Re: [Asterisk-Users] chan_capi-cm and DID
Hi armin, thx for the answer. I have connected the BRI on a HiPazt4000 and i still have the same issue. So i think i have a problem with my ISDN line. I will contact my provider. May be a reset of the line will solve the problem. rich. --- Armin Schindler [EMAIL PROTECTED] wrote: On Tue, 17 Jan 2006, richard Coco wrote: Hi Armin, thx for your feedback, but what do you mean with Did you load the card with config for DID on that port? I have loaded the modules with: modprobe capi modprobe kernelcapi modprobe divacapi modprobe divas and then loaded divactrl like this: divactrl load -f ETSI I suppose that this is ok (it works without did)? Or have i forgotten something? With divactrl load -f ETSI you load the card to PtMP (which is the default) on all four ports. Use divactrl load -c 1 -SeparateConfig -u1 where the '1' of -u1 means second port. E.g. -u is first port, -u1 -u2 -u3 is port 2,3,4. When using -SeparateConfig, the X-extension is available for many options. E.g., you can put port 3 and 4 into NT-mode, or even run another protocol (1TR6, JAPAN, QSIG,...) on other ports. See divactrl load -h for all options. Armin thx in advance.. --- Armin Schindler [EMAIL PROTECTED] wrote: On Mon, 16 Jan 2006, richard Coco wrote: Hi all, i have asterisk 1.0.9 with an Eicon Diva 4bri and chan_capi-cm-0.6. I have 2 NTBAs (one with did and one without). When using the one without did, i am able to place outgoing and incoming calls. When i use the NTBAs with did i have a layer 2 error. Anyone an idea? Did you load the card with config for DID on that port? What are your divactrl parameters? (Or do you use Eicon Package with xml based config?) Armin -- Executing Dial(SIP/2004-9634, CAPI/g1/43XX) in new stack data = g1/43XX parsed dialstring: 'g1' '43XX' '' capi request group = 2 parsed dialstring: 'g1' '43XX' '' == EICON: Call CAPI/EICON/43XX-6 (pres=0x00, ton=0x00) CONNECT_REQ ID=001 #0x000c LEN=0065 Controller/PLCI/NCCI= 0x1 CIPValue= 0x10 CalledPartyNumber = 8043XX CallingPartyNumber = 00 80 22EyeBeam22 3c20043e CalledPartySubaddress = default CallingPartySubaddress = default BProtocol B1protocol = 0x1 B2protocol = 0x1 B3protocol = 0x0 B1configuration= default B2configuration= default B3configuration= default BC = default LLC = default HLC = default AdditionalInfo BChannelinformation= 00 00 Keypadfacility = default Useruserdata = default Facilitydataarray = default -- Called g1/43XX CONNECT_CONF ID=001 #0x000c LEN=0014 Controller/PLCI/NCCI= 0x201 Info= 0x0 -- EICON: received CONNECT_CONF PLCI = 0x201 DISCONNECT_IND ID=001 #0x0011 LEN=0014 Controller/PLCI/NCCI= 0x201 Reason = 0x3302 DISCONNECT_RESP ID=001 #0x0011 LEN=0012 Controller/PLCI/NCCI= 0x201 CAPI INFO 0x3302: Protocol error layer 2 == EICON: CAPI Hangingup == EICON: Interface cleanup PLCI=0x201 == No one is available to answer at this time my capi.conf looks like: [DID] controller=1,2,3,4 isdnmode=did incomingmsn=* softdtmf=on relaxdtmf=on accountcode= context=DID echocancel=yes ;echocancelold=yes devices=2 group=1 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and
Re: [Asterisk-Users] speex in asterisk 1.0.10
Yeah, Paul. I guess you're right.. Just tested speex and got complains from my customer :S..Maybe this codec is not suited for our network ;).. Regards, Stevanus [EMAIL PROTECTED] wrote: Quick question - what is the point of speex? Do we really need it as an option? PaulH - Original Message - From: "stevanus" [EMAIL PROTECTED] To: "Asterisk Users Mailing List - Non-Commercial Discussion" asterisk-users@lists.digium.com Sent: Thursday, January 19, 2006 3:37 PM Subject: [Asterisk-Users] speex in asterisk 1.0.10 Hi, Does anyone know how to configure speex in asterisk 1.0.10? I've successfully installed it but cannot get any idea how to set the quality, etc.. Thanks Regards, Stevanus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and Linux-HA
Hi Srs., we have installing two machines with Asterisk and Linux-HA. I just copy conf files and voicemail files and more with rsync, and now I want to test with Linux-HA if asterisk is up. I'm installing Asterisk over Debian, but I haven't a status function in asterisk script. Any one help me to know how can I check if asterisk is up? If I switch off master machine or I cut network cable, second machine goes up OK, but if I switch on or replug cable in Main machine, all works fine but I realize that slave machine doesn't down asterisk. Any one has installed this system? regards, tron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spandsp-0.0.2pre22 not working!
hi, it seems that spandsp-0.0.2pre22 is not functioning right. downgrading to pre21 makes it work again. debug messages: Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Changed from phase 1 to 4 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW ???: Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Real-time Internet fax (T.38) Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW V.8 capable Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Prefer 64 octet blocks Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Reserved: 0x90 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Supported data signalling rates: Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: V.27ter fallback mode Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW 2D coding Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Scan line length: Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: 215mm Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Recording length: Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: A4 (297mm) Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Receiver's minimum scan line time: Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: 20ms at 3.85 l/mm: T7.7 = T3.85 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Reserved: 0x1 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Minimum scan line time for higher resolutions: T15.4 = T7.7 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Character mode Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW Reserved: 0x10 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: FLOW DIS: Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: 80 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: 00 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: ce Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: f4 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: 80 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: 80 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: 81 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: 80 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: 80 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: 80 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: 18 Jan 18 11:54:29 DEBUG[5157] app_rxfax.c: Jan 18 11:54:31 DEBUG[5157] app_rxfax.c: FLOW HDLC underflow in state 9 Jan 18 11:54:31 DEBUG[5157] app_rxfax.c: FLOW Changed from phase 4 to 3 Jan 18 11:54:31 DEBUG[5157] app_rxfax.c: FLOW HDLC carrier up Jan 18 11:54:31 DEBUG[5157] app_rxfax.c: FLOW HDLC framing OK Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW DCS: Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: 83 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: 00 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: 06 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: a4 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: 80 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: 80 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: 00 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW DCS with final frame tag Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW In state 9 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW ???: Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Real-time Internet fax (T.38) Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW V.8 capable Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Prefer 256 octet blocks Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Reserved: 0x80 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Supported data signalling rates: Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: V.27ter fallback mode Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW 2D coding Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Scan line length: Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: 215mm Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Recording length: Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: A4 (297mm) Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Receiver's minimum scan line time: Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: 20ms at 3.85 l/mm: T7.7 = T3.85 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Get at 9600bps, modem 1 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Changed from phase 3 to 5 Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Fast carrier up Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Fast carrier down Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Fast carrier up Jan 18 11:54:33 DEBUG[5157] app_rxfax.c: FLOW Fast carrier training failed Jan 18 11:54:34 DEBUG[5157] app_rxfax.c: FLOW Fast carrier down Jan 18 11:54:38 DEBUG[5157] app_rxfax.c: FLOW Fast carrier up Jan 18 11:54:38 DEBUG[5157] app_rxfax.c: FLOW Fast carrier training failed Jan 18 11:54:39 DEBUG[5157] app_rxfax.c: FLOW Fast carrier down Jan 18 11:54:39 DEBUG[5157] app_rxfax.c: FLOW Fast carrier up Jan 18 11:54:39 DEBUG[5157] app_rxfax.c: FLOW Fast carrier training failed Jan 18 11:54:41 DEBUG[5157] app_rxfax.c: FLOW Fast carrier down Jan 18 11:54:44 DEBUG[5157] app_rxfax.c: FLOW Fast carrier up Jan 18 11:54:44 DEBUG[5157] app_rxfax.c: FLOW Fast carrier training failed Jan 18 11:54:45 DEBUG[5157] app_rxfax.c: FLOW Fast carrier down Jan 18 11:54:45 DEBUG[5157] app_rxfax.c: FLOW Fast carrier up Jan 18 11:54:46 DEBUG[5157] app_rxfax.c: FLOW Fast carrier training failed Jan 18 11:54:47 DEBUG[5157] app_rxfax.c: FLOW Fast carrier down Jan 18 11:54:50
Re: [Asterisk-Users] OT: Network Wire Brand
On Wed, Jan 18, 2006 at 03:44:03PM -0800, calvis wrote: Sorry about the OT thread, but I am sure that someone could give me some advice. Nothing is more frustrated than doing a long cable run and then finding your cable is defective. OK, I have had it with the General Cable brand of network cable that we currently use for 5e cable runs. I am looking for something that is 100 percent reliable for doing cable runs. Does anyone have any recommendations? Panduit, it's kinda like the Rolls Royce of cables. The best one can get but it comes at a price. Kristian. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] speex in asterisk 1.0.10
On Thu, Jan 19, 2006 at 03:46:41PM +1100, [EMAIL PROTECTED] wrote: Quick question - what is the point of speex? Do we really need it as an option? Three points: 1. some nice features, see http://speex.org/ . Some are unique. Though not all are applicable to transcoded usage with Astrisk. 2. Like other codecs from Xiph.org (vorbis, theora, flac, the ogg container), it is probably the most patent-free of the decently-compressed codecs. 3. It is really handy when you want to stress the CPU. gsm requires much more calls to get to the same load level :-( Thus you'll typically find it in soft phones. You should expect it to give better quality than gsm. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Network Wire Brand
On Thu, 19 Jan 2006 00:16:19 -0600, Russ Price wrote Ross C wrote: We've always used Coleman and Belden with great results. We get a good deal on Coleman, so that's what we usually use--never had a problem. Berk-Tek is also good, I know a lot of cable installers who swear by it (I think Berk-Tek is actually owned by Nexans). Another one to consider would be Hitachi. I've had excellent results with Gigabit over the Hitachi Cat 5e cable that I installed in the office where I work. One other thing to consider - a good punchdown tool is a MUST. The Ideal that I used for the office wiring has worked very well. Use Krone cable and a genuine Krone tool It isnt the cheapest, but it is the best. Russ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Open WebMail Project (http://openwebmail.org) ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] making wakeup feature call phone number, not extension?
On Mon, Jan 16, 2006 at 08:17:28PM -0600, Moises Silva wrote: I dont know the wakeup feature. But what you want can be done with a web interface generating .call files with the timestamp of the day, hour and time when you want to hear the reminder. Just read in voip-info about the .call files and if you have doubts we will be glad to help you. generate an at daemon job that drops this file. See at(1) . echo script with params | at 6:00 -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF Simultaneous Inband and RFC2833 performedby Asterisk = Duplicate tones
I have seen the following effect in Asterisk, though: where it converts an inband DTMF (eg coming off a Zap channel) into an indication, it mutes the audio where that tone is. But sometimes it leaves a teeny bit of the tone behind. If you take such a call over say IAX to somewhere and then back out a Zap channel, you end up with the teeny remaining bit of the original tone, PLUS the regenerated tone. If you are very unlucky a remote DTMF receiver can hear two digits. The same thing can happen when a SIP ATA is configured to use rfc2833 but is also a little to lote with the filtering out of the DTMF. So sometimes it's not Asterisks fault at all ;-) And then there's some IVR's that don't notice it at all, while others are totally unusable. -- Andreas Sikkema BBned NV Software EngineerPlaneetbaan 4 +31 (0)23 70743422132 HZ Hoofddorp ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Brief silences during calls
Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] A problem in recieving voice on one side
Dear All, I am having a problem in a scenario I am doing, I have two branches, every branch has has an [EMAIL PROTECTED] that deals with each branch locally and a trunk connected to a central asterisk, now if any branch wants to call another branch it goes from the local asterisk@ home --> to the central asterisk server and then forwarded --> to the remote [EMAIL PROTECTED] server --> to the phone, this works and rings and the call is up but the problem lies in that one side can hear the voice and sends voice but the other side can send voice and not hear anything coming, any ideas where to begin, I would like to highlight some data below: [EMAIL PROTECTED] latest version on both sides. Central asterisk uses Asterisk 1.2.1. phones support reinvite some I am using reinvite=yes If you need any more data I will supply it, I wasn't sure what to put or even where to start, and I didn't want it to be a very long mail. -- Thx MAG ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP hardphones with xml/html/xhtml/microbrowser support?
