[Asterisk-Users] Re: VoIP gateway (2 FXO, 2 FXS)
Does anyone know a good (and stable) voip gateway product with 4 ports (2 fxo and 2 fxs), with the following requirements: * being able to connect analog phones to the FXS ports, and communicate over SIP with an REGISTRAR/PROXY server (SER in our case). * being able to connect the FXO port to local office PSTN network, and dial to that office pstn number and getting an internal dialtone, or forward ability to the SIP gateway. So employees can call to the local pstn number, and enter an international phone number which is routed over the SIP gateway (SER). The following are results with 2 products I tried, without any success. I used http://www.voip-info.org/wiki-VOIP+Gateways to order the following, * Ovislink VoIP-422 * Welltech 3702A I've tested them, and came across the following problems, * Ovislink product - Problem #1 adding sip accounts worked like a charm, they register perfectly with our sip gateway (SIP Express Router). But when we make a call from an analog phone (connected to a FXS port), the SIP packets (INVITE, etc) do NOT include the authentication details (SER sends 'Proxy Authentication Required'), the DIGEST username is just blank and From is elite@ (no idea where that came from, probably hardcoded). I've tried linking a callerid/name with that FXS port, without a difference. The same problem arises when we call the office pstn number (pstn port connected to FXO port of ovislink box). We get an internal dialtone (of the ovislink), and when the enter a number, it also doesn't send the auth details in the SIP INVITE packet it sends to SER. - Problem #2 As a 'quickfix' I configured SER to NOT look at the auth details, and just process the call anyway. When the call is answered, and SER sends the SIP/2.0 200 OK, the Ovislink does NOT send the ACK (but I can see the incoming OK packet in the ovislink console). Quite buggy indeed.. or i'm misconfiguring the device, but i'm sure I got everything right. Anyone else with some experiences ? * Welltech product Dialplan issues, I created the necessary routes to route everything over IP.. but it still sends incoming PSTN calls (FXO port, LINE1), to the analog phone connected on the FXS port (TEL1). Calls made from the analog phone are routed over the LINE1/FXO port. I specifically changed all the reference to FXO to IP, and STILL it's sending the calls over the FXO port. Anyone got some luck with either of these products, or has another product that fullfill our needs ? Thanks in advance. --__--__-- Take a look at the Planet VIP-450 http://www.planet.com.tw/product/product_dm.php?product_id=195menu_id=3 . We use the VIP-400 (H.323 version), which has lots of flexibility in the dial plan, IDs, etc. You can download the complete manual from the Planet site. Pros: Inexpensive, good voice quality, doesn't crash, excellent hardware reliability, good support for configuration problems. Cons: Many minor bugs and shortcomings (more subtle than your Ovislink and Welltech problems), no support at all for getting these fixed, unless a big customer of Planet happens to experience the same trouble! Good luck, Stewart ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk and Linejacks
Hi Greg! Sorry, but I can't help you. I've never tried asterisk with the linejack. regards, klaus greg wrote: I found a message from you to the asterisk users mailing list from 2001. I was wondering if you got (or still have) an asterisk system working with the linejack? If so, would you be willing to assist me with mine? I seem to have things working, and * says that caller ID is coming in, but I can't get * to actually answer the call. Thanks, Greg -- NetIO.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] fax still fails, ideas sought! Re: rxfax/spandsp fails to decode
Stephen J. Wilcox wrote: Okay having taken in some suggestions and googled this topic to death I'm still stuck - anyone got any ideas? To recap, the faxes are coming in via a digium E1 card but failing to train properly or if they manage it sending a garbled and very truncated fax. A number of folks have suggested clock sync issues.. my zaptel.conf is set to use the PRI as primary clock, i have no evidence of issues altho dont know how to check (other than the call quality is fine, no clicks, no pri down/ups). If you wanevidence of problems, look a the spandsp log. You have the evidence, but you are ignoring it. What can i try? I really should make spandsp smart enough to actually detect the timing jumps and put out a nice big FIX YOUR TIMING, AND NO COMPLAINTS UNTIL YOU DO * message. :-) Steve On Mon, 12 Jul 2004, Stephen J. Wilcox wrote: Hi, I just sent this to Steve Underwood, but then found a bunch of posts on the mailing list about similar issues.. does anyone have the fix? I'm running asterisk CVS-HEAD-06/28/04-18:13:13, spandsp 0.0.1k, libtif 3.5.7 one thing i just noticed is that calls come in with format '72' which is G711A-law or LinearPCM.. it uses PCM for the call, i assume this is ok the results of RxFAX vary, it sometimes saves the file in which case i get errors: Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 0 (got 2383, expected 1728). Fax3Decode2D: (FakeInput): Bad code word at scanline 1 (x 137). and the resulting tif looks to be only a few rows long or more commonly it just fails entirely.. i paste the output below so you can see. is there anything obvious i'm doign wrong here? TIA! Steve. -- Executing RxFAX(Zap/1-1, /var/spool/asterisk/faxes/20040712-183339.tif) in new stack Changed from phase 0 to 1 Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 31 37 31 31 36 35 34 35 34 38 30 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: DCS: 83 00 86 90 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 5ms Get at 9600 Changed from phase 3 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1699.90 (64) Training error 56.874846 Training succeeded (constellation mismatch 44.212022) Fast carrier trained Fast carrier down Trainability test failed - longest run of zeros was 14 FTT: 44 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.33 (64) Training error 51.989152 Training succeeded (constellation mismatch 37.988826) Fast carrier trained Fast carrier down Trainability test failed - longest run of zeros was 15 FTT: 44 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1700.32 (64) Training error 60.898646 Training succeeded (constellation mismatch 46.138793) Fast carrier trained Fast carrier down Trainability test failed - longest run of zeros was 17 FTT: 44 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1795.61 (4) Fast carrier down Fast carrier up Coarse carrier frequency 1789.60 (4) Fast carrier down -- Channel 0/1, span 1 got hangup -- Hungup 'Zap/1-1' Regards, Steve ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G.729 - ZAP ?
Hi, I am trying to replace my Cisco 5300 gateway with my new Zap TE405P card. Incoming calls and outgoing calls between my cisco and my SIP phone works fine on G.729. Recording messages in the asterisk voice-mailbox also works fine from both my SIP phone as well as PSTN - Cisco - Asterisk. I have licensed the digium G.729A codec. When I connect my ISDN PRI to my Zap card and I call in, Asterisk does not like the G.729 anymore and will not send the call to my phone, claiming the phone does not support the codec asterisk wants, as I forced it to G.729. For some reason incoming and outgoing calls will ALWAYS use G.711a. Is there anywhere else I need to look to tell ZAP to use G.729 preferrably ? The translator is loaded... [codec_gsm.so] = (GSM/PCM16 (signed linear) Codec Translator) == Registered translator 'gsmtolin' from format GSM to SLINR, cost 1 == Registered translator 'lintogsm' from format SLINR to GSM, cost 4 [codec_g729a.so] = (Annex A/B (floating point) G.729/PCM16 Codec Translator) == G.729 Host-ID: 1b:a1:18:82:47:6f:a8:f7:33:de:7d:77:e8:1d:60:15:53:ec:49:aa == Found license 'G729-700241AB' providing 5 channels == Found total of 5 G.729 licenses == Registered translator 'g729tolin' from format G729A to SLINR, cost 3 == Registered translator 'lintog729' from format SLINR to G729A, cost 14 [codec_lpc10.so] = (LPC10 2.4kbps (signed linear) Voice Coder) == Registered translator 'lpc10tolin' from format LPC10 to SLINR, cost 3 == Registered translator 'lintolpc10' from format SLINR to LPC10, cost 9 [codec_adpcm.so] = (Adaptive Differential PCM Coder/Decoder) == Registered translator 'adpcmtolin' from format ADPCM to SLINR, cost 1 == Registered translator 'lintoadpcm' from format SLINR to ADPCM, cost 1 [codec_ulaw.so] = (Mu-law Coder/Decoder) == Registered translator 'ulawtolin' from format ULAW to SLINR, cost 1 == Registered translator 'lintoulaw' from format SLINR to ULAW, cost 1 [codec_alaw.so] = (A-law Coder/Decoder) == Registered translator 'alawtolin' from format ALAW to SLINR, cost 1 == Registered translator 'lintoalaw' from format SLINR to ALAW, cost 1 [codec_a_mu.so] = (A-law and Mulaw direct Coder/Decoder) == Registered translator 'alawtoulaw' from format ALAW to ULAW, cost 1 == Registered translator 'ulawtoalaw' from format ULAW to ALAW, cost 1 [format_g723.so] = (G.723.1 Simple Timestamp File Format) == Registered file format g723sf, extension(s) g723 [format_wav.so] = (Microsoft WAV format (8000hz Signed Linear)) == Registered file format wav, extension(s) wav *CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC G723 - - - - - - - - - - - GSM - - 2 2 4 2 11015 -23 ULAW - 5 - 1 4 2 11015 -23 ALAW - 5 1 - 4 2 11015 -23 G726 - 7 4 4 - 4 31217 -25 ADPCM - 5 2 2 4 - 11015 -23 SLINR - 4 1 1 3 1 - 914 -22 LPC10 - 7 4 4 6 4 3 -17 -25 G729A - 7 4 4 6 4 312 - -25 SPEEX - - - - - - - - - - - ILBC - 8 5 5 7 5 41318 - - !!! Help Anybody??? Digium ??? I need this G.729 to work as G.711 is too much a bandwidth hog and my Cisco 5300 is dying... Walter Klomp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura 3000 PSTN disconnect in the UK
Anyone else got a Sipura 3000 in the UK? Apart from CID not working it also seems to not notice any of the line state changes on the PSTN when the remote party terminates the call. It only recognises the offhook signal which gets sent much later. Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP connections do not hang up
Hi, Well, the Problem is not the ZAP Channel but the SIP Channel, because it occurs no matter what channel I use the phone outside. Maybe this graph is more descriptive: 1. ZAP or SIP == 2. Asterisk == 3. SIP (thru internet, provider sipgate) == 4. PSTN The connections on 1. hang up correctly, as seen in the log, but the SIP connection of 3. does NOT hangup. Regards, Florian PS: Believe me, I'm searching for over one week in the whole internet for a solution, but did not find it. - Original Message - From: Jean-Yves Avenard [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 30, 2004 11:46 PM Subject: Re: [Asterisk-Users] SIP connections do not hang up -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 If you just bothered to search this list in the past 12 hours, you would have found a solution around that: to summarize: Add in zapata.conf: busydetect=yes busycount=6 The maximum it will take for asterisk to see the person hanged-up is after 6 busy dial-tones. On 31/07/2004, at 6:58 AM, Florian Rau wrote: I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone using Zap Channel) using sipgate to a number in public network. When I'm hanging up before the other person picked up the phone, the line is not closed correctly. The phone keeps on ringing until timeout (of Sipgate I assume) and it even costs my money, if the other person picks up the ringing phone, even if I already hung up. - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCsHLXeDVKqIr3GURArjyAJ9p97F/wWIiIesaYo85QfHut8zbzQCgj2l2 uuKZxyJoaSmpI9V9I+ojnJc= =Y8jQ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaphfc hardware sound trouble
On Fri, 30 Jul 2004 [EMAIL PROTECTED] wrote: ###zapata.conf context=default context=alex pridialplan=unknown echocancel=yes echocancel=yes echocancelwhenbridged=yes immediate=yes Why do you define echocancel and context twice? a) when i try to make an inbound call to msn I get the following message on the cli prompt -- Going to extension s|1 because of immediate=yes -- Extension 's' in context 'alex' from '17109904' does not exist. Rejecting call on channel 2, span 1 You should not use immediate=yes for a TE interface since that instructs asterisk to go to s extension (which is useful for NT interfaces but not for TE ones). I have set pridialplan=local and have the msn in exten = msn,... set including the area code. To only match calls from my cellular phone I use exten = msn/cellphonenumber,... That works fine for me. If it still does not work, turn on some debugging and try to catch what is happening when a call comes. b) the combination of my configuration with zaphfc and the acer pci isdn card seems to cause some other trouble Kernel: 2.4.21-0.13mdk Jul 29 23:46:49 faar kernel: sync lost, pci performance too low!!!. I had that problem running RedHat Linux 9.0 with kernel 2.4.20-31.9. It was probably occured by a buggy driver for the IDE controller integrated on my mainboard, that made the hard disk access interrupts taking too much time from the pci bus. The problem disappeared after upgrading kernel to 2.4.26. d) as long as the line works, I have clearly audible clicks/cracks in the line (zaphfc) that didn't occur using capi and the avm fritz! pci 2.0 - I don't have any sound problems on iax via voiptel.org or internally using sip This problem I still have. This is what I have found out: First turn with echocancel=no in zapata.conf. The echo cancelling does not work any good at all if some audio data is missing. That reintroduces echo while talking to analog phones, of course, but could help finding the source of your problem. Then check that your zaptel timing is all right. Try running zttest (in the zaptel directory) and watch the output. I still have problems with this one running zaphfc-0.1.0-RC2k on my new Intel Pentium 4 2,4 GHz system but timing is excellent on my old AMD K6-233 MHz system... Regards from Sweden, Tobias Jönsson ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium FXO Interfaces don't support groundstart???
