[Asterisk-Users] Missed Call List on SIP Phones
I have a SNOM 190 phone at home. It works well. On the display it shows the time, 3 soft key labels and also the number of missed calls. If I see a missed call I can use the CallLog soft key to see a list of calls that are divided into:- Missed, Received and Dialled I have extensions.conf set up so that when I receive a call at home it dials the SNOM 190 at home and also Firefly on my computer at work. If I answer the call on Firefly the call, obviously, shows up as a missed call on the SNOM 190. Is there any SIP technique to change this behavior and notify the SNOM phone that the call has been answered? Ian C ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] fax with asterisk
GFI MailSecurity's HTML threat engine found HTML scripts in this email and has disabled them. From couple of weeks i am working on asterisk fax but was not successful. I am able to receive only half fax documents and its ending with blurred lines. I have tried with almost all the versions of spandsp 0.0.2pre4,pre10 etc but of no use. I have digium card to which i have plugged in 2 pstn lines and i have removed echocancellation too(zapata.conf) and also enabled g7111 alawUlaw codecs. When ever i receive a fax it will come upto some extension(half) and ends.I even checked the pstn lines but its working fine. Please look into it and help me to get rid off. BELOW ARE THE LOGS I HAVE GOT : = Spawn extension (zapincoming, fax, 0) exited non-zero on 'Zap/2-1' Changed from phase 0 to 1 Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down 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 Slow carrier down Slow carrier up Slow carrier down Slow carrier up Slow carrier down Slow carrier up TSI: 43 35 32 30 39 35 33 33 32 20 20 20 20 20 20 20 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: XX DCS: 83 00 c6 f0 80 80 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Get at 9600 Changed from phase 3 to 5 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1699.76 (66) Training error 15.399144 Training succeeded (constellation mismatch 10.791423) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 2 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Fast carrier down Fast carrier up Coarse carrier frequency 1699.81 (66) Training error 3.356434 Training succeeded (constellation mismatch 2.404210) Fast carrier trained Feb 11 19:27:21 NOTICE[20570]: chan_sip.c:7531 handle_request: Fast carrier down Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 955 (got 1906, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 962 (got 2596, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 963 (got 1731, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 965 (got 2176, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 967 (got 1730, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 969 (got 2657, expected 1728). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 971 (got 1723, expected 1728). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 972 (got 1723, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 973 (got 1729, expected 1728). Fax3Decode2D: (FakeInput): Uncompressed data (not supported) at scanline 974 (x 1340). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 974 (got 1340, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 975 (got 1741, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 976 (got 1729, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 979 (got 1795, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 980 (got 1795, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 981 (got 2032, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 982 (got 2699, expected 1728). Fax3Decode2D: (FakeInput): Uncompressed data (not supported) at scanline 983 (x 262). Fax3Decode2D: Warning, (FakeInput): Premature EOL at scanline 983 (got 262, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 984 (got 2026, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 986 (got 1731, expected 1728). Fax3Decode2D: Warning, (FakeInput): Line length mismatch at scanline 987 (got 1730, expected 1728). Fax3Decode2D: Warning,
Re: [Asterisk-Users] Why echo occurs
quote who=Steve Underwood Wrong. Look at any cellular phone or IP phone. They all have echo cancellers. If you switch these cancellers off the results are generally bad. What they need to remove is the acoustic spill from the earpiece to the mike. This can be a surprisingly strong signal. While acoustic spill can be an issue, I do not believe it is the primary source of 90% of the echo experienced. I do not know of any IP phone that contains an echo canceler other than speaker phones. Find a situation where you think the echo is acoustic spill, then try it with a hands free head set. If you notice, the echo is a repeat once type of echo. Not the fading echo of a loop, that acoustic spill would cause. All the echo that I have been talking about, you hear yourself once, just delayed. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
Robert Hajime Lanning wrote: quote who=Steve Underwood Wrong. Look at any cellular phone or IP phone. They all have echo cancellers. If you switch these cancellers off the results are generally bad. What they need to remove is the acoustic spill from the earpiece to the mike. This can be a surprisingly strong signal. While acoustic spill can be an issue, I do not believe it is the primary source of 90% of the echo experienced. I do not know of any IP phone that contains an echo canceler other than speaker phones. Can you show me an ad for an IP phone which doesn't say it includes an echo canceller? A real phone, I mean. Not some thrown together half baked softphone, many of which do a very poor job. Find a situation where you think the echo is acoustic spill, then try it with a hands free head set. Sounds like you haven't worked with this very much. If you notice, the echo is a repeat once type of echo. Not the fading echo of a loop, that acoustic spill would cause. Who introduced a loop into the discussion? All the echo that I have been talking about, you hear yourself once, just delayed. Yep. That's the way they usually are. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Meetme
Nitesh Divecha wrote: Hey All, Just finished installing Asterisk and configured all the necessary parameters to start. I cant seem to find the Meetme application in my asterisk directory. I downloaded asterisk from CVS and installed it and all my Snom phones are working and voicemail too. I am getting error: - Feb 11 17:10:19 WARNING[13042]: pbx.c:1280 pbx_extension_helper: No application 'Meetme' for extension (sip, 5557, 1) == Spawn extension (sip, 5557, 1) exited non-zero on 'SIP/phone1-f88d' Do I need zaptel to be installed? http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe Please observe * The MeetMe application needs a timer to work. There are different ways to get the timer to work, but it won't work by default if you haven't got a Digium Zaptel hardware interface card installed. At this time only zaptel devices may be used. If you do not have a Zaptel device see the ztdummy instructions for timing. Your problem could be different but that answers you Question Exactly. Hope that helps. David Any help will be appreciated. Nitesh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Possible to use CAPI PBX as interface to analog phone?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello everyone, I've got an old AVM all-in-one ISDN box lying around. Currently, it's attached to my server via USB and CAPI4Linux and additionally has an analog DECT phone attached to one of its TAE ports. As I'm planning to switch to VOIP via cable internet, I'm now thinking about the following setup: [Internet] - Cablemodem - Computer w/ Asterisk - AVM box - Phone Two questions remain: 1. Is this possible with Asterisk? The AVM driver supports CAPI 2.0. 2. If not, is it at least theoretically possible to interface with the analog phone through the PBX from Linux? In that case, I would consider writing the necessary software myself. Thanks in advance for your answers, Yours, Florian - -- Preserve wildlife - pickle a squirrel today! -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) iD8DBQFCDdhr7CzyshGvatgRAmA9AKDNvfDqrY0WGODWjd3kfpHfdXXGDgCfefzn du5DeURNLLYhcmH9RZWOlnE= =2pYZ -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
quote who=Rich Adamson The sidetone is 'always' generated within analog and digital phones. It never comes from any source outside the phone. In analog phones, it derived from the hybrid within the phone. On digital phones, its basically firmware. I never said that sidetone was generated outside the phone. The hybrid is the conversion from the dual channel (4 wire, transmit/receive) to the single channel (2 wire, the POTS line). The audio injection point that I was talking about in my previous email, is the location of the hybrids. The hybrid is supposed to automaticaly cancel echo, but it takes precise impedance matching to pull it off. In an analog phone, the sidetone is a side-effect of the hybrid. In a digital phone, the sidetone is on purpose. The conversion from four-wire (analog or digital) to two-wire requires the use of a hybrid (physical component in analog phones, mostly firmware in digital phones). The hybrid is an analog device. When I am talking digital, I am talking about technology like ISDN. In a single bearer channel, I get 56Kbps out and 56Kbps in. I do not see an echo of the output on the input. (This would cause massive issues when used as a data call.) The echo comes when and if I hit a conversion to analog then hit a hybrid. If the conversation is happening purely digital end to end, then you will not get echo. Just like IP to IP. The 'inefficiencies' of that hybrid is the source of echo, regardless of where they happen to be in the end-to-end communications path. Since it is impossible to know what each telephone company or long distance carrier has engineered, its not possible to guess at where hybrids might exist in that path. It is fair to say the number of hybrids is very small now compared to twenty years ago, but they do exist at least at both ends of a communications path. This is true, as long as the path has an analog 2-wire leg. Though where the ends are that the hybrid is located could be lopsided. Say I have a PRI into the PSTN. I call a friend who has POTS service. Now days, the path will be digital from my PRI all the way to my friend's central office. At that point it gets split off the trunk, converted to analog, passed through a hybrid, and placed on the wire pair to my friend's house. Then, through the hybrid in his phone. So, the echo I hear is from the hybrid in the central office and the echo my friend hears is from the hybrid in his phone, which is so close to him, that it becomes sidetone. The previous paragraph is based on where I live (Silicon Valley), the location of the central office hybrid maybe different, depending on your local infrastructure. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
quote who=Steve Underwood Can you show me an ad for an IP phone which doesn't say it includes an echo canceller? A real phone, I mean. Not some thrown together half baked softphone, many of which do a very poor job. I haven't once talked about soft phones. I don't use them. I am talking about hardphones that talk SIP. Take the grandstream phones. Put them back to back, and I gaurentee you will never hear echo, unless you are in the same room. Then you can put the handsets together and get all the screech you want. I have not found anywhere that is says it has an echo canceler. Who introduced a loop into the discussion? I did. Because acoustic spill would most likely cause a loop. Why do I get the feeling you are trolling? You are the only one that brought up acoustic spill. Which, by the way, is usualy controled by directional mics and adjusted gains. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] how to pop up called number details using phpscripts in agi scripts
On 00:07, Fri 11 Feb 05, Robert Rozman wrote: Hi, Covide looks interesting. Is this a killer combination of groupware and Asterisk I was looking for ? Is it open source ? Do you have any more english info ? Thanks in advance, regards, Rob. Hi, We believe it's the killer app for all groupware needs etc. The english website is under construction, since we'll be at the cebit in Hannover so we need to have english and german version. For now, you can look at the preview website: www2.covide.net And yes, Covide is GPL software. Thats why i could post the tarball :))) Michiel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff
On 14:10, Fri 11 Feb 05, Remco Barende wrote: Hi list! I'm currently using a HFC-S card for my ISDN BRI line with bristuff. The instability is driving me crazy however. I'm having continuous problems where inbound calls will not work after some time of operation (the number then appears as not in use to the caller) or also outbound calls do not work. The solution is to unload the modules, stop asterisk, re-load the modules and start asterisk again. The machine (Athlon64) already hung several times when unloading the modules (I guess the same bug/problem is is reported for SMP boxes). This problem occurs every single day and giving me really grey hairs. If I ditch the HFC-S card and replace it with another card that will work with mISDN or chan_capi will this solve my problems? Thanks for any hints / tips! Remco Hi, We had the same trouble. It made me trash the HFC-S cards and now we are running on 2 Fritz! cards on the default Debian asterisk install. Even the chan_capi is included in Debian, and it works great. Dont test the i4l drivers tho, they will give you the same trouble as the zaphfc driver. For the first time in 3 months we now have our * box up and running without any issues for more than a day. So yes, from my point of view, installing chan_capi will solve your issues as it solved mine. Michiel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
Robert Hajime Lanning wrote: quote who=Steve Underwood Can you show me an ad for an IP phone which doesn't say it includes an echo canceller? A real phone, I mean. Not some thrown together half baked softphone, many of which do a very poor job. I haven't once talked about soft phones. I don't use them. I am talking about hardphones that talk SIP. Take the grandstream phones. Put them back to back, and I gaurentee you will never hear echo, unless you are in the same room. Then you can put the handsets together and get all the screech you want. I have not found anywhere that is says it has an echo canceler. Maybe http://www.grandstream.com.cn/BT100_Spec_cn.pdf is a bit confusing if you can't read Chinese, but I think G.168 should be easy to identify :-) Who introduced a loop into the discussion? I did. Because acoustic spill would most likely cause a loop. Why do I get the feeling you are trolling? You are the only one that brought up acoustic spill. Which, by the way, is usualy controled by directional mics and adjusted gains. Why do I get the feeling you haven't a clue what you are talking about? :-) Acoustic spill gives basically the same effects as hybrid echo, except acoustic spill tends to be more variable over time. Hybrid echo also bounces back and forth when both ends are causing echo, but the first echo is so much stronger than the subsequent ones that you tend not to notice them. I have worked on echo cancellation, and I know the acoustic spill issue is serious. In early GSM phones it was often easy to fool the canceler, and GSM to GSM calls would suffer really awful echo. They seem to have improved the cancelers a lot in the last few years, and its rare to get this problem today. This is a broad issue. Echo cancelers have generally improved a lot. The latest version of G.168 is a very different document from the early versions, and incorporates tests for a lot of the problem issues found in earlier canceler designs. How do you control acoustic spill within a phone through the use of directional microphones? Adjusting gains mitigates the issue a bit, but is hardly a solution. These are just bodges, not solutions. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as B2BUA - New Application!!!
