does zaptel 1.4.12 work with asterisk-1.6.0-rc4? No. Asterisk 1.6.0 now _only_ supports DAHDI. It looks like we first upgrade to zaptel 1.4.12, and then to dahdi. We can go back to this release of zaptel if we have problems with dahdi. Or if we go back to zaptel, do we go back to 1.6.0-beta9 also? You can upgrade directly to DAHDI. However, if you have trouble with DAHDI and need to go back to Zaptel, then I would go back to Asterisk 1.4 instead of using an old beta of 1.6.0. Many things have been fixed since 1.6.0-beta9. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail from an unknown caller
When I get a voice message from an unknown caller it will say Message from telephone number and just not say any number. I was wondering if I can manually set the caller ID in this case to be something that the Voicemail app will recognize so it will read out Message from an unknown caller ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ASTERISK supported Video phone
Hi folks, Can some one recommend a good video phone for asterisk (SIP.IAX2). I need better quality, duarability. and should support various video codec's .(Codec auto negotiation support id prefferble) thanks in advance, Tharanga ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Z-Wave or Zigbee for Office or Home automation using XML Browser enabled Screen Phone
Has anyone seen or done an XML phone application integration using Z-Wave or Zigbee (or legacy Crestron) for Office or Home automation, lighting thermostatic control, alarm systems etc? If you've seen Crestron systems you may know that they are VERY expensive, and proprietary control panels (some with color screens) and special wiring add up to many tens of thousands of dollars. Times are changing, and it's now becoming obvious that this would be MUCH easier, cheaper, more flexible, and reconfigurable etc using XML browser screen phones back to a control interface on the MESH network using technolgies like Zigbee Z-Wave (or Crestron if already present). You could even integrate a control IVR using DISA. Press 1 to ask Alison what's turned on. If you're doing thermostatic control, you could Press 2 to ask Alison what's hot or not :-) Does anyone have any experience with an integration like this? Care to share your story, or point us to your blog? Thanks -Karl p.s. Here's a possible programmable interface for programmatic control: http://www.hawkingtech.com/products/productlist.php?CatID=43FamID=119ProdID=392 Using it may be a kludge. No idea. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
