Re: [asterisk-users] No matching peers message has gone (1.8.23.1)
Hi Ish,
I assume you are using Fail2Ban to monitor the logs for dictionary attacks - If
so, the following regex should work for 1.8:
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Wrong password
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - No matching peer
found
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Username/auth name
mismatch
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Device does not
match ACL
Registration from '.*' failed for 'HOST(:[0-9]{1,5})?' - Peer is not supposed
to register
-
Regards,
AJ Stanfield
t: 0161-850-4001
e: a...@dmcip.com
w: http://www.dmcip.com
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, 4 November, 2013 3:36:06 PM
Subject: Re: [asterisk-users] No matching peers message has gone (1.8.23.1)
Hi
Thanks for the quick response. I'll read all the change logs from now on, I
promise!
Ish
On 4 November 2013 15:29, Joshua Colp jc...@digium.com wrote:
Ishfaq Malik wrote:
Hi
Ever since we upgraded our asterisk servers to 1.8.23.1, we no longer
get the 'no matching peer' error when we get a dictionary SIP attack.
Now the logs always show a 'wrong password' when there actually isn't a
matching peer.
We even have alwaysauthreject = yes in our sip.conf.
Has anyone else noticed this phenomenon?
This is on purpose. To fix some exposure issues the code was changed to have an
internal peer (albeit one that can never successfully be authenticated against)
that gets used if no real peer is found. This reduces the chance (by a lot) of
the code exposing information in some off nominal cases.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] Linksys PAPT2
Log into the Linksys GUI, Look at what SIP account it is registering to asterisk with then run sip show users ? - Original Message - From: Hassan Abdalla hagga...@gmail.com To: asterisk-users@lists.digium.com Sent: Saturday, 26 May, 2012 6:08:59 PM Subject: [asterisk-users] Linksys PAPT2 Hello people, We have 4 asterisk server acting in which 2 are running as gateway, the problem that i am facing is not asterisk related, we are using linksys PAP2T firmware 5.1.6 5.1.8, as gateway to some GSM providers in Africa, we have now reached the point where we must put credentioal into asterisk directly rather than using ATA with analoge cards. hence the QTY of ATA and our needs are growing. We have every possible available soluation to find the SIP passwords inside linksys PAP2T without joy, we have used various asterisk-password decrypters but all failed. any idea will be helpful, Thanks in advance, Regards Hassan -- _ -- 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] Digium IP Phones
We've just had one of each delivered for us to play with in our lab (Literally an hour ago!). Not had chance to play with them yet, But initial thoughts are they look good. Build quality seems fine for the price. I'll form more of an opinion when i get chance to play with them properly tomorrow. I don't think the SDK is available yet (I've not been able to find it on the digium site). I'm itching to get my hands on it though! My first thought when seeing the D70 and looking at the screen for the speed dial keys was I hope we can use this screen in for the apps, It's perfect for a tetris clone. :) Cheers, AJ. - Original Message - From: Danny Dias ing.diasda...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, 10 May, 2012 2:38:02 AM Subject: [asterisk-users] Digium IP Phones Hello, Im looking to buy a digium phone D70 unit just for testing on lab; to really understand the phone and features. I cant find any website with opinions; any here? Are they really valuable to the price? (D70 quite expensive) Does the SDK for building apps is usable? Can you build powerfull apps? Examples? Many thanks -- _ -- 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] Fwd: Flashphoner
Boo, and i felt so special for a few minutes this morning! :( -- AJ [YOUR AD HERE] - Original Message - From: Steven Howes steve-li...@geekinter.net To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 27 April, 2012 9:28:18 AM Subject: [asterisk-users] Fwd: Flashphoner Thought this deserved a name and shame! ;) Steve Begin forwarded message: From: Pavel Ismailov pavel.ismai...@gmail.com Date: 27 April 2012 06:58:07 GMT+01:00 To: steve-li...@geekinter.net Subject: Flashphoner Hello! My name is Pavel Ismailov and I`m CEO of www.flashphoner.com project. We noticed that you quite active in Asterisk-user mail list, and would like to offer you buy signature in your messages for some monthly price. Is it interested for you? -- Thanks, Pavel Ismailov skype: pavel.ismailov www.flashphoner.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] Experience with virtual servers?
