Re: [asterisk-users] Voicemail Forwarding
- Original Message - Is that user trying to forward to xxx in the same context? On Fri, Dec 17, 2010 at 5:57 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: Experiencing a problem when users attempt to forward a voicemail from within VoiceMailMain(Option 8) I see the console message: Couldn't not find mailbox XXX in context default As why are running in a multi-tenant environment voicemail.conf has been separated into individual contexts. The users retrieve their email by dialing an extension which calls VoiceMailMail(x...@vmcontext) so how do I instruct Asterisk to use that context when forwarding voicemails ? Yes exactly that indeed. Though Asterisk appears to ignore which context the user is in and selects default instead. Beginning to think that it is a bug. -- Thanks, Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Alcatel digital phone's
Hi, I'm sorry if this is already asked somewhere on the list but I couldn't find it. We have an old PBX system controlled by our Telecom provider. There are analog phones but also digital alcatel phone's connected to it. These are not ip based but legacy digital phone's. Is there a way how we can connect them to our own Asterisk PBX? The old PBX is going to be removed, so it has to be a solution: Digital alacatel phone - directly connected to Asterisk. Is there some hardware gateway or something we can use? We looked at the Grandstream GXW4024 gateway for our analog phones but I'm not sure the digital one's can connect to that one as well. Kind regards, Sander Naudts -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Forwarding
--[ UxBoD ]-- wrote: - Original Message - Yes exactly that indeed. Though Asterisk appears to ignore which context the user is in and selects default instead. Beginning to think that it is a bug. I got it figured out. In your voicemail.conf, search for the option searchcontexts=yes And enable it. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Alcatel digital phone's
Sander Naudts wrote: Hi, I'm sorry if this is already asked somewhere on the list but I couldn't find it. We have an old PBX system controlled by our Telecom provider. There are analog phones but also digital alcatel phone's connected to it. These are not ip based but legacy digital phone's. Is there a way how we can connect them to our own Asterisk PBX? The old PBX is going to be removed, so it has to be a solution: Digital alacatel phone - directly connected to Asterisk. Short answer NO! What you are calling legacy digital phones are not universal, and for many years have been integrated with the host system. This is generally true for business systems from 2 lines and six stations to large systems with hundreds of phones. Is there some hardware gateway or something we can use? The only gatewaywill be your existing switch or another of the same generation. When the switch is removed, why would the phones not be? the analog phones, if they are not special, but POTS phones that could be used anywhere on a loop start line in a business or home could be reused, but you may find that you will not want to. We looked at the Grandstream GXW4024 gateway for our analog phones but I'm not sure the digital one's can connect to that one as well. No they cannot. Better plan on replacing all the Alcatel phones with IP ones. John Novack Kind regards, Sander Naudts -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target
John Reynolds wrote: Has anyone seen or heard of this? Know how to resolve to expected behavior? I appreciate any pointers. John, Without seeing any of your dial plan or any of the output from your console during the failed transfer, nobody is going to be able to help. Why don't you start by posting the relevant part of your code that does the dialing and shows up the console output during a test transfer? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Alcatel digital phone's
There is a product from Citel (the TVA) that we're currently using with Toshiba phones. I know they also support Avaya, Nortel, and Panasonic, but am not sure if they do any other brands. They more or less convert your old digital phones to SIP. They have have full compatibility information on their website... -Jon - Original Message - From: John Novack jnov...@stromberg-carlson.org To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, December 18, 2010 6:48:57 AM Subject: Re: [asterisk-users] Asterisk and Alcatel digital phone's Sander Naudts wrote: Asterisk and Alcatel digital phone's Hi, I'm sorry if this is already asked somewhere on the list but I couldn't find it. We have an old PBX system controlled by our Telecom provider. There are analog phones but also digital alcatel phone's connected to it. These are not ip based but legacy digital phone's. Is there a way how we can connect them to our own Asterisk PBX? The old PBX is going to be removed, so it has to be a solution: Digital alacatel phone - directly connected to Asterisk. Short answer NO! What you are calling legacy digital phones are not universal, and for many years have been integrated with the host system. This is generally true for business systems from 2 lines and six stations to large systems with hundreds of phones. Is there some hardware gateway or something we can use? The only gatewaywill be your existing switch or another of the same generation. When the switch is removed, why would the phones not be? the analog phones, if they are not special, but POTS phones that could be used anywhere on a loop start line in a business or home could be reused, but you may find that you will not want to. We looked at the Grandstream GXW4024 gateway for our analog phones but I'm not sure the digital one's can connect to that one as well. No they cannot. Better plan on replacing all the Alcatel phones with IP ones. John Novack Kind regards, Sander Naudts -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to install the new cdr-stats?
