Re: [asterisk-users] VOIP PBX replacement suggestions?
--Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Daniel Seagraves Sent: Wednesday, June 06, 2012 2:40 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VOIP PBX replacement suggestions? The boss wants to move from landline service to VOIP service as a cost-cutting measure. We have one voice line and one fax line. The telco is billing over $100 a month for the two. We're using Hylafax for faxing and a PBX for the voice line. Our existing PBX is an Intertel Axxess box with the old v5 processor. The management and voicemail computer died years ago (PSU burned up). I'm worried that it's going to die before too much longer. We have the IPRC and several IP Phone+ devices. It's my understanding that the IP Phone+ speaks only a proprietary Intertel protocol and can never be used with any non-Intertel equipment. I would like to dump the entire Intertel box and move to Asterisk instead, but my budget for this project is exactly $0. I can't afford to buy new devices. The boss is leaning toward getting digital voice service from the local cable monopoly. They want to charge us $30 a month per line to start, and we will have to sign a 3 year contract. The monopoly in question has a reputation for very poor service, but they are a monopoly so my boss sees them as the only alternative. My worry is that if we sign that contract, we are trapped with both the intertel and the cable monopoly, and if I exceed the capacity of the intertel (or it just dies) I am SOL. My questions then are as follows: 1) Is there a way I don't know about to get Asterisk to talk to either the IPRC or the IP Phone+ directly in such a way that gets calls from one to the other? 2) Are there any reputable VOIP providers that provide business service at a rate less than $30 per line per month? The boss is adamant that we need unlimited minutes. === Where do you get your IP connection? The cable monopoly? There are several companies you can get service from. One is Teliax.com Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Incoming fax cuts ADSL line
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of James Sharp Sent: Wednesday, May 16, 2012 11:19 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] OT - Incoming fax cuts ADSL line On 5/16/2012 12:07 PM, Tim Nelson wrote: - Original Message - Hi, I'm facing a strange situation. Though it's not directly related to Asterisk, I do think it is interesting to this mailing list. The setup is a single line which is split between an ADSL modem/routeur and a fax machine (Asterisk was removed from the equation). Any time the fax machine rings (incoming fax), the ADSL service is troubled to the VPN users are disconnected. It can be reproduced at will. I've changed the ADSL filter twice (a different unit, then a different model) without any visible change. What could explain this ? I've experienced this quite a few times, and after working with a local telco, it has become policy to not place ADSL on lines where fax is going to be used. I'm unsure of the exact technical reasons behind this other than 'the fax signals/frequencies interfere with the ADSL signalling/frequencies used on the circuit'. It sounds like you might want to separate your fax/ADSL lines. --Tim You might also be able to limit the Fax machines maximum transmission rate so the modem's transmission spectrum doesn't inch up into where the ADSL service is. ADSL is transmitted at a relatively low frequency using phase modulated carriers to achieve the bandwidth. It could be about 32 different phase/level locations on 360 degree/level pie chart or vector scope. The actual frequencies of the carrier are moderately low, maybe 100 to 200 kcps. Voice is low density. Faxes and modems are high density and loud. They can splatter or have harmonics that can confuse the local DSL demodulator. As others have said, the best thing to try is the best filters you can get between the phone line and the DSL demod, and maybe two filters in series. If that doesn't work, put the fax on a different line than the DSL, which could cost you money. Paying for better filters or two of them is less expensive than separate lines. Or move the DSL to an alternate existing voice only line, since you probably don't want to change the fax number Contents of this message were dredged from foggy memory. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] USB or Ethernet based FXO device ?
We have run some tests on the Xorcom equipment, mostly the PRI port cards, running up to 16 ports in a chassis. They work. I see no problem in Xorcom as FXO ports. We will be installing a lot of them as PRI ports soon.. CF _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Marco Signorini Sent: Wednesday, August 31, 2011 10:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] USB or Ethernet based FXO device ? Hi. I was following this thread. We normally use Patton SmartNode SN4112 series to interface to FXO ports. But I'm looking for something different for a future setup. Xorcom USB channel banks seems something quite interesting. Is there anyone that could/would share experiences using that? We need to replace an old PBX interfaced to 50 FXS and 8 BRI ISDN in Italy. My concern is about reliability of USB Any success stories with it? Tips and tricks? Thank you and regards, Marco Signorini. mailto:marco.signor...@ingegnitech.com -- http://www.ingegnitech.com/images/logo.gif INGEGNI Tech S.r.l. site http://www.ingegnitech.com/ http://www.ingegnitech.com mail mailto:i...@ingegnitech.com i...@ingegnitech.com _ Gilles wrote: On Tue, 30 Aug 2011 10:41:30 -0500, Carlos Chavez mailto:cur...@telecomabmex.com cur...@telecomabmex.com wrote: Actually Xorcom makes USB channelbanks of up to 32 FXO/FXS ports. Thanks for the tip. It looks like the smallest option is 8 FXO ports: www.xorcom.com/telephony-interfaces/astribank-models.html -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello image001.gif-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dragging the dialup customers along, possible?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C F Sent: Monday, August 29, 2011 3:18 PM To: t...@ewebforce.net; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dragging the dialup customers along, possible? From what you are asking it appears that you are trying to run similar to a fax (modulation and demodulation) over VoIP. Try again, the fact that you succeeded twice was pure luck, and as far as I understand that didn't even work out. Switch back to TDM. Your dial up modems want that magic thing called timing and no jitter that only TDM will give you. === This is more of a whimsical statement than a scientific one, but I would think in today's world, there would be a real small box that would take in IP and put out TDM with good timing with a moderate buffering window. Obviously, the IP has to actually get to the box in a timely fashion, like today , but a TDM circuit has to be up also. A box that would take in IP data..., look for valid ascii, and otherwise put out TDM modem tones with no data content for 1 second and then pick up the data as it catches up. Better a laggy modem connection than no data at all. CF -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
There is a T1 over Ethernet scheme that runs a T1 over Ethernet, Up all the time. It consumes 1.5 +/- megs 24/7. I would suspect that was what was being offered. Cary _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman Sent: Thursday, July 07, 2011 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Anybody doing PRI over IP? _ From: eric weaver ecwea...@gmail.com Sent: Thursday, July 07, 2011 4:41 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Anybody doing PRI over IP? A carrier I like will be introducing PRI over IP, presumably going thru some sort of gateway box (I'm guessing by Adtran but no data yet). Has anybody set up successfully to work directly with such a feed without bothering to take it down to T1 and use a T1/PRI card? Thanks eric I agree with others that likely what you are getting is a product that is SIP based and it is just being priced and bundled to compete with a PRI connection as most bussiness owners and phone guys know what a PRI is.. We have pri's into gateways that run on our VOIP network and we have sip trunks and we mix services out to our customers based on what the routes require. Most of our up line CLEC's can now deliver their TDM and SIP services in both forms so in most cases we take the SIP version and where the vendor does not support SIP correctly we take their PRI version and convert it to SIP ourselves on our gateways. zktech -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anybody doing PRI over IP?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Thursday, July 07, 2011 4:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Anybody doing PRI over IP? - Original Message - There is a T1 over Ethernet scheme that runs a T1 over Ethernet, Up all the time. It consumes 1.5 +/- megs 24/7. I would suspect that was what was being offered. Wow, that sounds horrifically inefficient. *MAYBE* keep the D channel open all the time if necessary, but what about all the channels that are not in use? Just waste 64k for each channel just because? Do you have any information on this technology, or the name of the vendor that offers it? You've piqued my interest. :) --Tim The reason for this scheme as proposed to us was for FAX use rather than other means. The people who proposed it however within a week converted to Plan B. So, I have no further info. CF -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] gmkioliyuiuuuioihlllkppuipppppookkkgkkkhklfgko
-- I know we've come a long way, We're changing day to day, But tell me, where do the children play? Yusuf Islam, Nov. 1970 AKA CAT STEVENS. -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk fails when DNS or internet fails
It seems to me: That every Asterisk system that is being used for PBX or other internal as well as external use should have a local DNS, run either on the same box or on an adjacent box. A simple BIND installation is low overhead. If remote phones use it for DNS then if they are on the net, they have all info they need to make a call anywhere, and commonly called locations are either in DNS or cached. Then there are no DNS based failures unless the Net is down or the Asterisk box is down. Well, now, I should do as I preach and install Bind on our Asterisk box. However since we do have multiple DNS locally, we are effectively following this advice. C. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free CNAM
BE WARY OF THIS ONE! If you click the link it comes up with a simple block Text Message US GOVERNMENT I doubt the US Government has any thing to do with it but... something is fishy here. Cary -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael R. Wally Sent: Sunday, May 29, 2011 6:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Free CNAM FreeCNAM.org is providing a free CNAM API for Open Source PBX users. This API queries a private CNAM database, and returns standard 15-Character CNAM results. Any entry not already in the database will be queued for investigation, and added to the database as soon as information is located. This system has access to several CNAM backends, and is not a party to any use-limiting or no-caching agreements. The API is: http://freecnam.org/dip?q=2024561414 You can monitor the stats, including the current queue size, at freecnam.org API Results will continually improve as the database grows, so please be patient with limited results at this early stage. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free CNAM
Excuse me! Doing that with http://freecnam.org/dip?q=9038874180 Comes up with WAL-MART which is correct, so I guess my mistrust was paranoid. (But just because you are paranoid doesn't mean that someone isn't trying to infiltrate your computer or steal your email address or what ever!) And the returned page is perfectly clean as far as HTML code is concerned! I DO suggest changing the example so as not to alarm suspicious minds. Cary -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael R. Wally Sent: Sunday, May 29, 2011 6:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Free CNAM FreeCNAM.org is providing a free CNAM API for Open Source PBX users. This API queries a private CNAM database, and returns standard 15-Character CNAM results. Any entry not already in the database will be queued for investigation, and added to the database as soon as information is located. This system has access to several CNAM backends, and is not a party to any use-limiting or no-caching agreements. The API is: http://freecnam.org/dip?q=2024561414 You can monitor the stats, including the current queue size, at freecnam.org API Results will continually improve as the database grows, so please be patient with limited results at this early stage. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free CNAM
Thanks I had just sent off a mea culpa, and I have a gold plated tin foil hat that I love and will continue to wear! :-) Cary -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Sunday, May 29, 2011 7:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Free CNAM You are doing a CNAM lookup on that 202 number. Change the URL to a number you know, and it will do a CNAM lookup on it. You can take your tinfoil hat off now. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Sunday, May 29, 2011 8:05 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Free CNAM BE WARY OF THIS ONE! If you click the link it comes up with a simple block Text Message US GOVERNMENT I doubt the US Government has any thing to do with it but... something is fishy here. Cary -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michael R. Wally Sent: Sunday, May 29, 2011 6:48 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Free CNAM FreeCNAM.org is providing a free CNAM API for Open Source PBX users. This API queries a private CNAM database, and returns standard 15-Character CNAM results. Any entry not already in the database will be queued for investigation, and added to the database as soon as information is located. This system has access to several CNAM backends, and is not a party to any use-limiting or no-caching agreements. The API is: http://freecnam.org/dip?q=2024561414 You can monitor the stats, including the current queue size, at freecnam.org API Results will continually improve as the database grows, so please be patient with limited results at this early stage. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free CNAM
GRIN! C. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: Sunday, May 29, 2011 8:48 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Free CNAM On 05/29/2011 09:37 AM, Michael R. Wally wrote: So how long till its an adaptive telemarketing blocker based on the query velocity of the numbers ? The system uses real Telco CNAM Dips. Any generic names you get are from the subscriber's carrier itself. We can only provide what we ourselves get. I tried it, but it returns the same kind of junk that some of the databases do. For example, on a Florida number, it just says FLORIDA instead of the proper name (some of the CNAM databases have the right name). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Free CNAM
The system uses real Telco CNAM Dips. Any generic names you get are from the subscriber's carrier itself. We can only provide what we ourselves get. There's more than one CNAM database (aren't there seven?). I would have hoped that a service such as this would look at a bunch of them and choose the one that had the best result. == I am the original skeptic in this thread. (and, horrors, have top posted besides!) But, What do you want for free? Oh, Asterisk for starters of course, but... don't look a gift horse in the mouth. Say Thanks, I hope it grows. So how about some one in the open source community come up with a few lines of code to add it to Asterisk? C. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Wednesday, May 04, 2011 11:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Discussion: Are we ready to leave 1.4 behind? 2011/5/5 Flavio Goncalves fla...@asteriskguide.com snip but stuffing Asterisk with many new features on each version does not seem to be contributing to the stability of the code or the migration to newer versions. yes but it seems to me that code stability is improving. Maybe next 1.10.0 version will be production-ready from day 1 ? Flavio E. Goncalves www.asteriskguide.com Compare to which version of Windows. Patches are a never ending process Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Occasional call from asterisk
