Re: [asterisk-users] small homebrew pbx
On 15/06/15 07:46, lu...@sulweb.org wrote: Hello all, Given the requirements above, what's a cheap but working PCIe card / USB adapter I could buy for this kind of PBX? Do I need things like echo cancellation? Do I need FXS ports? Thanks in advance, Lucio. I would get hold of some lower-power hardware, that system seems hugely over-specified for what you want to do. A raspberry pi a Cisco SPA-3102 would be a good solution. Cisco don't make the 3102 any more but there are still plenty of them around. I believe Grandstream still make ATAs as well but I've never thought very highly of them. As others have said, it's an FXO port you need. You want to avoid transcoding on low power hardware such as a raspberry pi so set everything for a codec such as g711a or g711u (Asterisk, the IP phones you use and the SPA3102). -- _ -- 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] Updating to 11.7.0
On 19/12/13 17:15, David Lee (digium) wrote: On Dec 19, 2013, at 10:34 AM, Jerry Geis ge...@pagestation.com mailto:ge...@pagestation.com wrote: [snip] Looking that up, it says add to asterisk.conf [options] live_dangerously = yes After doing this, and stopping and starting I still get the message. I'm having the same issue, even with it set to yes and restarting Asterisk, it still give that warning. Whats up? You want to avoid danger, so set live_dangerously = no. I appreciate that and I do understand why but that setting doesn't work as described, it seems to do nothing. While we're at it, what's the recommended alternative method to replace using asterisk -rx in bash scripts now? cheers, Paul. Jerry -- David M. Lee Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com http://www.digium.com www.asterisk.org http://www.asterisk.org -- _ -- 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] Updating to 11.7.0
On 05/03/14 12:56, Paul Hayes wrote: I appreciate that and I do understand why but that setting doesn't work as described, it seems to do nothing. While we're at it, what's the recommended alternative method to replace using asterisk -rx in bash scripts now? cheers, Paul. Apologies for replying to my own post but I since found this: https://issues.asterisk.org/jira/browse/ASTERISK-23084 and after testing it turns out that even though asterisk -rx always shows the warning, the command is still executed so no where near as bad a problem as I initially thought. I guess the live_dangerously setting wont eventually have an affect on asterisk -rx (it doesn't really make any sense to since if someone is already in as root then you are screwed!). cheers, Paul. -- _ -- 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 Registration Hijacking
On 20/01/12 01:36, eherr wrote: It is also register on an AudioCodes MP-118. Thanks, -E Is the Audiocodes gateway accessible online? Have you set a strong password for it's web interface (and cli if it has one)? It is possible someone is breaking into that and getting the SIP password out of it. cheers, Paul. -- _ -- 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] local channels and g729a voice quality
On 16/01/12 07:59, Roi Stork wrote: I also asked my provider to test call me using their Cisco as5300 system and g729 codec and compared it with ulaw. The difference is unnoticable. ^^ this doesn't make any sense, the difference *should* be very much noticeable. g729 is a lower quality codec (in terms of audio quality). -- _ -- 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] Connecting to an Old Phone System
On 06/01/12 13:14, Dan Journo wrote: Is there such a thing as an ISDN30e PCI card which can be used with a copy of Asterisk, that can act like a voip gateway between the old phone system, and our asterisk box? Yes Digium sell 2 port PRI cards that support E1. TE200 series. I use them like this to connect to old ISDN PBX. There's no need for an extra box in the middle, just put the ISDN PCI card directly into your Asterisk system. cheers, Paul. -- _ -- 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] Change port from 5060 on Snom phone
On 06/01/12 16:17, Ishfaq Malik wrote: Hi Does anyone know how to change the target port on a Snom phone. I have tried adding :new port number to the end of the registrar but this doesn't work. It should do. Try putting registrarip:port into Outbound Proxy and leave the Registrar box just set to Registrar. Can you email me off list (since this isn't really Asterisk related and a snom support issue, which I can help with) with some details and ideally a SIP trace? cheers, Paul. Advanced - SIP/RTP - Network identity(port) is something else before anyone suggests it. Thanks in advance Ish -- _ -- 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] Good, inexpensive wireless VOIP handsets ?
On 27/08/11 10:14, Gordon Henderson wrote: On Sat, 27 Aug 2011, Alan Lord (News) wrote: On 26/08/11 19:02, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. We've been using the Siemens Gigaset 685IP range for over three years and I'm (still) very pleased with them: +1 The current generation is the N300 or N300A (A = with answering machine). These have the advantage of being able to do 3 SIP calls at once. You also don't get a handset with it so you can choose whatever handset you want (in theory doesn't even have to be a Gigaset handset as long as it is GAP compatible but you'll have a better time if you do use Gigaset ones, or at least one). These definitely work well with Asterisk for me. So if you really need 6 calls at once then you can get 2 base stations and 6 handsets. If you can live with 6 handsets but only 3 of them in use at once, then a single base will do the job. cheers, Paul. -- _ -- 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] Trouble with *8 Pickup
is this bug already reported at the issue tracker/jira? Is someone working on it? Karsten https://issues.asterisk.org/jira/browse/ASTERISK-18225 -- _ -- 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] Trouble with *8 Pickup
On 15/08/11 15:41, Ishfaq Malik wrote: On Mon, 2011-08-15 at 15:32 +0100, Paul Hayes wrote: is this bug already reported at the issue tracker/jira? Is someone working on it? Karsten https://issues.asterisk.org/jira/browse/ASTERISK-18225 That's a different issue to what we have been discussing... The last comment seems to be the same thing. -- _ -- 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] Trouble with *8 Pickup
On 12/08/11 08:46, Ishfaq Malik wrote: Have you seen it in any other versions of 1.8 or is it something that has happened in the latest release? I've not specifically seen this issue with other versions of Asterisk but then I've never tried to replicate it. The only time I've seen this with 1.8.5 is when I've purposely replicated it after reading your post. I have had much, much worse problems with pickup in previous versions of 1.8 and in the 1.6 branch where pickup will occasionally lock chan_sip altogether. This is a known issue and is in Jira and is fixed in 1.8.5. This issue doesn't really seem to cause any problems other than some stuck SIP channels. It's in Jira too: https://issues.asterisk.org/jira/browse/ASTERISK-18225 For the minute I can live with this and a nightly cron job to restart Asterisk to drop the stuck channels. Bit of a bodge I know but it works till someone fixes the issue. cheers, Paul. -- _ -- 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] Trouble with *8 Pickup