-BEGIN PGP SIGNED MESSAGE- Hash: RIPEMD160 [EMAIL PROTECTED] wrote: | On Wed, 18 Jan 2006, Hirosh Dabui wrote: | | look there http://snom.com/wiki/index.php/Xmlobjects for snom | 360... | | | nice... any hope for snom 320? | | -Dan ___ | i think not, coz it makes no sense on a small display... Hirosh - -- snom technology AG Dipl.-Ing. Hirosh Dabui PGP Key-ID: 0x30A34758 mailto:[EMAIL PROTECTED] http://snom.com -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.2 (GNU/Linux) iD8DBQFDz2ohAO47/DCjR1gRA/7PAJ43rm7PCWQZ5mkkj0u5vh/pNbu71QCfWLnh xhFEZ21gMzPmTMIf/90psUk= =tAyA -END PGP SIGNATURE- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sipura ata 3000 UK (BT) CAllerid
Under the regional tab (admin/advanced) select ETSI FSK WITH PR (UK) for caller id method. Make sure PSTN CID For VoIP CID: is set to yes in the pstn tab to pass on the cid to asterisk. Chris - Original Message - From: Conrad Wood [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 18, 2006 9:00 PM Subject: [Asterisk-Users] sipura ata 3000 UK (BT) CAllerid Hi I wonder whether anyone got the Sipura ata 3000 to decode British Telecoms callerid and pass it to asterisk? The userguide seems to suggest that this is not possible, is that right? Conrad ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Brief silences during calls
You might try using a tool like Ethereal to look at what's happening on the network. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Mimmus Sent: 19 January 2006 10:13 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Brief silences during calls Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR Accounting Question
I
have aproblem with the cdr.
We terminate through
a pstn provider to the pstn network.
The problem is now
the cdr accounts the connection to the gateway. Coz the gateway is answering our call and then forward to the pstn
number.
So i have billsecs
all the time even it is only ringing or so.
Somebody has a
solution for that?
-- Executing Dial("SIP/1000114-fcf8", "SIP/[EMAIL PROTECTED]|60")
in new stack -- Called [EMAIL PROTECTED]
-- SIP/connect.xxx.de-c61d is making progress passing it to
SIP/1000114-fcf8 -- SIP/connect.xxx.de-c61d answered
SIP/1000114-fcf8 -- Attempting native bridge of
SIP/1000114-fcf8 and SIP/connect.xxx.de-c61d
Regards
rene
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RE: [Asterisk-Users] SMS to fixed phone line
It seems that ETSI standard ES 201 912 documents the protocol which is (may be?) used in Australia. If anyone is interested it can be downloaded from http://www.etsi.org/services_products/freestandard/home.htm, after filling in some soul sucking registration details :) James -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of James Harper Sent: Thursday, 19 January 2006 15:08 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] SMS to fixed phone line Telstra (Australian Telco) has recently introduced a feature to allow the sending of SMS direct to fixed analogue lines, with an appropriate handset. As best as I can figure out, this appears to use CID type signalling, or at least on a line that otherwise has no CID on it, CID is sent, but with a standard modem I can only receive the date, time, and phone number (eg normal CID info). After that the phone rings, but Telstra will just call the number and use 'Text to Speech' to read the message out when a user answers. Does anyone know anything more about this in Australia or, failing that, if they do the same thing anywhere else in the world? My guess is that either: 1. the whole message is transmitted in the CID period, but my modem doesn't hear it, but then I don't know how Telstra would know that the message has been received. 2. Some indicative signalling takes place in the CID, which then triggers the handset to hide rings from the user and use normal modem signalling to transfer the message. If it has been around for a while outside Australia, is there an SMS module for Asterisk which would make use of it? I think that being able to receive (and probably send - haven't even started looking at that yet but it is supported in the same way) SMS messages would be a really nifty thing to be able to do from a phoneline, and would save me buying a $600 GSM modem to do the same thing! Thanks James ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds
[EMAIL PROTECTED] wrote:
On Wed, 18 Jan 2006, Javier Oviedo wrote:
Jan 18 18:06:17 NOTICE[16386]: chan_sip.c:11213 do_monitor:
Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11
seconds
Jan 18 18:06:17 WARNING[17340]: file.c:583 ast_readaudio_callback:
Failed to write frame
== Spawn extension (default, 331222, 3) exited non-zero on
'SIP/172.25.92.153-085340d0'
The channels has RTP activity because I hear the voicemail message
The problem is that no RTP is coming from the other side (ie towards
Asterisk). This check is in case the other side has disappeared
suddenly. It doesn't help Asterisk to know that its transmitting. It
could transmit for hours and hours to nowhere and never know the other
side is gone. (that's UDP for you).
Best is to fix the original source so as to not do silence suppression. If
you can't do that, you can remove or lengthen the rtp timeout by adjusting
rtptimeout= and rtpholdtimeout= in the sip.conf file.
Regards,
Steve
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Hi Steve, thanks for your response, my h323 endpoints have the silence
suppression option set a off. I remove the rtptimeout and
rtoholdtimeout options in the sip.conf file and now I obtain the
following error:
*CLI -- Executing Set("SIP/X.X.X.X-09f3ebf8", "LANGUAGE()=es")
in new stack
-- Executing SetCallerID("SIP/X.X.X.X-09f3ebf8", "331222") in new
stack
-- Executing VoiceMail("SIP/X.X.X.X-09f3ebf8", "u331223") in new
stack
-- Playing 'vm-theperson' (language 'es')
-- Playing 'digits/3' (language 'es')
-- Playing 'digits/3' (language 'es')
-- Playing 'digits/1' (language 'es')
-- Playing 'digits/2' (language 'es')
-- Playing 'digits/2' (language 'es')
-- Playing 'digits/3' (language 'es')
-- Playing 'vm-isunavail' (language 'es')
-- Playing 'vm-intro' (language 'es')
-- Playing 'beep' (language 'es')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/331223/INBOX/msg0011 format:
wav49, 0x9ef4f60
Jan 19 11:51:11 WARNING[19282]: app.c:653 ast_play_and_record: No audio
available on SIP/X.X.X.X-09f3ebf8??
-- User hung up
I think that it's a rare behavior of asterisk because the problem only
ocurs in "Not Response" case study but not in "Busy" or "Unavailable"
responses.
Thanks in advance!
Regards
___
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RE: [Asterisk-Users] Advice Of Charge (AOC) ?
Pisac wrote: I'm very surprised that Asterisk (PBX !) do not support AOC. Setting some variable with AOC informations should be enough. Storing AOC in CDR would be perfect. Arg! I noticed just now that Asterisk breaks all my report applications, extracting accounting data from an Alcatel PBX downstream to Asterisk: PRI PSTN --- Asterisk --- E1 cable --- Alcatel PBX In fact, there are no accounting data in the Alcatel PBX since I put Asterisk in front of it. I hope this will not abort the full Asterisk project in my company! Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still the LDAP Realtime extension
Hi, On Wed, 18 Jan 2006 21:18:47 -0200 Juan Carlos Castro y Castro [EMAIL PROTECTED] wrote: | OK, I've got the new schema installed and I was able to create | oxyPBXAccountSIP objects. | | Now, how do I generate the MD5 values to put in the realmedPassword field? md5sum -b user:realm:secret for sip, realm is in sip.conf Manuel -- __ Manuel Guesdon - OXYMIUM ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] LDAP direct authentication Problem
Hi, On Wed, 18 Jan 2006 18:50:33 +0530 Chandan Mishra [EMAIL PROTECTED] wrote: | Hi | I need to authenticate all the asterisk users from the LDAP server instead | of from sip.conf. | If anybody already have done this then please guide. | | I tried to integrate authenticate asterisk users from LDAP using the open | source project astirectory1.2.0. | After using the astirectory1.2.0 , now when the asterisk starts then it | registeres with the LDAP. Following logs shows it. | | Jan 18 18:36:20 WARNING[26190]: res_config_ldap.c:641 parse_config: LDAP | RealTime Host: ldap://192.168.0.16 | Jan 18 18:36:20 WARNING[26190]: res_config_ldap.c:642 parse_config: LDAP | RealTime User: synapse\dirsearch | Jan 18 18:36:20 WARNING[26190]: res_config_ldap.c:643 parse_config: LDAP | RealTime Base DN: dc=synapse,dc=com [...] | actually ldap_search_s(ldap, ldapbasedn, LDAP_SCOPE_ONELEVEL, query2, NULL, | 0, res) function is not execution successfully in the file | res_config_ldap.c. | | I am not able to find the reason. There's a problem with base dn configuration when you have multiple parts (dc=,dc=). I have to submit a fix (really soon now :-) Manuel -- __ Manuel Guesdon - OXYMIUM [EMAIL PROTECTED] 14 rue Jean-Baptiste Clement - 93200 Saint-Denis - France Standard: +33 811 093 286 - Fax: +33 1 7473 3971 - LD: +33 1 7473 3980 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CDR Accounting Question
René Enskat [Teamware GmbH] a écrit : I have a problem with the cdr. We terminate through a pstn provider to the pstn network. The problem is now the cdr accounts the connection to the gateway. Coz the gateway is answering our call and then forward to the pstn number. The gateway is configured wrong. If it's an asterisk gateway, avoid using Answer() Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell PowerEdge 830 server
Have you seen if this equipment share IRQ for the resto of PCI Slots. I want to install one TDM2400P with 24 FXS Port and one TDM04B with 4 FXO ports but I want to know if that equipment has voltage connector for TDM2400P and it doesn't share IRQ in two PCI Slots. regards, tron -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Kerry Garrison Enviado el: miércoles, 18 de enero de 2006 5:38 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Dell PowerEdge 830 server To be specific, I installed [EMAIL PROTECTED] 2.2 which is CentOS 4.2, Asterisk 1.2.1, Asterisk Management Portal, Flash Operator Panel, etc etc. That site has about 15 users with half of them having both on-site and off-site extensions (setup using AMP's Users and Devices mode). This site is not using any real-time functions. They do use the meet-me rooms fairly heavily. The system has a TDM400 with 4 FXO ports on it and the phone lines are in a hunt group that does a rollover on the 5th call to Teliax on the pay as you go plan which provides 10 additional channels. Does that help? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jennifer Hales Sent: Tuesday, January 17, 2006 7:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell PowerEdge 830 server Hello Kerry, Many thanks for your information. Do you mind giving some more details on your setup? What version of Asterisk are you using? How many users do you have? Are you using real-time? And what Asterisk features are you providing? Feel free to reply off list if you wish. Kind regards Jenn Hales -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Wednesday, January 18, 2006 1:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell PowerEdge 830 server Go into the BIOS, disable all unneeded peripherals like floppy controller, serial ports, parallel ports, etc. It should work fine, I have one at a decent sized installation. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jennifer Hales Sent: Tuesday, January 17, 2006 5:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Dell PowerEdge 830 server Hello all, We are looking at using a Dell PowerEdge 830 Server for an Asterisk installation. Does anyone have experience using this server with Asterisk? Any feed back would be appreciated. Kind regards Jenn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Dropping incompatible voice frame
I am now getting these messages on a second box running a different version of Asterisk. If anyone has any idea what is causing these, or how to avoid them I would be very grateful. 157 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 158 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 159 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 160 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 161 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 162 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 163 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw 164 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw Thanks, Joe ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem configuring Asterisk
Hi All, I tried with different configurations and referred many articles to configure Asterisk with a Vonage account I have but all my attempts failed. I am a newbie and hope this mailing list will help fixing my problem and configure Asterisk. The error I get after I make a call to outside number like 18007633555 is -- Accepting AUTHENTICATED call from 59.93.69.218, requested format = 1024, actual format = 1024 -- Executing Dial(IAX2/[EMAIL PROTECTED]/2, SIP/[EMAIL PROTECTED]:5061|30|r) in new stack -- Called [EMAIL PROTECTED]:5061 == No one is available to answer at this time Jan 19 06:26:36 NOTICE[29115]: chan_sip.c:4045 sip_reg_timeout:-- Jan 19 06:26:38 WARNING[29115]: pbx.c:1948 ast_pbx_run: Timeout, but no rule 't' in context 'outgoing' -- Hungup 'IAX2/[EMAIL PROTECTED]/2' I am attaching my iax.conf,sip.conf and extensions.conf please check them and help me. Thanks, Manoj. sip.conf Description: Binary data iax.conf Description: Binary data extensions.conf Description: Binary data ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iaxmodem and Efax?