Hi Binni, We run a number of Asterisk servers on virtual machines. I'm not heavily involved in the virtualisation side of the business so i'm afraid i can't give you much advice on it, Past saying it is possible to have an Asterisk System up and running reliably on virtual machines. Our virtualisation platform is KVM based. Hopefully someone with more knowledge than me will be able to help!. Cheers, AJ. - Original Message - From: Brynjolfur Thorvardsson bi...@itanet.nu To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, 20 April, 2012 1:51:03 PM Subject: [asterisk-users] Experience with virtual servers? Hi All Does anybody have experience with running Asterisk on virtual servers? I have been experimenting with two suppliers and I am not altogether happy with sound quality etc. Is it perhaps foolish to try and install a “production” Asterisk server on a virtual machine? With dedicated servers being comparatively cheap (although still several times more expensive than virtual servers), perhaps that is the way I should be going? I have heard someone mention “Asterisk friendly” VPS providers, how can you tell if they are or aren’t friendly? We currently have our Asterisk server running on a five year old single AMD CPU 32 bit machine with 512Mb and that works fine. Even the cheapest virtual server vendors offer servers that seem much more powerful but after testing I am not so sure any more! Any info would be very welcome! Regards Binni -- _ -- 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] Caller ID problem
Hi Anam, Hope this helps explain Asterisk version numbering: http://leifmadsen.wordpress.com/2011/08/29/asterisk-10-asterisk-1-hh10/ Easy to get confused!. Cheers, AJ. - Original Message - From: Satria Anamarta anam.satri...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 16 April, 2012 12:10:27 PM Subject: Re: [asterisk-users] Caller ID problem Thanks Danny. I test it with blind transfer and hey, you're right, the caller ID passed successfully, but the attended transfer doesn't. What version did you refer to by saying 10.x ? Asterisk? Shoudn't current version of asterisk is 1.x and should move to 2.x instead of a big jump to 10.x ? Thanks :) BR, Anam Totally newbie On 4/16/12, Danny Nicholas da...@debsinc.com wrote: Do a blind transfer instead of attended transfer - the under the covers changes in 10.X handle this for attended transfers, but to the best of my knowledge, the blind transfer is the only solution in the 1.X tree. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satria Anamarta Sent: Sunday, April 15, 2012 10:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Caller ID problem Hi, I'm running asterisk 1.8.7.0 FreePBX 2.8.1 IP Phone Yealink T20 Trustrpid and sendrpid is on the sip.conf Let say I pickup a call on ext A using *8, the caller's caller ID successfully passed to my phone. I decide to pass the call to ext B. On phone B, it display ext A not the original's caller ID. I want on phone B it display the caller's caller ID. Is there any solution for this? I already googling this for around a week but found no solution yet :( Thanks and BR, Anam -- Sent from my mobile device -- _ -- 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 -- Sent from my mobile device -- _ -- 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] cross ivr is comming in my ivr system
- Original Message - From: A J Stiles asterisk_l...@earthshod.co.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 4 April, 2012 10:48:02 AM Subject: Re: [asterisk-users] cross ivr is comming in my ivr system On Wednesday 04 April 2012, Jagadish Thoutam wrote: hi all, i have gradwell DID i am using it for inbound dialing with IVR when ever customer call my DID some times other IVR is cumming on my IVR that IVR is not even related with my server .can u please help me on this Sorry. This list is only for questions that make sense. -- AJS Answers come *after* questions. Not everyone who comes here is going to speak English perfectly as their first language. Taking snide little digs at someone because of their English skills is not what this userlist is about and doesn't benefit the community in the slightest (If anything, It damages it.) For what its worth, I understood his question entirely. He has a DDI provided by Gradwell, Which when dialed leads into an IVR (I assume running on an Asterisk server, Hence why he's posted the question here.) Occasionally this number is hitting an IVR system that is not his own (Upstream Call routing issue?). Jagadish, When the unknown IVR is being played are you seeing any traffic at all through the Asterisk CLI (Or in the logs?) - Does the call hit the server at all?. Which Gradwell service are you using, And how are you connecting it to your server? Which version of Asterisk are you using?. Cheers, AJ. -- _ -- 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] [NAT] SSH vs. OpenVPN?
