[asterisk-users] RTP IP re-write
I am having a problem trying to get a particular softphone working on my setup. The machine it runs on has more than one interface. When the softphone registers, it registers fine, and asterisk is given the correct IP for registration. Whenever RTP is set-up however, the client gives the wrong IP to connect to and I get the inevitable problem with one-way media. Is there any way of forcing that SIP account to have the rtp always sent to a particular IP. (I know that this still may not work, because the device is probably listening on the wrong interface as well, but it's worth a try). I haven't been able to get a response from the vendor of the softphone. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP IP re-write
Joshua Colp wrote: Thomas Kenyon wrote: I am having a problem trying to get a particular softphone working on my setup. The machine it runs on has more than one interface. When the softphone registers, it registers fine, and asterisk is given the correct IP for registration. Whenever RTP is set-up however, the client gives the wrong IP to connect to and I get the inevitable problem with one-way media. Is there any way of forcing that SIP account to have the rtp always sent to a particular IP. (I know that this still may not work, because the device is probably listening on the wrong interface as well, but it's worth a try). It's not possible to do this as you describe but if you set nat=yes the RTP module will lock on to the source of the incoming media after a certain number of packets. This does require that the softphone send packets to Asterisk and that they make it, of course. Cheers, Thanks, works perfectly :-) I should have known that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Peculiar problem with failover provision.
I have noticed a peculiar problem recently with the way that the failover operates in my dialplan. I normally have: 1,Dial(SIP/provider-1/extension) n,Dial(SIP/provider-2/extension) (or something similar). This has up until now worked flawlessly. If there is an error with the first provider, the call is completed with the second one. Now, what is happening is, if the remote party hags up first, then the call progresses to the next priority and re-dials them. Is this a change in default behaviour? Do I need to add a particular flag / config directive to my dialplan I am running Asterisk 10.6.0. Thanks for any help in solving this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Peculiar problem with failover provision.
Eric Wieling wrote: You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on the internet do it the same way. It is still wrong. When you do a Dial on the dialplan you need check the value of DIALSTATUS or HANGUPCAUSE before dialing again. Both variables will give you some indication of why the first call ended. Then your dialplan logic can decide how to proceed. Thanks for your help. In previous versions of asterisk it worked, and iirc after the called party hung up, the dialplan only progressed if there was a particular flag used with Dial (g?). It's going to cause a heck of a headache but I'll look into doing this properly in the week. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon Sent: Monday, September 24, 2012 7:00 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peculiar problem with failover provision. I have noticed a peculiar problem recently with the way that the failover operates in my dialplan. I normally have: 1,Dial(SIP/provider-1/extension) n,Dial(SIP/provider-2/extension) (or something similar). This has up until now worked flawlessly. If there is an error with the first provider, the call is completed with the second one. Now, what is happening is, if the remote party hags up first, then the call progresses to the next priority and re-dials them. Is this a change in default behaviour? Do I need to add a particular flag / config directive to my dialplan I am running Asterisk 10.6.0. Thanks for any help in solving this. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for SIP
On 15/6/10 06:22, Randy R wrote: By the way, I am currently testing this product from Skype. I would like to be able to receive calls ona Skype name on our pbx. 1) It works beautifully and you don't have to do anything in particular. 2) It's disproportionally expensive which is why I want Skype for Asterisk to work. SfS costs $5 per month per channel just to test the beta! I find that insane, but I wanted to test it. In October, they will begin charging for Skype Manager (required for SfS) and a per seat charge for that. SfA also requires Skype Manager, and only works with users that were created with it. (At Skypes insistance afaict). The only architectures supported by SfA at the moment are x86 and x86-64. Also afaik, video still doesn't work with it. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Phishing attempt posing as digium
Did anyone else just get what looks like a phising attempt pretending to be from digium? It appears to be full of links to http://app.en25.com/e/er.aspx I must admit, it looks genuine. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
Jeff LaCoursiere wrote: On Fri, 15 Jan 2010, Hans Witvliet wrote: If you connect your pc with GB-lan card to an dual-ported ip-phone, you and up with an 100Mbps lan connection to your pc. Only way to avoid that, is to insert a cheap second lan-card in your pc, and connect your phone to the second lan, so your pc will act as an switch, instead of your phone... I'm curious - how have you managed to connect a second LAN card and have it bridge your (presumably onboard) ethernet? Does Windows have such capability? Right click on the interface and choose bridge connections. But I guess the OP was running XUbuntu, and though relatively complicated I guess you could get it to do that. Not all that complicated. IIRC it's just. brctl addbr br0 brctl addif eth0 brctl addif eth1 Then configure br0 as your interface. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about g729
Tilghman Lesher wrote: On Tuesday 01 December 2009 19:10:08 Alex Balashov wrote: All calls. Landy Landy wrote: You only need to purchase 10 licenses, if all 10 clients will be making calls at the same time. Ok. Does this apply only for outbound calls using a voip provider and/or applies to calls within the lan? An additional clarification: it only applies to calls in which codecs need to be transcoded. If you have a g729 call bridged to another g729 call, then no license is used in that call path. Also, the only consideration, isn't the endpoints. If the call is being recorded or you are in a conference, then the call needs to be transcoded for mixing purposes. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
Kevin P. Fleming wrote: Lee Howard wrote: I've heard of people who go to casinos and come home with a couple thousand bucks winnings, too. But the truth is that invariably the vast majority of people who gamble don't win. http://hylafax.sourceforge.net/docs/fax-over-voip.pdf Everyone wants to see if they're lucky. The smart ones, however, don't trust luck. FAX over SIP does not necessarily mean FAX over VOIP. FAX over SIP can also describe T.38, which is not as much of a gamble as FAX over VOIP :-) True, although I've yet to find a provider in this country (UK) that supports T.38. He may be better off porting the number to a fax2email service (although ime they are worth play testing first before you put any real work on them, eg. recently I've found one that doesn't support Fine Print or higher res faxes). AFAICT, to get a (real) fax machine using T.38, you either need to buy one that already supports it (never seen one, but I am assured they exist), Buy an ATA that supports it, or move to callweaver. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 and Fax
Dan Journo wrote: How do these fax2email providers run their service? I've not the faintest Idea, the provider I use afaict outsource it. Do they all use physical lines rather than use the internet? Thanks Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problems using chan_sebi and Huawei E169G
Martin Stubbs wrote: Hi, If I connect to the USB modem with minicom and issue the ATDxxx; command with a semicolon at the end to signify a voice call I get the same error response. Could someone else with this type of USB modem tell me if that command should work in minicom? I had exactly this problem with a 3UK E169, in the end after trying a million and one things (including crossflashing to various different providers), I replaced it with an unlocked Vodafone UK one. (apparently vodafone ES ones work fine as well, I also have what I think is a german vodafone one, that works). Thanks for the patch, I also have a little addition that allows the dongle to roam. (very simple change but essential on the network I am using it on). Do you know if there's anywhere set up that people can collaborate on this? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Should digium build a 2FXO / 2FXS 4-port daughterboard?