Hello, I did some tries with efax and it worked pretty well, however I sent faxes to the same machine, so I don't know how good is this combination in real life faxes. Regards, Tamas Carlos Chavez wrote: Has anyone tried to use an Asterisk server with iaxmodem and efax? I have installed both and they work well on their own. Efax is working with an external modem connected to a Cisco ATA186. The only thing I change is the device to /dev/ttyIAX. When I dial my fax extension it rings but it never answers. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Loud Tone issue, still having problems
Hello Everone,I hope everyone is having a good day. I am having a problem with my asterisk box. When I call the box from a land line or cell phone and I press a number I hear a very loud tone and then it comes back and says the person is unavailable. The loud tone I hear is very annoying. I am not sure why this is happening. Any ideas? Thanks! Randy ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sipura ata 3000 UK (BT) CAllerid
I wonder whether anyone got the Sipura ata 3000 to decode British Telecoms callerid and pass it to asterisk? The userguide seems to suggest that this is not possible, is that right? It's definately possible... A quick google returned the following (from http://voxilla.com/PNphpBB2-printview-t-7101-start-0.html) In the [admin, advanced] PSTN tab, you should have: Detect CPC: Yes Detect Polarity Reversal: Yes Min CPC Duration: 0.085 (you've got this one right) Detect Disconnect Tone: Yes Disconnect Tone: [EMAIL PROTECTED],[EMAIL PROTECTED];20(*/0/1+2) I don't have one (had real issues with the 2000 and have shied away from sipura stuff since) but all posts I can find seem to agree it works. I haven't paid much attention to the spa3k and BT support, but if memory serves correctly, the options noted above were added in one of the later sipura versions. So, if you don't see the options, upgrade the firmware. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Processor Size
Can someone give me an idea of the processor power I will need for 1 x TDM240 with 2xquad FXO's and 8 sip phones/ATA's on a quite 100Mbit LAN. The machine we have available of hand is a P4 1GHz with 768MB RAM. Tx Marnus van Niekerk -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Connection pooling
I wrote a connection pooling patch because asterisk is not usable with MSSQL without it. If you're using, or would like to use, MSSQL I recommend you to check it out. http://bugs.digium.com/file_download.php?file_id=8809type=bug Just so you know, this is a diff against 1.2.1 and it's been only tested with 1.2.1 (because that's what we are using in production). To enable pooling, you need to edit your res_odbc.conf file and add pooled = yes poolsize = 10 To the entries you want to be pooled This also adds an additional command odbc showpool that shows detailed information about the connection pools. We've been running with this patch on two machines and it seems to be working just fine. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Processor Size
Marnus, Sounds like a good configuration to me. You might want to upgrade the RAMs to over a gig, but thats about it. Joash www.kahuna.nl From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marnus van Niekerk Sent: Thursday, January 19, 2006 2:23 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Processor Size Can someone give me an idea of the processor power I will need for 1 x TDM240 with 2xquad FXO's and 8 sip phones/ATA's on a quite 100Mbit LAN. The machine we have available of hand is a P4 1GHz with 768MB RAM. Tx Marnus van Niekerk -- Opportunity is missed by most people because it isdressed in overalls and looks like work.Thomas Alva Edison - Inventor of 1093 patents,including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Processor Size
Marnus - I assume you mean a Pentium "3" 1GHz processor? I don't believe there was a 1GHz P4. You should be alright with that machine, you can always add more RAM, RAM is pretty inexpensive these days. Cory J AndrewsVOIPSupply.com454 Sonwil DriveBuffalo, NY 14225++voice - 716.630.1555 X22email - [EMAIL PROTECTED]AIM - B2CORY - Original Message - From: Marnus van Niekerk To: asterisk-users@lists.digium.com Sent: Thursday, January 19, 2006 8:23 AM Subject: [Asterisk-Users] Processor Size Can someone give me an idea of the processor power I will need for 1 x TDM240 with 2xquad FXO's and 8 sip phones/ATA's on a quite 100Mbit LAN.The machine we have available of hand is a P4 1GHz with 768MB RAM.TxMarnus van Niekerk-- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Processor Size
Thatshouldbefine.Ihaverunmorewithmuchless. As a rule disable any uneeded services (chkconfig is your friend). Also it has been mentioned time and time again DO NOT RUN X on the machine. The load placed on the machine will interupt your Asterisk process and you will get choppy sound at best. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Marnus van NiekerkSent: Thursday, January 19, 2006 8:23 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Processor Size Can someone give me an idea of the processor power I will need for 1 x TDM240 with 2xquad FXO's and 8 sip phones/ATA's on a quite 100Mbit LAN.The machine we have available of hand is a P4 1GHz with 768MB RAM.TxMarnus van Niekerk-- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sipTAPI and usernames
I have installed the sipTAPI from http://sourceforge.net/projects/siptapi/ when I use user names like joash.herbrink in Asterisk, it is not working when I change the sip username to my internal extension, like 1006, it works fine. Anybody any idea as to why this is? met vriendelijke groet, Joash Herbrink Technical Consultant Control the flow De Kahuna groep helpt organisaties met het zakelijk gebruik van Internet. Kahuna Network Solutions levert beheerde oplossingen die de beveiliging, performance en beschikbaarheid van netwerk- en Internetinfrastructuur verbeteren. Kahuna Business Solutions levert oplossingen voor het verbeteren van on-line Customer Relationship Management (eCRM). Specialisaties: E-mail management en Web Self Service. Kahuna Telecom is de service provider op het gebied van breedband Internet, point-to-point verbindingen en vaste telefonie oplossingen. Kahuna IP-communications richt zich op het verbeteren vanSpraak-, Data- en Beeldcommunicatie doorinnovatieve inzet van middelen op basis van IP. Maanlander 14a/bm: +31 6 53 80 28 20 3824 MP Amersfoort e: [EMAIL PROTECTED] t: +31 33 4500370ext 1006URL: www.kahuna.nl f: +31 33 4500371 Voor support e-mailt u naar [EMAIL PROTECTED] of belt u in dringende gevallen naar +31 33 4500373. Kahuna is winnaar van de ICT Company Award 2002, de ComputerPartner Award 2003, en staat 4e in de DeloitteTouche fast 50. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Processor Size
Thanx for all the answers. I do intend using chkconfig to disable everything and definately no X. This machine will be asterisk only - nothing else. (Except httpd and mysql for web based cdr reports.) M Alexander Lopez wrote: Thatshouldbefine.Ihaverunmorewithmuchless. As a rule disable any uneeded services (chkconfig is your friend). Also it has been mentioned time and time again DO NOT RUN X on the machine. The load placed on the machine will interupt your Asterisk process and you will get choppy sound at best. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Marnus van Niekerk Sent: Thursday, January 19, 2006 8:23 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Processor Size Can someone give me an idea of the processor power I will need for 1 x TDM240 with 2xquad FXO's and 8 sip phones/ATA's on a quite 100Mbit LAN. The machine we have available of hand is a P4 1GHz with 768MB RAM. Tx Marnus van Niekerk -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SAN Devices
On Wed, 2006-01-18 at 14:26 -0500, Adam Robins wrote: Anyone out there using small-midsized (2-4 TB) SAN solution among multiple Asterisk systems? I don't have the budget for an EMC-caliber solution, and can't seem to find much else out there. Have a look at www.dothill.com. SANnet II boxes are fully redundant and NEBS Level III certified. Afaik they are below EMC's price level. Sun's StorEdge 35xx are oem'ed from Dot Hill. See http://www.sun.com/storage/workgroup/ Or you could build your own with software from e.g. www.openfiler.org Regards, Patrick ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Simultaneous Inband and RFC2833 performed by Asterisk = Duplicate tones
[EMAIL PROTECTED] wrote: I have seen the following effect in Asterisk, though: where it converts an inband DTMF (eg coming off a Zap channel) into an indication, it mutes the audio where that tone is. But sometimes it leaves a teeny bit of the tone behind. Yes, that is correct. By the time that DTMF detector has determined that the tone exists, some of the tone has already been passed on. This occurs because the Asterisk DSP runs in parallel with the audio path, not in the middle of it, so it cannot 'delay' the audio to be able to mute it retroactively. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk 1.2.2 RPMS for CentOS 4.x
yes, asterisk work with centos - Original Message - From: Eric Bishop To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 19, 2006 1:47 AM Subject: Re: [Asterisk-Users] asterisk 1.2.2 RPMS for CentOS 4.x will they work with CentOS 4.2? On 1/19/06, Andrew McRory [EMAIL PROTECTED] wrote: I have compiled a set of RPMS from svn and put them in the regular place.Link:ftp://ftp.linuxsys.com/pub/releases/CentOS-4.0/asterisk-1.2.2/ Best Regards,--Andrew McRory - President/CTOLinux Systems Engineers, Inc. - http://www.linuxsys.comLocated in beautiful Tallahassee, FloridaOffice850-224-5737 Office850-575-7213Mobile850-294-7567___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Visita www.tutopia.com y comienza a navegar ms rpido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PHPAGI
i'm finding a little script example in phpagi, to do a query in mysql, how i do that, beacause i'm tired of finding information about that, and the code of php dont work for me anybody have a little example on how do that??? __ Visita http://www.tutopia.com y comienza a navegar más rápido en Internet. Tutopia es Internet para todos. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk least cost routing expert needed
We need an expert in least cost routing (LCR) for an Asterisk project. Please provide references and a resume of your experience. Contact us at [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PHPAGI
Here is a simple mysql snippet in php. Straight from the PHP manual.
http://www.php.net
$link = mysql_connect('HOST', 'UID', 'PASS') or die (Could not connect);
mysql_select_db('DATABASE_NAME') or die (Could not select database);
$query = SELECT * FROM table;
$result = mysql_query($query) or die(Query failed);
while ($line = mysql_fetch_array($result))
{
var_dump($line)
}
mysql_free_result($result);
mysql_close($link);
anybody have a little example on how do that???
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Re: [Asterisk-Users] PHPAGI
thanks for the reply!
i have installed [EMAIL PROTECTED] i need install more soft? odbc etc...?, i think if [EMAIL PROTECTED]
have inside the phpmyadmin i dont need more installed, this is true?
thanks
Vladimir
- Original Message -
From: Mark Ackroyd [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, January 19, 2006 9:27 AM
Subject: RE: [Asterisk-Users] PHPAGI
Here is a simple mysql snippet in php. Straight from the PHP manual.
http://www.php.net
$link = mysql_connect('HOST', 'UID', 'PASS') or die (Could not connect);
mysql_select_db('DATABASE_NAME') or die (Could not select database);
$query = SELECT * FROM table;
$result = mysql_query($query) or die(Query failed);
while ($line = mysql_fetch_array($result))
{
var_dump($line)
}
mysql_free_result($result);
mysql_close($link);
anybody have a little example on how do that???
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__
Visita http://www.tutopia.com y comienza a navegar más rápido en Internet.
Tutopia es Internet para todos.