Hi Gilles, You can't tunnel UDP through SSH. Some of the newer Grandstream handsets support OpenVPN and are a bit cheaper than the Snom alternatives. - Regards, AJ Stanfield t: 0161-850-4001 e: a...@dmcip.com w: http://www.dmcip.com - Original Message - From: Gilles codecompl...@free.fr To: asterisk-users@lists.digium.com Sent: Tuesday, 31 January, 2012 12:32:20 PM Subject: [asterisk-users] [NAT] SSH vs. OpenVPN? Hello In case a NAT firewall prevents using STUN to open SIP/RTP ports, a solution is to first connect the phone to the Asterisk server through a tunnel, and then have data go through the tunnel. Are there hardphones that support OpenVPN? If none, what about SSH? Is this a good alternative to use VoIP with SIP? If you've tried either or both solutions, I'm interested in any feedback. 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 -- _ -- 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] Peer doesn't answer
Hi Arlen, I'm interested in seeing what setup you settled on to get decent voice quality over the Sat link? Which codec are you using, and what is the bandwidth usage?. Are you doing just one concurrent call, Or multiple?. - Regards, AJ Stanfield - Original Message - From: Arlen Nascimento arlen.nascime...@gmail.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, 18 January, 2012 12:29:23 PM Subject: Re: [asterisk-users] Peer doesn't answer Hi guys, the problem was too many NATs on the way. Although the server had a valid ip, it was behind a nat, as soon as I set ip directly on the server, things worked fine. Also, despite the huge delay, if the link has qos, the quality is very good. On Mon, Jan 16, 2012 at 9:06 AM, Sammy Govind govoi...@gmail.com wrote: I'm only expecting NAT issues if not the latency issues. SIP traces of any such calls will make more sense. On Mon, Jan 16, 2012 at 6:05 PM, Arlen Nascimento arlen.nascime...@gmail.com wrote: the client is aware of the adverse environment and this is the only solution for him On Mon, Jan 16, 2012 at 9:00 AM, Flavio Miranda flaviormira...@hotmail.com wrote: Unless you are doing test with SIP under adverse environmet, that is not the point, but, if you intend to have Communication, you should worry about this detail. Basic infra-estructure is the first thing to think in any new project. Good luck! Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Mon, 16 Jan 2012 07:58:34 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Peer doesn't answer It is a satellite connection, so ping is about 500ms. I know it is not ok to keep a normal conversation, that is not the point. On Sun, Jan 15, 2012 at 10:34 PM, Flavio Miranda flaviormira...@hotmail.com wrote: Hi Arlen, A reasonable time to Voip calls is about 250 ms. What about the Ping test end-to-end ? Att, Flavio Roberto Miranda MSN:flaviormira...@hotmail.com Skype: flaviormiranda Date: Sun, 15 Jan 2012 21:53:46 -0400 From: arlen.nascime...@gmail.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Peer doesn't answer Hi all, i'm implementing an asterisk server that will have several peers connected by satellite links. When qualify=yes or some value (from 3000 to 5), 'sip show peers' shows the peer as unreachable. In this case i can place calls from the phone in the satellite link, but can't call to it. When i turn off qualify, the status changes to unmonitored. In this case, I can make calls in both directions but the call is never established. The phone keeps ringing until 'ring time' expires even when I answer the call on the phone/softphone. Any thoughts? Regards, -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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 -- Arlen Nascimento -- _ -- 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:
Re: [asterisk-users] vigor 2920 problems
Hi John, We've had similiar issues with customers behind the 2920 connecting to a hosted asterisk system. If you rebooted a phone it often didn't re-register, Checking the NAT sessions table on the router revealed stale nat sessions open for the phone. On the advice of Dreytek we found a fix by lowering the NAT session timeout from the default of 24hrs down to 5 minutes and installing the latest release of the firmware (3.3.7) it may not be available on the UK Site at the moment (It wasn't when we did the upgrade!) but it can be got from ftp://ftp.draytek.com/Vigor2920/Firmware/v3.3.7/ It may help, It may not - But its quick easy fix if it does. Regards, AJ. - Original Message - From: John Taylor j...@vetsurgeon.org.uk To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, 21 November, 2011 10:20:14 AM Subject: [asterisk-users] vigor 2920 problems One of our clients has a Draytek Vigor 2920- their natted Snom phones behind it are registered to an Asterisk 1.4 server on an external public IP. I've set QOS, bandwidth management and turned off the SIP ALG via telnet but I'm still having some problems with some of the phones losing registration if Asterisk is restarted. I can see the phones sending SIP REGISTER messages, but they never arrive at the server; this happens in about half of the phones- with no consistency as to which lose registration. It looks like the router is swallowing the messages, or there's some kind of NAT problem. Other clients at other sites are fine. The problem clears if the phone is rebooted (renegotiates a new nat path?) Any help warmly appreciated. 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 -- _ -- 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 - CRM Integration