Gordon Henderson wrote: On Wed, 9 Sep 2009, Alec Davis wrote: Definitely, 10 votes from me. For the home user, 2xFXO + 6FXS, in a single slot small profile box is ideal, but only able to offer 2xFXO + 4xFXS at the moment. SIP phones don't exactly have the appropriate WIFE factor. A standard off the shelve, no frills phone does the job. You're kidding, right? Or maybe my wife is just more demanding... 6 FXS's at home? Ye gods, think of the wiring! We went DECT 8 years ago and haven't looked back... I do confess to having 1 FXS port though - we have an old BT rotary dial phone which sits in the 70's corner, along with a lava lamp... Gordon I must just be a freak then, at home I have 4 SIP phones (excluding unused ones), and a linecard with 6xFXS and 1xFXO (admittedly I only use 4 of the FXS ports). Hmm, I wonder how cheap the PA6488-based phones will be. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Should digium build a 2FXO / 2FXS 4-port daughterboard?
Gordon Henderson wrote: Wire free house - well the living area. The office is a different matter! I like wires, even prefer them. Just so long as no-one can see them. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Skype for Asterisk???
Julian Lyndon-Smith wrote: Nope - but you are also running on an unsupported version of asterisk, so I am not surprised. From the readme: ===[ Installation Overview ]=== It is required that the proper version of Asterisk is installed prior to installing Skype For Asterisk. Skype For Asterisk is currently supported on: Asterisk 1.4 versions = 1.4.25 Asterisk 1.6.0 versions = 1.6.0.6 Asterisk 1.6.1 versions = 1.6.1.5 Ah didn't spot that, if you are running 1.6.1, you need a version that isn't available yet. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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-announce] Asterisk project changes Music-On-Hold provider
Asterisk Development Team wrote: As posted on blogs.digium.com today: http://blogs.digium.com/2009/08/18/asterisk-music-on-hold-changes/ the Asterisk project has changed providers for Music-On-Hold (MOH) content distributed with/for Asterisk. In addition to the change for future Asterisk releases, we have also opted to rebuild historical releases with the new MOH content, in an effort to eliminate unnecessary distribution of the old MOH content. Great to hear, although I am a bit suspicious, the asterisk-sounds package in http://downloads.asterisk.org/pub/telephony/asterisk/releases/ still has a Mar 06 timestamp. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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-announce] Asterisk project changes Music-On-Hold provider
Please ignore my stupid reply to this, I was having issues with weasles at the time. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Skype for Asterisk???
Michael Graves wrote: I wonder if that was not a codec specific issue, but rather the matter of their license to the p2p technology provided by JoltID? Since Skype has recently dveloped their own codec (SILK) they could easily drop support for any codec that they previously licensed from outside. I think that the failure to ge a new license on a codec would not be a major issue for them. Failure to renew the license on the p2p transport technology is a much more significant problem. Michael That's probably what it was, It does appear to be trying to remove Jolt support. http://www.theregister.co.uk/2009/07/31/skype_joltid/ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Skype for Asterisk???
Geoff Lane wrote: On Tuesday, August 18, 2009, Gordon Henderson wrote: I was under the impression that Three (who I guess you're using) placed a regular call over their network then Skyped it at their HQ - rather than have the Skype client actually reside in the handset.. (And I'm suspecting their 3G limitation is that they want to use their own 3G network rather than pay Orange for the call over their 2G network) I am using Three. At first I got two Three S1 Skypephones and was allegedly one of the first business customers to take up those phones. One handset was replaced under warranty for a basic Nokia (can't remember the model number) that offered Bluetooth and could run the Skype application. In both cases, AIUI you run the Skype client on your handset, which uses 3G/HSDPA data bandwidth to connect to Skype via a NAT router in Three's network. FWIW and IMO the S1 is rubbish. However some tell me that the S1 is part of the problem I have using Skype and that the S2 is considerably better. It's kinda a mixture of both, the client on the handset sets up the call that is a regular voice call through their gateway. A bit (although not much) like the way international roaming SIMS make calls. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Skype for Asterisk???
Casey Boone wrote: I would have happily bought 20 channels at $10/channel, but at most will be buying only a single channel now :\ That does sound a bit pricey, although it it's as stable as the latest beta, I wont be buying it at all. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Skype for Asterisk???
Michael Graves wrote: Pricing is a very legitimate way to minimise support effort. It winnows down the market size to a point where the company offering the goods can sustain the projected per user support issues. You can always drop the price later on when you have a better handle on the per user support issue. Michael You make it sound like you're saying it's expensive because it doesn't work :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie: How to find the serial number ofDigium card?
Lee, John (Sydney) wrote: Thanks Tilghman. I learnt it the hard way - I never imagined I need to jot down the serial number of a PCI card :-( I've had a linecard that's been unregistered now for 4 years or more, because it's in a production server. It does of course mean that I didn't get any HPEC licenses. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Question: How to contribute to Asterisk-addons
uzuki Hironobu wrote: Hi, I am a beginner who began to use Asterisk in this July. Last week, I made two addons for PostgreSQL (cdr_addon_postgresql.c and res_config_postgresql.c), because I use not usual MySQL but PostgreSQL. Err, cdr_pgsql and res_config_pgsql are part of the main asterisk tree anyway. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Host-ID.
I'm about to change the motherboard in my server machine, (Different chipset). The most notable thing that will change, is the onboard network card (eth2) will be an atheros one instead of realtek. If I change the mac address of eth2 to read the same as the old one, will my host-id stay the same? TIA for any help with this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Host-ID.
Danny Nicholas wrote: AFAIK, host-id is tied to ip address and linux uname, so that's all that should matter. It's definately not tied to uname, otherwise it'd change every time I built a new kernel. Basing it on IP address would be extremely foolish, since most people use one of 3 ranges for their internal network with servers generally being .1-10 or .250-254, and for external connections too many people are on dynamic IPs. It is appears to be tied to the adapter address of eth0, I just don't know if the adapter addresses of other interfaces make a difference. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thomas Kenyon Sent: Friday, August 07, 2009 3:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Host-ID. I'm about to change the motherboard in my server machine, (Different chipset). The most notable thing that will change, is the onboard network card (eth2) will be an atheros one instead of realtek. If I change the mac address of eth2 to read the same as the old one, will my host-id stay the same? TIA for any help with this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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 2009 - October 13 - 15 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 2009 - October 13 - 15 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] Host-ID.