2011/8/11 Ishfaq Malik i...@pack-net.co.uk mailto:i...@pack-net.co.uk On Thu, 2011-08-11 at 14:47 +0100, --[ UxBoD ]-- wrote: Ah, now this is interesting as one of our clients had the same problem the other day; in our case when they performed the *8 they got an extension unavailable from a completely different dialplan! This was on Asterisk 1.6 though with Snom phones. In the case of this server I was looking at, the only time this error occurred was when the pickup request happened in the same second as a dialplan step change so by the time the pick up of the channel was attempted, it no longer existed. -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 It's not just a snom/asterisk thing. I can replicate this with various phones and Asterisk 1.8.5. In fact with some phones the symptoms seemed worse where the phone *8 had been dialled on didn't hang up but thought it was on a call (while the caller had gone through to whatever the next dial plan priority was, a Queue in my test case). It makes perfect sense to me that a pickup should fail if your Dial has finished and * is stepping onto the next priority but a nicer Warning such as Trying to pickup a non-existent channel would be better. My test code was simply this: exten = 123321,1,Dial(SIP/5502,5) same = n,Answer same = n,Wait(1) same = n,Queue(booking,thHr) If you time the *8 just right so it is being handled during the end of the Dial then I got: [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL [Aug 11 16:26:18] ERROR[18458]: astobj2.c:110 INTERNAL_OBJ: user_data is NULL [Aug 11 16:26:18] WARNING[18458]: chan_sip.c:6429 sip_fixup: No SIP tech_pvt! Fixup of SIP/5501-01da failed. [Aug 11 16:26:18] WARNING[18458]: channel.c:6462 ast_do_masquerade: Fixup failed on channel SIP/5501-01daMASQ, strange things may happen. cheers, Paul. -- _ -- 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] snom and srtp
On 03/08/11 03:15, James Perkins wrote: Hi, I am running asterisk 1.8.5.0 and have compiled in the srtp module All but Snom phones are working. I have set the srtp tag on the snoms to 80 and RTP/SAVP to mandatory and they worked for a few hours. This morning all snoms are reporting this when trying to make a call (this is snom calling snom). What firmware version have you got on the snom phones? It needs a pretty new version to work properly. I wrote some notes when I got this working here: http://blog.provu.co.uk/item/212/catid/3 Although that was back on Asterisk 1.8.4.1. The same server is currently on 1.8.4.3 and still working OK. cheers, Paul. -- _ -- 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] Strange network issue
On 28/07/11 02:58, Mike Diehl wrote: Any ideas? Mike. I'd go on site if possible and see what actually happens at 19:00. Set up a wireshark trace capturing all traffic through their router. -- _ -- 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] Lightning and thunder (Claude Hayn
On 27/07/11 19:41, Claude Hayn wrote: The office manager freaks out each time and starts randomly rebooting devices in no particular order including the UPS, PBX, Asterisk Gateway, firewall and router. Ahh that old chestnut. That's never a good thing, try to tell them not to do this, although I know it's hard, I have customers who love to reboot things too. If it is the Asterisk system coming online too early causing problems on the old PBX, does the UPS you are using have a power-on delay feature? In some UPS you can set delays for various sockets on them. Designed for situations like this and also so everything doesn't try to power up at once causing a power surge. cheers, Paul. -- _ -- 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] Securing Asterisk
On 23/07/11 18:38, CDR wrote: I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user information, and also, b) disable any response to any REGISTER packet altogether. Can somebody please write patch? Or should we go broke trying to stop the flood of criminals coming from abroad? Federico Not looking for an argument here but you are asking for a solution to a problem that doesn't exist. If you'd done your job properly in the first place you'd have put some basic intrusion detection on such as fail2ban, OSSEC or just a basic bash script of your own writing. The solution is already there and it's not trying to bodge Asterisk into a firewall application. If you'd done that (and instructions on how to are literally all over the Internet and this mailing list) then your customer wouldn't be $50,000 down, you'd still have your full pay and you'd not be asking for people to break Asterisk's SIP implementation (even more :P ) in order to stop you having to do things the right way. Sorry if the truth hurts... -- _ -- 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] Fwd: Re: Securing Asterisk
Original Message Subject: Re: [asterisk-users] Securing Asterisk Date: Wed, 27 Jul 2011 09:28:54 -0700 From: Myles Wakeham my...@techsol.org To: p...@provu.co.uk On 07/27/2011 09:23 AM, asterisk-users-requ...@lists.digium.com wrote: On 23/07/11 18:38, CDR wrote: I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user information, and also, b) disable any response to any REGISTER packet altogether. Can somebody please write patch? Or should we go broke trying to stop the flood of criminals coming from abroad? Federico Not looking for an argument here but you are asking for a solution to a problem that doesn't exist. If you'd done your job properly in the first place you'd have put some basic intrusion detection on such as fail2ban, OSSEC or just a basic bash script of your own writing. The solution is already there and it's not trying to bodge Asterisk into a firewall application. If you'd done that (and instructions on how to are literally all over the Internet and this mailing list) then your customer wouldn't be $50,000 down, you'd still have your full pay and you'd not be asking for people to break Asterisk's SIP implementation (even more :P ) in order to stop you having to do things the right way. Sorry if the truth hurts... +1 to Paul on this. Security is Job #1 for any IT professional. If you don't implement IDS, Firewalls, Fail2Ban, etc. you only have yourself to blame. Whether the target is Asterisk, or some old version of Apache, MySQL, or some vulnerability in Linux Kernel, etc. the hackers want a way in. Its YOUR JOB to secure your server. Even if Asterisk built some heavy security into their software, it would probably get in the way of us folk that have legitimate need for other functionality. Security is one of those things that most programmers think of as either an after-thought, or some constraint/expense that they don't want to deal with. The problem is that it should be the FIRST thing IT folk think of before putting the technology online. Anyway enough ranting... Well said Paul. Myles -- - Myles Wakeham Director of Engineering Tech Solutions USA LLC www.techsolusa.com Phone +1-480-451-7440 -- _ -- 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] Securing Asterisk - How to avoid sending, SIP/2.0 603 Declined
On 23/07/11 04:48, Bruce B wrote: Quote,/How do the users register to begin with, if their REGISTER requests won't be processed unless their IP is already known to be a registrant? :-)/ Well, unfortunately I don't have the luxury of knowing their IP and the closest I know is their IP range. Then I don't understand what the point would be. You'll have to leave Asterisk responding to all Register requests (and to be fair all the attacks I've seen have been done by sending Register requests anyway). I use OSSEC on my Asterisk systems to handle iptables rule generation on the fly. You could write your own rule(s) for that to block source IP addresses sending you Invites when they aren't Registered. cheers, Paul. -- _ -- 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] a=sendonly Music On Hold ignored
On 19/07/11 08:20, Michael wrote: On the AsteriskNow system, it gives an OK, but nothing happens, there's no music and after some time, the call even drops for empty RTP. That's the log there: What does the Asterisk CLI show when this happens on your AsteriskNow system? -- _ -- 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] Re : Re : Direct RTP with Asterisk
On 20/06/11 13:18, Eric Wieling wrote: If you can't ping between the two end points, then you can't do direct RTP. precisely. If 10.10.9.1 isn't reachable from the network that 10.10.8.1 is on then 10.10.8.1 isn't going to be able to send RTP to 10.10.9.1. You need to add routes to the routers on both networks telling them how to reach the other networks. cheers, Paul -- _ -- 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] PAP2T provisioning via SRV record?