Wow, what a disaster of an open source project. Install docs are impossible to use. Many, many inaccuracies. i think you just need someone set it up for you ... think of it as an air conditionning system, you can use it but can never install it on your own unless you're from the field. ___ -- 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] Asterisk - CRM Integration
No. The problem is the docs are all wrong on Covide's project. The web site says one thing, the readme another. Neither are correct. well you may be correct but we must admit one thing, it takes a lot of dedication to start continue a real project ... and only for that every developper must get all of our respect. ___ -- 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] need a monitor for asterisk
is registerattempts=0 in your sip.conf ? ___ -- 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] Asterisk - CRM Integration
Do you plan to use it in a call center or casual business office ? Call Center I still think your best bet is vicidial. ___ -- 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] Asterisk - CRM Integration
vicidial ... vicidialNOW ___ -- 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] Manual Wardialer
can tell you that it is an actual asterisk application (with a conf file) and html for the web interface (and obviously a database). I'd like to start something similar with python/Mysql under the hood a web page at the front end (ajax) ... but so far my time does not help. I wonder if there are any projects like this around. I believe the most delicate part would be interfacing with asterisk ? ___ -- 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] Manual Wardialer
exten = wardial,1,NoOp(wardialing: i didn't mean i wanted a wardialer ! i meant a simple predictive dialer (that is no rich features, only to be able to make transfers internaly externaly to show data of called lead to agent) ___ -- 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] Manual Wardialer
Please contact me off-list if you would like to test out my beta i'd be glad to do that. It is connected to a TDM PRI. Where are you calling? I pay a penny a minute but would be glad to eat that cost for real world testing. I also use PRI links only ... the thought of testing a wardialer makes me feel like the bill is gona be very fat. but maybe we could run the tests using sip accounts ... at least you know how much minutes you can afford. ___ -- 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] (no subject)
http://www.soft-switch.org/unicall/mfcr2/ch02.html ___ -- 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] (no subject)
Make sure you get a helpful tech on the phone. Many times they will just dismiss you with we cannot do that even though they may be able to. i always say if you pay your bills you should get the support you diserve. every provider is almost always willing to help out his clients if they express their needs with precision. one more thing : nothing compares to having a friend working at the providers company so get yourself one. Again, a reply to my reply. Note to self: stop hitting send before completing thoughts. you shoudl add something like this to your base code .. if finish-email == 'yes': keyboard.hit(enter) else: keyboard.write(text) :) ___ -- 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] Dell 1950
i suggest a look at digium's hardware compatibility list on thier website. i would also worry about concurrent calls thus concurrent recordings, 48 with your actual card which i guess is acceptable load to your hardware but i am not an expert in this. recording to disk (even scsi ones) will make your server unstable when lots of calls are being recorded, then your option is to record to RAM (or go for OrecX). then empty RAM to disk when load is low since you've got plenty of RAM. gsm conversion will lower the size of wav recordings by 10. ___ -- 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] PRI hangup certain outgoing calls
I can dial out to other numbers without issue. Calling the number from a separate PSTN phone works fine. I have had such problem last week. from PRI interface (i get busy tone as fast as i finish typing last digit hit dial) from analog interface I can make the call without problem. my provider never could solve it (well so far) I myself never could understand what it could be. is this a special number (call center or something) ? because mine is a call center number ... so maybe they had a wrong setup on thier servers or something. ___ -- 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] Manual Wardialer
you saying : You make it sound as if only one person would be dialing one number at a time he's saying but I want to be on the line and if possible complete / talk on certain calls. ___ -- 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] Manual Wardialer
some people use a war dialer to provide call centers with numbers for their campaigns ... if number called rings the number is valid if it doesn't its invalid discarded. i wonder if that is legal .. its basically a scan of the network for valid numbers (that is potential buyers). i once was contacted by a company who offered this kind of service but i didn't trust them ... the numbers without names is ugly. ___ -- 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] Manual Wardialer
