[asterisk-users] Skype channel beta
On Fri, Sep 26, 2008 at 6:00 AM, Darrick Hartman [EMAIL PROTECTED] wrote: Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. Good question Lots of questions about this one. There's definitely a demand for it so I can see why Digium would be interested in exploring this option. Time will tell how well it will work. I'm personally not too excited about bolt-on binaries which are probably not compatible with uClibc (and therefore Astlinux). That leaves us in the same place as we are with codec_g729. We're at the mercy of whoever creates these binaries to produce one that will work for us. Yeah, another good point. I think just like a lot of things, the Skype-Asterisk bridge idea isn't for everyone. Right now I'm testing a piece of Mac software that allows me to bridge Skype and SIP by connecting the Skype client with X-Lite on one box. It works fine, but with a little lag (maybe 1sec) , obviously. I'm sure things will be easier when this new system goes public. I'd be happy to license a channel or a few channels if the price is right, just like I did with g729. I don't need to konw what's in the black box, but I know many of you out there do for various reasons. We'd all need to know what the implication of the peer to peer model are on your asterisk box. As time goes on, we'll learn more about the details, financial model, etc. Until then, I suppose it's inevitable that some (not directed at those on this thread) will immediately start bitching about things that are not actually decided yet. And what about video? Tune in Friday at Noon Eastern with all the bitching, conjecture and fears or joy. r http://voipusersconference.org ___ -- 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] ZAP not answering call
Paul Hales wrote: Just to check - have you got the right modules plugged into the right sort of lines? Yes - if you have a look at my zapata.conf snippet the zap chanell has signalling=fxs_ks Also - some analog phone interfaces are NOT standard. :( This could be the problem. However, what I did not mention is that there is a old Trixbox server here and it can connect up and answer the line... I can not replicate this even if i take all the settings out of zapata.conf and other relavant files on the tixbox. I just wish we had a standard PSTN line here to test with, as we only have ISDN lines. But the line modules have to be (to work with standard phone lines, of course) Well these analogue lines currently just got to fax's / alarms / modems etc. No PSTN connections -- *Daniel Johnson* Systems Administrator / Systems Development Scanning Systems Australia *Office:* +61 7 3387 *Facsimile:* +61 7 3387 5588 *E-mail:* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] *Website:* http://www.scanningsystems.com.au PaulH Daniel Johnson wrote: Hi, I am trying to interface our old PBX(Siemens) to asterisk via some analogue ZAP lines. The problem is that Asterisk never successfully answers the call. See debug ouput below. If I connect FXO - FXS on the same card and make a call it all works fine. So the card is not faulty. I see there are some stange (to me) messages in the debug. I have done search on google and tried all suggestions but they do not fix. eg. busydetect=no callprogress=no hanguponpolarityswitch=yes Some people suggest its to do with callerID. (How can I tell if the old PBX sends callerID?) have tried turning callerid in asterisk on/off - changing settings. etc. callerid=yes/no cidstart=ring/palarity cidsignalling=v23/bell/dtmf Does anyone have any other ideas? [Sep 26 12:31:54] VERBOSE[7905] logger.c: -- Starting simple switch on 'Zap/53-1' [Sep 26 12:31:54] NOTICE[7905] chan_zap.c: Got event 18 (Ring Begin)... [Sep 26 12:31:55] NOTICE[7905] chan_zap.c: Got event 2 (Ring/Answered)... [Sep 26 12:31:57] NOTICE[7905] chan_zap.c: Got event 18 (Ring Begin)... [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:1] Goto(Zap/53-1, ivr-2|s|1) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Goto (ivr-2,s,1) [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:1] Set(Zap/53-1, LOOPCOUNT=0) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:2] GotoIf(Zap/53-1, 0?begin) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:3] Ringing(Zap/53-1, ) in new stack [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:4] Answer(Zap/53-1, ) in new stack [Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Took Zap/53-1 off hook [Sep 26 12:31:57] DEBUG[7905] chan_zap.c: No echo training requested [Sep 26 12:31:57] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:5] Wait(Zap/53-1, 1) in new stack *[Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED Polarity on channel 53, state 6* *[Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to IDLE on channel 53, state 6 [Sep 26 12:31:57] DEBUG[7905] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1669362190 [Sep 26 12:31:57] WARNING[7905] chan_zap.c: Ring/Off-hook in strange state 6 on channel 53 [Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED Polarity on channel 53, state 6 [Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to IDLE on channel 53, state 6 [Sep 26 12:31:58] DEBUG[7905] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0, honp= 0, pdelay= 600, tv= -1669361310* [Sep 26 12:31:58] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:6] BackGround(Zap/53-1, ssa/welcome) in new stack [Sep 26 12:31:58] VERBOSE[7905] logger.c: -- Zap/53-1 Playing 'ssa/welcome' (language 'en') *[Sep 26 12:31:58] WARNING[7905] chan_zap.c: Ring/Off-hook in strange state 6 on channel 53* [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:7] Set(Zap/53-1, TIMEOUT(digit)=3) in new stack [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Digit timeout set to 3 [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:8] Set(Zap/53-1, TIMEOUT(response)=5) in new stack [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Response timeout set to 5 [Sep 26 12:31:59] VERBOSE[7905] logger.c: -- Executing [EMAIL PROTECTED]:9] WaitExten(Zap/53-1, |) in new stack [Sep 26 12:32:00] DEBUG[7905] chan_zap.c: Ignore switch to REVERSED Polarity on channel 53, state 6 [Sep 26 12:32:00] DEBUG[7905] chan_zap.c: Ignoring Polarity switch to IDLE on channel 53, state 6 [Sep 26 12:32:00] DEBUG[7905] chan_zap.c: Polarity Reversal event occured - DEBUG 2: channel 53, state 6, pol= 0, aonp= 0,
Re: [asterisk-users] Server Dimensioning
On Thu, 25 Sep 2008, Philipp Kempgen wrote: Jon Weisman schrieb: I'm planning on getting a Dell PowerEdge 1950. All I can tell is that I have bad experiences with those Dell PowerEdges. A standard Debian Etch install (2.6.18 kernel I think) didn't even have the driver to run the network interface. At least Dell doesn't seem to play nice with Debian. It's not that Dell doesn't play nice with Debian, but that Debian use what they regard as a stable kernel - which is usually relatively old and lacking the drivers for Dells bleeding edge hardware. I've installed Debian Woody, Sarge and Etch on various Dell hardware over the years and there are sites out there producing Debian netinst CDs with a kernel having the right Dell drivers supplied. However, given the past history of problems I've seen people writing about on this list, I'd be very suspicious of using Dells with plug-in cards. Dells themselves are fine, but it seems there are IRQ issues with some of their systems... (Search the archives) Gordon ___ -- 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 on VMware Workstation 6
yes i have ztdummy loaded. i assume that is what i want. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Gibson Sent: Wednesday, September 24, 2008 8:40 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk on VMware Workstation 6 Do you have ztdummy loaded in the VM? Thanks, Matt G : http://www.voipphreak.ca : http://www.ratemydialplan.com From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael J. Liberatore Sent: Wednesday, September 24, 2008 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk on VMware Workstation 6 Hi, i am running a small personal asterisk server for my business, and instead of getting a dedicated machine to run linux which would waste power and money i decided to run it on my windows xp sp2 machine. The machine is barely used but it does have some crucial programs i need to run in windows so reformating or dual booting is not an option. Its basically a iax2 connection to my voip provider and a sip connection to my phone. It does work well, but the calls especially the voicemail are all garbarled alot. Its definetly not the provider or internet connection because i use this provider for many clients asterisk setups and i also even setup a temp. asterisk setup on this very pc to test to make sure it was infact vmware causing the problem. I upgraded from vmware player to the latest vmware workstation hoping that would fix the problem since its a better system but it hasnt. I also installed and compiled the vmware tools when i installed workstation version. Is this a known issue with vmware? Is there a way to correct the issue either on the windows/vmware side or on the asterisk/linux side? Any other ways to do this project? i looked into astwind or something but either couldnt get it to work or it was unreliable. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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 on VMware Workstation 6
Its really just a very minor system I am running, its sole purpose is a vm basically. Well a VM that can redirect calls based on number. I would prefer to just run it on this windows machine doing nothing most of the time. Id rather not buy an appliance, maybe if its $100 but I would rather just grab an old celeron pc I have laying around and use that, but I am trying to do this green and since this windows pc is running 24/7 anyways (cause I never know when I will need to connect to it) I figured it was a good shot. Maybe a different virtualization software like virtual pc would run better. I think some tweaking is what I need though, I don't care if the call quality is great, I just want it usable. Thanks -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Howes Sent: Thursday, September 25, 2008 7:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk on VMware Workstation 6 Hi, Agreed. Asterisk on a VM appears to work sometimes, only if magic is involved. It is not the way to run anything for a business. Steve On 25 Sep 2008, at 02:36, Dean Collins wrote: Mike, Buy an asterisk appliance like http://www.taa.com/products-vdex-40.html problem solved. If you are worried about good call quality it's either a dedicated pc or a dedicated appliance, one or the other. Cheers, Dean From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Michael J. Liberatore Sent: Wednesday, 24 September 2008 8:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk on VMware Workstation 6 Hi, i am running a small personal asterisk server for my business, and instead of getting a dedicated machine to run linux which would waste power and money i decided to run it on my windows xp sp2 machine. The machine is barely used but it does have some crucial programs i need to run in windows so reformating or dual booting is not an option. Its basically a iax2 connection to my voip provider and a sip connection to my phone. It does work well, but the calls especially the voicemail are all garbarled alot. Its definetly not the provider or internet connection because i use this provider for many clients asterisk setups and i also even setup a temp. asterisk setup on this very pc to test to make sure it was infact vmware causing the problem. I upgraded from vmware player to the latest vmware workstation hoping that would fix the problem since its a better system but it hasnt. I also installed and compiled the vmware tools when i installed workstation version. Is this a known issue with vmware? Is there a way to correct the issue either on the windows/vmware side or on the asterisk/linux side? Any other ways to do this project? i looked into astwind or something but either couldnt get it to work or it was unreliable. thanks mike This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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 ___ -- 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] sip forking needed for ekiga 3.0