On 13/06/11 19:44, Mike Diehl wrote: Hi all, I'm trying to provision my PAP2T's to use a SVR lookup to find the Asterisk server. I'm using a provisioning file that contains an element like: Proxy_1_ _sip._udp.example.com/Proxy_1_ However, the PAP doesn't seem to be able to find my server with this hostname. The DNS records are in place because my Polycom and Grandstream servers work just fine. What else do I need to do to get the PAP to work this way? TIA, There's a setting in the Line 1 and Line 2 page called Use DNS SRV which is set to No by default for some reason. Set this to yes and set the proxy to example.com. So something like: Use_DNS_SRV_1_yes/Use_DNS_SRV_1_ Proxy_1_example.com/Proxy_1_ cheers, Paul. -- _ -- 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] Connect intercom to Asterisk?
On 07/06/11 09:47, Gilles wrote: Hello I just read this article about a kid in England who built a box with a 3G SIM card: www.dailymail.co.uk/sciencetech/article-1394448/Doorbell-tricks-burglars-thinking-youre-home-invented-schoolboy-Laurence-Rook-13.html When someone rings your intercom, the box will call your cellphone so you can answer just like you were home. The company I work for sell SIP door entry phones but I'll not post anything here since this is not a commercial list! We also used to have a DECT door entry system that would do the same thing as that kid has invented a few years ago. It got a small bit on a TV program here in the UK called The Gadget Show. I spent the day with the film crew guys setting up the kit but unfortunately didn't get to meet the rather nice woman who presents the show - they filmed that part another day and edited it all together. Anyway, I don't really see that this is particularly unique, I also know of companies who already make GSM/3G door entry devices too although they are generally aimed at the business market rather than for home use (pricing design reflects this). cheers, Paul. -- _ -- 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] Playing with sipvicious ..
On 01/06/11 16:13, Allen David Niven wrote: what does ossec give u that fail2ban does not ? thx and cheers Replied to list so others can find this in the future if they want to. I haven't spent a lot of time investigating fail2ban as I was already using ossec before I saw much talk about fail2ban with Asterisk. Anyway as far as I can see my main advantage is that OSSEC has multiple levels of incidents. So I can create rules to send emails out for unusual activity that might not necessarily require an IP block but needs checking out. My fear with something that just watches Asterisk logs for a very specific known attack metric and then blocks IP(s) based on that is what happens when the attackers start doing something different? Fail2ban may well do all this as well, I don't know but I find OSSEC does it very well and the XML rules and log decoders are very versatile. cheers, Paul. -- _ -- 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 1..8 multiple queue
On 26/05/11 23:18, Satish Patel wrote: Thanks, I went through this example before. I was confuse and wondering how should I add third queue in this picture? From the example: *CLI database put queue_agent 0001/available_queues support^sales support^sales is a list of queues. Put as many in the list as you need. E.G. sales^support^tech cheers, Paul. -- _ -- 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] standalone PRI-to-SIP converter
On 27/05/11 16:10, Michelle Dupuis wrote: I'm looking for recommendations for standalond PRI to SIP converters. (Needs to be outside the asterisk box - so a PCIe card won't do) I've used redfone but this project doesn't need the redundancy features... Thanks! A 2nd Asterisk box with a PCIe card in it then? :) cheers, Paul. -- _ -- 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] Reporting Tool: To show who is login, queue, ... etc
On 26/05/11 15:03, Justin Sherrill wrote: Queuemetrics is neat-looking. However, it requires MySQL, and I'm using Postgres. Does anyone have a recommendation for a different product for reporting usage that's not tied to MySQL? It uses JDBC so it should work with any storage engine you can get (or make) a JDBC connector for. Although all the installations I've done of it have used Mysql. Drop the guys at Queuemetrics an email, they are very helpful. cheers, Paul. -- _ -- 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] Sending call to specific IP address
On 23/05/11 22:30, Elliot Murdock wrote: Hello, I am wondering how to send a call to a specific IP address that is different than the host of the URI. For example, an invite to the URI is j...@phone.com mailto:j...@phone.com needs to be sent to the IP address 123.456.789.255, not to the IP address of phone.com http://phone.com. How is this done? Thanks, Elliot Unless I'm misunderstanding the question, the owner of phone.com should use DNS SRV records to advertise where to send SIP traffic to if it is not the same as the A record for phone.com. E.G. paul@barney:~$ host provu.co.uk provu.co.uk has address 81.187.73.2 paul@barney:~$ host -t SRV _sip._udp.provu.co.uk _sip._udp.provu.co.uk has SRV record 0 1 5060 pbx.provu.co.uk. cheers, Paul. -- _ -- 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 per-call heartbeat?
On 24/05/11 12:08, Tony Mountifield wrote: OK, thanks. Sounds like there was some kind of issue at the ITSP then. I have seen this happen with broken SIP-ALGs in routers too. The ITSP sends the BYE but for some reason a broken SIP ALG will not deliver the packet to the right place. The ITSP will resend the BYE several times if they don't receive the responding OK message (or some error such as 481 etc) but after a few attempts it's pointless them continuing. Since SIP is UDP, this situation must occur from time to time, and I wondered if it is possible to configure any kind of per-call SIP heartbeat so that a dead call could automatically be identified with a 481 response much sooner. SIP session timers is what you need for that. Implemented in Asterisk 1.8. That's useful to know. Planning on moving from 1.2 to 1.8 over the next few months. Setting absolute timeouts on all calls might help too: http://www.the-asterisk-book.com/unstable/funktionen-timeout.html Although it's a bit of a balancing act, it can be used to limit these things to a couple of hours rather than having stuck calls going on for days. cheers, Paul. Cheers Tony -- _ -- 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] Unable to REGISTER to the Asterisk v1.8.3.3 server via SIP/TLS
