[asterisk-users] Call popups with Thunderbird (and potentially other PIMs)
Some of you may be interested in a project I've been keeping myself busy with whilst convalescing - or rather, two closely related projects. callPopPy is an incoming call popup notifier written in Python. It's probably most useful to Linux users, but was written to be portable and has been tested on Windows XP. It uses the starPy library to interact with Asterisk's AMI. callPopPy looks up incoming numbers in a SQLite database and displays any associated name found. Which is where the second project comes in... Squalit is an extension for Mozilla Thunderbird, the purpose of which is to populate callPopPy's database. It exports phone numbers and display names from Thuderbird's address book. It can export individual contacts or entire address books, and can be configured to automatically update the database periodically. The two stage approach means callPopPy can potentially be integrated with other PIMs as well. Any feedback would be gratefully received. You can find more at http://www.oak-wood.co.uk/callpoppy Cheers Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Thunderbird extension using AMI to dial
Hi I've just added direct support for AMI to a forthcoming version of TBDialOut, a Thunderbird extension for dialling direct from Thunderbird's address book. If anyone fancies testing it I'd be grateful for any feedback. If you feel like casting a critical eye over the code, or doing some translating, even better. AMI support is available in TBDialOut 1.7.0pre1, which can be found either at http://www.oak-wood.co.uk/tbdialout/ or from the 'Development channel' at the bottom of the page at https://addons.mozilla.org/en-US/thunderbird/addon/tbdialout/ Thanks for your help Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Flood of REGISTERs - attack?
On 13/04/10 00:27, Tom Stordy-Allison wrote: Yep - this is the same codebase - the attack that I had from an EC2 yesterday and the day before, all had the User-Agent: friendly-scanner too. Looks like they are branching out Go with Joshua Steins blog post - it worked perfect for me and got it off my back. Unfortunately, it hasn't worked here. Took me ages to figure why iptables -t nat -A PREROUTING -i ppp0 -s 62.149.239.97 -p udp --dport 5060 -j REDIRECT --to-port 5071 didn't redirect the traffic. Turns out (I think) that only new connections are sent to the nat table, and this ones been established for several days now. If anyone can shed light on how to reset the connection tracking I'd be interested, but only academically now. Instead I just stopped asterisk and ran Joshua Stein's script on 5060. But it didn't do the trick. The bot showed no sign whatsoever of letting up. My other line of defence is the following rate limiting in iptables. Is this likely to interfere with actual day to day operations of Asterisk, given a small and not very busy installation? Basically it will drop packets if it has seen more than 20 in the last 30 seconds, or more than 10 in the last 2 seconds from the same host. # rate limit external SIP connections to Asterisk iptables -A INPUT -i ppp0 -p udp --dport 5060 -m recent --name SIP --rcheck --seconds 30 --hitcount 20 -m limit --limit 1/sec --limit-burst 3 -j LOG --log-prefix Dropped (sip rate lim 1): iptables -A INPUT -i ppp0 -p udp --dport 5060 -m recent --name SIP --update --seconds 30 --hitcount 20 -j DROP iptables -A INPUT -i ppp0 -p udp --dport 5060 -m recent --name SIP --rcheck --seconds 2 --hitcount 10 -m limit --limit 1/sec --limit-burst 3 -j LOG --log-prefix Dropped (sip rate lim 2): iptables -A INPUT -i ppp0 -p udp --dport 5060 -m recent --name SIP --update --seconds 2 --hitcount 10 -j DROP iptables -A INPUT -i ppp0 -p udp --dport 5060 -m recent --name SIP --set -- _ -- 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] Flood of REGISTERs - attack?
I'm currently receiving over 200 SIP REGISTER requests per second from a machine apparently in Italy, host97-239-149-62.serverdedicati.aruba.it. This has continued for several days, and ab...@staff.aruba.it are unresponsive. I've had a couple of similar incidents recently, the others originating from uk2.net. I have an ADSL connection and responding to these REGISTERS was consuming all my outbound bandwidth. I am now dropping the packets but still some 600kbps of inbound bandwidth is consumed by this. The packets look something like this: REGISTER sip:62.3.200.113 SIP/2.0 Via: SIP/2.0/UDP 62.149.239.97:5086;branch=z9hG4bK-2570753370;rport Content-Length: 0 From: test sip:t...@62.3.200.113 Accept: application/sdp User-Agent: friendly-scanner To: test sip:t...@62.3.200.113 Contact: sip:1...@1.1.1.1 CSeq: 1 REGISTER Call-ID: 3778139552 Max-Forwards: 70 I'm guessing the 'friendly-scanner' bit is sarcastic, as there is little that is friendly about this behaviour. Has anyone else experienced this? Is this intended as a DOS attack, or is it a dictionary attack? Or something else? What is the best strategy for dealing with it? For now I have started rate limiting SIP connections to Asterisk, but what is a reasonable rate for each host to be allowed? This is a small SOHO installation. Thanks Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DISA and the # key
My dial plan includes a number of service codes that include #, largely because I wanted to mirror BT's codes as I know them already. Example, check the divert on an an extension: *#21 (as opposed to BT's *#21#). Everything was working fine. I've just got around to setting up DISA and find that I can't access these extensions as # is interpreted as the end of the dialled string. Is there any way to alter this behaviour, or do I need to go through the dialplan and change anything with a # in? ___ -- 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] Two peers, same IP and port