I will be releasing a dialer under the GPL shortly that has a very low profile, AJAX web interface, and will be able to do just what you want. what programming language will you use under the hood ? ___ -- 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] Need comments on CRM development / Asterisk Customization
Vicidial has call center / CRM integration with asterisk ... with many years of bug reporting ... it open source. On Sat, Apr 26, 2008 at 12:11 PM, Kashif Naeem [EMAIL PROTECTED] wrote: Hello All, A company has two requirements: 1) They are looking to develop its own CRM 2) Second thing is that they want to develop enhancements / new features in Asterisk like Thirdlane. What are your comments about technology to be used. Which one would be most beneficial in future ? PHP, JSP, ASP ? Can anyone suggest existing easy and generic CRM ? Regards -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: [EMAIL PROTECTED] MSN: [EMAIL PROTECTED] Gmail: [EMAIL PROTECTED] Skype: kashif.naeem 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. ___ -- 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] yum install for specific kernel, how? And zaptel on fedora core 8
I am facing the problem that zaptel is not going online when booting, most people run it from /etc/rc.local ... ___ -- 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] yum install for specific kernel, how? And zaptel on fedora core 8
Install kernel-version.src.rpm with the following command: rpm -Uvh kernel-version.src.rpm this is from http://fedoraproject.org/wiki/Docs/CustomKernel ubuntu is better/safer/faster/has 5 year of updates ... you name it. On Sat, Apr 26, 2008 at 4:31 PM, bilal ghayyad [EMAIL PROTECTED] wrote: Dear All; If i need to download the source for the kernel 2.6.23.1-42.fc8-i686 (i do not need the latest one, i need this kernel specifically), what the command to be used? Also, any one tried to run zaptel 1.4.10 + asterisk 1.4.19 on fedora core 8? I am facing the problem that zaptel is not going online when booting, I tried a lot of solutions but did not work. Any one has idea? Do I need specific kernel to be used to work with zaptel 1.4.10? Any help? Regards Bilal Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ ___ -- 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] yum install for specific kernel, how? And zaptel on fedora core 8
I thought most people ran it from /etc/rc.d/init.d/zaptel here is what README file says : Installation Note: If using `sudo` to build/install, you may need to add /sbin to your PATH. -- make make install # To install init scripts and config files: #make config -- if you don't run make config you will be using /etc/rc.local ___ -- 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] yum install for specific kernel, how? And zaptel on fedora core 8
By what criteria do you form the opinion that Ubuntu is better, safer, and faster? well I don't know what's your favourite but compared to fedora ... i guess it is. ___ -- 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] yum install for specific kernel, how? And zaptel on fedora core 8
By what criteria its a like a car you've got to drive it to feel it.. so give it a shot ___ -- 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] Manual Wardialer
if wardialer is the correct term this must be a predictive dialer ... which is simply a dialer that dials a list of numbers you supply to him ... then you need to configure if for when to pass the call to you when to hung up ... vicidial is a good project to start with ___ -- 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] Quality problems with ISDN PRI
D is for Disturbing other poeple. On Sat, Apr 26, 2008 at 10:39 PM, Benny Amorsen [EMAIL PROTECTED][EMAIL PROTECTED] wrote: Steve Totaro [EMAIL PROTECTED] writes: But then that gets back to my Intel C2D show as two procs. 2 x 2 = 2. Or is C2D not four cores? D is for duo. /Benny ___ -- 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] Quality problems with ISDN PRI
I still hope someone would enlighten us by his experience in doing call recordings without recording to RAM Drive. ___ -- 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] ATA FXO / FXS - can forward to sip ?
i can only think of an asterisk box the right dialplan. ___ -- 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] Quality problems with ISDN PRI
he's capturing the audio at the network layer i'd better stay with my 3Gigs RAM drive ___ -- 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] Quality problems with ISDN PRI
There are much better solutions than doing a RAM drive. While it may be stable (not in my experience, I advise using different servers for different tasks (with redundancy obviously). A phone switch should be just that, a recording server should also be just that (in demanding environments). hi, still hoping you will give us some insight about remote recording server. ___ -- 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] Dell SC1430 + Digium TE110P = Digital Noise in PRI
The Digium cards are known to steal IRQ's. The Sangoma cards do not. Arthur Miller Sr. Sales Associate VoIP Supply, LLC. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. ___ --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] OT: DELL Platforms