Hi, It looks to me that the 401 unauth packets aren't getting back to the phones. Which suggests a network/router/nat issue rather than anything wrong with the asterisk or phone configuration. Cheers, Paul. On 8 May 2011, at 01:59, GNUbie gnu...@gmail.com wrote: Hello all, I have installed the .deb packages of the Asterisk v1.8.3.3 from the upstream project on my Debian GNU/Linux Squeeze server and bought the Comodo's PossitiveSSL SSL certificate to be used for my SIP/TLS exercise. After setting up everything and trying to fix this problem, I am still getting a 401 Unauthorized SIP message. So as of this writing, I still cannot successfully REGISTER to my Asterisk box. Below are the snippets of my Asterisk and SNOM 300 configurations including the logs for your reference. I hope anyone from this community can help me solve this problem. A HOWTO of a similar scenario will help a lot. Thank you in advance. Regards, GNUbie - - - ASTERISK v1.8.3.3 - - - [ /etc/asterisk/sip.conf ] [general] ... ... tlsenable=yes tlsbindaddr=0.0.0.0 tlscertfile=/etc/asterisk/keys/pbx.domain.com.pem tlscipher=ALL tlsclientmethod=tlsv1 tlsbindport=5061 externtlsport=5061 externtcpport=5061 tcpbindaddr=0.0.0.0 tcpbindport=5061 tcpenable=yes srvlookup=yes [361] username=361 secret=*** callerid=361-tls361 mailbox=361@family context=family transport=tls port=5061 type=friend host=dynamic dtmfmode=rfc2833 canreinvite=no nat=yes qualify=yes autoframing=yes encryption=yes *CLI core show version Asterisk 1.8.3.3-1digium1~squeeze built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-04-22 17:50:44 UTC *CLI sip show settings Global Settings: UDP Bindaddress: 0.0.0.0:5060 TCP SIP Bindaddress: 0.0.0.0:5060 TLS SIP Bindaddress: 0.0.0.0:5061 Videosupport: No Textsupport: No Ignore SDP sess. ver.: No AutoCreate Peer: No Match Auth Username: No Allow unknown access: No Allow subscriptions: Yes Allow overlap dialing: Yes Allow promsic. redir: No Enable call counters: No SIP domain support: Yes Realm. auth: No Our auth realm pbx.domain.com Use domains as realms: No Call to non-local dom.: Yes URI user is phone no: No Always auth rejects: Yes Direct RTP setup: No User Agent: Asterisk rocks! SDP Session Name: Asterisk PBX 1.8.3.3-1digium1~squeeze SDP Owner Name: root Reg. context: (not set) Regexten on Qualify: No Caller ID: asterisk From: Domain: Record SIP history: Off Call Events: Off Auth. Failure Events: Off T.38 support: No T.38 EC mode: Unknown T.38 MaxDtgrm: -1 SIP realtime: Disabled Qualify Freq : 6 ms Q.850 Reason header: No Network QoS Settings: --- IP ToS SIP: CS0 IP ToS RTP audio: CS0 IP ToS RTP video: CS0 IP ToS RTP text: CS0 802.1p CoS SIP: 4 802.1p CoS RTP audio: 5 802.1p CoS RTP video: 6 802.1p CoS RTP text: 5 Jitterbuffer enabled: Yes Jitterbuffer forced: No Jitterbuffer max size: 200 Jitterbuffer resync: 1200 Jitterbuffer impl: fixed Jitterbuffer log: No Network Settings: --- SIP address remapping: Enabled using externhost Externhost: pbx.domain.com externaddr: 11.22.33.44:0 Externrefresh: 10 Localnet: 192.168.101.0/255.255.255.0 Global Signalling Settings: --- Codecs: 0x60e (gsm|ulaw|alaw|speex|ilbc) Codec Order: ulaw:20,alaw:20,gsm:20,speex:20,ilbc:30 Relax DTMF: No RFC2833 Compensation: No Symmetric RTP: No Compact SIP headers: No RTP Keepalive: 0 (Disabled) RTP Timeout: 15 RTP Hold Timeout: 0 (Disabled) MWI NOTIFY mime type: application/simple-message-summary DNS SRV lookup: Yes Pedantic SIP support: Yes Reg. min duration 1800 secs Reg. max duration: 3600 secs Reg. default duration: 120 secs Outbound reg. timeout: 20 secs Outbound reg. attempts: 0 Notify ringing state: Yes Include CID: No Notify hold state: No SIP Transfer mode: open Max Call Bitrate: 384 kbps Auto-Framing: No Outb. proxy: not set Session Timers: Refuse Session Refresher: uas Session Expires: 1800 secs Session Min-SE: 90 secs Timer T1: 3000 Timer T1 minimum: 100 Timer B: 192000 No premature media: Yes Max forwards: 70 Default Settings: - Allowed transports: UDP Outbound transport: UDP Context: default Force rport: No DTMF: rfc2833 Qualify: 0 Use ClientCode: No Progress inband: Never Language: MOH Interpret: default MOH Suggest: Voice Mail Extension: asterisk *CLI sip show peer 361 * Name : 361 Secret : Set MD5Secret : Not set Remote Secret: Not set Context : family Subscr.Cont. : Not set Language : AMA flags : Unknown Transfer mode: open CallingPres : Presentation Allowed, Not Screened Callgroup : Pickupgroup : MOH Suggest : Mailbox : 361@family VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Max forwards : 0 Dynamic : Yes Callerid : 361-tls 361 MaxCallBR : 384 kbps Expire : -1 Insecure :
Re: [asterisk-users] best current version and motherboard/CPU compatibilities
On 04/05/11 18:17, || dave cantera Mobile wrote: paul, doug, I had several AMD athlons 64bit... no problems running centos, suse. they seem solid on 1.4.xx... had a few intel celerons and P4s. they were good as well. guess I was Lucky back then! thanks for supporting the list! daveC don't get me wrong, I use AMD almost exclusively in my desktop PCs. Usually with Debian or Ubuntu. It's just back in the late 1990s/early 2000s I had an AMD K6-2 300 (if I remember correctly) and it wasn't a good experience at all. I certainly wouldn't build an Asterisk system using 10+ year old CPUs either (unless it's just a toy) :) For the record most Asterisk systems I build are currently using Intel Atom hardware but also some PowerPC. cheers, Paul. -- _ -- 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?
On 05/05/11 05:41, Cary Fitch wrote: Flavio E. Goncalves www.asteriskguide.com http://www.asteriskguide.com Compare to which version of Windows… Patches are a never ending process Cary Fitch I think this attitude is half the problem. Asterisk is not a desktop computer operating system. It is the engine for a telephone system, a telephone system needs to be much more reliable than a desktop PC if it is going to continue to compete in a growing industry. I agree with the comments on concentrating more on stability than new features. It's hard because it is new features that make good stories and are easier to shout about in order to get a product better known. For now I am sticking with 1.4 mainly (although I am using 1.6 where I need BRI connectivity) but my plan is to move to 1.8 when I feel I have tested it enough and it's been around for long enough to be proven. -- _ -- 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] Cordless VoIP Phones and Access Point hand-off?