We were getting a lot of those. We installed IPTables with blocking of everything outside of North America and they all but vanished. No direct evidence, but a pretty good empirical guess that they were related to hackers trying to get paths to the US. CF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian Henning Sent: Thursday, April 07, 2011 4:54 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Occasional call from asterisk Hi, Now and then our SIP phones ring with asterisk showing as the caller-ID. Upon picking up the receiver, there is about five seconds of silence and then the channel is closed (hangup). Can anyone offer some insight? Here's relevant snippets from my extensions.conf and Master.csv log: This line shows up in Master.csv: ,,1-NOANSWER,inbound,,DAHDI/1-1,SIP/505-0150,Dial,SIP/5 01SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr,2011-04-07 21:37:05,2011-04-07 21:37:16,2011-04-07 21:37:21,16,5,ANSWERED,DOCUMENTATION,1302212225.444, Here's [inbound] from extensions.conf: [inbound] exten = s,1,Answer exten = s,n,Ringing exten = s,n,Set(CALLERID(num),9${CALLERID(num)}) exten = s,n,Dial(SIP/504SIP/506,5,tTgr) exten = s,n,Goto(1-${DIALSTATUS},1) exten = 1-ANSWER,1,Hangup exten = _1-.,1,Dial(SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506,10,tTgr) exten = _1-.,n,Goto(2-${DIALSTATUS},1) exten = 2-ANSWER,1,Hangup exten = _2-.,1,Voicemail(499@default,u) exten = _2-.,2,Hangup The idea is that first 504 and 506 ring, then if neither of them answer, everyone rings. Works great most of the time. I have a hunch that maybe this happens if the inbound caller hangs up while the first Dial() is ringing, but I would've expected to see the first Dial (to 504 and 506) show up in the Master.csv log, and it's not there. (The preceding line of the log is a call from almost an hour earlier). In that case though I'd expect to see 1-CANCEL in the log instead. Perhaps if the caller happens to hang up right between the two Dial() commands?.. As an aside, the Set(CALLERID...) bit doesn't work. The idea was to prepend a 9 so that a SIP user could use the redial feature of the phone's call log to return a missed call (automatically including the 9 for outside line). Unfortunately the 9 does not get prepended. Thanks in advance for any and all advice! ~Brian -- Brian Henning, Software Engineer /\Pine Research Instrumentation //\\ 5908 Triangle Drive ///\\\ Raleigh, NC 27617 USA || ||phone: 919.782.8320 fax: 919.782.8323 email: bhenn...@pineinst.com -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
You are a bad person! ;-) CF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen Sent: Thursday, March 31, 2011 10:53 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk and fail2ban On Thu, Mar 31, 2011 at 10:42:52AM -0500, JR Richardson wrote: I have F2B set to ban after 1 attempt. The most I have seen in the logs is 4-5 attemps before ban is applied. I am calling scripts that apply the ban to a cisco access-list, so there is script/telnet/config delay but it is very minimal and works very well. So I forge one SIP packet and I get you to block the IP address of your SIP trunk (or your IAX trunk)? Cool! -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and fail2ban
Obviously, the other side of the world wants connections to your side, no matter what side you are on. :-) Cary -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Administrator TOOTAI Sent: Tuesday, March 29, 2011 3:21 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] asterisk and fail2ban Le 29/03/2011 19:34, Sherwood McGowan a écrit : On 3/29/2011 12:25 PM, Steve Edwards wrote: On Tue, 29 Mar 2011 12:10:59 -0500, Sherwood McGowan First thing I'd do is restrict the ip blocks your sip endpoints can register/call from in sip.conf (or your database's table for sip endpoints) On Tue, 29 Mar 2011, Gilles wrote: Thanks for the idea, but it's not possible, as the Asterisk must be accessible for road warriors and receive SIP calls from anyone. Really? How many callers are you expecting from North Korea, Libya, China, Iran, etc? Thanks Steve, you just emailed exactly what I was going to say... Remember guys, there's a LOT of IP blocks out there that are almost definitely not going to be somewhere you expect to receive SIP traffic from. Well, I can tell you that our servers in europe those days are mainly attacked by US IP ranges (remember last year the problem with amazon cloud). They now disappear here in europe but lots of other US networks quickly replace them :-( -- Daniel -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [zapata.conf] What is wink?
Wink, I think is a start protocol aks wink start. It is like a flash, but happens as part of the predialing/dialing process. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Brummell Sent: Tuesday, March 01, 2011 7:47 AM To: Asterisk Users Mailing List - Non-Commercial Discussion; asterisk-users@lists.digium.com Subject: Re: [asterisk-users] [zapata.conf] What is wink? http://www.carrieraccessbilling.com/telecommunications-glossary-w.asp _ From: asterisk-users-boun...@lists.digium.com on behalf of Gilles Sent: Tue 3/1/2011 7:08 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] [zapata.conf] What is wink? Hello I couldn't find information about what wink is in zapata.conf: www.voip-info.org/wiki/view/Asterisk+config+zapata.conf#TimingParameters Does someone know what it is, and how it differs from flash? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Caller ID
We do not get caller ID (name) on our telco lines. However we have a few single line extensions with consumer type handsets that ring at odd hours with Asterisk before the phone is picked up, and Out of Area after it is picked up. I have read that Asterisk is what is reported by Asterisk for 0 length caller ID number. But since we don't subscribe to Caller ID Name, I am wondering where the Out of Area is coming from? Could these be hacking attempts via IP? Perhaps they are doing the caller ID name? It only happens to a few extensions as far a we know. TIA for any input or knowledge. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Caller ID
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, March 01, 2011 11:31 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Caller ID _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Tuesday, March 01, 2011 11:23 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Caller ID We do not get caller ID (name) on our telco lines. However we have a few single line extensions with consumer type handsets that ring at odd hours with Asterisk before the phone is picked up, and Out of Area after it is picked up. I have read that Asterisk is what is reported by Asterisk for 0 length caller ID number. But since we don't subscribe to Caller ID Name, I am wondering where the Out of Area is coming from? Could these be hacking attempts via IP? Perhaps they are doing the caller ID name? It only happens to a few extensions as far a we know. TIA for any input or knowledge. IP Hacking should not apply on your Telco lines. I'd start with your CDR file (/var/log/asterisk/cdr-csv/Master.csv) and go from there. I had in mind a hacking attempt IP calling extension lines that exist, but thanks, I will look at the cdrs. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk caller ID
We are getting a lot of calls identified as Asterisk or out of area in the middle of the night. From other posts on the list, I have assumed these are null Caller ID calls and Asterisk is plugging in pseudo ID. Is that correct? It seems to me that Asterisk should simply say no caller ID or No ID or something besides Asterisk. In any case, we are trying to filter them with little success. When we do a LEN(CALLERID(num) we get 13, when we expect 10 The call pattern is 1 call followed by a second abut 1 minute later followed by 1 about 10 minutes later. Does anyone have any ideas to contribute? Thanks Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Google Voice outbound Caller ID broken
What kind of broken are you seeing. It could be the ID is pseudo ID and may never reflect the actual caller. CF _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Gentle Sent: Thursday, February 24, 2011 8:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Google Voice outbound Caller ID broken Anybody else noticed that caller id for outbound calls via Google Voice seems to be broken? It seems to be a Google Voice problem though, not an asterisk issue. -- Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] REFER and dialplan broken (as documentedinchan_sip.c on line 11951)
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Wednesday, February 23, 2011 3:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] REFER and dialplan broken (as documentedinchan_sip.c on line 11951) I've exhausted every option without paying someone to fix this, so asterisk might as well be commercial software. It is free if you can use it. You can pay for all the help you want to or have the money to pay for. The Asterisk Software Charity Society went bankrupt about 2500 years ago. You can pick some name from the mail list and demand they fix the issue you perceive. But you probably won't be able to wait for results. There are 3 legs to any transaction. Speed, Quality, Price. You get to pick any two. The other party gets to set the 3rd one. You can't set all 3. Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] REFER and dialplan broken(asdocumentedinchan_sip.c on line 11951)
It is free if you can use it. You can pay for all the help you want to or have the money to pay for. The Asterisk Software Charity Society went bankrupt about 2500 years ago. You can pick some name from the mail list and demand they fix the issue you perceive. But you probably won't be able to wait for results. There are 3 legs to any transaction. Speed, Quality, Price. You get to pick any two. The other party gets to set the 3rd one. You can't set all 3. Cary Except in Wisconsin. Even there. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] On-Hold Music
And there is also ASCAP: American Society of Composers, Authors and Publishers. Also a smaller than either group: SESAC. Cary -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Monday, February 14, 2011 2:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] On-Hold Music On Monday 14 February 2011 08:23:08 Danny Nicholas wrote: Might not be your question to answer, but if I did get a BMI license, this would allow me to use virtually any music I wanted for MOH? The answer is, as long as the music publisher for each piece of music has an agreement with BMI to license their music, yes. You can verify each individual title here: http://www.bmi.com/search/ -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
How amusing that you follow that statement by being too lazy to trim all of the irrelevant crud after your comment by pressing ctrl-shift-end followed by delete. It works in Outlook. Tom This is the problem, everyone has a personal goal. One side wants fast replies at the top, with no interest in the repetitive, redundant signature/disclaimers content below. The other wants total historical readability or questions and answers in top to bottom readability in every message. And, this is a type of list that is used by 1000s of individuals, not people from a single company. We are just lucky we don't have someone posting in sentences that read from right to left. :-) Also most (all?) mail clients don't allow setting preferences based on the source of the message. I.E. Top post for email, bottom post for the cooking list and bottom post for the Asterisk list. And then almost no one trims anything no matter what their preferences/beliefs are, and yells at others for top or bottom posting or interleaving, usually while violating some other list rule or general net etiquette. How about just no quoting or only the actual last message you are replying to? The list doesn't require any quoting. Contribute your thoughts, and leave it at that. Everyone has the previous posts on their computer, if they don't know the history, let them go back and read. Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
Paul Belanger wrote: Moderation would be another option (personally opinion). Regardless, we should all now be aware of the rules [1] of the mailing lists. All we can do now is hope people respect them. [1] http://www.asterisk.org/community/rules -- Paul Belanger Digium, Inc. | Software Developer With that type of trimming and my own trimming, bottom posting works for me, as well as top posting. There is little difference. But with 5 screens of text, , 7-10 repeated messages multiple signature lines and other tripe, bottom posting is a PITA. So if others trim, I am happy to bottom post. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
Hi, I thought this kind of discussion didn't exists in asterisk list. I guess most tech list members will argue on top-vs.-bottom subject at some point :) anton It only exists when someone starts the discussion. The top posters never start it. Bottom posting wouldn't be bad if the posters trimmed all but the message they are replying to from the reply. (See what I did?) Top posting puts the recent replay clearly as the first thing on the screen. I often have messages with 5 to 10 screens of previous posts to wade through to get to the bottom. If repliers would trim old content from their replies, it would save a lot of readers time and make bottom posting very efficient. I went through this type of discussion 20-25 years ago because we were all on 2400 to 14,400 baud modems, and trimming was demanded. Then Rich Text and HTML came along and you should have heard the screaming about bloated messages. Six-ten line signatures with ascii-art were not really appreciated either. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Top Posting
Tom Rymes wrote: On Jan 15, 2011, at 9:29 AM, Don Kelly wrote: That said, of course I want to follow this list's etiquette. I've posted a couple times asking how I can interleave responses in Outlook or what other approach can I take to make it practical to stop top-posting. Any suggestions? Don: Outlook-QuoteFix: http://home.in.tum.de/~jain/software/outlook-quotefix/ I found that program last night after reading one of the pages linked in this thread. The program isn't supported on OL 2007 and newer, but there is a link on the page to a macro for newer versions. Wish I had known about it years ago! Also, http://mailformat.dan.info/config/outlook.html shows the general steps needed to make Outlook approximate standards. HTH, Tom Thanks. As far as making bottom posting work, http://home.in.tum.de/~jain/software/outlook-quotefix/ makes it so much simpler and better! I just installed it, and will try to keep using it. If I don't want the previous material at the top, I will delete it. But, personally, I really prefer top posting or the previous material deleted. However... Time will tell. Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip attack.. fail2ban not stopping attack
Simply to reduce the attack, and then improve the defense: If you don't need traffic from some area that is attacking you, just put the whole area in IPTables. A list is available on VOIP-INFO.org. Cull out what you want to allow. Then tune Fail2Ban at your leisure. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum E1 Ports on Asterisk ?