Shotgunning the use of IP addresses is foolish at best and lazy programming at worst. Imagine if the poeple writing browsers did that! The internet could end up with double or triple the traffic for no extra benefit not to mention the additinoal load on web servers etc. It's not particularly difficult to determine the best IP address for a piece of client software to use. Check the local machines default gateway, apply the subnet mask and then compare it against all the local IP's. Regards, Greyman. ___ -- 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] Astricon people please post the announcement
On Sep 25, 2008, at 9:00 PM, Darrick Hartman wrote: Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. Lots of questions about this one. There's definitely a demand for it so I can see why Digium would be interested in exploring this option. Time will tell how well it will work. I'm personally not too excited about bolt-on binaries which are probably not compatible with uClibc (and therefore Astlinux). That leaves us in the same place as we are with codec_g729. We're at the mercy of whoever creates these binaries to produce one that will work for us. Darrick According to them today, if the user initiates calls with someone outside the pbx, it will not go through the pbx. The user can register both to skyp and the asterisk also register the user. So, if the user initiates contact to another it is peer to peer outside the pbx. smime.p7s Description: S/MIME cryptographic 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 on VMware Workstation 6
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Michael J. Liberatore wrote: Its really just a very minor system I am running, its sole purpose is a vm basically. Well a VM that can redirect calls based on number. I would prefer to just run it on this windows machine doing nothing most of the time. Id rather not buy an appliance, maybe if its $100 but I would rather just grab an old celeron pc I have laying around and use that, but I am trying to do this green and since this windows pc is running 24/7 anyways (cause I never know when I will need to connect to it) I figured it was a good shot. Maybe a different virtualization software like virtual pc would run better. I think some tweaking is what I need though, I don't care if the call quality is great, I just want it usable. Coming in from the side here... Why not use XEN and have asterisk in a xen VM, which some people here claim to have working, and then have a second VM for windows. I have a windows XP Pro VM running under xen just in case I need to use a windows based application, or whatever, from time to time... It works well for me, I haven't tried asterisk in a VM as yet though... but might one day... Regards, Adam - -- Adam Goryachev Website Managers www.websitemanagers.com.au -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFI3JsIGyoxogrTyiURAi2UAKCGuoNdby+4hSipuVnfaBi6onXfdQCgquSV Yp4eDzhjNg48M+G3lhgdNrA= =KgqM -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] Astricon people please post the announcement
On Thu, Sep 25, 2008 at 05:25:52PM -0700, Fred Posner wrote: On Sep 25, 2008, at 11:06 AM, Steve Anness wrote: So what a minute. They will charge us to use Skype with our Asterisk servers? Yes, I think I shall move along. Steve I talked with both Skype and Digium today at Astricon for a while on this... it's actually going to be amazing. The license for Skype will be the same way you license g.729. So yes, it's not free... but you're only paying for in use channel capabilities... There are already a number of such Skype connectors. Some of them claim to be free (that is: no charge). Some of them take money. I'm not sure if Skype/eBay sees any of that. They tend to at least bend Skype's license if not break it completely (e.g: run a client in a XNest server is a common trick to work around the requirement in the license of the Skype API for an interactive client). So now that we have a blessed client for which Skype/eBay gets payed, what happens to those others? Will they still be legal? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] FAX t.38 on Asterisk 1.6?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Arturo Ochoa wrote: Ok, so it's clear now that this feature is missing on Asterisk, but as Russell states, it's on the roadmap. So, Can you guys give an alternate idea on what to do on this scenario: One customer has this situation: The headquarters are located on MTY, Mexico. They have 2 landlines on Edinburgn TX, USA because they have presence on that city. When one person calls to the landlines on TX, the call goes to MTY. If that call is a FAX it's answered by asterisk+iaxmodem+hylafax and MTY receives an email with the FAX on PDF. The problem goes when someone in MTY wants to send a FAX to some other TX customer. At first I suggested that if they have the document on the computer it'll be easy to use hylafax or even a webportal called AvantFAX. The problem it's the kind of documents they send are not computer documents... they really need the FAX machine... Any ideas on this ?? I'm just catching up on some old posts here, so this might not be relevant anymore... How about accepting the call into rxfax, and then creating a call file which will re-send the fax to it's real destination. ie, asterisk will receive the fax, and then send it, instead of just transparently acting as a gateway ? perhaps, in between the receive and send you might use scp to copy the file to the remote asterisk server, and then ssh to create the call file on the remote asterisk Regards, Adam - -- Adam Goryachev Website Managers www.websitemanagers.com.au -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFI3KbpGyoxogrTyiURAsImAKDSEEJw5GBOYGOJKhUB/VwNyOXGxwCgrS24 DFeGdcqgA0PdzHApN16jLHw= =E1Y8 -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] Astricon people please post the announcement
On Thu, 25 Sep 2008 18:00:00 +0100, Tim Panton wrote: It's essentially a channel driver. Licensed per channel in the same way that the g729 codec is. which would mean that us freebsd folks are going to be left out. oh well. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- 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] OT: Do You Know What the Problem With CDMA is?
On Thu, 25 Sep 2008 12:21:41 -0400, Jason Aarons \(US\) wrote: A lot of places you still can't get GSM in the US.it has improved...but GSM 3G coverage is lacking compared to EVDO/CDMA. which isn't usually a problem as all 3G phones i've seen also use GSM, and the phones switch to GSM when 3G coverage isn't available. You start to explain about GSM and their eyes open wide as they realize they need a unlocked GSM phone from a electronics shop and SIM chip from actually, if you're using a gsm/3G phone, and your carrier has a roaming agreement with a malaysian carrier (there're 3 big ones and 1 small one, by the way), then it shouldnt matter. of course, they'll sock roaming charges on you. some company named Digi sold in 7-Eleven and some scratch off cards for refills using SMS. that's just one of the three, and its a prepaid gsm card you're referring to. you could've also picked up a celcom or a maxis prepaid card, or not worry about that and just roam with a gsm phone. In reality my roaming fees for Intl are too high, I'll get a pre-paid in-country phone before I get phone bill for Intl roaming. My data connection syncs email all day long. i hear the vodaphone 3G service hits you a fixed monthly fee for use anywhere in the world for a data/3G connection. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- 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] T38 fax gateway announcement
Hi, there is http://bugs.digium.com/view.php?id=13405 updated version of fax (T38) gateway. Your bug reports and questions are welcome. Thank you in advance. Best regards Daniel. ___ -- 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] Astricon people please post the announcement
I have tryed skip2pbx 580€ yeastar 60 €, the quality is the way behind of a good sip provider, thay are simply not suitable for business, i hope it would not be the case of asterisk addon. Also i wonder if skype auto relay will be disabled (bandwith), wait and see... ___ -- 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] FAX t.38 on Asterisk 1.6?
Hi, if you are interested in t.38 gatewaying you may try fax gateway that has been posted recently: http://bugs.digium.com/view.php?id=13405. I'm looking forwards seeing any reports. Best regards Daniel. On Fri, Sep 26, 2008 at 11:10 AM, Adam Goryachev [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Arturo Ochoa wrote: Ok, so it's clear now that this feature is missing on Asterisk, but as Russell states, it's on the roadmap. So, Can you guys give an alternate idea on what to do on this scenario: One customer has this situation: The headquarters are located on MTY, Mexico. They have 2 landlines on Edinburgn TX, USA because they have presence on that city. When one person calls to the landlines on TX, the call goes to MTY. If that call is a FAX it's answered by asterisk+iaxmodem+hylafax and MTY receives an email with the FAX on PDF. The problem goes when someone in MTY wants to send a FAX to some other TX customer. At first I suggested that if they have the document on the computer it'll be easy to use hylafax or even a webportal called AvantFAX. The problem it's the kind of documents they send are not computer documents... they really need the FAX machine... Any ideas on this ?? I'm just catching up on some old posts here, so this might not be relevant anymore... How about accepting the call into rxfax, and then creating a call file which will re-send the fax to it's real destination. ie, asterisk will receive the fax, and then send it, instead of just transparently acting as a gateway ? perhaps, in between the receive and send you might use scp to copy the file to the remote asterisk server, and then ssh to create the call file on the remote asterisk Regards, Adam - -- Adam Goryachev Website Managers www.websitemanagers.com.au -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFI3KbpGyoxogrTyiURAsImAKDSEEJw5GBOYGOJKhUB/VwNyOXGxwCgrS24 DFeGdcqgA0PdzHApN16jLHw= =E1Y8 -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 ___ -- 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] Push presence from one asterisk to another
Hello, I would like to push presence from one asterisk to another. Here is my scenario: Office A has 3 users: extension 100,101,103 Office B has 3 users: extension 200,201,203 Now 200 would like to see on his phone (BLF) when user 100 is on the phone. Asterisk of Office A and Asterisk of Office B can talk to each other. The phone of Office A can not talk to the Asterisk of Office B. So the solution where 200 subscribes to a hint extension on Asterisk of Office A is not possible. I have several ideas but I would like to see how others would solve this. Any minor idea is welcome. Best regards, Loic Didelot. ___ -- 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] setting DNID
Hi, I'm using Asterisk 1.2. I have to redirect a call coming from a line with DIDs to an ATA devices but keeping the DNID just as Asterisk would be DNID-transparent. I need this because the machine connected to my ATA needs to know which DID was called from outside. Anybody knows if DNID can be modified? Thank you. Giorgio. ___ -- 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] Get Call Length of Calls
Hi I am using show cannels verbose to get info about my current sip calls. However, the time displayed is always zero. Any hints ? ___ -- 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] Astricon people please post the announcement
On 26 Sep 2008, at 11:17, Grygoriy Dobrovolskyy wrote: I have tryed skip2pbx 580€ yeastar 60 €, the quality is the way behind of a good sip provider, thay are simply not suitable for business, i hope it would not be the case of asterisk addon. Also i wonder if skype auto relay will be disabled (bandwith), wait and see... The Asterisk team said that a) the skype for asterisk code does not act as a supernode - i.e. it only routes traffic for local users, this was one of their requirements. b) they _think_ that in the case where both ends of a skype to skype call are 'local' the huge majority of the bandwidth remains local. c) there will be configuration options controlling which of the transport methods skype for asterisk will use. So you can disable skype over port 443 if you want to ensure that port is available for your ssl webserver (for example) Tim. ___ -- 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] Incoming URL handling Problem (Asterisk problem ?)