Danny Nicholas wrote: Editing my original comment, linux uname should have been linux hostname. Tilghman, can you elaborate a bit more? It's definitely not based on that either since changing your hostname doesn't change your Host-ID. In case anyone was wondering, I changed the adapter address on the new board so that it matched the old one and got udev to make sure it had the same name. Then started asterisk and my licenses were in tact. I didn't check what the host-Id was before doing this though. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Friday, August 07, 2009 11:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Host-ID. On Friday 07 August 2009 10:11:23 Thomas Kenyon wrote: Danny Nicholas wrote: AFAIK, host-id is tied to ip address and linux uname, so that's all that should matter. It's definately not tied to uname, otherwise it'd change every time I built a new kernel. Basing it on IP address would be extremely foolish, since most people use one of 3 ranges for their internal network with servers generally being .1-10 or .250-254, and for external connections too many people are on dynamic IPs. It is appears to be tied to the adapter address of eth0, I just don't know if the adapter addresses of other interfaces make a difference. Yes, it's based on all of them, and they should always present in the same order. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Skype for Asterisk: Public Beta available
Pascal Bruno wrote: Well I think thats what the problem was, I dont have it named as eth0. So if your NIC is not labeled eth0 you cannot use skypeforasterisk??? Why cant it just scan you nic handles? Can someone point me to where I can change the NIC name in the source file or something??? I don't know about centos, but in debian the file /etc/udev/rules.d/70-persistent-net.rules decides which interfaces are named what. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Skype for Asterisk: Public Beta available
Pascal Bruno wrote:
Unfortunately for me, I cannot register my license. Kept saying:
Could not generate Host-ID.
Make sure that you have eth0 enabled.
Any help would be appreciated
It uses the same licensing scheme as the G.729 licenses (so as soon as
you need to upgrade the machine, or set up LACP or VPN or any other type
of virtual interface or in the case of G.729 you change the codec to a
newer version {since you've upgraded to a new version of asterisk that
doesn't support older ones} that doesn't support the old name for the
codec, you need to re-register).
Or as in your case, it doesn't like the names of the network interfaces.
It's all a total PITA.
Fwiw, the Skype channel driver stopped working on my machine a while
ago. I never did track down the cause.
When res_skypeforasterisk starts, 39 res_skypeforasterisk processes
start and 1 skypewatcher service starts.
If I start it manually after asterisk has started, usually asterisk
segfaults, (not always).
Although Sometimes it starts up properly but can't log anyone in, Either
the user is stated as Logged Out or Connection Error, usually if I type
skype show users I get the following error message:
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
(Debian 5.0.2 x64 running kernel 2.6.30.2, asterisk 1.6.1.1 and
skypeforasterisk-1.6.1_0.9.10-x86_64)
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Re: [asterisk-users] Skype for Asterisk: Public Beta available
Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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] Latest chan_mobile
Carlos Ruiz Diaz wrote: @Steve: I considered the hardware separation between servers but when I exposed the idea it was immediately discarded because it is mandatory to have all in a box. Well, I'll start the migration then. Thank you. I doubt this helps anyone, but today I built the newest stable kernel (2.6.30.2) and the latest bluez libs (bluez-4.46) and obviously rebuilt dahdi and asterisk-addons. Without any config changes chan_mobile is working for incoming calls, picking up the handset is answeing the calls, and there is 2 way audio (which wasn't working before). Oddly when a call finishes, the mobile disconnects for a while and then reconnects again and there is terrible audio with outgoing calls, (scratchy and with a few seconds delay). This is definite progress (and doesn't require a separate box). This is all with a Cambridge Silicon Radio USB2 dongle and a nokia e61. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Latest chan_mobile
Carlos Ruiz Diaz wrote: That is exactly what happens to me. Still looking for a solution. Well, it's a step forward from what I was getting before. Have you tried with different USB adapters and handsets? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile help.
Sasa Bobek wrote: Could not agree more. I had chan_mobile up and running with an older version of Trix, but never managed to recreate it with the latest versions. Other people I talked to even suggested that it was made on purpose. With elastix the only problem I had was the missing mobile.conf.example, but you can create one from the Trix instructions from scratch or download it from the SVN. I've got a spare machine I can play with that on, I wish I could get it working on the server machine though. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Skype
DHAVAL INDRODIYA wrote: Hello All, can anybody tell me how can i integrate asterisk and skype users so that skype users can dial my asterisk number or dial internal dialplan form skype regars Dhaval Chan_celiax can apparently interface with a copy of the skype client running on the same machine, (I've not tried it so don't know how well it works). Other than that there is I gether an online SIP to Skype service (that someone will probably mention in a moment). As Alex suggests, digium are working on their own channel driver. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile help.
Razza wrote: I'm running centos, so tried a yum upgrade but nothing was marked for upgrade. I've reinstalled bluez-libs.i386 0:3.7-1.1. I've tried a different dongle, but still get the same message. I tried setting up chan_mobile again about a week ago. (admittedly last time I'd tried it, it had a different name and was using callweaver). I had exactly the same problems as the first time, No audio (although on one attempt I had 1 way audio, but strangely the phone and the deskphone and other mobile appeared to be conferenced together), answering desk phone didn't answer call on mobile, hanging up didn't hang up etc. Although I think I'm using bluez-4.40, the USB dongle is also a CSR one like the OP, asterisk 1.6.1.1, with a nokia e61. Oh a piece of advice, when the computer is automatically connecting to the phone, you can't find it by calling it :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
Jeff LaCoursiere wrote: On Sun, 28 Jun 2009, randulo wrote: On Sat, Jun 27, 2009 at 11:06 AM, Olivieroza-4...@myamail.com wrote: Hi, Has anyone tried it ? Is there any available pricelist ? It is possible no one wants to answer this due to the NDA they had to sign? Though they have written me back twice to say coming soon I am still waiting for the software... So you'd rather have it even when it hasn't been finished? I'm sure that as soon as it is complete and stable there will be pricing and availability announced. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [Asterisk-Users] Prices of g729 codec
Kevin P. Fleming wrote: - Chris Mason (Lists) li...@masonc.com wrote: licenses on them, usually $100 each time, and when I install the real hardware for the client, I can't transfer the licenses. If I scrap that Our support department is very accomodating when it comes to handling licensing issues like this; I'm surprised to hear that you can't transfer the licenses as we do that exact thing all the time. If you have a specific support ticket number where you requested this and it was declined, please email it to me off-list. I have a few lots of G.729 licenses and a digium branded linecard. I can't even email support, presumably since they were bought such a long time ago there is no longer a record that they were sold. (oh and I haven't registered the card, and haven't wanted to since it would involve taking down the machine it's in to find out the S/N.) (I have a 4 channel license for G.729 which I can't use with the most recent versions of codec_g729, presumably because the Product line calls it Digium-G729 rather than G.729 Codec. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38 support
Jay Ray wrote: Does asterisk support T38 passthrough now? What version onwards? I thought it came in at 1.6.0 . ANy ideas on how to configure it for a host? There are lots of guides to this on t'internet. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Grandstream, call pickup, ...
Peder wrote: Decent product, but their support and development are horrible. I showed them that their SIP over TCP implementation was broken and their reply was use udp Such a shame it sounds like it has gone down hill, previously when I've spoken to them the standard response was that they'll pass my comments on to the development team. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Grandstream, call pickup, ...
Doug wrote: Linksys looks good in comparison. I've found (in the past) linksys support to be quite good (although not used much of their voice products). Certainly they are in my experience much better than the kit they sell. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Fax Driver
Steve Underwood wrote: I've had a kinda-working-but-not-production-ready SIPmodem for ages, which does allow audio and T.38 from the same HylaFAX system, but I haven't found the time to complete it. Regards, Steve It's good to know that it's not been completely shelved, we are all grateful for your hard work. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] h extension and channel variables
Is there a method to fetch the ${EXTEN} of the channel that has been
hung up when exten h is started?