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zoel Hairi - Yahoo Sent: Wednesday, December 22, 2010 5:50 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Maximum E1 Ports on Asterisk ? Hi All, Just a little over thought. Sorry if someone already asked about this before. Is it possible to put all 16 Ports of E1 in One Asterisk Server ? And if it's not possible is there any suggestion or alternative for me to use more than 320 lines of outgoing calls on One Asterisk Server ? Thanks ZH The general answer is Yes, maybe. I suggest you look at the Xorcom.com website for their load test data. Using a well sized server with best practice tweaks is important. It appears that bigger is not always better. For instance it seems to hurt or at least give no benefit to use a quad core processor. We just ran tests that indicates Xorcom's 3000 model would handle 16 CAS T1s. CAS T1s produce a very high interrupt rate. PRI T1s don't cause nearly as high rate. The right choice of cache, motherboard, processor and tweaks, are essential. You would be leading the pack. I think. .Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum E1 Ports on Asterisk ?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Wednesday, December 22, 2010 6:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Maximum E1 Ports on Asterisk ? On Wed, Dec 22, 2010 at 8:50 AM, Zoel Hairi - Yahoo zoelha...@yahoo.co.id wrote: Hi All, Just a little over thought. Sorry if someone already asked about this before. Is it possible to put all 16 Ports of E1 in One Asterisk Server ? And if it's not possible is there any suggestion or alternative for me to use more than 320 lines of outgoing calls on One Asterisk Server ? Thanks ZH Zoel It is possible to do what you are asking. In general the issue is raised about having all your eggs in one basket where one server or hardware failure can drop all of your lines for a period of time. External solutions like Xorcom and Redfone are great ways of abstraction. The concurrent call load on a server relies on the work to be done on each call. If you are using multiple codecs and recording the calls in another file format with other complex dialplan or AGI scripts then one server may not handle the calls well. If everything is ALAW and just dialing though then this would not be a problem for one server. If you search the list for sizing concurrent and load you will find more information. One very nice thing is that testing is very easy with or without the E1 hardware, try running the TDMoE channels between two servers and run a SIPp or other test to see the issues in a lab. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ In a previous post I also mentioned Xorcom. They do have a unique fail over ability with their Astribank systems. With dual servers, separate chassis and power supplies for the 4 port T1/E1 cards, USB interconnections, and redundant power supplies for the Astribanks, system downtime can be minimized, and if there is a failure, repair would be at worst, no screwdriver needed. If system failure would be idling 200 - 400 people, avoiding system down time would be a major objective. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, December 22, 2010 11:56 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Vacancy - Asterisk MySQL Support Engineer 45KSouth London On Wed, 2010-12-22 at 11:23 -0500, C. Savinovich wrote: 45K ? With 45K I can barely pay for gas, tolls, and breakfast. If you guys are such a fast growing company, probably you can pay better salaries. CS And you have to know Kalamino! :) You know Kalamino? I haven't seen him since prison in Budapest! Tell him hello! :-) CF -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk as a caller ID
In a 1.4.24 system, out of several lines, one of ours gets 1 or more random calls a day with Asterisk as the caller ID. I have just seen this described in the last couple of weeks, but at the time it wasn't happening to us, and I the explanation didn't stick with me. Can anyone give me a pointer to this feature? Searching the message base for Asterisk seems futile. Thanks! Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible??
I am not sure where you are and what legal conventions are involved. Are you saying the Telco (and legal restrictions) say you cant send calls to the internet via the AS5300 but you can if Asterisk does it directly? What is the logic in that? Or are they saying your Telco to Asterisk trunks have to be SS7 controlled? Or are you concerned about Asterisk handling the TDM to IP conversion in an adequate manner? I am not sure/aware myself that Asterisk will do a modem to IP conversion. I think in your example the AS5300 is doing that. What is the Telcos problem in doing what the customer was doing before? Feel free to correspond directly if you want to. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of José Pablo Méndez Soto Sent: Wednesday, November 24, 2010 7:31 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Incoming calls through SS7 for data modemtransmissions - possible?? Hello, We are working on implementing a solution for a medium service provider. They were previously using a Cisco AS5300 gateway with some PRI trunks to receive modem calls, then route them out the Internet. The Telco they were buying the trunks to discovered this configuration and restricted them due to legal conventions, and stated that in order to continue doing this, they would have to talk SS7 directly. We are planning on solving this by placing an Asterisk server with some TE410 cards talking SS7 to Telco, and another 4 ISDN ports talking to the AS5300 for the dial-up to complete after authenticating against a RADIUS server. My questions is: can we use only Asterisk to complete/terminate the dial-up connection, removing the AS5300 out of the picture? Current topology to be set-up: Telco -- SS7 -- TE410P-AsteriskServer -- ISDN -- AS5300 -- Internet Ideal topology: Telco -- SS7 -- TE410P-AsteriskServer -- Internet Some posts talk about zapRAS being able to accomplish this, not quite sure though Sounds like possible: http://lists.digium.com/pipermail/asterisk-users/2004-January/026956.html mailto:asterisk-users@lists.digium.com Sounds like not possible: http://lists.digium.com/pipermail/asterisk-users/2009-November/240202.html Thanks in advance, José Pablo Méndez -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Spam
I have been pounded with new, mostly text spam in the last few weeks. Tonight I realized that the address that is being spammed is a personal one I use for this list. Has anyone else noticed new spam in the last 2-3 weeks? Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GSM and SS7 Questions
In regard to #2, any T1 card should work. But the problem is you need SS7 software and SS7 connectivity in addition to the T1 card. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Sent: Wednesday, November 17, 2010 2:31 PM To: asterisk-users Subject: [asterisk-users] GSM and SS7 Questions I have two questions for the group. #1 - I'm looking to use some GSM SIM cards with my Asterisk PBX. Can anyone recommend a gateway? I need about 10-15 SIM slots. #2 - I'm also looking to connect Asterisk to an SS7 signaled DS1 (24 channels) for inbound and outbound voice calls. Can anyone offer any suggestions for cards to use there? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] T1 with Robbed Bit Signaling
Has anyone here used T1s with RBS with asterisk? Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 with Robbed Bit Signaling
So you are using RBS for CO to Asterisk, and SIP for Asterisk to subscriber. How does the Telco send you the called party info and you to them? DTMF? What signaling system is being used? EM or? We are trying to get DS0 EELs from the Telco via T1s and are not sure what their trunk scheme will be. We are guessing it will be RBS T1s with DTMF from the customer premises. To enhance the question: Is anyone on list getting customer loop EELs from a US (ATT) Telco, and if so, what flavor and other details? Cary Fitch boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere On Tue, 16 Nov 2010, Cary Fitch wrote: Has anyone here used T1s with RBS with asterisk? Cary Fitch All of my T1s are RBS. No PRI service here. Am using both Digium and Sangoma cards. LaCoursiere So you are using RBS for CO to Asterisk, and SIP for Asterisk to subscriber. How does the Telco send you the called party info and you to them? DTMF? What signaling system is being used? EM or ??? We are trying to get DS0 EELs from the Telco via T1s and are not sure what their trunk scheme will be. We are guessing it will be RBS T1s with DTMF from the customer premises. To enhance the question: Is anyone on list getting customer loop EELs from a US (ATT) Telco, and if so, what flavor signaling, and other details? Cary Fitch -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] T1 with Robbed Bit Signaling
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, November 16, 2010 2:58 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] T1 with Robbed Bit Signaling How many T1s per server? And, the T1s would be DS0s from customer's homes. Cary We are trying to get DS0 EELs from the Telco via T1s and are not sure what their trunk scheme will be. We are guessing it will be RBS T1s with DTMF from the customer premises. To connect two remote offices? Why don't you use SIP/IAX over the net or over a tunnel? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to construct a call center on asterisk
That is a broad question. If it is a 10-20 person call center, you may do it. If it is a high density center with lots of lines, requiring fail over capability, etc. so that hundreds of employees are not sitting around during down time, I would suggest designing it by buying it from a manufacturer that has done it before. I know of a company whose name starts with X . There may be others. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phuong Hoang Sent: Monday, November 15, 2010 8:01 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to construct a call center on asterisk Hi all Now, i want to construct a call center on asterisk but i don`t know how to do. Can anyone help me? Thanks and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to construct a call center on asterisk
Sorry, I don't have any experience in the call center area. Cary _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phuong Hoang Sent: Monday, November 15, 2010 10:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to construct a call center on asterisk Thanks Cary, I constructed IVR (Interactive Voice Response) system on asterisk and now i want to construct a call center to support IVR system.The call center that i want to construct has 7-8 employees so that i don`t want to buy it. I found on internet and guided using Open SIP Server with asterisk to construct a call center but i don`t know how to do now. Do you know using Open SIP Server with asterisk to do?please help me! Thanks and best regards. Phuong Hoang. On Tue, Nov 16, 2010 at 10:32 AM, Cary Fitch ca...@usawide.net wrote: That is a broad question. If it is a 10-20 person call center, you may do it. If it is a high density center with lots of lines, requiring fail over capability, etc. so that hundreds of employees are not sitting around during down time, I would suggest designing it by buying it from a manufacturer that has done it before. I know of a company whose name starts with X . There may be others. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phuong Hoang Sent: Monday, November 15, 2010 8:01 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to construct a call center on asterisk Hi all Now, i want to construct a call center on asterisk but i don`t know how to do. Can anyone help me? Thanks and best regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big practical systems