Glare cannot be prevented on two way trunks (it is physically impossible because the two ends are separated in distance and therefore separated in time and any independent decision to use it at one end is never seen instantly at the other end). Ground start does not decrease glare at all (it actually increases it) and use of ground start to eliminate glare is a common myth. This is because use of ground start (which uses only one side of the pair to earth ground to start a request for service) increases the time to mark a central office line busy when it is seized from the Customer Premise Equipment (CPE), owing to its clunky signaling (150ms earth ground on the ring of the line) and the fact that it uses only one half of the current to start the line as loop start. Since it increases the time to signal the distant end, it increases glare. Its only benefit is to the central office because it stops a second seizure to the central office when a call disconnects from the central office end first, which would otherwise find a request (loop) as soon as the disconnect was effected. This is why ground start was introduced by the Bell System (when they owned both the PBX and the CO) since it would reduce the attempt load on the central office from large business users by 25% or more saving a lot of central office gear for a relatively small expenditure on the PBX end. Ground start has some ugly drawbacks, since it reduces signaling range, requires the normally isolated floating pair to be referenced to earth ground (which exposes the circuit to longitudinal spikes, noise and lightning) and requires the circuit to be muted during the imbalanced condition that occurs when the ring conductor is momentarily grounded to draw dial tone. Digium is right to leave it out. Most other informed, modern manufacturers do likewise. Ground start signaling referred to in T1 (which is an absurd label since there is no ground placed on a T1) is really after the Grey Code (only one signaling bit transitions at a time) and has nothing to do with glare or ground start signaling and is just a carry over label. Glare can be reduced by changing the hunt order from either end and to employ faster signaling. The former method decreases the likelihood that both ends will compete for the circuit at the same time and the latter reduces the window that a commitment has been made at one end and is still not known by the other end. Typically, the CO is set to hunt ascending and the CPE descending and this is still employed even in ISDN circuits. This is a terminal hunt and NOT a round robin hunting sequence. If you want to absolutely eliminate glare, use one way (incoming/outgoing only) circuits. I believe asterisk has a feature to set the hunt order preference. The disconnect problems you experienced with your Agilent PBX may be more likely related to the guard interval that a circuit is left alone at your end after it is used. Though ground start will appear to fix it, there are some issues of CO message rate three way calling that have caused grief (the CO interprets the next call as a flash for a three way call and holds the circuit rather than disconnecting it). This phenomena may have been misdiagnosed as glare, since the message unit 3-way calling was imposed as a default feature in certain jurisdictions. Increasing the guard interval to 2 or 3 seconds will suffice, or specify to the carrier that the 3-way calling is to be denied for your lines. Hope this helps. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 08, 2004 4:55 AM Subject: [Asterisk-Users] Digium FXO Interfaces don't support groundstart??? Hi All, I was surprised to be told by a Digium support person today that Digium's FXO interfaces (X100P, TDM400P FXO modules) don't support groundstart signalling. This surprises me because as far as I know in a typical PBX configuration with analog trunk lines, groundstart signalling is the only way to prevent Glare. I just purchased two TDM400P's for a system I'm building to replace our office PBX (Altigen). Since there are no statements anywhere on Digium's website about lack of groundstart support (Actually, to the contrary they boast about all the signalling support in their sales slick), I now need to decide if I want to return the products and switch to a T1 / channel bank configuration. I remember when we setup our current Altigen PBX, we had problems with glare and disconnect detection and so I went through the process of figuring out what was going on and learning about groundstart. After we switched to groundstart everything worked great. In a high use system, it's highly likely that a trunk will experience glare, which is annoying for incoming callers and system users. I'm just a bit baffled as to why Digium wouldn't support groundstart on cards designed to be PBX trunk lines. Someone please tell me I'm missing something.
RE: [Asterisk-Users] Sipura 3000 PSTN disconnect in the UK
Chris Stenton [EMAIL PROTECTED] wrote: Anyone else got a Sipura 3000 in the UK? Apart from CID not working it also seems to not notice any of the line state changes on the PSTN when the remote party terminates the call. It only recognises the offhook signal which gets sent much later. The SPA-3000 is not rated for UK use yet. It will apparently get a firmware update in September 2004 to cover the UK. I wouldn't expect any of it to work until then, so I'm waiting for the firmware before placing an order. Apparently it works and works well in North America, so I have every confidence that it'll work in the UK - once the firmware update is released. I have never had any trouble out of my SPA-2000's interaction with UK-orientated phones. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New to IP-PBX
I have been seeing reccomendations for using asterisk as a soft-pbx with the reccomendation being to use regular analog phones via FXS rather than SIP. Is this still a big issue? Or is this a left-over from previous bad experiences? I have been doing demos with SIP phones, and some IAXYs to whet their apetites, and people are really biting at the feature set I can provide, and I have run into no problems yet, but I would love to know at what threshold of SIP phones does the system start to have problems. One of the primary drivers for using FXS rather than SIP is that traditional pbx sales people sell their products based primarily on least cost. Historically, they use to sell features, least cost call routing, toll bypass, and other such things as they use to be popular sales attractions. Given what has happened to long distance costs, the least cost call routing kinds of things are not much of a concern on the part of the buyer any more. As a result, the business oriented buyer (not technical people) are far more oriented towards initial cost and features because that's one of the things they can understand. If you search the * list you'll find all kinds of postings relative to I can configure a cheaper asterisk then you can, and if initial cost is a serious factor for the business buyer, then FXS is likely the approach. However, with that said, how you communicate with the business buyer will make all the difference in the world. If you structure you sales pitch around cost, you're heading for FXS's. If you change that pitch, selling solid well-defined sip phones is a piece of cake. So, if you understand your customer's actual requirements and the stability of their network infrastructure, selling sip into an account should be easy in most cases. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RAID affecting X100P performance...
Andrew Kohlsmith wrote: On Friday 30 July 2004 19:51, Mike Benoit wrote: Tuning these [PCI latencies] should allow you to give your TDM cards long burst lengths, and make your IDE devices very premptable... I would have figured you want very short burst lengths to prevent any one device from hogging the PCI bus and delaying your VOIP data. IIRC the VOIP traffic is very short anyway (1000 interrupts a second, 8 bit PCM data or 64 bytes of actual VOIP data (sampled at 8000Hz, that's 8 8 bit samples every interrupt) each way... I think. Yes, but by lowering the available time that another device can tie up the PCI bus, you increase the chances that your TDM cards get their interrupts handled very quickly. And since you know that your TDM card interupts are very short (but very frequent), you want a way to prioritize them. So shorten the burst lengths everything else is allowed, and allow your TDM card to tie up the bus whenever it wants. Working with the 2.6 kernel and some of the new stuff by Ingo Molnar, you can actually get interrupt priorities, and the possibility of allowing specific RT threads preempting lower priority interrupts. Basically, to give Asterisk ideal conditions to to echo cancel, you want to make sure that: 1) PCI bus is given to TDM card as soon as possible when it wants to raise an interrupt. 2) Make sure you OS can handle the interrupt soon enough. Use #1 to tune PCI latencies (yes, an IDE or Network event can tie up your PCI bus for many usec, delaying your TDM interrupts). For #2, you have to live with linux-2.4 being good enough, but if your willing to live on the edge, do check out some of the new recent work that's going on around 2.6, mainly driven by the pro-audio users. On a 600Mhz Eden, users are reporting 42us latencies on RT user-space threads. Basically working out to the almost theoretical minimum hardware capabilities. Again - none of these will guarantee asterisk echo-cancel/dropouts will be fixed, but the all work towards giving asterisk the best conditions for it to do what it is trying to do. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Hello, Eric Bart wrote: Flash don't work for sip This affirmation is too broad, it might not work with X-lite, but flash will work with may sip devices, including cheap ones (grandstreams, sipuras, etc). From: Jozeph Brasil [EMAIL PROTECTED] I have one X100P installed with two SIP extensions using X-Lite, I just would like to transfer the call to another SIP extension; Just a Flash+Extension+Hangup CALL... -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: RAID affecting X100P performance...
Andrew Kohlsmith wrote: On Friday 30 July 2004 19:51, Mike Benoit wrote: Tuning these [PCI latencies] should allow you to give your TDM cards long burst lengths, and make your IDE devices very premptable... I would have figured you want very short burst lengths to prevent any one device from hogging the PCI bus and delaying your VOIP data. IIRC the VOIP traffic is very short anyway (1000 interrupts a second, 8 bit PCM data or 64 bytes of actual VOIP data (sampled at 8000Hz, that's 8 8 bit samples every interrupt) each way... I think. Yes, but by lowering the available time that another device can tie up the PCI bus, you increase the chances that your TDM cards get their interrupts handled very quickly. And since you know that your TDM card interupts are very short (but very frequent), you want a way to prioritize them. So shorten the burst lengths everything else is allowed, and allow your TDM card to tie up the bus whenever it wants. Working with the 2.6 kernel and some of the new stuff by Ingo Molnar, you can actually get interrupt priorities, and the possibility of allowing specific RT threads preempting lower priority interrupts. Basically, to give Asterisk ideal conditions to to echo cancel, you want to make sure that: 1) PCI bus is given to TDM card as soon as possible when it wants to raise an interrupt. 2) Make sure you OS can handle the interrupt soon enough. Use #1 to tune PCI latencies (yes, an IDE or Network event can tie up your PCI bus for many usec, delaying your TDM interrupts). For #2, you have to live with linux-2.4 being good enough, but if your willing to live on the edge, do check out some of the new recent work that's going on around 2.6, mainly driven by the pro-audio users. On a 600Mhz Eden, users are reporting 42us latencies on RT user-space threads. Basically working out to the almost theoretical minimum hardware capabilities. Again - none of these will guarantee asterisk echo-cancel/dropouts will be fixed, but the all work towards giving asterisk the best conditions for it to do what it is trying to do. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Adding SIP Based Termination
Hi, I've been reading some manuals and have added a bunch of SIP Accounts for outbound calls into my Asterisk Setup. The local extensions are working perfectly. The problem I am facing at the moment is that, if I try and make outbound calls using a SIP Account, it rings thrice and then there is a disconnection tone. [sip.conf] register = userid:[EMAIL PROTECTED] [prov] type=friend secret=passwd username=userid host=ip disallow=all allow=gsm allow=ulaw allow=alaw allow=G726 [/sip.conf] [extensions.conf] [sip] exten = 101,1,Dial(SIP/netstation,20,tr) exten = 102,1,Dial(SIP/sahil-akl,20,tr) exten = _34.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) [/extensions.conf] A paste from my CDR file: ,2123,3400911126476699,sip,NetStation 2123,SIP/netstation-bed4,SIP/prov-ecf9,Dial,SIP/[EMAIL PROTECTED]|20|tr,2004-08-01 01:00:54,,2004-08-01 01:01:04,10,0,NO ANSWER,DOCUMENTATION Any help would be greatly appreciated. Cheers, Sahil ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail Not releasing
I have the same issue with IAX2. I get messages anywhere from 5 min to 45 min of silence. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 30, 2004 9:59 PM Subject: [Asterisk-Users] VoiceMail Not releasing About twice a week we have a caller that comes in and hangs up on voicemail. We have 2 x100ps each with their own irq. When the caller hangs up asterisk does not release the line. The line rings busy, sometimes I can do a soft hangup Zap/1 and release the line sometimes I have stop asterisk and remove and re-insert the modules. I am running RC1 on debian. Is this a bug, or something I have setup wrong. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RES: [Asterisk-Users] Softphone - Freeware?!