Hello, I use an Asterisk box with the following configuration: Operating System : linux Fedora Core 4 (2.6.17-1.2142_FC4smp #1) Asterisk 1.4.18 I use the following asterisk command to send url to client : Dial(IAX2/ciwww/[EMAIL PROTECTED],,,https://xx..it/es/crm/dashboard.php?codice_ordine=xxx-xx-xxx;) I've a problem using the Incoming URL handling feature with my IAX2 client softphone. I've dumped my lan traffic and I've filtered the correct URL: (this is part of my dump (libcap/ASCII) Begin ..pU..U...U.UU.U.UU...UT.U..T..TUU.U...U.U...UU.UU..UUU.UU.UU.TU.UU.U.UU.UUUU.UU..U.TUTT.UUUT.U.U.U..U...UU.UU.U.UU.T.UU..U ..U...AD.UUUU.U.U...UUU...U.UU.UU.UU..U..U.U...UU.U..UUU..UU.U.....U.UUUU..UUU.U..U..U..UUU...UUU.U...U.U.UU.T.UU.UUU.U ...%. .p...UU.UUU...UU...UUTTTUUU......UU...UU...UUU. .. ...(... ...shttps://xx..it/es/crm/dashboard.php?codice_ordine=xxx-xx-xxx;.. .s... ...u.. dd .I*P.. End the incoming call works fine, but I can't see the url. When the client (Zoiper Biz softphone 2.16 on Windows Vista Windows 2000) receives the call, it does not open any browser and it does not generate any warning. Possible Asterisk Problem ? Can you help me ? Best Regards, Fabio ___ -- 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] Dell (was: Re: Server Dimensioning)
Gordon Henderson schrieb: However, given the past history of problems I've seen people writing about on this list, I'd be very suspicious of using Dells with plug-in cards. Dells themselves are fine, but it seems there are IRQ issues with some of their systems... (Search the archives) Exactly. See http://www.digium.com/en/docs/misc/compatibility_notes.php for Digium cards. But the list is not complete. I had problems with other Dell PowerEdge models as well. Others have reported everything works fine for them. 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] Astricon people please post the announcement
On 26 Sep 2008, at 04:36, Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. I asked Mark about that. They expect to have text to work right, when associated with a voice call. It is less clear what happens it it is _just_ a text session. Olle tells me that 1.6 can do text only calls (he's been working on an asterisk for the deaf project) so there is a decent chance they will get it to work. Tim. ___ -- 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] Push presence from one asterisk to another
Loic Didelot schrieb: I would like to push presence from one asterisk to another. Here is my scenario: Office A has 3 users: extension 100,101,103 Office B has 3 users: extension 200,201,203 Now 200 would like to see on his phone (BLF) when user 100 is on the phone. Asterisk of Office A and Asterisk of Office B can talk to each other. The phone of Office A can not talk to the Asterisk of Office B. So the solution where 200 subscribes to a hint extension on Asterisk of Office A is not possible. I have several ideas but I would like to see how others would solve this. Any minor idea is welcome. It's not quite production ready yet. Junghanns' BriStuff can do it via ESEL (extension state export logic). Basically that's a connection between the AMIs. In Asterisk 1.6 you could do it via DEVSTATE(). http://www.asterisk.org/blog/8 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] sip forking needed for ekiga 3.0
On Fri, 2008-09-26 at 08:43 +0100, Grey Man wrote: It's not particularly difficult to determine the best IP address for a piece of client software to use. Oh? Check the local machines default gateway, apply the subnet mask and then compare it against all the local IP's. Yeah? And if more than one matches? Then what? Have you read the whole thread here? b. signature.asc Description: This is a digitally signed message part ___ -- 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] Get Call Length of Calls
i! Not about this directly, but an alternative. If you need the length of finished calls, work with the system. Use a specific call to the date command, so it's easy to evaluate the time info or some other tool to give you an absolute of time. Then at the end of the call use another system call to that program and subtract the two values. You could use bc for this. It even works with decimal numbers. One or two shell-scripts will do this trick. HTH. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] Friday 2008-09-26 12:00:00 Asterisk + Skype on your box
I'm a little surprised no one wants to say anything on IRC this morning about this. I know many of you here are interested. Mark was talking about this three years ago and it was exciting news then as it is now (IMO). Maybe Mark will join us, although I believe he's got a long flight today or maybe even last night. Regardless of who is there, there will be some interesting discussion of chan_skype beta, Astricon and anything else that came up this week in VoIP. Worst case, go see John Todd's shirt on http://youtube.com/voiceroute http://www.voipusersconference.org PSTN (724) 444-7444 Enter 22622# #1 SIP [EMAIL PROTECTED] DTMF 22622# 1# If you have an account, enter your PIN# in the place of the 1# so I know who's calling IRC: #voip-users-conference on freenode.net Twitter: voipusers Bookmarks: http://Delicious.com/voipusersconference Skype: voipusersconference You guessed it, Twitter id's are too short. ___ -- 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] setting DNID
Giorgio Incantalupo schrieb: I'm using Asterisk 1.2. Anybody knows if DNID can be modified? Not sure about 1.2 but at least in 1.4 you can set CALLERID(dnid). 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] Astricon people please post the announcement
Get Olle to call in for once in his life! Mark did say IM and video, IM first. It's all gonna happen. (just not right away) On Fri, Sep 26, 2008 at 3:34 PM, Tim Panton [EMAIL PROTECTED] wrote: On 26 Sep 2008, at 04:36, Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. I asked Mark about that. They expect to have text to work right, when associated with a voice call. It is less clear what happens it it is _just_ a text session. Olle tells me that 1.6 can do text only calls (he's been working on an asterisk for the deaf project) so there is a decent chance they will get it to work. Tim. ___ -- 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] Queue Calls getting stuck in there
i upgraded to 1.4.21.2-2 and set the autofill to on and it solved the problem.. yet i kept the failover settings incase it happens again.. so if it happens.. the fail over will redirect the caller to the same queue but the conditions will apply like it was a new call. i found out that there is a problem with the SIP hints.. sometimes my agent has busy hint while it's not onthe phone!!! AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Thu, 25 Sep 2008 20:36:50 -0500 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Queue Calls getting stuck in there Did you ever solve this? I am experiencing the same issue with 1.4.21.2. I have turned autofill on and tried incoming limits, but no luck. It happens at least once per day. Agents will be available but calls will just sit there until one of the waiting agents logs off and back in. Andrew Tariq .. wrote: the Autofill thing didn't solve the problem.. i have another server hosted in the USA with Asterisk 1.4.20-1 on it.. it doesn't have that problem.. in the server i'm talking about the only way i found to avoid this problem is to set a time out for the queue then the user is rotated into the same queue again.. that will give the waiting users a chance to go delivered.. i'm already questioning my agents about the delay in answering the calls so i set the time out to 3 seconds where the caller will be rotated in turns and the queue will be working fine.. the Autofill worked with this slution pretty well .. plus when the stuck caller gets rotated his chance of getting connected to an agent go higher as he won't be stuck forever.. but i need a better solution for this problem.. so im thinking of installing the Asterisk 1.4.20-1 which i haven't faced any problem with since i installed it. Regards From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Sat, 13 Sep 2008 22:00:14 -0700 Subject: Re: [asterisk-users] Queue Calls getting stuck in there Try the autofill=yes setting available in queues.conf Original Message Subject: [asterisk-users] Queue Calls getting stuck in there From: Tariq .. Date: Sat, September 13, 2008 5:53 pm To: Asterisk Users Greetings, i have a problem with my asterisk .. i'm using Asterisk 1.4.19-1 with FreePBX 2.4.1.1 and TrixBox the problem is that i'm having is the following.. a call comes to a Queue.. the caller must be forwarded to one of the free members who are waiting.. but instead of going to a member.. the caller stays in the queue without being forwarded.. i tried to play with the timeout and fail over times but the caller stays in the queue no matter what.. following are my Queues.conf , Extensions.conf, SIP.conf for one of my queues ___ -- 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 _ Stay up to date on your PC, the Web, and your mobile phone with Windows Live. http://clk.atdmt.com/MRT/go/msnnkwxp1020093185mrt/direct/01/ ___ -- 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] Get Call Length of Calls
Hi Thanks for the hint, however I do already have a cdr tool for finished calls. core show channels verbose does show the duration of calls in real time. However, it does not work all the time, I.e. at times it works great other times it just displays 0 for the call duration, although the call is up and running. On Fri, Sep 26, 2008 at 4:37 PM, Julien Claassen [EMAIL PROTECTED] wrote: i! Not about this directly, but an alternative. If you need the length of finished calls, work with the system. Use a specific call to the date command, so it's easy to evaluate the time info or some other tool to give you an absolute of time. Then at the end of the call use another system call to that program and subtract the two values. You could use bc for this. It even works with decimal numbers. One or two shell-scripts will do this trick. HTH. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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 -- -- With Regards Ali Jawad System Administrator http://www.alijawad.org Phone : +961-01-559031 Mobile : +961-03-041705 Confidentiality Notice: The contents of this E-mail are intended for the named recipient only. It may contain confidential and privileged information. If you received it in error, please notify us immediately and then destroy it. Internet communications are not secure and therefore I do we do not accept legal responsibility for the contents of this message. Also, and though we provide every effort to keep our network free from viruses, you would need to check this E-mail and any attachments for viruses as we can take no responsibility for any computer virus which might be transferred by way of this E-mail. ___ -- 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] sip forking needed for ekiga 3.0