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joel Maslak Sent: Sunday, November 07, 2010 2:56 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Big practical systems I believe this looks like a standard channel bank. Asterisk generates all audio. Ring and hook status are sent out of band. Dial tones are in-band. Ringback, busy, congestion are in-band audio. I would think a standard T1 card would be fine. That said, I would verify this with the LEC. === Does anyone know if ATT EELs delivered to a CLEC would be PRI, or Robbed bit? Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
My guess is they are looking for 10 digit phone numbers as extensions. Are they all from 1 IP address or from many? If from many, they are likely many serial scan or from a list of suspected VOIP numbers. If from one, and that random, then from a list of suspected VOIP numbers. Since you listed a phone number as part of your signature… I might guess hackers might soon add that number to a scan list. It is one thing to randomly run 2,XXX-, to 999-999-, with skips for the “dead zones,” (0-XXX-XXX-) etc. but another to hit suspected VOIP numbers. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Murphy Sent: Sunday, November 07, 2010 8:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Why are the hackers scanning for these? Hey, I'm going thru logs, and I see some very common and interesting things that the hackers are looking for. In a whole bunch of scans, I've noticed that the first guess or two for sip accounts is usually a 10-digit number. I'm asking myself, why these numbers? Are they looking for a voip trunk? Or is it just like a serial number for the scan? What? Here's some examples: 2648061411 3190339404 2685608247 3358171034 2092652562 2206598858 Just trying to follow the advice: Know thy Enemy murf Steve Murphy ParseTree Corp. 57 Lane 17 Cody, WY 82414 ✉ m...@parsetree.com ☎ 307-899-5535 http://www.wisestamp.com/email-install?utm_source=extensionutm_medium=emailutm_campaign=footer Signature powered by http://www.wisestamp.com/email-install?utm_source=extensionutm_medium=emailutm_campaign=footer WiseStamp http://s.wisestamp.com/pixel.png?p=mozillav=2.0.3t=1289138760949u=949715e=4286 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Journo Sent: Sunday, November 07, 2010 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Why are the hackers scanning for these? Here's some examples: 2648061411 3190339404 I'm getting exactly the same. Odds of getting a working number, are like the odds of winning the lottery. My guess is they are either trying to find a voip trunk, or they are trying to make cold calls to the extensions on my system. Sales or something similar. We got pounded last weekend, but installed a list of distant IPs in IPTABLES and see nothing this weekend. We have no need to be contacted by any sites more than 2500 miles away, and not too many from within 2500 miles. ;-) Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
I've just switched my outbound ip address a week ago. Not static, but dhcp on TimeWarner cable. I've registered only with another of our offices. The outbound calls are all pstn bound through Teliax. But somehow my log is filling up with registration requests over this new ip address from a bunch of addresses. How can these guys find my new ip address? Or are they just scanning all ip addresses in creation? sean -- _ Follow the money Just like for Spam, there is money in Sip-Hacking. Anyone that has SIP traffic to move (selling the service) has money. If they can move it for free, even more money. A few servers running Hacking programs (SIPVicious) or e-mail server hacking programs is no big deal and bandwidth at colo centers is unlimited. Then they convert to BOT controllers and have free computers and bandwidth world wide. They generate a database of public IP addresses (DHCP, whatever) and have a target of poorly protected IPs to troll. Lucky you. ;-) Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why are the hackers scanning for these?
Adding on more thoughts: Think what Google has done in Mapping the Earth, Mapping the Web, and now working on Google Voice and Google Mail. Every one of those makes money either directly and/or synergistically with other components. Now consider someone with telephone interests or spam interests. In this modern database and filtering and probing age, load in ARIN or RIPE IP Ranges, start building database data and filters, and let it run... And the other IP areas too. Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Big practical systems
I don't want to start the How many calls can Asterisk handle? discussion or How many angels can stand on the point of a pin? discussion either. But can anyone contribute some practical knowledge of systems that take in channel bank T1s or DS3s from far away, and process the calls? I am looking for real world, been there, done that, or check the 'Belchfire Systems GigaFiber 65536' system. Not to start the discussion, but Is there a board that will take a DS3 (672 channels) and a system that will handle the calls, or is that a silly question? Is there an IP box that would take the DS3 and then a system that would handle the calls? My guess would be yes because the actual call load would be far lower than 672 calls. Maybe 100-150 or so simultaneous. Each line/call would have to have absolute caller ID. In other words, PSTN call handling. Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big practical systems
Yes. Adtran makes excellent gear. The MX 2800 is good for breaking a channelized DS3 into PRIs. Thanks, will look at that. Ah, a DS3/T1 mux. I was looking for a DS3 PC Card... it would have 672 channels but the system doesn't need to handle but 20% of them at one time. If you're just talking 150 calls, you could do that with two 4x port cards in a single PC. I thought you were talking a lot bigger. ==I mean DS3 with 672 channel paths. There are 672 subscribers out there. I am saying that only a percentage of them are talking at peak times. We need to supervise 672 lines and expect 15% to talk at the same time. Each line/call would have to have absolute caller ID. In other words, PSTN call handling. Ummm, there's no such thing as absolute caller ID. You wanna try that question again? callerID is not legally binding, is not used by billing, anybody can spoof it. ===I mean we have to provide service and know what line is calling, not just provide anonymous service to a lot of people. We can't just mux a bunch of lines in to the Asterisk box with no identification. Cary -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Phones slow to ring
Watch the console as you dial. Dial the number and #. The ring should be instant. Or if not, look at the console and report what you see. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jy Sent: Thursday, November 04, 2010 5:32 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Phones slow to ring I am new to asterisk and using it for a research project. Have set up an server (version 1.6.2.6) and 2 SIP phones (Linksys spa901) which are registering fine with the server. They are able to call one another, however, the problem is it takes roughly 8-10 seconds for the called phone to ring. I have a really simple dialplan using only 4 digit extensions and have turned off callerid. Both phones are on the same subnet and I have enabled nat and keepalive. Does anyone have an idea what could be wrong here or idea on how to debug this problem? Thanks, John -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Under heavy attack
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jon pounder Sent: Tuesday, November 02, 2010 10:24 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] FW: Under heavy attack I'm still on old-fashion copper-wire and have yet to experience the joy of SIP Trunk-ing and the type of issues discussed in this thread. My thought to share here is that outgoing calls should be easy for thoroughly authenticated users and impossible for others... Probably more can-o-worms than help. Sorry if this is so. nothing new here, this is just the digital equivalent of a wats line with a weak access code for outbound access. the difference is code guessing can be a lot more aggressive now, and finding the inbound path is simpler. == Each system needs to be configured according to its purpose and needs. Simply these are phone systems, not e-mail or web servers. You may want to be able to get mail from (almost) anywhere in the world, same for web services. But for a phone system you may have very different needs. One can visualize the differences between a national or international VOIP provider, a 4 person office in Little Rock, AR, a local SIP provider in Houston, TX and an international sales company with offices in Rome Italy. A small sip system used with an upstream VOIP provider should be invisible to 99.% of the world's population. (Excepting any other trusted peers.) If there was a wide spread peering network and an individual system wanted/needed to access and be accessed like email then it would be a different world. We could all be robo-call spammed just like email. :-( But leaving small systems open for attack from 99. percent of the world is just begging for trouble. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Under heavy attack
I was going to point out a failing of the attackers, but figured they read the list and don't need any more tips. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Monday, November 01, 2010 12:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: Under heavy attack And obviously these attackers read our emails on lists like this and adjust their sick strategies accordingly. Zeeshan A Zakaria -- www.ilovetovoip.com www.pbxforall.com (beta) On 2010-11-01 12:02 PM, Jamie A. Stapleton jstaple...@computer-business.com wrote: Only 100? We had a single server over 300. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Saturday, October 30, 2010 9:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Under heavy attack My count has reached 100 for the day. The server serves doesn't serve international calls anywa... Sat, Oct 30, 2010 at 9:33 PM, Joel Maslak jmas...@antelope.net wrote: No. It seems that opening ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Under heavy attack
-BEGIN PGP SIGNED MESSAGE- Hash: SHA256 Here's my take on the attack... Sigh... http://www.stuartsheldon.org/blog/2010/11/sip-brute-force-attacks-escalate-o ver-halloween-weekend/ Stu They were trolling for SIP account IDs, not really trying to register. It was a coordinated bot or spoofed source attack not The Halloween Club doing tricks. Any small system should: Use IPTABLES and block any parts of the world you don't need access to/from. Start with any Class A address that is probing your system. Make your SIP IDs 8-12 characters in length, and use at least alpha numerical characters, some special characters if you like a little more variety. bear3579 b3e5a7r9 Bear3579 La3579ke Or more. Do the same for passwords. 6543office 7659home Etc. Are these perfect? No, but they are human friendly, and require the exploiter to hack a 16 to 24 character combination ID and Password that has 36 or more characters in the character set. Of course some dashes or periods or commas or others can be added. And when you see an attack if it isn't from a network on your planet, put the whole network in IPTABLES. (And get the world country delegations for IP addresses and block all not on your planet.) $.02 Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: Under heavy attack
I know there was talk on VUC recently about some kind of realtime RBL for SIP. Has anything progressed? It would be SO easy for asterisk users to contribute to a blacklist and also do a lookup in realtime to see if an IP has been blacklisted. A little bit of joined up thinking in the community could eliminate this issue. Would also be another major + for Asterisk as a platform.. Regards Brian Some systems need to communicate with the world. Other only with their own network, and a few selected outside addresses. If anyone from Amsterdam or Nigeria or Malaysia (and 100 other countries) is trying to get on our system, we are surprised! Vail Colorado, not so much. :-) Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Under heavy attack