On 05/05/11 00:02, Ira wrote: Not that it applies but I recently installed a Snom M3 and it seems to behave like you want. When I walk out of range and then back in the call is usually still there. I've not tested past that so it might hang up after an unknown timeout. Ira The difference here is that the M3 is a DECT phone. So the SIP leg of the call terminates on the base station itself. If the handset goes out of range of the base then the base can decide to just keep the call going. Wifi is a bit different because Asterisk will see the RTP stream stop and qualifies being lost. There are probably work arounds but personally I believe DECT is a far superior protocol for voice and hand-over usually works better. -- _ -- 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] Issue with Asterisk Aastra 57i at v3.2
On 05/05/11 04:37, Richard Kenner wrote: I recently tried to update my Aastra 57i to version 3.2 and ran into a problem. It won't properly register and says contact mismatch. I added sip contact matching: 2 to aastra.cfg, but that didn't help. When I look at the SIP trace, but I see is the Aastra sending a REGISTER and Asterisk replying with the 401. The phone then sends the REGISTER again, this time with the hash. Asterisk now replies OK, but sends an OPTION packet FIRST and I think that confuses the Aastra. Has anybody seen this? Is there any way to have the packets sent in the proper order? -- Since I was keen to see if there was a phone bug I've just tested this here. I am using firmware 3.2.1.43 on my 57i which I have just downloaded from aastra.co.uk this morning and Asterisk 1.4.25.1. Asterisk does indeed send an Options before the OK but my 57i doesn't seem to mind. See the SIP debug trace below. Perhaps you need to upgrade firmware on the Aastra phone? Or turning off qualify for this peer might work-around it for you. Reliably Transmitting (no NAT) to 192.168.2.73:5060: OPTIONS sip:2002@192.168.2.73:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK6c97be12;rport From: asterisk sip:asterisk@192.168.2.201;tag=as71d2aacd To: sip:2002@192.168.2.73:5060;transport=udp Contact: sip:asterisk@192.168.2.201 Call-ID: 0d0ecb8721126fdc43a44660792b63b6@192.168.2.201 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 05 May 2011 10:43:00 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- sark500*CLI --- Transmitting (no NAT) to 192.168.2.73:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.73;branch=z9hG4bKb06503cf2bc96500f;received=192.168.2.73 From: sip:2002@192.168.2.201:5060;tag=893258dbbd To: sip:2002@192.168.2.201:5060;tag=as6fe265b2 Call-ID: eb5b051757397d5d CSeq: 18419 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Expires: 160 Contact: sip:2002@192.168.2.73:5060;transport=udp;expires=160 Date: Thu, 05 May 2011 10:43:00 GMT Content-Length: 0 Scheduling destruction of SIP dialog 'eb5b051757397d5d' in 32000 ms (Method: REGISTER) sark500*CLI --- SIP read from 192.168.2.73:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.201:5060;branch=z9hG4bK6c97be12;rport=5060;received=192.168.2.201 From: asterisk sip:asterisk@192.168.2.201;tag=as71d2aacd To: sip:2002@192.168.2.73:5060;transport=udp;tag=2437297184 Call-ID: 0d0ecb8721126fdc43a44660792b63b6@192.168.2.201 CSeq: 102 OPTIONS Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Server: Aastra 57i/3.2.1.43 Supported: path Content-Length: 0 -- _ -- 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] Issue with Asterisk Aastra 57i at v3.2
On 05/05/11 13:41, Richard Kenner wrote: Asterisk does indeed send an Options before the OK but my 57i doesn't seem to mind. That's odd. It does for me. Perhaps you need to upgrade firmware on the Aastra phone? The problem occured when I DID upgrade it! Precisely to the one you mentioned. OK, I just wasn't sure if it was precisely 3.2.1.43 you had. In that case it suggests it is some setting you have applied to the phones that is causing it. Can you post the local.cfg server.cfg files from the phone (removing the passwords from there first)? Or turning off qualify for this peer might work-around it for you. I'm sure it would, but all peers are those phones, so that's not an acceptable workaround. But it might allow your users to make phone calls while you fix the issue properly. cheers, Paul. -- _ -- 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] audiohook.c: Failed to get 160 samples from write factory
On 05/05/11 14:04, Jonas Kellens wrote: Hello list, what does this mean : [May 5 *14:58:12*] DEBUG[8770] chan_sip.c: This call was answered elsewhere[May 5 14:58:12] DEBUG[8770] chan_sip.c: ### It's the cause code, buddy. The cause code!!! [May 5 14:58:12] DEBUG[8770] chan_sip.c: This call was answered [snip] see rfc3326 section 3.1. Call Completed Elsewhere. It's used so that phones in ring/hunt groups don't record a missed call if the call is answered by someone else. I was looking forward to Asterisk supporting this for a while :) cheers, Paul. -- _ -- 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] audiohook.c: Failed to get 160 samples from write factory
On 05/05/11 14:16, Olle E. Johansson wrote: We've had that for quite some time. There's an option to Dial() and one for Queue() to enable it. Check the documentation. /O yes my only problem with the 'c' option for the Dial command is that it still seems to add the Reason header if the call hasn't actually been answered elsewhere :) I.E. 10 phones in a ring group, someone calls up, no one is here so no phones answer the call. Caller eventually gives up, no phones record a missed call. Unless it's changed from when I last tried it, or I was doing something wrong (quite possible!). Wasn't aware the same option existed for Queue though. cheers, Paul. -- _ -- 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] Password to be ecrypted?
On 03/05/11 09:09, Robles Román, José Miguel wrote: Perhaps using one-way hash functions (http://en.wikipedia.org/wiki/Cryptographic_hash_function) like MD5 or SHA-x, even if you get the file with passwords and the code that checks them, it would be difficult to find a collision (a password that matches the hash). This is the way in which apache, for example, stores passwords (see htpasswd). In order to maintain compatibility, the configurarion could be [...} secret_sha2 = ... Regards, José Miguel I thought this already existed: http://www.voip-info.org/wiki/view/Asterisk+sip+md5secret Although I have to admit, I've never tried using it. cheers, Paul. -- _ -- 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] best current version and motherboard/CPU compatibilities
On 04/05/11 17:10, || dave cantera Mobile wrote: doug, why are you shaking!?!?... do you have a better recommendation? daveC AMD K6 CPU brings back some pretty bad memories from me too. Doug Lytle wrote: C F wrote: model name : AMD-K6(tm) 3D processor *shudder* Doug -- _ -- 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 shouldn't I use 1.8?
On 25/03/11 14:36, Douglas Mortensen wrote: Now that we've hashed out some thoughts on the most stable version of asterisk, I'd like to ask the question as to why I should NOT use 1.8? What are specific reasons? For instance a few days back I was speaking with James at Rhino Equipment. He said that he has no real data on why I shouldn't use 1.8. They just follow a practice of not jumping on the newest version. I agree with what Jonathan also said in this thread but that is also a good enough reason on it's own. Data doesn't yet exist to say whether it's stable enough. I like to err on the side of caution with phone systems as they cost lots of money when they go wrong! -- _ -- 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 Invite and Asterisk API/Variable
On 24/03/11 05:49, Olivier CALVANO wrote: The To, To:sip:081169x...@91.121.xxx.xxx;user=phone, can i get it into a variable for sent it at a API ? You want the sip_header function: http://www.voip-info.org/wiki/view/Asterisk+func+sip_header cheers, Paul. -- _ -- 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 registration DoS but no logs in messages
On 17/03/11 05:37, Patrick wrote: Dear mailing list, I've a Asterisk 1.4.21.2~dfsg-3+lenny1 package installed on my debian and I've a strange behavior. After some days running normally, my asterisk is under heavy attack, however, there is nothing logged in the console (logging from debug - error) or file (level from notice -error) I can see that there is also a peak on the network traffic. My first guess is that I'm suffering from a SIP registration DoS, but, as there is nothing logged about a not matching peer or incorrect password logged to file, my fail2ban script is not blocking the attacker. I normally restarts Asterisk and logs are restarting to log attacks, but, today, it's not working FYI, I've checked and my loggers are not muted and the logging level is at least notice. I've also reloaded my loggers but no effect. Do you already have experienced such situation ? Is there any known issue with logging module stopping while Asterisk is DoS'ed ? Best regards, Patrick It's possible that fail2ban has already blocked the incoming registration attempts but the attacker is still blindly sending packets to you. Often a sign the attacker is using an old version of sip-vicious, you can often stop such things by using the svcrash.py script they now provide. Check your iptables logs. cheers, Paul. -- _ -- 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] ADA: DOA?