On Fri, Sep 26, 2008 at 2:36 PM, Brian J. Murrell [EMAIL PROTECTED] wrote: Check the local machines default gateway, apply the subnet mask and then compare it against all the local IP's. Yeah? And if more than one matches? Then what? Use one of them! And if the network set up is too complex that that still causes problems do what the Ekiga guy told you and set the IP address you want to use in the config file. It's not a difficult situation. I have servers with 15 public IP addresses on them and manage to run SIP services no problems. I've read enough of the thread to know the Asterisk issue you are trying to describe is loop detection and not forking. Asterisk does support forking: Dial(SIP/user1SIP/user2) is forking. Not being able to handle duplicate requests from different IPs is loop handling and you'll already find bugs open about that. Oh? I like to make sure I've done my homework before ascending to sarcasm especially when I'm the one that requested Thots in the first place... Greyman. ___ -- 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] Astricon people please post the announcement
On Fri, Sep 26, 2008 at 03:44:01PM +0200, randulo wrote: Get Olle to call in for once in his life! Mark did say IM and video, IM first. It's all gonna happen. (just not right away) On the topic of #pidgin they say, amomng others, Pidgin does NOT support voice or video. Likewise we should state on #asterisk Asterisk does NOT support text chats. There are some awkward methods for sending some text messages over some channels (SMS in european POTS, SIMPLE and simpler texxt messages in SIP, XMMP for Jingle, and well, probably nothing in IAX. Bristuff ads even a few more bits there). But do we actually care routing those messages from one place to another? This is a major limitation of Asterisk for me. Text messages require much lower a bandwith and a text connection is much easier to setup. Hence it can work even when a voip connection is lousy. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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] sip forking needed for ekiga 3.0
On Fri, 2008-09-26 at 14:54 +0100, Grey Man wrote: On Fri, Sep 26, 2008 at 2:36 PM, Brian J. Murrell [EMAIL PROTECTED] wrote: Yeah? And if more than one matches? Then what? Use one of them! And if the one I choose to use doesn't work because of some kind of policy routing or filtering, etc.? And if the network set up is too complex that that still causes problems do what the Ekiga guy told you and set the IP address you want to use in the config file. Uhm. That's not an option. There is no option to do that. Should there be? Perhaps. But hey, I'm just here to report the failure (i.e. to properly accept multiple legal registrations through this so called SIP forking) that the Ekiga guys are telling me about. I've read enough of the thread to know the Asterisk issue you are trying to describe is loop detection and not forking. Asterisk does support forking: Dial(SIP/user1SIP/user2) is forking. Not being able to handle duplicate requests from different IPs is loop handling and you'll already find bugs open about that. I will relay your description of loop detection back on to the Ekiga guys. I'm just the messenger here. b. signature.asc Description: This is a digitally signed message part ___ -- 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] sip forking needed for ekiga 3.0
A machine with more than one default gateway is a VERY special case (used for load-balancing or possibly failover). Most systems will not allow it. I mean... logically, it's odd. Default means when not applied to any other special rule, I choose this one.Not this two. Not this three. This one. It used to be called the Gateway of Last Resort. Last being final and not penultimate. With that being said, if you somehow manage to get by the internal consistency checks and more than one interface (and by interface, I also mean alias, as those are 'virtual interfaces') matches the default gateway, your machine is misconfigured and internet traffic will not properly flow. I know you're just the messenger here, and it's not your fault. But the message is wrong. Ekiga has tried to solve a problem (that of determining a 'best path' for SIP to allow data flow in a NAT or filtered scenario) using poorly thought-out logic. While there may be any number of SIP proxies out there (SER is one of them, and I know that's what the Ekiga service uses) that might be able to handle a mistake on the client side with ease and grace, there's no guarantee that they all will, and assuming they will simply because your test environment allows it is lazy. The RFCs are there for a reason. All SIP forking is UAS territory. Not UAC territory. N. Brian J. Murrell wrote: On Fri, 2008-09-26 at 08:43 +0100, Grey Man wrote: It's not particularly difficult to determine the best IP address for a piece of client software to use. Oh? Check the local machines default gateway, apply the subnet mask and then compare it against all the local IP's. Yeah? And if more than one matches? Then what? Have you read the whole thread here? b. ___ -- 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] sip forking needed for ekiga 3.0
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J. Murrell Sent: Friday, September 26, 2008 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] sip forking needed for ekiga 3.0 I've read enough of the thread to know the Asterisk issue you are trying to describe is loop detection and not forking. Asterisk does support forking: Dial(SIP/user1SIP/user2) is forking. Not being able to handle duplicate requests from different IPs is loop handling and you'll already find bugs open about that. I will relay your description of loop detection back on to the Ekiga guys. I'm just the messenger here. Can you provide a SIP trace of the signalling taking place? What are you getting back from the Asterisk box? - Brad ___ -- 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] Astricon people please post the announcement
2008/9/26 randulo [EMAIL PROTECTED] Get Olle to call in for once in his life! Mark did say IM and video, IM first. It's all gonna happen. (just not right away) http://lists.digium.com/mailman/listinfo/asterisk-users Video ? that could be really nice but limited to pc/macasteriskwhatever. There are tonns of 3G phones on the market, so why not to adapt software fot the videocalls over wifi ? such a client is my dream for about a year, and i dont care it it would be a skype or else. A new product for that purpose is not a solution, but adapting software to existing 3G phones will open a HUGE market recently created and closed for 3G operators w/licence. Any suggestions ? ___ -- 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] Sip reload casuing issues
carl Lougher wrote: Howdy, Running asterisk 1.4.13 Sometime when running a sip reload the clients are unable to make and receive calls.. Any pointers? That can happen when Asterisk is contacting DNS servers to resolve host names and there are delays in responses (which is done with a sip reload). Try to replace all hostnames with IP Addresses and you will see the problem go away. Andres http://www.neuroredes.com No errors in debug or asterisk console so far.. Cheers, Taff.. ___ -- 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] sip forking needed for ekiga 3.0
On Fri, 2008-09-26 at 10:16 -0400, SIP wrote: The RFCs are there for a reason. All SIP forking is UAS territory. Not UAC territory. From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras asks: I repeat, Ekiga is doing something perfectly legal. The real question is why does Asterisk think it is the same request when the from tag is different ? b. signature.asc Description: This is a digitally signed message part ___ -- 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] sip forking needed for ekiga 3.0
Brian J. Murrell wrote: On Fri, 2008-09-26 at 10:16 -0400, SIP wrote: The RFCs are there for a reason. All SIP forking is UAS territory. Not UAC territory. From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras asks: I repeat, Ekiga is doing something perfectly legal. The real question is why does Asterisk think it is the same request when the from tag is different ? b. Oh yes. It's perfectly legal. It's also a) NOT SIP forking, b) Lazy, and c) Poorly designed. Sending multiple requests and hoping and praying that the recipient will ignore two of them (it will NOT in many cases -- specifically set out by the RFC -- see MESSAGE) because the tag is different doesn't make it any less poorly designed just because it's not specifically written that it can't be done. N. ___ -- 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] sip forking needed for ekiga 3.0
On Fri, 2008-09-26 at 10:41 -0400, SIP wrote: Oh yes. It's perfectly legal. It's also a) NOT SIP forking, b) Lazy, and c) Poorly designed. Sending multiple requests and hoping and praying that the recipient will ignore two of them (it will NOT in many cases -- specifically set out by the RFC -- see MESSAGE) because the tag is different doesn't make it any less poorly designed just because it's not specifically written that it can't be done. The Ekiga developer points out: Have a look at this link : http://www.faqs.org/rfcs/rfc3261.html And look how an UAS (Asterisk in this case) is supposed to handle merged requests : 8.2.2.2 Merged Requests If the request has no tag in the To header field, the UAS core MUST check the request against ongoing transactions. If the From tag, Call-ID, and CSeq exactly match those associated with an ongoing transaction, but the request does not match that transaction (based on the matching rules in Section 17.2.3), the UAS core SHOULD generate a 482 (Loop Detected) response and pass it to the server transaction. The same request has arrived at the UAS more than once, following different paths, most likely due to forking. The UAS processes the first such request received and responds with a 482 (Loop Detected) to the rest of them. In our case, the From tag is different, so it should not detect a loop. This excerpt show that any client or server should be able to receive merged requests. There is obviously a bug here in Asterisk. I have no idea how (in-)correct it is, again, just being the messenger. b. signature.asc Description: This is a digitally signed message part ___ -- 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] Astricon people please post the announcement