We have about 8-10 boinking us. They generally run a 1- peer attack and a few alphas like common words or eieio We use large, complex peer IDs and passwords, so they have a long way to go. I am happy to help keep them busy. I also send messages to their network abuse address. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Saturday, October 30, 2010 6:11 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Under heavy attack On Sat, 2010-10-30 at 14:28 -0400, Zeeshan Zakaria wrote: My main asterisk server is under unusual heavy attack, and so far Fail2Ban has blocked about 30 IPs, from various different countries. At this time it is blocking about 1 IP address every few minutes. Just wondering if anybody else is also experiencing unusually increased hack attempts today? Just 30 ? I got 1593 different IP's on my personal blacklist who constantly are looking if i may lower my guards. Though 82.101.63.5 and 132.68.58.60 are rather busy tonight... hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ISDN SS7
I do not have knowledge of the SS7 vendors for Asterisk. Using redundant 56k data channels, we handle calls via 6 DS3s (672 X 6 calls) from the PSTN on a commercial telephone switch, with no issues at all. SS7 can support any number of simultaneous calls depending only on the bandwidth of the SS7 channels. SS7 is always done on a redundant channel basis since it is so important. Cary _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang Sent: Sunday, October 24, 2010 12:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ISDN SS7 Hi cary, Can you recommend me what add-on vendors I should use ? Can a open source solution such as chan_ss7 or libss7 support many conncurrent calls (for example 240 calls) ? Thanks _ From: Cary Fitch ca...@usawide.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Sun, October 24, 2010 9:33:28 AM Subject: Re: [asterisk-users] ISDN SS7 SS7 is an inter-telco system using a separate network for all signaling. You must have an SS7 network connection before anything will work. Then the T1 Spans run 24 64k audio paths. The SS7 net exchanges the call data and connection info between the switches. Asterisk doesn't support SS7 natively although I believe there are one or more add-on vendors. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of huu giang Sent: Sunday, October 24, 2010 10:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ISDN SS7 Hi all, I'm being requested to deploy an IVR service using SS7. I've deployed Asterisk before using ISDN connection, but never with SS7. Can anyone explain me the different between using ISDN and SS7 ? What need I do now to change to use SS7 ?. Many thanks, Giang -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Needed in Dallas, Texas, Network Voice Engineer and Field Service Install Technician
His post may have been of interest to some outside of DFW, and I appreciated your post less than his. But, enjoy. C == A very vast majority of people on here are not in Dallas (and indeed probably a majority in the US). So stop filling their mailboxes with this crap. Incase you hadn't noticed Asterisk Users Mailing List - Non-Commercial Discussion -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Blacklisting
We would be interested. Spam is a harder problem to fight due to volume and the ability of any idiot to set up free email accounts. But anyone blasting SIP systems is a pure commercial crook. Tagging and strangling them should be a clear cut project. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Thursday, October 21, 2010 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] SIP Blacklisting Hi, Given the recent increase in SIP brute force attacks, I've had a little idea. The standard scripts that block after X attempts work well to prevent you actually being compromised, but once you've been 'found' then the attempts seem to keep coming for quite some time. Older versions of sipvicious don't appear to stop once you start sending un-reachables (or straight drops). Now this isn't a problem for Asterisk, but it does add up in (noticeable) bandwidth costs - and for people running on lower bandwidth connections. The tool to crash sipvicious can help this, but very few attackers seem to obey it.. The only way I can see to alleviate this, is to blacklist hows *before* they attack. This means you wont ever be targeted past an initial scan. Is there any interest in a 'shared' blacklist (similar to spam blacklists, but obviously implemented in a way that is more usable with Asterisk/iptables)?. Clearly it raises issues about false positives etc, but requiring reports from more than X hosts should alleviate this. There's all the usual de-listing / false-listing worries as with any blacklist, but the SMTP world has solutions we could learn from. Leaving a 'honeypot' running on a single IP address has revealed a few hundred addresses in less than a month. I am fairly certain these are all 'bad' as this host isn't used for anything else. There is obviously a wealth of data (and attacks) out there that would be good to share. Anyone have any thoughts? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dials a trunk when off hook
I am not sure that can be done literally by Asterisk because most phones/adapters give dial tone when off hook, but Asterisk doesn't know the phone is off hook until a send button is pushed, several seconds pass after some keys are pressed, or the # button is pressed. However many of the adapters can be set to autodial. I would look for a phone or adapter that has autodial ability. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Baha @ SH Sent: Friday, October 22, 2010 6:11 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] dials a trunk when off hook How can I let asterisk immediately dials a trunk when off hook? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] fraud advice
As a practical matter, on anything that can generate endless billings, there should be a dumb trap that compares current usage to history (last month) and if usage exceeds 2/1 or 3/1 for instance then usage is choked or denied enough to cause the user to complain or perhaps generate a message to call customer support, (or call your cell phone!) Then if it is valid, raise last month's reference enough to let current calling continue. If it isn't valid you have found a problem and saved your or your customer's caboose. As to who to complain to, gather all info possible and report to everyone you can find. Someone may investigate, but there isn't likely anyone who will absolve the problem. Some will just take the report and ... as far as you are concerned, do nothing. There isn't much a local police dept. can do about a hacker in Western Slobovia cracking your server. Generally the FBI doesn't take matters of less than $10,000. But it sounds like you may meet that test. But they could take months or years or never finding the culprit and finding the culprit will likely net you nothing financial for you will be 1/10,000 of the fraud they did. This is a problem like spam in email. But this has cash costs to the server operator/customer. Passwords need to be un-crack-able, and there should be usage alarms, as described above. Depending on the situation even a single counter to your upstream billable sip server for all usage would likely trip on excessive usage and save your bacon. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Thursday, October 14, 2010 8:11 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] fraud advice Hi, Embarrassed as I am to write this, I am hoping for some advice. One of our very first PBX installs, now six years old, was taken advantage of over the past few weeks. A victim of sipvicious, I assume, that managed to guess one of the SIP passwords. 4000 calls to various middle eastern destinations have been placed, which ended up being sent over our customer's PSTN trunk, and of course there was no warning until the bill came today. Unfortunately the bill only covered the first few days of this fiasco, and was only $700. I am afraid the one that is on the way will be tens of thousands. ONE CALL on the bill that just arrived was $200 (80 minutes to Sierra Leone). I'm sure this started out as a single scan. It must have been posted, because I have at least ten IP addresses now that were placing calls via the same peer. They are from all over the world. So what is the accepted procedure? I'm in the US Virgin Islands, so do I go to the FBI? Police? Is their some telecom fraud body to report such things to? Does any one ever get any relief from such events? I'm basically sick to my stomach right now. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big time system
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stuart Elvish Sent: Saturday, June 26, 2010 6:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Big time system Hi All, I am not sure if my comments will be helpful but here goes. snip Let me know if you need anymore pointers. Also happy to consult but you would need to contact me off list for that... Stuart Elvish === Thanks, your info is most helpful, as is the other info I have received, some of it in private messages. Your snapshot description of a ~4000 user system and architecture is a good starting point for our planning. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Big time system
Thanks for the feed back, but the rates are more or less predetermined. ATT rates would be $.0007 per minute for local calls. The operation would be providing local phones wired to houses with copper pairs. What I am looking for is the best ways to handle those lines when brought to a local switch site. The actual switch might not be there but back hauled, might be a TDM switch, a concentrator (TNT, etc) 10 ganged Asterisk systems, or tin can and string. I see some talking about TNTs in this forum. Those are 672 lines or in some versions double that, what is used behind them to do the processing, etc. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Friday, June 25, 2010 9:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Big time system On Thu, Jun 24, 2010 at 11:24 PM, Cary Fitch ca...@usawide.net wrote: But, we have an opportunity to get into a big time telecom activity. It would have 2000 to 30,000 user lines per city, and we would like to have those brought back to a central location for control and because transport can be more economical than remote site rentals, maintenance and personnel. I would say you need to make an RFP process to first negotiate your calling rate extremely low with the major vendors of the country where you're operating. If this is US, you're talking Qwest, ATT, Verizon, and the ilk, and you negotiate an extremely low minute rate in return for giving them a guaranteed minimum revenue. And while you're at it, you ask them how they suggest you design the architecture over their national network. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Big time system
We are an asterisk user... small time system 50-100 users or so. But, we have an opportunity to get into a big time telecom activity. It would have 2000 to 30,000 user lines per city, and we would like to have those brought back to a central location for control and because transport can be more economical than remote site rentals, maintenance and personnel. We could take the local lines into concentrators (TNTs or equivalent) and bring back IP to a central site, or put servers at the remote cities. Our object is to serve as a central office switch for subscribers on standard telco service loops. This isn't a How many lines can I handle using a Belchfire 2600 processor? type question but a request for pointers to big time systems. There would be no IP path to the end user, just copper. Thank you Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why asterisk down when inet server down?
