Re: [Asterisk-Users] List of Motherboards or Servers that are testedok with Asterisk and Digium boards
Atari 600XL.16K ram...lol On 18 Nov 2005 at 19:00, [EMAIL PROTECTED] wrote: I went from a Vic20 to a CPC6128...both great items PaulH - Original Message - From: Julian Lyndon-Smith [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 18, 2005 6:49 PM Subject: Re: [Asterisk-Users] List of Motherboards or Servers that are testedok with Asterisk and Digium boards Man, looking back it was a gas - the 16k wobbly rampack. You spent 30 minutes looking at a blank screen whilst loading Horace goes skiing (or some other c*appy game you wanted to hack) making incantations to the tapedrive god in the vain hope that you wouldn't get an I/O error. It wasn't a gas then. ;) Always coveted a 48k Spectrum. Got a CPC6128 instead .. Julian. Matt Riddell wrote: Julian Lyndon-Smith wrote: Asterisk is cool. But maybe not that cool. Hey, don't you know that the dev team gets all the cool toys ;) You can tell I started coding on a ZX81. Woohoo go the ZX81!!! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 trunking wackyness
Hi I was doing some bandwidth testing, and my incomming usage is 36% more than my outgoing bandwidth. The setup is IAX2 trunking using GSM codec. Is there any obvious reason I am overlooking to figure out why there is such a big difference between the two.? I am using CVS-head September 3rd, maybe there is a version skew? Any suggestions will be appreciated. Thanks Clive ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2 trunking wackyness
On 21 Sep 2005 at 19:48, Matt Riddell wrote:
Clive wrote:
Hi
I was doing some bandwidth testing, and my incomming usage is
36% more than my outgoing bandwidth.
In my case the calls come in separately (i.e. untrunked) and get trunked by
the Asterisk machine and sent out. This causes an imbalance.
Are your calls coming from many to one or one to one?
--
Cheers,
Matt Riddell
Hi
My setup is: telco-asterisk(voip)-asterisk{ITSP}telco
so there should be an almost balanced transmit and receive rate on
the voip leg.
My suspicion is that perhaps the packets are not getting trunked on
the ITSP side.
regards
Clive
___
http://www.sineapps.com/news.php (Daily Asterisk News - html)
http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss)
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Motherboard and processor recommendations
Hi I discovered that most onboard raid controllers are really software raid, and it uses the cpu to perform raid functions. I am not sure how much extra load this introduces, but anyway, its still not ideal when you need your cpu for transcoding voip stuff. my 2c. regards Clive On 8 Sep 2005 at 12:01, Soner Tari wrote: Thanks Tzafrir and canuck15 for your comments. Yes I don't think the NIC will be saturated, and I'll search the quality of the Onboard RAID. I guess I have to learn more about canuck15's comments though, because I am actually questioning what happens to the board when you're adding onboard peripherals and whether that would create problems with, say, Digium cards. I remember I've read comments on the list saying that some chipsets/motherboards cannot handle the interrupt frequency that Digium cards demand, thus miss some interrupts. So, even though a regular desktop user would not notice any problems, an Asterisk server would suffer a lot. But I'm afraid there is no rule of thumb on such matters (except Xeon motherboards?). The load on the computer will never be too high, but my purpose in asking about processor preference is that if there is any processor dependant dsp routines (such as G729 codec), then I thought that I might have problems. As another example, I don't know the details of the echocancelers on Asterisk (all 5 of them), but perhaps their performance is more satisfactory on, say, a P4 2.4 machine rather than, say, an AMD64, even though I'd expect AMD64 to be a more powerful processor. So I am questioning code compatibility/performance based on processor type rather than processor load. If that's irrelevant, please disregard this question (I need to learn more about dsp routines). Thanks again for your answers, Soner - Original Message - From: canuck15 [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, September 08, 2005 3:46 AM Subject: RE: [Asterisk-Users] Motherboard and processor recommendations Regarding Chipsets/Motherboards. I would stay FAR away from cheap ones. Any chipset/motherboard that electrically and logically separates some PCI slots (ie. interrupts) from onboard peripherals (network controller, VGA, USB etc.) makes compatibility issues with Digium cards much less likely. Many of the newer Intel chipsets do this. The Xeon chipsets/motherboards are the best IMHO because they usually have PCI-X slots connected directly to the memory controller hub, that you can put your Digium card(s) in, which are completely separate from the peripherals and PCI slots on the I/O controller hub. -Original Message- From: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 07, 2005 4:59 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Motherboard and processor recommendations On Wed, Sep 07, 2005 at 11:02:58PM +0300, Soner Tari wrote: Hi All, For sometime now I've been searching the wiki and googling, but I think I'm missing some of the very important answers. So I'll have to ask this to the list. I'm trying to decide on the right motherboard and processor. Here are my questions: 1. Would I have problems with all-onboard motherboards (Onboard VGA, LAN/GLAN, Sound, SATA, RAID) ? I've read the comment about an Onboard VGA on wiki. Considering the exceptional quality of graphics you'll need with Asterisk, and VGA-compatible adapter would suffice. The on-board one would be more than enough. Ditto for the sound card, at least in most cases. As for the network adapter: Are you going to get anything close to saturating the card? I figure that the efficiency of the network adapter and its driver will not be your bottleneck. Most of the WAN-oriented systems would have worked fine with an old 10Mbps card, probably without a noticable performance hit (right?). So their quality is not much of an issue. If you have the extra space, you can always add an extra one in an expansion slot. But it should not be required. An extra raid controller is something you may consider. But then-again, if it is a cheap software-based raid, it is practically the same as using linux for that (but with more problematic drivers). But it is for you to decide if it is worth the extra cost. 2. Which chipset should I prefer: Intel, SiS or VIA? I've read the old SiS chipset problem on wiki. There is much voodoo about this. There are good and bad boards made with each of those chipsets. In fact, for practically each model of board that has been sold for over a month or so, you'll probably find someone in this list who had bad experience with it. 3. Which processor has the least support problems: P4 (478 or LGA775, or even EMT64) or AMD64 ? For example, in G729 config file Athlon comment
Re: [Asterisk-Users] Japanese ISDN BRI card for asterisk (INS64) where to start?