On Fri, Sep 26, 2008 at 11:59:35AM +0300, Tzafrir Cohen wrote: On Thu, Sep 25, 2008 at 05:25:52PM -0700, Fred Posner wrote: On Sep 25, 2008, at 11:06 AM, Steve Anness wrote: So what a minute. They will charge us to use Skype with our Asterisk servers? Yes, I think I shall move along. Steve I talked with both Skype and Digium today at Astricon for a while on this... it's actually going to be amazing. The license for Skype will be the same way you license g.729. So yes, it's not free... but you're only paying for in use channel capabilities... There are already a number of such Skype connectors. Some of them claim to be free (that is: no charge). Some of them take money. I'm not sure if Skype/eBay sees any of that. They tend to at least bend Skype's license if not break it completely (e.g: run a client in a XNest server is a common trick to work around the requirement in the license of the Skype API for an interactive client). So now that we have a blessed client for which Skype/eBay gets payed, what happens to those others? Will they still be legal? I wonder if http://narod.ru/disk/2812178000/asterisk-skype.gz.html is legal. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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 on VMware Workstation 6
On Wednesday 24 September 2008 19:28:17 Michael J. Liberatore wrote: Hi, i am running a small personal asterisk server for my business, and instead of getting a dedicated machine to run linux which would waste power and money i decided to run it on my windows xp sp2 machine. The machine is barely used but it does have some crucial programs i need to run in windows so reformating or dual booting is not an option. One option might be to run in the opposite vmware direction. That is, run Linux as the native OS and run Windows within a vmware instance. That gives you the Windows compatibility for your applications, while at the same time providing the critical hardware timing for your Asterisk instance. -- Tilghman ___ -- 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 Plan Issues
Steve Murphy wrote: On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote: Greetings, i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox.. i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls while it is possible in the opposit direction.. i have the same exact settings for the extensions.conf i tried with another friend of mine.. and connected to his server.. and it didn't allow him to send me calls.. so my question is.. is it possible that my server is not accepting any context ? it only runs the ones that come default with Trixbix.. like chanspy, ext-local, from-trunk... and so on.. what can i do to avoide this problem?? i can't rebuild a new box this one is a production server and i wasn't making tests.. i was connecting two of my employer's servers with each other.. regards Tariq-- You might try a trixbox users mailing list. There might be a few trixbox users hanging around in this group who might be able to help, but your chances are much better in that list. murf The server that is not accepting calls is not behind a NAT firewall by any chance is 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
Re: [asterisk-users] Astricon people please post the announcement
Brian J. Murrell wrote: And so will this channel driver also allow Skype to use my resources (CPU, bandwidth -- i.e. Internet for which many have usage caps, etc.) the way the Skype client does? The Skype engine in Skype For Asterisk does not currently have 'relay' support, so it does not route calls or media any calls that it is not involved in. However, this will be present in the production release of the product, but when it appears we will also document its behavior and the configuration options that can be used to control it. -- 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] server and 2 uniden phones no ringing
I have a box running asterisk 1.4.17 that had been working. it has 2 uniden phones connected on it. This was working and now the phones dont ring when calling each other. below is the sip debug. I cant see why the other phone does not ring? I also tried changing the canreinvite for no to yes but that made no difference after restarting. Very simple network. server, linksys router and 2 phones. 192.168.1.X for everything. Any ideas? Jerry [522] type=friend username=522 secret=522 dtmfmode=RFC2833 host=dynamic context=smvoice-sip callerid=522 522 522 qualify=no canreinvite=no nat=no disallow=all allow=ulaw allow=alaw allow=gsm [532] type=friend username=532 secret=532 dtmfmode=RFC2833 host=dynamic context=smvoice-sip callerid=532 qualify=no canreinvite=no nat=no disallow=all allow=ulaw allow=alaw allow=gsm demobox*CLI --- SIP read from 192.168.1.75:5060 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKz8184a8a520f49fe1a53a4d9647a51fb7 Call-ID: [EMAIL PROTECTED] CSeq: 420456 INVITE From: 532 sip:[EMAIL PROTECTED];tag=6619ac3b4bbd705d7102c4565d72e1bc To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Session-Expires: 300 Content-Type: application/sdp User-Agent: Uniden SIP Phone p2 Ver BS4.77 Max-Forwards: 70 S [Kdemobox*CLI upported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK Content-Length: 269 v=0 o=- 1794556993 298723 IN IP4 192.168.1.75 s=- c=IN IP4 192.168.1.75 t=0 0 m=audio 30006 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,16,17,18 a=sendrecv a=ptime:20 - ?--- (14 headers 13 lines) --- ? [Kdemobox*CLI Sending to 192.168.1.75 : 5060 (no NAT) ? [Kdemobox*CLI Using INVITE request as basis request - [EMAIL PROTECTED] ? [Kdemobox*CLI --- Reliably Transmitting (no NAT) to 192.168.1.75:5060 --- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKz8184a8a520f49fe1a53a4d9647a51fb7;received=192.168.1.75 From: 532 sip:[EMAIL PROTECTED];tag=6619ac3b4bbd705d7102c4565d72e1bc To: sip:[EMAIL PROTECTED];tag=as15ac0056 Call-ID: [EMAIL PROTECTED] CSeq: 420456 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=2ebfd7ed Content-Length: 0 ? [Kdemobox*CLI Scheduling destruction of SIP dialog '[EMAIL PROTECTED]' in 32000 ms (Method: INVITE) ?Found user '532' ? [Kdemobox*CLI --- SIP read from 192.168.1.75:5060 --- ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKz8184a8a520f49fe1a53a4d9647a51fb7 CSeq: 420456 ACK To: sip:[EMAIL PROTECTED];tag=as15ac0056 Call-ID: [EMAIL PROTECTED] From: 532 sip:[EMAIL PROTECTED];tag=6619ac3b4bbd705d7102c4565d72e1bc User-Agent: Uniden SIP Phone p2 Ver BS4.77 - ?--- (7 headers 0 lines) --- ? [Kdemobox*CLI --- SIP read from 192.168.1.75:5060 --- INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 192.168.1.75:5060;branch=z9hG4bKvf10df6048844205ddfc1a15bace4c673 CSeq: 420457 INVITE Call-ID: [EMAIL PROTECTED] From: 532 sip:[EMAIL PROTECTED];tag=6619ac3b4bbd705d7102c4565d72e1bc To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060 Session-Expires: 300 Content-Type: application/sdp User-Agent: Uniden SIP Phone p2 Ver BS4.77 Max-Forwards: 70 S [Kdemobox*CLI upported: sip-cc, sip-cc-01, replaces, timer Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, PRACK Content-Length: 269 Proxy-Authorization: Digest realm=asterisk, nonce=2ebfd7ed, algorithm=MD5, uri=sip:[EMAIL PROTECTED], username=532, response=301dfbf68f00b164f64effa90188bf58 v=0 o=- 1794556993 298723 IN IP4 192.168.1.75 s=- c=IN IP4 192.168.1.75 t=0 0 m=audio 30006 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15,16,17,18 a=sendrecv a=ptime:20 - ?--- (15 headers 13 lines) --- ? [Kdemobox*CLI Sending to 192.168.1.75 : 5060 (no NAT) ?Using INVITE request as basis request - [EMAIL PROTECTED] ? [Kdemobox*CLI Found user '532' ? [Kdemobox*CLI Found RTP audio format 0 ? [Kdemobox*CLI Found RTP audio format 8 ?Found RTP audio format 18 ?Found RTP audio format 101 ?Peer audio RTP is at port 192.168.1.75:30006 ? [Kdemobox*CLI Found audio description format PCMU for ID 0 ?Found audio description format PCMA for ID 8 ?Found audio description format G729 for ID 18 ?Found audio description format telephone-event for ID 101 ? [Kdemobox*CLI Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) ?Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) ? [Kdemobox*CLI Peer audio RTP is at port 192.168.1.75:30006 ?Looking for
Re: [asterisk-users] Astricon people please post the announcement
- Tzafrir Cohen [EMAIL PROTECTED] wrote: There are some awkward methods for sending some text messages oversome channels (SMS in european POTS, SIMPLE and simpler texxt messages in SIP, XMMP for Jingle, and well, probably nothing in IAX. Bristuff ads even a few more bits there). But do we actually care routing those messages from one place to another? This is a major limitation of Asterisk for me. Text messages require much lower a bandwith and a text connection is much easier to setup. Hence it can work even when a voip connection is lousy. The specific thing here (that makes handling text messages within the framework of the more complicated protocol attractive) is *addressibility*. If you already have a path to someone, why should you be forced to *discover* another path to them for some other, simpler protocol? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- 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] Dial issue
Hi, when I make a call I need that the caller can** hang up by dialing ***(H option in Dial command), the call but it don´t work. Command EXEC DIAL Zap/g1/433391|20|H In CLI... -- AGI Script Executing Application: (DIAL) Options: (Zap/g1/433391|20|H) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/433391 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/510093-082160f0 (--- At this moment I press * several times, but nothing happens Then I hung up the phone--) -- Hungup 'Zap/1-1' Any Ideas? ___ -- 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 retention
Asterisk version 1.2.27 We are running into issues where people are not deleting their voicemails and it is filling up the storage for voicemail. We would like to run a script that dumps all voicemail that are older than X days. Can we simply check the date time stamp on the message directory and delete those files older than X days or will that mess up the sequence of the voicemails? Anyone have a smooth way of doing this in 1.2? Thanks Phil ___ -- 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] Bizarre international call problem.