Hmmm, Flash work for IAX? -Mensagem original- De: Eric Bart [mailto:[EMAIL PROTECTED] Enviada em: sábado, 31 de julho de 2004 02:26 Para: [EMAIL PROTECTED] Assunto: Re: [Asterisk-Users] Softphone - Freeware?! Flash don't work for sip - Original Message - From: Jozeph Brasil [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 30, 2004 4:17 PM Subject: RES: [Asterisk-Users] Softphone - Freeware?! I have one X100P installed with two SIP extensions using X-Lite, I just would like to transfer the call to another SIP extension; Just a Flash+Extension+Hangup CALL... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] one extention, multiple phones
Is it possible to get a few 7960's and asterisk to allow all of the 7960 phones to use one extentsion and can only be used by one person at a time, have it indicate on the other 7960's when one of the others has the line engaged. Basicly so like I can setup a rule when an incoming call comes from IAX to divert to this extension, it will ring the extension (thus all phones), and allow me to place a call on hold on one phone and pick it up on another and the original phone would acknowledge that the call has been picked up and disengage. Can I do this without call parking? Basicly the same model as having a bunch of phones on a pstn line with each phone having a hold button. The goal here is to allow me to pick up a call on any of the 7960's anywhere in my house and be able to move from room to room as needed by placing the call on hold and picking it up on one of the other phones in the house. /\ . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . \ / - ASCII Ribbon Campaign . Sean McKay - [EMAIL PROTECTED] X - NO HTML/RTF in e-mail . Team Lead, bahamut web team / \ - NO Word docs in e-mail . ircd-qa team member ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one extention, multiple phones
Sean McKay wrote (on Jul 31): The goal here is to allow me to pick up a call on any of the 7960's anywhere in my house and be able to move from room to room as needed by placing the call on hold and picking it up on one of the other phones in the house. If this is the intention, then you probably want to use call parking. AFAIK, there's no mechanism to inform a phone that another phone currentl has the line, though you could probably do some clever XML pages for the 7960 coupled up to the manager interface to do close to the same. Chris. -- == [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RAID affecting X100P performance...
On Saturday 31 July 2004 00:20, Steven Critchfield wrote: Actually, it isn't VoIP data yet, VoIP is Voice over Internet Protocol. The 1000hz interupt is still just digitizing the audio off the PSTN link. When it comes time to read/write VoIP data, it is likely 20ms of audio, plus headers and IP encapsulation. If you are lucky, your LAN/WAN card supports DMA and is reading the packet on it's own out of memory. Duh, yes, you're right... How were my numbers for the TDM data coming off the Zap hardware? 1000Hz sampling off the PSTN (You're grabbing 8 samples though aren't you?) -- 8 bit PCM data * number of channels * number of samples, right? So 64 bytes / interrupt per channel in use, one-way, 128 bytes for both directions? And are the TE400/405P cards the only Digium cards that handle DMA? -A. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Thanks for the correction I didn't know that SIP would do. As I understood the R key will send the flash signal. However does it really act as a transfer ? For the zap transfer, as said in : http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20zap%20transfer when the transferer hangs up each parties are disconnected. Is it what you are experiencing ? With my app when the transferer hangs up the others parties stay connected ... I'm wondering whether it's useful or not :) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: RAID affecting X100P performance...
On Friday 30 July 2004 20:26, Aidan Van Dyk wrote: Yes, but by lowering the available time that another device can tie up the PCI bus, you increase the chances that your TDM cards get their interrupts handled very quickly. And since you know that your TDM card interupts are very short (but very frequent), you want a way to prioritize them. So shorten the burst lengths everything else is allowed, and allow your TDM card to tie up the bus whenever it wants. Yup I read the article and I understand what he's doing now. Nice little bit of information to have onhand. Regards, Andrew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one extention, multiple phones
I'm guessing then CCM could handle this, and I could simply link asterisk to the CCM as a gateway and acchieve the reult I'm looking for On Sat, 31 Jul 2004, Chris Luke wrote: Sean McKay wrote (on Jul 31): The goal here is to allow me to pick up a call on any of the 7960's anywhere in my house and be able to move from room to room as needed by placing the call on hold and picking it up on one of the other phones in the house. If this is the intention, then you probably want to use call parking. AFAIK, there's no mechanism to inform a phone that another phone currentl has the line, though you could probably do some clever XML pages for the 7960 coupled up to the manager interface to do close to the same. Chris. -- == [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one extention, multiple phones
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Saturday 31 July 2004 09:41 am, Chris Luke wrote: Sean McKay wrote (on Jul 31): The goal here is to allow me to pick up a call on any of the 7960's anywhere in my house and be able to move from room to room as needed by placing the call on hold and picking it up on one of the other phones in the house. If this is the intention, then you probably want to use call parking. AFAIK, there's no mechanism to inform a phone that another phone currentl has the line, though you could probably do some clever XML pages for the 7960 coupled up to the manager interface to do close to the same. Chris. Sure there is. You can only have one extension per call by default. Wait a minute! Good ol' Transfer works fine. You just transfer it to the extension you want. No reason you cannot have an extension for each room which all ring. The call goes to the extension which answers first. You can have conference calls too. Now every room can join in. You can have conference rooms that only allow certain rooms even. - -- Steve They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (GNU/Linux) iD8DBQFBC7DRljK16xgETzkRAiogAJ91YDCw5AJ6J7WYBRyPv73AX6kWywCfWGc6 zfveLFeNmNg1/VbTS/k0s1M= =JEnC -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Which version of MySQL works with cdr_addon_mysql?
I'm having problems compiling cdr_addon_mysql with MySQL 3.23.58 I get the following errors: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:107: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:107: (Each undeclared identifier is reported only once cdr_addon_mysql.c:107: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:417: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 I have loaded: MySQL-3.23.58-1.i386.rpm MySQL-client-3.23.58-1.i386.rpm MySQL-devel-3.23.58-1.i386.rpm MySQL-shared-3.23.58-1.i386.rpm Does cdr_addon_mysql work with this version of MySQL or do I need to install version 4? Thanks Malcolm Bader ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adding SIP Based Termination
On Sun, 1 Aug 2004 [EMAIL PROTECTED] wrote: exten = _34.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) The r at the end of this line tells asterisk to generate a ringing sound for you to hear. In other words, the ringing you're hearing isn't coming from the far end SIP device. Taking the r out will probably help you get a little closer to the solution.. another thing you can do is turn sip debug on while you try to place the call and see what happens. Watch for responses from the far end sip server, etc. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Hi Eric, Eric Bart wrote: Thanks for the correction I didn't know that SIP would do. As I understood the R key will send the flash signal. However does it really act as a transfer ? For the zap transfer, as said in : http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20zap%20transfer when the transferer hangs up each parties are disconnected. This is for ZAP channels, the original question was: have one X100P installed with two SIP extensions using X-Lite, I just would like to transfer the call to another SIP extension; Just a Flash+Extension+Hangup CALL He wants to transfer a call from one SIP extension to another... All sip devices that I know off (I'm not talking about soft phones, I do not use them, so I can say anything about them) have a way to transfer a call to another sip device by themselves (without the help of asterisk). Grandstream phones have a 'transfer' key. If you press that key and then dial the extension you like to transfer and then hangup (just like the original poster asked), it will just work. Its a blind transfer, and you better dial the desired extension right, because if you made a mistake, the call will be lost in limbo as some other users are reporting (a grandstream feature/bug) Sipuras can do this to: just by flashing the analog phone. They are capable of consultative transfers also (they let you talk to the destination party before transferring the call) I tried them both, transferring an inbound call from a ZAP FXO line to a sip extension and it works great, no hangups, no problems. With sipuras I can do consultative transfers also, I use them all the time. You can also achieve the same results by using asterisk transfer feature (T or t options in the dial command). In this case the transfer will be allways blind. It works perfect with ZAP FXO and SIP FXS for me. If you want consultative transfers with asterisk, you can sort of have it by using parking: you can dial '#' to transfer, then send the call to the parked calls extension, and the parked extension will be read back to you. Then you hangup and talk to the extension you want the call to be transferred: 'you have Bob on the extension 702'. The other party can now dial that extension and talk to Bob. Its not a consultative transfer as regular phone users are accustomed, but it works. And if the parked call times out, it will ring back the extension that parked it on the first place. And I'm sure it works also with other technologies as IAX2 or CAPI. Is it what you are experiencing ? With my app when the transferer hangs up the others parties stay connected ... I'm wondering whether it's useful or not :) Maybe your application cann fill the gap for sip devices that are not capable of consultative transfers by themselves... Best regards, -- Nicolas Gudino House Internet S.R.L. Buenos Aires - Argentina ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Which version of MySQL works with cdr_addon_mysql?
Hi- I've used MySQL, both version 3 and 4 with no trouble. I copied the following from my notes: * MySQL version 4 - use the files in /usr/src. Move to new directory /usr/src/mysql. Install by using rpm -U for each, in this order: * shared-compat * client * devel * server - and follow on screen instructions to create root pw Can't remember whether shared-compat is needed on version 3, but definitely in version 4. Hope this helps, Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Malcolm Bader Sent: Saturday, July 31, 2004 8:00 AM To: astrerisk users Subject: [Asterisk-Users] Which version of MySQL works with cdr_addon_mysql? I'm having problems compiling cdr_addon_mysql with MySQL 3.23.58 I get the following errors: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:107: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:107: (Each undeclared identifier is reported only once cdr_addon_mysql.c:107: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:417: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 I have loaded: MySQL-3.23.58-1.i386.rpm MySQL-client-3.23.58-1.i386.rpm MySQL-devel-3.23.58-1.i386.rpm MySQL-shared-3.23.58-1.i386.rpm Does cdr_addon_mysql work with this version of MySQL or do I need to install version 4? Thanks Malcolm Bader ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk scalability?
Hi I plan to setup an asterisk box to function as a SIP gateway forwarding lots of calls to/from a backend of several other asterisk boxes, each with a TE410 card for PSTN connectivity. It will only gateway the calls into the PSTN gateways. No transcoding is planned - only plain ALAW. How many concurrent calls would you think this can handle? I'm asked to plan a system that can handle 1000 concurrent calls... thanks for any input regards roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk does not disconnect SIP call
Hello everybody, my situation is the following: I have an ISDN telephone connected to a HFC ISDN card on an asterisk server. The asterisk server is behind a NAT, but all the ports (i.e. 5060 and the range specified in rtp.conf) are forwarded to the asterisk machine. I am using the German SIP provider Sipgate.de. The sip commands show that I am registered properly with Sipgate. My problem is that when I want to call via the Sip provider a real phone number (ISDN phone SIP), I get a ring tone. When I now decide to hang up (f.e. when nobody answers), the called telephone continues to ring forever. This error shows up: app_dial.c : 362 wait_for_answer: Unable to forward frame If the other party answers and I am the first one to hang up, the call sometimes does not get cancelled as well. The called party has to hang up first to really disconnect the call. This error is not yet reproducable, as I said, sometimes it works and asterisk hangs up correctly. I am using the lastest cvs version of asterisk that automatically installs with the install script of the brifstuff von www.junghanns.net. Any suggestions how to solve it? Christopher ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling * on OpenBSD 3.5
Fantastic - Many thanks!