Hi It looks to me that the intel board is the same as the dialogic board. Clive On 29 Aug 2005 at 11:43, Mick Hastings wrote: Hi All, I currently run asterisk in our office (in Japan) and use a cisco PRI gateway for connection to the PSTN. I would like to setup some more systems for our smaller offices (in Japan) that would use BRI and preferably using a PCI card in the asterisk box and not a seperate Cisco gateway (expensive). HOWEVER, Japan has this INS64 protocol for their BRI lines and im not sure what cards are available that are compatible with asterisk and Japanese BRI (INS64). I know that it is supported by Cisco (like they support Japanese T1 PRI (INS1500)) but it just adds to the cost and is another piece of hardware. I tried searching the archives and only found a few references to INS64 and it didnt sound too promising. I then searched the net and found this Intel/Dialogic board: BRI/80-PCI BRI/PCI Series High-Density ISDN Basic Rate Interface Boards (for details see: http://www.intel.com/network/csp/products/7007web.htm) It seems to support INS64 but appears to only have windows drivers. Has anybody used this cards with asterisk? is it possible? or even likely that it would be supported by any of the linux ISDN drivers? I also noticed some other mentions of 'ISDN protocol converters' What are these specifically? (im guessing they convert between US BRI standards and INS64), how much are they? where do I get one? Has anybody out there got an asterisk system running with INS64 connections to their box? If so could you please let me know how you are doing it, else can anybody offer any information as to where I should start to look for more informaion this topic? I really appreciate the help. cheers, Mick Hastings ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Packet loss concealment and G729
Hi Does anyone know where one can get hold of a G729 codec for asterisk which effectively can do packet loss concealment using Steve Kann's wonderful new Jitter buffer. The 2 versions that I know of, (digium's and the IPP one) do not perform great at PLC, especially with 4 or 5% loss. Thanks in advance. Clive ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * CVS-HEAD and ASTCC Intermittent issue
Hi I have something similar, what happends is that intermittantly, (especially when DTMF tones are played) the call does not hang up when the timeout expires. It looks like it is related to your issue. Please let us know if you find any answers to this bug. Thanks Clive On 18 Jul 2005 at 12:02, seehoe yee wrote: Hie! I've installed Asterisk CVS-HEAD with ASTCC. The problem i'm facing is that the astcc.agi script completes when the recipient picks up the call. When the astcc.agi completes is returns 0 bill time but both end still able to talk. It occurs intermittently, any one facing the same issue? Asterisk Console - == Spawn extension (sip, 77, 2) exited non-zero on 'SIP/1112-15a3' -- Executing Answer(SIP/1112-9696, ) in new stack -- Executing DeadAGI(SIP/1112-9696, astcc.agi|1112|) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/astcc.agi -- Playing 'digits/4' (language 'en') -- Playing 'astcc-pin' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/40' (language 'en') -- Playing 'digits/2' (language 'en') -- Playing 'digits/20' (language 'en') -- Playing 'digits/3' (language 'en') -- Playing 'digits/20' (language 'en') -- Playing 'digits/3' (language 'en') -- AGI Script Executing Application: (DIAL) Options: (Local/[EMAIL PROTECTED]|30|HL/n(78:6:3)) -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/|30|tr) in new stack -- Called [EMAIL PROTECTED] -- Called -- Local/[EMAIL PROTECTED],1 is ringing -- SIP/-0c16 is ringing -- SIP/-0c16 answered Local/[EMAIL PROTECTED],2 -- Local/[EMAIL PROTECTED],1 stopped sounds -- Local/[EMAIL PROTECTED],1 answered SIP/1112-9696 -- AGI Script astcc.agi completed, returning 0 == Spawn extension (sip, , 1) exited non-zero on 'SIP/1112-9696' Regards See Hoe ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi ASTCC trouble
On 10 Jul 2005 at 22:01, Armin Schindler wrote: On Sun, 10 Jul 2005, Clive wrote: Hi all I am wondering if anyone has had a similar trouble to this: The timeout arguments in the dial command does not work. The caller does not get disconnected when the timeout reaches zero. I am not sure if this is a chan_capi issue, or a asterisk issue. I am using CVS-head and chan_capi CVS head also. Any suggestions or help will be appreciated. Thanks Clive Ok, just did some testing on the dial command using only iax2 and it does disconnect the call, so this may be a chan_capi issue. As far as I know, the timeout and hangup logic is done within Asterisk e.g. dial-application. chan-capi does not know anything about a timeout, so I don't know how this can be the location of the problem. Armin Hi On doing some tests, I have found that the timeout works fine only if the caller does not dial any DTMF tones , like for an IVR system. If DTMF tones are dialled during the call, the timeout doesn't work. another piece to add to the puzzle..:) very wacky, but hopefully this may help find the bug best regards Clive ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi ASTCC trouble
Hi all I am wondering if anyone has had a similar trouble to this: The timeout arguments in the dial command does not work. The caller does not get disconnected when the timeout reaches zero. I am not sure if this is a chan_capi issue, or a asterisk issue. I am using CVS-head and chan_capi CVS head also. Any suggestions or help will be appreciated. Thanks Clive ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi ASTCC trouble
Hi all I am wondering if anyone has had a similar trouble to this: The timeout arguments in the dial command does not work. The caller does not get disconnected when the timeout reaches zero. I am not sure if this is a chan_capi issue, or a asterisk issue. I am using CVS-head and chan_capi CVS head also. Any suggestions or help will be appreciated. Thanks Clive Ok, just did some testing on the dial command using only iax2 and it does disconnect the call, so this may be a chan_capi issue. Any suggestions will be great.:) thanks Clive ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2 Trunking - CVS-Head
Hi Is anyone successfully using iax2 trunking with CVS head ? The reason I am asking is that I have heard there may be some audio problems, which I would like to know about before sending customer's calls over a iax2 trunked connection. Thanks in advance. Clive ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Eicon equipment, BRI Server or PRI?