The nearest thing I can think of is to set another variable to the
extension and pick that up. Would that be a reliable method though?
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Re: [asterisk-users] h extension and channel variables
On 5/26/2009 10:57, Thomas Kenyon wrote:
Is there a method to fetch the ${EXTEN} of the channel that has been
hung up when exten h is started?
The nearest thing I can think of is to set another variable to the
extension and pick that up. Would that be a reliable method though?
Which is clearly a bad idea, since an intervening call would change this.
My Best idea so far is to change the CallerID to the exten (although it
may be desirable to keep it in tact, it's not as important in this case).
Does anybody have any suggestions?
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Re: [asterisk-users] h extension and channel variables
On 5/26/2009 14:08, Marco Sambo wrote:
I set a variable CalledID to ${EXTEN} before dial it. So in h extension
I can use ${CalledID}.
Thanks for the response.
In that case if there is an intervening call that is shorter, then the
$calledID will be wrong.
I found a better approach than using the h, extensions.
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Re: [asterisk-users] 1.6.0.9 sip.c: Serious Network Trouble ??
sean darcy wrote: I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 over this weekend. I'm getting: I asked the same thing just over a week ago and didn't get a response. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP error message
As of today, during startup I get lots of the following: ERROR[2704] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data Does anyone know what it means? This is with Asterisk 1.6.0.9. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma Wanpipe Driver Compile for DAHDI Failure
Tzafrir Cohen wrote: On Mon, May 04, 2009 at 07:55:04PM -0500, Atlanticnynex wrote: I don't like DAHDI anyway... even if it is just the name. Gets me confused with DUNDi and other fail acronyms. What's there not to like about DAHDI? It's a fun game: It doesn't support my linecard (I know it's not digium's fault) so I'm stuck using an old version of asterisk. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POS modems
Steve Underwood wrote: Hi, If anyone is interested in the low speed modems needed for POS applications (V.22, V.22bis, V.22bisFC and V.29FC) please contact me. I had some spare time while travelling, and finally got the V.22bis code I started a long time ago into a start where its basically functional. I'm now looking for input about exactly what application software expects from these modems, so I can plan the remainder of the code. Steve Does this mean that sipmodem is still on the back seat? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] POS modems
Steve Underwood wrote: Hi, If anyone is interested in the low speed modems needed for POS applications (V.22, V.22bis, V.22bisFC and V.29FC) please contact me. I had some spare time while travelling, and finally got the V.22bis code I started a long time ago into a start where its basically functional. I'm now looking for input about exactly what application software expects from these modems, so I can plan the remainder of the code. Steve Careful, soon you'll want to get your head round V.150.1 and we won't see you for months. Since none of the bank providers over here allow customers to use software EFTs, I can't think of a useful application for a v22 soft modem. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
Atis Lezdins wrote: Ok, our setup is the following: Inbound call arrives from SIP provider to Asterisk 1.4.19 Asterisk Dials Callweaver (1.2.0 as I recall) on localhost CallWeaver uses RxFax, which causes call to be switched to T.38, Asterisk does T.38 passtrough. CallWeaver executes shell script at the end which emails the .tiff file to recipient. User prints document to Hylafax Desktop client. Document is sent to Hylafax server Hylafax executes shell script specified in SendFaxCmd. Shell script creates callfile for CallWeaver CallWeaver dials destination number to Asterisk Asterisk forwards call to SIP operator CallWeaver uses TxFax to send .tiff file already generated by CallWeaver. As we are currently on stable 1.4 version, we chose to use CallWeaver for this, but we plan to simplify whole setup when migrating to Asterisk 1.6, which would take over CallWeaver functions. I will try this later, it looks straight forward enough. Does Asterisk 1.6 SendFax command autonegotiate T.38 (in the way callweaver does)? Can I use a faxmachine in a linecard to terminate to T.38 on a remote host in asterisk 1.6? (Well I suppose I can put it in it's own context, use ReceiveFax to make a tiff, and SendFax the image afterwards, but that does seem clumsy). I guess in Callweaver you'd use the T38Gateway command. Is there anything of interest in digiums new Fax For Asterisk software? (Or is it just a version of ReceiveFax and SendFax that doesn't rely on spandsp?) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T38modem in loopback mode does not work on asterisk 1.4.20.1
Atis Lezdins wrote: On Fri, Apr 17, 2009 at 4:31 PM, Michael mich...@networkstuff.co.nz wrote: The problem is that there is no reliable, or really any viable way to achieve this when using T.38 as the carrier uplink. Could You explain this? I really don't understand Your point. On voip-info there is a how to using T38modem. Congrats to anyone who can get it working. Well, i initially wrote that howto, after numerous hours of unsuccessful compilations and wrong versions, but T38modem didn't prove to work with our provider. Later, one Russian guy managet to get this working with he's provider. That may explain why I couldn't get it to work :-). I have accounts with 2 providers that are known to support T.38, the At the moment, the most notable error I'm getting is on startup t38 reports: error loading avcodec - avcodec: cannot open shared object file: No such file or directory That's why i put CallWeaver (which basically has the same T.38 stack as Asterisk 1.6) in it's place. I shold have another play with that, I have a box dedicated to it after all. The setup you describe does not have a audio data path connection to Hylafax and I wonder why the convoluted method when the same could be achieved using Callweaver alone and some custom scripting. Why would the audio data path would be necessary? In our setup CallWeaver effectively acts as modem, and talks T.38 with provider. Fax information data path to be pedantic. Data from Hylafax to CallWeaver is passed as TIFF image - thus no data/quality loss. How does this work? What do you use as the modem in hylafax? Regards, Atis ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP attacks
I have been receiving a lot of hack attempts today (home and work) multiple SIP registration requests (none of them managed to find a relevant username before fail2ban kicked in). Is this happening to a lot of people now? I only have SIP available externally for enum purposes, is it possible on a host which is specified as dynamic to choose a valid hostmask in sip.conf on a per peer/user basis? TIA for any response to this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP attacks
Tilghman Lesher wrote: Yes, you can use the permit/deny labels to specify an IP mask that is eligible to authenticate: deny=0.0.0.0/0 permit=192.168.0.0/16 permit=172.16.0.0/12 permit=10.0.0.0/8 By the way, after the slash, you can use either CIDR notation or a netmask. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please help test the gender detection module at 575-613-4392
