Re: [asterisk-users] Call sip:u...@domain.com?
On Mon, 20 Dec 2010 12:39:44 -0600, Kevin P. Fleming kpflem...@digium.com wrote: You've missed a very important point here: you are using a *SIP* endpoint to call a *SIP* URI. The endpoint can do that directly, and doesn't need any help from Asterisk to do it. If you wanted to be able to restrict/control such calls, you'd need to use a SIP proxy... but Asterisk is not a proxy. Asterisk is a Back-to-Back User Agent, which means whatever URI the endpoint sends to Asterisk terminates there, and Asterisk constructs an outbound URI of some form, connecting the two channels together. Thanks much Kevin. I found this article helpful to have a better understanding of what a B2BUA is compared to an SIP proxy: www.voip-info.org/wiki/view/Asterisk+SIP+not-proxy One advantage I see in using Asterisk even when the two end-points are SIP, is that I end up with a single application to handle calls between end-points (SIP, VOSP, and FXO) and provide additional features like voice-mail, etc. But I could use a good article/book to better understand my options, how Asterisk is different from the alternatives (Freeswitch, openSIPS, etc.) www.amazon.com/s/ref=nb_sb_noss?url=search-alias%3Dstripbooksfield-keywords=voip Thank you. -- _ -- 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] Call sip:u...@domain.com?
On Tuesday 21 Dec 2010, Gilles wrote: But I could use a good article/book to better understand my options, how Asterisk is different from the alternatives (Freeswitch, openSIPS, etc.) www.amazon.com/s/ref=nb_sb_noss?url=search-alias%3Dstripbooksfield-keyword s=voip The same way Ubuntu, Slackware, CentOS c. differ from each other. They are all using the Linux kernel and the X Window System under the bonnet. Well, every Free and Open Source telephony system is using Asterisk (and Linux) under the bonnet. The differences are in the user configuration tools. -- AJS -- _ -- 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] Call sip:u...@domain.com?
On 21 Dec 2010, at 14:20, A J Stiles wrote: Well, every Free and Open Source telephony system is using Asterisk (and Linux) under the bonnet. The differences are in the user configuration tools. Uh, no? S -- _ -- 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] Call sip:u...@domain.com?
On Tue, 21 Dec 2010 14:20:55 +, A J Stiles asterisk_l...@earthshod.co.uk wrote: The same way Ubuntu, Slackware, CentOS c. differ from each other. They are all using the Linux kernel and the X Window System under the bonnet. Well, every Free and Open Source telephony system is using Asterisk (and Linux) under the bonnet. The differences are in the user configuration tools. According to this article, it appears that what really makes a B2BUA different from an SIP register/proxy is that a B2BUA can manage media (voicemail, etc.) while an SIP proxy doesn't: www.tinyurl.com/Asterisk-vs-OpenSIPS -- _ -- 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] Call sip:u...@domain.com?
On Fri, 17 Dec 2010 17:54:00 +, Roger Burton West ro...@firedrake.org wrote: How would you _expect_ to be able to specify a destination server from a telephone keypad? Thanks guys for the infos. My goal was to learn how to configure Asterisk so it could call SIP URI (u...@domain) using XLite, but didn't consider the issue of regular phones, which only have a keypad. I'll read up about Freenum, ENUM/E164, SIPBroker etc. to learn how to map a SIP URI to a digit-only number. Thank you. -- _ -- 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] Call sip:u...@domain.com?
On 12/17/2010 06:25 AM, Gilles wrote: On Thu, 16 Dec 2010 11:54:31 +0100, Gillescodecompl...@free.fr wrote: Now, I'd like to be able to call any number on the Net that is advertised as sip:u...@domain.com, such as those: I mean: Do I really have to first create a section in sip.conf each time a user needs to call a number on a new SIP server? http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net You've missed a very important point here: you are using a *SIP* endpoint to call a *SIP* URI. The endpoint can do that directly, and doesn't need any help from Asterisk to do it. If you wanted to be able to restrict/control such calls, you'd need to use a SIP proxy... but Asterisk is not a proxy. Asterisk is a Back-to-Back User Agent, which means whatever URI the endpoint sends to Asterisk terminates there, and Asterisk constructs an outbound URI of some form, connecting the two channels together. You should probably take a step back and ask yourself what value Asterisk would bring being in the middle between your SIP softphones and some random SIP endpoint out on the Internet. Once you determine that, you'll know whether it's worth trying to construct a solution for this or not. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- 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] Call sip:u...@domain.com?