Hi As far as I know, only the server versions of Eicon work with asterisk (using chan_capi). There are a few other BRI cards that work with asterisk. Junghanns cards seem to work the best from the little I have seen. Good luck. Regards Clive On 27 Jun 2005 at 23:19, [EMAIL PROTECTED] wrote: Hello Everyone, I am once again wondering about EICON. I have had no success with the Diva Pro or Diva Pro PCI, so my question is, does anyone use an Eicon Server BRI card on Asterisk? Or would I be better off trying to get a split PRI? Regards, Greg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC/ 'L' option hangup wackyness
Hi I am using CVS-HEAD-05/09/05 and astcc. The call sometimes does not hang up at all, does not even get the warning notices, as it should, since the L option specifies that the caller gets played a message like 1 minute before the call is disconnected, and then the call should end. Its basically not working, although I seem to think that if only one call is being processed it does work, but with multiple calls confuses the system somewherealthough I am not sure of this. If anyone has some suggestions, it will be appreciated. By the way, it worked fine with asterisk 1.0.2 , I am not sure what changed in the CVS since then. Thanks Clive ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX ATA's
I would also be interested in a multi-port ata that supports iax. The only single port ata I know of (besides the iaxy) that supports iax is the PA168 from china. cheers Clive On 27 Apr 2005 at 11:15, Rod Bacon wrote: What sort of price are they asking for a 4-port gateway? - Original Message - From: Joseph [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, April 27, 2005 8:53 AM Subject: Re: [Asterisk-Users] IAX ATA's There is Taiwan company Soundwin that seems to me are willing to support asterisk protocol in their equipment. I was looking for 1FXO x 3-4FXS in one unit. http://www.soundwin.com/ I just exchanged few email with Sam at [EMAIL PROTECTED] and was able to convince them to add support for IAX2; they seems to me listen to the end user so I suggest some of you drop him an email and express your interest in their product if they will support asterisk protocol. --quote We have plan to implement IAX or IAX2 in our product line including 2 -8 port VoIP Gateway in Q3. Thanks your information and we would pay more attention in Asterisk community. -end quote- -- #Joseph On Tue, 2005-04-26 at 16:46 -0400, Garrett Smith wrote: Does anyone know of a quality alternative to the Digium IAXy? I have a customer experiencing numerous issues such as over heating with the older IAXyÿs and the new IAXy is not yet available. Can anyone recommend an alternative? Thanks, Garrett Smith [EMAIL PROTECTED] B2 Technologies/ VoIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 (716) 250-3408 Direct (716) 630-1548 Fax (716) 903-9495 Cell AOL IM: B2sales Specializing in New and Used equipment from vendors including Cisco Systems, Juniper, Adtran, Dialogic, Lucent, Nortel, Sipura, Granstream, Snom, Mediatrix, Carrier Access, Digium, Zultys, IPDialog and more. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codecs and * pass through...
Etienne, howzit I am not 100% sure about this, but Net2phone do not always use standard SIP as the protocol. They have their own proprietry protocol as well, so perhaps your phone is trying to talk on the proprietry protocol. For G723.1 passthrough, you just allow it, and it should work fine, as long as you do not try playing any voice prompts to the channel. good luck. regards Clive = Phone I.T. http://www.phonehome.co.za On 13 Apr 2005 at 8:52, Etienne Pretorius wrote: Hello all, I came a cross a problem yesterday that I don't quite know how to solve. I am trying to use * to connect to net2phone, and have a net2phone MAX IP-10 connect to net2phone. From the settings on http://www.voip-info.org/ it was easy to get asterisk to connect to the network - acting like a net2phone device/user. Anyway the problem arose when attempting to call the MAX IP-10 device through the net2phone network. They seem to be using the G732.1 codec. I have in my settings in sip.conf allow=G732.1 or what ever flavour of the like and still I can not talk to the two devices. I googled a bit and came across the fact of * being able to do a pass through - well I was not successful and this subject is either simple or not well documented. The devices are using SIP and there is a bridge initiated, but there is no audio and no voice being passed through... I have tried connecting as the receiving device a GrandStrem Budge Tone-100 and still no luck. So all that I am inquiring is has anyone successfully done a pass through and if so can someone please guide me through some of the settings. I have set the [net2phone] with a canreinvite=yes - that a post on a forum also suggested, and that also did not work. On a separate issue: When the Grandstream Budge Tone-100 is connected on the internal network then the audio and the voice in both directions work fine. But when the device is connected on a separate network - ie on an other ADSL line, then the device doesn't send voice packets although is receives packets. I have opened up IPTABLES, to allow udp 5060 and udp 1:2 in both directions on any interface and the problem still persists. (SIP phone: Grandstream Budge Tone 100 connects to * and the call is answered by a Softphone X-Lite with all the codecs enabled. As far as I can tell thy both are speaking with a G711 codec ULaw/ALaw). So can anyone please give me a guideline or some advise on where to look to solve the problem. -- Kind Regards Etienne ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Petition for IAX firmware
Why dont you just get the netweb phone which already has iax support On 5 Apr 2005 at 17:13, Sean Kennedy wrote: denon wrote: Hi all, I've put together a quick petition, in hopes that we can possibly persuade Sipura (or any other large-scale IP handset manufacturer) to include firmware support for IAX. The IAXy has proven that an IAX product is in demand, and very useful, and I think we'd all like to see a handset manufacturer follow Digium's lead. I'm not particularly endorsing Sipura, however I do know that they have seriously considered support for IAX, and have decided to hold off until the demand is there. I'm hoping that with some numbers, we can prove to them that the demand is already here, and that IAX is already a viable technology. I'd like to encourage everyone to show your support -- hopefully Sipura, and/or other manufacturers will see these hard names and numbers, and realize it's time to move something into production. Petition: http://www.petitiononline.com/IAXPhone Thanks, -d Signed. Sean Kennedy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] at-320 phone configuration difficulty