But why can't my phones call. The outgoing lines are PRI/DAHDI T1. No sip. No iax. Why does the asterisk machine have to resolve any address? The internal phones can't even call each other, even though they have hard ip addresses. Same for doing DHCP for handing out addresses to your phones... All the phones have manual ip addresses. No DHCP. sean Do the phones find the sip server by IP or by domain name. I.e. 1.1.1.1 Or sip.yourdomain.com If domain name, what are they using for DNS? Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ring splash
The ring splash is a long standing feature of call forwarding. Of course somewhere in the Asterisk code a change could be made to extend the time required to detect a valid ring. But, how about just unplugging the pots lines from the PBX with a quick restore ability? Unplug lines at the NID, or open bridging clips or whatever applies. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, May 26, 2010 11:36 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ring splash Something new to me. Recently installed a 1.4.30 box for a small office with four POTS lines in a hunt (Digium TDM410P). Had the telco put a call forward option on the main line of the hunt. They dial a feature code from their desk phones (Polycom IP450) that results in forwarding the main number to our VoIP service. This is all to let them try out our dialtone service before porting the number to us and ditching the POTS lines. So we perform some test calls and they all go through fine, and everyone is happy, BUT everytime a call comes through it ALSO causes the POTS line to ring, and a ghost call rings all the phones in the office (the desired result of an inbound call from POTS). When they answer it they get fast busy because it isn't actually a real call. I spoke to the telco this morning about it and they said oh yeah - that is a ring splash that lets the customer know that a call was forwarded. They said this was a feature of their DMS-100, it has worked that way for twenty years, and they can't turn it off. So to the question - can the TDM410P somehow tell the difference between a ring splash and an actual inbound call? I think in the meantime I will send inbound POTS calls to an auto attendant that will eventually hang up, but would love a more elegant solution ;) Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NPA NXX Database
http://www.localcallingguide.com/ will give you lots of info. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly Sent: Tuesday, May 18, 2010 12:14 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] NPA NXX Database Has anyone had good results with an on-line database that returns a LATA based on NPA NXX? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MATH
Clue, If a caller keys in 4 5 3 will some variable return 453? I ASSume yes, since you can make menu selections with DTMF, obviously you can process the results further or in other ways than that. Cary -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Sunday, January 31, 2010 2:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MATH On Sun, 31 Jan 2010, Thomas Perron wrote: does dtmf any any variable that i can capture and use w/ some logic like in the case of a gotoif Anyone have a clue what this means? Anyone? Anyone? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] disable comfort noise
Something else that is flaky, missing or otherwise irritatingly broken in the piece of shit that is 'Asterisk'. [Cary Fitch] It is an open source project. When can we count on your contribution of a comfort noise generator that will not be a piece of s--t? Can you have that by Monday? CF -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem
As a guess, they can both talk to the server, but can't talk to each other. What is common to that is they may be trying to reinvite each other, and there is no path through the respective routers/firewalls to the other. So if reinvite is set to yes, set it to no, in both phone profiles on the server. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves Arikoglu Sent: Monday, January 25, 2010 7:28 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] sip.conf with versatel and two NICs very strangeproblem Hi My System is: Asterisk 1.6 running on a Dell Server with two network interfaces. eth1 (IP 10.26.208.252) is connected to a versatel sip-router tha has the local ip 10.26.208.252 and the external ip 89.244.x.y eth0 of the server is configured to 10.26.192.107 The Problem: SIP registration works, phone rings in- and outbound, but there is no audio, nor the caller neither the callee can hear anything. So i am quite sure that is has something to do with firewalls, natting and so on but i?ve read hundreds of pages and tried thousands of setting but i cant get audio to work.. the strange thing is... when i call the versatel-sip-number from my mobile phone, i see the call coming in in the cli, i see the voiceprompts that asterisk plays, but even there I cant hear anything on my mobile. next strange thing: i defined 2 sip-extensions. both are registered... everything is fine... routes are ok, they can call out and can be called from external and from internal (sip phones call each other).. but the same... no audio. but when one sip extension calls a wrong number... the cannot be completed message is hearable. i configured a queue with moh and even this works... but why cant to sip-phones talk to each other? why cant an external caller hear any audio? if i make sip debug, i see traffic (and due to extension is calling i think that on the sip-level everything is okay...) how can i see, which port and interface is chosen for audio when a call comes in? thanks, yves -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 10/100 voip phones and gigabit connection
If the phone is first, then it slightly limits the PC and rebooting the phone causes loss of contact with the PC. If the PC is first you have to have dual ports on it (a few bucks of hardware, plus configuration costs), then rebooting the PC causes the phone to loose contact with the world. Not good if you are on the phone and need to reboot. Two separate feeds would work best, but cost more. Dual wall jacks with green-PC and blue-Phone jacks could then be used. The phone will be on the desk, the PC may be under it. Two jacks, two cables. If there were a standard for two Ethernet connections in a cable... that could work, but might interfere with Power Over Ethernet. I wouldn't want to be like Bill Gates saying 640K memory is enough for anyone circa-197?, but isn't two 100 meg connections enough for any single desk? The phone doesn't really need more than 10 meg. An advantage of the separate net for the phone, is that it would make POE easy for the phones, and eliminate a lot of wall warts under every desk. Plug in the phone and it works. YMMV. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP over VPN -- no audio to other remote/VPNconnected phones
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan McCormack Sent: Monday, January 11, 2010 1:05 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] SIP over VPN -- no audio to other remote/VPNconnected phones Hello, I am having a problem with my current SIP over VPN setup. We have a server running asterisk at our office. All the phones in the office are on the same network / local to this server. We also have two employees with home offices using SIP phones over VPN to connect to the asterisk server. These phones have no problem with calls to the phones in the office, however there is no audio when trying to place a call from one remote phone to the other. I'm assuming this is a routing issue, but I have no been able to come up with a solution. Thanks for any advice. [Cary Fitch] One thought: if you are using reinvite try turning that off. That will be a clue. It would seem that both phones are on the local net via VPN, and should be able to talk to each other if they can talk to anyone in the office. (As you know.) So look for clues as to real paths to each other. Cary Fitch -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Please remove me from the mailing list.
I have been doing this (whatever that is), since about 1976, involving many facets, including posting on #1 CBBS out of Illinois, usenet in the 90s, and more. It is not possible to get people to follow all the RFC rules and customs much less the -- CR sigdashes. There are a lot of relative newbies much less oldies who never heard of such, or run 10 different mail options from gmail to hotmail to sendmail to I have no idea what it is, I am just a member, poster, customer, or-something mail user. In the 90s, a very well like member of a BBS type system (MajorBBS/Worldgroup) went ballistic when people started using HTML. The other people on the net finally just told him We don't care, we are not staying in the dark ages. Like it or lump it.. I am on numerous lists where 75% to 100% of the posts are top posts. If someone bottom posts people who want to go to the bottom and read. Most of us start at the earliest post and read message by message.. and don't want to rescroll through 10-20 messages over and over. I know that it would be nice to have the last message have all the text inline, but that doesn't happen either. And then there are always 5 more messages in the same thread later today. I don't even use a sig file. I just type my name. But to see if it works: -- Cary Fitch IMHO, top-posting isn't the problem, but just an obvious symptom of the real problem, which is failure to edit/strip the quotes to the bare minimum. When a thread gets hijacked by top-posters, who bang out their thoughts without even scrolling down to see all the garbage below, another problem also becomes apparent, and that is the failure of many MUAs to honor 'sigdashes', which is the convention of preceeding your sigfile with a line that is 'dash dash space CR'. A compliant MUA will strip that line and everything after it when quoting for a reply or forward. Note for the list admin: Please preceed your message-footer with a sigdashes line! -- Rick Green Those who would give up essential Liberty, to purchase a little temporary Safety, deserve neither Liberty nor Safety. -Benjamin Franklin As for our common defense, we reject as false the choice between our safety and our ideals. -President Barack Obama 20 Jan 2009 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialplans Holiday Dates
Perhaps make the dates a database entry? The fixed dates would stay the same each year and you would adjust only the floating dates. Or, there are really few holidays in the year. (Unless you are a government or a bank) Simple intercept code in the dialplan would handle most businesses. Just write 5-10 cloned lines of date traps in the code to pass the calls or send them to a closed handler. That is less disk/system intensive that doing a disk access, except they would likely be in cache anyway. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Myles Wakeham Sent: Thursday, December 31, 2009 9:46 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Dialplans Holiday Dates I have a working dialplan for our phone system with Mon-Fri, business hours identification, etc. But what I'm lacking right now is support for company holiday dates. What I'd like to do is to create a database of these dates and just update them as new years rollover. I suspect others have done this sort of thing with Asterisk before, but I've not found any resources so far. Does anyone have a suggestion as to how to approach this? I'm running Asterisk 1.4.2. Thanks Myles -- === Myles Wakeham Director of Engineering Tech Solutions USA, Inc. Scottsdale, Arizona USA http://www.techsolusa.com Phone +1-480-451-7440 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
Watch the calls on the console. Try both ways. Document what you see and your codec settings on both the phone, and sip.conf. You may have to tell the phone that the only codec it can use is G.729, don't just make that first choice. Make it the only choice. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... I thought I already did that - which is how they now get some (but not yet all) of their calls on G.729. scratching head Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
And tell Asterisk that G.729 is the only codec for that number as well! Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
Sorry, I can't resist. How do I join the Mail List Nazi Corp? Do I have to be invited, or can I just self appoint myself? Asking neophyte questions are objected to by some, top posting by those who blast others, etc. How about leaving member chastisement to the sponsor of the list? Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Nearly everyone can be criticized for something, Asking dumb questions, top posting, bottom posting and leaving 3 pages of crap to scroll through, answering questions that were answered 5 posts down, because they didn't review the newer messages before posting, and more. Be charitable and kind. Have a nice day. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Howes Sent: Friday, November 13, 2009 9:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: hi Dan Its not just you mate. He's doing it to everyone, and sadly the list server is too clever to accept forged unsubscribes.. Steve On 13 Nov 2009, at 15:22, Dan Journo wrote: Please stop emailing me personally. If no one replies to a post, it means that everyone is busy or they think you should read through the documentation before posting. If you can't figure out simple things like Music on hold from the documentation, then i dont think VOIP is for you. -Original Message- From: aster...@opensourcesolution.in [mailto:aster...@opensourcesolution.in ] Sent: 13 November 2009 09:18 To: Dan Journo Subject: hi Dan Hi dan, sorry for sending u personal mail. i am a beginner in asterisk, i had configured a minimum dial plan in which i had made two extentions n made call between two extentions via soft phone (X- Lite). now i am begining with CALLER -ID, MUSIC ON HOLD, QUEUE.plz if u have any good link or documentation than share it with me. Regards, Pawan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
Slightly paraphrasing a very old and wise saying: Give a man a fish, he eats for a day. Teach him how to fish, he eats for a lifetime. -- JohnM I see no teaching, just no help. He doesn't eat today or tomorrow either. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FW: hi Dan