Hello all! It's my try to make b2bua from asterisk. It's patched asterisk and some AGI script for it. What it support? Full vovida's b2bua radius emulation, radius failover, LCR, Call failover, Codec based routing, Session-Timeout and much other things that can be useful. Any suggestions and critics welcome! http://b2bua.berlios.de Best regards, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Initializing two ISDN cards in isdn4linux
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello! After LOTS of research on this list and internet in general I managed to get an old Teles PCI card working with Asterisk throught ISDN4Linux. No echos, no delays, simply perfect -- electronic poetry ! :) eheheheheh I just didn't get it to work with CAPI and chan_capi but, since isdn4linux is doing such a good job, I'll kept it. However, what I really wanted to do, was to connect TWO of those cards (that I already have and already tested one at a time) working with Asterisk. (that's because I have two independent BRI lines) Asterisk isn't the problem. The problem lies with the card initialization in Debian and isdn4linux. For the single card I was using with isdntool for initialization, wich works fine but has no support for two cards. Can anyone tell me exactly how to initialize the ISDN system manually ??? It all starts with modprobe -v hisax type=21,21 (loading hisax and telling it that we'll use two teles pci cards) and then ? what else ??? Thanks in advance Miguel Gonçalves -BEGIN PGP SIGNATURE- Version: PGPfreeware 7.0.3 for non-commercial use http://www.pgp.com iQA/AwUBQg30kO+AwW6HpSS6EQLHFgCg19JJkyZ9KmjpF6Lp0VKuMOAYvAUAn0lz VckU3wyjqUK/svPNQRJQeA41 =wi67 -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@home .05 release questions on setup.
I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: samedi 12 février 2005 06:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. 1. There is a help file you can run from the Linux command line help-aah. This will tell you how to change the passwords. On a clean install it tells you this in the motd. 2. Not sure about this second one. I made some big changes in asterisk for this release. It now runs as asterisk not as root and it uses amportal to start not the startup files in /etc/init.d I think only a clean install will fix this. 3. A lot of changes in FOP too the config files are in a different place could cause this problem. Sorry about all the changes. As we get closer to a 1.0 release of [EMAIL PROTECTED] a lot of this will stabilize. --- Ariel Batista [EMAIL PROTECTED] wrote: Hello, Great job on the [EMAIL PROTECTED] project. Looks great this new version is really nicer looking. But I have a few questions. 1) For the new web access http://localIP/maint how and where do I change the password. 2) Since I don't use the Amp section for setup the .conf files I use my own. How do I get the asterisk server running status up. I have it running and works but shows up as not running on the web page. 3) I upgraded my system from the older .04 by downloading the new tar and running your script. Then I copied my .conf files back and rebooted. I had already changed my password and logins names before this. Asterisk is up and running without any issue's. But the Flash Operator panel comes up flashing and I can't seem to get it to work. I feel you have done a great job and I would like to thank you for your setup to us. I will be sending you a donation soon. I am at a small self employed computer consultant that has limited funds at present. This is one of the best setups for Asterisk that I have seen. I feel your name does not do it right due to it can be used for SOHO's and other setups. It's great keep up the good work. You actually make AMP work. P.S. one more question do you have an area in the freenode for chat? If you don't I would love to help out in it. Something like Asterisk-athome would be nice. Ariel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Initializing two ISDN cards in isdn4linux
On 12:20, Sat 12 Feb 05, JunkMail wrote: It all starts with modprobe -v hisax type=21,21 (loading hisax and telling it that we'll use two teles pci cards) and then ? what else ??? try adding protocol=2,2 (for euroisdn, replace with your type) -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 3000 configuration
hello try: exten = 8908,1,Dial(h323/8908,20,Ttr) ! harry --- Scott Henderson [EMAIL PROTECTED] a écrit : I am trying to add a Polycom IP 3000 to our Asterisk system and am not getting anywhere. h323.conf [8908] type=friend host=192.168.104.25 secret=polycom context=crv-default callerid=Conference Room Polycom extensions.conf exten = 8908,1,Dial(h323/polycom,20,Ttr) ; Polycom exten = 8908,2,Hangup I have tried setting the Asterisk system as both gatekeeper and gateway in the polycom config. To date nothing seems to work and Polycom is now on a week return a support call to the reseller that sold us the unit. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Initializing two ISDN cards in isdn4linux
Sure ! But that's not the complete initialization of the isdn system. With modprobe -v hisax type=21,21 protocol=2,2 ALONE, not even the first card answers calls... I just wonder what else does isdntool does to initialize the isdn system... Thank you for your reply. M.G. - Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 12:41 PM Subject: Re: [Asterisk-Users] Initializing two ISDN cards in isdn4linux On 12:20, Sat 12 Feb 05, JunkMail wrote: It all starts with modprobe -v hisax type=21,21 (loading hisax and telling it that we'll use two teles pci cards) and then ? what else ??? try adding protocol=2,2 (for euroisdn, replace with your type) -- Michiel van Baak ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Initializing two ISDN cards in isdn4linux
On 13:05, Sat 12 Feb 05, JunkMail wrote: Sure ! But that's not the complete initialization of the isdn system. With modprobe -v hisax type=21,21 protocol=2,2 ALONE, not even the first card answers calls... I just wonder what else does isdntool does to initialize the isdn system... Thank you for your reply. M.G. Weird. That's all I did to get my two Fritz! cards working. Only did the modprobe. Are you sure your * modem.conf is setup correctly ? I remember I installed isdnutils package, but only to be able to see calls in /var/log/isdn/isdnlog Everytime I started the isdnutils it complained about no configs. So my guess is you dont need to fiddle around with isdntool. My experience is limited to pci devices tho, dont know if you have ISA or pci. Good luck. -- Michiel van Baak http://lunteren.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D Two of the most famous products of Berkeley are LSD and BSD. I don't think that this is a coincidence. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound calls on a busy Zap/1: BUSY vs. CHANUNAVAIL
Hi,
I'm using the macro below in extensions.conf for most of my outbound
calls. One issue with my current configuration is that when I make an
outbound call it doesn't properly detect that my PSTN line (Zap/1) is
busy with another call and then overflow to my outbound IAX
connections. I think the root cause is that DIALSTATUS gets reported
as BUSY. The debug output is below. My desired result would be for
DIALSTATUS to get set to CHANUNAVAIL so it would then try any other
trunks that I have configured. Further, I don't want simply change
the logic to try the IAX on a DIALSTATUS=BUSY because then a truely
busy destination number would get re-dialed three times.
Thanks for any help,
John
[macro-dial-pstn-iax]
;
exten = s,1,SetGlobalVar(FOUNDME=ANSWER)
exten = s,2,Dial(${PSTN}/${ARG1},${ARG2})
exten = s,3,SetGlobalVar(FOUNDME=${DIALSTATUS})
exten = s,4,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?5:19)
;
exten = s,5,GotoIf($[${LEN(${ARG1})} = 10]?7:6)
exten = s,6,GotoIf($[${LEN(${ARG1})} = 7]?9:11)
exten = s,7,SetVar(NumToDial=1${ARG1})
exten = s,8,Goto(s,12)
exten = s,9,SetVar(NumToDial=1908${ARG1})
exten = s,10,Goto(s,12)
exten = s,11,SetVar(NumToDial=${ARG1})
;
exten = s,12,SetGlobalVar(FOUNDME=ANSWER)
exten = s,13,Dial(${IAXCO1}/${NumToDial},${ARG2}) ; try server 1
exten = s,14,SetGlobalVar(FOUNDME=${DIALSTATUS})
exten = s,15,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?16:19)
;
exten = s,16,SetGlobalVar(FOUNDME=ANSWER)
exten = s,17,Dial(${IAXCO2}/${NumToDial},${ARG2}) ; try server 2
exten = s,18,SetGlobalVar(FOUNDME=${DIALSTATUS})
exten = s,19,Goto(s-${DIALSTATUS},1)
;
; returns here if busy on PSTN
;
exten = s,103,SetGlobalVar(FOUNDME=BUSY)
exten = s,104,Goto(s-BUSY,1)
;
; returns here if busy on IAXCO1
;
exten = s,114,SetGlobalVar(FOUNDME=BUSY)
exten = s,115,Goto(s-BUSY,1)
;
; returns here if busy on IAXCO2
;
exten = s,118,SetGlobalVar(FOUNDME=BUSY)
exten = s,119,Goto(s-BUSY,1)
;
exten = s-BUSY,1,BackGround(the-party-you-are-calling)
exten = s-BUSY,2,BackGround(is-curntly-busy)
exten = s-BUSY,3,SetGlobalVar(FOUNDME=BUSY)
exten = s-BUSY,4,Goto(s-CLEANEXIT,1)
;
exten = s-CANCEL,1,BackGround(canceled)
exten = s-CANCEL,2,SetGlobalVar(FOUNDME=CANCEL)
exten = s-CANCEL,3,Goto(s-CLEANEXIT,1)
;
exten = s-CHANUNAVAIL,1,BackGround(channel)
exten = s-CHANUNAVAIL,2,BackGround(is-curntly-unavail)
exten = s-CHANUNAVAIL,3,SetGlobalVar(FOUNDME=CHANUNAVAIL)
exten = s-CHANUNAVAIL,4,Goto(s-CLEANEXIT,1)
;
exten = s-NOANSWER,1,BackGround(nbdy-avail-to-take-call)
exten = s-NOANSWER,2,SetGlobalVar(FOUNDME=NOANSWER)
exten = s-NOANSWER,3,Goto(s-CLEANEXIT,1)
;
exten = s-ANSWER,1,SetGlobalVar(FOUNDME=ANSWER)
exten = s-ANSWER,2,Goto(s-CLEANEXIT,3)
;
exten = s-.,1,BackGround(something-terribly-wrong)
exten = s-.,2,SetGlobalVar(FOUNDME=ERROR)
exten = s-.,3,Goto(s-CLEANEXIT,1)
;
exten = s-CLEANEXIT,1,GotoIf($[${ARG3} = RT]?3:2)
exten = s-CLEANEXIT,2,Hangup
exten = s-CLEANEXIT,3,NoOp
zapata.conf
[channels]
language=en
context=inbound-analog
signalling=fxs_ks
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
echocancel=64
txgain=0.0
rxgain=0.0
channel = 1
Asterisk CVS-HEAD-01/30/05-15:35:12, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer [EMAIL PROTECTED]
=
-- Executing Macro(SIP/22-6189, dial-pstn-iax|537|70|HR)
in new stack
-- Executing SetGlobalVar(SIP/22-6189, FOUNDME=ANSWER) in new stack
-- Setting global variable 'FOUNDME' to 'ANSWER'
-- Executing Dial(SIP/22-6189, Zap/1/537|70) in new stack
Feb 12 08:22:22 NOTICE[18856]: app_dial.c:884 dial_exec_full: Unable
to create channel of type 'Zap' (cause 0)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing SetGlobalVar(SIP/22-6189, FOUNDME=BUSY) in new stack
-- Setting global variable 'FOUNDME' to 'BUSY'
-- Executing Goto(SIP/22-6189, s-BUSY|1) in new stack
-- Goto (macro-dial-pstn-iax,s-BUSY,1)
-- Executing BackGround(SIP/22-6189,
the-party-you-are-calling) in new stack
-- Playing 'the-party-you-are-calling' (language 'en')
-- Executing BackGround(SIP/22-6189, is-curntly-busy) in new stack
-- Playing 'is-curntly-busy' (language 'en')
-- Executing SetGlobalVar(SIP/22-6189, FOUNDME=BUSY) in new stack
-- Setting global variable 'FOUNDME' to 'BUSY'
-- Executing Goto(SIP/22-6189, s-CLEANEXIT|1) in new stack
-- Goto (macro-dial-pstn-iax,s-CLEANEXIT,1)
-- Executing GotoIf(SIP/22-6189, 0?3:2) in new stack
-- Goto (macro-dial-pstn-iax,s-CLEANEXIT,2)
-- Executing Hangup(SIP/22-6189, ) in new stack
== Spawn extension (macro-dial-pstn-iax, s-CLEANEXIT, 2) exited
non-zero on 'SIP/22-6189' in macro 'dial-pstn-iax'
== Spawn extension (intern, 537, 1) exited non-zero on 'SIP/22-6189'
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[Asterisk-Users] Delay on zap channel
I'm using Asterisk on a system described as below: Asterisk version 1.0.5 on Linux Debian version 3.0 (unstable) with kernel version 2.6.10 (hardware: PC, i386 class). My Asterisk works with a phone card Digium TDM400P, where 2 FXS and 2 FX0 modules are provided. It works, but I notice an annoying delay on incoming calls from analog phones: system answers only after about 2 seconds, even if the first command is answer(). If I request an external line and therefore I dial 0, I have to wait 2 seconds to listen to the external line tones and to be allowed to dial an external telephone number. Asterisk should answer and give external line free tones immediately, instead. This issue happens on incoming calls from local internal phones, that is from classic phones connected to FXS modules. How can this be fixed? Is it possible to have Asterisk answering (almost) immediately when it receives a dial tone for the first digit? Note that my Linux box is not overloaded: there is no possible latency due to system performance. More, I cannot set on hold my active calls by using the R button. The only way to put on hold an active call is to hang up for a very short time. How can I configure Asterisk to understand my phone's R button, so that it puts on hold current active voice call? Thank you in advance, Stefano Arata Italy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MGCP, Asterisk Cisco VG200
Hello: I want to receive calls from my SIP proxy and re-route them to one of the analog lines on my Cisco VG200 ia MGCP and Asterisk. Inbound SIP calls will arrive with the five digit called number preficed with an m by the proxy. I'd like to them match these calls against a rule like exten = _mX,1,Dial(MGCP/aaln/[EMAIL PROTECTED]) . This however results in an error. My mgcp.conf looks like: ; mgcp audit endpoint aaln/[EMAIL PROTECTED] (vg200) [general] port = 2427 bindaddr = 128.100.10.10 (asterisk server) [vg200] host = 128.100.10.11 canreinvite = no line = aaln/2 line = aaln/1 Can anyone explain how this should work? Thanks,Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@home .05 release questions on setup.