Mishehu try 19750407 Also to get palmtool to work you need to play with the debug settings on the phone first. koltov Clive On 2 Apr 2005 at 0:30, I put the Who? in Mishehu wrote: Hi guys, I just got a Netweb 401 (AT-320) phone. It came with firmware 1.38 on it, and it has since been updated after failed attempts to configure, and now has 1.42 (IAX2) from centrality (P1688S). According to voip-info, atcom's docs, etc, there are two passwords for it - one is 1234, and the superuser password is supposed to be 12345678. Only 1234 works, and I get codec configuration, IP configuration, firmware/ringtone/dialplan update options. But nowhere do I find where to set information about my asterisk box I want this phone to connect to. I've tried using Palmtool 1.42, and anytime I try to query the phone's settings, I get Cannot connect to Palm1. The person who sold me sent no documentation or discs with it, and now on top of it, all the buttons such as Local Num and Local IP are all switched around. I am very unhappy, and have wasted 4 hours already trying to work on this. If anybody can assist, I'll be very grateful. -mishehu ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC dialstatus confusing billing issue
Hi I wonder if anyone else has noticed this, or has an explanation When a call ends with dialstatus=cancel ,one would expect that the call never went through, BUT it seems that sometimes a call does go through sucessfully, and ends with dialstatus=cancel and I have no idea why.??..very strange. The problem this introduces to ASTCC is that it does not bill for these calls if it sees the Cancel. This can be fixed easily, but I am still baffled why the cancel comes through, and I am wondering if anyone else has had a similar experience. Regards Clive ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk@home scary log
On Thu, 10 Feb 2005 17:49:23 +, Clive Carter [EMAIL PROTECTED] wrote: I hesitated before sending this, as I have been flamed before for being a beginner. but I am newish to linux/asterisk, and I am running an ssh server. It is still running with default settings, (I dont know yet how/where to change it), and I CAN logon remotely as root. (Haven't figured out how to 'su' yet !) This is using the Rapid Xorcomm v 1.0 cd, which I believe (may be wrong) is based on a very recent version of Debian ? Perhaps xorcom have changed the default setting ? Hey Clive. I thought it was mentioned earlier before in the thread, but if not, all you need to do is edit your sshd_config file. In Debian, this is located at /etc/ssh/sshd_config, but it could be different for other distros. Open that up in a text editor and then locate the line that says PermitRootLogin yes, and change that to PermitRootLogin no. Save it, and then restart SSH. On Debian, you type in /etc/init.d/ssh restart, but on other distros it might be different. Note that you'll have to be root to edit that file and restart that service. -- Dana Thanks for that. I did not see it before, and I was afraid to ask in case I got jumped on again ! Thanks again -- -- Clive Email : [EMAIL PROTECTED] Tel : 08444844790 Alt : 08450043366 Fax : 08444844813 SIP : [EMAIL PROTECTED] Mobile : 07031945504 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk@home scary log
I'm not sure what happens when you do a fresh compile and install of OpenSSH, but every distro I've ever worked with (Red Hat, Gentoo, Slackware, Vector, Tao, Yellow Dog, Debian, Knoppix, SuSe, Linspire, FreeBSD, OpenBSD, Darwin, OS X) has allowed root logins via SSH by default. Maybe they're changing that on newer versions of some distros. I dunno. I'll call bullshit on that. I know for a fact that Debian does NOT allow root logins except from console. Hell Debian isn't allowing root logins from X anymore due to the likely hood for you to try and use root for more than administration. I hesitated before sending this, as I have been flamed before for being a beginner. but I am newish to linux/asterisk, and I am running an ssh server. It is still running with default settings, (I dont know yet how/where to change it), and I CAN logon remotely as root. (Haven't figured out how to 'su' yet !) This is using the Rapid Xorcomm v 1.0 cd, which I believe (may be wrong) is based on a very recent version of Debian ? Perhaps xorcom have changed the default setting ? -- Clive Email : [EMAIL PROTECTED] Tel : 08444844790 Alt : 08450043366 Fax : 08444844813 SIP : [EMAIL PROTECTED] Mobile : 07031945504 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax2-jitter-trunking?
There is a new patch in the mantis for jitter buffering together with
trunking.
On 6 Feb 2005 at 18:45, Mark Eissler wrote:
AFAIK, trunk=yes is not a global option. You set it within a context.
Also, using the jitter buffer with trunk=yes is not recommended since
its broken right now.
-mark
On Feb 6, 2005, at 12:45 PM, Rich Adamson wrote:
Two cvs-head asterisk boxes with iax2 working fine (without register
statements).
When two calls are placed simultanously from system A - B and the
packets
are sniffed on the wire, I see the two calls using two different udp
packets. At the top of iax.conf I have trunk=yes and jitterbuffer=yes
(at both ends).
I was expecting to see both calls handled within a single udp packet,
but that's not happening. Each iax2 packet is 79 bytes using ethereal.
I've tried the trunk=yes both within the inbound context and at the top
of the iax.conf file (assuming the one at the top would be used for all
outbound iax calls that don't reference a context). Calls are placed
with:
exten = _2.,1,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:1})
Is trunking dependent upon the use of 'register'? Or, dependent on the
above exten=_2., referencing a context (instead of the IP directly)?
--
Mark Eissler, [EMAIL PROTECTED]
Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI for Data and Voice
Dave, howzit You can use asterisk with a quad E1 card to divide your E1. So anyone who dials in using 1234 for example, route to your portmaster and anyone who dials in using 1235 use for IVR/voip, whatever. Good luck Regards Clive On 29 Jan 2005 at 15:11, David Norton wrote: Hi, Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRIs of 15 channels each, or plugging the PRI into the * box and it send the data calls to the portmaster, or handles them itself? Any advice would be much appreciated Regards David Norton ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoIP-to-TDM processing on-card?