My point was the two previous posters could have ignored the request and made no post at all. That they were violating a rule by top posting to tell a person not to bug them. And, someone criticized me for an off topic post and of course there have been 15-20 more. And some have top posted and interleave posted, and etc. And, it will all die down in a day or so. It is Friday night, time to turn off the computer and click Mark all as read on Monday morning. Be charitable and kind. Have a nice weekend. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards Sent: Friday, November 13, 2009 4:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] FW: hi Dan On Fri, 13 Nov 2009, Cary Fitch wrote: Sorry, I can't resist. Next time please try harder. How do I join the Mail List Nazi Corp? And of course, lacking any sense of history, let's blame it on the Nazis. Asking neophyte questions are objected to by some, top posting by those who blast others, etc. Not at all. If you had any sense of Pawan's history you may have chosen sides differently: Date: Tue, 27 Oct 2009 09:34:18 + Subject: [asterisk-users] Installing Asterisk Pawan states he is reading an Asterisk book and requests suggestions on which OS he should use. He received helpful responses from Dan Journo, PATRICK KANGETHE, John Novack, and Hans Witvliet. 30 minutes later he posts a brilliant tome Subjected installing consisting of 2 words -- installing asterisk. He received less than helpful responses from Steve Howes, Alex Balashov, and Pascal Bruno. Date: Wed, 28 Oct 2009 14:07:30 + Subject: [asterisk-users] deploying asterisk Pawan states he had just finished the installation requirement of asterisk and now feels competent to piss off 40 executives with his first installation. He received helpful responses from Danny Nicholas, Darrick Hartman, Steve Edwards (me), and Alex Balashov. Date: Mon, 02 Nov 2009 09:37:42 + Subject: [asterisk-users] hardware requirements for asterisk Pawan request help with hardware requirements. Curiously, he implies that he can read and has just finished my chapters of asterisk. He receives helpful responses from Alex Balashov and Hans Witvliet. Date: Fri, 06 Nov 2009 04:33:09 + Subject: [asterisk-users] asterisk,libpri,zaptel Pawan requests help installing Asterisk. Date: Fri, 06 Nov 2009 17:08:02 + Subject: [asterisk-users] problem while compiling asterisk tar file Pawan requests help in compiling gtk. He receives helpful responses from Jimmy Godbout, Danny Nicholas, Steve Howes, Jason Parker. Date: Sat, 07 Nov 2009 17:29:57 + Subject: [asterisk-users] help in installing asterisk Pawan requests help in compiling Asterisk. Date: Sun, 08 Nov 2009 06:20:46 + Subject: [asterisk-users] how to check version of asterisk Pawan requests help to determine the version of Asterisk he installed. He receives helpful responses from Alex Balashov, Tzafrir Cohen, and C. Savinovich. Date: Mon, 09 Nov 2009 17:11:47 + Subject: [asterisk-users] how to configure softphones in asterisk Pawan requests help configuring a softphone. He does not indicate that he has done any research, tried anything or received any error messages. He receives helpful responses from Matt Riddell and Danny Nicholas. He receives less that helpful responses from Alex Balashov, Steve Howes, and C. Savinovich in response to emailing them privately. Date: Tue, 10 Nov 2009 18:16:50 + Subject: [asterisk-users] how to configure softphones in asterisk Pawan solicits help configuring a softphone. He does not indicate that he has done any research, tried anything or received any error messages. He receives helpful responses from Alex Balashov and Barry L. Kline. Date: Thu, 12 Nov 2009 06:31:35 + Subject: [asterisk-users] soft phone (X-lite) not able to register with asterisk Pawan solicits help configuring a softphone. 20 minutes later he posts the same request. He receives a helpful response from ABBAS SHAKEEL. Date: Fri, 13 Nov 2009 08:47:41 + Subject: [asterisk-users] CALLER-ID, MUSIC ON HOLD, HOLD, QUEUE Pawan indicates he has succeeded in placing a call between 2 extensions and now wants someone to complete is dialplan. He receives a helpful response from Leif Neland and less than helpful responses from Steve Howes and Steve Edwards. He also invites a flame-fest by soliciting help privately from several list members. All this in the last 2 weeks. Some people have no one within 250 miles of where they are to learn from or learn better by working with code than reading inscrutable examples from different versions, and other inanimate pages of examples that have wrong variables, etc. Distance is no defense to ignorance. If you have the ability to email, you have access to all the resources you need
[asterisk-users] POTS 4K linear codec
I am not sure what the problems are and the reasons for the basic 64K modems used in VOIP are. I understand the compressed codecs that get the bandwidth down to 20-30 K. And perhaps the 64K units give much better potential audio than you would get on a normal POTS line. But, as phone circuits VOIP/SIP doesn't seem to perform as well as plane old phones. Multiple transcodings cause issues. Today a cell phone or a POTS line phone can send DTMF clearly enough to operate a credit card or other interactive tone based system at the far end. With SIP it is sometimes chancy. Is there a plain 64K codec that would simply pass through the SIP server and be handed off to a PRI or phone co. trunk on a T1 on the other side of the SIP server? Digital 64K telco sounds very good as a phone conversation. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Number Portability
Your chances are likely slim to none. But good luck. First to port numbers you have to be a recognized carrier, which for the most part means getting numbers from NANPA : North American Numbering Plan Administration. To do that you have to be certified by your state PUC or be a CMRS (cell phone) carrier. They would give you a block of 10,000 numbers designated to the rate center of the ILEC in question. Then you designate on of those numbers as a local routing number (LRN) which is like a pathfinder number for ported numbers. And, you work out an Interconnection agreement with the local Telco (probably with them kicking and screaming for months or a year) because they really don't want you there, and you aren't a big cell phone company, but a local wire line competitor, which then is approved by the state PUC. What some others have done is to operate as a PBX Service provider or some other business term. They get a PRI from the local company, and become the agent for the customer, move the service delivery to their PRI, and then distribute the calls to the appropriate customer via SIP and Asterisk or other solution. That has worked in Casa Grande, AZ for one place. (Not ours.) Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Saturday, October 31, 2009 8:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OT - Number Portability Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural monopoly. From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them, and a set of DID numbers we purchase from them. Do we have to be a full CLEC to participate as a carrier? Does this imply we must have an SS7 connection to the PSTN? Thanks for any info, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - Number Portability
Two more comments. Yes, to join the PSTN call distribution system you must have SS7. While rural ILECs are not exempt from number portability, there is a court injunction that saves them from having to transport the call out of their local rate center, so getting calls from a distant RILEC to a central point is at a cost to the requesting carrier. There are other complexities. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Saturday, October 31, 2009 8:34 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] OT - Number Portability Sorry for the off-topic, but perhaps this will be of interest to other asterisk based ITSPs. We are starting service in a rural area where the ILEC has the rural monopoly. From what we have read in the FCC docs this does NOT exempt them from number portability, but what does it take for us to qualify to receive their numbers? To date we simply have a few voice trunks to them, and a set of DID numbers we purchase from them. Do we have to be a full CLEC to participate as a carrier? Does this imply we must have an SS7 connection to the PSTN? Thanks for any info, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Grandstream 2010
The Grandstream 286s automatically re register when a connection is restored. Our Grandstream 2010s don't. Does anyone know of a setting that makes them reregister? I has tweaked Watchdog timer and anything that looked promising. Cary Fitch Affordable Telecom ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can dial long distance but not local?
Perhaps send it as 10 digits or 1+? Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Wednesday, October 07, 2009 3:58 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Can dial long distance but not local? AsteriskNOW 1.4.26.2 with a Digium TE205P connected to an ISDN PRI (single span). I'm sure I just have something goofed up in the dialplans? I have a bunch of Polycom 331 IP phones connecting to the server. I can dial the other extensions in the system fine and I can dial long distance outgoing but cannot seem to get it to dial local (7 digit) calls. I see this in the CLI: -- Executing [...@macro-dialout-trunk:19] Dial(SIP/801-09b6e498, DAHDI/g1/5551212|300|) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/5551212 -- Channel 0/1, span 1 got hangup, cause 28 -- Hungup 'DAHDI/1-1' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [...@macro-dialout-trunk:20] Goto(SIP/801-09b6e498, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-chanunav...@macro-dialout-trunk:1] GotoIf(SIP/801-09b6e498, 1?noreport) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) -- Executing [s-chanunav...@macro-dialout-trunk:3] NoOp(SIP/801-09b6e498, TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 2Cool - failing through to other trunks) in new stack Also when I check the PRI DEBUG I see an Error 28 which indicates an invalid number format. But I'm just sending 5551212, which should be o.k. I'm a newbie at this.any suggestions welcomed. -Ben- image001.gif___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voiced E-mail
Does anyone have info or starter points on how to take emails from an external POP3 or IMAP server and cause them to be voiced by Asterisk? It is our e-mail server, so we can do anything to it. My question is concept or products required to get asterisk to do the job. Text-to-voice converter? Program to strip email down to just to, from, text, special mail box, have it call user, or have user call in? Whatever anyone that has done something like this would suggest. Thank you. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New thread - SIP over VPN
Last week I did a Microsoft VPM from one XP computer to another via Verizon broadband wireless. SIP worked ok, but BLF on a Grand Stream 2010 didn't work. In addition to the VPN the phone was behind a NAT router. The phone was already set up behind the NAT Router, the only difference was to get the connectivity via Wireless VPN. There could have been some missing ports in the VPN environment. The audio was good, but there were times it lost clarity, likely to wireless bandwidth/lag/jitter issues. I decided that couldn't be my main business phone. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657
FWIW: From old, old memory, DTMF was 60 ms on, 40 ms off, way back when. With modern technology, shorter durations could work. Most phones of all types don't make a standardized tone burst but produce tones only while the button is pressed. Fast punching will produce short tones. On the other hand, a redialed number will be very well formatted. Reliability of TT data transfer for audio applications (over the phone voice mail, credit card, IVR, etc) would be better if the phones would run button pushes through the redial buffer/formatter. But they don't. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher Sent: Saturday, September 19, 2009 9:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657 On Saturday 19 September 2009 01:07:54 Rajkumar S wrote: I have an occasional problem where DTMF is not recognized, ie if clients type a digit while in menu the system does not register it. In my C server I saw a log line like this today: DTMF end '1' has duration 57 but want minimum 80, emulating on IAX2/a16-q1-9657 Is the above message an indication of this problem? How can I fix it? It isn't evidence of this problem, but it might be indicative of it. What this message says is that the DTMF lasted for 57ms, but Asterisk normally doesn't detect DTMF that lasts for under 80ms, so it is increasing the duration of the DTMF to compensate (because as a digital signal, DTMF is reliable, but when sent as audio, it might not be). What it probably indicates is that the DTMF sent to your system is _incredibly_ short, and if a DTMF detector is employed, it's possible that the DTMF audio is simply too short to be reliably detected. -- Tilghman Lesher Digium, Inc. | Senior Software Developer twitter: Corydon76 | IRC: Corydon76-dig (Freenode) Check us out at: www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
That doesn't happen on all phones. Either find a way to block that feature on the phone, or change phones for that location. I assume you don't want the user to know that phone's local number. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Monday, August 31, 2009 1:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inquiry:How to hide Caller Id Thank you for your reply . Yes , he is seeing his own number on his phone upon going off hook and before dialing any number . Can you please do me favor and confirm if it is not a feature of Asterisk that I can disable it ? Regards H.Motamedi On Mon, Aug 31, 2009 at 6:53 AM, Matt Riddell li...@venturevoip.com wrote: On 31/08/09 5:49 PM, hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Hi, If he is seeing his own number on his display before he has dialed any numbers then it is probably a feature of the phone - in which case you need to disable it there. If you're talking about an incoming call then it's different. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Help me testing this webphone atwww.VisionVoIP.com
I just tried it on 3 different numbers. Dialed as 10 digits NPANXX I was told I am sorry but you can only dial within North America..etc. C Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Zeeshan Zakaria Sent: Sunday, August 30, 2009 9:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Help me testing this webphone atwww.VisionVoIP.com Thank you everyone who tested the webphone, but I haven't got input from anybody. Most of the calls made were unfortunately unsuccessful, but I would like to know what error you got? Did the webphone stayed in Loading... state and never completed its loading, or your local firewall blocked, it, or something else happened? -- Zeeshan A Zakaria On Sun, Aug 30, 2009 at 1:52 AM, Zeeshan Zakaria zisha...@gmail.com wrote: Greetings everyone, I've been trying to make this java based webphone work for everybody visiting my website, but seems like for many users it doesn't work. In order to get a better idea what is the success rate of this webphone, I would appreciate help from anybody who could make a few calls from it within North America and if it doesn't work, send me what error you get, or if it works, tell me it sounds right, no echoing etc. I am keeping calling free for now for testing purposes. The webphone is located at http://www.visionvoip.com Thanks, -- Zeeshan A Zakaria -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to hide Caller Id
A Google of that model showed a discontinued Telstra corded phone. But in any case SNOM and Grandstream phones Do show the number before you pick up the handset. I would suggest you use a Grandstream 286 voip adapter and a standard corded or wireless phone so that the caller doesn't have a display to see. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of hadi motamedi Sent: Monday, August 31, 2009 1:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Inquiry:How to hide Caller Id Sorry for mis-typing in phone type . Please be informed that the current phone type our subscribers are using is TP6000 ones . On Mon, Aug 31, 2009 at 7:24 AM, Paul Hales pdha...@optusnet.com.au wrote: I couldn't find any information on this brand of phone on the internet at all. PaulH hadi motamedi wrote: Sorry for lack of enough information . I mean my subscriber when goes off hook he will see his own number displayed on his phone . I need to disable this feature on my Asterisk .The phone type is ANABELL phone . Please do me favor and let me know how can I disable this feature on my Asterisk ? Looking forward your reply Regards H.Motamedi On Mon, Aug 31, 2009 at 6:28 AM, Matt Riddell li...@venturevoip.com mailto:li...@venturevoip.com wrote: On 31/08/09 5:24 PM, hadi motamedi wrote: Dear All Can you please do me favor and let me know how I can hide the subs number being displayed on his phone when he goes off hook ? I mean when the subs goes off hook he sees his assigned number on his phone and I need to disable this feature . I don't know from which configuration file this feature is coming so please let me know how can I disable it . You're not really giving enough information. Who sees the number? Where do they see it? What type of phone? What is a subs? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to stop IVR once system receives DTMF?