On Thu, 16 Dec 2010 17:05:35 -0500, Jamie A. Stapleton jstaple...@computer-business.com wrote: Just add something like this to your dialplan: exten=1234,1,Dial(SIP/u...@domain.com) Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com. Thanks Jamie, but isn't there a universal way to solve this, so that users can dial any SIP number without first having to create an extension for that specific number? -- _ -- 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] Call sip:u...@domain.com?
Le 17/12/2010 07:45, Gilles a écrit :
On Thu, 16 Dec 2010 17:05:35 -0500, Jamie A. Stapleton
jstaple...@computer-business.com wrote:
Just add something like this to your dialplan:
exten=1234,1,Dial(SIP/u...@domain.com)
Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com.
Thanks Jamie, but isn't there a universal way to solve this, so that
users can dial any SIP number without first having to create an
extension for that specific number?
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
phone connected on it: how to send alpha characters or @ ?
--
Daniel
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Re: [asterisk-users] Call sip:u...@domain.com?
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
phone connected on it: how to send alpha characters or @ ?
Thanks Daniel. I added that line above, told Asterisk to reload the
dialplan, and typed the following in XLite:
9*031...@ekiga.net
This is to perform an echo test
http://wiki.ekiga.org/index.php/Fun_Numbers
I guess something else must be done to Asterisk for this to work:
==
CLI
-- Executing [9*031...@my-phones:1] Dial(SIP/6011-00a1b67c,
SIP/*031600) in new stack
[Dec 17 11:43:14] WARNING[306]: chan_sip.c:2923 create_addr: No such
host: *031600
[Dec 17 11:43:14] WARNING[306]: app_dial.c:1183 dial_exec_full: Unable
to create channel of type 'SIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/6011-00a1b67c' status is
'CHANUNAVAIL'
==
I also tried this, same result:
exten=_9.,1,Dial(SIP/ippi_outgoing/${EXTEN:1})
Do I need to add something in sip.conf, or some other configuration
file?
Thank you.
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Re: [asterisk-users] Call sip:u...@domain.com?
On Thu, 16 Dec 2010 11:54:31 +0100, Gilles codecompl...@free.fr wrote: Now, I'd like to be able to call any number on the Net that is advertised as sip:u...@domain.com, such as those: I mean: Do I really have to first create a section in sip.conf each time a user needs to call a number on a new SIP server? http://wiki.ekiga.org/index.php/Connecting_Asterisk_to_ekiga.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Call sip:u...@domain.com?
Le 17/12/2010 12:48, Gilles a écrit :
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
phone connected on it: how to send alpha characters or @ ?
Thanks Daniel. I added that line above, told Asterisk to reload the
dialplan, and typed the following in XLite:
9*031...@ekiga.net
This is to perform an echo test
http://wiki.ekiga.org/index.php/Fun_Numbers
I guess something else must be done to Asterisk for this to work:
==
CLI
-- Executing [9*031...@my-phones:1] Dial(SIP/6011-00a1b67c,
SIP/*031600) in new stack
[Dec 17 11:43:14] WARNING[306]: chan_sip.c:2923 create_addr: No such
host: *031600
[...]
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
You should read info on voip.org to learn basis of Asterisk.
--
Daniel
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Re: [asterisk-users] Call sip:u...@domain.com?
On 10-12-17 06:48 AM, Gilles wrote:
On Fri, 17 Dec 2010 10:16:00 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Then create a prefix for SIP calls
exten=_9.,1,Dial(SIP/${EXTEN:1})
and you dial 9u...@domain.com from XLite
Remember that calling sip URL is not as easy with a phone. Imagine you have an
ATA with DECT or POTS
phone connected on it: how to send alpha characters or @ ?
Thanks Daniel. I added that line above, told Asterisk to reload the
dialplan, and typed the following in XLite:
9*031...@ekiga.net
This is to perform an echo test
http://wiki.ekiga.org/index.php/Fun_Numbers
You have to tell it the host to request the extension from. All you're doing is
dialing SIP/*031600, which with that format, is going to try and call [*031600]
as defined in sip.conf.
You're missing the host that you want to call. The format needs to be
SIP/*031600@some_hostname
What you're trying to do is essentially what FreeNum was designed for:
http://www.freenum.org
We discuss it in this chapter here:
http://ofps.oreilly.com/titles/9780596517342/ch12.html
Leif.