As far as I am aware, Atacomm is in the process of building a DSP based card which would work with asterisk I just checked, its going to be called a ipVolution TDM120 On 20 Jan 2005 at 14:42, Michael Baird wrote: On Thu, 2005-01-20 at 14:05 -0500, Olson, Dana wrote: I did look there. If you read my follow up, I screwed up the original question. What I want is a card with multiple T1 ports that do the processing on the card, and not on the system CPU. I'm not aware of any cards with DSP's on board for Asterisk (nice thought), the Digium cards I have rely on the PC's CPU to handle the calls. Is there a mailing list for Asterisk where people treat each other in a civil manner? __ Dana Olson It's only one guy who seems to attack each poster for not posting in a manner of which he approves (there is one/two of these fellows on every mailing list), don't let him ruin your day, this list is quite helpful and many guys will give you a good answer without the extra attitude. Regards Michael Baird ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC - error on call end
Hi I get an error popping up when the call ends as follows: DBD::mysql::db do failed: Unknown column 'callstart' in 'field list' at /var/lib/asterisk/agi-bin/astcc.agi line 90, STDIN line 32. Does anyone else get this same thing? Looks as if my database table is wrong, or something else is up...not sure Thanks Clive ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] UK * group
Ben wrote Is there a UK Asterisk users group? Would be interested in contacting others in the UK who use asterisk for either home or business applications. If there is, could someone provide me with some contact details, else anyone who's also interested, contact me off list. Add me to the list ! (And willing to help organize if there isn't one already) -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 08444844790 Alt : 08450043366 SIP : [EMAIL PROTECTED] Mobile : 07031945504 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] capi help please..(capi not installed - No such device or address (6) )
Yay, I got it working.! I added CAPI verbose reason reporting to the kernel, and modprobed as follows: capi, kernelcapi, divacapi, divas and then loaded divactrl and it works!!, yaynow to figure out how to get Asterisk to load with capi...my .conf files seem to be wrong Thanks for everyones help. regards Clive On 9 Jan 2005 at 12:39, Philipp Ebneter wrote: Hi Clive, I ran into a similar problem: I also have a eicon 4bri and tried to install it on a dell server with redhat as 3. The problem I have is that I always got a error message when doing modprobe capi. The module is compiled in the kernel (it shows up with lsmod). I have not found any solution so far - in the end I had to install asterisk on a different machine. I guess there must be some interrupt problem and maybe this is also the root of your problem. regards philipp Clive wrote: Asterisk wont load because capi wont load. When I do capiinfo I get: capi not installed - No such device or address (6) I have a eicon 4bri card installed, with the patched kernel with the melware.de files. No irq conflicts., and the dmesg shows this: capifs: Rev 1.1.4.1 CAPI-driver Rev 1.1.4.1: loaded capi20: started up with major 68 kcapi: capi20 attached capi20: Rev 1.1.4.2: started up with major 68 (middleware+capifs) I have modprobed capi with no error messages. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] capi help please..(capi not installed - No such device or address (6) )
Hi I wonder if anyone has had a similar trouble. I have googled and no solutions. Asterisk wont load because capi wont load. When I do capiinfo I get: capi not installed - No such device or address (6) I have a eicon 4bri card installed, with the patched kernel with the melware.de files. No irq conflicts., and the dmesg shows this: capifs: Rev 1.1.4.1 CAPI-driver Rev 1.1.4.1: loaded capi20: started up with major 68 kcapi: capi20 attached capi20: Rev 1.1.4.2: started up with major 68 (middleware+capifs) I have modprobed capi with no error messages. If anyone has any suggestions, they will be appreciated. Thanks Clive ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk PBX Manager
I have this, it comes as a webmin module. I also got it with the intention of bundling it for clients. It costs $300 , and the license is tied to the NIC. While it wont do EVERYTHING, it will probably be sufficient for the user to set up extensions/phones/menus/voicemail/conferences. One thing that I am not happy with, is that it allows raw editing of the conf files. Gawd help us if a user gets into that lot. I emailed Third lane, and they replied staright away with an address where I could download an evaluation. I'd publish the url here, but there must be a reason why they don't show it on their web site. Oh, and by the way (this from a beginner), I found it by searching on the WIKI Clive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rtp compile error
Hi Just uploaded source from cvs (CVS-HEAD-11/27/04-12:56:51) Zaptel and libpri make install works ok, but I get the following error when running make install in asterisk directory rtp.c : in function 'ast_rtp_bridge': rtp.c : 1552 internal compiler error : Illegal instruction Please submit a full debug report ... make *** [rpt.o] : Error 1 What have I done wrong ? (Its got to be me, never do anything right !) Thanks -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] rtp compile error
Hi Just uploaded source from cvs (CVS-HEAD-11/27/04-12:56:51) Zaptel and libpri make install works ok, but I get the following error when running make install in asterisk directory rtp.c : in function 'ast_rtp_bridge': rtp.c : 1552 internal compiler error : Illegal instruction Please submit a full debug report ... Looks like n'owt to do with asterisk this is your compiler bailing out. Seems like its a hardware error. I reran it a bit later, and it stopped in another place. Waited a while, reran again and this time it went through. Machine seems to have been giving a few unusal errors lately. Must check cpu fan/hard disc thanks -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Paul Mahlers Book
Anybody know of a UK supplier of Voip Telephony with Asterisk by Paul Mahler ? I've searched on the web, and the only suppliers I can find are US based, and the postal charge is as much as the book. cheers -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE : -lssl
Found and fixed the problem. Did not have libssl-dev installed. Thanks -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SipTone II
Anybody used the above phone with asterisk I have one working ok for calls, but having a problem with voice mail. Using either the 'Voice mail function key' or dialing 88 (for my system) just gets me to Call Terminated Asterisk CLI shows the error message 'unable to get User name' My Grandstream works ok, asking for User name, then Password Any ideas ? -- Clive On Fri, 19 Nov 2004 09:44:13 -0800, Michael Swan [EMAIL PROTECTED] wrote: At 02:55 AM 11/19/2004 +, you wrote: Hi Clive, I've been using a SipTone II for quite a while. Great phone but kind of pricey. I got the VM key working by configuring the Voicemail Server item in the Phone Configuration web interface section as follows: sip:[EMAIL PROTECTED] where voicemailextension is the extension number for accessing voicemail in * and asterisk.company.com is the domain name or IP address of your * machine. I'm using Firmware version: SipTone 1.2.0 rc Z_8. Hope this helps. Michael Swan Neon Software, Inc. Hello, I had the same problem with the SipTone - it's just a matter of setting the dtmfmode in the sip.conf file. I think I set it to inband - I remember setting it to either that or rfc2833 or whatever that rfc number is - the correct number is available in the sip.conf fdile itslf. Just fiddle with the dtmf mode - either inband or rfc and u'll be fine. Hope this helps. Shireen Thanks guys. Tried all suggestions above and some of my own. Nothing worked, Tried every combination OF INFO, RFC2833, Inband on phone and in Sip. No good In desparation I reset EVERYTHING to defaul, rebooted, then put all my data back in. IT WORKS ! Must have made a typo or something in the phone setup, but I'm damned if I could find it. Only thing that stopped working now is the VM button, even tho that is set up as per Michaels instructions. I can get at the voicemail by dialling 88 anyway, so I am leaving it alone :-) Thanks again. -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] UK available SIP phone?