Hello list, I have a customer who is interested in standardizing on dell servers for asterisk deployments. Has anyone had success with a particular configuration? Anything specifically to watch out for? Thank you for your time, Art Arthur Miller Sr. Sales Associate VoIP Supply, LLC. 454 Sonwil Drive Buffalo, NY 14225 716-250-3871 OFFICE 716-630-1548 FAX [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. ___ --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] NAT/Qualify/RTP bug
Got a really wierd problem her. Maby it's a bug. But before i report it, i'll try my luck here. I have one asterisk server on public ip. I have two identical hardphones on two different LAN's. The firewall are different. Both are configured in asterisk with nat=yes and qualify=yes. For one phone everything works. SIP and audio is sent to the global address of the client. But for the other it's a bit different. SIP messages are sent to the global address of the client. You can call in and out. But the audio (RTP) is sent to the local address found in the SIP packets. The only thing that is different is the firewalls. How can a firewall, or anything else, tell asterisk to use the ipaddress in the sip packets instead of the global address, when i have told asterisk nat=yes Is this a bug? Or something i've missed. PS: i'v tried nat=route, same results Thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT question
/etc/group /etc/passwd /etc/shadow The line looks like a yp line. It tells the pam module to search the NIS server for users, groups and password On Saturday 08 January 2005 05:28, Michael Levenson wrote: Can someone help me answer this question? Where would you most likely find a file with the line +::? What does it do? I have been racking my brain a buddy of mine is testing me and I don't want him to win. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do i talk to the IAXy...? (Newbie Alert)
I think you need that provisioning tool from digium. And you need a unix system to compile and run it. I dont think theres any port to your OS. Sorry... On Saturday 08 January 2005 08:08, Daiku wrote: Hi, hoping that experienced hands will quickly show me the right way: after a fruitless web search i am turning to this list with my rather elementary question: is there any other way to communicate with the IAXy besides using special utility software that needs to be compiled under UNIX? Here is the story: about two months ago, after some not very satisfactory attempts at using SIP (my phone adapter and router don't seem to be able to handle SIP's special requirements for free passage through umpteen ports), i decided to try out hat i think is the conceptually better alternative anyway, IAX, and signed up with Diamondcards, a phone service provider using the IAX protocol. And today the post office delivered the IAXy adapter that i ordered from Digium (i had it shipped surface mail since i live in Okinawa and would have had to pay ovcer 50 bucks for air delivery). To my surprise there was no manual, not even a single sheet with instructions, in the package, but i quickly found a setup guide on Digium's website. However, i can't make much sense of the rather sparse instructions about provisioning the box (what's the difference between configuring and provisioning?). All i want to do is to tell the device my user ID, my password, and the server i want to connect to, but, as i found out at http://ruk.ca/article/2501 , while the Sipura unit has a friendly web-based configuration tool, the IAXy requires compiling a small Unix utility which is then used to provision the IAXy. Well, my two computers are Macs from the OS 7 era, so there is no way i can compile or run that utility to talk to the IAXy. And although i played around a bit with the telephone and even got the dial tone to come up once by punching some random number on the keypad, i am not optimistic that it is actually possible to configure the IAXy via the telephone's key pad - i couldn't find anything about this topic on the web. So... how can i get the IAXy to work? Thank you in advance: H. Daiku -- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Help With Configuration From Odbc
Hi. I can't figure this one out. Hope someone can help me. [EMAIL PROTECTED]:# cat /etc/odbc.ini [Asterisk] Description=PostgreSQL asterisk Driver=PostgreSQL Trace=No TraceFile=/tmp/odbc.log Database=asterisk ServerName=localhost UserName= Password= Port=5432 Protocol=7.4 ReadOnly=No RowVersioning=No ShowSystemTables=Yes ShowOidColumn=Yes FakeOidIndex=Yes ConnSettings= [EMAIL PROTECTED]:# cat /etc/odbcinst.ini [PostgreSQL] Description=PostgreSQL ODBC driver for Linux and Windows Driver=/usr/local/lib/psqlodbc.so Setup=/usr/lib/odbc/libodbcpsqlS.so Debug = 1 CommLog = 1 [EMAIL PROTECTED]:# echo select * from ast_config where filename='iax.conf' and commented=0 order by filename,cat_metric desc,var_metric asc,category,var_name,var_val,id | isql Asterisk lot of output from table SQLRowCount returns 39 39 rows fetched So the odbc thingy works! [EMAIL PROTECTED]:# cat res_config_odbc.conf [settings] table = ast_config connection = myconn [EMAIL PROTECTED]:# cat res_odbc.conf [myconn] dsn=Asterisk username=X password=X preconnect=yes [EMAIL PROTECTED]:# cat extconfig.conf [settings] agents.conf = odbc enum.conf = odbc extensions.conf = odbc iax.conf = odbc iaxprov.conf = odbc queues.conf = odbc sip.conf = odbc zapata.conf = odbc And asterisk answers: [res_odbc.so] = (ODBC Resource) == Parsing '/etc/asterisk/res_odbc.conf': Found Jan 1 02:21:11 NOTICE[32024]: res_odbc.c:133 load_odbc_config: registered database handle 'myconn' dsn-[Asterisk] Jan 1 02:21:11 NOTICE[32024]: res_odbc.c:379 load_module: res_odbc loaded. [res_config_odbc.so] = (ODBC Configuration) Jan 1 02:21:11 NOTICE[32024]: config.c:888 ast_config_register: Registered Config Engine odbc == Parsing '/etc/asterisk/extconfig.conf': Found Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: agents.