Darren Wiebe wrote: Pretty cool. I'm almost offended though as I'm not usually guessed as a female of the species. :) I am a male and was detected as a male, so I'm feeling a bit left out. :-p ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] licensed g729
Michael Graves wrote: On Sun, 15 Feb 2009 10:23:50 +0800, Nhadie wrote: Hi All, If i buy 20 g729 and install to my asterisk, if 20 calls are already engaged using g729. would the next call then revert to using the other codec, in this case ulau and alaw? Yes, if you set the codec preferences this way. Allow both but prefer G.729. And presuming that the end-points do likewise. Since when? It certainly used to be that if you'd used up all the licenses, next call would still negotiate g.729 and they'd be no audio and lots of errors in the console. (albeit this was a long time ago). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi-linux 2.1.0.4 released
On 2/3/2009 17:34, Asterisk Team wrote: The Asterisk development team has released dahdi-linux 2.1.0.4 This release is available for immediate download from http://downloads.digium.com/pub/telephony/dahdi-linux. This release fixes a regression from dahdi-linux 2.1.0 in which it was possible for the kernel to panic when conferencing channels together. Ah, that explains it. :-) Please see http://bugs.digium.com/view.php?id=14183 for more information. The complete change log can be read at: http://downloads.digium.com/pub/telephony/dahdi-linux/releases/ChangeLog-2.1.0.4 Thanks for your continued support of Asterisk! I can't get this to build, the following error is produced: CC [M] /usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.o /usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.c: In function 'xproto_get': /usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.c:96: error: implicit declaration of function 'module_refcount' make[3]: *** [/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.o] Error 1 make[2]: *** [/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp] Error 2 make[1]: *** [_module_/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.28' make: *** [modules] Error 2 This is on a P4 machine with gcc-4.1.2, (mot sure what else to include really, DAHDI Tools 2.1.0.2, asterisk 1.6.0.3). TIA for any help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi-linux 2.1.0.4 released
Shaun Ruffell wrote: Thomas Kenyon wrote: I can't get this to build, the following error is produced: CC [M] /usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.o /usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.c: In function 'xproto_get': /usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.c:96: error: implicit declaration of function 'module_refcount' make[3]: *** [/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp/xproto.o] Error 1 make[2]: *** [/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi/xpp] Error 2 make[1]: *** [_module_/usr/src/asterisk/1.6/dahdi-linux-2.1.0.4/drivers/dahdi] Error 2 make[1]: Leaving directory `/usr/src/linux-2.6.28' make: *** [modules] Error 2 This is on a P4 machine with gcc-4.1.2, (mot sure what else to include really, DAHDI Tools 2.1.0.2, asterisk 1.6.0.3). TIA for any help. is CONFIG_MODULE_UNLOAD defined in your kernel config? No, does it need to be for building them then? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi-linux 2.1.0.4 released
Shaun Ruffell wrote: Thomas Kenyon wrote: No, does it need to be for building them then? It shouldn't, but the module_refcount function is only defined in kernels that are configured to allow module unloading. This probably needs a mantis issue to make sure the drivers build when the kernel is not configured to allow modules to unload. Thanks. Sorted now. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dahdi caused Kernel to segfault
Tzafrir Cohen wrote: On Mon, Jan 26, 2009 at 08:53:11PM +, Rony Ron wrote: Hi the same happened here also with different distros (ubuntu and fedora 9) each time i run dahdi start the kernel crash. What exactly do you mean by crash? An error, or the system completely crashes / hangs? I got both, very pretty. It's not happened since, but still a worry. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] the FXS ports of Digium and damaging if connected to Tel Line
Geoff Lane wrote: If you take care, it's possible to cut off the end of the tang that sticks out of the socket while leaving enough of the tang to lock the plug in place. You may not even need to clip the end off, with the last lot of RJ-11 plugs I ordered the tangs were short enough to snap in the socket on a TDM400P. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 codec
On 1/19/2009 12:03, michel freiha wrote: Dear All, I have the following CPU info on my asterisk server: Linux switch1.domain.net http://switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 2008 i686 i686 i386 GNU/Linux I need to install G729 on the asterisk server just to pass through and not for encoding...Which G729 package do you advice me to install? I tried several packages with no luck Regards If you only need it to be used as passthrough, you don't need any, just the format interpreter that comes with asterisk. It is worth noting that if you have conference calls, you use the Page function or want to record calls, then you will need to install a codec (since in these situations the call is transcoded inside asterisk). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 0800 UK number
Danny Nicholas wrote: Why are you using a text message when you could be recording a message and sending it out? This would possibly be clearer than a read-and-callback scenario? Do you think so? Remembering that most people, if they pick up the phone to hear a recorded message will immediately hang up without listening to it, then there's the few that will listen and not get it or understand, either because they have answered their phone in a crowded room expecting a real person to be on the end or because a lot of the time you don't hear a recorded message the first time you listen to it (even if you're expecting it to be recorded). I think a text message is a much more elegant way (presumably relayed though an SMSC so that sending the message to all users doesn't take a day to do). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dahdi caused Kernel to segfault
Yesterday, a low-duty production server that I maintain core-dumped. At the time there were only around 2 calls going through it. The strace on the screen made it look like it was caused by Dahdi. The machine is running asterisk-1.6.0.3 dahdi-linux-2.1.0.3 dahdi-tools-2.1.0.2 asterisk-addons-1.6.0 Kernel version 2.6.28 There is a genuine TDM400P (populated witrh 2xFXO cards and 2xFXS cards. Has anyone had a similar issue? This has only happened once, but I am a bit worried. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 0800 UK number
Julian Lyndon-Smith wrote: The only fly in the ointment is that my server is in 01702, but I need a local number (01376) for political reasons That's hardly a problem, (If the call is to be presented using VoIP) more or less any provider will give you a local number from another area. I have an 0800 number with voiptalk (an ex-BT number) and that seems to be pretty reliable. Although they did put the price up a year ago without telling me. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP URI: Allison Smith, Music-on-Hold Parody--outstanding.
Karl Fife wrote: Somebody requested a path to listen without termination charges. Here's a SIP URI: (a SIP What??) karlonh...@sip.kfife.com mailto:karlonh...@sip.kfife.com or 3605195...@74.92.179.65 mailto:3605195...@74.92.179.65 Thanks -Karl Very Funny :-) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy strangeness...