Hi Everyone, I am trying to install the new cdr-stats from http://www.cdr-stats.org/ for Asterisk 1.6 but it's installation instructions are all garbled. It mentions both sqlite and mysql and there are no organized documentation. Not to mention that the apache port 8000 and port 9000 are also confusing and I don't know why is it not easy enough as adding a /var/www/html/cdr-stats rather than tampering with apache settings. I have a standard install of Asterisk 1.6 with addons and I already have the asteriskcdrdb database and everything is logged fine. I would appreciate it if someone can provide their commands for the install or step-by-step instructions. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] transfer from sip to dahdi, connects caller to MOH stream and not target
On Sat, Dec 18, 2010 at 6:51 AM, Doug Lytle supp...@drdos.info wrote: John Reynolds wrote: Has anyone seen or heard of this? Know how to resolve to expected behavior? I appreciate any pointers. John, Without seeing any of your dial plan or any of the output from your console during the failed transfer, nobody is going to be able to help. Why don't you start by posting the relevant part of your code that does the dialing and shows up the console output during a test transfer? Doug Doug and Darrick, Thanks, I work on this a bit more and get back with more info. John R. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to install the new cdr-stats?
Bruce B wrote: Hi Everyone, I am trying to install the new cdr-stats from http://www.cdr-stats.org/ for Asterisk 1.6 but it's installation instructions are all garbled. It mentions Interesting, I'll have to take a peek at it when I get home. I'll let you know what I come up with, hopefully before the weekend ends. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Specifying DID for outbound calls
The host I am working with has two accounts from the same DID provider. Incoming calls work correctly and dial the appropriate extensions. This also allows incoming calls to be billed appropriately to the individual DID accounts. Outgoing calls from either extension default to the first DID, i.e. calls from either extension have the same callerID. How can an extension specify separate outgoing contexts so the correct number is associated with it, also allowing the SIP provider to recognize the difference for billing purposes, or is there a better way? In short I'm looking to associate an outgoing call from an extension with a specific number. Here's the sip.conf for both accounts as they are using IP routing, I'm assuming I do not have to perform auth based to separate the two accounts for outgoing calls: [vitel-inbound] type=friend dtmfmode=auto host=inbound18.vitelity.net context=inbound allow=all insecure=very [vitel-outbound] type=friend dtmfmode=auto host=outbound.vitelity.net context=outbound insecure=very allow=all Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Specifying DID for outbound calls
On Sat, Dec 18, 2010 at 4:03 PM, Stephen Reese rsre...@gmail.com wrote: The host I am working with has two accounts from the same DID provider. Incoming calls work correctly and dial the appropriate extensions. This also allows incoming calls to be billed appropriately to the individual DID accounts. Outgoing calls from either extension default to the first DID, i.e. calls from either extension have the same callerID. How can an extension specify separate outgoing contexts so the correct number is associated with it, also allowing the SIP provider to recognize the difference for billing purposes, or is there a better way? The outgoing caller-id is probably just the extension number, so the provider is setting it to a default (usually the main billing number). You can set what Asterisk sends as the outbound Caller-ID in the outbound context before the Dial statement. Make sure your provider will honor what you set, as many filter what you can send to only the DIDs they provide for you. Take a look here for more information on setting the caller-id in the dialplan: http://www.voip-info.org/wiki/view/Asterisk+func+callerid -Jonathan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Specifying DID for outbound calls
The outgoing caller-id is probably just the extension number, so the
provider is setting it to a default (usually the main billing number). You
can set what Asterisk sends as the outbound Caller-ID in the outbound
context before the Dial statement. Make sure your provider will honor what
you set, as many filter what you can send to only the DIDs they provide for
you.
Take a look here for more information on setting the caller-id in the
dialplan:
http://www.voip-info.org/wiki/view/Asterisk+func+callerid
-Jonathan
Thanks for the heads up, I have been setting the caller-ID but the
trouble I'm running into is specifying the which number to call out
as. How can an extension specify a different number? See below for my
current extension.conf, thanks.
[default]
exten = 201,1,Dial(SIP/201@,30)
exten = 201,n,Voicemail(2...@default)
exten = 201,n,Hangup
exten = 202,1,Dial(SIP/202,30)
exten = 202,n,Voicemail(2...@default)
exten = 202,n,Hangup
include = inbound
include = outgoing
[inbound]
exten = 3012323434,1,Goto(default,201,1)
exten = 3013232322,1,Goto(default,202,1)
[outgoing]
exten = _1NXXNXX,1,Set(CALLERID(num)=3012323434)
exten = _1NXXNXX,n,Set(CALLERID(name)=User1)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
[outgoing2]
exten = _1NXXNXX,1,Set(CALLERID(num)=3013232322)
exten = _1NXXNXX,n,Set(CALLERID(name)=User2)
exten = _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
--
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