Hi, Anybody here from the UK using Asterisk at home? I'm looking for a SIP phone which will work with Asterisk and not leave me broke! I got one of the Tecom ones from Solwise but it refuses to login to Asterisk server for some reason. May have to send it back. What are the other options please? Thanks Mike I use Grandstream Budge Tones. They are cheap, and some people say they look it, but they work ! I have also got ipDialogs SipTone II. They are twice the price, and although I have got the basic functions working, for some reason they just will not connect to VoiceMail HTH -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Voicemail Issue
Hello , When comedian mail prompts for login info , no matter what I dial on the phone , nothing is sent to * . I'm using a budgetone 102 , with the latest firmware (1.0.5.16). I have set dtmfmode=Info in sip.conf. I'm not sure if its the phone or * that is the issue. Any assistance would be appreciated. I have Budget Tone 1's working with no problem. (Same software level) The message button does not work (known problem). You need to set up an extension in extension.conf and dial that e.g. exten = 8000,1,Wait(1) exten = 8000,2,VoiceMailMain exten = 8000,3,Hangup Dial 8000 and it should work. If you have done this and still got a problem, send me your Voicemail.conf. sip.conf,extensions.conf and I'll have a look -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SipTone II
Anybody used the above phone with asterisk I have one working ok for calls, but having a problem with voice mail. Using either the 'Voice mail function key' or dialing 88 (for my system) just gets me to Call Terminated Asterisk CLI shows the error message 'unable to get User name' My Grandstream works ok, asking for User name, then Password Any ideas ? Hi Clive, I've been using a SipTone II for quite a while. Great phone but kind of pricey. I got the VM key working by configuring the Voicemail Server item in the Phone Configuration web interface section as follows: sip:[EMAIL PROTECTED] where voicemailextension is the extension number for accessing voicemail in * and asterisk.company.com is the domain name or IP address of your * machine. I'm using Firmware version: SipTone 1.2.0 rc Z_8. Hope this helps. Michael Swan Thanks for your reply Michael. I don't think I explained my problem very well. I was dialling '88' (my voicemail extension) from the Siptone to get to Voicemail. I have tried your tip , but with the same result The CLI shows Executing VoiceMailMain(SIP/2004-43c0,) in new stack Playing 'vm-login' WARNING [655381] app-voicemail.c:2748 vn-execmain : Couldnt read Username Spawn extension (internal,88,2) exited non-zero on 'SIP/2004 - 9f48' - (internal is the context, 2004 is the extension of the SipTone) The problem is that I do not hear the vm-login message. The PHONE has hung up with a message 'Call Terminated' before that comes through. The Grandstreams have no problem getting voicemail, and just to check out, I disconnected the Siptone from the circuit, set up one of my Grandstreams with the 2004 extension, and it worked ok. I have no problem making internal/external calls with it. Its just Voicemail. Relevant bits of config files SIP [2004] type=friend secret=2004 host=dynamic mailbox=2004 dtmfmode=rfc2833 context=internal VoiceMail 2004=2004,2004 Extensions,conf exten = 88,1,Wait(1) exten = 88,2,VoiceMailMain exten = 88,3,Hangup Thanks -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SipTone II
Anybody used the above phone with asterisk I have one working ok for calls, but having a problem with voice mail. Using either the 'Voice mail function key' or dialing 88 (for my system) just gets me to Call Terminated Asterisk CLI shows the error message 'unable to get User name' My Grandstream works ok, asking for User name, then Password Any ideas ? -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Analog calls not working
Hi
I have battled my way through setting up linux,
and then installing Asterisk. I have got 90% of the way there.
Asterisk registers with my IAX provider ok, my SIP phones can send and
receive calls
to each other, and out over the network.
Voicemail is working ok.
The last thing I want is to direct incoming analog calls to a SIP phone,
and to send all calls starting with a '9' out through my analog line.
I have been scratching my head over this for 2 days, and cannot find the
answer anywhere.
Can anyone help please ?