Geraint Lee wrote: Hello there... Noticed some strangeness going on with mixmonitor and chanspy, the called (External SIP) party seem to be responding before the calling party (Internal SIP) on call recordings and also when you listen in using chanspy. as far as the agent (calling party) is conserned the conversation is perfectly normal... just not the recordings that are produced, or any spying that's going on at the time. This is happening on mixmonitor recordings even if you're not listening in on chanspy too. Any suggestions? I don't have any suggestions, but this is similar to something I am experiencing with Chanspy in 1.4.21.1. If I spy on a call, then progressively throughout the call a delay is introduced. By the end of the call I can be listening to sound that is 10 seconds out of sync. (Then I don't get to hear the end of the call when the call is finished). This also leaves stale channels open. (the entry in show channels doesn't go away until the asterisk process is restarted). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phishing attempt
Matt Riddell wrote: On 6/11/2008 8:37 p.m., Thomas Kenyon wrote: Jeff LaCoursiere wrote: I didn't realize only 4% of the world's population lived in North America! Learn something every day. Sorry that was my bedtime maths, the figure is just over 4.5%. 4.5611893661578069635904176186202% To be slightly more accurate :) Or to be outright pedantic 4.5380853065046927068273204778542%. According to figures from the US census bureau for figures projected as being the start of this month and 8:39 GMT this morning respectively. World: 6,733,867,928 USA:305,588,671 According to google's figures (July 2007 est.) USA: 301,139,947 World: 6,602,224,175 Title: Census Bureau Home Page US Census Bureau Skip this top of page navigation FAQs Subjects A to Z Help SEARCH: Skip this left side navigation New on the Site Data Tools American FactFinder [EMAIL PROTECTED] Catalog Publications Are You in a Survey? About the Bureau Regional Offices Doing Business with Us Related Sites Skip this center section 2010 Census NewsBecome a Census Taker American Community Survey Census 2000 People Households Estimates Projections Housing Income | StateMedianIncome Poverty HealthInsurance International Genealogy More Business Industry Economic Census GetHelpwithYourForm EconomicIndicators NAICS Survey of Business Owners Government E-Stats ForeignTrade|ExportCodes LocalEmployment Dynamics More Geography Maps TIGER Gazetteer More Newsroom Releases Facts For Features MinorityLinks BroadcastPhotoServices Embargo/News Release Subscription More Special Topics Census Bureau Data and Emergency Preparedness CensusCalendar Training ForTeachers Students StatisticalAbstract FedStats USA.gov Skip this left side navigation Data Finders Population Clocks U.S. 305,588,703 World 6,735,039,780 08:43 GMT (EST+5) Nov 06, 2008 Population Finder Enable _javascript_ for access to this tool or see data available from the American Community Survey. city/ town, county, or zip or state Select a state Alabama Alaska Arizona Arkansas California Colorado Connecticut Delaware District of Columbia Florida Georgia Hawaii Idaho Illinois Indiana Iowa Kansas Kentucky Louisiana Maine Maryland Massachusetts Michigan Minnesota Mississippi Missouri Montana Nebraska Nevada New Hampshire New Jersey New Mexico New York North Carolina North Dakota Ohio Oklahoma Oregon Pennsylvania Puerto Rico Rhode Island South Carolina South Dakota Tennessee Texas Utah Vermont Virginia Washington West Virginia Wisconsin Wyoming Find An Area Profile with QuickFacts For the following combo box, to make a selection, press enter then alt plus down arrow and use the up and down arrows. Select a state to begin Select a state Alabama Alaska Arizona Arkansas California
Re: [asterisk-users] Phishing attempt
Thomas Kenyon wrote: Or to be outright pedantic 4.5380853065046927068273204778542%. I apologise for attaching the files, It was unintentional. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phishing attempt
John Todd wrote: It's a legitimate mail from Digium. Bit of a pooh survey. 1. Does your business use an Open Source PBX in North America? \ Err well, no, like 96% of the world, I don't live in North America. 2. Does your business plan to install an Open Source PBX? \ Err, No, only if we find something wrong with the existing Open Source PBX. Doesn't really tell you much, presumably you are only interested in opinions from people in North America. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phishing attempt
Jeff LaCoursiere wrote: I didn't realize only 4% of the world's population lived in North America! Learn something every day. Sorry that was my bedtime maths, the figure is just over 4.5%. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cheapest 4 port FXO
Gordon Henderson wrote: On Sat, 25 Oct 2008, Joseph L. Casale wrote: X100P. Yeah I saw these but they are single port and I need at least 2 ports. I only have 1 free pci slot as well. OpenVox. Gordon I don't know if this is still the case, but nxtvox used to be much cheaper, although I don't know if their 8-port version of the TDM400 will work with dahdi. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The skype channel...
Dean Collins wrote: You had to sign up with a form and Digium was going to get back to you. Don't know if they got back to people yet..but I didn't hear from them yet. I don't have the form url anymore. Regards, Dean Collins Judging by their emails, it looks like at least the first wave of beta testing has begun. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] The skype channel...
Julien Claassen wrote: Hi! So no way to get in anymore? that's too bad. It would have been the first real skype accessibility for blind people working like myself on linux. If anyone does remember or can retrieve the URL for the signup form, please tell me anyway, perhaps it's still possible. Kindest regards Julien Send em your details on the form on their website, there is more than one round of beta testing. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722 between Eyebeam and a Polycom IP650
Steve Underwood wrote: If I were building a terminal, I'd make mine announce 8000, but accept 8000 or 16000 to try to maximise compatibility. It seems people don't do that. Looking at debug output from 1.6 (using a grandstream), it looks like that is what it does. ___ -- 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] Transcoding G.729 files
Does anyone know of a utility I can use to transcode a group of files from G.729 format to something playable on a PC (GSM or WAV). I know I can convert them individually from the CLI, but I have quite a lot I need to do. ___ -- 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] Transcoding G.729 files
Alex Balashov wrote: SOX will do it if you install its G.729 format library. As far as converting a group of files, that's what scripting is for, i.e. for FILE in `find . -type f -name '*.g729'`; do NFILE=$(echo $FILE | sed 's/\.g729/\.wav/g') sox [some args] $FILE ... $NFILE ... done Thanks, didn't know sox could support g.729. ___ -- 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 1.6.0rc6 make menuselect failed.
Sean Bright wrote: Thomas Kenyon wrote: In trying to upgrade my test machine from 1.6.0beta9 to 1.6.0rc6 when I try to make menuseletc I get the following error. This is using gcc 4.1, libgtk 2.0, on an intel Core2Duo machine running an up to date Debian etch. Asterisk builds okay (not tried running it yet) menuselect_gtk.c: In function ârun_menuâ: menuselect_gtk.c:311: warning: implicit declaration of function Would you mind opening a bug in mantis (http://bugs.digium.com/) and include the config.log in your asterisk source directory as well as the one in the menuselect sub-directory as attachments to the bug? I've seen this problem crop up before and I would like to get it worked out. Thanks, Thanks, I've opened it, id 0013472 http://bugs.digium.com/view.php?id=13472 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.6.0rc6 make menuselect failed.