Nope, works fine. Several people have already downloaded and installed it yesterday. Try again. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Daniel Eboa Sent: Saturday, February 12, 2005 7:32 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: samedi 12 février 2005 06:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. 1. There is a help file you can run from the Linux command line help-aah. This will tell you how to change the passwords. On a clean install it tells you this in the motd. 2. Not sure about this second one. I made some big changes in asterisk for this release. It now runs as asterisk not as root and it uses amportal to start not the startup files in /etc/init.d I think only a clean install will fix this. 3. A lot of changes in FOP too the config files are in a different place could cause this problem. Sorry about all the changes. As we get closer to a 1.0 release of [EMAIL PROTECTED] a lot of this will stabilize. --- Ariel Batista [EMAIL PROTECTED] wrote: Hello, Great job on the [EMAIL PROTECTED] project. Looks great this new version is really nicer looking. But I have a few questions. 1) For the new web access http://localIP/maint how and where do I change the password. 2) Since I don't use the Amp section for setup the .conf files I use my own. How do I get the asterisk server running status up. I have it running and works but shows up as not running on the web page. 3) I upgraded my system from the older .04 by downloading the new tar and running your script. Then I copied my .conf files back and rebooted. I had already changed my password and logins names before this. Asterisk is up and running without any issue's. But the Flash Operator panel comes up flashing and I can't seem to get it to work. I feel you have done a great job and I would like to thank you for your setup to us. I will be sending you a donation soon. I am at a small self employed computer consultant that has limited funds at present. This is one of the best setups for Asterisk that I have seen. I feel your name does not do it right due to it can be used for SOHO's and other setups. It's great keep up the good work. You actually make AMP work. P.S. one more question do you have an area in the freenode for chat? If you don't I would love to help out in it. Something like Asterisk-athome would be nice. Ariel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] bristuff-0.2.0 RC7 and RC7a
Is anybody familiar with the recent bristuff packages released ? There is only a 3 hour difference in release time between them and the CHANGES files are the same. Also what's strange RC7 has 163K and RC7a has only 87K. Ideas anyone ? Regards, Ivan ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with # Transfer from queue
Ryan Stark wrote:
Hi I'm having trouble # transfering queue calls.
in extensions.conf I have:
[macro-queue]
;
; Places caller in queue
; ${ARG1} - Queue name to place caller in.
; ${ARG2} - Voicemail Extention
; ${ARG3} - Caller ID to Set.
exten = s,1,DBget(temp=nm/on) ; Get Night key, if not existing,goto 102
exten = s,2,Playback(custom/500/10)
exten = s,3,SetCallerID(${ARG3})
exten = s,4,DigitTimeout(0)
exten = s,5,ResponseTimeout(0)
exten = s,6,Queue(${ARG1}|t)
exten = s,7,Voicemail(su${ARG2})
exten = s,102,Voicemail(su${ARG2})
exten = s,107,Voicemail(su${ARG2})
in queues.conf I have:
[mainq]
member = Agent/10
member = Agent/11
member = Agent/12
member = Agent/13
member = Agent/14
member = Agent/15
member = Agent/16
member = Agent/17
member = Agent/18
member = Agent/20
member = Agent/21
retry = 0
timeout = 20
announce-holdtime = yes
joinempty = yes
announce-frequency = 90
reportholdtime = yes
In features.conf I have:
[general]
parkext = 7000
parkpos = 7001-7020
context = parkedcalls
parkingtime = 90
transfer = #
Normal direct calls are transferable but not the ones from the queue,
both parties can hear the DTMF. This same config worked on my old
asterisk box, but when I moved everything over to this version:
CVS-HEAD-01/19/05-15:06:17 I can't transfer from the queues. Is this
something that might be fixed if I update my src tonight and recompile?
Thanks,
-Ryan
depending on which directly the call is traveling the option is 'T or
t, have you tried both?
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[Asterisk-Users] Re: Codec Issue on IAX trunk? (Solved)
Hi Rich - Those type changes to iax.conf require a full stop of asterisk followed by a cold asterisk startup. A restart from the CLI won't cut it. Ahh! That's a very important piece of information! Were you previously doing the CLI restart? I did lots a CLI reloads, and few cold restarts to the ast33 machine, but no cold restarts on the ast551 machine until after business hours (it's a production machine), and then I did a cold restart at the same time I did the recompile. That explains things. Hopefully, I won't have to make any more changes like this, although I guess I could use the restart when convenient command. Now I just have to figure out why a cold restart is needed. Thanks Again, Noah ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Flash Pane - Monitor Parked Calls?
Need help with how to configure for parked calls in the Flash Operator Panel's op_buttons.cfg file ... I've looked on the wiki, google and asternic's site and can't seem to find how to setup op_buttons.cfg to monitor parked calls. For example, if someone parks in 701, I'd like to see that represented on the panel. I've tried a number of things ... this is what I have now and it does not work ... [701] Position=12 Label=Park 701 Extension=701 Context=parkedcalls Icon=1 Any help would be great! Thanks, Bruce [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Problem with # Transfer from queue
*snipped depending on which directly the call is traveling the option is 'T or t, have you tried both? i think i 'directly' need to go find some coffee! (meant direction, sorry) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Speech Recognition
Hi I dont know jack about speech recognition, however since this topic came up anyonw know how spinvox do speech ercognition, in fact its so good it converst the speech to text and sends the voicemail as a SMS, I think a awesome addone to the sms module in asterisk. Iqbal On 2/12/2005, Race Vanderdecken [EMAIL PROTECTED] wrote: Ahem, Being one who has programmed, consulted and argued to points beyond violence about the subjects of your first paragraph, I shall now expound. Expounding begins: I worked on several projects with a company named Intellivoice that did so called voice dialing, voice activated dialing, VAD, as a bread and butter product in the PSTN/T-1 world. The product was good at about 3-5 recognitions so long as they were distinct enough that your well trained dog could understand them as different commands. I was first hand witness to many sales and customer meetings, I rode in the car of the inventor and ate lunch with the VAD developers and beat them often with questions about how they did it and why it did not work. Personally, I have a Mid-Western trained Mid-Atlantic accent, i.e. no accent to speak of, so speech and voice recognition engines like me. I am even tempered and have been working in telecommunications, 22 wpm Morse code, to Tech Plus, to before NETBIOS, SNB, and 256K twisted pair Ethernet on 9DB, through voice and right back into VoIP before it was an acronym. I have been to college to study communications. I have an ear for dialects and can place most people in 100 mile range within their State. I coded the Persona project. I have pushed Sphinx down Festivals throat, and I have worked with Dave. I was working to create voice X/HTML/XML browser before they were committees. I am pushed speech and voice and dictation since I got my hands on a computer. I love speech recognition and generation, period. So, when I say that you are out of your mind if you think you can get VAD or SAD to work across the wire if there is an analog device in the path you should take heed. VAD on cel-phones works now because the reco is in the phone, for the most part. VAD over the analog wires can be done but is of no use to anyone unless they like to scream at the phone from time to time. By the way screaming at voice-reco engines only makes the angry. So angry in fact that they will either repeatedly ask you to please say the name again until you calm down or they will deliberately misdial the number for you. Machines just don't like to be yelled at, ask Woody Allen about the time beat up his television and the elevator incident. If you are Digital from speaker to reco then you have a chance. If you are G.711 all the way you have a chance. And by chance I mean if you use grammar based recognition and have a caller with an IQ greater then the first two digit of their Area code. Zhong's thesis work is interesting and I will state he is on the right track and I enjoin him to continue his work. Neural works are the right path, but he needs to re-read Strousstrup on objects in Tries. But short utterances don't work in communication models; see trying to communicate with teenage son and HDLC. My Expounding ends. Yes, what you want can be done, and it would be easy, no, I say trivial to accomplish. Follow the dynamic context example. Listen for reco, then create a dynamic context, then forward to that dynamic context. Easy peasy. Please contact me, [EMAIL PROTECTED], if you would like to create a project to do this. I would love to see VAD on VoIP come to fruition in my lifetime. Race The Tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Rozman Sent: Saturday, January 29, 2005 12:16 PM To: Jon Radon; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Speech Recognition Hi, probably I won't be much of help, but I'm also looking for speech recognition solution. But we're actually looking at two problems: - one would be so called voice dialing (similar to celular phones) - one records its own spoken names and speaks them after to call certain person - this problem is much easier to solve. Recently I have found interesting project that could be easily integrated for such functionality (http://www.princeton.edu/~lzhong/DNN.html). I'd like to start doing this but don't know much about Asterisk and its eagi interface to get sound out of it. I guess some with more insight could easily integrate this code. This solution could be probably used for simple 1 word recognition tasks (like speak name for outgoing call, or maybe say sales to get sales department - but as said this is speaker dependent solution. - using speaker independent solutions for other stuff. I guess that Sphinx is at the moment most serious candidate. There is already some work on connecting speech recognition to MH and I'm sure that guys will help with other uses. What would be most desirable from Asterisk community is
RE: [Asterisk-Users] Asterisk@home .05 release questions on setup.