For the purposes of the archive, this is what I did..
Edited /usr/src/asterisk/Makefile
Just after:-
ifeq (${OSARCH},Darwin)LIBS+=-lresolvendififeq (${OSARCH},FreeBSD)LIBS+=-lcryptoendifLIBS+=-lssl
I added:-
ifeq (${OSARCH},OpenBSD)LIBS=-lcrypto -lpthreadendif
And it compiled just fine..
Is this something us OpenBSD fans will always have to do - or should I submit this to the -dev list?
Michael.[EMAIL PROTECTED] wrote:
On Fri, Jul 30, 2004 at 06:33:14PM -0700, [EMAIL PROTECTED] wrote: Hi, Has anyone had any success?
sched.o: In function `sched_context_create': /usr/src/asterisk/asterisk/include/asterisk/lock.h:299: undefined reference to `pthread_mutexattr_init' /usr/src/asterisk/asterisk/include/asterisk/lock.h:300: undefined reference to `pthread_mutexattr_settype' /usr/src/asterisk/asterisk/include/asterisk/lock.h:301: undefined reference to `pthread_mutex_init' sched.o: In function `sched_context_destroy': Modify the above Makefile (or CFLAGS) to include -pthread for gcc (or possibly,-lpthread, but I think in general for BSD's its -pthread). and
/usr/lib/libssl.so.8.0: undefined reference to `ERR_load_strings' /usr/lib/libssl.so.8.0: undefined reference to `EVP_md2' /usr/lib/libssl.so.8.0: undefined reference to `HMAC_CTX_init' /usr/lib/libssl.so.8.0: undefined reference to `EVP_get_digestbyname' collect2: ld returned 1 exit status gmake: *** [asterisk] Error 1 Try adding -lcrypto to the above Makefile entry (and/or CFLAGS) as well.
Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem
Yes, NAT is a problem. Due to the changes on the Broadvoice side, my router will not work anymore. I will change the router for linksys which works for other people. This was also recommended by James Jones from broadvoice tech support. One more time, thanks for your support James. Bart, Tommorow I will remove NAT and will connect my modem directly. Thanks Thanks Chris, I will try more things tommorow. The thing that broadvoice is awesome I know. It worked for me till last Sunday. It worked with nat=on settings. Then, on Sunday it broke completely. I used it with qos running on linux.(simple HTB) Thanks Bart (simple - Original Message - From: Bartosz Wegrzyn [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 6:54 PM Subject: Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem I took your advice and also created two entries for incoming calls to overcome the DNS problem. NOW I will track incoming call to my * box. I see first how asterisk registers:(before the call) 20:10:37.521845 ns.0.10.10.in-addr.arpa.5060 147.135.0.128.5060: udp 379 (DF) [tos 0x18] 20:10:37.581972 147.135.0.128.5060 ns.0.10.10.in-addr.arpa.5060: udp 352 I am calling into my box. Asterisk picks up. My default messeage starts playing, but I cannot hear it. I look at the tcpdump output to see the packets flow: 20:25:36.492039 147.135.0.128.5060 ns.0.10.10.in-addr.arpa.5060: udp 352 20:25:39.538052 147.135.0.128.5060 ns.0.10.10.in-addr.arpa.5060: udp 718 20:25:39.538544 ns.0.10.10.in-addr.arpa.5060 147.135.0.128.5060: udp 474 (DF) [tos 0x18] 20:25:39.539005 ns.0.10.10.in-addr.arpa.5060 147.135.0.128.5060: udp 475 (DF) [tos 0x18] 20:25:39.540286 ns.0.10.10.in-addr.arpa.5060 147.135.0.128.5060: udp 655 (DF) [tos 0x18] 20:25:39.540799 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.560257 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.580238 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.600240 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.620242 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.640239 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.660103 147.135.0.128.5060 ns.0.10.10.in-addr.arpa.5060: udp 454 20:25:39.660268 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.673145 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.680241 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.700238 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.704918 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.720239 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.724855 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.740239 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.744548 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.760238 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.780239 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.800239 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.813456 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.815961 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.817976 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 Looks like that broadvoice and I talk to each other. But I still cannot hear anything. To test it deeper I setup an extension that will call one of my phones at home. I call again to my * box and press nine. On my asterisk it looks like this: -- Executing Ringing(SIP/192.168.0.3-08d6b540, ) in new stack -- Executing Goto(SIP/192.168.0.3-08d6b540, menu|s|1) in new stack -- Goto (menu,s,1) -- Executing DigitTimeout(SIP/192.168.0.3-08d6b540, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(SIP/192.168.0.3-08d6b540, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround(SIP/192.168.0.3-08d6b540, 3) in new stack -- Playing '3' (language 'en') -- Executing BackGround(SIP/192.168.0.3-08d6b540, closed) in new stack -- Playing 'closed' (language 'en') == CDR updated on SIP/192.168.0.3-08d6b540 -- Executing Dial(SIP/192.168.0.3-08d6b540, Zap/2|30|m) in new stack -- Called 2 -- Started music on hold, class 'default', on SIP/192.168.0.3-08d6b540 -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 answered SIP/192.168.0.3-08d6b540 -- Stopped music on hold on
Re: [Asterisk-Users] one extention, multiple phones
I think you have to use parking. - Original Message - From: Sean McKay [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 31, 2004 9:36 AM Subject: [Asterisk-Users] one extention, multiple phones Is it possible to get a few 7960's and asterisk to allow all of the 7960 phones to use one extentsion and can only be used by one person at a time, have it indicate on the other 7960's when one of the others has the line engaged. Basicly so like I can setup a rule when an incoming call comes from IAX to divert to this extension, it will ring the extension (thus all phones), and allow me to place a call on hold on one phone and pick it up on another and the original phone would acknowledge that the call has been picked up and disengage. Can I do this without call parking? Basicly the same model as having a bunch of phones on a pstn line with each phone having a hold button. The goal here is to allow me to pick up a call on any of the 7960's anywhere in my house and be able to move from room to room as needed by placing the call on hold and picking it up on one of the other phones in the house. /\ . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . \ / - ASCII Ribbon Campaign . Sean McKay - [EMAIL PROTECTED] X - NO HTML/RTF in e-mail . Team Lead, bahamut web team / \ - NO Word docs in e-mail . ircd-qa team member ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Thanks I don't understand why sipura can do consultative transfer and why grandstream can't. They're both SIP, aren't they ? If you want consultative transfers with asterisk, you can sort of have it by using parking: you can dial '#' to transfer, then send the call to the parked calls extension, and the parked extension will be read back to you. Then you hangup and talk to the extension you want the call to be transferred: 'you have Bob on the extension 702'. The other party can now dial that extension and talk to Bob. Its not a consultative transfer as regular phone users are accustomed, but it works. That's how my app work. Except that it's automatic. Blind transfer to 76 for parking. Blind transfer to 77 for unparking to the second party. Maybe your application cann fill the gap for sip devices that are not capable of consultative transfers by themselves... Yep. Maybe I'll add conferencing transfer. Blind transfer to 74 will put everyone in a conf. As soon as one quits the conf, the conf will turn down to a normal call. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adding SIP Based Termination
Hi, I've had a look at it and the timeout error is what happens straight after the phone disconnects: Aug 1 04:07:13 WARNING[106511]: pbx.c:922 pbx_substitute_variables_temp: The use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo' Aug 1 04:07:20 WARNING[5126]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Aug 1 04:07:23 WARNING[106511]: pbx.c:1924 ast_pbx_run: Timeout, but no rule 't' in context 'sip' Aug 1 04:10:01 WARNING[109583]: pbx.c:1924 ast_pbx_run: Timeout, but no rule 't' in context 'sip' Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 219.88.229.122;branch=z9hG4bKb552afc0d1c81034 From: Sahil Gupta sip:[EMAIL PROTECTED];tag=f7e5481bb929c765 To: sip:[EMAIL PROTECTED];tag=as269fa212 Call-ID: [EMAIL PROTECTED] CSeq: 28952 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 219.88.229.122:5060 Any ideas on that error? A quick search on google didn't bring up much. Cheers, Sahil Quoting Greg Hill [EMAIL PROTECTED]: On Sun, 1 Aug 2004 [EMAIL PROTECTED] wrote: exten = _34.,1,Dial(SIP/[EMAIL PROTECTED],20,tr) The r at the end of this line tells asterisk to generate a ringing sound for you to hear. In other words, the ringing you're hearing isn't coming from the far end SIP device. Taking the r out will probably help you get a little closer to the solution.. another thing you can do is turn sip debug on while you try to place the call and see what happens. Watch for responses from the far end sip server, etc. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Silence suppression (was: Re: RAID affecting X100P performance...)
In article [EMAIL PROTECTED], Mike Benoit [EMAIL PROTECTED] wrote: I also discovered my SPA-2000's silence suppression was causing a good chunk of choppiness (much more so then any SS should), so I disabled that too. Asterisk requires that SIP devices have silence suppression disabled. It uses the incoming audio stream as a timing reference for the outgoing stream. If the incoming stream stops (due to suppressed silence), then the outgoing stream stops too. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Which version of MySQL works with cdr_addon_mysql?
In article [EMAIL PROTECTED], Malcolm Bader [EMAIL PROTECTED] wrote: I'm having problems compiling cdr_addon_mysql with MySQL 3.23.58 I get the following errors: cc -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql -c -o cdr_addon_mysql.o cdr_addon_mysql.c cdr_addon_mysql.c:50: warning: parameter names (without types) in function declaration cdr_addon_mysql.c:50: warning: data definition has no type or storage class cdr_addon_mysql.c: In function `mysql_log': cdr_addon_mysql.c:107: `mysql_lock' undeclared (first use in this function) cdr_addon_mysql.c:107: (Each undeclared identifier is reported only once cdr_addon_mysql.c:107: for each function it appears in.) cdr_addon_mysql.c: In function `usecount': cdr_addon_mysql.c:417: `mysql_lock' undeclared (first use in this function) make: *** [cdr_addon_mysql.o] Error 1 Those errors are nothing to do with the version of MySQL. They indicate that you are compiling a recent version of cdr_addon_mysql with an older version of Asterisk. Specifically, a version of Asterisk from before the BSD portability fixes were added. Try fetching the latest CVS HEAD version of Asterisk, compile and install it, and then try compiling cdr_addon_mysql again. Oh, and I'm using MySQL 3.23.58 with no problems at all. Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?