This is the message I get when I try -
CLI output
Nov 10 15:45:13 DEBUG[81926]: Check for res for 2001
Nov 10 15:45:13 DEBUG[81926]: Call from user '2001' is 1 out of 0
Nov 10 15:45:13 DEBUG[81926]: build_route: Contact hop:
sip:[EMAIL PROTECTED];user=phone
Nov 10 15:45:13 VERBOSE[294931]:
[1;37;40mAsterisk Ready.[0;37;40m-- Executing
[1;36;40mDial[0;37;40m([1;35;40mSIP/2001-3d7a[0;37;40m,
[1;35;40mZap/1/07970856261[0;37;40m) in new stack
Nov 10 15:45:13 NOTICE[294931]: Unable to create channel of type 'Zap'
Nov 10 15:45:13 VERBOSE[294931]: == Everyone is busy at this time
---
result of zap show channels
Chan ExtensionContextLanguageMusicOnHold
1inbound-analogen
-
result of zap show channel 1
File Descriptor : 28
Span:1I
Extension:
Caller ID string:no
Destroy:0
Signalling Type:FXS Kewlstart
Owner:None
Real:None
Callwait:None
Threeway:None
Confno:-1
Propagated Conference:-1
Real in Conference:0
DSP:noI
Relax DTMF:yes
Dialing/CallwaitCAS0/0
Default law:ulaw
Fax Handled:no
Pulse Phone:no
Echo Cancellation:128 taps, currently off
Actual Confinfo:Num/0, Mode/0x
Actual Confmute:No
-
ZTCFG result
Channel Map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured
---
ZAPTEL.CONF
loadzone = uk
defaultzone=uk
fxsks=1
ZAPATA.CONF
signalling=fxs_ks
echocancelwhenbridged=yes
relaxdtmf=yes
rxgain=0.0
txgain=0.0
context=inbound-analog
channel = 1
SIP.CONF
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
allow=alaw
allow=gsm
register = ID:[EMAIL PROTECTED]
context = internal
(rest is configuration for SIP Phones)
EXTENSIONS.CONF
.
.
[internal] ; context for SIP phones
exten = 01952XX,1,Dial(Zap/1/${EXTEN})
exten = 01952XX,2,Hangup
.
.
[inbound-analog]
exten = _0[1-9].,1,Dial(${OFFICE},15,Ttm) ;OFFICE is SIP phone
exten = _0[1-9].,2,VoiceMail(u${OFFICEVM}); OFFICEVM is mailbox for
OFFICE phone
exten = _0[1-9].,3,Hangup
TIA
Clive
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CVS RPMs for Mandrake 10 (Zaptel and, Asterisk)
Dear Scott I am new user of Mandrake 10 And very excited at the idea to work with Asterisk but, as you can imagine. I am currently blocked because of the kernel 2.6.. the Wildcard X100P drivers . I would be more than happy to get test your source RPMs for zaptel and asterisk And so would I !! -- Clive Email : [EMAIL PROTECTED] Alt : [EMAIL PROTECTED] Tel : 0845 0043366 Alt : 01952 402032 SIP : [EMAIL PROTECTED] Mobile : 07970 856261 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] User problem
The user is me !
I decided to try out *, and installed the xorcom Rapid installation
With the help of the wiki and reading posts on this list, I have got to
the stage where I have got an internal pbx, and able to dial out
and receive calls.
I am know trying to incorporate one analog line into the system.
I have a x100P.
If I use the Rapid front end and look in zttools, the X100p is showing
as a RED alarm. Don't know if that has anything to do with my problem
ztcfg reports
Channel 01 : FXS Kewlstart (Default) (Slaves : 01)
1 Channel{s) configured.
Zaptel.conf
fxsks=1
(loadzone and default zone stuff)
Zapata.conf
signalling = fxs_ks
Channel = 1
If Channel=1 is changed to ;Channel =1, Asterisk loads ok
However, when Channel =1 is enabled, asterisk won't start
Gives following errors
ERROR[16384]: chan_zap.c: 6181 mkintf : Unable to get parameters
ERROR[16384]: chan_zap.c :9109 setup_zap : Unable to register channel '1'
WARNING[16384] : loader.c:334 ast_load_resource : Chan_zap.so :
load_module failed, returning -1
Unregistered channel type 'Tor'
Unregistered channel type 'Zap'
So what have I done wrong, or not done ??
Thanks for reading
--
Clive
Email : [EMAIL PROTECTED]
Alt : [EMAIL PROTECTED]
Tel : 0845 0043366
Alt : 01952 402032
SIP : [EMAIL PROTECTED]
Mobile : 07970 856261
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Eezee phone?
Hi This phone is based on the PA1688 chip. The rumour is that IAX2 support will be available by December...we will have to wait and see. Regards Clive On 1 Nov 2004 at 13:16, Kanuri, Seshu (Company IT) wrote: The link refers to an expired auction. It is no longer listed as having IAX2. That claim was withdrawn till IAX2 on it is stabilized by the Chip manufacturer. NOTICE: If received in error, please destroy and notify sender. Sender does not waive confidentiality or privilege, and use is prohibited. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX trunking clarification...
Hi I would think that it is possible, but never been implemented in other protocols. Quintum has something like trunking they call packet saver, and they use h.323 goodluck with your presenation. Clive On 26 Oct 2004 at 10:23, Chris Bshaw wrote: Hi I am putting together a presentation on VoIP technology, and I just wanted to make sure I get my facts straight. I have read that one of the features of the IAX protocol is that in can trunk multiple calls in a single UDP stream. Anywhere I have read this, it seems to be implied (but never stated explicitly) that this is something that is not possible with other protocols (eg: SIP, H.323, Skinny, MGCP etc.) Is this correct? Thanx muchly in advance. Chris Bradshaw. _ FREE pop-up blocking with the new MSN Toolbar - get it now! http://toolbar.msn.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] chan_mISDN
Hi I am just wondering if chan_mISDN is a worthwhile alternative to zaphfc which I am having issues with. I have 2 hfc-s modem cards in my asterisk box. Any comments or advice will be appreciated. Thanks Clive On 19 Oct 2004 at 11:16, Brian West wrote: Well the error does give you some clue on whats wrong and it's done that way to give you exactly what you need to do: Use AST_DEFINE_STATIC rather than AST_MUTEXT_INITIALIZER Check out the other apps and compare them to chan_mISDN and you'll get what you need to change.. its only one line if I recall. bkw -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Erwan DESVERGNES Sent: Tuesday, October 19, 2004 10:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] chan_mISDN Did someone have succeed to compile chan_misdn ??? Ive got an error when in try to compile chan_misdn.c:68: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function) thanks _ Erwan Desvergnes - ANDIUM - 82/86 rue Château Gaillard 69100 Villeurbanne Tel. 04 37 43 44 45 / Fax 04 37 43 44 44 E-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-2000's rebooting every hour or so...