Howdy, Who knows? If I do an md5sum on the 0.5 iso, I get 9d5657b7c833830b8a1fd1f024215d46 asteriskathome-0.5.iso They don't tell you the right md5 that I can tell, so nobody must care if the downloads work. Good day, Ralph On Sat, 2005-02-12 at 13:32 +0100, Daniel Eboa wrote: I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech Recognition
Iqbal wrote: Hi I dont know jack about speech recognition, however since this topic came up anyonw know how spinvox do speech ercognition, in fact its so good it converst the speech to text and sends the voicemail as a SMS, I think a awesome addone to the sms module in asterisk. If it works really well, there is probably a human operator involved. A number of systems that try to look automated actually rely on human operators. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Initializing two ISDN cards in isdn4linux
PROBLEM SOLVED (read the rest) Weird. That's all I did to get my two Fritz! cards working. Only did the modprobe. Er... that's really not enough (in this case, at least). Asterisk will only be happy with the initialization when you can see all four channels listed in the imon utility (from the isdn4k-utils package). At that point you can fool around with /dev/ttyI0...3 in the minicom as described in the http://www.voip-info.org/wiki-Asterisk+ISDN4Linux page and asterisk will be happy. Only modprobe alone won't do that ! Are you sure your * modem.conf is setup correctly ? Yes, that's important. My modem.conf includes this : group=1 msn=41 incomingmsn=41,2***4 device = /dev/ttyI0 device = /dev/ttyI1 device = /dev/ttyI2 device = /dev/ttyI3 I remember I installed isdnutils package, but only to be able to see calls in /var/log/isdn/isdnlog Everytime I started the isdnutils it complained about no configs. So my guess is you dont need to fiddle around with isdntool. I still don't know HOW isdntool does the initialization of the ISDN system, however, after fiddling some more with isdntool I found out that it let's me write my own modprobe command. That's how I solved the problem : -- Ran isdntool -- Selected ISDN-Settings : View / Edit -- Selected Hardware -- Selected Teles PCI -- But instead of accepting the default modprobe prompted, I wrote the correct one. After this, isdntool did it's thing and I started Asterisk, who was now answering calls on all four channels. My experience is limited to pci devices tho, dont know if you have ISA or pci. Like I said -- two equal Teles PCI cards. One bought in 1996 and other bought in 1998. At a time I was about the throw out both of them; but now I'm tempted to buy a used one from anyone who wants to sell. :) Thank you for the help ! Best Regards Miguel Gonçalves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can agents login be permanent across Asterisk restarts ?
Asterisk wrote: persistentmembers = yes That is only in CVS HEAD, and it does not apply to logins generated via AgentCallbackLogin, it applies to dynamic members added via AddQueueMember. There is, however, some analogous persistence in chan_agent (again CVS HEAD only) that may do what the OP wants as long as they are using Agent channels and not direct channels for their queue members. I haven't tried it though, since I don't use chan_agent. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Codec Issue on IAX trunk? (Solved)
Those type changes to iax.conf require a full stop of asterisk followed by a cold asterisk startup. A restart from the CLI won't cut it. Ahh! That's a very important piece of information! Were you previously doing the CLI restart? I did lots a CLI reloads, and few cold restarts to the ast33 machine, but no cold restarts on the ast551 machine until after business hours (it's a production machine), and then I did a cold restart at the same time I did the recompile. That explains things. Hopefully, I won't have to make any more changes like this, although I guess I could use the restart when convenient command. Now I just have to figure out why a cold restart is needed. I haven't tried to keep track of the code changes involving reloads (or cli restarts for that matter), but having been around * for a fair amount of time and having been caught with making changes that have had no affect, I'll usually play it very safe and simply stop / start asterisk for many different changes. Iax and sip def's in particular. Reloads are fine for lots of things, but experience is the only way to know what's acceptable at this point. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech Recognition
Does anyone know of a speech recognition module (like say yes or no, or numbers) I guess due to the complexity of speech recognition it might just be found in commercial applications or am I wrong like always? What's wrong with the old and non-fancy IVR? Voice recognition menus only piss people off. If you're setting up a call center where you want as many as possible of the customers to ABANDON their calls, go on... roy ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] return code of app in dialplan
Hi, I'll probably kick myself when I read the replies to this... How do I test the return code of an app in the dialplan? I need to test if the app, MYSQL() in this case, returned -1 or 0. It's easy to see after-the-fact in the log output, but I need the result in the dialplan, I just can't find which variable stores the actual return-code. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PLEASE HELP Adit 600 went kaput?
After months of setting up Asterisk. I completed the final testing last night We would go live on Monday. Or so I thought. As I moved the Adit 600 back out of the way, sliding it six inches. I noticed the major light was red as were both T1 and T2 lights ( eventhough only one t1 port is being used ) -- I have no idea what these lights mean or if they were lit before when things were working. Then I go a phone call and theres no dial tone. I have restarted asterisk, check all the connections to and from the CB, diconected, reconnected and repowered the CB, still no dial tone. I realize that it hard to help me with such a problem, however, if anyone can help shed some light on what the problem could be I would greatly appreciate it. Thanks, Richard __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
The sidetone is 'always' generated within analog and digital phones. It never comes from any source outside the phone. In analog phones, it derived from the hybrid within the phone. On digital phones, its basically firmware. I never said that sidetone was generated outside the phone. Your original posting said the sidetone was coming from the distant phone and did not even come close to implying that sidetone is something always engineered into the local phone, regardless of whether its analog or digital. Sidetone is always local phone generated by design. The hybrid is the conversion from the dual channel (4 wire, transmit/receive) to the single channel (2 wire, the POTS line). The audio injection point that I was talking about in my previous email, is the location of the hybrids. The hybrid is supposed to automaticaly cancel echo, but it takes precise impedance matching to pull it off. A 100% perfect hybrid would never generate any feedback or echo. But, to date no one has been successful at designing such a beast. So a better way to say that is imperfections in the hybrid can cause echo as opposed to the hybrid is supposed to automatically cancel echo. There is no such thing as an echo canceller in a hybrid. In an analog phone, the sidetone is a side-effect of the hybrid. Not true at all. Sidetone _is_ designed into the hybrid in analog phones on purpose and has been for for at least 30 years. In a digital phone, the sidetone is on purpose. Just exactly like the analog phones. The conversion from four-wire (analog or digital) to two-wire requires the use of a hybrid (physical component in analog phones, mostly firmware in digital phones). The hybrid is an analog device. Not true. Better take a look at the Silicon Labs chip sets that are used in the digium TDM card (as one example). The hybrid is 100% digital. When I am talking digital, I am talking about technology like ISDN. In a single bearer channel, I get 56Kbps out and 56Kbps in. I do not see an echo of the output on the input. (This would cause massive issues when used as a data call.) The echo comes when and if I hit a conversion to analog then hit a hybrid. If the conversation is happening purely digital end to end, then you will not get echo. Just like IP to IP. Say I have a PRI into the PSTN. I call a friend who has POTS service. Now days, the path will be digital from my PRI all the way to my friend's central office. At that point it gets split off the trunk, converted to analog, passed through a hybrid, and placed on the wire pair to my friend's house. Then, through the hybrid in his phone. So, the echo I hear is from the hybrid in the central office and the echo my friend hears is from the hybrid in his phone, which is so close to him, that it becomes sidetone. In most current-day CO hardware (line cards included), the echo that we hear from the distant end is almost always associated with the phone hybrid, not the CO hardware. (Cellular and other types of non- telco stuff can be very different, and shouldn't be used as the basis for evaluting echo.) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] return code of app in dialplan
Gary Reuter wrote: I need to test if the app, MYSQL() in this case, returned -1 or 0. You can't. If it returns -1, execution stops and if there is a channel active, it is hung up. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 7912G: Takes the same firmware as 7940/60?
Does anyone know if the 7912G (which the wiki says can do either sccp or sip) uses the 7940/60 sip firmware? I ask this because the only firmware I can seem to find on TAC for the 7912G is sccp, no sip...if it takes it's own firmware and doesn't use 7940/60 firmware, can someone point me to the right location for it? Thanks, Marty Mastera M3 Resources [EMAIL PROTECTED] Phone: 303.680.1283 x200 FAX: 206.666.1786 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Speech Recognition
On Feb 12, 2005, at 17:58, Roy Sigurd Karlsbakk wrote: Does anyone know of a speech recognition module (like say yes or no, or numbers) I guess due to the complexity of speech recognition it might just be found in commercial applications or am I wrong like always? What's wrong with the old and non-fancy IVR? Voice recognition menus only piss people off. If you're setting up a call center where you want as many as possible of the customers to ABANDON their calls, go on... How true that is... faced with customer-unfriendly service like that (especially when they don't offer a choice to get a human at all) I start hitting keys like 0 or # or * until something happens... jens ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.
Daniel Eboa wrote: I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. Not as of approx 5:)) PM yesterday. I downloaded, burned, and in last stage of the install ( compiling * ) right now. I don't remeber if there was a md5sum for the iso, but a binary error in hte download or bad hardware ( cd burner ) are the twom main causes of this problem. Try another download. Rod -- --- [This E-mail scanned for viruses by Declude Virus] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] French CallerID
I have two phones which are callerid num and name capable connected to asterisk 1.0.3. Both of these phones will display number and name of caller when available and when connected to the French phone company (France Télécom). However, one of these phones will not show it on asterisk connected via a TDM400 FXS port where the other one does. The one not working is a new phone, a Siemens C200 DECT. Does this matter to anyone, IOW should I submit a report? I'm not thrilled with the fact that CID doesn't work but it isn't mission critical. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC vs AreskiCC
Hello All, Could someone please give me your impressions as to which Calling Card application is better? I'm trying to decide on the one that I will implement while I'm learning about my new Asterisk server that I have just installed on Fedora 2. Thanks in advance, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax.conf config and iax based clients
correct your dialplan. something like this [from-iax] exten = 105,1,Dial(IAX2/QIax1,20) exten = 106,1,Dial(IAX2/QIax2,20) exten = 107,1,Dial(IAX2/QIax3,20) hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ASTCC vs AreskiCC
Astcc is mysql driven w/ perl based web ui Areski is same concept based on postgres w/ a php frontend also tied in w/ Areski other scripts for reports and such ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
quote who=Steve Underwood Maybe http://www.grandstream.com.cn/BT100_Spec_cn.pdf is a bit confusing if you can't read Chinese, but I think G.168 should be easy to identify :-) ok, I did miss that. Then again, the grandstream does have a speaker phone. I guess the problem is that I don't know of a SIP hardphone that doesn't have a speaker phone. Acoustic spill gives basically the same effects as hybrid echo, except acoustic spill tends to be more variable over time. Hybrid echo also bounces back and forth when both ends are causing echo, but the first echo is so much stronger than the subsequent ones that you tend not to notice them. I have worked on echo cancellation, and I know the acoustic spill issue is serious. In early GSM phones it was often easy to fool the canceler, and GSM to GSM calls would suffer really awful echo. They seem to have improved the cancelers a lot in the last few years, and its rare to get this problem today. This is a broad issue. Echo cancelers have generally improved a lot. The latest version of G.168 is a very different document from the early versions, and incorporates tests for a lot of the problem issues found in earlier canceler designs. I would expect it to be a problem in the GSM (cell) phones. They are too small to get proper acoustic separation. I am talking about the phones that are physically designed the same as analog phones. Why do we not hear this echo in the analog device? But, we do when it is digital. This type of echo would always be far end, as the near end would always be seen as sidetone. How do you control acoustic spill within a phone through the use of directional microphones? Adjusting gains mitigates the issue a bit, but is hardly a solution. These are just bodges, not solutions. You can say the same about echo cancelers. They patch the symptom, not the cause. -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Finding exact build version
What's the recommended way to show my exact build of Asterisk -- down
to the minor-minor version number?
I ask because I am setting up a small testbed and need to keep myself
straight and would prefer something more authoritative than a post-it
note and my addled memory.
If I do Asterisk -V, I seem to only get the point release and one
decimal beyond, e.g.: 1.0.5 shows as 1.0{compiledate}
Thanks,
/rg
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[Asterisk-Users] Compiling asterisk
While trying to compile asterisk, I get the following errors - -- bison ast_expr.y --name-prefix=ast_yy -o ast_expr.c ast_expr.y:118: unrecognized: %locations ast_expr.y:118: Skipping to next % ast_expr.y:149: invalid @-construct ast_expr.y:149: $. is invalid ... ... ... - Any help in solving this would be greatly appreciated. thanks BK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
quote who=Rich Adamson Your original posting said the sidetone was coming from the distant phone and did not even come close to implying that sidetone is something always engineered into the local phone, regardless of whether its analog or digital. Sidetone is always local phone generated by design. I went back and read my original post. I did not say that the sidetone was coming from the far end, but I was completely unclear in what I was saying. We have echo as A hears his own voice, but the timing makes it perceived as sidetone. Should have been more like: (using terms in the original email) A hears his own voice coming from his mic's injection point, which is close enough to his speaker to make the delay short enough, so as it is perceived as sidetone. I wonder if sidetone was in the original spec when creating the hybrid, or was it added as a feature when they could not get rid of it. Did they get a 100% working hybrid, then say hey, I can't hear myself!? A 100% perfect hybrid would never generate any feedback or echo. But, to date no one has been successful at designing such a beast. So a better way to say that is imperfections in the hybrid can cause echo as opposed to the hybrid is supposed to automatically cancel echo. There is no such thing as an echo canceller in a hybrid. True. I used the wrong wording. Not true at all. Sidetone _is_ designed into the hybrid in analog phones on purpose and has been for for at least 30 years. My guess on that is above. Not true. Better take a look at the Silicon Labs chip sets that are used in the digium TDM card (as one example). The hybrid is 100% digital. I probably shouldn't have made a blanket statement. There really isn't anything we can't simulate in digital, anymore. And I doubt that sidetone is purposely put into the TDM cards. It just comes down to that we can't get rid of it. (hybrid imperfections) -- END OF LINE -MCP ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.