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Re: [Asterisk-Users] Asterisk and Linux-HA
Doesn't heartbeat take care of this? It's been awhile since I've configured it. If two servers join back together as master, one of them shuts down its services. Maybe I'm just wishfully thinking.. There's also a directive to determine if a secondary should fail back over to the master if it comes back up. All your 'shared' services should be controlled by heartbeat -- you shouldn't have to do anything except use the supplied init scripts. On 1/19/06, Tron [EMAIL PROTECTED] wrote: Hi Srs., we have installing two machines with Asterisk and Linux-HA. I just copy conf files and voicemail files and more with rsync, and now I want to test with Linux-HA if asterisk is up. I'm installing Asterisk over Debian, but I haven't a status function in asterisk script. Any one help me to know how can I check if asterisk is up? If I switch off master machine or I cut network cable, second machine goes up OK, but if I switch on or replug cable in Main machine, all works fine but I realize that slave machine doesn't down asterisk. Any one has installed this system? regards, tron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Linux-HA
Yes, I know, but I have I was think that heartbeat use status function in init asterisk script to check if asterisk is alive, but status function is for redhat.Are there any similar function in Debian?. And in respect of slave, when slave get all resources and master wakeup, maste request for all resources, slave give it all resources, but asterisk continues alive in slave. My questions are: What I need to say to heartbeat that asterisk is alive or dead and why when slave give all resources to master doesn't goes down itself asterisk service. My init script is on /etc/hearbeat/resource.d and in /etc/init.d/. Any idea? regards, tron -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Gary Richardson Enviado el: jueves, 19 de enero de 2006 16:28 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Asterisk and Linux-HA Doesn't heartbeat take care of this? It's been awhile since I've configured it. If two servers join back together as master, one of them shuts down its services. Maybe I'm just wishfully thinking.. There's also a directive to determine if a secondary should fail back over to the master if it comes back up. All your 'shared' services should be controlled by heartbeat -- you shouldn't have to do anything except use the supplied init scripts. On 1/19/06, Tron [EMAIL PROTECTED] wrote: Hi Srs., we have installing two machines with Asterisk and Linux-HA. I just copy conf files and voicemail files and more with rsync, and now I want to test with Linux-HA if asterisk is up. I'm installing Asterisk over Debian, but I haven't a status function in asterisk script. Any one help me to know how can I check if asterisk is up? If I switch off master machine or I cut network cable, second machine goes up OK, but if I switch on or replug cable in Main machine, all works fine but I realize that slave machine doesn't down asterisk. Any one has installed this system? regards, tron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Linux-HA
Could you possibly use the redhat init scripts instead? Or at least duplicate the functionality under Debian. (I'm not too familiar with Debian, so I don't know how it does such things). On 1/19/06, Tron [EMAIL PROTECTED] wrote: Yes, I know, but I have I was think that heartbeat use status function in init asterisk script to check if asterisk is alive, but status function is for redhat.Are there any similar function in Debian?. And in respect of slave, when slave get all resources and master wakeup, maste request for all resources, slave give it all resources, but asterisk continues alive in slave. My questions are: What I need to say to heartbeat that asterisk is alive or dead and why when slave give all resources to master doesn't goes down itself asterisk service. My init script is on /etc/hearbeat/resource.d and in /etc/init.d/. Any idea? regards, tron -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Gary Richardson Enviado el: jueves, 19 de enero de 2006 16:28 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] Asterisk and Linux-HA Doesn't heartbeat take care of this? It's been awhile since I've configured it. If two servers join back together as master, one of them shuts down its services. Maybe I'm just wishfully thinking.. There's also a directive to determine if a secondary should fail back over to the master if it comes back up. All your 'shared' services should be controlled by heartbeat -- you shouldn't have to do anything except use the supplied init scripts. On 1/19/06, Tron [EMAIL PROTECTED] wrote: Hi Srs., we have installing two machines with Asterisk and Linux-HA. I just copy conf files and voicemail files and more with rsync, and now I want to test with Linux-HA if asterisk is up. I'm installing Asterisk over Debian, but I haven't a status function in asterisk script. Any one help me to know how can I check if asterisk is up? If I switch off master machine or I cut network cable, second machine goes up OK, but if I switch on or replug cable in Main machine, all works fine but I realize that slave machine doesn't down asterisk. Any one has installed this system? regards, tron ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Linux-HA
Heartbeat monitors only 'life' of cluster members. Service should be monitored by a custom script. Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Richardson Sent: Thursday, January 19, 2006 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Linux-HA Doesn't heartbeat take care of this? It's been awhile since I've configured it. If two servers join back together as master, one of them shuts down its services. Maybe I'm just wishfully thinking.. There's also a directive to determine if a secondary should fail back over to the master if it comes back up. All your 'shared' services should be controlled by heartbeat -- you shouldn't have to do anything except use the supplied init scripts. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Linux-HA
Then, say you that I must to do a script that check if asterisk is alive? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Mimmus Enviado el: jueves, 19 de enero de 2006 16:51 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk and Linux-HA Heartbeat monitors only 'life' of cluster members. Service should be monitored by a custom script. Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Richardson Sent: Thursday, January 19, 2006 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Linux-HA Doesn't heartbeat take care of this? It's been awhile since I've configured it. If two servers join back together as master, one of them shuts down its services. Maybe I'm just wishfully thinking.. There's also a directive to determine if a secondary should fail back over to the master if it comes back up. All your 'shared' services should be controlled by heartbeat -- you shouldn't have to do anything except use the supplied init scripts. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fritz card technology German *
Thanks for all the posts everyone So yes, we have 4 channels, so I'm going to need 2x Fritz cards -- but I would rather not have to apply patches just to get the two PCI cards to work in the same box The price difference between the cards you guys mentioned is interesting I have also heard about BERONET isdn cards? a single Beronet 4-channel card would suffice I think? Thing is, whatever the legacy system in place already is (this is not a fresh operation) must have some sort of minor PBX in place, where all the phones are plugged in. So I would have to remove that and could use a TDM card to plug the phones in? These phones, isdn etc -- probably aren't analog -- probably don't work with a TDM card right? So I think what you were suggesting John is ISDN channel cards and a TDM in the same machine? with * just bridging calls between the two? Interesting. :-S Chris Earle - Original Message - From: John Daragon [EMAIL PROTECTED] To: Chris Earle (CBL) [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 18, 2006 6:45 AM Subject: Re: [Asterisk-Users] Fritz card technology German * Chris Earle (CBL) wrote: Hi all, I've been working with * for a long time now, but only with analog FXS/FXO systems. I am venturing towards setting up a box in Germany now and I believe that requires a Fritz card? Do I even have to use the Fritz cards? Why not a Digium card The AVM Fritz card is a single connection (2 x 64 kbps) passive ISDN card. It's well supported by chan_capi, but running more than one of them in a PC requires a driver patch. You can't use a Digium card because Digium doesn't make an ISDN2 card. We have 2 ISDN lines ( -- 6 handsets) so I'm guessing that will require 2 Fritz PCI cards (they have 1 port only). Then there's some sort of channel bank that sends the calls out to the extensions. Does this make any sort of sense? By 2 lines I guess you mean 4 channels ? i.e. 4 simultaneous calls ? If you mean 2 channels, then you only need 1 fritz card. Could someone confirm with me that this is the right direction to go -- ISDN lines, Fritz cards/Asterisk box, Channelbank/telco-box, extension handsets.. On the handset side you could use a couple of TDM4xx cards, or just use SIP phones. jd -- John Daragon [EMAIL PROTECTED] argv[0] limited (Asterisk implementation consultancy) Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Linux-HA
On 1/19/06, Tron [EMAIL PROTECTED] wrote: Yes, I know, but I have I was think that heartbeat use status function in init asterisk script to check if asterisk is alive, but status function is for redhat.Are there any similar function in Debian?. And in respect of slave, when slave get all resources and master wakeup, maste request for all resources, slave give it all resources, but asterisk continues alive in slave. My questions are: What I need to say to heartbeat that asterisk is alive or dead and why when slave give all resources to master doesn't goes down itself asterisk service. My init script is on /etc/hearbeat/resource.d and in /etc/init.d/. Any idea? There's a check_asterisk.pl plugin availble for nagios that you be able to take some code/concepts from to achieve what you're looking for. -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Linux-HA
Yes, heartbeat is good at monitoring system and network availability, but to monitor applications as well, you need to jump through hoops and do some custom development. A shame really because without that it's useless. Also, heartbeat only works in a primary/secondary fashion. Ie you can't have both systems in your cluster serving asterisk at the same time. This is because you only have one virtual IP address, and when the master fails, the secondary gets the virtual IP adddress. Once again, useless if you want to load balance your asterisk systems. That's what we wanted, and why we didn't go down this path. You also might want to check and see what the effect is on Asterisk of having your IP address change without restarting asterisk. Doug. -Original Message- From: Tron [mailto:[EMAIL PROTECTED] Sent: Thursday, January 19, 2006 9:00 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Asterisk and Linux-HA Then, say you that I must to do a script that check if asterisk is alive? -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Mimmus Enviado el: jueves, 19 de enero de 2006 16:51 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Asterisk and Linux-HA Heartbeat monitors only 'life' of cluster members. Service should be monitored by a custom script. Mimmus -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Gary Richardson Sent: Thursday, January 19, 2006 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk and Linux-HA Doesn't heartbeat take care of this? It's been awhile since I've configured it. If two servers join back together as master, one of them shuts down its services. Maybe I'm just wishfully thinking.. There's also a directive to determine if a secondary should fail back over to the master if it comes back up. All your 'shared' services should be controlled by heartbeat -- you shouldn't have to do anything except use the supplied init scripts. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Incoming fax on voipbuster
Hello,
I'm trying to receive a fax to my inbound number from voipbuster.
Asterisk receives the call and starts the rxfax application successful,
but then nothing happens. The calling party is still hearing a ringing
tone, or sometimes nothing. Voicecalls are working correct and without
problems.
For testing I've add a local number (300) to the dialplan. When I call
this number everything works OK, I hear the fax beeps.
The config and the dialplan for the localnumber and the voipbuster are the
some. What could be wrong?
Using: Asterisk 1.2.1 / SpanDSP 0.2.pre22
My Settings:
--- sip.conf ---
[general]
port=5060
nat=yes
insecure=very
localnet=192.168.1.0/255.255.255.0
externip=myexternalip
dtmfmode=auto
disallow=all
allow=alaw
register = myusername:[EMAIL PROTECTED]/myphonenumber
[voipbuster]
type=peer
fromuser=myphonenumber
username=myusername
secret=mypassword
host=sip1.voipbuster.com
context=voipbuster
language=nl
-
--- dialplan ---
[voipbuster]
exten = myusername,1,Wait(10)
exten = myusername,2,Answer()
exten = myusername,3,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = myusername,4,SetVar([EMAIL PROTECTED])
exten = myusername,5,rxfax(${FAXFILE})
exten = myusername,6,system(/usr/local/sbin/mailfax ${FAXFILE}
${EMAILADDR} ${CALLERIDNUM} ${CALLERIDNAME})
[default]
exten = 300,1,Wait(10)
exten = 300,2,Answer()
exten = 300,3,SetVar(FAXFILE=/var/spool/asterisk/fax/${UNIQUEID}.tif)
exten = 300,4,SetVar([EMAIL PROTECTED])
exten = 300,5,rxfax(${FAXFILE})
exten = 300,6,system(/usr/local/sbin/mailfax ${FAXFILE} ${EMAILADDR}
${CALLERIDNUM} ${CALLERIDNAME})
-
Thanks for the help.
René
The Netherlands.
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[Asterisk-Users] DTMF # ?
Can the # be used as a valid key press for a user in a dial plan? if so how can the asterisk recognize it as a valid key press? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] phpagi
hello wath is wrong with this code???
$dbconn = mysql_connect(127.0.0.1,vladimir,vladimir);
mysql_select_db(cuentas);
$inDB = false;
//$link = mysql_connect(127.0.0.1, vladimir, vladimir) or die(Could
not connect);
//$db = mysql_select_db(cuentas, $link) or die(Could not select
database);
$rc = execute_agi( ANSWER );
sleep(1); // Wait for the channel to get created and RTP packets to be sent
// On my system the welcome you would only hear 'elcome' So I paused for 1
second
$rc = execute_agi( STREAM FILE user \\ );
$rc = execute_agi( STREAM FILE please-enter-the \0123456789\ );
// obtenga datos diga number.gsm espere x tiempo y espere 6 digitos.