[EMAIL PROTECTED] (Rich Adamson) writes: Like *, it also has an internal dialplan, however understanding the various interactions requires some experimentation, as each of the interfaces seem to be considered a gateway, and part of the dialplan directs calls to gw0, gw1, gw2 (etc) which correspond to physical interfaces in most cases. I felt some pangs of guilt turning all that stuff off, but I couldn't think of any time I'd want two dialplans in series. The box was truly targeted for the residential user where existing phones interface on one side, the pstn line on the other side, and the default call is sent to the voip interface. Disconnected (or failed) ethernet results in a relay flipping, tying the fxs directly to the fxo. Same with power failure. Nice. I think the cut-through from the fxs to the fxo (and backwards) is via a digital connection. In normal use you appear to end up getting hit by the digitization delays. As far as I can tell the relay cut-through is only used for power failure. Initial tests did not show any signs of echo, very good volume and audio quality, and would probably be a good choice for small quantities of pstn lines (particularily soho and residential users). I still notice some low-volume problems with FXO-asterisk-grandstream-bt101 even though I bumped the FXO incoming (and outgoing) gains to +12dB. (To keep calls from the FXO-asterisk-FXS a reasonable volume I needed to drop the gain of the fxs port to -15 (from the factory of -3). Somebody with a real phone VU meter needs to have a look at the Sipura-3000 FXO. I can't believe it is off that much. Might the Grandstream BT-101 be really low in volume and I'm just mistakenly blaming the volume problem on the Sipura? The only downside I've seen thus far (not much experience as yet) is that * calls to the pstn line are cut through immediately, so one hears the initial dialtone from the pstn and the sending of the dtmf tones on all outgoing calls. Kind of annoying, but there might be some config option to handle it; I've just not found it as yet. (If anyone knows how to handle that, sure would appreciate a suggestion.) Given the choice between hearing dead air and hearing the tones, I think I'd rather hear the tones. At least I know something is happening. -wolfgang -- Wolfgang S. Rupprechthttp://www.wsrcc.com/wolfgang/ openbsd amd64 http://www.wsrcc.com/wolfgang/ftp/asterisk-openbsd35.patch ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Trunk doesn't work Adit 600/T100P
Hi ! I am connecting to Adit 600 thru a T100P card I have configured 1-16 FXS channels and 17-24 FXO. Everything looks fine on Asterisk side I get a tone on all FXS channels, but when I try to dialout thru one of the FXO channels 17-24 it doesn't connect to the POTS line and echoes back my voice. I use fxsls and fxols for the T1 channels and ls on Adit side. Whats wrong here ? here is my Adit conf voip-pbx print config - -Cactus.lite configuration file -Created on 01/01/1999 at 00:02:49 for adnan -This file is valid for the following configuration only: - -CardType - -SLOT AT1x2 Code Revision: 1.3.1 -SLOT 1FXSx8 -SLOT 2FXSx8 -SLOT 3FXSx8 -SLOT 4FXSx8 -SLOT 5FXOx8 -SLOT 6FXSx8 - -Note: Lines beginning with '-' will be ignored as comme -by the CLI. Before downloading, review the sections of -configuration file delimited by these comments and delet -the commands that are not needed (e.g. 'set ip address' -and 'add user' are likely candidates for deletion). - -While downloading, a character and line delay of 5 ms is -recommended. - -Turning off verification messages. set verification off -Setting local off. set local off -Disconnecting all connections. disconnect a disconnect 1 disconnect 2 disconnect 3 disconnect 4 disconnect 5 disconnect 6 -Setting users. add user adnan -Setting network id. set id voip-pbx -Setting primary and secondary clock sources. set clock1 a:1 set clock2 internal -Setting IP addresses. set ethernet ip address 192.168.7.151 255.255.255.192 set ip gateway 0.0.0.0 -Setting the SNMP MIB-II System Group objects. set snmp getcom public set snmp setcom public set snmp trapcom public -Setting slot a. set a:1 up set a:1 fdl none set a:1 lbo 1 set a:1 framing esf set a:1 id CAC DS1# 01 set a:1 linecode b8zs set a:1 loopdetect on set a:1 threshold min15 uas default set a:1 threshold min15 ses default set a:1 threshold min15 es default set a:1 threshold min15 sefs default set a:1 threshold min15 les default set a:1 threshold min15 css default set a:1 threshold min15 bes default set a:1 threshold min15 dm default set a:1 threshold min15 lcv default set a:1 threshold min15 pcv default set a:1 threshold day uas default set a:1 threshold day ses default set a:1 threshold day es default set a:1 threshold day sefs default set a:1 threshold day les default set a:1 threshold day css default set a:1 threshold day bes default set a:1 threshold day dm default set a:1 threshold day lcv default set a:1 threshold day pcv default set a:1:1-24 signal ls set a:1:1-24 type voice set a:2 down set a:2 fdl none set a:2 lbo 1 set a:2 framing esf set a:2 id CAC DS1# 02 set a:2 linecode b8zs set a:2 loopdetect on set a:2 threshold min15 uas default set a:2 threshold min15 ses default set a:2 threshold min15 es default set a:2 threshold min15 sefs default set a:2 threshold min15 les default set a:2 threshold min15 css default set a:2 threshold min15 bes default set a:2 threshold min15 dm default set a:2 threshold min15 lcv default set a:2 threshold min15 pcv default set a:2 threshold day uas default set a:2 threshold day ses default set a:2 threshold day es default set a:2 threshold day sefs default set a:2 threshold day les default set a:2 threshold day css default set a:2 threshold day bes default set a:2 threshold day dm default set a:2 threshold day lcv default set a:2 threshold day pcv default set a:2:1-24 signal ls set a:2:1-24 type voice -Setting slot 1. set 1:1-8 signal ls set 1:1-8 txgain 0 set 1:1-8 rxgain 0 set 1:1-8 linelength short -Setting slot 2. set 2:1-8 signal ls set 2:1-8 txgain -3 set 2:1-8 rxgain -6 set 2:1-8 linelength short -Setting slot 3. set 3:1-8 signal ls set 3:1-8 txgain -3 set 3:1-8 rxgain -6 set 3:1-8 linelength short -Setting slot 4. set 4:1-8 signal ls set 4:1-8 txgain -3 set 4:1-8 rxgain -6 set 4:1-8 linelength short -Setting slot 5. set 5:1-8 signal ls set 5:1-8 txgain 0 set 5:1-8 rxgain 0 -Setting slot 6. set 6:1-8 signal ls set 6:1-8 txgain -3 set 6:1-8 rxgain -6 set 6:1-8 linelength short -Making connections. connect a:1:1-8 1:1-8 connect a:1:9-16 2:1-8 connect a:1:17-24 5:1-8 -Turning verification on. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?
The box was truly targeted for the residential user where existing phones interface on one side, the pstn line on the other side, and the default call is sent to the voip interface. Disconnected (or failed) ethernet results in a relay flipping, tying the fxs directly to the fxo. Same with power failure. Nice. I think the cut-through from the fxs to the fxo (and backwards) is via a digital connection. In normal use you appear to end up getting hit by the digitization delays. As far as I can tell the relay cut-through is only used for power failure. It's actually a relay, and you can hear/feel it. The cut-through actually works by either removing power, or, removing the cat5 cable. However, it wouldn't have a clue whether a layer-3 box (including *) were down. Initial tests did not show any signs of echo, very good volume and audio quality, and would probably be a good choice for small quantities of pstn lines (particularily soho and residential users). I still notice some low-volume problems with FXO-asterisk-grandstream-bt101 even though I bumped the FXO incoming (and outgoing) gains to +12dB. (To keep calls from the FXO-asterisk-FXS a reasonable volume I needed to drop the gain of the fxs port to -15 (from the factory of -3). Somebody with a real phone VU meter needs to have a look at the Sipura-3000 FXO. I can't believe it is off that much. Might the Grandstream BT-101 be really low in volume and I'm just mistakenly blaming the volume problem on the Sipura? That's odd; sort of sounds like BT101 problem. Using C7960's, the volume was excellent (without touching anything). Using an analog set on the fxs port was very hot, and dropping the fxs gain slightly improved that to what a non-technical user would suggest is normal. (I did use a $3500 transmission test set on as well.) Given the choice between hearing dead air and hearing the tones, I think I'd rather hear the tones. At least I know something is happening. I'd suspect that non-technical users would raise a small issue with the tone feedback (at least in the US), as their not acustomed to hearing that on normal calls. Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium FXO Interfaces don't support groundstart???
First of all I use this in humor: Frank you ignorant slut! I have to disagree on your analysis. I worked in telephone COs (DMS250, Stromberg/Carlson) and with PBXes for over a decade. Glare can and is controlled by ground start signaling. It does so because the ground is tested for (or supposed to be) prior to dialing. It's called the pre-seize condition. On a T1 using robbed bit signaling, tip and ring conditions are converted into A/B signaling states in the channel modules of a channel bank. Ground start was the prefered signaling system for what was called Feature Group D trunks between Other Common Carriers and the RBOCs. Before FGD was available, we used loop start. We had incoming and outgoing trunk groups, hence no glare... Needless to say expensive. Because FGD had ground start, to cut interconnect costs, we went there as soon as it was made available. The 150ms pulse you described is called wink start, which was funky. I most commonly say it on systems using EM signaling. Gawd I hated those! YA know, the asterisk list has been for a lot of walks down memory lane :) Frank Cofer wrote: Glare cannot be prevented on two way trunks (it is physically impossible because the two ends are separated in distance and therefore separated in time and any independent decision to use it at one end is never seen instantly at the other end). Ground start does not decrease glare at all (it actually increases it) and use of ground start to eliminate glare is a common myth. This is because use of ground start (which uses only one side of the pair to earth ground to start a request for service) increases the time to mark a central office line busy when it is seized from the Customer Premise Equipment (CPE), owing to its clunky signaling (150ms earth ground on the ring of the line) and the fact that it uses only one half of the current to start the line as loop start. Since it increases the time to signal the distant end, it increases glare. Its only benefit is to the central office because it stops a second seizure to the central office when a call disconnects from the central office end first, which would otherwise find a request (loop) as soon as the disconnect was effected. This is why ground start was introduced by the Bell System (when they owned both the PBX and the CO) since it would reduce the attempt load on the central office from large business users by 25% or more saving a lot of central office gear for a relatively small expenditure on the PBX end. Ground start has some ugly drawbacks, since it reduces signaling range, requires the normally isolated floating pair to be referenced to earth ground (which exposes the circuit to longitudinal spikes, noise and lightning) and requires the circuit to be muted during the imbalanced condition that occurs when the ring conductor is momentarily grounded to draw dial tone. Digium is right to leave it out. Most other informed, modern manufacturers do likewise. Ground start signaling referred to in T1 (which is an absurd label since there is no ground placed on a T1) is really after the Grey Code (only one signaling bit transitions at a time) and has nothing to do with glare or ground start signaling and is just a carry over label. Glare can be reduced by changing the hunt order from either end and to employ faster signaling. The former method decreases the likelihood that both ends will compete for the circuit at the same time and the latter reduces the window that a commitment has been made at one end and is still not known by the other end. Typically, the CO is set to hunt ascending and the CPE descending and this is still employed even in ISDN circuits. This is a terminal hunt and NOT a round robin hunting sequence. If you want to absolutely eliminate glare, use one way (incoming/outgoing only) circuits. I believe asterisk has a feature to set the hunt order preference. The disconnect problems you experienced with your Agilent PBX may be more likely related to the guard interval that a circuit is left alone at your end after it is used. Though ground start will appear to fix it, there are some issues of CO message rate three way calling that have caused grief (the CO interprets the next call as a flash for a three way call and holds the circuit rather than disconnecting it). This phenomena may have been misdiagnosed as glare, since the message unit 3-way calling was imposed as a default feature in certain jurisdictions. Increasing the guard interval to 2 or 3 seconds will suffice, or specify to the carrier that the 3-way calling is to be denied for your lines. Hope this helps. - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 08, 2004 4:55 AM Subject: [Asterisk-Users] Digium FXO Interfaces don't support groundstart??? Hi All, I was surprised to be told by a Digium support person today that Digium's FXO interfaces (X100P, TDM400P FXO modules) don't support groundstart signalling.
Re: [Asterisk-Users] queue_log question: which endpoint was connected?