Hi I have also have the Sipura rebooting itself. I changed the codec from G723.1 to G729 and this seems to have helped fix the problem. I have the latest firmware...2.0.10(e) I think..?? Hope this helpsstrange stuff though. regards Clive On 14 Oct 2004 at 14:48, Mike Benoit wrote: I thought it originally started happening after a firmware upgrade to 2.0.10e, so I downgraded to 2.0.10d, and the problem continued. I'm in the process of moving them to a cooler place and putting a fan on them just to rule out overheating, which I've heard can be a problem. On Thu, 2004-10-14 at 12:36 -0700, Dooz Owings wrote: try to run a firmware update on one and see if it works, just a guess. What all have you tried ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Benoit Sent: Thursday, October 14, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SPA-2000's rebooting every hour or so... I realize this is slightly off-topic here, but I know quite a few people on this list use Sipura products. Has anyone else experienced the same rebooting problem I'am? I have about 8 SPA-2000's and about half of them just started rebooting 4-8times/day in the last month or so. (they used to be rock solid) I already emailed Sipura support, but they seem to be on strike as of late. Here is the debug output from just one of the devices: (I've trimmed it for size, it happens more often than what is shown) Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 21:34:14 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:20:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 22:01:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 22:01:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 23:21:17 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 23:21:17 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 00:41:20 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 14 00:41:20
RE: [Asterisk-Users] SPA-2000's rebooting every hour or so...
Hi Mine used to reboot on every call Clive On 15 Oct 2004 at 0:15, Mike Benoit wrote: How often was it rebooting before, do you know? Mine seem to be rebooting almost exactly 1hour apart, which is the registration expire time. I've just recently changed it to 6hrs, so I'll see if that makes a difference. On Fri, 2004-10-15 at 08:26 +0200, [EMAIL PROTECTED] wrote: Hi I have also have the Sipura rebooting itself. I changed the codec from G723.1 to G729 and this seems to have helped fix the problem. I have the latest firmware...2.0.10(e) I think..?? Hope this helpsstrange stuff though. regards Clive On 14 Oct 2004 at 14:48, Mike Benoit wrote: I thought it originally started happening after a firmware upgrade to 2.0.10e, so I downgraded to 2.0.10d, and the problem continued. I'm in the process of moving them to a cooler place and putting a fan on them just to rule out overheating, which I've heard can be a problem. On Thu, 2004-10-14 at 12:36 -0700, Dooz Owings wrote: try to run a firmware update on one and see if it works, just a guess. What all have you tried ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mike Benoit Sent: Thursday, October 14, 2004 10:36 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] SPA-2000's rebooting every hour or so... I realize this is slightly off-topic here, but I know quite a few people on this list use Sipura products. Has anyone else experienced the same rebooting problem I'am? I have about 8 SPA-2000's and about half of them just started rebooting 4-8times/day in the last month or so. (they used to be rock solid) I already emailed Sipura support, but they seem to be on strike as of late. Here is the debug output from just one of the devices: (I've trimmed it for size, it happens more often than what is shown) Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:18:33 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:33:37 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 08:53:52 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 12 21:34:14 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:20:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:28:31 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:48:55 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:53:08 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:54:23 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:55:15 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:56:46 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 11:59:05 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:C200 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 12:40:58 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 14:00:51 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 15:20:54 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H73720143 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 16:44:57 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 18:01:03 192.168.1.189 System started: [EMAIL PROTECTED], reboot reason:H0 Oct 13 19:21:06
[Asterisk-Users] Bristuff wackyness - not answering
Hi Is anyone having this weird trouble with bristuffcalls don't get answered, but on the debug it looks as if the call is answered. The call will just hear ringing. I am using -BRI-stuffed-0.1.0-RC4a Occassionally the system starts working on its own, other times I have to stop and rmmod everything and start them all up again. These error messages appear on the CLI: == Primary D-Channel on span 2 down Sep 5 02:13:04 WARNING[213005]: chan_zap.c:1942 pri_find_dchan: No D-channels available! Using Primary on channel anyway 6! WARNING[213005]: chan_zap.c:6902 zt_pri_error: PRI: !! Not good - head of queue has not been transmitted yet WARNING[213005]: chan_zap.c:6902 zt_pri_error: PRI: !! Got reject for frame 8, retransmitting frame 8 now, updating n_r! kernel: zaphfc: empty HDLC frame received. Any help or pointers will be appreciated. Thanks and regards Clive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] error 1 and 2 during make of asterisk with SUSE 8.2 and 9.1
Paul wrote: Hi, i'm traying to compile asterisk on my pc, a laptop whit SUSE 9.1 and a desktop with SUSE 8.2, with a teles S0 16/3 PnP. With Kernel 2.4 (Desktop) Asterisk run but it's umpossible to compile the driver ISDN-utils for Teles. With kernel 2.6 I can't compile zaptel (not necessary with my laptop) and asterisk, in both cases I receve errors during make or make linux26 (I saw the notes on http://www.voip-info.org/wiki+Asterisk+Zaptel+Installation). These r my notes from compiling on SUSE 9.1 Bit painful until u know what to do :-) Install the kernel souces from yast Then you need to install this rpm which is ONLY on the DVD, not on the CDs - sigh kernel-syms-2.6.4-52.i586.rpm Then run the yast online updater to get the latest kernels and sources reboot then in /usr/src/linux make cloneconfig make prepare make modules Then make a symlink from /usr/src/linux to /usr/src/linux-2.6 Then you can build all the * stuff ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