Daniel Eboa wrote: I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. I just burned the CD and it installed just fine on my test box. Regards. Daniel. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: samedi 12 février 2005 06:00 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. 1. There is a help file you can run from the Linux command line help-aah. This will tell you how to change the passwords. On a clean install it tells you this in the motd. 2. Not sure about this second one. I made some big changes in asterisk for this release. It now runs as asterisk not as root and it uses amportal to start not the startup files in /etc/init.d I think only a clean install will fix this. 3. A lot of changes in FOP too the config files are in a different place could cause this problem. Sorry about all the changes. As we get closer to a 1.0 release of [EMAIL PROTECTED] a lot of this will stabilize. --- Ariel Batista [EMAIL PROTECTED] wrote: Hello, Great job on the [EMAIL PROTECTED] project. Looks great this new version is really nicer looking. But I have a few questions. 1) For the new web access http://localIP/maint how and where do I change the password. 2) Since I don't use the Amp section for setup the .conf files I use my own. How do I get the asterisk server running status up. I have it running and works but shows up as not running on the web page. 3) I upgraded my system from the older .04 by downloading the new tar and running your script. Then I copied my .conf files back and rebooted. I had already changed my password and logins names before this. Asterisk is up and running without any issue's. But the Flash Operator panel comes up flashing and I can't seem to get it to work. I feel you have done a great job and I would like to thank you for your setup to us. I will be sending you a donation soon. I am at a small self employed computer consultant that has limited funds at present. This is one of the best setups for Asterisk that I have seen. I feel your name does not do it right due to it can be used for SOHO's and other setups. It's great keep up the good work. You actually make AMP work. P.S. one more question do you have an area in the freenode for chat? If you don't I would love to help out in it. Something like Asterisk-athome would be nice. Ariel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Mobile Wireless IP Phone
Hi! I would like to have feedback on wireless (wifi / 802.11b) IP phone to use with Asterisk PBX. Can you sugest model, The best and also the worst to use. Thanks, eric. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sound Problem
I have been using Asterisk to make phone calls and when i tried to use it today the volume was unexpectedly very low. Changing the volume in the volume control didn't effect it. I believe the problem lies with Asterisk and not the volume control. I appreciate any feedback of where and what to check. __ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff
- Original Message - From: Michiel van Baak [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 11:57 AM Subject: Re: [Asterisk-Users] chan_capi or chan_mISDN vs bristuff On 14:10, Fri 11 Feb 05, Remco Barende wrote: Hi list! I'm currently using a HFC-S card for my ISDN BRI line with bristuff. The instability is driving me crazy however. I'm having continuous problems where inbound calls will not work after some time of operation (the number then appears as not in use to the caller) or also outbound calls do not work. The solution is to unload the modules, stop asterisk, re-load the modules and start asterisk again. The machine (Athlon64) already hung several times when unloading the modules (I guess the same bug/problem is is reported for SMP boxes). This problem occurs every single day and giving me really grey hairs. If I ditch the HFC-S card and replace it with another card that will work with mISDN or chan_capi will this solve my problems? Thanks for any hints / tips! Remco Hi, We had the same trouble. It made me trash the HFC-S cards and now we are running on 2 Fritz! cards on the default Debian asterisk install. Even the chan_capi is included in Debian, and it works great. Dont test the i4l drivers tho, they will give you the same trouble as the zaphfc driver. For the first time in 3 months we now have our * box up and running without any issues for more than a day. So yes, from my point of view, installing chan_capi will solve your issues as it solved mine. Hi, could you give some more info about your setup. How do you get 2 fritz cards working (I thought it works only on 2.4 kernels ) ? What capi drivers do you use ? Thanks, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sound Problem
You need to tell us what type of device you are using to make the phone calls. Are you using a ZAP FXS, a softphone, a sip phone, or an iax phone. Also, how are you terminating the call. Is it via a ZAP FXO device like a t100p, is it another VOIP phone, or is it via a service provider like iax.cc or nufone? Cheers, Jon. On Saturday 12 February 2005 02:20 pm, chawki hammoud wrote: I have been using Asterisk to make phone calls and when i tried to use it today the volume was unexpectedly very low. Changing the volume in the volume control didn't effect it. I believe the problem lies with Asterisk and not the volume control. I appreciate any feedback of where and what to check. __ Do you Yahoo!? Yahoo! Mail - Helps protect you from nasty viruses. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What quad/octo BRI cards are best/stable for EuroISDN and Asterisk ?
Hi, I'm currently deciding on what card to pruchase for octo/quad BRI card to use with Asterisk on EuroISDN lines. I'm aware of at least two options (Junghanns or Beronet), but don't know how stable and well supported they are. Which ones are better supported ? Any experiences? Any advice ? How tos ? What would you buy ? Thanks in advance, Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How stable are cheap HFC-s cards in NT mode ?
Hi, I'd like to use one card to interface with existing ISDN pbx output. How stable are those cards for this ? Where can I find more info how to setup ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] soho fax suggestions?
Need to replace our older soho fax machine with something more current. Would like to run the fax line through *, but haven't been able to make spandsp work correctly with digium TDM04b card. Our fax volume is very low (maybe a few per week), but we have multiple offices in three geographic locations and would like to be able to email the images to the correct location. For planning purposes, is it appropriate to think in terms of purchasing a t38 capable box even if its not supported by * today? (I'm well aware of the bounty and Steve's work.) If now is the time to purchase a t38 capable fax machine, anyone have any suggestions on a low-volume soho-sized box? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.
I've downloaded 2x and burned 2 cds and get an error invalid compressed format (err=2) system halted message both times. It'd be nice to have a MD5 to verify my download is OK. It'd narrow down the problem to either the download or the burn, wouldn't it? - Original Message - From: Roderick A. Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 11:55 AM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. Daniel Eboa wrote: I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. Not as of approx 5:)) PM yesterday. I downloaded, burned, and in last stage of the install ( compiling * ) right now. I don't remeber if there was a md5sum for the iso, but a binary error in hte download or bad hardware ( cd burner ) are the twom main causes of this problem. Try another download. Rod -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] uninstall Asterisk?
Hello All, I followed all of the steps to install Asterisk on my Fedora2 and it worked great. Now I want to uninstall Asterisk because I want to make a fresh install along with some additional modules. I have found that there does not seem to be a make uninstall to go with the make install How to I remove all of the Asterisk files easily? Thanks, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Speech Recognition
Spinvox have a distinct advantage over most telephony applications in that their speech recognition does not have to occur in realtime - it simply records the speech and then processes it afterwards. I strongly suspect since the company always acts very tight-lipped about their technology that it relies heavily on human operators. Think about it if the average translation of voicemail to text message was to take 2 mins, that would cost them about 20p per message if they use minimum wage UK workers. Make that under 3p per message in the more likely scenario that they are using a call-centre in India. Since they charge you 25p per message this is a feasible business model, and one that hasn't got to rely on any bleeding edge technologies. -- Adam Holt Bayham Systems Ltd Web:http://www.bayhamsystems.com/ Email: [EMAIL PROTECTED] Address: No. 1 Farnham Road, Guildford, Surrey, GU2 4RG, United Kingdom Hi I dont know jack about speech recognition, however since this topic came up anyonw know how spinvox do speech ercognition, in fact its so good it converst the speech to text and sends the voicemail as a SMS, I think a awesome addone to the sms module in asterisk. Iqbal ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk@home scary log
The bottom line for those asterisk readers that have actually read this far is to use complex lenthy passwords where possible, and some sort of alerting mechansim when xx number of passwords are guessed incorrectly (such as an account lockout mechanism with alerts as just one of many available choices). I tend to disagree with you regarding the exact length. An alerting mechanism is there, in the logs. Most linux distros have some nice log watchers. However it still requires that someone actually monitors them, as boring as it is. Can anyone recommend a watching tool for this? I know I can write a script myself but if there is a convenient Linux method that is prepackaged, that would be good. Specifically nice would be a mechanism like the one referred to above: some sort of alerting mechansim when xx number of passwords are guessed incorrectly (such as an account lockout mechanism with alerts as just one of many available choices). Incidentally, I know this thread is somewhat off topic but it has been very helpful to me and since reading it I have checked my /var/log/secure logs and found that our system has been scanned for ssh-password guessing several times over the last few months. So thanks! That scanning has been going on for a long time, and the script kiddies that doing it are using pre-staged/pre-written password lists looking for the very simple passwords (eg, root/root, root/blank, root/test). They usually stop after about 30 to 60 different attempts, one right after the other. A small number of hackers will try other password guessing methods as noted in an earlier post. There are some open source syslog scanning tools, but I don't know of any off hand that do a nice job at managing thresholds. Might try google to see what's available. If your * box is exposed to the Internet, you might want to take a look at 'netstat -an' or 'netstat -a' to see what ports/services are actually exposed. For sshd, you will see entries in the /var/log/secure log like: Feb 10 11:41:16 asterisk sshd[23033]: User root not allowed because not listed in AllowUsers Feb 10 11:42:36 asterisk last message repeated 2 times Feb 10 11:41:22 asterisk sshd[23033]: Failed password for illegal user root from 1.2.3.4 port 53262 ssh2 Feb 10 11:40:58 asterisk sshd[22993]: Failed password for myuserid from 1.2.3.5 port 53255 ssh2 Writing a script to scan through the log entries and develop your own thresholds (based on what your system has exposed) is not that difficult. In the above, the first three entries are of no real value as root isn't allowed ssh access. But, the fourth entry was an attempt to guess the password for 'myuserid'. If you receive more then four or five failed attempts against a valid userid, then send a text message to your cell phone, email to your mail box, or whatever action you want to take (considering how serious this might be to you). If you are running firewall s/w on the machine, add the source IP of the attempt to whatever table is used to block that user. If you're running Cisco routers, execute a tcl script to add the IP to your access list. Lots of different choices depending on how serious this is to your environment and the resources available to you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
When echo occurs (the type where I hear myself echoing as I talk) what is bouncing against. Is it the other caller's equipment, the central office or something in between? When you are talking via 4 wire or VoIP phones there is a seperate outbound audio channel and inbound audio channel, niether the twain shall meet no echo Except for POTS lines (2 wire)... where you have one audio channel going in both directions. So you have these: (fixed font spacing needed) A Straight POTS B -- speaker-speaker || mic mic A talks into mic and the audio is injected into the single audio channel. A almost immediatly hears his voice in his own speaker, as the distance between the mic and the speaker is short. B hears A's speach a bit later traveling through the long line. We have echo as A hears his own voice, but the timing makes it perceived as sidetone. A ISDN/VoIP to POTS B -- speaker--===O---speaker || mic mic A talk into mic and the audio is sent as a seperate channel down the line. At some point this channel is injected into the single channel of the POTS line for B. The return channel to A picks up everything on the single channel POTS line (wanting to get B's audio, but also getting A's injected mic channel.) The distance between A's mic, the injection point and A's speaker combines to make the delay. This delay causes the echo to be heard as an echo and not a sidetone. * some (not all) VoIP/ISDN phones will simulate sidetone by sampling the mic and sticking it directly in the speaker. This is done because us humans are used to the POTS technology and think the line is dead if we do not hear it. The same goes for comfort noise generation. If the line is active we expect analog white noise on it. -- END OF LINE That summary is sort of reasonable for soft phones, but not very accurate for hard phones (analog or digital). The sidetone is 'always' generated within analog and digital phones. It never comes from any source outside the phone. In analog phones, it derived from the hybrid within the phone. On digital phones, its basically firmware. The conversion from four-wire (analog or digital) to two-wire requires the use of a hybrid (physical component in analog phones, mostly firmware in digital phones). The 'inefficiencies' of that hybrid is the source of echo, regardless of where they happen to be in the end-to-end communications path. Since it is impossible to know what each telephone company or long distance carrier has engineered, its not possible to guess at where hybrids might exist in that path. It is fair to say the number of hybrids is very small now compared to twenty years ago, but they do exist at least at both ends of a communications path. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax.conf config and iax based clients
Hi, I am a newbie in asterisk. trying to configure firefly third party edition to connect to aserisk 1.0.3 im able to authenticate but cannot dial extensions. I have been reading the documentation cant seem to find the correct configs. Attached the error message and configs. What am I missing? *CLI Urgent handler Feb 12 15:52:05 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect attempt from 33.44.5.55, request '[EMAIL PROTECTED]' does not exist Urgent handler Urgent handler Feb 12 15:52:14 NOTICE[16537]: chan_iax2.c:5718 socket_read: Rejected connect attempt from 33.44.5.55, request '[EMAIL PROTECTED]' does not exist Urgent handler Urgent handler iax.conf [general] bindport=4569 bindaddr=2.3.4.5 bandwidth=low jitterbuffer=no tos=lowdelay [QIax1] type = friend host = dynamic accountcode = iaxy secret = 12345678 contex = from-iax disallow = all allow = ilbc allow = gsm auth = md5 trunk = no qualify = no [QIax2] type = friend host = dynamic accountcode = iaxy secret = 12345678 contex = from-iax disallow = all allow = ilbc allow = gsm auth = md5 trunk = no qualify = no [QIax3] type = friend host = dynamic accountcode = iaxy secret = 12345678 contex = from-iax disallow = all allow = ilbc allow = gsm auth = md5 trunk = no qualify = no extension.conf [general] static = yes writeprotect = yes [bogon-calls] exten = _.,1,Congestion [from-iax] exten = 105,1,Dial(IAX2/[EMAIL PROTECTED],20) ;exten = 105,2,Voicemail(u2000) ;exten = 105,102,Voicemail(b2000) exten = 105,103,Hangup exten = 106,1,Dial(IAX2/[EMAIL PROTECTED],20) ;exten = 106,2,Voicemail(u2001) ;exten = 106,102,Voicemail(b2001) exten = 106,103,Hangup exten = 107,1,Dial(IAX2/[EMAIL PROTECTED],20) ;exten = 107,2,Voicemail(u2002) ;exten = 107,102,Voicemail(b2002) exten = 107,103,Hangup TIA, Wesley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Can agents login be permanent across Asterisk restarts ?