Hello, is there a way I can obtain the IP endpoint address when the telephone is called from app_queue? I even tried creating a pseudo number, so that instead of having my queue call straight (say) OH323/1234 I call a number on asterisk where I log the call id and then do the dialling. Of couse the OH323 call id I find in queue_log is different from the one I log there from AGI, so this attempt was useless. I'd try and hack app_queue directly, but can anybody tell me where to find the number dialed by queue_app? is there in the channel datastructure? thanks l. In data Fri, 30 Jul 2004 19:35:13 +0300, Michael Manousos [EMAIL PROTECTED] ha scritto: The IP of the connected endpoint can be obtained from the OH323_RADDR variable. For incoming H.323 calls you can get the name of the channel and the IP address inside the dialplan, write them to a file and process them later. For outgoing H.323 calls [Dial(OH323/...)], you can't do it from the dialplan. In that case the OH323_RADDR variable is accessible only through the Dial() app. Anyway, it seems that the name of the OH323 channels needs to be more useful (added to my TODO list). Any help will be greatly appreciated. Thanks l. Michael. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Creato con M2, il rivoluzionario client e-mail di Opera: http://www.opera.com/m2/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?
Wolfgang S. Rupprecht [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] (Rich Adamson) writes: Like *, it also has an internal dialplan, however understanding the various interactions requires some experimentation, as each of the interfaces seem to be considered a gateway, and part of the dialplan directs calls to gw0, gw1, gw2 (etc) which correspond to physical interfaces in most cases. I felt some pangs of guilt turning all that stuff off, but I couldn't think of any time I'd want two dialplans in series. It saves having to wait for an inter-digit timeout to expire when dialling via the FXS port. The SPA-[123]000 dialplan will recognise your dial string and send it immediately to Asterisk (if so configured). Asterisk will take it from there. You can apparently use the SPA-3000 dialplan to specify that the call should go via its FXO port, without going via Asterisk. This could be useful for emergency services. I don't have a SPA-3000 yet, so I can't say what happens if you try to route an emergency call via the FXO port and that port is in use. Perhaps it sends the call to Asterisk instead. I'll find out when I get mine and play with it. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP Cisco ATA 186 Help
Does anybody has the expirience configuring Asterisk with Cisco ATA 186 MGCP firmware ? I have Cisco software v3.1.1 atamgcp (Build 040629A) Asterisk 1.0-RC1 On ATA i only put domain test. mgcp.conf looks like this [test] host = 192.168.195.55 context = default line = aaln/2 line = aaln/1 Asterisk CLI shows this: Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485 __mgcp_xmit: mgcp_xmit returned -1: Address family not supported by protocol family *CLI Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt: Maximum retries exceeded for transaction 1 on [test] Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt: Maximum retries exceeded for transaction 2 on [test] Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response: Transaction 2 timed out Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response: Transaction 1 timed out *CLI Jul 31 16:05:44 NOTICE[135449600]: chan_mgcp.c:1474 find_subchannel: Gateway 'test' (and thus its endpoint '*') does not exist mgcp debug MGCP Debugging Enabled *CLI MGCP read: RSIP 1 [EMAIL PROTECTED] MGCP 1.0 RM: restart from 192.168.195.55:2427Verb: 'RSIP', Identifier: '1', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Jul 31 16:06:03 NOTICE[135449600]: chan_mgcp.c:1474 find_subchannel: Gateway 'test' (and thus its endpoint '*') does not exist -- WBR - Dmitry Baranov Phone: +(372) 6 880 000 STV Internet Fax:+(372) 6 880 550 Valge 6 Mobile: +(372) 5 012 825 Tallinn, Estonia - ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk scalability?
Hi Roy- I've done a lot of load testing with asterisk and TE410P's. My guess, with no transcoding, is that you might be able to handle 8 E1's max on the PSTN side absolute max (ie: 2 TE410P's). This assumes you have a fast processor.If you're using T1's, scale these numbers up accordingly, as there are fewer channels per span. If this answer is lower than you might expect, consider that every byte of data has to pass through the processor. The 410's are capable of bus-mastering, and so are an improvement over the T400P's, but still I think you run into horsepower limitations. Regards Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California London England www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk Sent: Saturday, July 31, 2004 8:25 AM To: Asterisk Users Subject: [Asterisk-Users] Asterisk scalability? Hi I plan to setup an asterisk box to function as a SIP gateway forwarding lots of calls to/from a backend of several other asterisk boxes, each with a TE410 card for PSTN connectivity. It will only gateway the calls into the PSTN gateways. No transcoding is planned - only plain ALAW. How many concurrent calls would you think this can handle? I'm asked to plan a system that can handle 1000 concurrent calls... thanks for any input regards roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk scalability?
Roy Sigurd Karlsbakk wrote: Hi I plan to setup an asterisk box to function as a SIP gateway forwarding lots of calls to/from a backend of several other asterisk boxes, each with a TE410 card for PSTN connectivity. It will only gateway the calls into the PSTN gateways. No transcoding is planned - only plain ALAW. How many concurrent calls would you think this can handle? I'm asked to plan a system that can handle 1000 concurrent calls... Search the archives and the wiki. Look for a thread a few months ago called Asterisk on 64-bit Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Best Linux for Asterisk
I am running * CVS head on Gentoo/i586 and Gentoo/Sparc64 (US60 2x450/1GB RAM), they are running great. On sparc64 * does not compile out-of-the-box, some hackings in the Makefiles are needed, Ming-Wei ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adding SIP Based Termination
On Sun, 1 Aug 2004 [EMAIL PROTECTED] wrote: Hi, I've had a look at it and the timeout error is what happens straight after the phone disconnects: Aug 1 04:07:13 WARNING[106511]: pbx.c:922 pbx_substitute_variables_temp: The use of 'EXTEN-foo' has been deprecated in favor of 'EXTEN:foo' Aug 1 04:07:20 WARNING[5126]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call [EMAIL PROTECTED] for seqno 102 (Non-critical Request) Aug 1 04:07:23 WARNING[106511]: pbx.c:1924 ast_pbx_run: Timeout, but no rule 't' in context 'sip' Aug 1 04:10:01 WARNING[109583]: pbx.c:1924 ast_pbx_run: Timeout, but no rule 't' in context 'sip' Reliably Transmitting (no NAT): SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 219.88.229.122;branch=z9hG4bKb552afc0d1c81034 From: Sahil Gupta sip:[EMAIL PROTECTED];tag=f7e5481bb929c765 To: sip:[EMAIL PROTECTED];tag=as269fa212 Call-ID: [EMAIL PROTECTED] CSeq: 28952 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 to 219.88.229.122:5060 Any ideas on that error? A quick search on google didn't bring up much. 403 forbidden usually means you didn't authenticate correctly to the other SIP endpoint. IIRC, your sip.conf section for your this provider included host, secret, and username. Sometimes you need to use fromuser and fromdomain as well -- sometimes you're expected to identify yourself as [EMAIL PROTECTED] or whatever instead of using [EMAIL PROTECTED] (this is what asterisk will use by default). You would make asterisk identify itself the other way by using fromuser=12345 and fromdomain=siptermination.com in the appropriate section of your sip.conf. Give that a try and let us know what happens.. Another thing you could try would be to make a softphone like x-ten lite, msn messenger, or one of the linux varieties connect to your provider. Sometimes they're a little easier to get working because they don't have so many little things you can tweak. After you have a known good configuration there, you could do a sip debug or network packet dump to see the communication it's making to the provider, and then compare that with what asterisk says when it talks to the provider. Greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Cisco ATA 186 Help
I've got it to work in the past. I've upgraded to SIP, seems to work better. Is there a reason you MUST have MGCP? Duane - Original Message - From: Dmitri Baranov To: [EMAIL PROTECTED] Sent: Saturday, July 31, 2004 12:38 PM Subject: [Asterisk-Users] MGCP Cisco ATA 186 Help Does anybody has the expirience configuring Asterisk with Cisco ATA 186MGCP firmware ? I have Cisco software v3.1.1 atamgcp (Build 040629A) Asterisk 1.0-RC1 On ATA i only put domain test. mgcp.conf looks like this [test] host = 192.168.195.55 context = default line = aaln/2 line = aaln/1 Asterisk CLI shows this: Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485 __mgcp_xmit:mgcp_xmit returned -1: Address family not supported by protocol family *CLI Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt:Maximum retries exceeded for transaction 1 on [test] Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt: Maximumretries exceeded for transaction 2 on [test] Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response:Transaction 2 timed out Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response:Transaction 1 timed out *CLI Jul 31 16:05:44 NOTICE[135449600]: chan_mgcp.c:1474find_subchannel: Gateway 'test' (and thus its endpoint '*') does notexist mgcp debug MGCP Debugging Enabled *CLI MGCP read: RSIP 1 [EMAIL PROTECTED] MGCP 1.0 RM: restart from 192.168.195.55:2427Verb: 'RSIP', Identifier: '1', Endpoint:'[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Jul 31 16:06:03 NOTICE[135449600]: chan_mgcp.c:1474 find_subchannel:Gateway 'test' (and thus its endpoint '*') does not exist-- WBR - Dmitry Baranov Phone: +(372) 6 880 000 STV Internet Fax: +(372) 6 880 550 Valge 6 Mobile: +(372) 5 012 825 Tallinn, Estonia -___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] learning from the audio folks
Besides playing with Asterisk, i'm also using Linux for all kinds of multimedia things, especially recording music, mixing, etc. In order to use Linux as a digital audio workstation, there are a few things that one must do: use low-latency kernels, use pre-emption, use apps that run with real-time privileges, etc. For example, among audio Linux users, the CK (Con Kolivas) and LCK (Locosoft CK) patches are popular: http://members.optusnet.com.au/ckolivas/kernel/ http://www.plumlocosoft.com/kernel/ These patches provide O(1) scheduler, pre-emption, low latency, variable Hz, and other improvements that the audio community found not only useful, but actually required to do any kind of serious audio work with Linux. Some of those patches were integrated into kernel 2.6, so the CK patch for 2.6 is smaller than LCK. Also, JACK, the professional audio daemon for Linux, has options for running with real-time privileges. It crossed my mind that Asterisk performs a job quite similar to JACK. The problems that the audio community see with JACK (dropped audio frames, jitter, etc.) are not unheard of to Asterisk users. Therefore: - does it makes sense to experiment with the kernel audio patches? - if Asterisk doesn't already do that (correct me if i'm wrong), does it make sense to make it run with real-time privileges, just like JACK? (i have no idea how JACK accomplishes that, to me it's just a command-line option that makes it a lot more reliable) Anyone running Asterisk on top of a 2.4 LCK kernel? -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] different pridialplan for different channels in zapata.conf
Hello, I read the previous postings in asterisk-users mailinglists and I didnt found any postings related to my problematic topic. Problem: If I need to set different pridialplan for different channels. For example: group1 has first 15 channels and all calls what are sendt via this group are pridialplan=national group2 has next 15 channels and all calls what are sent via this group are pridialplan=international Is there any way to present it so, as I described? Thanks for the attention. Best Regards: Key Aavoja /* Never argue with an idiot. They drag you down to their level, then beat you with experience.*/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Softphone - Freeware?!
Eric Bart wrote: Thanks I don't understand why sipura can do consultative transfer and why grandstream can't. They're both SIP, aren't they ? They use different sip stacks... and yes, they are both sip. Try to mix in the same environment sipuras, grandstreams, snoms, uniden, saysons, cisco, asterisk, etc. and you will notice that SIP is not the same thing for every vendor. Sipura sip stack is rock solid and very compatible with other vendors. Grandstreams sip implementation is flacky, you will have a crashed phone every now and then. Some versions of snom firmaware work better with asterisk than others. SIP devices have a life on their own..:) Nicolas ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PrePaidCID does core dump..
I am trying to get PrepaidCID working and, it shows it connecting to the database correctly. I call the extension and it Asterisk does a core dump. Can anyone help me? Norm
RE: [Asterisk-Users] one extention, multiple phones
You can easily ring different phones at the same time within the dial
command. For example,
SIP/4024${PRITRUNK1}/16505551212${PRITRUNK1}/1411212
A blind transfer will move the call to the next phone. Or you can park the
call.