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: enum.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: extensions.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: iax.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: iaxprov.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: queues.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: sip.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: zapata.conf to odbc Jan 1 02:21:11 NOTICE[32024]: res_config_odbc.c:190 load_module: res_config_odbc loaded. [skipping res_adsi.so] [skipping chan_modem.so] [chan_sip.so] = (Session Initiation Protocol (SIP)) Jan 1 02:21:11 NOTICE[32024]: config.c:764 __ast_load: Loading Config sip.conf via odbc engine == Parsing '/etc/asterisk/res_config_odbc.conf': Found Jan 1 02:21:11 WARNING[32024]: res_config_odbc.c:103 config_odbc: SQL select error! [select * from ast_config where filename='sip.conf' and commented=0 order by filename,cat_metric desc,var_metric asc,category,var_name,var_val,id] What is wrong? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help With Configuration From Odbc
Sorry about this. Just figured it out. In res_odbc.conf its supposed to be pre-connect and not preconnect. On Saturday 01 January 2005 02:30, Arthur B Olsen wrote: Hi. I can't figure this one out. Hope someone can help me. [EMAIL PROTECTED]:# cat /etc/odbc.ini [Asterisk] Description=PostgreSQL asterisk Driver=PostgreSQL Trace=No TraceFile=/tmp/odbc.log Database=asterisk ServerName=localhost UserName= Password= Port=5432 Protocol=7.4 ReadOnly=No RowVersioning=No ShowSystemTables=Yes ShowOidColumn=Yes FakeOidIndex=Yes ConnSettings= [EMAIL PROTECTED]:# cat /etc/odbcinst.ini [PostgreSQL] Description=PostgreSQL ODBC driver for Linux and Windows Driver=/usr/local/lib/psqlodbc.so Setup=/usr/lib/odbc/libodbcpsqlS.so Debug = 1 CommLog = 1 [EMAIL PROTECTED]:# echo select * from ast_config where filename='iax.conf' and commented=0 order by filename,cat_metric desc,var_metric asc,category,var_name,var_val,id | isql Asterisk lot of output from table SQLRowCount returns 39 39 rows fetched So the odbc thingy works! [EMAIL PROTECTED]:# cat res_config_odbc.conf [settings] table = ast_config connection = myconn [EMAIL PROTECTED]:# cat res_odbc.conf [myconn] dsn=Asterisk username=X password=X preconnect=yes [EMAIL PROTECTED]:# cat extconfig.conf [settings] agents.conf = odbc enum.conf = odbc extensions.conf = odbc iax.conf = odbc iaxprov.conf = odbc queues.conf = odbc sip.conf = odbc zapata.conf = odbc And asterisk answers: [res_odbc.so] = (ODBC Resource) == Parsing '/etc/asterisk/res_odbc.conf': Found Jan 1 02:21:11 NOTICE[32024]: res_odbc.c:133 load_odbc_config: registered database handle 'myconn' dsn-[Asterisk] Jan 1 02:21:11 NOTICE[32024]: res_odbc.c:379 load_module: res_odbc loaded. [res_config_odbc.so] = (ODBC Configuration) Jan 1 02:21:11 NOTICE[32024]: config.c:888 ast_config_register: Registered Config Engine odbc == Parsing '/etc/asterisk/extconfig.conf': Found Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: agents.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: enum.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: extensions.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: iax.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: iaxprov.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: queues.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: sip.conf to odbc Jan 1 02:21:11 NOTICE[32024]: config.c:1092 read_ast_cust_config: Binding: zapata.conf to odbc Jan 1 02:21:11 NOTICE[32024]: res_config_odbc.c:190 load_module: res_config_odbc loaded. [skipping res_adsi.so] [skipping chan_modem.so] [chan_sip.so] = (Session Initiation Protocol (SIP)) Jan 1 02:21:11 NOTICE[32024]: config.c:764 __ast_load: Loading Config sip.conf via odbc engine == Parsing '/etc/asterisk/res_config_odbc.conf': Found Jan 1 02:21:11 WARNING[32024]: res_config_odbc.c:103 config_odbc: SQL select error! [select * from ast_config where filename='sip.conf' and commented=0 order by filename,cat_metric desc,var_metric asc,category,var_name,var_val,id] What is wrong? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Iaxy
Hope this is the right maillinglist. I would like to know how i can secure the iaxy. Or is the the sad truth that anyone with an iaxyprov program can change any box not behind a firewall? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy playing dead again
So i guess were screwed. These unusable thingies are quite expensive. Good thing i only bought two. Now im really nervous about the isdn pri card i bought. Gonna try it out tonight. Hope its different. Is the software for iaxy open source. Then maby it can be fixed. On Wednesday 22 December 2004 20:22, Erik Espinoza wrote: As much as I appreciate the work done by Digium on Asterisk, it appears as though the IAXy is not ready for prime time. 1) IAXy has no security of any kind, anyone with iaxyprov can reprovision your device without so much as a password!!! 2) The IAXy doesn't work with regular dhcp, it uses bootp (thus never renews an address, which confuses quite a bit of dhcp servers) 3) Supports only two codecs, pcm/ulaw 4) Not configurable via http 5) No default IP (usually not a problem, if the damn thing would do dhcp!!!) 6) Cost almost twice as much as Sipura SPA-1001 I'm mentioning this in the mailing list because when I had issues getting the IAXy to get an ip from a Microsoft DHCP Server, Digium instructed me to ask in the mailing list or on irc. Digium charges quite a heavy premium for their equipment, and gives away their software. Weird how Asterisk is the coolest thing since sliced bread but their hardware is somethin right outta the trash heap. On Wed, 22 Dec 2004 11:12:00 -0600, Jay Milk [EMAIL PROTECTED] wrote: Sounds like a thermal problem -- which most intermittent problems are. Had this happen with a network switch in my home office. Pull it out, disconnect it and put it in a cool spot for a few hours. If the problem goes away, see whether you can stabilize the environmentals. -Original Message- From: Wilson Pickett [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 22, 2004 10:26 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] IAXy playing dead again It's happened before, cleared up and now it happened again. The IAXy, working for a total of about 6 months. Symptoms: Registered with asterisk and even receives calls (the LED shows it's ringing) but phones connected to it are dead. Same phones work connected directly to the phone line. Cable swapped out, no difference. Endless re-provision (with normal looking output) and power recycling. This really looks like a dead FXS - except - this has happened before and it came back. Comments? Suggestions? As a TV repairman once told me, The most obscene word in technology is 'Intermittent' ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAXy playing dead again
But whats the future of iaxy. Are these problems being fixed. Or is the whole project dropped? On Thursday 23 December 2004 01:38, Kristian Kielhofner wrote: Arthur B Olsen wrote: So i guess were screwed. These unusable thingies are quite expensive. Good thing i only bought two. Now im really nervous about the isdn pri card i bought. Gonna try it out tonight. Hope its different. Is the software for iaxy open source. Then maby it can be fixed. I hate to say this, but I have pointed out all of these SEVERE limitations in the iAXY before, and it stinks to hear them again. Overall, Digium makes very good hardware. The iAXY is definitely an exception to that. Please don't form an opinion of Digium solely off of the iAXY, as it leaves much to be desired. No DNS, bootp only, Cisco switch troubles, bad provisioning, etc. all plague the iAXY. They are really no competition when compared to something from Sipura. The only advantage they have is IAX, and that doesn't come close to making up for all of the other problems and big price difference. Sorry about that. -- Kristian Kielhofner ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Point to Point VOIP
I am looking for a setup something like the following. I have two offices, one located in the US and one in Australia. I would like to implement a solution whereby I would install a gateway in each of the two offices. When calls are made to a few numbers in the US, the calls would be routed over the gateway to the one in Australia. The gateway in Australia would dial out to a pre-defined number/set of numbers to complete the call. What is the minimum hardware/software configuration I would need to complete this sort of setup? I am relatively new to the concepts behind VOIP, so any help would be greatly appreciated. Is there anyone with a similar setup to this that has any suggestions/tips? Thanks, Jacob Jacob Arthur, MCP ATS [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Anyone else having Broadvoice Problems?
I have also been having problems today registering... I contacted them, but they have no known issues. It finally did register on it's own. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andre Normandin Sent: Wednesday, July 21, 2004 8:44 PM To: Asterisk-Users Subject: [Asterisk-Users] Anyone else having Broadvoice Problems? Suddenly my broadvoice will no longer register. It was working fine for over 1 month without a single problem, now I get a SIP registration timed out message. I called them, and I was told that they are experiencing problems, and they hoped to have it resolved ASAP. I called them at around 10 AM EST this morning. It's now 8:30 EST PM, and I still have not heard back, and the problem is not resolved. Is anyone else having problems with their broadvoice account? - Andre ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Using a DNS name for externip in sip.conf
I though that the externip was used within the sip communications, so it is sent as is, and resolved on the other side. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dennis Cartier Sent: Wednesday, July 14, 2004 9:51 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Using a DNS name for externip in sip.conf Does anyone know if the 'externip=' in sip.conf is resolved just once at startup or on an on going basis? I would like to use a DNS name through one of the dynamic DNS providers, but if the DNS updates, and asterisk continues using the old resolved value, this could get tricky. Thanks, Dennis ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