queues.conf ; Persistent Members ;Store each dynamic agent in each queue in the astdb so that ;when asterisk is restarted, each agent will be automatically ;readded into their recorded queues. Default is 'yes'. ; persistentmembers = yes Julian Robert Rozman wrote: Hi, I noticed that agents logins (agentcallbacklogin) are reset if Asterisk is restarted. Can this be avoided in some way ? Regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
Robert Hajime Lanning wrote: quote who=Eric Bishop Just out of interest, When echo occurs (the type where I hear myself echoing as I talk) what is bouncing against. Is it the other caller's equipment, the central office or something in between? When you are talking via 4 wire or VoIP phones there is a seperate outbound audio channel and inbound audio channel, niether the twain shall meet no echo Wrong. Look at any cellular phone or IP phone. They all have echo cancellers. If you switch these cancellers off the results are generally bad. What they need to remove is the acoustic spill from the earpiece to the mike. This can be a surprisingly strong signal. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Why echo occurs
Eric Bishop wrote: Just out of interest, When echo occurs (the type where I hear myself echoing as I talk) what is bouncing against. Is it the other caller's equipment, the central office or something in between? With 2-wire analogue line you will have echo from the hybrid at the local exchange. If the far end has a 2-wire analogue line you will have echo from the hybrid in their phone. Whatever kind of phone is at the other end there will be echo from the acoustic coupling between earpiece and mic (in cellular phones and IP phones this is usually eliminated by a local echo canceller). If you try to cancel these echos at your end it will only work if the path has a precisely constant length. If the path length changes (e.g. the far end is an IP phone), the echo canceller's training will keep falling apart. If the path length is constant, and the canceller well training it should do a very good job of eliminating the echo from the local analogue loop. It won't give more than about 30dB of suppression of the echo from beyond the A-law/u-law section of the link, due to the inherent distortion of the codec. It is normally necessary to suppress small residual signals to avoid hearing a weak echo from the distant phone when its user is silent. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Still stuck trying to make Asterisk read MySQL
I've been working with RealTime configuration from MySQL Server, and have had good results. You might check it out. You can do a search for 'realtime' on the Wiki and get some good documentation on how to set it up. I think in the extconfig.conf file, not only do you need to identify the engine (ODBC in your case), but you also need to identify the actual table you used for your Voicemail configuration. If I recall correctly, the default is a table named 'voicemail' and since you are using a different name, you need to specify the name in the extconfig.conf file so it can find it. beonice ([EMAIL PROTECTED]) wrote: I've been continuing to experiment with MySQL. I'm having absolutely no luck getting asterisk to read voicemail configuration data and mailbox configuration data from mysql tables instead of from voicemail.conf. The default Asterisk setup that reads from voicemail.conf and extensions.conf works fine. I'm using Asterisk CVS-v1-0-12/12/04-15:58:29 on a Whitebox Enterprise Linux box. I'm not using any telephony hardware or SIP phones. I've just got a voicepulse DID talking to asterisk via IAX. I've got mysql downloaded and installed and have successfully got the contributed script reading from my asterisk_vm database to set up the extensions.conf, as per the instructions at: http://www.voip-info.org/wiki-Asterisk+extensions+from+mysql Now I'm trying to get Asterisk to look up voicemail configs from the asterisk_vm database. In order to do this, I've been following the instructions at: http://www.voip-info.org/wiki-Asterisk+voicemail+database So, I've: 1) Updated the /usr/src/asterisk/apps/Makefile to have USE_MYSQL_VM_INTERFACE=1 and recompiled asterisk, with make clean; make; make install 2) Updated voicemail.conf to have the appropriate entries: dbuser=username ;; Yes I changed this to my username dbpass=password ;; Yes I changed this to my password dbhost=localhost dbname=asterisk_vm 3) Created the users table in the asterisk_vm database. +-++--+--+---+---+++ | context | mailbox| password | fullname | email | pager | options| stamp | +-++--+--+---+---+++ | default | | 1234 | Moron Tester | [EMAIL PROTECTED] | | attach=yes | 20050211131641 | +-++--+--+---+---+++ 4) Updated extensions.conf to have the following line: exten = ,1,VoiceMail(u) I tried restarting asterisk at this point, called in and tried to leave voicemail for extension (and mailbox) . Here's the message I get: *CLI Feb 11 13:21:36 WARNING[18393]: app_voicemail.c:1539 leave_voicemail: No entry in voicemail config file for '' So I dug around some more and found http://www.voip-info.org/wiki-Asterisk+res_config Decided to try these instructions as well. So: 5) I created the ast_config table as directed: Here is the data: ++++---++--+--+-+ | id | cat_metric | var_metric | commented | filename | category | var_name | var_val | ++++---++--+--+-+ | 1 | 0 | 0 | 0 | voicemail.conf | default | | | ++++---++--+--+-+ 6) I edited /etc/asterisk/configs/res_odbc.conf to contain: [mysql1] dsn = MySQL-asterisk username = myuser password = mypass pre-connect = yes [mysql1] dsn = asterisk_vm username = myuser ;; changed to my userid on mysql password = mypass ;; changed to my password on mysql pre-connect = yes [mysql2] dsn = MySQL2-asterisk username = myuser2 password = mypass2 enabled = no [ENV] VAR=VALUE 7) Inserted glue to tell asterisk where to look: ; /etc/asterisk/res_config_odbc.conf [settings] table = ast_config connection = mysql1 8) Rerouted Asterisk's config engine: ; /etc/asterisk/extconfig.conf [settings] ;queues.conf = odbc voicemail.conf = odbc 9) I modified the sample script load_res_config.pl and ran it, it successfully updated my ast_config table, stuffing in all the settings that I'm used to seeing in voicemail.conf. 10) I restarted asterisk _again_. I get the exact same message. Feb 11 14:18:40 WARNING[18528]: app_voicemail.c:1539 leave_voicemail: No entry in voicemail config file for '' I'm totally out of ideas now. Anyone else got a clue to lend me? Thanks, Maya __ Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. http://mobile.yahoo.com/maildemo ___ Asterisk-Users mailing list
Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.
Roger Hanson wrote: I've downloaded 2x and burned 2 cds and get an error invalid compressed format (err=2) system halted message both times. It'd be nice to have a MD5 to verify my download is OK. It'd narrow down the problem to either the download or the burn, wouldn't it? The other day I was getting problems with downloading files over 12mg in size. They all were failing the checksum. Found out it was my driver for the nic card in my Linux box. I was using an RealTec. Changed the nic to an Intel and no problems after that. - Original Message - From: Roderick A. Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 11:55 AM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. Daniel Eboa wrote: I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. Not as of approx 5:)) PM yesterday. I downloaded, burned, and in last stage of the install ( compiling * ) right now. I don't remeber if there was a md5sum for the iso, but a binary error in hte download or bad hardware ( cd burner ) are the twom main causes of this problem. Try another download. Rod -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] *@home .5 Double Dial Tone
Using digit networks x100p clone card. On both incoming and outgoing calls, once the call is connected a second dial tone is generated. Any ideas? I have tried both jacks on the x100p clone; both produce the same result. -Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Asterisk as b2bua
Hello. LCR means least cost routing, and it's billing system problem where to route a call, not b2bua's. But currently I dunno any free billing system that support it, so i moved this logic to b2bua. On Sat, 12 Feb 2005 07:05:39 +0330, mohammad [EMAIL PROTECTED] wrote: Hi Mike; Thanks for your new application, but I think it would be better if you put everything under the radius. I mean for example LCR (radius based call routing). have you any plan to do that? Warmest Regards Mohammad ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ast_data does not patch
Hello all, I have just been trying to install the latest ast_data from: http://svn.asteriskdocs.org/res_data/ast_data/ into my cvs version of Asterisk and have found that the install patching fails. - patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c patching file apps/app_directory.c patching file channels/chan_sip.c Hunk #2 succeeded at 621 (offset 9 lines). Hunk #3 FAILED at 1480. Hunk #4 succeeded at 1549 (offset 11 lines). Hunk #5 succeeded at 1617 (offset 18 lines). Hunk #6 succeeded at 1972 (offset 11 lines). 1 out of 6 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej patching file channels/chan_iax2.c Hunk #2 succeeded at 593 with fuzz 2 (offset 13 lines). Hunk #3 FAILED at 944. Hunk #4 succeeded at 4441 (offset 57 lines). Hunk #5 FAILED at 5234. 2 out of 5 hunks FAILED -- saving rejects to file channels/chan_iax2.c.rej patching file Makefile patching file pbx.c Hunk #6 succeeded at 1390 (offset 18 lines). Hunk #8 succeeded at 1439 (offset 18 lines). Hunk #10 succeeded at 1508 (offset 18 lines). patching file asterisk.c Hunk #2 succeeded at 1922 (offset 76 lines). -- Does anyone know how to get in touch with the developer or have another viable and working option that will allow me to dynamically place my users information in a MySQL database? Thanks, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk as B2BUA. New application!!!
Hello all! It's my try to make b2bua from asterisk. It's patched asterisk and some AGI script for it. What it support? Full vovida's b2bua radius emulation, radius failover, LCR, Call failover, Codec based routing, Session-Timeout and much other things that can be useful. Any suggestions welcome! http://b2bua.berlios.de Best regards, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE HELP Adit 600 went kaput?