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean McKay
Sent: Saturday, July 31, 2004 5:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] one extention, multiple phones
Is it possible to get a few 7960's and asterisk to allow all
of the 7960 phones to use one extentsion and can only be used
by one person at a time, have it indicate on the other 7960's
when one of the others has the line engaged. Basicly so like
I can setup a rule when an incoming call comes from IAX to
divert to this extension, it will ring the extension (thus
all phones), and allow me to place a call on hold on one
phone and pick it up on another and the original phone would
acknowledge that the call has been picked up and disengage.
Can I do this without call parking?
Basicly the same model as having a bunch of phones on a pstn
line with each phone having a hold button. The goal here is
to allow me to pick up a call on any of the 7960's anywhere
in my house and be able to move from room to room as needed
by placing the call on hold and picking it up on one of the
other phones in the house.
/\ . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
. . . . . .
\ / - ASCII Ribbon Campaign . Sean McKay - [EMAIL PROTECTED] X -
NO HTML/RTF in e-mail . Team Lead, bahamut web team / \ - NO
Word docs in e-mail . ircd-qa team member
___
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Re: [Asterisk-Users] Softphone - Freeware?!
I don't understand why sipura can do consultative transfer and why grandstream can't. They're both SIP, aren't they ? They use different sip stacks... and yes, they are both sip. Maybe the sipura transfer is using a sip reinvite or some other SIP command. Does the consultative transfer works when the other parties are not attached to a sipura phone (ie when a sipura phone try to make a consultative transfer from a grandstream to a snom) ? From what you said, I believe that asterisk is not managing these consulative transfers and is not aware of. These are inter-phone communications (peer to peer). Each peer has to understand each other, which is not easy when mixing multiple technologies. Does it sounds right for you ? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Successfully Using $135 Avaya sip phone
Brian Elton wrote: I think I am the first to use the $135 Avaya 4602 SIP phone, but I need some support from the community to fix one problem I have with it. The phone stops working after about 20-30mins if I have mailbox=context in Asterisk; when I do have mailbox=contect in asterisk the sip debug returns 481 extension does not exist. Anyone willing to help me figure out why? I have two debugging suggestions for you: 1) Upgrade to latest CVS 2) Try using ethereal to look at the SIP packets going back and forth before the phone stops working Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem
I am ready to close that topic. Finally, I replaced my router from Multitech for Linksys. It solved all the problems related to NAT and incoming calls issues. My router model is Linksys BEFSX41. Thanks for your help, asterisk people. Bart, Yes, NAT is a problem. Due to the changes on the Broadvoice side, my router will not work anymore. I will change the router for linksys which works for other people. This was also recommended by James Jones from broadvoice tech support. One more time, thanks for your support James. Bart, Tommorow I will remove NAT and will connect my modem directly. Thanks Thanks Chris, I will try more things tommorow. The thing that broadvoice is awesome I know. It worked for me till last Sunday. It worked with nat=on settings. Then, on Sunday it broke completely. I used it with qos running on linux.(simple HTB) Thanks Bart (simple - Original Message - From: Bartosz Wegrzyn [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 6:54 PM Subject: Re: [Asterisk-Users] broadvoice/asterisk incoming calls problem I took your advice and also created two entries for incoming calls to overcome the DNS problem. NOW I will track incoming call to my * box. I see first how asterisk registers:(before the call) 20:10:37.521845 ns.0.10.10.in-addr.arpa.5060 147.135.0.128.5060: udp 379 (DF) [tos 0x18] 20:10:37.581972 147.135.0.128.5060 ns.0.10.10.in-addr.arpa.5060: udp 352 I am calling into my box. Asterisk picks up. My default messeage starts playing, but I cannot hear it. I look at the tcpdump output to see the packets flow: 20:25:36.492039 147.135.0.128.5060 ns.0.10.10.in-addr.arpa.5060: udp 352 20:25:39.538052 147.135.0.128.5060 ns.0.10.10.in-addr.arpa.5060: udp 718 20:25:39.538544 ns.0.10.10.in-addr.arpa.5060 147.135.0.128.5060: udp 474 (DF) [tos 0x18] 20:25:39.539005 ns.0.10.10.in-addr.arpa.5060 147.135.0.128.5060: udp 475 (DF) [tos 0x18] 20:25:39.540286 ns.0.10.10.in-addr.arpa.5060 147.135.0.128.5060: udp 655 (DF) [tos 0x18] 20:25:39.540799 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.560257 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.580238 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.600240 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.620242 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.640239 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.660103 147.135.0.128.5060 ns.0.10.10.in-addr.arpa.5060: udp 454 20:25:39.660268 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.673145 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.680241 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.700238 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.704918 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.720239 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.724855 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.740239 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.744548 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.760238 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.780239 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.800239 ns.0.10.10.in-addr.arpa.18938 147.135.0.128.14384: udp 172 (DF) [tos 0x18] 20:25:39.813456 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.815961 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 20:25:39.817976 147.135.0.128.14384 ns.0.10.10.in-addr.arpa.18938: udp 172 Looks like that broadvoice and I talk to each other. But I still cannot hear anything. To test it deeper I setup an extension that will call one of my phones at home. I call again to my * box and press nine. On my asterisk it looks like this: -- Executing Ringing(SIP/192.168.0.3-08d6b540, ) in new stack -- Executing Goto(SIP/192.168.0.3-08d6b540, menu|s|1) in new stack -- Goto (menu,s,1) -- Executing DigitTimeout(SIP/192.168.0.3-08d6b540, 5) in new stack -- Set Digit Timeout to 5 -- Executing ResponseTimeout(SIP/192.168.0.3-08d6b540, 10) in new stack -- Set Response Timeout to 10 -- Executing BackGround(SIP/192.168.0.3-08d6b540, 3) in new stack -- Playing '3' (language 'en') -- Executing BackGround(SIP/192.168.0.3-08d6b540, closed) in new stack -- Playing 'closed' (language 'en') == CDR updated on SIP/192.168.0.3-08d6b540 -- Executing Dial(SIP/192.168.0.3-08d6b540,
Re: [Asterisk-Users] learning from the audio folks
On Sat, 2004-07-31 at 12:27, Florin Andrei wrote: - if Asterisk doesn't already do that (correct me if i'm wrong), does it make sense to make it run with real-time privileges, just like JACK? (i have no idea how JACK accomplishes that, to me it's just a command-line option that makes it a lot more reliable) I mean, a la SCHED_FIFO: http://www.samspublishing.com/articles/article.asp?p=101760seqNum=4 -- Florin Andrei http://florin.myip.org/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?
Wolfgang S. Rupprecht [EMAIL PROTECTED] wrote: [EMAIL PROTECTED] (Rich Adamson) writes: Like *, it also has an internal dialplan, however understanding the various interactions requires some experimentation, as each of the interfaces seem to be considered a gateway, and part of the dialplan directs calls to gw0, gw1, gw2 (etc) which correspond to physical interfaces in most cases. I felt some pangs of guilt turning all that stuff off, but I couldn't think of any time I'd want two dialplans in series. It saves having to wait for an inter-digit timeout to expire when dialling via the FXS port. The SPA-[123]000 dialplan will recognise your dial string and send it immediately to Asterisk (if so configured). Asterisk will take it from there. You can apparently use the SPA-3000 dialplan to specify that the call should go via its FXO port, without going via Asterisk. This could be useful for emergency services. I don't have a SPA-3000 yet, so I can't say what happens if you try to route an emergency call via the FXO port and that port is in use. Perhaps it sends the call to Asterisk instead. I'll find out when I get mine and play with it. Yes, the dialplan for the fxs line can look like: (*xx|[34569]11:@gw0|0|00|[2-9]xx:@gw0|1xxx[2-9]xxS0|.) where 911 is sent to gw0 (the fxo port), calls to Nxx (local calls) go to gw0, and 1+ calls (long distance) go to a voip box (* in my case) The above is from a test spa3000 that is not in production, so the actual dialplan is not complete as yet. The fxs dialplan is limited to a single line, and can only have a limited number of characters in that line. There are an additional 8 dialplans that appear to be oriented around how to deal with incoming pstn calls, and routing those to * (or whatever). Haven't played with these at all as yet. The doc suggests these 8 dialplans can also be tied in with user assigned pin numbers, allowing a user to call into the spa3000 via the pstn, enter their pin, and be routed to * (or different voip providers). Rather sophisticated little box. :) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail Not releasing
Steve Totaro wrote: [I think you'll find that inline-posting makes treads easier to read] - Original Message - From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, July 30, 2004 9:59 PM Subject: [Asterisk-Users] VoiceMail Not releasing About twice a week we have a caller that comes in and hangs up on voicemail. We have 2 x100ps each with their own irq. When the caller hangs up asterisk does not release the line. The line rings busy, sometimes I can do a soft hangup Zap/1 and release the line sometimes I have stop asterisk and remove and re-insert the modules. I have the same issue with IAX2. I get messages anywhere from 5 min to 45 min of silence. Look in your voicemail.conf for maxsilence and silencethreshold: ; How many seconds of silence before we end the recording maxsilence=10 ; Silence threshold (what we consider silence, the lower, the more sensitive) silencethreshold=128 Nick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP Cisco ATA 186 Help
My ATA with V3.0 firmware works fine. Check that test can be resolved by your DNS or is in /etc/hosts. You might just want to put the IP address directly. Dmitri Baranov wrote: Does anybody has the expirience configuring Asterisk with Cisco ATA 186 MGCP firmware ? I have Cisco software v3.1.1 atamgcp (Build 040629A) Asterisk 1.0-RC1 On ATA i only put domain test. mgcp.conf looks like this [test] host = 192.168.195.55 context = default line = aaln/2 line = aaln/1 Asterisk CLI shows this: Jul 31 16:05:40 WARNING[135449600]: chan_mgcp.c:485 __mgcp_xmit: mgcp_xmit returned -1: Address family not supported by protocol family *CLI Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt: Maximum retries exceeded for transaction 1 on [test] Jul 31 16:05:41 WARNING[135449600]: chan_mgcp.c:595 retrans_pkt: Maximum retries exceeded for transaction 2 on [test] Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response: Transaction 2 timed out Jul 31 16:05:41 NOTICE[135449600]: chan_mgcp.c:2234 handle_response: Transaction 1 timed out *CLI Jul 31 16:05:44 NOTICE[135449600]: chan_mgcp.c:1474 find_subchannel: Gateway 'test' (and thus its endpoint '*') does not exist mgcp debug MGCP Debugging Enabled *CLI MGCP read: RSIP 1 [EMAIL PROTECTED] MGCP 1.0 RM: restart from 192.168.195.55:2427Verb: 'RSIP', Identifier: '1', Endpoint: '[EMAIL PROTECTED]', Version: 'MGCP 1.0' 2 headers, 0 lines Jul 31 16:06:03 NOTICE[135449600]: chan_mgcp.c:1474 find_subchannel: Gateway 'test' (and thus its endpoint '*') does not exist ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Hiring Setup
Sorry to just jump in on the list like this, but I'm in a hurry. I've become lost in the mumbo-jumbo of the Wiki. Is there anyone on this list who would set up a Asterisk system for me? I have a Fedora Core 2 fresh install on an HP Pavilion 6553, along with a single POTS line. I'll buy the hardware I need to add to it to get Asterisk to work. We also have a couple of computers, including 2 PCs and 1 Mac. We want to use these to answer calls. So what I need is someone to assist me in the installation of Asterisk (shouldn't be too hard, I've installed things on Linux before) and more importantly, write some configuration files that will let me do a basic menu system with redirects to the appropriate computer, or voicemail if the computer isn't on. We need three or four extensions but they are split between two computers. Nothing fancy--it's only one line, so no hold or queueing needed. Please e-mail a quote off-list to [EMAIL PROTECTED] if you're interested. Thanks for your time, sorry for the intrusion. Adam Ernst cosmicsoft ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] one extention, multiple phones
On Sat, 31 Jul 2004, Paul Mahler wrote:
You can easily ring different phones at the same time within the dial
command. For example,
SIP/4024${PRITRUNK1}/16505551212${PRITRUNK1}/1411212
A blind transfer will move the call to the next phone. Or you can park the
call.