if ( !$rc[result] )
$rc = execute_agi( GET DATA number 15000 6);
$usuario = $rc[result];
$rc = execute_agi( SAY DIGITS $usuario \\ );
if ( !$rc[result] )
$rc = execute_agi( GET DATA access-code 15000 4 \\ );
$codigo = $rc[result];
$rc = execute_agi( SAY DIGITS $codigo \\ );
$query = SELECT codigo, password from usuarios where codigo
='$usuario';
$tmp = mysql_query($query);
if(mysql_num_rows($tmp) 0) {
list($usuario2, $codigo2) = mysql_fetch_array($tmp);
}
$rc = execute_agi( SAY DIGITS $codigo2 \\ );
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[Asterisk-Users] (newbie) using dtmf during a call
hi, im complete new with asterisk, so.. i want to be able using dtmf during a call, for execute a application. Now i'm still making phonecalls trough a sip-adapter (Linksys pap2) with my soundcart from server, and i receive in the asterisk-console putting some digits from a analog-telephon: Console receive digit 1 e.g. Now how can i execute some application with this digit?. I have tried, to execute with a shellscript to exploit my applications from the asterisk-console but this console seems to be not really shell-compatible? I want to use it for a art-performance, so im using asterisk not for a company. ..and sorry for my bad english thanks for any hints for a very asterisk-newbie Moritz Wettstein ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Connection TDM400P to UK PSTN
Okay, sorry to hash out this discussion again, but it's starting to drive me crazy Successfully got the adapters to allow the BT phones to ring on lines coming out of a TDM.. but now my latest problem is echo. I have done tweaking of the gains in North and South America, and after a bit of work have gotten echo to go away, but this seems to just not want to go away. On an incoming call from the POTS, everything on my end sounds perfect, but on the internal extension phone, there is an echo when you speak. An almost perfect copy of what you say. If I turn down the gains on that channel, it doesn't seem to do much, or causes other volume issues. Help! In my research and hunting, I am starting to worry that the US-bought digium cards have IMPEDENCE issues in the UK with the BT Lines etc? That would seem to explain why the echo is so incessant. I have even tried changing Echo Cancellers to MARK3. Right, this is Asterisk 1.0.9, Zaptel 1.0.9.2 on Debian Suggestions / Experiences in UK appreciated -- Chris Earle System Solutions Specialist, - Original Message - From: John Novack [EMAIL PROTECTED] Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Wednesday, August 24, 2005 10:03 AM Subject: Re: Connection TDM400P to UK PSTN The jacks on the TDM are ( incorrectly ) referred to as RJ45, correctly they are 8 position modular. The line, either in or out is on the two CENTER pins. NONE of the other 6 pins are used. Though I am not in the UK, from what I know you don't use the two center pins for a single line connection, so you will need to fashion some sort of adapter to connect. Frankly, using the two center pins ( A Bell System brain blizzard) wasn't the smartest idea. It makes the modular plug into, with the addition of just a little moisture, a really good spark gap when a ring signal or small induction of lightning is applied. I have seen many a modular plug turned black and useless since the introduction of modular in the US in the early 70's Good luck John Novack Graham Kiff wrote: I'm a complete Asterisk novice and have an installation based on the [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]CD. I've installed my TDM400P with 2 x FXO 2 x FXS, but every time I try to dial out, I get a message No circuits available. Can someone confirm the pinouts for connecting the FXO's to a UK BT Line - I have RJ11 connectors on the back of my TDM400P card, so ideally I'd like to know the pin mappings from a standard BT plug to RJ11. Cheers Graham ___ 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 -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PHPAGI
i have installed [EMAIL PROTECTED] i need install more soft? odbc etc...?, i think if [EMAIL PROTECTED] have inside the phpmyadmin i dont need more installed, this is true? I don't use [EMAIL PROTECTED] , but if phpmyadmin is installed you be pretty sure that you have all you need to connect to a mysql database. Mark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dell PowerEdge 830 server
I don't know, I only tested it with a single TDM400. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tron Sent: Thursday, January 19, 2006 3:28 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell PowerEdge 830 server Have you seen if this equipment share IRQ for the resto of PCI Slots. I want to install one TDM2400P with 24 FXS Port and one TDM04B with 4 FXO ports but I want to know if that equipment has voltage connector for TDM2400P and it doesn't share IRQ in two PCI Slots. regards, tron -Mensaje original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Kerry Garrison Enviado el: miércoles, 18 de enero de 2006 5:38 Para: 'Asterisk Users Mailing List - Non-Commercial Discussion' Asunto: RE: [Asterisk-Users] Dell PowerEdge 830 server To be specific, I installed [EMAIL PROTECTED] 2.2 which is CentOS 4.2, Asterisk 1.2.1, Asterisk Management Portal, Flash Operator Panel, etc etc. That site has about 15 users with half of them having both on-site and off-site extensions (setup using AMP's Users and Devices mode). This site is not using any real-time functions. They do use the meet-me rooms fairly heavily. The system has a TDM400 with 4 FXO ports on it and the phone lines are in a hunt group that does a rollover on the 5th call to Teliax on the pay as you go plan which provides 10 additional channels. Does that help? Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jennifer Hales Sent: Tuesday, January 17, 2006 7:55 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell PowerEdge 830 server Hello Kerry, Many thanks for your information. Do you mind giving some more details on your setup? What version of Asterisk are you using? How many users do you have? Are you using real-time? And what Asterisk features are you providing? Feel free to reply off list if you wish. Kind regards Jenn Hales -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kerry Garrison Sent: Wednesday, January 18, 2006 1:29 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Dell PowerEdge 830 server Go into the BIOS, disable all unneeded peripherals like floppy controller, serial ports, parallel ports, etc. It should work fine, I have one at a decent sized installation. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jennifer Hales Sent: Tuesday, January 17, 2006 5:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [Asterisk-Users] Dell PowerEdge 830 server Hello all, We are looking at using a Dell PowerEdge 830 Server for an Asterisk installation. Does anyone have experience using this server with Asterisk? Any feed back would be appreciated. Kind regards Jenn ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dropping incompatible voice frame
On Thursday 19 January 2006 13:48, Joseph Rothstein wrote: I am now getting these messages on a second box running a different version of Asterisk. If anyone has any idea what is causing these, or how to avoid them I would be very grateful. 157 Jan 19 11:59:50 NOTICE[6070]: Dropping incompatible voice frame on Local/[EMAIL PROTECTED],2 of format ulaw since our native format has changed to alaw I have had a similar issue but was saying :of format slin since our native format has changed to ulaw whatever. The problem was: wrong configuration of FXO port dialplan(spa3000). Kind of - simultaneous use of PSTN dialplan and Call Forward Settings on User tab... This is just a guess since your info is not enough. benchev ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] (newbie) using dtmf during a call
not sure if i get your point, but may be something like this:
exten = s,1,Answer()
exten = s,2,Background(dial_some_digits);
exten = _X.,MyApplication(${EXTEN})
where MyApplication can be a custom application, and ${EXTEN} is a
magic variable that will hold the dialed digits
regards
On 1/19/06, moritz [EMAIL PROTECTED] wrote:
hi, im complete new with asterisk, so..
i want to be able using dtmf during a call, for execute a application.
Now i'm still making phonecalls trough a sip-adapter (Linksys pap2) with
my soundcart from server, and i receive in the asterisk-console putting
some digits from a analog-telephon: Console receive digit 1 e.g.
Now how can i execute some application with this digit?.
I have tried, to execute with a shellscript to exploit my applications
from the asterisk-console but this console seems to be not really
shell-compatible?
I want to use it for a art-performance, so im using asterisk not for a
company.
..and sorry for my bad english
thanks for any hints for a
very asterisk-newbie
Moritz Wettstein
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Re: [Asterisk-Users] Re: Connection TDM400P to UK PSTN
Chris: I had the same problem and gave up. (Gloucestershire) If I have the gains right down low (just enough so that the DTMF tones are recognised), the echo is acceptable, but audio at the far end is very low. I think that forwarding my POTS line to the VOIP line is the only sensible option, or back to my original thought of using an SPA3000. BTW, I tried building a matching network, briefly, but failed with that too. For the moment, I'm just using Asterisk for VOIP - but very pleased with that. Roger Chris Earle (CBL) wrote: Okay, sorry to hash out this discussion again, but it's starting to drive me crazy Successfully got the adapters to allow the BT phones to ring on lines coming out of a TDM.. but now my latest problem is echo. I have done tweaking of the gains in North and South America, and after a bit of work have gotten echo to go away, but this seems to just not want to go away. On an incoming call from the POTS, everything on my end sounds perfect, but on the internal extension phone, there is an echo when you speak. An almost perfect copy of what you say. If I turn down the gains on that channel, it doesn't seem to do much, or causes other volume issues. Help! In my research and hunting, I am starting to worry that the US-bought digium cards have IMPEDENCE issues in the UK with the BT Lines etc? That would seem to explain why the echo is so incessant. I have even tried changing Echo Cancellers to MARK3. Right, this is Asterisk 1.0.9, Zaptel 1.0.9.2 on Debian Suggestions / Experiences in UK appreciated -- Chris Earle System Solutions Specialist, - Original Message - From: John Novack [EMAIL PROTECTED] Newsgroups: gmane.comp.telephony.pbx.asterisk.user Sent: Wednesday, August 24, 2005 10:03 AM Subject: Re: Connection TDM400P to UK PSTN The jacks on the TDM are ( incorrectly ) referred to as RJ45, correctly they are 8 position modular. The line, either in or out is on the two CENTER pins. NONE of the other 6 pins are used. Though I am not in the UK, from what I know you don't use the two center pins for a single line connection, so you will need to fashion some sort of adapter to connect. Frankly, using the two center pins ( A Bell System brain blizzard) wasn't the smartest idea. It makes the modular plug into, with the addition of just a little moisture, a really good spark gap when a ring signal or small induction of lightning is applied. I have seen many a modular plug turned black and useless since the introduction of modular in the US in the early 70's Good luck John Novack Graham Kiff wrote: I'm a complete Asterisk novice and have an installation based on the [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]CD. I've installed my TDM400P with 2 x FXO 2 x FXS, but every time I try to dial out, I get a message No circuits available. Can someone confirm the pinouts for connecting the FXO's to a UK BT Line - I have RJ11 connectors on the back of my TDM400P card, so ideally I'd like to know the pin mappings from a standard BT plug to RJ11. Cheers Graham ___ 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 -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fritz card technology German *
Chris Earle (CBL) wrote: So yes, we have 4 channels, so I'm going to need 2x Fritz cards -- but I would rather not have to apply patches just to get the two PCI cards to work in the same box Than don't use two Fritz! cards, but two hfc-s cards, or the 4-port cards mentioned below. The price difference between the cards you guys mentioned is interesting I have also heard about BERONET isdn cards? a single Beronet 4-channel card would suffice I think? It is not a 4-channel, but a 4-port card, i.e. 8-channels, just like the junghanns and sirrix cards. Thing is, whatever the legacy system in place already is (this is not a fresh operation) must have some sort of minor PBX in place, where all the phones are plugged in. So I would have to remove that and could use a TDM card to plug the phones in? These phones, isdn etc -- probably aren't analog -- probably don't work with a TDM card right? So I think what you were suggesting John is ISDN channel cards and a TDM in the same machine? with * just bridging calls between the two? An even better approach is the one outlined on the junghanns.net. Use the existing PBX and plug it into the other two ISDN ports of the 4-port beronet/junghanns/Sirrix card, and use asterisk as the middleman. Adding new phones would be done either as extensions of the old PBX, or as SIP-phones to asterisk -- Best regards Peer Oliver Schmidt PGP Key ID: 0x83E1C2EA ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] phpagi
$dbconn = mysql_connect(127.0.0.1,vladimir,vladimir);
snip
kinda looks ok, but I would seriously consider putting as much code as you
can into the dialplan rather than forking a PHP script for all the user
interaction.
For example, I use asterisk to handle loads of fax 2 email accounts. on 4
E1's. Each E1 goes to
[incoming]
exten = _XXX.,1,Answer
exten = _XXX.,2,SetAccount(${DNID})
exten = _XXX.,3,AGI(callhandle.php)
exten = _XXX.,4,GotoIf($[${application} = none]?6:5)
exten = _XXX.,5,Goto(${application},s,1)
exten = _XXX.,6,Hangup
The callhandle.php script looks in a database for the number that's being
called and sets variables like application, email address (and speed --
that's still buggy at the moment)
I set up further applications or contexts to handle the calls
[unknownnumber]
exten = s,1,Playback(unknownnumber)
exten = s,2,Hangup
[fax2email]
exten = s,1,SetAccount(${DNID})
exten = s,n,Wait(2)
exten = s,n,Wait(2)
exten = s,n,Wait(2)
exten = s,n,Wait(2)
exten = s,n,Wait(2)
exten = s,n,Hangup
exten = fax,1,Set(FAXFILE=/var/lib/asterisk/fax/${UNIQUEID}.tif|${SPEED})
exten = fax,2,rxfax(${FAXFILE})
exten = fax,3,Hangup
exten = h,1,System(/var/lib/asterisk/agi-bin/email.php ${EMAIL} ${FAXFILE}
${RESCALE})
Now, I have only been use asterisk for about 6 months and this may not be
the best solution, but when I started I tried to get *everything* into the
PHP scripts and just use the dialplan for as little as I could. It just
didn`t work, I wasted hours on it. I guess some mistakes you just have to
learn yourself.
Some simple advice (that I have learnt) :
use the dialplan as much as possible.
develop and test your scripts outside of asterisk before using them.
bookmark http://www.voip-info.org/tiki-index.php?page=Asterisk , I refer to
it everyday.
Mark
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Re: [Asterisk-Users] (newbie) using dtmf during a call
http://www.voip-info.org/wiki-Asterisk+config+features.conf2006/1/19, Moises Silva
[EMAIL PROTECTED]:not sure if i get your point, but may be something like this:
exten = s,1,Answer()exten = s,2,Background(dial_some_digits);exten = _X.,MyApplication(${EXTEN})where MyApplication can be a custom application, and ${EXTEN} is amagic variable that will hold the dialed digits
regardsOn 1/19/06, moritz [EMAIL PROTECTED] wrote: hi, im complete new with asterisk, so.. i wantto be able using dtmf during a call, for execute a application.