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Re: [asterisk-users] Call sip:u...@domain.com?
On Fri, 17 Dec 2010 14:36:58 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
Thanks for the tip but I wanted to be able to call _any_ SIP number,
not just Ekiga, so needed a destination-agnostic solution.
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Re: [asterisk-users] Call sip:u...@domain.com?
On Fri, 17 Dec 2010 09:00:39 -0500, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
You have to tell it the host to request the extension from. All you're doing
is
dialing SIP/*031600, which with that format, is going to try and call
[*031600]
as defined in sip.conf.
You're missing the host that you want to call. The format needs to be
SIP/*031600@some_hostname
Thanks for the tip. Elsewhere, someone suggested adding this code in
extensions.conf, which solved the problem:
===
[macro-dialsipuri]
exten = s,1,Set(dialuri=${CUT(ARG1,\;,1)})
exten = s,n,Verbose(Calling SIP URI ${dailuri})
exten = s,n,Verbose(--- From: ${CALLERID(all)})
exten = s,n,Dial(SIP/${dialuri},60,tr)
exten = s,n,Congestion()
[internal]
...
exten = _[a-z].,1,Macro(dialsipuri,${ext...@${sipdomain})
exten = _[A-Z].,1,Macro(dialsipuri,${ext...@${sipdomain})
===
What you're trying to do is essentially what FreeNum was designed for:
http://www.freenum.org
I'll read up on Freenum, but I was just trying to do something that I
thought was very simple, namely make a phone call over the Net, ie.
have XLite send an INVITE to Asterisk, which would then forward the
INVITE to the remote server, which would ring the phone. I expected
Asterisk users to make direct calls routinely, but maybe it's not that
frequent.
We discuss it in this chapter here:
http://ofps.oreilly.com/titles/9780596517342/ch12.html
Thanks Leif. I was going through the 2nd edition, which doesn't seem
to deal with direct, Internet dialing. I'll go through that Chapter 12
in the 3rd edition.
Thank you.
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Re: [asterisk-users] Call sip:u...@domain.com?
On Fri, Dec 17, 2010 at 04:52:32PM +0100, Gilles wrote: Thanks for the tip but I wanted to be able to call _any_ SIP number, not just Ekiga, so needed a destination-agnostic solution. How would you _expect_ to be able to specify a destination server from a telephone keypad? -- _ -- 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] Call sip:u...@domain.com?
Le 17/12/2010 16:52, Gilles a écrit :
On Fri, 17 Dec 2010 14:36:58 +0100, Administrator TOOTAI
ad...@tootai.net wrote:
Domain part disappear.
exten=_9*.,1,Dial(SIP/${EXTEN:1...@ekiga.net)
In Xlite call 9*031600
Thanks for the tip but I wanted to be able to call _any_ SIP number,
not just Ekiga, so needed a destination-agnostic solution.
You can use SipBroker.
http://www.sipbroker.com/sipbroker/action/providerWhitePages
--
Daniel
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[asterisk-users] Call sip:u...@domain.com?
Hello At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT set up with a VOSP trunk that I can use to make/receive calls to/from the PSTN. Now, I'd like to be able to call any number on the Net that is advertised as sip:u...@domain.com, such as those: www.voip-info.org/wiki/view/Phone+Numbers Do I need to register a second trunk (FWD, etc.) through which those calls will be made? Can't my VOSP perform both tasks (landline + Internet calls)? Can I just let my Asterisk server connect to the remote SIP server through the SRV DNS record and have it dial the extension? Any example appreciated, thank you. -- _ -- 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] Call sip:u...@domain.com?
Just add something like this to your dialplan: exten=1234,1,Dial(SIP/u...@domain.com) Then, when you dial 1234 on your XLite, it will connect you to u...@domain.com. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gilles Sent: Thursday, December 16, 2010 5:55 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Call sip:u...@domain.com? Hello At this point, I have an Asterisk 1.4 + PC running XLite behind a NAT set up with a VOSP trunk that I can use to make/receive calls to/from the PSTN. Now, I'd like to be able to call any number on the Net that is advertised as sip:u...@domain.com, such as those: www.voip-info.org/wiki/view/Phone+Numbers Do I need to register a second trunk (FWD, etc.) through which those calls will be made? Can't my VOSP perform both tasks (landline + Internet calls)? Can I just let my Asterisk server connect to the remote SIP server through the SRV DNS record and have it dial the extension? Any example appreciated, thank you. -- _ -- 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