In trying to upgrade my test machine from 1.6.0beta9 to 1.6.0rc6 when I try to make menuseletc I get the following error. This is using gcc 4.1, libgtk 2.0, on an intel Core2Duo machine running an up to date Debian etch. Asterisk builds okay (not tried running it yet) menuselect_gtk.c: In function ârun_menuâ: menuselect_gtk.c:311: warning: implicit declaration of function âgtk_tree_view_set_enable_tree_linesâ menuselect_gtk.c:312: warning: implicit declaration of function âgtk_tree_view_set_grid_linesâ menuselect_gtk.c:312: error: âGTK_TREE_VIEW_GRID_LINES_BOTHâ undeclared (first use in this function) menuselect_gtk.c:312: error: (Each undeclared identifier is reported only once menuselect_gtk.c:312: error: for each function it appears in.) make[1]: *** [menuselect_gtk.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/rc/asterisk-1.6.0-rc6/menuselect' make: *** [menuselect/gmenuselect] Error 2 ___ -- 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] iLBC and G729 codecs
Edgar Guadamuz wrote: I notice that I have only format_ilbc.so but not codec_ilbc.so... is it due to the compilation or there is some way to create the module? That's the format interpreter, for the codec you need to select it in make menuselect befor ecompiling asterisk (subject to libraries). ___ -- 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] Reliable wireless SIP phones
Geraint Lee wrote: I've used several hitachi dmp330's they work great, roam between wireless access points with no loss of audio or connection for that matter. it will be a great shame if hitachi stop producing them, they are the most reliable wireless sip phones i've come accross... stay well away from pirelli phones, they are very buggy. Cheers Geraint I have a pirelli in use, I haven't had any complaints with them being buggy (was a pain in the arse to pair to an AP), the biggest issue the user has is that apparently the battery lasts less than an hour. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
Steve Repo wrote: I agree! I bought a GXP1200 (business class phone) and it's buggy. Can't use the message button (404 not found).. and some other features (404 not found). I have requested help from Grandstream and so far nothing. I've never heard of that problem, ar eyou sure the 404 response isn't coming from asterisk? (It works on all the GXP-2000's I have). The Lines all work, the BLF keys mostly work (It's better than not having any), The Transfer, Conference, Message (can have a separate mailbox setup for each line) etc. buttons all work. It's a shame they don't have a more standard headset port (like the Polycom). I havemn't experienced any problems with crashes or sound quality. (Although as I stated before, the G.722 codec isn't discernably clearer than the G.711 ones). I don't really think they test important features supported by their phones just the basic ones (dial out/dial in) and that about it :) Apparently there are problems with some hardware versions, (I have 3 different versions that have been okay) and most versions of the firmware. I'm on 1.1.6.16 and it's been fine. Don't see why it is worth spending almost twice the money to get a handset without features that (in the case of BLF) are needed. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
Michael Graves wrote: On Thu, 4 Sep 2008 00:25:12 +0530, Steve Repo wrote: I have a Grandstream GXP1200 and eager to try this codec. I've heard good things about the quality. Anyone tried it with asterisk? I can't until 1.6 is released. Steve I can't speak to Asterisk but I really like that codec. It's in my Polycom IP650s. I hope yours works better than it did on the BudgeTones. There, while the codec is supported, the hardware limits the call quality. Michael -- I can't speak for the GXP1200, but with the GXP2000s I have, you cannot tell the difference between G.711 and G.722. The only Polycom I have is an IP501 (that I bought to test, found to be solid, reliable, feature barren and awkward). Sadly this is very limited wrt its codec support. (supports even fewer than cheap chinese handsets). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G722 and Asterisk 1.6
Michael Graves wrote: On Wed, 03 Sep 2008 21:19:40 +0100, Thomas Kenyon wrote: I can't speak for the GXP1200, but with the GXP2000s I have, you cannot tell the difference between G.711 and G.722. The only Polycom I have is an IP501 (that I bought to test, found to be solid, reliable, feature barren and awkward). Sadly this is very limited wrt its codec support. (supports even fewer than cheap chinese handsets). Hardware phones usually support fewer codecs than soft phones, almost all support at least G.711u/a and G.729 families. I don't know about that, the Grandstream phones support 8 different codecs, the old chinese PA168-based phones supported 6, as do the newer AR1688-based handsets, the linksys SPA922 has 4, the SPA962 has 5, the old Snom 300 has 7 (including G.711). The IP501 has 3. I know it's a considerably better handset than say the PA168 or AR1688-based handsets, but then they are $45 and the 501 is £125. I'd also be more sold on it if it had half the features of the GXP2000 (which is only a little over half the price). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 - 1.6
Chris Maciejewski wrote: Hi, You can find some info about differences between 1.4 and 1.6 here: http://svn.digium.com/view/asterisk/branches/1.6.0/UPGRADE.txt?view=markup Kind regards, Chris Although reading the 1.4 UPGRADE.txt isn't a bad thing either, since all the things that were marked deprecated in 1.4 that hadn't been changed, will need to be changed for 1.6. ___ -- 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?
JR Richardson wrote: Asterisk 1.6 currently has T.38 origination and termination support. It does not yet have fax gateway support. -- Russell Bryant Russell, Can you please clarify what you mean. I think there is still a bit of confusion as to what termination and gateway and Asterisk 1.6 is all about, capability, functionality, call flow to what application, library requirements, spandsp versioning. And when do you think we can expect to see stable solutions for each. Thanks. JR Err, Origination would be the fax originates from the server and then gets sent as T.38. Termination would be an incoming T.38 signal comes in and gets interpreted at the server. and Gateway would be when you have a signal from a fax machine that the server then converts to a T.38 signal (as in the T38Gateway app in callweaver). Or am I missing something? ___ -- 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] Max amount of concurrent calls on and iax trunk
Mattias Andersson wrote: I agree bandwidth is the limit, however the reason to use IAX is it is saving bandwidth. I am runging 2 Trixbox CE with IAX over a 2 Mbit line. I have never had any isues with the IAX trunk. I wish that I could get son good IAX phones to the office, tan would we skip on Trixbox and run the phones directly over a VPN Chanel. SIP over VPN are giving more hassle then IAX sound wise. In this setup, there can be a bandwidth saving with keeping both the servers so that you can use trunking=yes in the trunk definition in iax.conf. ___ -- 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 1.4.21.1: Bugs in IAX
Michael J. Liberatore wrote: Are you sure your using 1.4.21.1 and not 1.4.21? I am pretty sure the major bug they fixed in .1 was the iax2 and cli bugs you listed below. Atleast they were supposed to fix it. 1.4.21.1 seems to have fixed the iax crash bug that I was experiencing in 1.4.21. The CLI prompt no longer returns when there is output to the console, but that's hardly a problem. There is a show-stopper in it for me though. From 1.4.19.1 to 1.4.21(.1) for some reason chan_alsa has stopped working. (which is used for an overhead PA). It reports Read Error: Resource Temporarily unavailable. I will revert back to 1.4.19.1 when it's out of use. ___ -- 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] Major problem with 1.4.21 asterisk
Steve Totaro wrote: Just out of curiosity, why did you feel they needed an upgrade? Thanks, Steve For me it was a bit of slapdash behaviour on my part. I'd hreceived a few complaints that sounded like they were probably asterisk' fault (most interesting one being that one person could hear in the background someone elses conversation who was in the next office in their call occasionally). Rather than looking through bugtracker to see if they'd been reported and then look into more detail at what was going on, I just updated to the latest release version to see if anything got magically fixed. (The new release cheme gives a bit of confidence that it is generally not a bad idea). Oops. ___ -- 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] Major problem with 1.4.21 asterisk
Michael J. Liberatore wrote: Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. thanks mike I seem to have exactly the same problem, have rolled back to 1.4.19.2 . Although on my machine I needed to kill -9 the process before it finally died. (process is launched by safe_asterisk). 1.6.0b9 (running at home) doesn't suffer this. ___ -- 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] Major problem with 1.4.21 asterisk
Thomas Kenyon wrote: Michael J. Liberatore wrote: Hi, i upgraded the other ay to 1.4.21 from 1.4.19 and started having major iax2 problems. All of a sudden calls wouldnt come in on the iax2 DID, and we couldnt make calls out even though everything looked ok. Also there was usually a hung iax2 channel when this happened. Stopping asterisk also wouldnt work, i would do a Stop now and it would just go back to the cli prompt. I would do a ? and it wouldnt work. I would have to kill asterisk via ps and then restart it via init.d and then iax2 would start working again for a short while (maybe a few hours) I reinstalled 1.4.19 and the problems went away. There appears to be a major bug in 1.4.21 but i am not sure. thanks mike I seem to have exactly the same problem, have rolled back to 1.4.19.2 . Although on my machine I needed to kill -9 the process before it finally died. (process is launched by safe_asterisk). 1.6.0b9 (running at home) doesn't suffer this. I forgot to mention that for the 10 to 20 minutes (at a time) asterisk 1.4.21 is working, chan_alsa also appears to have stopped working (well produces chan_alsa.c:693 alsa_read: Read error: Resource temporarily unavailable). ___ -- 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] Grandstream Busy Light Fields
Jan Prunk wrote: Hello ! I am having troubles setting up Busy Light Fields (BLF) in asterisk 1.4.18 The things work up to 80%, I can transfer the call by BLF button and I can see the green (free) status and red (busy) status. What I cannot do is to accept the call when someone rings a remote extension. The BLF button starts to blink in red telling me that the call is ringing on remote extenson, but if I press it, my phone starts ringing that extension instead of accepting the call. I can only acept the call from other sides if I enter *8 (send) button. Here I am sending an example from my sip.conf file (I have sip accounts from 60-90) : I may have it wrong, but I've found that PickUp can be a complete pain in the arse. In the same arrangement, I have an agi script called for (what in yours would be _**6X,1, which picks up a group of extensions (well all 4 lines on the [EMAIL PROTECTED] (internal queue context) and the extension number itself. This seems to work for incoming calls, but not internal calls. (Although I haven't really looked into this, I do know that PickUp doesn't work ifn certain circumstances, such as the extension being picked up using a Macro to which handles the dialling for instance). Now hopefully someone will come along and explain all the bits I got wrong. :-) ___ -- 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] MiixMonitor filename for queue calls.