Hi, all. We've got a PoS legacy PBX at my company that doesn't have call
accounting. I figured, Hey, why not stick a dual-span T1 Asterisk-based
system in the middle? Then, I just passively pass in-bound calls to the
PBX, and outbound calls to the PSTN. I can then have Asterisk do all the
call accounting, and everything should Just Work. Right?
Well, not so much.
My outbound dialing rule was incredibly complex:
exten = _X.,1,Dial(${PASSTHROUGHTRUNK}/${EXTEN})
And everything seemed to be working ducky, until I went to call Germany
and got -- a local cell phone number. Needless to say, this puzzled me
greatly. A quick look at my log, though, showed that all calls dialed
with 011 were being submitted from the PBX to the Asterisk box without
the 011. (Ironically, if I dial the number with 011011 in front, it
goes through fine.)
So I'm confused: any ideas on how this worked when the PBX was hooked
straight to the PSTN? Is there some SS7 signal or something that says,
This is an international call, when the number has no 011 preface? I'd
hate to have to revert, but I will if need be... *sigh*
Thanks for any insights. I'm totally flummoxed.
-Ken
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Re: [asterisk-users] server and 2 uniden phones no ringing
Jerry Geis wrote: I have a box running asterisk 1.4.17 that had been working. it has 2 uniden phones connected on it. This was working and now the phones dont ring when calling each other. below is the sip debug. I cant see why the other phone does not ring? I also tried changing the canreinvite for no to yes but that made no difference after restarting. Very simple network. server, linksys router and 2 phones. 192.168.1.X for everything. Any ideas? Jerry snip Based on the SIP debug included here, it appears that Asterisk is not receiving a response to the INVITE it is sending to 522 (192.168.1.99). Since the phone is not ringing, it makes me suspect that for some reason the linksys is preventing the INVITE from reaching the phone. If you can look at a packet capture on the linksys, you may want to verify that the linksys isn't modifying or blocking the INVITE that Asterisk is sending. Mark Michelson ___ -- 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] sip forking needed for ekiga 3.0
Brian J. Murrell wrote: On Fri, 2008-09-26 at 10:16 -0400, SIP wrote: The RFCs are there for a reason. All SIP forking is UAS territory. Not UAC territory. From http://bugzilla.gnome.org/show_bug.cgi?id=553810 Damien Sandras asks: I repeat, Ekiga is doing something perfectly legal. The real question is why does Asterisk think it is the same request when the from tag is different ? b. Asterisk ignores tags in To and From headers unless you have pedantic=yes set in sip.conf. Mark Michelson ___ -- 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] Voicemail retention
On Fri, 26 Sep 2008, Asterisk User List wrote:
Asterisk version 1.2.27
We are running into issues where people are not deleting their
voicemails and it is filling up the storage for voicemail. We would
like to run a script that dumps all voicemail that are older than X
days.
Can we simply check the date time stamp on the message directory and
delete those files older than X days or will that mess up the sequence
of the voicemails?
Anyone have a smooth way of doing this in 1.2?
Standard unix stuff:
cd /voicemail
find . -ctime +7 -type f -exec rm {} \;
However you'll need to do some work to stop it deleting the announcments.
Left as an excercise to the user because you really ought to know this
stuff. IMO.
Gordon
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Re: [asterisk-users] server and 2 uniden phones no ringing
snip Based on the SIP debug included here, it appears that Asterisk is not receiving a response to the INVITE it is sending to 522 (192.168.1.99). Since the phone is not ringing, it makes me suspect that for some reason the linksys is preventing the INVITE from reaching the phone. If you can look at a packet capture on the linksys, you may want to verify that the linksys isn't modifying or blocking the INVITE that Asterisk is sending. Mark Michelson Mark, Thanks, I looked at the Linksys WRT54G wireless router. I see nothing that would stop the invite. I have disabled the firewall on the server. I have updated to 1.4.21.1 Still same behavior. I can call into the dialplay and hear audio of some playback wave file. I just cant call another phone. it never rings. The linksys router is set as the default and DHCP is turned off as my server is providing DHCP. Any other toughts? Jerry ___ -- 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] Voicemail retention
Asterisk User List schrieb: Asterisk version 1.2.27 We are running into issues where people are not deleting their voicemails and it is filling up the storage for voicemail. We would like to run a script that dumps all voicemail that are older than X days. Can we simply check the date time stamp on the message directory and delete those files older than X days or will that mess up the sequence of the voicemails? Anyone have a smooth way of doing this in 1.2? Thanks Phil hello afaik is there a script in the asterisk source tree in the folder tools which does exactly what you need. sorry but i dont remember the name of it. best regards Stefan Schmidt ___ -- 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] Extremely OT: I need someone who can parse a MS Word or PDF or RTF document
Hi, I have a complex job totally unrelated to asterisk. I only post here because there are so many bright people on the list. Sorry, but someone may need a buck so write me if you are interested. Otherwise, ignore. We have as input a newsletter type document, originally in MS Word (but obviously can be exported as PDF or RTF if that helps). I need someone who can parse these documents and spit out a series of mysql INSERT statements that result form the position, context, symbology (color, bold, etc), of the text. It's a big challenge, but it might be an interesting job for someone who thinks they are ready and able to take it on. Basically this is taking a human readable text and turning it into a bunch of database SQL inserts. Please contact me OFF LIST if you are interested in bidding on the job. 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] Asterisk on VMware Workstation 6
Your idea (and adam's to run xen) is a very good idea. I have considered it but I'd rather not do a complete reinstall on this xp machine, but if I can deal with that then it would prob work well. I am going to play with the settings, etc to try to get this working first though. Or like I mentioned maybe I will try virtual pc. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tilghman Lesher Sent: Friday, September 26, 2008 12:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk on VMware Workstation 6 On Wednesday 24 September 2008 19:28:17 Michael J. Liberatore wrote: Hi, i am running a small personal asterisk server for my business, and instead of getting a dedicated machine to run linux which would waste power and money i decided to run it on my windows xp sp2 machine. The machine is barely used but it does have some crucial programs i need to run in windows so reformating or dual booting is not an option. One option might be to run in the opposite vmware direction. That is, run Linux as the native OS and run Windows within a vmware instance. That gives you the Windows compatibility for your applications, while at the same time providing the critical hardware timing for your Asterisk instance. -- Tilghman ___ -- 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 This E-mail, including any attachments, may be intended solely for the personal and confidential use of the sender and recipient(s) named above. This message may include advisory, consultative and/or deliberative material and, as such, would be privileged and confidential and not a public document. Pursuant to 42 CFR, any information in this e-mail identifying a former, present, or potential client of Straight Narrow is confidential. If you have received this e-mail in error, you must not review, transmit, convert to hard copy, copy, use or disseminate this e-mail or any attachments to it and you must delete this message. You are requested to notify the sender by return e-mail. ___ -- 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] Astricon people please post the announcement
Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. Keeping in mind that the product has not yet entered beta testing... at this time, all chat messages are ignored by the Skype For Asterisk product. We have discussed today the possibility of being able to send chats to Skype users (sort of a way to do 'screen pop' information for a call you are about to send them), but haven't got any plans at the moment for incoming chat messages. -- 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
Re: [asterisk-users] Extremely OT: I need someone who can parse a MS Word or PDF or RTF document
randulo schrieb: I have a complex job totally unrelated to asterisk. I only post here because there are so many bright people on the list. Sorry, but someone may need a buck so write me if you are interested. Otherwise, ignore. We have as input a newsletter type document, originally in MS Word (but obviously can be exported as PDF or RTF if that helps). RTF would probably help a lot. I need someone who can parse these documents and spit out a series of mysql INSERT statements that result form the position, context, symbology (color, bold, etc), of the text. It's a big challenge, but it might be an interesting job for someone who thinks they are ready and able to take it on. Basically this is taking a human readable text and turning it into a bunch of database SQL inserts. Out of curiosity: Why? 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] Astricon people please post the announcement
2008/9/26 Kevin P. Fleming [EMAIL PROTECTED] Brian J. Murrell wrote: And so will this channel driver also allow Skype to use my resources (CPU, bandwidth -- i.e. Internet for which many have usage caps, etc.) the way the Skype client does? The Skype engine in Skype For Asterisk does not currently have 'relay' support, so it does not route calls or media any calls that it is not involved in. However, this will be present in the production release of the product, but when it appears we will also document its behavior and the configuration options that can be used to control it. -- 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 Will it be packed into the base asterisk package, or to asterisk-addons? or into some third party ? Would it be possible to buy some comminication licences use them while disabling the 'relay' function ? ___ -- 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] Fax with asterisk
On Thu, Sep 25, 2008 at 7:34 AM, Rizwan Hisham [EMAIL PROTECTED] wrote: The fax is originated from a fax machine connected to an ata which supports t38. That would be great if Asterisk had true T.38 support. It can pass the T.38 packets it receives to another SIP endpoint (it will do this even if the other device doesn't suppor tT.38 -- which cause the call to drop) but it cannot originate nor terminate T.38 traffic. If you have a VoIP provider or Cisco gateway that support T.38 then that's all you need but if you want to terminate the calls yourself on a T1/E1 T.38 does not help when using Asterisk. ___ -- 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] Extremely OT: I need someone who can parse a MS Word or PDF or RTF document