On 06/10/10 20:25, Ken D'Ambrosio wrote: Hey, all. While ADA can still be downloaded, that's about all that I see. No development, no recent mention, and -- perhaps worst of all -- it appears not to work properly under 64-bit systems. So, assuming Digium's abandoned it, are there any suggestions of alternatives? Right now, I'm replacing a Shoretel system, and I'd *dearly* love to avoid the incredibly fat client they have; if there's something slender -- roughly in the same line as ADA -- I'd be very interested, even if it's not free. Thanks, -Ken It would seem to be a dead project yes, I can't understand why Digium bought Click2Dial, re-branded it ADA and then stopped doing anything with it. I even tried to ask a Digium employee at a VoIP show in the UK about a year ago what was going on with it but they skirted the question and tried selling Switchvox to me (which might actually, inadvertently answer the question ;) ). cheers, Paul. -- _ -- 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 flood attacK
On 03/10/10 21:19, Greg Saunders wrote: Hello all. I was recently the victim of a SIP flood attack. I'm wondering what is the best method to prevent such things in the future. Many thanks Greg do one of the following: - use deny permit lines in sip.conf /or iax.conf to restrict any remote Registrations from known IP address ranges only. Or use iptables rules to do something similar. - use a log scanning tool such as fail2ban or ossec which can react on multiple registration fails and block ip addresses in iptables - enforce strict password policy on all users on the system I think simply relying on alwaysauthreject is very dangerous as it's only a matter of time before the attackers catch on to this and carry on attacking regardless. Sure there's less chance of them getting a correct username/secret combination but in the meantime, the register attempts are practically a DoS attack. Plus that setting further breaks the SIP RFC. I also think that assuming that the attackers will eventually get in one way or another is wise. So put in place appropriate measures to limit the damage they can do (daily spend limits with SIP providers, blocking international and/or premium rate numbers etc...). cheers, Paul. -- _ -- 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] minimum card for dahdi timing source ?
On 02/10/10 17:24, mancyb...@gmail.com wrote: Hi All, for a vicidial server which uses only voip, which is the minimum telephony card which would provide the required clock timing source for conferences to work properly ? Maybe the Digium TDM410PLF card without any daughter card would do the job ? Thank you very much for supporting. Have a nice week-end, Mike The cheapest device I've seen to provide a hardware timing source is the USB voice sync tool from Sangoma: http://www.sangoma.com/products/hardware_products/specialty_tools.html I know of at least one person using this with Vicidial successfully. cheers, Paul. -- _ -- 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] Playing with sipvicious ..
On 18/08/10 17:10, Gordon Henderson wrote: ... using it as a tool and understanding what it does... So one part of it's toolset identifys valid SIP accounts - and I was under the impression that alwaysauthreject=yes was supposed to stop this... However, it sends a request for a highly probably non-existent account, then sends requests for probably existing accounts and I guess compares the results - account not found vs. bad username or password... It thus trivially, and very quickly finds valid accounts when fed with a list of accounts to try in the first place (e.g. 100-999, or 1000-, etc.) I wonder if it's time to introduce yet another parameter to it - which will cause asterisk to return the same error code for all 3 conditions - and return the not found error, even on bad username or password. It breaks the RFC even more, but might it be worth it? (I've just had 30GB of sipvicious traffic sent to my hosted servers in a 12-hour period - it came from what looked like a VPS host in France - trivially firewalled out, but even dropping the packets didn't stop the flood! It's so badly written it appears to just ignore any return codes that it doesn't want, or even no returns at all!) Gordon I've been playing with this a fair bit recently too, if only to gain myself a better understanding of the attacks so as to be able to prevent them better. I found that when sending Registers to Asterisk (I was testing with 1.4 since all my deployments are 1.4), alwaysauthreject does actually stop it from being able to determine real extension numbers. However I also found that making it send Invite requests means it can determine real extensions that are currently Registered. I've been using OSSEC to block source IPs that attacks come from. So far it seems to work well. Once you start silently dropping the inbound SIP traffic from the attacker they seem to go away very quickly (once the door is shut, no point them carrying on). I'm yet to see a more intelligent attack using distributed source IPs but I'm sure it'll happen. The scans I see happening usually come from random dynamic DSL addresses and the like from all over the place (inc within the UK) so I suspect these are virus infected zombies, so a distributed attack is surely easily possible. Something else I noticed is that once OSSEC is doing it's job (or whatever other automatic blocking script you use), the attacks stop. I have my systems set up to email me when an attack is blocked and after a few days, the attacks stop. Which I interpret as a sign that attackers are maintaining lists of known vulnerable IP addresses, which is common for things like ssh attacks, spam relays etc... I don't believe modifying Asterisk code to send non-RFC compliant relies is a good idea, I prefer the security layer to be handled by something else on top of Asterisk. I have also seen attacks exploiting bugs in Asterisk too, I'm not going to go into them here for obvious reasons but I guess these types of attacks will get more commonplace once people start getting a bit wiser to the current fairly basic port scan and extension enumeration attacks. cheers, Paul. -- _ -- 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 Hardwares
On 16/08/10 11:46, Tino wrote: Hello, Can antbody recommend devices that can be used along with my Asterisk server Paging Amplifier SIP enabled Paging Gateway VOIP SIP loudspeaker Also , please recommend video phone sets that suppot paging, intercom (autoanswer) Thanks A Snom PA-1 should cover all those requirements. cheers, Paul. -- _ -- 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] Small PC to build and run Asterisk
On 14/06/10 18:11, Gordon Henderson wrote: On Mon, 14 Jun 2010, Chris Bagnall wrote: Actually, the Atom seems to be surprisingly powerful. We have a couple of Atom boxes with transcoding and conferences enabled without issue. I wouldn't pretend it'll cope with hundreds of conference participants, but with ~10 or so it seems to be fine. I'll second the Atoms - I have several in the data centre handling VoIP, virtual PBXs, etc. And you can now get fanless motherboards. Bliss. Even using a few as general purpose LAMP servers too - the data centre I use doesn't charge per amp, but it's coming, and already there in most big places - in the UK, anyway - it seems Amps cost more than Gb) Likewise with transcoding - we've only really tested up to ~30 channels with G.711 to GSM, not any of the heavier CPU workload translations (e.g. iLBC or G.729). For a small to medium office (e.g. 30 extensions, 10 concurrent calls) it works fine, even with a little conferencing and transcoding. I do that with a 500MHz AMD Geode ... (no transcoding though - benchmarked it to 85 concurrent calls, handling the media streams - limit these boxes to 60 extensions though) Gordon +1 for Atom based systems. I use them too (although we build the systems ourselves). It really is quite powerful hardware, has no problems transcoding, conferences, multiple ports of ISDN BRI or a single PRI. I'd be happy using one of these up to 50 extensions with 15-20 concurrent calls. I use an Atom based system at home too which is running OpenVZ with things like apache tomcat, asterisk, nfs server etc... all running at the same time. cheers, Paul. -- _ -- 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] Dual Atom mobo - call capacity
On 11/06/10 01:19, Michelle Dupuis wrote: I'm looking for a small formfactor mobo for an install that needs to handle 25 phone sets (no transcoding). I found a new dual atom 1.66GHz mobo - anyone know what kinds of call volume that will handle? MD Any of the Atom CPU systems will /easily/ handle 25 concurrent calls (and with a 25 extension system, 25 concurrent calls is very unlikely). I use the single core Atom 230 CPUs for systems of this size. Something to bear in mind is how the system will be used, max concurrent calls isn't really that great a performance factor, call arrival rates are more relevant, the CPU time is spent setting up and tearing down calls. Simply having calls in progress with no transcoding uses a tiny amount of CPU in comparison to the work involved setting up and routing a new call. cheers, Paul. -- _ -- 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] OpenVPN/SNOM 820: a review.