Now i'm still making phonecalls trough a sip-adapter (Linksys pap2) with my soundcart from server, and i receive in the asterisk-console putting some digits from a analog-telephon: Console receive digit 1
e.g. Now how can i execute some application with this digit?. I have tried, to execute with a shellscript to exploit my applications from the asterisk-console but this console seems to be not really
shell-compatible? I want to use it for a art-performance, so im using asterisk not for a company. ..and sorry for my bad english thanks for any hints for a very asterisk-newbie
Moritz Wettstein ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list
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Re: [Asterisk-Users] Fritz card technology German *
Chris Earle (CBL) wrote: I have also heard about BERONET isdn cards? a single Beronet 4-channel card would suffice I think? Yes. Beronet and Junghanns both have the same cards. (they just 'work' different, junghanns uses zap interfaces, beronet mISDN) So, as already mentioned, you have 2 good options: - 4x BRI card (Beronet or junghanns) - 2x HFC PCI card (uses zap, and are cheap!) Regarding the phones, I only use sip phones, so no idea on that.. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Sound Issue
On Wed, 18 Jan 2006, Kevin wrote: Thanks for the reply Steve, I was able to try what you suggested, but no, that did not solve the issue. Now, I didn't think about this before (and this might sound dumb, but I am new to asterisk) but the phones I am using right now are all on SIP, so they are coming in over the Ethernet to the server. The T1 card right now isn't even plugged in since I was just setting up the dial plan before we installed anything. Should something be plugged in order to get the sound to work since, from what I understood, asterisk is looking for the clock cycle from zaptel? Or is there another way to disable that besides moving chap_zap.so? Hi, You probably need to change your span lines so that all the second parameters are 0 - in other words, don't try to get clock off the ports. I think that will get it going. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF Simultaneous Inband and RFC2833 performedby Asterisk = Duplicate tones
On Thu, 19 Jan 2006, Andreas Sikkema wrote: The same thing can happen when a SIP ATA is configured to use rfc2833 but is also a little to lote with the filtering out of the DTMF. So sometimes it's not Asterisks fault at all ;-) And then there's some IVR's that don't notice it at all, while others are totally unusable. Obviously we can't do anything about the ATA. But I did have an idea for Asterisk... Which is that when the DTMF detector spots DTMF in an audio frame, that it passes along on the channel a kind of merged frame which is both the detected DTMF AND the actually audio. Then, when the frame arrives at an application of a channel that wants the indication, it can use that. On the other hand, if the frame arrives at a channel that needs to send inband DTMF, the indication can be ignored and the original audio passed on instead. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF # ?
It's mapped to blind transfer in features.conf -- If you want to use the blind transfer feature, which I find easier than my phones' transfer features, remap it to ## in features.conf. That way if you hit # it dtmfs through to the target IVR, but you can hit ## real quick to get the transfer function. Or maybe I could refer you to the notes in the wiki: ;) http://www.voip-info.org/wiki-Asterisk+config+features.conf Using the blindxfer in [featuremap] section you can redefine the transfer key. For example, if the blindxfer is set to ##, transfer only happens when you press the # key twice very quickly. This solves a problem using Asterisk phones to call IVR systems such as those used by banks and credit card companies - Enter you account number followed by the # key. Moj chris songer wrote: Can the # be used as a valid key press for a user in a dial plan? if so how can the asterisk recognize it as a valid key press? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Brief silences during calls
check irq supply on the * server -- When you run zttest do you maintain over 98%? or like Steve suggested, the network may have congestion or other errors ethereal may help you figure out. I had a polycom 500 that was doing this to my user, 301s and 501s wouldn't do it. Not sure if that was network issues or something with the polycom itself. Moj Mimmus wrote: Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with rxfax - Dropping incompatible voice frame?
Hi, I'm having problems with the rxFax app. One of the messages that appear in my console is: Executing Set(SIP/something, FAXFILE=/var/spool/asterisk-fax/1137692307.5.tif) in new stack -- Executing RxFAX(SIP/something, /var/spool/asterisk-fax/1137692307.5.tif) in new stack Jan 19 12:38:30 NOTICE[12008]: channel.c:1906 ast_read: Dropping incompatible voice frame on SIP/something of format slin since our native format has changed to ulaw Dropping incompatible voice frame on SIP/something of format slin since our native format has changed to ulaw This seemed particularly important, but I can't really say whyCould this be why my faxes are often interrupting during transmission and giving me errors on my PSTN fax machine that is used for sending the fax? Mike ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfer and zap
Hello, some problems with transfer and zap... one hfc-card in NT mode and one fritz isdn-card in server. there is one gigaset SX353 isdn phone on the hfc-card. anybody calls from external via capi and the call is bridged to the zap-device. if you want to transfer the call via R-button on the isdn-phone the caller get the music-on-hold. you get a dialtone and dial - if the called person gets on phone - i will hang up the phone. but the call did'nt transfer - the moh to the first caller will not stop. how can i transfer the call? i want to transfer back the origin call to the asterisk-server in an extension for faxtransfer, so if anyone call and a faxtone is there i want transfer it back, so that asterisk answer the call. any ideas? sorry for my bad english ;-) Marcel Pennewiss ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound issue with Asterisk
Hey Steve and everyone, I looked at the configuration, and unless I am missing something I don't think they are configured # ztcfg -vv Zaptel Configuration == Channel map: 0 channels configured. In the zapata.conf file, it is the sample version, but I didn't notice anything in there that related to what you said. Or is it in a different file or location? I am in the office now so I am able to provide some more information about the issue that I am having. Here is the kernel if this helps Fedora core 4 -- 2.6.11-1.1369_FC4smp I know that ztdummy is at least loaded now. Also as stated before there is nothing plugged into the T1 card. So I wasn't sure if that was causing a problem or not which is why I enabled ztdummy but it was not the first time I e-mailed you. # lsmod | grep ztdummy ztdummy 7748 0 zaptel192516 6 ztdummy,wct4xxp If I look at the connections from tcpdump, I see my phone call coming in, but no traffic is being sent back to the phone. With an Echo() test, I see the traffic going back and forth, but when I call into a menu, then there is nothing. Thanks, Kevin I ran a sip debug as well but I felt it was better at the end of the e-mail: -- SIP read from 64.7.189.14:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.7.161.26:5060;branch=z9hG4bK2b87d1ab;rport From: asterisk sip:[EMAIL PROTECTED];tag=as4a36d77b To: sip:[EMAIL PROTECTED];tag=a0efbf44ecab5900 Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Grandstream BT100 1.0.6.7 Contact: sip:[EMAIL PROTECTED] Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Supported: replaces Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' -- SIP read from 64.7.189.14:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Supported: replaces Call-ID: [EMAIL PROTECTED] CSeq: 19606 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 337 v=0 o=budgeTone-PubIP 8000 8000 IN IP4 64.7.189.14 s=SIP Call c=IN IP4 64.7.189.14 t=0 0 m=audio 5004 RTP/AVP 2 8 4 18 15 97 9 a=sendrecv a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:15 G728/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 --- (13 headers 16 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 64.7.189.14 : 5060 (non-NAT) Found peer 'budgeTone-PubIP' Reliably Transmitting (no NAT) to 64.7.189.14:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c;received=64.7.189.14 From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588 To: sip:[EMAIL PROTECTED];tag=as6f00184d Call-ID: [EMAIL PROTECTED] CSeq: 19606 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Proxy-Authenticate: Digest realm=asterisk, nonce=351ca5f6 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from 64.7.189.14:5060: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKa4b619218b6ad43c From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588 To: sip:[EMAIL PROTECTED];tag=as6f00184d Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 19606 ACK User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Length: 0 --- (11 headers 0 lines)--- -- SIP read from 64.7.189.14:5060: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bK1b2220ace977c3a7 From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Supported: replaces Proxy-Authorization: Digest username=budgeTone-PubIP, realm=asterisk, algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=351ca5f6, response=3748b6120c7f4ecc4873cbdaf178d507 Call-ID: [EMAIL PROTECTED] CSeq: 19607 INVITE User-Agent: Grandstream BT100 1.0.6.7 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Content-Type: application/sdp Content-Length: 337 v=0 o=budgeTone-PubIP 8000 8001 IN IP4 64.7.189.14 s=SIP Call c=IN IP4 64.7.189.14 t=0 0 m=audio 5004 RTP/AVP 2 8 4 18 15 97 9 a=sendrecv a=rtpmap:2 G726-32/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=rtpmap:15 G728/8000 a=rtpmap:97 iLBC/8000 a=fmtp:97 mode=20 a=rtpmap:9 G722/16000 a=ptime:20 --- (14 headers 16 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 64.7.189.14 : 5060 (non-NAT) Found peer 'budgeTone-PubIP' Found RTP audio format 2 Found RTP audio format 8 Found RTP audio
Re: [Asterisk-Users] Brief silences during calls
Just look through the devices settings for suppress silence or transmit silence and don't supress or prevent transmission... this is a common problem inX-LiteRobOn 1/19/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: check irq supply on the * server -- When you run zttest do you maintainover 98%?or like Steve suggested, the network may have congestion orother errors ethereal may help you figure out.I had a polycom 500 that was doing this to my user, 301s and 501s wouldn't do it.Not sure if that was network issues or something withthe polycom itself.MojMimmus wrote: Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users --Mojo [EMAIL PROTECTED]Office Manger, Horan Company, LLC(907) 747- x112___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: MeetMe Listen Only flag (|m)
In article [EMAIL PROTECTED], Dan Austin [EMAIL PROTECTED] wrote: Tony wrote: I should tidy it up and submit it, but haven't got round to it :-( Let us know if you can. I'm already maintaining a grocery list of patches to make MeetMe viable in my orginization, so one more won't kill me. I should be able do so this weekend. That's the plan, anyway :-) I'll post the Mantis bug# when I've submitted it. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SAN Devices
Adam Robins wrote: Anyone out there using small-midsized (2-4 TB) SAN solution among multiple Asterisk systems? I don't have the budget for an EMC-caliber solution, and can't seem to find much else out there. I designed a virtualized san and have been running it in production for the last two years... Speaking from experience... stay away from EMC! We have several storage systems in production.. from multiple vendors... and I've had nothing but problems with the CX line of emc systems. Performance problems... hardware/crashing problems.. (they run embedded xp you know) and dead fibre port problems. If I didn't have two of everything.. mirroring across cabinets with IpStor we would have had serious problems. Just my two cents on the issue of 'EMC-caliber' storage... Jared ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Disabling zap echo cancellor from dialplan
Anybody knows if it's possible to disable zap echo cancellor from dialplan only for certain outbound calls ?? I share the same phone lines for voice calls and faxes. Iaxmodem works fine for me only turning off the echo cancellor, but I need it for voice calls. Any ideas ? maxx ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with rxfax - Dropping incompatible voice frame?