Ed Nunez wrote:
I have found the answer to my question.
It's also worth noting (I'm sure you spotted it), That you have 2
priority 1 entries for 8484 in your extensions.conf.
extensions.conf
exten =
8484,1,Set(MONITOR_FILENAME=QUEUE-NOC-${CALLERID(NUM)}-${STRFTIME(${EPOCH)
exten = 8484,1,answer
exten = 8484,2,Queue(noi-noc)
___
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[asterisk-users] MixMonitor Not recording whole calls
I have calls being recorded via mixmonitor which are not being recorded in their entirety. The calls are incoming G.729 calls recorded in G.729 format (which I know means a lot of licenses, and a bit of runtime, but the load on the server isn't great and it does save disk space). They seem to stop recording if the call it placed on hold for an extended period of time. Does anyone know what is happening? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G.722?
Steve Underwood wrote: Michael Graves wrote: Which flavor of G.722 has been implemented in Asterisk? And starting with what release version? The only flavour with a defined RTP format is the full 64kbps one. Steve I was going to say strawberry, but to try to answer his other question, there is a codec as from release 1.6.0 (coming soon, if you look at the changelog there has been a lot of work on it reasonably recently), The latest 1.4.x release doesn't have a codec or format interpreter, It is listed as a codec, with no translation paths. It will probably work in passthrough mode in much the same way as H.264 video. (although I don't know this for sure). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium cards with sangoma cards
wassim darwish wrote: Hi: Iam an Asterisk user and i have a Sangoma A200 with 4 fxo modules and i want to buy Digium card with 4 fxo modules and insert it on the PCI besides the sangoma card ,so i will have 8 fxo channels on my asterisk box ,Is that right? Does Asterisk make errors if there is two different cards ? Thanks in advance; I know this sounds silly, and isn't what you are after. Why don't you buy a daughterboard with an additional 4 fxo modules for your existing A200? I don't know which option is cheaper, but upgrading the A200 has got to be better for IO bandwidth on your machine. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream
Phibee Network Operation Center wrote: I have a problem connecting a Grandstream ipphone to an asterisk. The ipphone is behind a nat router, I redirected UDP 5060 and 5004 to my phone. It connects well to the asterisk server. I can call outside and receive calls from outside without any problems. But if I call from this ipphone to another ipphone connected on the same asterisk server, using internal dialing, I can hear my correspondant, but he cannot. Do you have any idea? Thanks for advance. This is usually a NAT issue, what happens if asterisk is kept in the media path? (by setting canreinvite=no the peer declaration for your handset in sip.conf). ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Chanspy
Mike wrote: Hi, I`m trying to use Chanspy for a customer that wants to listen to his employees so he can train them better (or so he claims). In any case, it looks simple but there is something I`m not doing right. When a call is incoming, I set SPYGROUP using Set(SPYGROUP=1234) When I use, on another phone, Chanspy(|qg(1234)) I know it's unlikely, but could some of the dialplan changes from 1.6 have accidentally filtered backwards into the 1.4 tree? ie. Chanspy(|qg(1234)) becomes Chanspy(,qg(1234)) Unlikely I know, but probably worth a shot. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] g729 encoder/decoder
Peder @ NetworkOblivion wrote: That makes sense. A call from 729 to 711 would require one encoder and one decoder, right? So if you have 10 licenses, is it 10 total encoders+decoders, or 10 calls (some may require encode, or decode, or both)? Because I had 10 licenses, but my encoders+decoders was more than 10 and calls worked fine. However I also ran out of licenses when neither number was =10. 1 license = 1 encoder + 1 decoder. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Expected iax user behaviour
I have a provider which terminate using iax setting the phone number dialled as the username. If I have in iax.conf the following: (basically what I have with some names changed and codec stuff removed) [phone-number-1] type-user username=phone-number-1 context=incomingcontext [phone-number-2] type-user username=phone-number-2 context=incomingcontext [providername] type=friend username=providername secret=somepassword host=ip of host call is coming in from context=incomingcontext trunk=yes Then the incoming call is authenticated with the [phone-number-1] credentials and then gets recognised by asterisk as [providername] . Is this expected behaviour? (and a way of adding additional credentials to users). Will concurrent calls be properly trunked? If this works how it looks like it does, then it will make my life a lot simpler. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX user identification.
Is there a way to set up a user/peer in iax.conf where it matches incoming calls based entirely on IP? I have a provider that sets the username (as well as the extension) to the phone number that has been dialled, I'd prefer calls from that provider to all be identified as the same trunk. TIA for any help with this. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] which phones to use ??
Gordon Henderson wrote: Not just firmware, but hardware... The latest 1.1.5.15 Seems to be good - on newer phones, but if you've got older phones, then 1.1.1.14 is the one for you ... 1.1.5.15 seems to have improvements in sound quality, but there are still a few unresolved issues that made me roll-back to 1.1.4.25 . ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FXO Cards - T38
Fernando Berretta wrote: Tzafir, I'm sorry, my question wasn't clear. Apparently Asterisk 1.6.0b2 and b4 has support for t38 because of some modifications on app_fax so the questions are: 1 - If I use Asterisk 1.6.0b2 o b4 and a fax is received from a FXO Card and this FXO port is forwarded to other ATA/Gateway is asterisk going to transmit this fax using t38 ? Excuse my ignorance, but don't ATAs generally only support T.38 Origination? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