I know of someone who was involved in a software project like this - scanning paper documents and importing files into a massive searchable database for a large legal company. Many of the documents were more than 1000 pages long. The amount of money spend on the project was stunning. PaulH On Fri, 2008-09-26 at 22:23 +0200, randulo wrote: Hi, I have a complex job totally unrelated to asterisk. I only post here because there are so many bright people on the list. Sorry, but someone may need a buck so write me if you are interested. Otherwise, ignore. We have as input a newsletter type document, originally in MS Word (but obviously can be exported as PDF or RTF if that helps). I need someone who can parse these documents and spit out a series of mysql INSERT statements that result form the position, context, symbology (color, bold, etc), of the text. It's a big challenge, but it might be an interesting job for someone who thinks they are ready and able to take it on. Basically this is taking a human readable text and turning it into a bunch of database SQL inserts. Please contact me OFF LIST if you are interested in bidding on the job. 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
[asterisk-users] Audio Files
Hello there, I wan to know what is the files that have the control of the quality the sound, When I call a extension, and reproduced a file gsm, or I tolk why another extension, have noise... I thinks that is because have bad quality in the .conf. Thanks. Abel ___ -- 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] Audio Files
Hi! I think all - at least all PSTN - calls have the same quality in means of bitrate, number of channels and samplerate. It's 8kHz, 16bit and mono. About noise, I didn't have problems with that. Seems it's not really about quality. Probably it would be helpful, if you tell us, which extensions/protocol you used. Kindest regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- 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] Astricon people please post the announcement
Grygoriy Dobrovolskyy wrote: Will it be packed into the base asterisk package, or to asterisk-addons? or into some third party ? Would it be possible to buy some comminication licences use them while disabling the 'relay' function ? Skype For Asterisk will be distributed as a separate package. We do not know yet what (if any) requirements will have to be handled for disabling the relay functionality. -- 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
Re: [asterisk-users] Push presence from one asterisk to another
Philipp Kempgen wrote: Junghanns' BriStuff can do it via ESEL (extension state export logic). Basically that's a connection between the AMIs. In Asterisk 1.6 you could do it via DEVSTATE(). http://www.asterisk.org/blog/8 Asterisk 1.6.1 will have distributed device state as well, although the current mechanisms for distribution (OpenAIS) are designed only for use over a low latency LAN connection, not VPNs or WAN links. -- 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
Re: [asterisk-users] users.conf behavior
Dave Poirier wrote: I have an Asterisk server running 1.4.20 and I have all my users in users.conf. Inside users.conf I used... #include ww-users.conf Thats seems to work great with one exception... The exception is that anytime anyone updates their voicemail password, Asterisk rewrites users.conf combines ww-users.conf and it removes my include line from users.conf. Is that expected behavior? I guess that I would have expected it to know to write the changes to the corresponding include file. Is there a better place to put the include? Maybe a better way to handle breaking my users up by location? Should I be using and include in the users.conf This is a flaw in the design of the config file rewriting logic in Asterisk 1.4.x; it's been redesigned in Asterisk 1.6 and doesn't have this problem. Unfortunately the code changes required to do this were too invasive to risk putting into 1.4. If you are going to use the Asterisk GUI (which is what rewrites users.conf), you should just leave all the users in the single users.conf file; there's really not much value in splitting them up. -- 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
Re: [asterisk-users] Bizarre international call problem.
My outbound dialing rule was incredibly complex:
exten = _X.,1,Dial(${PASSTHROUGHTRUNK}/${EXTEN})
And everything seemed to be working ducky, until I went to call Germany
and got -- a local cell phone number. Needless to say, this puzzled me
greatly. A quick look at my log, though, showed that all calls dialed
with 011 were being submitted from the PBX to the Asterisk box without
the 011. (Ironically, if I dial the number with 011011 in front, it
goes through fine.)
So I'm confused: any ideas on how this worked when the PBX was hooked
straight to the PSTN?
I seem to end up answering these 'PoS legacy PBX' questions, so here goes...
You have handsets connected to your proprietary PBX. Most domestic
things you dial on your proprietary PBX handsets get passed directly
through to your asterisk box without getting mangled by your
proprietary PBX. International calls that are prefixed by 011 are
getting mangled by your proprietary PBX. Are you already getting to
what I'm going to suggest?
Modify your proprietary PBX to not mangle your international calls.
Asterisk is doing what its told when it gets a proper number to dial,
as you demonstrated by your extra 011 padding work-around. Your
problem is not with Asterisk, your problem is with your PBX. You could
even have your workaround be to buy a VoIP hard phone, hook it to your
Asterisk, and have people dial internationally with that phone. Then
buy some more VoIP hard phones, and stop buying any more handsets for
your proprietary PBX. Do that a few more times, then put your
proprietary PBX on eBay.
Problem solved ;)
___
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Re: [asterisk-users] Audio Files
- Original Message - From: Julien Claassen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 26, 2008 8:03 PM Subject: Re: [asterisk-users] Audio Files Hi! I think all - at least all PSTN - calls have the same quality in means of bitrate, number of channels and samplerate. It's 8kHz, 16bit and mono. About noise, I didn't have problems with that. Seems it's not really about quality. Probably it would be helpful, if you tell us, which extensions/protocol you used. Kindest regards Julien Well, I had installed the sample with gmake, and I add my own extension, exten = 269544,1,dial(Sip/user1,20) exten = 269544,2,hangup() and exten = 269544,1,dial(Sip/user2,20) exten = 269544,2,hangup() exten = 1,1,Playback(Wellcome) exten = 1,2,hangup() So, When I call from user1 to user2, have noise, If I call from user1/user2 to extension 1 the Playback have noise to. but, If I call to inexitent extension like the asterisk reproduced a error sound and not have noise.. What's is wrong?? Abel ___ -- 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 on VMware Workstation 6
One option might be to run in the opposite vmware direction. That is, run Linux as the native OS and run Windows within a vmware instance. That gives you the Windows compatibility for your applications, while at the same time providing the critical hardware timing for your Asterisk instance. I second this idea. You don't want Asterisk down because it's Patch Tuesday again. While you're at it, use VMWare Server 2.0, which doesn't cost anything more than Player and might work better. ___ -- 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] iPhone Sip App
Has anyone seen or know of a iphone/ipod sip client that may be in the works? ___ -- 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] Split incoming call volume across queues on several asterisk servers
Hi, I was wondering if there is anyway to split, say, 300 calls that come in from the SIP provider across 10 asterisk servers with 30 agents each, without having the telco do the splitting. Is there any way to do call distribution, e.g. we send an incoming call to a similar queue on the next asterisk server if all agents on the first asterisk server are busy and the queue already has a certain number of calls in it? Thanks, -- Dr. Haider Raza ___ -- 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] Split incoming call volume across queues on several asterisk servers
You can set up a proxy to round-robin/load-balance the incoming calls across three servers. If you need to do this with a view to queue utilisation, an outside process can be set up to mediate this via the Manager API and provide this information to the proxy process in real time. A proxy can also be set up to roll calls over to another Asterisk server if that server returns an error status code because all the agents are unavailable, such as 486 Busy or temporarily unavailable. You can, also, of course, do this in the Asterisk dial plan itself - fiddle with the timeout values on the Queue() app. However, in this paradigm, the first Asterisk box is going to have to cross-connect the call to others in the series, in a daisy chain. But if you can avoid media handling in such scenarios (i.e. use re-INVITEs), that shouldn't be too bad. Haider Raza wrote: Hi, I was wondering if there is anyway to split, say, 300 calls that come in from the SIP provider across 10 asterisk servers with 30 agents each, without having the telco do the splitting. Is there any way to do call distribution, e.g. we send an incoming call to a similar queue on the next asterisk server if all agents on the first asterisk server are busy and the queue already has a certain number of calls in it? Thanks, -- Dr. Haider Raza ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] iPhone Sip App
- Forrest Beck [EMAIL PROTECTED] escribió: Has anyone seen or know of a iphone/ipod sip client that may be in the works? http://www.voip-info.org/wiki/view/Apple+iPhone+%252FiPod+Touch+and+SIP+:+SIPHON Regards, -- Guillermo Salas M. Telconet S.A. Calle 15 y Avenida 24 Esq Edificio Barre #2 Primer Piso Telefono : +593 5 262 7815 Celular : +593 9 985 5138 International : +1 360 968 1701 e-mail: [EMAIL PROTECTED] www : http://www.manta.telconet.net SIP : [EMAIL PROTECTED] Linux User: 255902 Beat me, whip me, make me use Windows! Please avoid the Top Posting, see http://es.wikipedia.org/wiki/Top-posting ___ -- 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] iPhone Sip App
I have used RF.com with my iPhone. Works well. Sent from my iPhone Eric Moniz On Sep 26, 2008, at 10:11 PM, Forrest Beck [EMAIL PROTECTED] wrote: Has anyone seen or know of a iphone/ipod sip client that may be in the works? ___ -- 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] Split incoming call volume across queues on several asterisk servers