On Thu, 19 Jan 2006, [iso-8859-1] Michaël Gaudette wrote: Executing Set(SIP/something, FAXFILE=/var/spool/asterisk-fax/1137692307.5.tif) in new stack -- Executing RxFAX(SIP/something, /var/spool/asterisk-fax/1137692307.5.tif) in new stack Jan 19 12:38:30 NOTICE[12008]: channel.c:1906 ast_read: Dropping incompatible voice frame on SIP/something of format slin since our native format has changed to ulaw FAXing doesn't work over Voice-over-IP channels. Or, hardly ever works. Or, works occasionally but unreliably. Check the spandsp FAQ. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Linux-HA
On Thu, Jan 19, 2006 at 04:41:29PM +0100, Tron wrote: Yes, I know, but I have I was think that heartbeat use status function in init asterisk script to check if asterisk is alive, but status function is for redhat.Are there any similar function in Debian?. And in respect of slave, when slave get all resources and master wakeup, maste request for all resources, slave give it all resources, but asterisk continues alive in slave. My questions are: What I need to say to heartbeat that asterisk is alive or dead and why when slave give all resources to master doesn't goes down itself asterisk service. My init script is on /etc/hearbeat/resource.d and in /etc/init.d/. The standard init.d scripts of Debian don't have status functions. However the latest version in Unstable has a home-brewed one. You can get it directly from the SVN: the init.d script: http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/asterisk.init?op=filerev=0sc=0 Its default config file: http://svn.debian.org/wsvn/pkg-voip/asterisk/trunk/debian/asterisk.default?op=filerev=0sc=0 And generally the pkg-voip team is also a place to ask questions regarding the integration oof Asterisk and Debian. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disabling zap echo cancellor from dialplan
Massimo De Nadal wrote: Anybody knows if it's possible to disable zap echo cancellor from dialplan only for certain outbound calls ?? I share the same phone lines for voice calls and faxes. Iaxmodem works fine for me only turning off the echo cancellor, but I need it for voice calls. There is not currently any way to do that, but the echo canceller should turn itself off when it hears the magic tone that FAX machines generate for that very purpose. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Disabling zap echo cancellor from dialplan
On Thursday 19 January 2006 12:52, Massimo De Nadal wrote: I share the same phone lines for voice calls and faxes. Iaxmodem works fine for me only turning off the echo cancellor, but I need it for voice calls. Any ideas ? IAXModem (and the device you're connecting to) should be sending out the echo canceller disable tone (standard tone, I forget the specific standard body # it is referenced in) -- I have my fax machines, security system and stamp machine work all the time this way, and you can see in the dmesg output that the echo canceller for channel 'x' is disabled due to the tone. Make sure you didn't disable the tone detection in zconfig.h. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] help
) Transmitting (no NAT) to 64.7.189.14:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 64.7.189.14;branch=z9hG4bKd592db0d7856fb18;received=64.7.189.14 From: Budge Tone sip:[EMAIL PROTECTED];tag=b72941c93fe74588 To: sip:[EMAIL PROTECTED];tag=as324cfd6f Call-ID: [EMAIL PROTECTED] CSeq: 19608 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Max-Forwards: 70 Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- == Spawn extension (local, 500, 1) exited non-zero on 'SIP/budgeTone-PubIP-7e44' Destroying call '[EMAIL PROTECTED]' -- Message: 6 Date: Thu, 19 Jan 2006 19:47:19 +0200 From: Rob Lith [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Brief silences during calls To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=iso-8859-1 Just look through the devices settings for suppress silence or transmit silence and don't supress or prevent transmission... this is a common problem inX-Lite Rob On 1/19/06, Mojo with Horan Company, LLC [EMAIL PROTECTED] wrote: check irq supply on the * server -- When you run zttest do you maintain over 98%? or like Steve suggested, the network may have congestion or other errors ethereal may help you figure out. I had a polycom 500 that was doing this to my user, 301s and 501s wouldn't do it. Not sure if that was network issues or something with the polycom itself. Moj Mimmus wrote: Where can I investigate the origin of brief silences during calls from/to my SIP phone? Only rare pauses of 0.5-1.0 sec when I'm not able to hear anything. Thanks Mimmus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mojo [EMAIL PROTECTED] Office Manger, Horan Company, LLC (907) 747- x112 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060119/22 939450/attachment.html -- Message: 7 Date: Thu, 19 Jan 2006 17:47:29 + (UTC) From: [EMAIL PROTECTED] (Tony Mountifield) Subject: [Asterisk-Users] Re: MeetMe Listen Only flag (|m) To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] In article [EMAIL PROTECTED], Dan Austin [EMAIL PROTECTED] wrote: Tony wrote: I should tidy it up and submit it, but haven't got round to it :-( Let us know if you can. I'm already maintaining a grocery list of patches to make MeetMe viable in my orginization, so one more won't kill me. I should be able do so this weekend. That's the plan, anyway :-) I'll post the Mantis bug# when I've submitted it. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org -- Message: 8 Date: Thu, 19 Jan 2006 12:52:36 -0500 From: Jared Watkins [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] SAN Devices To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Adam Robins wrote: Anyone out there using small-midsized (2-4 TB) SAN solution among multiple Asterisk systems? I don't have the budget for an EMC-caliber solution, and can't seem to find much else out there. I designed a virtualized san and have been running it in production for the last two years... Speaking from experience... stay away from EMC! We have several storage systems in production.. from multiple vendors... and I've had nothing but problems with the CX line of emc systems. Performance problems... hardware/crashing problems.. (they run embedded xp you know) and dead fibre port problems. If I didn't have two of everything.. mirroring across cabinets with IpStor we would have had serious problems. Just my two cents on the issue of 'EMC-caliber' storage... Jared -- Message: 9 Date: Thu, 19 Jan 2006 18:52:35 +0100 From: Massimo De Nadal [EMAIL PROTECTED] Subject: [Asterisk-Users] Disabling zap echo cancellor from dialplan To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-15; format=flowed Anybody knows if it's possible to disable zap echo cancellor from dialplan only for certain outbound calls ?? I share the same phone lines for voice calls and faxes. Iaxmodem works
[Asterisk-Users] Automatic redial on Hangup
I know this is a somewhat odd application, but we have a very good
reason for needing it.
Basically, I want asterisk to automatically redial a caller unless they
exit the system properly.
Here are some pertinate sections of the dialplan.
[AUTOBCSTART]
EXTEN=001,1,Meetme9${ENC}|pq)
EXTEN=001,2,Goto(CLEANEXIT|001|1)
EXTEN=h,1,Goto(AUTOREDIAL|S|1) -I've also used dial(LOCAL/..)
[AUTOREDIAL]
EXTEN=s,1,ChanIsAvail9ZAP/g1)
EXTEN=s,2,Dial(Zap/g1/${CID}
EXTEN=s,3,Goto(AUTOBCSTART|001|1)
EXTEN=s,102,NoOp(${CID})
EXTEN=s,103,Dial(IAX2/myuser@teliax/1${CID}|30|TtG(AUTOBCSTART^001^1))
Some notes.
The CID variable is set earlier and does pass through.
The problem is that asterisk attempts the dial and imediately hangs up
the IAX channel. I am not so worried about the ZAP section in AUTOREDIAL
because there is nothing hooked up yet. It will match IAX dial string
when it works. I have pasted relevant CLI output below (slightly edited
for security). I know the dial command works if I run it directly from
an extension. I want the caller to be automatically reconnected to the
conference unless the caller leave the conference cleanly. If there is
another way to to this, I would entertain suggestions.
-- Executing MeetMe(IAX2/teliax-4, 1|pq)
== Parsing '/etc/asterisk/meetme.conf': Found
-- Created MeetMe conference 1023 for conference '1'
-- Hungup 'Zap/pseudo-1195863188'
== Spawn extension (AUTOBCSTART, 001, 14) exited non-zero on
'IAX2/teliax-4'
-- Executing Dial(IAX2/teliax-4, LOCAL/[EMAIL PROTECTED])
-- Called [EMAIL PROTECTED]
== Spawn extension (AUTOBCSTART, h, 1) exited non-zero on 'IAX2/teliax-4'
-- Hungup 'IAX2/teliax-4'
-- Executing ChanIsAvail(Local/[EMAIL PROTECTED],2, Zap/g1)
-- Executing NoOp(Local/[EMAIL PROTECTED],2, returnscorrect)
-- Executing Dial(Local/[EMAIL PROTECTED],2,
IAX2/myuser@teliax/thevariablevalue|30|TtG(AUTOBCSTART^003^1))
-- Called [EMAIL PROTECTED]/17655460916
-- Hungup 'IAX2/teliax-3'
== Spawn extension (AUTOREDIAL, s, 103) exited non-zero on
'Local/[EMAIL PROTECTED],2'
Thanks
BEN
___
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[Asterisk-Users] Zapata.conf and Realtime
I asked this a few days ago, and haven't gotten an answer (or seen my message in the archive, yet). Since there were some email problems the other day, I will just pose the question again. I would like to know if there is a way to have a table, like zapata_conf in a DB, and have asterisk realtime pull the information out, like it does for voicemail, sip.conf, and iax.conf, etc. If anyone has done this and has a schema that I could use, I would be very happy. TYVMIA Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SAN Devices
Adam Robins wrote: Anyone out there using small-midsized (2-4 TB) SAN solution among multiple Asterisk systems? I don't have the budget for an EMC-caliber solution, and can't seem to find much else out there. http://www.coraid.com/ for a slightly different approach to large storage capacity. Rod -- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AGI Tx error 510
My php script called from my extensions.conf is working fine, however each time I run it with agi debug mode on I see the message AGI Rx Done AGI Tx 510 Invalid or unknown command which does not come from my script, apart from the Rx Done where the 'Done' is the message from my exit() statement. If I omit the exit() statement I get AGI Rx AGI Tx 510 Invalid or unknown command AGI Rx AGI Tx 510 Invalid or unknown command output after the last transmission to STDERR. It appears to be cosmetic, just wondering if I have anything to be concerned about here? -- Colin Beckingham (613) 374-1391 Skype: it4gh_ http://www.it4gh.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fw: chanspy
Hi, I was only able to ChanSpy Agent channels. How do I monitor outgoing calls? Thank youDov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Three Way Calling with HFC PCI Card
Hello, On Di, 2006-01-03 at 16:31 +0100, Giovanni Miano wrote: Use meetme app Unfortunately meetme is no solution for me. If nobody can help me, is there at least anybody who has the same problem? As far as I can see there are lots of people using the HFC PCI card, is nobody using Three-Way-Calling? It would be really helpful to know if the problem is with zaptel +asterisk or just with my setup. Thanks in advance :) Henry -- Hi! I'm a .signature virus! Copy me into your ~/.signature to help me spread! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re:[Asterisk-Users] Sound issue with Asterisk
Hi everyone and Steve, Well the problem I wrote about is fixed. Here is what I did to resolve the issue. I was running kernel 2.6.11-1.1369_FC4smp before. I went and upgraded to 2.6.14-1.1656_FC4smp along with the development files (which I finally found were not installed). After I installed the new kernel, recompiled asterisk, added my SIP and Extensions back in and all the menus in the demo, echo, etc, were all working. Thanks Steve for helping me and hopefully this will help someone out as well. Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] loading zaptel drivers automatically upon reboot
You could try removing the asterisk config, and zaptel config from boot. and just placing it in the /etc/rc.local file i did this for a while modprobe zaptel modprobe wcfxo. etc etc... sleep 2 /usr/sbin/asterisk Yes, I did exactly that, but when I boot zaptel doesn't load wct1xxp. Then asterisk dies with error code 1. But if I stop the service zaptel, and start it again, then zaptel loads wct1xxp, and everything works fine. Then as I said, I don't get why zaptel doesn't load wct1xxp when boots. Regards, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley Sent: Tuesday, January 17, 2006 9:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] loading zaptel drivers automatically upon reboot I have always just used make config.. then edit /etc/sysconfig/zaptel to comment out the modules that i dont need. On Sat, Jan 14, 2006 at 03:55:50PM -0500, Carlos Alperin wrote: After install everything on the supposedly right place, my conclusion is that zaptel doesn't load wct1xxp module. That's easy to test: before you restart zaptel, look at /proc/zaptel . if /proc/zaptel exists, zaptel was loaded . if /proc/zaptel/1 exists and reports those 24 channels, then wct1xxp has loaded and identified your card. Another possible reason: make sure that the zaptel init.d script runs before the asterisk one. It needs to have a lower start number. Use 'chkconfig --list asterisk' and 'chkconfig --list zaptel' to verify that. Then, that is the reason for Asterisk to fail loading. However I change the MODULES RMODULES on the zaptel on /etc/init.d /etc/sysconfig, it continuous same way. Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Saturday, January 14, 2006 2:36 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] loading zaptel drivers automatically upon reboot On Fri, Jan 13, 2006 at 09:39:09PM -0500, Carlos Alperin wrote: That is right for zaptel. But you still has to do modprobe wctdm on rc.local before to load asterisk. rc.local is run after the standard init.d scripts. Thus if you load asterisk in an init.d script, you'd be loading the zaptel modules too late. Just add another init.d script. See the skeleton in /etc/init.d (there's a README there IIRC). -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do you deal with subprefixes with LCR?
Hi List, I am working on least cost routing code on the moment, and I am stumbling on a problem. Say you have provider A having: Prefix XXX0.10 Prefix XXXYYY 0.20 And provider B having Prefix XXX0.15 You're stuck, because you cannot decide if provider B's XXX prefix also covers XXXYYY numbers or not. If it doesn't, it would be a waste to try and contact it. Or maybe worse, you might be dialing a destination which /does/ work but is not displayed in the rates list and could be billed a lot more. At the moment, the way I am dealing with this is by trying the longest prefixes first. So in this case, the preference order would be: Prefix XXXYYY 0.20 (Provider A) Prefix XXX0.10 (Provider B) Prefix XXX0.15 (Provider C) However there is also a problem with this approach. Say a 'provider C' comes along with the following price list: Prefix 0.30 Prefix 0.30 Prefix 0.30 Now some '' numbers might be chosen first when potentially provider A's 'XXX' prefixes were cheaper! Any ideas on how to deal with this? Cheers, Jean-Michel. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] red alarm?