Did a cable come out of the Adit(like the T1 cable???)? There should be a 'craft' port to hook up a serial port with a term program and you can poke around and see what alarms it's reporting. What alarms is * reporting? Lyle - Original Message - From: Richard Reina [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 11:05 AM Subject: [Asterisk-Users] PLEASE HELP Adit 600 went kaput? After months of setting up Asterisk. I completed the final testing last night We would go live on Monday. Or so I thought. As I moved the Adit 600 back out of the way, sliding it six inches. I noticed the major light was red as were both T1 and T2 lights ( eventhough only one t1 port is being used ) -- I have no idea what these lights mean or if they were lit before when things were working. Then I go a phone call and theres no dial tone. I have restarted asterisk, check all the connections to and from the CB, diconected, reconnected and repowered the CB, still no dial tone. I realize that it hard to help me with such a problem, however, if anyone can help shed some light on what the problem could be I would greatly appreciate it. Thanks, Richard __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Installation of Zatel
Hello, Is it ok to install Zaptel afterwards and go ahead and install Asterisk? For some reason I install Asterisk first and I wanted to use Conference Bringing which requires Meetme. Can I install Zaptel on top of Asterisk? Any help will be appreciated. Nitesh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE HELP Adit 600 went kaput?
--- Lyle Giese [EMAIL PROTECTED] wrote: Did a cable come out of the Adit(like the T1 cable???)? t1 cable is connected on bothe ends. There should be a 'craft' port to hook up a serial port with a term program and you can poke around and see what alarms it's reporting. Do you mean hook a monitor to? What alarms is * reporting? * seems to behave as if nothings wrong. There are no errors. Lyle - Original Message - From: Richard Reina [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 11:05 AM Subject: [Asterisk-Users] PLEASE HELP Adit 600 went kaput? After months of setting up Asterisk. I completed the final testing last night We would go live on Monday. Or so I thought. As I moved the Adit 600 back out of the way, sliding it six inches. I noticed the major light was red as were both T1 and T2 lights ( eventhough only one t1 port is being used ) -- I have no idea what these lights mean or if they were lit before when things were working. Then I go a phone call and theres no dial tone. I have restarted asterisk, check all the connections to and from the CB, diconected, reconnected and repowered the CB, still no dial tone. I realize that it hard to help me with such a problem, however, if anyone can help shed some light on what the problem could be I would greatly appreciate it. Thanks, Richard __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage. Learn more. http://info.mail.yahoo.com/mail_250 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom IP 3000 configuration
I see that typo I made for this suggestion, but the real problem is that the system doesn't seem to register with Asterisk. I can't dial out or even if I fix the error in my config will I be able to dial the extension. This phone just doesn't seem to want to work with Asterisk. I have found some old posts where people got this phone to work but they never post the solution so i am hopeful someone has the answer. Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK harry gaillac wrote: hello try: exten = 8908,1,Dial(h323/8908,20,Ttr) ! harry --- Scott Henderson [EMAIL PROTECTED] a crit : I am trying to add a Polycom IP 3000 to our Asterisk system and am not getting anywhere. h323.conf [8908] type=friend host=192.168.104.25 secret=polycom context=crv-default callerid="Conference Room Polycom" extensions.conf exten = 8908,1,Dial(h323/polycom,20,Ttr) ; Polycom exten = 8908,2,Hangup I have tried setting the Asterisk system as both gatekeeper and gateway in the polycom config. To date nothing seems to work and Polycom is now on a week return a support call to the reseller that sold us the unit. -- Scott Henderson Finite Technologies Incorporated 3763 Image Drive, Anchorage, Alaska 99504 Phone: 907.339.8085 ext 6101, Fax: 907.333.4482 http://www.finite-tech.com http://www.chillywall.com http://www.virtuale.cc http://www.mphage.com Current Local Time: http://www.worldtimeserver.com/time.asp?locationid=US-AK ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Dcouvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Crez votre Yahoo! Mail sur http://fr.mail.yahoo.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MGCP, Asterisk Cisco VG200
See my comments inline: snip ; mgcp audit endpoint aaln/[EMAIL PROTECTED] (vg200) [general] port = 2427 bindaddr = 128.100.10.10 (asterisk server) [vg200] Whatever host name that you put here should be resolvable either by /etc/hosts or DNS lookup. If not, set it to the IP address of the host, i.e. [128.100.10.11]. Hope this helps. snip ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Is no one using MySQL on stable asterisk?
beonice wrote: I'm still (doggedly) trying to get asterisk to read my voicemail configuration from MySQL. I'm using the stable release of Asterisk, from back in December, before realtime was included. If anyone has got it to work, please contact me ... I've posted details, but everyone who's responded so far has been working with the newer version that uses realtime. Use ast_data. it worked for me as of 1.0.2. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE HELP Adit 600 went kaput?
Did a cable come out of the Adit(like the T1 cable???)? On thing that is odd is that although the t1 cross over cable is plugged in to both * and the Adit. Both t1 and t1 leds on the Adit are red. How can they both have the same status if one is hooke up and on is not? Could my cross over cable have some loose wiring? There should be a 'craft' port to hook up a serial port with a term program and you can poke around and see what alarms it's reporting. What alarms is * reporting? Lyle - Original Message - From: Richard Reina [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 11:05 AM Subject: [Asterisk-Users] PLEASE HELP Adit 600 went kaput? After months of setting up Asterisk. I completed the final testing last night We would go live on Monday. Or so I thought. As I moved the Adit 600 back out of the way, sliding it six inches. I noticed the major light was red as were both T1 and T2 lights ( eventhough only one t1 port is being used ) -- I have no idea what these lights mean or if they were lit before when things were working. Then I go a phone call and theres no dial tone. I have restarted asterisk, check all the connections to and from the CB, diconected, reconnected and repowered the CB, still no dial tone. I realize that it hard to help me with such a problem, however, if anyone can help shed some light on what the problem could be I would greatly appreciate it. Thanks, Richard __ Do you Yahoo!? Meet the all-new My Yahoo! - Try it today! http://my.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do you Yahoo!? Yahoo! Mail - You care about security. So do we. http://promotions.yahoo.com/new_mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE HELP Adit 600 went kaput?
On February 12, 2005 07:31 pm, Richard Reina wrote: On thing that is odd is that although the t1 cross over cable is plugged in to both * and the Adit. Both t1 and t1 leds on the Adit are red. How can they both have the same status if one is hooke up and on is not? Could my cross over cable have some loose wiring? Unplug it and plug in a loopback plug (pin 1-5, pin 2-6) -- if the T1 alarm doesn't go away, the T1 controller itself is kaput. If it goes green (or off), then your wire is suspect. You can certainly have both T1 controllers showing alarm if you never turned the second one off. Honestly it sounds as if you didn't do *any* basic diagnostics here. Tell us what you *have* tried, and we can suggest other possible tests. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk@home .05 release questions on setup.
I tried downloading to a different PC - the download speed was much faster (or was it the mirror I was using?) - 4MB/sec. But again, I burned the image and got the same error. I suppose it's Nero that's messing it up? I've never had any problems on the many .iso's I've burned before with Nero. I guess I could try a different batch of CD-R's. Is a network install possible for [EMAIL PROTECTED] Someone send me a good CD and I'll paypal you a few bucks? - Original Message - From: Ariel Batista [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 4:50 PM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. Roger Hanson wrote: I've downloaded 2x and burned 2 cds and get an error invalid compressed format (err=2) system halted message both times. It'd be nice to have a MD5 to verify my download is OK. It'd narrow down the problem to either the download or the burn, wouldn't it? The other day I was getting problems with downloading files over 12mg in size. They all were failing the checksum. Found out it was my driver for the nic card in my Linux box. I was using an RealTec. Changed the nic to an Intel and no problems after that. - Original Message - From: Roderick A. Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 11:55 AM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. Daniel Eboa wrote: I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. Not as of approx 5:)) PM yesterday. I downloaded, burned, and in last stage of the install ( compiling * ) right now. I don't remeber if there was a md5sum for the iso, but a binary error in hte download or bad hardware ( cd burner ) are the twom main causes of this problem. Try another download. Rod -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Installation of Zatel
you cannot since asterisk checks for the existance of zaptel on initial compile and you will have chan_zap missing. On Sat, 12 Feb 2005 15:30:11 -0800, Nitesh Divecha [EMAIL PROTECTED] wrote: Hello, Is it ok to install Zaptel afterwards and go ahead and install Asterisk? For some reason I install Asterisk first and I wanted to use Conference Bringing which requires Meetme. Can I install Zaptel on top of Asterisk? Any help will be appreciated. Nitesh ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michal Bielicki http://www.asterisk.com.pl/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] soho fax suggestions?
Rich Adamson wrote: Need to replace our older soho fax machine with something more current. Would like to run the fax line through *, but haven't been able to make spandsp work correctly with digium TDM04b card. Our fax volume This seems to be a problem with the current wctdm driver. It seems to be broken for audio going out. I used to be able to send faxes reliably using spandsp and a TDM40P card, but I no longer can. I haven't had time to look in detail at what is wrong. is very low (maybe a few per week), but we have multiple offices in three geographic locations and would like to be able to email the images to the correct location. For planning purposes, is it appropriate to think in terms of purchasing a t38 capable box even if its not supported by * today? (I'm well aware of the bounty and Steve's work.) If spandsp doesn't work now, spandsp won't work through a T.38 channel. If now is the time to purchase a t38 capable fax machine, anyone have any suggestions on a low-volume soho-sized box? It seems the T.38 in a number of units doesn't really work. I'm not clear how widespread that problem is, but since there are only a few suppliers of protocol stacks for these boxes I suspect it may be widespread. Regards, Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE HELP Adit 600 went kaput?
Andrew Kohlsmith wrote: Unplug it and plug in a loopback plug (pin 1-5, pin 2-6) -- if the T1 alarm doesn't go away, the T1 controller itself is kaput. If it goes green (or off), then your wire is suspect. I really love it when a poster asks such a question: Could my cross over cable have some loose wiring? This question was asked on a mailing list, to a group of people who have never met this person, never seen his equipment, never seen his cable, etc. As best I can tell, there are three possible answers to his question: no - This would be the most humorous (but least helpful) answer :-) maybe - Accurate, but useless yes - Accurate, and obvious Equivalent quality answers could have been obtained from a Magic 8-Ball, and the poster could have continued trying to solve his problem, rather than asking us. You did a good job actually trying to help him out in spite of his lack of ability to troubleshoot on his own G ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk addson
Matthew Boehm wrote: This is most likely due to the HUGE CHANGES in recent code in regards to linked-lists. Be patient. They are being fixed and optimized. -Matthew Matthew, Has there been any progress on this? I am still getting the same error (from CVS-HEAD as of a few minutes ago): app_addon_sql_mysql.c:164:77: macro AST_LIST_REMOVE passed 4 arguments, but takes just 3 Does anyone have a working copy of this code? Thanks. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intermediary jitter buffering
Hello, I understand that only the destination of a call should do jitter buffering. So, if IAX2/PhoneA calls IAX2/PhoneB through my server (no transfers), PhoneA and PhoneB need to perform their own jitter buffering, and Asterisk will just forward the frames, correct? What happens if the peer does not support jitter buffering, but is close by so there's no need for jitter buffering? My situation is that I have an Asterisk machine right in front of our provider's systems (same switch, 1ms latency). If they don't have jitter buffering, how can I force my Asterisk machine to jitter buffer calls from my users to them? Thanks, -Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@home .05 release questions on setup.
I had the same problems, I changed network card first, same problem then I changed the burner and everything started to work. Make sure that on the second pc you have different burner. Oh and I use the nero 6 ... With no problems. robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Hanson Sent: Saturday, February 12, 2005 7:48 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. I tried downloading to a different PC - the download speed was much faster (or was it the mirror I was using?) - 4MB/sec. But again, I burned the image and got the same error. I suppose it's Nero that's messing it up? I've never had any problems on the many .iso's I've burned before with Nero. I guess I could try a different batch of CD-R's. Is a network install possible for [EMAIL PROTECTED] Someone send me a good CD and I'll paypal you a few bucks? - Original Message - From: Ariel Batista [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 4:50 PM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. Roger Hanson wrote: I've downloaded 2x and burned 2 cds and get an error invalid compressed format (err=2) system halted message both times. It'd be nice to have a MD5 to verify my download is OK. It'd narrow down the problem to either the download or the burn, wouldn't it? The other day I was getting problems with downloading files over 12mg in size. They all were failing the checksum. Found out it was my driver for the nic card in my Linux box. I was using an RealTec. Changed the nic to an Intel and no problems after that. - Original Message - From: Roderick A. Anderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 11:55 AM Subject: Re: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. Daniel Eboa wrote: I downloaded the iso file of the last release, but unable to burn it on CD. Got error at 90%. Did anyone experience the same problem ? Maybe the iso file is corrupted. Not as of approx 5:)) PM yesterday. I downloaded, burned, and in last stage of the install ( compiling * ) right now. I don't remeber if there was a md5sum for the iso, but a binary error in hte download or bad hardware ( cd burner ) are the twom main causes of this problem. Try another download. Rod -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE HELP Adit 600 went kaput?