That's not what I want to do. With a traditional PBX or PSTN setup I can
have more than one phone sharing the same extension. What I want to do is
to be able to have one extension (or line) on say 3 phones. When I pick
up the phone on one, the other two are alerted that the line is engaged
and should give a visual indicator on the screen that the line is in use.
Say I want someone to join in on the conversation, I'd rather much have
them be able to just lift the receiver and begin to talk rather than have
to do conferencing.
This is done on PSTN (normal home phone), and I've seen it done on PBX's
such as the ATT Merlin and NT Meridian systems. Since the application I'd
be setting up is basicly upgrading from PSTN to VOIP I'd like to keep that
feature of being able to share a call (w/o conference) or pick up a call
on another phone without having to use transfer (blind or regular).
Also I'd like to say I believe this is possible with the CCM otherwise
how could an agent (operator) be able to monitor which extensions are in
use with the 7960 expansion device?
Paul Mahler
[EMAIL PROTECTED]
Signate, LLC
665 Third Street
Suite 100
San Francisco, CA
94107-1901
Asterisk Services and Training
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sean McKay
Sent: Saturday, July 31, 2004 5:37 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] one extention, multiple phones
Is it possible to get a few 7960's and asterisk to allow all
of the 7960 phones to use one extentsion and can only be used
by one person at a time, have it indicate on the other 7960's
when one of the others has the line engaged. Basicly so like
I can setup a rule when an incoming call comes from IAX to
divert to this extension, it will ring the extension (thus
all phones), and allow me to place a call on hold on one
phone and pick it up on another and the original phone would
acknowledge that the call has been picked up and disengage.
Can I do this without call parking?
Basicly the same model as having a bunch of phones on a pstn
line with each phone having a hold button. The goal here is
to allow me to pick up a call on any of the 7960's anywhere
in my house and be able to move from room to room as needed
by placing the call on hold and picking it up on one of the
other phones in the house.
/\ . . . . . . . . . . . . . . . . . . . . . . . . . . . . .
. . . . . .
\ / - ASCII Ribbon Campaign . Sean McKay - [EMAIL PROTECTED] X -
NO HTML/RTF in e-mail . Team Lead, bahamut web team / \ - NO
Word docs in e-mail . ircd-qa team member
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RE: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *?
Rich Adamson [EMAIL PROTECTED] wrote: Kevin Walsh wrote: You can apparently use the SPA-3000 dialplan to specify that the call should go via its FXO port, without going via Asterisk. This could be useful for emergency services. I don't have a SPA-3000 yet, so I can't say what happens if you try to route an emergency call via the FXO port and that port is in use. Perhaps it sends the call to Asterisk instead. I'll find out when I get mine and play with it. Yes, the dialplan for the fxs line can look like: (*xx|[34569]11:@gw0|0|00|[2-9]xx:@gw0|1xxx[2-9]xxS0|.) where 911 is sent to gw0 (the fxo port), calls to Nxx (local calls) go to gw0, and 1+ calls (long distance) go to a voip box (* in my case) As I said, I don't have one of these yet. Do you happen to know what the box would do if the dialplan said to route the call to :@gw0 and that port was already in use? If the call simply fails then that's a wasted facility, and I wouldn't use it; If Asterisk was in charge then it could loop the call back to the FXO or sent it via another route. I suspect that the SPA would try the :gw0 first and then fall back to the SIP link, either automatically or as an option, before giving up. Rather sophisticated little box. :) So it would seem. I can't wait to get my hands on one in September. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] learning from the audio folks
Florin Andrei [EMAIL PROTECTED] wrote: On Sat, 2004-07-31 at 12:27, Florin Andrei wrote: - if Asterisk doesn't already do that (correct me if i'm wrong), does it make sense to make it run with real-time privileges, just like JACK? (i have no idea how JACK accomplishes that, to me it's just a command-line option that makes it a lot more reliable) I mean, a la SCHED_FIFO: Asterisk will use SCHED_RR if you use the -p switch upon startup. SCHED_RR is an enhancement to SCHED_FIFO, as explained in the sched_setscheduler(2) manual page. The other patches you mentioned, in your previous article, are mostly included in Linux 2.6. Linux 2.6 is clearly far superior to 2.4, especially for the purposes of applications such as Asterisk. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 480i User Feedback With Asterisk (fwd)
For those that are interested, here is my report back to Sayson on the 480i -- Forwarded message -- Date: Sat, 31 Jul 2004 22:03:31 -0400 (EDT) From: [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: 480i User Feedback With Asterisk Seshu, I am using a 480i, and I am impressed with the phone on a whole, but obviously the firmware is lacking. Details follow. Hold button works, but holds the user at the phone, does not hold them at the PBX, allowing for music on hold. I would also like to see the hold button software addressable so that it could be used to park the call (transfer to the parking extension) rather than only putting the call on hold. Transfer seems to work fine, but would like to see blind transfer (transfer direct to the remote party, rather than the current behavior where you are bridged to the called party, then have to press transfer again to complete the transfer) Maybe a softbutton for blind transfer, and reserve the hard button for attended transfers? It would be nice if we could change the behavior of the transfer button to do this. Conference works, but I have some trouble dropping the correct party from conversation. Also, when conferenced, I am no longer able to send DTMF tones to the second line. The original caller hears the DTMF, but the second party does not. I am using RFC2833 DTMF. Redial seems to work a bit odd. It does redial a previous number, but not the last number dialed. Can't seem to find any rhyme or reason, but it seems do dial the last non PSTN number. For instance, if I dial 50 for voicemail, then dial 93934481, hang up, and press redial, it dials 50, not 93934481. When viewing the SIP config through the menu on the phone, it displays defaults, and not the settings specified via the TFTP configs. If I set qualify=1000 in my sip.conf, Asterisk will send a poke to the phone every few seconds, to make sure it is still alive. If I enable this option, the phone stops working after a few minutes. Asterisk shows the phone as unreachable, and I cannot dial any number from the phone. It will accept the input, but the dialplan does not timeout and dial, nor will it dial if I press the # sign. On the subject of dialplans, I am only able to dial 10 digits, on the 11th digit, the phone tries to dial. This is a bad thing when trying to dial long distance. It is basically impossible. The display occasionally shows L1 in the lower left hand corner of the display. As if someone had pressed the left/right arrows of the navigation pad. It shows up for less than a second, and then goes away. On to the good things now. ;-) Conference basically works, save for the caviats above. The backlight is very nice, and the surface of the LCD does not show fingerprints like the Cisco phones do. Sound quality is good with ULAW codec, but there is noticable echo in the call that is not present when using a Cisco phone, or a Softphone. Transfer seems to work fine. Message lamp and stutter dialtone works fine. I would like to have a way to turn off the stutter tone. If I have a message lamp, there is no need for a stutter tone. The web config is nice, but limited. I would like to be able to set up SIP configs through there. Also it only works from Internet Explorer on Windows. If you are running Netscape/Mozilla/Opera, you cannot use the web config. It should also be fully password protected, so rouge users cannot reboot the phone etc. Custom ringers are nice, and professional, recognize them all from Nortel phones. I would like to be able to specify a SIP Alert_Info so that I can have a different ring from different calling partys. For instance, incoming calls from the PSTN would have one ring, and extension to extension calls would have another ring. Look forward to hearing back from you. Nice work on this phone. If it hits the market with a complete firmware, at $200 or less, they will sell well within the Asterisk community. Jeremy Parr Senior Engineer, Network Services BGC Ltd. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PrePaidCID does core dump..
On Sat, Jul 31, 2004 at 04:36:51PM -0400, Norman Tomlnis wrote: I am trying to get PrepaidCID working and, it show's it connecting to the database correctly. I call the extension and it Asterisk does a core dump. Can anyone help me? If you'd like to read over http://www.mail-archive.com/[EMAIL PROTECTED]/msg43645.html for example it should be somewhat instructive of what you should send to the list in order to help other people work out what your problem is. Norm Thanks, Andrew Griffiths ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 480i User Feedback With Asterisk (fwd)
[EMAIL PROTECTED] wrote: For those that are interested, here is my report back to Sayson on the 480i Thanks for the report, some of us are very interested! Look forward to hearing back from you. Nice work on this phone. If it hits the market with a complete firmware, at $200 or less, they will sell well within the Asterisk community. That is quite true, there is a big market for a phone with these features under $200. The Avaya 4602 is looking promising as well, but the Sayson is even more feature-packed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 480i User Feedback With Asterisk 802.1Q?
Does anyone know if the 480i supports 802.1Q? -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Saturday, July 31, 2004 10:17 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] 480i User Feedback With Asterisk (fwd) [EMAIL PROTECTED] wrote: For those that are interested, here is my report back to Sayson on the 480i Thanks for the report, some of us are very interested! Look forward to hearing back from you. Nice work on this phone. If it hits the market with a complete firmware, at $200 or less, they will sell well within the Asterisk community. That is quite true, there is a big market for a phone with these features under $200. The Avaya 4602 is looking promising as well, but the Sayson is even more feature-packed. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] one extention, multiple phones
On Sat, 2004-07-31 at 20:11, Sean McKay wrote: Say I want someone to join in on the conversation, I'd rather much have them be able to just lift the receiver and begin to talk rather than have to do conferencing. This is done on PSTN (normal home phone), and I've seen it done on PBX's such as the ATT Merlin and NT Meridian systems. Since the application I'd be setting up is basicly upgrading from PSTN to VOIP I'd like to keep that feature of being able to share a call (w/o conference) or pick up a call on another phone without having to use transfer (blind or regular). Also I'd like to say I believe this is possible with the CCM otherwise how could an agent (operator) be able to monitor which extensions are in use with the 7960 expansion device? I think you're talking about two different functions. I'm a little bit rusty on Meridian (it's been a few years since I've worked on them), but I don't think it can do this either. Monitoring an extension for busy status is done all the time (IIRC Nortel calls this busy lamp field), but you still can't just pick up an in-use extension and start talking. You have to bridge the channels. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] one extention, multiple phones
On the Merlin Legend I believe the function you're talking about is on the MLX-20 receptionis's console. When the system is in hybrid-PBX mode, you can simply press a line button that's in-use and you can listen in on the conversation and even talk, it basically puts you in a conference with the other person without actually setting up a conference. Not sure what the equivalent of that in * is... or if there even is one... -Chris - Original Message - From: Reid A. Forrest [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 31, 2004 8:10 PM Subject: RE: [Asterisk-Users] one extention, multiple phones On Sat, 2004-07-31 at 20:11, Sean McKay wrote: Say I want someone to join in on the conversation, I'd rather much have them be able to just lift the receiver and begin to talk rather than have to do conferencing. This is done on PSTN (normal home phone), and I've seen it done on PBX's such as the ATT Merlin and NT Meridian systems. Since the application I'd be setting up is basicly upgrading from PSTN to VOIP I'd like to keep that feature of being able to share a call (w/o conference) or pick up a call on another phone without having to use transfer (blind or regular). Also I'd like to say I believe this is possible with the CCM otherwise how could an agent (operator) be able to monitor which extensions are in use with the 7960 expansion device? I think you're talking about two different functions. I'm a little bit rusty on Meridian (it's been a few years since I've worked on them), but I don't think it can do this either. Monitoring an extension for busy status is done all the time (IIRC Nortel calls this busy lamp field), but you still can't just pick up an in-use extension and start talking. You have to bridge the channels. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Compiling * on OpenBSD 3.5
I found that it was also necessary for me to add -lm to the LIBS line for it to work on my nice fresh OpenBSD 3.5 installation. -- Greg Broiles, JD, EA [EMAIL PROTECTED] (Lists only. Not for confidential communications.) Law Office of Gregory A. Broiles San Jose, CA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