Red alarm on PRI is a physical layer problem, as in your telco had an outage or soemone unplugged the cable. -Original Message-From: Dov Bigio [mailto:[EMAIL PROTECTED]Sent: Tuesday, January 17, 2006 1:02 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] red alarm? Hi, What is the meaning of: Jan 17 18:05:21 WARNING[2388]: chan_zap.c:6315 handle_init_event: Detected alarm on channel 2: Red AlarmJan 17 18:05:21 WARNING[2388]: chan_zap.c:1432 zt_disable_ec: Unable to disable echo cancellation on channel 2Jan 17 18:05:21 WARNING[2388]: chan_zap.c:6315 handle_init_event: Detected alarm on channel 3: Red AlarmJan 17 18:05:21 WARNING[2388]: chan_zap.c:1432 zt_disable_ec: Unable to disable echo cancellation on channel 3 This happened once today with my 30 channels, but then everything came backto normal. Thank you Dov ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Simultaneous Inband and RFC2833 performed by Asterisk = Duplicate tones
Hi. DTMF recognition could be device problem. For example I have a setup which confirm that: However I stuck and don't know in what direction to continue. Have such schema: Asterisk - sipura3000 - Land Line. When call comes from cellphone spa3k captures DTMF and in debug from spa3k I see first digit doubled almost each time. Jan 19 12:44:26 192.168.162.30 FXO:Digit=1 Jan 19 12:44:26 192.168.162.30 AUD:Stop PSTN Tone Jan 19 12:44:26 192.168.162.30 FXO:Digit=1 Jan 19 12:44:26 192.168.162.30 AUD:Stop PSTN Tone Jan 19 12:44:26 192.168.162.30 FXO:Digit=0 Jan 19 12:44:26 192.168.162.30 AUD:Stop PSTN Tone Jan 19 12:44:27 192.168.162.30 FXO:Digit=1 Jan 19 12:44:27 192.168.162.30 AUD:Stop PSTN Tone Jan 19 12:44:28 192.168.162.30 FXO:Digit=0 Jan 19 12:44:28 192.168.162.30 AUD:Stop PSTN Tone On cell phone typed 1010 during Asterisk IVR. BTW, tried on different celphones from different GSM providers. Thanks in advance for any advices. Quoting Kevin P. Fleming [EMAIL PROTECTED]: Max Glucksmann wrote: - RFC2833 standard configured on both end-points. If this is the case, but the end sending DTMF _also_ puts it inband, then it is broken. This is an either/or setting; it's either inband or out of band, but not both. Asterisk sometimes listens for inband DTMF even when RFC2833 has been specified because the SDP tells us what the peer wants to _RECEIVE_, not necessarily what it will send. There are devices out there that will tell us they want RFC2833 but send only inband... many devices are just broken and/or inconsistent. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom FW
Anyone know how to obtain firmware and starter .cfg files for Polycom phones? Despite registering at the Polycom web site, I can't locate this stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Zapata.conf and Realtime
On Thu, Jan 19, 2006 at 10:37:49AM -0800, Steven Ringwald wrote: I asked this a few days ago, and haven't gotten an answer (or seen my message in the archive, yet). Since there were some email problems the other day, I will just pose the question again. I would like to know if there is a way to have a table, like zapata_conf in a DB, and have asterisk realtime pull the information out, like it does for voicemail, sip.conf, and iax.conf, etc. If anyone has done this and has a schema that I could use, I would be very happy. I believe that the nature of the channel is not dynamic enough. For starters, check if a simple reload reloads parameters properly. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dundi Examples
On 1/16/06, John Falk [EMAIL PROTECTED] wrote: Can someone show me how to set up DUNDi, I will be using it to connect 14 asterisk servers internally. I don't want to use it on the external world. If anyone has any examples of connecting 2 or 3 (if their is a difference) machines in a DUNDi co-operation that would be helpful. Check out my paper on DUNDi -- its a bit old now, but I think its probably still relevant. I probably really need to update it. Oh wait... I kind of did. Check out www.asteriskdocs.org for the O'Reilly book Asterisk: The Future of Telephony and look for the section on DUNDi. Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom FW
Polycom are analy retentive when it comes to this. You should be able to get the older versions on their web site though. Doug. -Original Message- From: Bill Michaelson [mailto:[EMAIL PROTECTED] Sent: Thursday, January 19, 2006 2:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Polycom FW Anyone know how to obtain firmware and starter .cfg files for Polycom phones? Despite registering at the Polycom web site, I can't locate this stuff. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dundi Examples
The O'Reilly TFOT book is full of errors. Two that pop into my head instantly are it's referring to regcontext being able to execute dialplan commands upon SIP registration and it's use of auth= in sip.conf in the DUNDi section. I wouldn't trust it. -Original Message- From: Leif Madsen [mailto:[EMAIL PROTECTED] Sent: Thursday, January 19, 2006 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Dundi Examples On 1/16/06, John Falk [EMAIL PROTECTED] wrote: Can someone show me how to set up DUNDi, I will be using it to connect 14 asterisk servers internally. I don't want to use it on the external world. If anyone has any examples of connecting 2 or 3 (if their is a difference) machines in a DUNDi co-operation that would be helpful. Check out my paper on DUNDi -- its a bit old now, but I think its probably still relevant. I probably really need to update it. Oh wait... I kind of did. Check out www.asteriskdocs.org for the O'Reilly book Asterisk: The Future of Telephony and look for the section on DUNDi. Leif Madsen. http://www.leifmadsen.com http://www.oreilly.com/catalog/asterisk ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New astGUIclient/VICIDIAL release: 1.1.9
Hello, We've released another update to our Asterisk GUI Client suite: 1.1.9 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL client-side web app auto-dialer. This package is free as in GPL. (the suite is not an asterisk configuration tool) This package is geared towards Asterisk installations with SIP,IAX or Zap phones and Zaptel, IAX or SIP trunks. For this revision, we have focused on several security enhancements as well as fixing bugs and several new features like a favorites panel for astguiclient that will show realtime extension state and a Scripts tab in vicidial that will show a script to read with customer data filled in. We have also tested the suite on Asterisk versions through 1.2.2 All client web-apps and administration pages are available in English, Spanish and Greek, with rough translations of French, German, Italian and Portuguese for the client web-apps only. Check out the project blog for screenshots and more information: http://astguiclient.blogspot.com Let me know what you think. Thanks, MATT--- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Zaptel Load as module
Tzafir, Any suggestion about the chan.zap issue that doesn't load wct1xxp as module? I don't know if you send the answer to my last e-mail, just our e-mail server was under attack, and we lost a lot of e-mails. Thanks Carlos Alperin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cannot compile chan_bluetooth on Asterisk 1.2.1
The short of it: I am unable to compile chan_bluetooth on Asterisk 1.2.1 on CentOS 4.2. I installed using the [EMAIL PROTECTED] 2.2 iso. Server is a plain Celeron 2.93GHz box. Asterisk source is in /usr/src/asterisk, newest chan_bluetooth source is in /usr/src/asterisk-test/bluetooth/chan_bluetooth (I have two older versions in other directories). Steps taken: Followed the instructions here to a T: http://www.crazygreek.co.uk/content/chan_bluetooth. Basically, edit /usr/src/asterisk/channels/Makefile adding chan_bluetooth.so to CHANNEL_LIBS, and at the very bottom adding include /usr/src/asterisk-test/bluetooth/chan_bluetooth/Makefile. First tried the version by David Woodhouse, exact command used to download was cvs -d :pserver:anoncvs at cvs.infradead.org:/home/cvs co chan_bluetooth. Also tried the version at http://www.crazygreek.co.uk/data/pages/chan_bluetooth/latest.tar.gz. Lastly, wanted to try a newer version of Theo's code on the SVN server, which was down. Google helped me find r40 at http://rock.inode.at/ROCK-2.1/c/chan_bluetooth-r40.tar.bz2. Using the newest version (by David Woodhouse) gives me this error: make[1]: Entering directory `/usr/src/asterisk/channels' gcc -shared -Xlinker -x -o chan_bluetooth.so /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.o -lbluetooth gcc: /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.o: No such file or directory make[1]: *** [chan_bluetooth.so] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 Using an older version will at least try to compile, but giving many errors. I found by using the Makefile from an older version with the newest, it also tries to compile but with errors as well. The only difference I see in the Makefile is using a .tmp directory in the chan_bluetooth directory in order to compile. Here's the end of the error using that Makefile (I'd post the entire error, but it fills up the buffer and I can't copy it all, let me know if you need more than I posted): /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c: In function `load_module': /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3210: error: `sdp_session_t' undeclared (first use in this function) /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3210: error: `sess' undeclared (first use in this function) /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3221: warning: implicit declaration of function `hci_open_dev' /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3227: warning: implicit declaration of function `hci_read_voice_setting' /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3228: warning: implicit declaration of function `htobs' /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3244: warning: implicit declaration of function `hci_devba' /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3248: error: `BDADDR_LOCAL' undeclared (first use in this function) /usr/src/asterisk-test/bluetooth/chan_bluetooth/chan_bluetooth.c:3248: error: `SDP_RETRY_IF_BUSY' undeclared (first use in this function) make[1]: *** [/usr/src/asterisk-test/bluetooth/chan_bluetooth/.tmp/chan_bluetooth.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/channels' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk]# I have also tried a few things, moving the include statement up in the Makefile, adding #define ASTERISK_VERSION_NUM 010201 to the top of chan_bluetooth.c (also used 010200, and 00). In /usr/src/asterisk/include/asterisk/version.h, it kept being set to 00, I had to edit the Makefile in /usr/src/asterisk to force it to 010201 (after trying it with the 00 value first, of course). When compiling Asterisk, I will do a make clean then make. When making minor changes I would just do a make clean in /usr/src/asterisk/channels then a make in /usr/src/asterisk. The two errors above were after doing a complete make clean in /usr/src/asterisk, then a make. Hopefully I gave enough information, if I missed anything let me know. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI
I am having problem with T1 configuration. Following r the config.. Zaptel.conf # Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit # Zaptel Configuration File # # This file is parsed by the Zaptel Configurator, ztcfg # # It must be in the module loading order span = 1,1,0,esf,b8zs # fxsks=1-24 bchan = 1-23 dchan = 24 # Span 1: TE2/0/1 T2XXP (PCI) Card 0 Span 1 # ??: 1 TE2/0/1/1 FXSKS # ??: 2 TE2/0/1/2 # ??: 3 TE2/0/1/3 # ??: 4 TE2/0/1/4 # ??: 5 TE2/0/1/5 # ??: 6 TE2/0/1/6 # ??: 7 TE2/0/1/7 # ??: 8 TE2/0/1/8 # ??: 9 TE2/0/1/9 # ??: 10 TE2/0/1/10 # ??: 11 TE2/0/1/11 # ??: 12 TE2/0/1/12 # ??: 13 TE2/0/1/13 # ??: 14 TE2/0/1/14 # ??: 15 TE2/0/1/15 # ??: 16 TE2/0/1/16 # ??: 17 TE2/0/1/17 # ??: 18 TE2/0/1/18 # ??: 19 TE2/0/1/19 # ??: 20 TE2/0/1/20 # ??: 21 TE2/0/1/21 # ??: 22 TE2/0/1/22 # ??: 23 TE2/0/1/23 # ??: 24 TE2/0/1/24 loadzone = us defaultzone = us Zapata.conf signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national group=2 context=from-pstn ; Points to the default context of your extensions.conf channel = 1-23 I am using Asterisk@ home Asterisk version is 1.2.0 I created a trunk ZAP/g2 it say all circuits r busy.. I have Digium Wildcard TE205P If any one have working config please let me know. Thanks Ali ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400P zttest not working
Hi, I have a TDM400P running with only one FXO port running on a VIA EPIA MS1 (1Ghz Via Eden). When I run zttest from Zaptel 1.2.1 it hang and when I interrupt it with Ctrl-C that is the result: ¿anyone have some idea about why isn't working? Some additional info: # /usr/src/zaptel/zttest -v Opened pseudo zap interface, measuring accuracy... --- Results after 0 passes --- Best: 0.00 -- Worst: 100.00 -- Average: 100.00 Some info: # lsmod Module Size Used by wctdm 35264 1 zaptel188804 7 wctdm binfmt_misc12040 1 # cat /proc/interrupts CPU0 0: 705996 XT-PIC timer 1: 8 XT-PIC i8042 2: 0 XT-PIC cascade 5: 0 XT-PIC uhci_hcd:usb4, uhci_hcd:usb5 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi 10:100 XT-PIC ehci_hcd:usb1 11: 2834 XT-PIC uhci_hcd:usb2, uhci_hcd:usb3, eth0, wctdm 12:114 XT-PIC i8042 14: 6880 XT-PIC ide0 NMI: 0 LOC: 0 ERR: 0 MIS: 0 # info from dmesg Zapata Telephony Interface Registered on major 196 ACPI: PCI Interrupt :00:13.0[A] - Link [LNKA] - GSI 11 (level, low) - IRQ 11 Freshmaker version: 73 Freshmaker passed register test Module 0: Not installed Module 1: Not installed Module 2: Not installed Module 3: Installed -- AUTO FXO (FCC mode) Found a Wildcard TDM: Wildcard TDM400P REV I (1 modules) Registered tone zone 0 (United States / North America) Registered tone zone 0 (United States / North America) # lspci -vvv 00:13.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface Subsystem: Unknown device b119:0003 Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- Status: Cap+ 66Mhz- UDF- FastB2B- ParErr- DEVSEL=medium TAbort- TAbort- MAbort- SERR- PERR- Latency: 32 (250ns min, 32000ns max) Interrupt: pin A routed to IRQ 11 Region 0: I/O ports at ec00 [size=256] Region 1: Memory at e6402000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 Flags: PMEClk- DSI+ D1- D2+ AuxCurrent=55mA PME(D0+,D1-,D2+,D3hot+,D3cold+) Status: D0 PME-Enable- DSel=0 DScale=0 PME- Best Regards. Antonio. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