Andrew Kohlsmith wrote: Unplug it and plug in a loopback plug (pin 1-5, pin 2-6) -- if the T1 alarm doesn't go away, the T1 controller itself is kaput. If it goes green (or off), then your wire is suspect. If he gets a green light with a loopback plug wired like that, his controller is definatly screwed up :-) 1-4 2-5 That was how I always learned to wire a loop plug anyway. David ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Setting a Forward to an external number on your phone
If you want to set up call forwarding in a device independent way, see http://lists.digium.com/pipermail/asterisk-users/2003-July/016872.html Unfortunately, if you get Asterisk to do the forwarding, there is no way to tell just by looking at the phone that your calls are forwarded. If you use an IP phone like Cisco, Polycom or Snom, you can use the phone's built-in Call Forwarding button, and you can tell by looking at the phone whether the calls are forwarded or not. Rana Dutt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] IAX2-FWD
Guys.. Im using iaxphone softphone for testing my asterisk config and I noticed that FWD offers a IAX to FWD gateway using asterisk or any IAX softphone.. Has anybody configured iaxphone to use this iax to fwd gateway? I want to try this out before messing with asterisk and fwd. Thx! __ Anton Krall ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@home .05 release questions on setup.
I have problems getting into maintenance screen of AMP, What is the user I should use? I must be missing something easy ... Thanks robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PLEASE HELP Adit 600 went kaput?
On February 12, 2005 09:21 pm, David Coulson wrote: If he gets a green light with a loopback plug wired like that, his controller is definatly screwed up :-) 1-4 2-5 That was how I always learned to wire a loop plug anyway. You're absolutely right, I made a pretty big (and public) thinko... hahaha -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] anyone patched CVS Asterisk with ast_data?
Hello All, Has anyone been able to patch the latest CVS Asterisk with the ast_data from: http://svn.asteriskdocs.org/res_data/ast_data/ I am having troubles getting a patch version together. Any help would be greatly appreciated. Thanks, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] i want to load chan_h323.so
I use Fedora core 2, and openssl-0.9.7, expat-1.95.7 is installed by rpm packages. I downloaded pwlib-1.5.2 and openh323-1.12.2 at my home directory(/root/root_src), asterisk 1.0.4 at directory /usr/src/ and have installed successfully. Asterisk is executed normally, but module chan_h323.so cannot be loaded. The message is : # asterisk ?vvvgc . .some message . Asterisk Ready. *CLI> load chan_h323.so /root/root_src/openh323/lib/libh323_linux_x86_r.so.1.12.2: undefined symbol: _Z13vpb_dial_synciPc Unable to load module chan_h323.so *CLI> Please give me your solutions. Thank you for your reading. My install log is : # tar xvfz pwlib-1.5.2.tar.gz # tar xvfz openh323-1.12.2.tar.gz # cd /root/root_src/pwlib # ./configure # make # cd /root/root_src/openh323 # ./configure # make opt # cd /usr/src # export CVSROOT=:pserver:[EMAIL PROTECTED]:/usr/cvsroot # cvs login # cvs co -r v1-0 asterisk # echo $PWLIBDIR /root/root_src/pwlib # echo $OPENH323DIR /root/root_src/openh323 # echo $LD_LIBRARY_PATH /root/root_src/pwlib/lib:/root/root_src/openh323/lib # cd /usr/src/asterisk/channels/h323 # make # cd /usr/src/asterisk # make install ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] iax.conf config and iax based clients
Hi, I changed the dialplan and the same error. By the way the * server has public IP address and the firefly clients are behind firewall(iptables). here is the error and config chan_iax2.c:5718 socket_read: Rejected connect attempt from 33.44.5.55, request '[EMAIL PROTECTED]' does not exist chan_iax2.c:5718 socket_read: Rejected connect attempt from 33.44.5.55, request '[EMAIL PROTECTED]' does not exist iax.conf [general] bindport=4569 bindaddr=2.3.4.5 bandwidth=low jitterbuffer=no tos=lowdelay [QIax1] type = friend host = dynamic accountcode = iaxy secret = 12345678 contex = from-iax disallow = all allow = ilbc allow = gsm auth = md5 trunk = no qualify = no [QIax2] type = friend host = dynamic accountcode = iaxy secret = 12345678 contex = from-iax disallow = all allow = ilbc allow = gsm auth = md5 trunk = no qualify = no [QIax3] type = friend host = dynamic accountcode = iaxy secret = 12345678 contex = from-iax disallow = all allow = ilbc allow = gsm auth = md5 trunk = no qualify = no extension.conf [general] static = yes writeprotect = yes [bogon-calls] exten = _.,1,Congestion [from-iax] exten = 105,1,Dial(IAX2/QIax1,20) ;exten = 105,2,Voicemail(u2000) ;exten = 105,102,Voicemail(b2000) exten = 105,103,Hangup exten = 106,1,Dial(IAX2/QIax2,20) ;exten = 106,2,Voicemail(u2001) ;exten = 106,102,Voicemail(b2001) exten = 106,103,Hangup exten = 107,1,Dial(IAX2/QIax3,20) ;exten = 107,2,Voicemail(u2002) ;exten = 107,102,Voicemail(b2002) exten = 107,103,Hangup TIA WEsley correct your dialplan. something like this [from-iax] exten = 105,1,Dial(IAX2/QIax1,20) exten = 106,1,Dial(IAX2/QIax2,20) exten = 107,1,Dial(IAX2/QIax3,20) hth ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk+GNOMEMeeting=No Sound.
Hi all! I'm newie to asterisk and I've been trying to make it work in order to use it with Linux softphones (H.323, SIP or IAX, I don't mind) and none hardware phone. I'm using asterisk packages from Debian SID (my distribution), asterisk, asterisk-config, asterisk-sounds, asterisk-h323. I've still not tried with any IAX softphone (gnophone?) but with linphone (SIP) I've not luck (oRTP errors in console) even to p2p connection between 2 linphone client computers or sipomatic. I've tried GNOMEMeeting also. It works fine with a P2P client connections (ALSA works fine) but, even when I success connecting to an asterisk server, I haven't hear anything. I mean, I don't hear the demo successfull messages. I've looking the GNOMEMeeting logs and it says that it closes the sound channel as soon as it connects to the asterisk server. This is my h323.conf file: [general] port = 1720 bindaddr = 0.0.0.0 allow=all ; turns on all installed codecs context=default and my extensions.conf file: [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp IAXINFO=guest TRUNK=Zap/g2 TRUNKMSD=1 . . . [demo] exten = s,1,Wait,1 exten = s,2,Answer exten = s,3,DigitTimeout,5 exten = s,4,ResponseTimeout,10 exten = s,5,BackGround(demo-congrats) exten = s,6,BackGround(demo-instruct) . . . [default] include = demo . . . I've also can see how asterisk says it actually plays these sound files in the CLI. Any idea? Thanks in advance. -- Andrés Gómez García Ingeniero en Informática Telf: +34 981 91 39 91 Fax: +34 981 91 39 49 mailto:[EMAIL PROTECTED] http://personales.igalia.com/agomez IGALIA, S.L. http://www.igalia.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ast_data does not patch
Why not just use the built-in database features to do what you want? Its called RealTime. Lots of info on it on the wiki. -Matthew - Original Message - From: [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 4:56 PM Subject: [Asterisk-Users] ast_data does not patch Hello all, I have just been trying to install the latest ast_data from: http://svn.asteriskdocs.org/res_data/ast_data/ into my cvs version of Asterisk and have found that the install patching fails. - patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c patching file apps/app_directory.c patching file channels/chan_sip.c Hunk #2 succeeded at 621 (offset 9 lines). Hunk #3 FAILED at 1480. Hunk #4 succeeded at 1549 (offset 11 lines). Hunk #5 succeeded at 1617 (offset 18 lines). Hunk #6 succeeded at 1972 (offset 11 lines). 1 out of 6 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej patching file channels/chan_iax2.c Hunk #2 succeeded at 593 with fuzz 2 (offset 13 lines). Hunk #3 FAILED at 944. Hunk #4 succeeded at 4441 (offset 57 lines). Hunk #5 FAILED at 5234. 2 out of 5 hunks FAILED -- saving rejects to file channels/chan_iax2.c.rej patching file Makefile patching file pbx.c Hunk #6 succeeded at 1390 (offset 18 lines). Hunk #8 succeeded at 1439 (offset 18 lines). Hunk #10 succeeded at 1508 (offset 18 lines). patching file asterisk.c Hunk #2 succeeded at 1922 (offset 76 lines). -- Does anyone know how to get in touch with the developer or have another viable and working option that will allow me to dynamically place my users information in a MySQL database? Thanks, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk addson
I didn't write app_addon_sql_mysql.c so I have no clue as to its current state. -Matthew - Original Message - From: Kristian Kielhofner [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 7:46 PM Subject: Re: [Asterisk-Users] asterisk addson Matthew Boehm wrote: This is most likely due to the HUGE CHANGES in recent code in regards to linked-lists. Be patient. They are being fixed and optimized. -Matthew Matthew, Has there been any progress on this? I am still getting the same error (from CVS-HEAD as of a few minutes ago): app_addon_sql_mysql.c:164:77: macro AST_LIST_REMOVE passed 4 arguments, but takes just 3 Does anyone have a working copy of this code? Thanks. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ast_data does not patch
Thanks I'll look into it, but from the little that I read on RealTime, I was under the impression that it did not use MySQL or PostgreSQL which is a database feature that I was hoping to use. --Lonnie Why not just use the built-in database features to do what you want? Its called RealTime. Lots of info on it on the wiki. -Matthew - Original Message - From: [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Saturday, February 12, 2005 4:56 PM Subject: [Asterisk-Users] ast_data does not patch Hello all, I have just been trying to install the latest ast_data from: http://svn.asteriskdocs.org/res_data/ast_data/ into my cvs version of Asterisk and have found that the install patching fails. - patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c patching file apps/app_directory.c patching file channels/chan_sip.c Hunk #2 succeeded at 621 (offset 9 lines). Hunk #3 FAILED at 1480. Hunk #4 succeeded at 1549 (offset 11 lines). Hunk #5 succeeded at 1617 (offset 18 lines). Hunk #6 succeeded at 1972 (offset 11 lines). 1 out of 6 hunks FAILED -- saving rejects to file channels/chan_sip.c.rej patching file channels/chan_iax2.c Hunk #2 succeeded at 593 with fuzz 2 (offset 13 lines). Hunk #3 FAILED at 944. Hunk #4 succeeded at 4441 (offset 57 lines). Hunk #5 FAILED at 5234. 2 out of 5 hunks FAILED -- saving rejects to file channels/chan_iax2.c.rej patching file Makefile patching file pbx.c Hunk #6 succeeded at 1390 (offset 18 lines). Hunk #8 succeeded at 1439 (offset 18 lines). Hunk #10 succeeded at 1508 (offset 18 lines). patching file asterisk.c Hunk #2 succeeded at 1922 (offset 76 lines). -- Does anyone know how to get in touch with the developer or have another viable and working option that will allow me to dynamically place my users information in a MySQL database? Thanks, Lonnie ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk@home .05 release questions on setup.
Yeh this could be explained a little better. If you log into the concole and type the command help-aah It will bring up all of the commands available to change passwords for the various user names (AMP, maintenance etc) When you change the passwords it will give you the user name. Because [EMAIL PROTECTED] is not modifying the actual software just packaging it up into a superscript [EMAIL PROTECTED] doesn't have an overall username or password, it just uses the various names from each application. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Augustyn Sent: Saturday, February 12, 2005 10:10 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] [EMAIL PROTECTED] .05 release questions on setup. I have problems getting into maintenance screen of AMP, What is the user I should use? I must be missing something easy ... Thanks robert ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