--[ UxBoD ]-- wrote: Would be nice if the VPN support could be back ported to the 360s. Never going to happen, there isn't enough flash memory to store the code. The Snom370 has had OpenVPN support for quite a while though. cheers, Paul. -- _ -- 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] Odd bug in Siemens C460IP ?
Robert Lister wrote: Hello, I think I have encountered an odd bug in Siemens C460 IP/dect handsets, which is a bit annoying, and I'm not (yet) sure how to get round it without lots of hacks. Basically, on all external incoming calls, we set: exten = s,n,SIPAddHeader(Alert-Info: Bellcore-dr2) This causes handsets (i.e. Cisco 7960 / Grandstream / aastra) to set a different ring cadence so to differentiate between external and internal calls. Other handsets that do not support Alert-Info: just ignore the presence of this header. When this header is set in a call to the C460 IP, it does not alert, in fact it does not respond to any INVITE requests; asterisk just retries the requests a few times and then gives up. Anyone able to reproduce? I have firmware version 0107 / 041.00 I suppose as a workaround I could add an astDB entry for these extensions, and a bit of logic in the dialplan to tell asterisk not to add the header for extensions that have that flag set. Regards, Rob I can replicate this behaviour too using an S450IP when an Alert-Info header is present. I have reported the issue to Siemens so hopefully this will be fixed in a firmware update in the near future. cheers, Paul. ___ --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] Siemans SIP/PSTN phone S450
Adrian Marsh wrote: Hi All, Just added a Siemens DECT SIP/PSTN S450 phone to login to my A*k server, and I see Got SIP response 405 Method Not Allowed back from 192.168.3.64 but the phone seems to work ok. Any ideas where it falls over in the SIP protocol? I've included this in the debug below. It is in response to Notify packets because the Siemens phone doesn't support presence at the moment. It wont effect the operation of the phone or Asterisk at all. cheers, Paul. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --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] Siemens Gigaset DECT base provisioning
Olivier wrote: Hello, My goal is to provision C450IP or S450IP models. Has anyone a hint to provision them from configuration files ? Usually, we use dedicated menu embedded inside Gigaset handset. An http server also exists but I couldn't find any dhcp-tftp combination to configure them. Any clue ? Regards It's not currently possible but Siemens are working on new firmware for at least the S450IP model which will support auto-config using http. I'm not sure when it's due for release though. ___ --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] POE injector
Noah Miller wrote: I'm looking for 24 or 48 port IEEE802.3af POE injector. Any recommendation? Yes. For the price of one of those multi-port injectors, you can come close to the price of a new Netgear or 3Com PoE switch. The injectors typically add power to the unused pairs (mode B PoE). This means you can't use them on anything better than fastethernet. When switches do PoE natively, they put the power on the data carrying pairs (mode A PoE), so they can do gigabit ethernet. I think PowerDsine makes a PoE injector that uses mode A, and so it can do gigabit ethernet. - Noah The midspans that Phihong make are very good. They support gigabit pass through and have very good overload protection. cheers, Paul. ___ --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] blind transfer on hook-flash from SIP phone
Hi, I have a SIP phone which does not natively support SIP transfers (REFER etc...). So far all that is possible is to enable blind transfers using the t and T arguments in Dial from the # DTMF key. The phone has an R button on it and this can be setup to either send an RFC2833 hook flash message (value 16) or a SIP INFO message which you can edit the contents of (since there seems to be no standard way of signalling a hook flash in SIP INFO). However, Asterisk ignores the hook flash messages and I can't find anyway of getting it to treat the hook flash message in the same way as a # being sent. The only information I can find relates to detecting or sending hook flashes on zap channels. cheers, Paul. ___ --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] wifi sip phone real-world experiences?
Alex Crow wrote: Alban, Thanks! Where on earth did you source this? I can't seen to find hide nor hair of it here in the UK :( Alex On Mon, 2007-06-04 at 16:01 +0200, Alban wrote: Hi, I've tested several wifi phone (UtStarcom, Hitachi 3000 and 5000, and one Siemens). The Siemens is the best one, for a really cheaper price than hitachi. And was the only one which roams well between AP (same SSID, same channel) with WPA. Battery is still a problem, especially if the coverture is not very good everywhere. But that was the best one I could test... The reference is : Gigaset SL75 WLAN. Hope it helps Alban It's not available in the UK, Siemens pulled the UK varient of the product due to lack of demand. They'll only release something in a country if companies place orders for 1000's of the product. There simply isn't the demand for Wifi phones in the UK to sell them in a reasonable amount of time. I have one of them in my desk drawer which I've had for a good while, it's pretty good but as someone else said, not as good as DECT. cheers, Paul. ___ --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] wifi sip phone real-world experiences?
Zoa wrote: Gordon Henderson wrote: On Sun, 3 Jun 2007, Andrew Kohlsmith wrote: On Sunday 03 June 2007 4:30 pm, Alex Crow wrote: No frills, specs look good, price seems excellent! http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519 That's a terrible phone. I've tried them. the screen is pretty much useless, the buttons are *TINY*, the battery life horrible, and the ringtones gimmicky. I have to disagree on at least one point here - Battery life. I don't think 3 or 4 days standby and several hours of talk time makes for a horrible battery life. The F1000G has other faillings, but battery life isn't one of them! If you compare this battery life to a decent DECT phone, it's still miserable. I'm used to these dect phones : http://www.bang-olufsen.com/UserFiles/File/Products/Technical%20Specifications/BeoCom6000_en.pdf [snip] It's not really a fair comparison though, DECT was designed from the offset to be used with portable phones on low-power batteries. Wifi was never designed for this so it's comparatively power hungry. The F1000G in my opinion is still pretty good battery life for a wifi phone, I've seen some that will not last a day in standby. Looking at the OP's requirements list in the first post, there is nothing currently on the market which will cover anything like all those features (and do it well!). I've currently got several Nokia Wifi/GSM phones sat on my desk, they are difficult to configure and very quirky, frankly not even good enough to be considered a techie's toy. I am told 3rd party softphone clients such as TruPhone work a lot better than the built-in SIP client but I'm yet to test any of these. The main problem is they have a habit of constantly losing connection with my access points. Even the F1000G and F3000 phones I have here don't do that. I'm yet to be convinced that wifi in it's current state is any use for telephony at all. DECT works so much better, it just needs someone to make a fully functioning SIP DECT phone. The Siemens is good but they need to work on more SIP functions, although proper transfers should be possible soon. I also have some Philips DECT SIP equipment next to my desk to look at when I've got a chance! cheers, Paul. ___ --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:spa942 provisioning
Benko wrote: Hello! Sorry for the OT-thread, but i don't know where else too ask... Has anyone done http-provisioning of a Linksys SPA942 with client side ssl-authentication? Where do i get the CA from? I'm aware of the Sipura mass deployment howto on voip-info.org, but it doesn't cover the authentification part. Thanks Christian You get the server certificate from Linksys. You'll need to be a reseller or service provider though, or the reseller/service provider you buy from may be able to request one on your behalf. You need to use something like OpenSSL to generate a CSR to send to Linksys. cheers, Paul. ___ --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] F3000 registering to asterisk