I think the IVR audio must be playing in Background mode, not Play Mode. Try that. Background means play the sound and move on to the next instruction. Play means to play the sound and after it is over, move to the next instruction. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bharath B. Reddy Bynagari Sent: Monday, August 31, 2009 8:27 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] How to stop IVR once system receives DTMF? Hi, We are trying to implement a complex business logic in Asterisk. Executing Wait_For_Digit command after playing IVR. We want to stop the IVR once we receive the digit. It is not recognizing the Digit until it completes the IVR. How can we stop the IVR once we receive the digit? Thanks BB ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] caller id problem
Yes, the issue(s) is/are: 1. The VOIP provider may be masking the callerID for their own cost allocation reasons. That is some of the issue. 2. Your Asterisk box may forward some of the regular phone line calls with their caller ID. 3. Somehow, the number you want to use may leak through sometimes. :-) What you need to do is put in a simple, absolute CallerID(num) = 3216540987 type of statement before sending the call out. Make it apply to every call no matter what. That isn't the syntax but you get the idea. Of course you won't have true caller ID then, but do you want cheap or real? Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Nathan Sent: Friday, August 07, 2009 12:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] caller id problem I'm having a weird problem with CallerIDs and I can't tell if it is a problem with Asterisk, the telco, or the VOIP provider I'm using. Basically, I am using Asterisk as a proxy for my cell phone. People call in and the call gets forwarded to my personal number. The feature on my phone allows for unlimited phone calls from one number, any time, for $7/month, so I'm saving a bundle (I use it for outgoing too). However, whenever somebody calls in and the call is forwarded to my regular telco cell number, the number is coming up different e.g. instead of 478-9987 (made up number) it is coming in as 383-6894. Since it is now a different number I am getting charged for incoming calls and my neat trick is no longer working. I'd just like to know if anybody has an inkling as to where the problem might be. I've tried to use Asterisk to set the CallerID and nothing has changed. I have called both the telco and VOIP provider's tech support and they both seem to blame the other. To make things even more strange, over the course of dozens and dozens of calls, I have twice received a call from the correct number! That is the 478-9987 number, not the 383-6894. But I have no idea what the conditions where to make that happen. Additionally, it seems that most everybody else who gets a call from the Asterisk box receives the correct number, suggesting that the problem is with the telco. But I can't be certain, and besides their tech support is no help at all. I'm running out of options and I may need to switch providers. I know this is only loosely related to Asterisk, but any help would be greatly appreciated. Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk don't detects hang-up by phone
Assuming you are connected to a regular phone line, the hang up signal from the phone line would be a break or reversal of polarity of the DC signal on the phone line. (We connect to PRIs, so our signaling is on a data channel. I assume you don't. ) The first question you need to answer is Are you getting a voltage drop or polarity reversal when the other end disconnects? Asterisk has to have a signal to respond to. Some Telcos may not give that signal. Check your phone line with a meter. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of ABBAS SHAKEEL Sent: Thursday, August 06, 2009 7:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk dont detects hangup by phone Hello I have configured TDM400P with asterisk . The problem is that when i make a call to server. and while going on it dont detects call hang up. ie i called the Asterisk server and it start playing files that i indicated to do so in extensions.conf i suddenly put down the phone. now the server must detect that phone is hangup but it dont. How can i make server to detect this -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Setting up Outgoing Trunk
Are these trunks or PRI/ISDN circuits, or phone lines? If either of the first two, the callerID sent with the call should be their ID, which should be the appropriate number of digits your area telco expects. Depending on your agreement with them, they may be supplying the number, rather than accept what you send. If your connection is phone lines they are supplying the Line Number, and you have no control over that except by strategic use of the lines, etc. Or if there is further info or questions, explain the exact details. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of kumarshantanu Sent: Thursday, August 06, 2009 10:33 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Setting up Outgoing Trunk Hello Everybody, I have a genuine problem in Asterisk setup. I have three inbound trunks in my asterisk box, everything is working fine but the only problem is when any user make an out- going call through his/her extension it goes with same number labeled on this. Can we set each of these lines to have fixed outgoing numbers like if extn: 201 make an outgoing call the recipient should get different no and if extn: 202 make an outgoing call the recipient should get different one. Please can someone help me in this. Thanks Shantanu http://sigads.rediff.com/RealMedia/ads/click_nx.ads/www.rediffmail.com/sign atureline@middle? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Modem
Yes, they changed their name to Copaco for Compania Paraguaya de Comunicaciones. It's basically the same company ruling the whole country. : Oh, like ATT and Verizon here. :-( Please pardon the editorial comment, list. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] 800 number portability
There are national number rental agencies that lease out prime 800 numbers even down to the rate center level. They own the number, not the renter, and there is a contract that says so. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Thursday, July 16, 2009 7:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 800 number portability On Thu, 16 Jul 2009, Don Kelly wrote: Changing toll-free RespOrgs (Responsible Organizations) is different from number portability. That said, the owner of a toll-free number has the right to change RespOrgs, so the question is Who is the owner? The owner in this case is CallSource (www.callsource.com). Funny enough, it looks a lot like the kind of stuff you do, Don ;) So I guess my disconnect is that a party can own an 800 number, but have it routed by the RespOrg of their choice? In this case my client must be renting the 800 number from CallSource, and they are the actual owner, so are refusing to let it go. Does that sound right? Has your customer been buying simple toll-free service and owned the number all along, or are they buying some sort of enhanced service and the provider owns the number? I assumed it was simple 800 service (and in fact at first they told me it was ATT they were getting the service from). It seems that this is actually something enhanced. Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Error
Does anyone have any light to shed on: c_avpair_new: unknown attribute sip02 asterisk[7796]: rc_avpair_new: unknown attribute 1490026597 We are getting congestion errors on a Pri to telco, and not sure what is going on. Thanks Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error
Thanks, we agree.. have reset PRI on telco end and rebooted here and trouble cleared... for a while anyway. Our PRI card seems to have issues. CF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, July 14, 2009 11:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Error Cary Fitch wrote: Does anyone have any light to shed on: c_avpair_new: unknown attribute sip02 asterisk[7796]: rc_avpair_new: unknown attribute 1490026597 We are getting congestion errors on a Pri to telco, and not sure what is going on. Doing a google search gave an indication that it's a max connection error, but they were talking about ppdp http://osdir.com/ml/network.poptop/2004-04/msg00095.html Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calling non-extension numbers issue
One to few X's for that number? Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Monday, June 29, 2009 10:10 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calling non-extension numbers issue Your pattern appears to be set up in anticipation of a leading digit. What happens if you dial either 17706743900 or 97706743900? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Monday, June 29, 2009 9:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling non-extension numbers issue This is the configured pattern for local calls - _NXXNXX When I dial a local number from the device, I dial a number like 7706743900 and select internet call. Does what I dial not match the pattern? Danny Nicholas wrote: IMO, it is indeed connecting differently. When you dial an extension, say 1000, you get a match connection. When you dial the local number, you go to a different segment of the dialplan. You say that other softphones connect to local numbers correctly; have you checked things like truncation, pattern matching, etc. ? _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kayton Sapale Sent: Monday, June 29, 2009 9:35 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calling non-extension numbers issue That's the strange thing. Nothing shows when monitoring the service in debug. On the phone, however, I do see a connection time-out error. I guess this might indicate that the device is attempting to connect to the service in a way different from when just dialing an extension? Geraint Lee wrote: I have used an nokia n95 with asterisk without any problems (except for the actual phone deciding to restart itself every few hours - but that's nothing new with nokias!) Are you getting anything on the CLI that might point you in the right direction when the call is attempted? CHeers 2009/6/29 Kayton Sapale ksap...@speartek.com Hi everyone, This is my first post, so apologies if I have not included all details about the issue. I am using a Nokia e71 to connect to a corporate asterisk server and am having issue with dialing. I can dial all extensions and receive all types of incoming calls. I cannot however, dial local phone numbers. When putting the service into debug, it appears that the device does not enter into configuration when attempting to dial numbers that are not extensions. The assumption being made here is that the device is the issue, as other devices - softphones, cell phones and other internet phones - do not have this same issue. Has anyone had similar issues and some guidance on where to find a solution. Our admin and I are both searching for solutions, as we are both stuck on the problem. We are currently running asterisk version 1.4.18.1 Thanks! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Newbie, Question on making a PSTN call..
I understand the desire to try, but you are trying too hard. Getting a soft modem to work with Asterisk is. like trying to push a string up a 10 foot pipe. At the least, buy an inexpensive FXO device from someone like Grandstream and use it via Ethernet to work with Asterisk. If you have greater ambitions, buy any appropriate piece of hardware and start with that. Otherwise, You are going to have a lot of string in that pipe, before you see any come out the top. You won't get help on this because no one really knows how to do it or if it will work at all. I am trying to help, by getting you to try a better way. Good luck. Cary Fitch _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shiva Kumar Sent: Tuesday, June 16, 2009 12:27 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Newbie, Question on making a PSTN call.. Need help pls..Anyone? On Mon, Jun 15, 2009 at 10:58 PM, Shiva Kumar shi...@gmail.com wrote: Hello Asterisk-users, I am new to Asterisk. I got SIP Calls to work between two computers using a soft phone and asterisk in the middle. Since then, I have been trying to get my soft phone to make a PSTN call with terrible failure for about two days now. On Windows using asteriskwin32: I have a soft modem (Agere HDA Modem) in my laptop. Windows' default dialer is able to make a PSTN call by connecting the Phone's RJ line into my laptop's RJ 11. I am unsure what drivers to choose where and what parameters to change in tapi/fx configuration files etc. to get asterisk to use this modem to call out. Read plenty of articles about how asterisk cannot make a good phone call using a half duplex modem. But, This is for experimental purposes and I will be thrilled to just get my phone ringing before I go out to buy specific hardware. On my Ubuntu: Next up, I connected my phone(NOKIA N73) to my computer, ensured that I am able to connect to internet on my ubuntu. wvdial works good too. Again, I am unsure how to get asterisk to connect to this modem so that I can use my soft phones to make a call. Need help. Thanks in Advance. -- Shivku, http://blog.shivku.com -- Shivku, http://blog.shivku.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ANI
When Asterisk sends a call to a phone company via a PRI/Dahdi, does it actually send CALLERID(ANI), or only CALLERID(NUM)? Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users