But will this allow the proxy to handle a load of 300 simultaneous calls? I mean will the calls be sent off to other asterisk servers and the proxy be left load-free to route new calls? -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 10:37 PM, Alex Balashov [EMAIL PROTECTED]wrote: You can set up a proxy to round-robin/load-balance the incoming calls across three servers. If you need to do this with a view to queue utilisation, an outside process can be set up to mediate this via the Manager API and provide this information to the proxy process in real time. A proxy can also be set up to roll calls over to another Asterisk server if that server returns an error status code because all the agents are unavailable, such as 486 Busy or temporarily unavailable. You can, also, of course, do this in the Asterisk dial plan itself - fiddle with the timeout values on the Queue() app. However, in this paradigm, the first Asterisk box is going to have to cross-connect the call to others in the series, in a daisy chain. But if you can avoid media handling in such scenarios (i.e. use re-INVITEs), that shouldn't be too bad. Haider Raza wrote: Hi, I was wondering if there is anyway to split, say, 300 calls that come in from the SIP provider across 10 asterisk servers with 30 agents each, without having the telco do the splitting. Is there any way to do call distribution, e.g. we send an incoming call to a similar queue on the next asterisk server if all agents on the first asterisk server are busy and the queue already has a certain number of calls in it? Thanks, -- Dr. Haider Raza ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Split incoming call volume across queues on several asterisk servers
Proxies do not handle media, so, one can definitely handle 300 simultaneous calls. Haider Raza wrote: But will this allow the proxy to handle a load of 300 simultaneous calls? I mean will the calls be sent off to other asterisk servers and the proxy be left load-free to route new calls? -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 10:37 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You can set up a proxy to round-robin/load-balance the incoming calls across three servers. If you need to do this with a view to queue utilisation, an outside process can be set up to mediate this via the Manager API and provide this information to the proxy process in real time. A proxy can also be set up to roll calls over to another Asterisk server if that server returns an error status code because all the agents are unavailable, such as 486 Busy or temporarily unavailable. You can, also, of course, do this in the Asterisk dial plan itself - fiddle with the timeout values on the Queue() app. However, in this paradigm, the first Asterisk box is going to have to cross-connect the call to others in the series, in a daisy chain. But if you can avoid media handling in such scenarios (i.e. use re-INVITEs), that shouldn't be too bad. Haider Raza wrote: Hi, I was wondering if there is anyway to split, say, 300 calls that come in from the SIP provider across 10 asterisk servers with 30 agents each, without having the telco do the splitting. Is there any way to do call distribution, e.g. we send an incoming call to a similar queue on the next asterisk server if all agents on the first asterisk server are busy and the queue already has a certain number of calls in it? Thanks, -- Dr. Haider Raza ___ -- 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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Split incoming call volume across queues on several asterisk servers
I guess what I want to ask is...how do I setup a proxy? In a nutshell...how are calls transfered or handed off to other asterisk servers leaving the originating server free from all call handling once the transfer is done. What dialplan command would do that? Do I setup a trunk and then Dial the call to the trunk? Maybe write an agi script to connect to manager interfaces on the different asterisk servers to see who has a spot free on their queue and then transfer on a trunk. I guess what I am not clear on is, are IAX trunks between asterisk servers what I need to accomplish this (Using a proxy or daisy chained asterisk servers)? -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 11:36 PM, Alex Balashov [EMAIL PROTECTED]wrote: Proxies do not handle media, so, one can definitely handle 300 simultaneous calls. Haider Raza wrote: But will this allow the proxy to handle a load of 300 simultaneous calls? I mean will the calls be sent off to other asterisk servers and the proxy be left load-free to route new calls? -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 10:37 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You can set up a proxy to round-robin/load-balance the incoming calls across three servers. If you need to do this with a view to queue utilisation, an outside process can be set up to mediate this via the Manager API and provide this information to the proxy process in real time. A proxy can also be set up to roll calls over to another Asterisk server if that server returns an error status code because all the agents are unavailable, such as 486 Busy or temporarily unavailable. You can, also, of course, do this in the Asterisk dial plan itself - fiddle with the timeout values on the Queue() app. However, in this paradigm, the first Asterisk box is going to have to cross-connect the call to others in the series, in a daisy chain. But if you can avoid media handling in such scenarios (i.e. use re-INVITEs), that shouldn't be too bad. Haider Raza wrote: Hi, I was wondering if there is anyway to split, say, 300 calls that come in from the SIP provider across 10 asterisk servers with 30 agents each, without having the telco do the splitting. Is there any way to do call distribution, e.g. we send an incoming call to a similar queue on the next asterisk server if all agents on the first asterisk server are busy and the queue already has a certain number of calls in it? Thanks, --Dr. Haider Raza ___ -- 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 --Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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] Split incoming call volume across queues on several asterisk servers
Asterisk is not a SIP proxy. You would have to use another piece of software, such as Kamailio/OpenSIPS (formerly OpenSER). Haider Raza wrote: I guess what I want to ask is...how do I setup a proxy? In a nutshell...how are calls transfered or handed off to other asterisk servers leaving the originating server free from all call handling once the transfer is done. What dialplan command would do that? Do I setup a trunk and then Dial the call to the trunk? Maybe write an agi script to connect to manager interfaces on the different asterisk servers to see who has a spot free on their queue and then transfer on a trunk. I guess what I am not clear on is, are IAX trunks between asterisk servers what I need to accomplish this (Using a proxy or daisy chained asterisk servers)? -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 11:36 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Proxies do not handle media, so, one can definitely handle 300 simultaneous calls. Haider Raza wrote: But will this allow the proxy to handle a load of 300 simultaneous calls? I mean will the calls be sent off to other asterisk servers and the proxy be left load-free to route new calls? -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 10:37 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You can set up a proxy to round-robin/load-balance the incoming calls across three servers. If you need to do this with a view to queue utilisation, an outside process can be set up to mediate this via the Manager API and provide this information to the proxy process in real time. A proxy can also be set up to roll calls over to another Asterisk server if that server returns an error status code because all the agents are unavailable, such as 486 Busy or temporarily unavailable. You can, also, of course, do this in the Asterisk dial plan itself - fiddle with the timeout values on the Queue() app. However, in this paradigm, the first Asterisk box is going to have to cross-connect the call to others in the series, in a daisy chain. But if you can avoid media handling in such scenarios (i.e. use re-INVITEs), that shouldn't be too bad. Haider Raza wrote: Hi, I was wondering if there is anyway to split, say, 300 calls that come in from the SIP provider across 10 asterisk servers with 30 agents each, without having the telco do the splitting. Is there any way to do call distribution, e.g. we send an incoming call to a similar queue on the next asterisk server if all agents on the first asterisk server are busy and the queue already has a certain number of calls in it? Thanks, --Dr. Haider Raza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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/ http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now:
Re: [asterisk-users] Split incoming call volume across queues on several asterisk servers
I will now look into reinvites and openser. Thank you so much for your time and all the excellent advice. -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 11:59 PM, Alex Balashov [EMAIL PROTECTED]wrote: Asterisk is not a SIP proxy. You would have to use another piece of software, such as Kamailio/OpenSIPS (formerly OpenSER). Haider Raza wrote: I guess what I want to ask is...how do I setup a proxy? In a nutshell...how are calls transfered or handed off to other asterisk servers leaving the originating server free from all call handling once the transfer is done. What dialplan command would do that? Do I setup a trunk and then Dial the call to the trunk? Maybe write an agi script to connect to manager interfaces on the different asterisk servers to see who has a spot free on their queue and then transfer on a trunk. I guess what I am not clear on is, are IAX trunks between asterisk servers what I need to accomplish this (Using a proxy or daisy chained asterisk servers)? -- Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 11:36 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Proxies do not handle media, so, one can definitely handle 300 simultaneous calls. Haider Raza wrote: But will this allow the proxy to handle a load of 300 simultaneous calls? I mean will the calls be sent off to other asterisk servers and the proxy be left load-free to route new calls? --Dr. Haider Raza BM 5203 3508 North West 114 Av. Doral, Florida 33178 Mobile+(809)-659-0623 On Fri, Sep 26, 2008 at 10:37 PM, Alex Balashov [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: You can set up a proxy to round-robin/load-balance the incoming calls across three servers. If you need to do this with a view to queue utilisation, an outside process can be set up to mediate this via the Manager API and provide this information to the proxy process in real time. A proxy can also be set up to roll calls over to another Asterisk server if that server returns an error status code because all the agents are unavailable, such as 486 Busy or temporarily unavailable. You can, also, of course, do this in the Asterisk dial plan itself - fiddle with the timeout values on the Queue() app. However, in this paradigm, the first Asterisk box is going to have to cross-connect the call to others in the series, in a daisy chain. But if you can avoid media handling in such scenarios (i.e. use re-INVITEs), that shouldn't be too bad. Haider Raza wrote: Hi, I was wondering if there is anyway to split, say, 300 calls that come in from the SIP provider across 10 asterisk servers with 30 agents each, without having the telco do the splitting. Is there any way to do call distribution, e.g. we send an incoming call to a similar queue on the next asterisk server if all agents on the first asterisk server are busy and the queue already has a certain number of calls in it? Thanks, --Dr. Haider Raza ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com 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/ http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 --Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ --
