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