Neil Cherry wrote: [snip] How did you get access to the web config? What user and is it the default password/access code? type it's IP address into a web browser. Username: admin, password: psw is the default. cheers, Paul. ___ --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] Development news :: T38 passthrough support
The SPA-2100 is the only one to support T.38 at the moment though. SPA-2002 has the ability to support t.38 (i.e. it has the processing power required) but the firmware support isn't there yet. C F wrote: On 3/11/06, Kristian Kielhofner [EMAIL PROTECTED] wrote: Olle E Johansson wrote: Friends in the Asterisk.org community, There is a lot of cool stuff going on in Asterisk development, things that will change Asterisk and make it work better in your organisation, make it easier to sell in your area or give you more consulting oppurtunities - in short, functionality that will make a lot of sense for you users. However, developers can't really get anywhere without a dialog with the users. You know what you need, you know what is missing and how you would like to make Asterisk a better choice. I am planning to send out a description of new features now and then, to inform you about what is going on, but also to get some feedback. The bug tracker is not only a tool for developers, but also for testers and users to react to changes and contribute. *** ITU T.38 -- Fax over VoIP Olle, Let's say that I wanted to setup a complete environment to test this. I presume that I would need the following: Fax machine T.38 compliant ATA (Sipura claims this) Asterisk server T.38 compliant something - does this need to be a Cisco 5300 (or similar)? Can it be just another plain ATA and fax machine? Another ATA like the SPA line should work on the second end as well. Please suggest some possible hardware! Thanks! -- Kristian Kielhofner ___ --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 ___ --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 ___ --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] dipura 2002 auto dial or intercom
This called hot line or batphone (as it's like the phone the commissioner used to have in Batman that went straight through to Bruce Wayne). Set the dialplan to this: (S0:#) where is the number/SIP address you want to dial. Note, that's a zero after the S. Anton Krall wrote: Guys. Anybody using sipuras 2002 knows if there is a way to make the phones connected to it to autodial an extension when the phone is picked up? For example, if the phone is on a police booth (building entrance) and you want the guys to just pick up the phone and make the phone auto dial the receptionist extension without the guys having to dial anything (ala batphone). Is this possible with spa's? ___ --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 ___ --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] SPA-3000: Dual Registrations?
Are you trying to register both lines to the same user account in *? That wont work, a user can only be registered once at any time. Kerry Garrison wrote: We just posted an updated guide to the SPA-3000 a few days ago. The example uses AMP but all the settings are there: http://voipspeak.net -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Rich Adamson Sent: Tuesday, December 13, 2005 5:04 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA-3000: Dual Registrations? Brian Capouch wrote: I'm wondering if there's anyone out there who has successfully gotten an SPA-3000 to register, as its documentation would indicate, on both ports 5060 (for standard client FXS service) and 5061 (for the purpose of originating calls via SIP from the PSTN interface on the box). I can get one or the other to register, but with the current firmware (3.1.7) so far I haven't been able to get both. The second ones gives me an error: chan_sip.c:10823 handle_request_register: Registration from 'sip:[EMAIL PROTECTED]' failed for '192.168.1.113' - Username/auth name mismatch I have checked the settings 1000 times; spa3000 is what I have in both the SIP "stanza" name as well as the "username" parameter, and that is the name I'm using in the SPA config screen for "User" It works all right, even though, according to the average of the many conflicting explanations as to how these things are to be configured, it shouldn't. Yes, have had it working through many sipura firmware updates including the latest, and through many cvs-head updates over the last year or so. I'm out of town today and can't supply any sample config info, but it was very straight forward. I used different userid/secrets for the two registrations. Multiple associates and isp's (that I assist) also have it working fine. ___ --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 ___ --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 ___ --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] Linksys SPA-941 Admin Guide
we should be getting a limited number in a couple of weeks time. Proper stocks will be arriving in January - www.provu.com Paul. Senad Jordanovic wrote: [EMAIL PROTECTED] wrote: There is a review on the homepage at http://voipspeak.net It has been available for a few weeks, it is much nicer than the 841! Who has it for sale in UK? ___ --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 ___ --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] Anyone using Parlay VoXip SIP Gateway with Asterisk ?
I've used one with a Snom SIP server system it worked quite well but not tried it with * unfortunately. Voxtream support team are excellent though I'm sure they'll help you get it working. Robert Rozman wrote: Hi, we're having quite some problems with new hardware we're testing - Parlay Voxip ISDN-SIP gateway... So we're curious if anyone is using this in connection to Asterisk and what are experiences on this HW ? Thanks in advance, regards, Rob. ___ --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 ___ --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] Linksys PAP2: supported codecs
yes that's what i'm lead to believe as well. Only the SPA-2100 SPA-2002 support two simultaneous g.729 calls, the older/lesser models don't have the processing power required to encode two g.729 streams. Rich Adamson wrote: I don't think they want to solve it. It's the same with the Sipura boxes. Only SPA 2100 supports 2 G729 sessions. The archives suggest the original models didn't have enough processing power to handle the compute-intensive g729 codec. I'd have to guess that is a correct assessment from what I've seen. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sipura 2000
You could get in touch with the company who is providing the settings for the Sipura adaptor (should be able to find out who it is from the Settings URL) ask them to change the settings to be user-changable. The permissions for each setting is configured through the http configuration it sounds like they have yours all set to be read only by the user. I followed your steps to the letter but after resetting to factory defaults unfortunately it still doesn't record the configuration changes I do. 2005/11/9, Adam Moffett [EMAIL PROTECTED]: If you unplug the ethernet cable on a Sipura SPA and then reset the power it'll boot up in a diagnostic mode. When you pick up the phone that's connected to it you'll get a dialtone and there are speical codes you can dial to do various things. Reset it to factory defaults by dialing followed by 73738# full instructions are here: http://www.sipura.com/Documents/faq/Section_3.html#4 Once you do that the provisioning enable should be no and you can reconfigure the device however it needs to be. Hi, Thanks for your response. I checked the setting, and indeed it was set to yes. However, once I change it to no and click on apply but after rebooting it's enabled again (with all settings reverted to factory defaults, as usual). Maxi. 2005/11/8, Rusty Dekema [EMAIL PROTECTED]: It's possible that your SPA-2000 is set up to read a configuration file from a remote host every time it boots up, which would overwrite any changes you make. If you log in as admin and go to the advanced view, there is an option under the Provisioning tab called Provision Enable. Make sure that this is set to no and your changes should remain in place. -Rusty On 11/8/05, Maximiliano J. Goldsmid [EMAIL PROTECTED] wrote: Hello, I have a problem with my Sipura 2000. The problem is that it does not accept any change in the configuration. When I access to it, via browser or phone, and make any change, after clicking submit all changes all the changes I made dissapear and teh configuration remains with the original parameters. So I need to know how can I work it out. Thank you very much. Maxi ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