Is it possible to have two peers register to Asterisk from the same IP/port combination? I have a Zoom 5821 two port ATA that can support up to 4 VOIP accounts. I want to use it to provide two different extensions on an Asterisk system. In the past I have configured two port ATAs to use a different SIP local port for each account, but the Zoom unit does not appear to allow the SIP local port to be specified on an account by account basis. Can I get the unit to register two separate accounts on Asterisk from the same port and IP? -- Chris Hastie ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Music on Hold restart at beginning for each call
On Tue, 16 May 2006, Tim Sharp [EMAIL PROTECTED] wrote Chris, When I tried background it waited until the message was done before dialing, just like playback. Am I missing something? Wasn't my suggestion :) If I've understood what you're trying to I would go one of two ways: Rather than dial each of the four numbers sequentially, dial them simultaneously. This should hopefully speed up your average pick up time, but will loose any control over preference for who deals with the call. Or investigate queues. I don't have enough people to make it worth my while looking at these, so I've no idea if they're what you need, but they sound like they might be. -- Chris Hastie ___ --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
[Asterisk-Users] X100P not recognised on FreeBSD system
I've just received an OEM Wildcard X100P FXO card. Installing into my FreeBSD 5.4-RELEASE box it doesn't appear to be recognised at all. Since it's the first time I've put a PCI card in this machine I've just dropped a Netgear ethernet card in to make sure there isn't something fundamentally wrong with the motherboard, but that works fine. Is there anything else I should check / try before assuming the X100P is faulty? Output of pciconf -l -v below (after the Netgear card went back to where it belongs): [EMAIL PROTECTED]:0:0: class=0x06 card=0x31161106 chip=0x31161106 rev=0x00 hdr=0x00 vendor = 'VIA Technologies Inc' device = 'VT8375 ProSavageDDR PM266/KM266 CPU to PCI Bridge' class= bridge subclass = HOST-PCI [EMAIL PROTECTED]:1:0: class=0x060400 card=0x0080 chip=0xb0911106 rev=0x00 hdr=0x01 vendor = 'VIA Technologies Inc' device = 'VT8633 Apollo Pro 266 CPU to AGP Controller' class= bridge subclass = PCI-PCI [EMAIL PROTECTED]:16:0:class=0x0c0300 card=0x30381106 chip=0x30381106 rev=0x80 hdr=0x00 vendor = 'VIA Technologies Inc' device = 'VT82x UHCI USB 1.1 Controller (All VIA Chipsets)' class= serial bus subclass = USB [EMAIL PROTECTED]:16:1:class=0x0c0300 card=0x30381106 chip=0x30381106 rev=0x80 hdr=0x00 vendor = 'VIA Technologies Inc' device = 'VT82x UHCI USB 1.1 Controller (All VIA Chipsets)' class= serial bus subclass = USB [EMAIL PROTECTED]:16:2:class=0x0c0300 card=0x30381106 chip=0x30381106 rev=0x80 hdr=0x00 vendor = 'VIA Technologies Inc' device = 'VT82x UHCI USB 1.1 Controller (All VIA Chipsets)' class= serial bus subclass = USB [EMAIL PROTECTED]:16:3:class=0x0c0320 card=0x73801462 chip=0x31041106 rev=0x82 hdr=0x00 vendor = 'VIA Technologies Inc' device = 'VT6202 USB 2.0 Enhanced Host Controller' class= serial bus subclass = USB [EMAIL PROTECTED]:17:0:class=0x060100 card=0x31771106 chip=0x31771106 rev=0x00 hdr=0x00 vendor = 'VIA Technologies Inc' device = 'VT8235 PCI to ISA Bridge' class= bridge subclass = PCI-ISA [EMAIL PROTECTED]:17:1: class=0x01018a card=0x73801462 chip=0x05711106 rev=0x06 hdr=0x00 vendor = 'VIA Technologies Inc' device = 'VT82 EIDE Controller (All VIA Chipsets)' class= mass storage subclass = ATA [EMAIL PROTECTED]:17:5:class=0x040100 card=0x73801462 chip=0x30591106 rev=0x50 hdr=0x00 vendor = 'VIA Technologies Inc' device = 'VT8233/33A/8235/8237 AC97 Enhanced Audio Controller' class= multimedia subclass = audio [EMAIL PROTECTED]:18:0: class=0x02 card=0x738c1462 chip=0x30651106 rev=0x74 hdr=0x00 vendor = 'VIA Technologies Inc' device = 'VT6102 Rhine II PCI Fast Ethernet Controller' class= network subclass = ethernet [EMAIL PROTECTED]:0:0: class=0x03 card=0x73891462 chip=0x8d045333 rev=0x00 hdr=0x00 vendor = 'S3 Graphics Co., Ltd.' device = '86C420 ProSavage DDR' class= display subclass = VGA -- Chris Hastie ___ --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] X100P not recognised on FreeBSD system
On Fri, 19 May 2006, Dinesh Nair [EMAIL PROTECTED] wrote: On 05/19/06 16:30 Chris Hastie said the following: I've just received an OEM Wildcard X100P FXO card. Installing into my FreeBSD 5.4-RELEASE box it doesn't appear to be recognised at all. Since have you downloaded, compiled and installed the zaptel-bsd drivers ? if you haven't, instructions for getting them are at http://www.voip-info.org/wiki-FreeBSD+zaptel for the X100P, you'd need to kldload zaptel.ko and wcfxo.ko, though be warned that the wcfxo.ko driver has not had much development in yonks. Yes, I have these. The modules load, but ztcfg complains ZT_CHANCONFIG failed on channel 1: No such device or address (6) and as I said, it doesn't appear that the card has been recognised by the kernel. -- Chris Hastie ___ --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] X100P not recognised on FreeBSD system
On Fri, 19 May 2006, Dinesh Nair [EMAIL PROTECTED] wrote On 05/19/06 18:57 Chris Hastie said the following: Yes, I have these. The modules load, but ztcfg complains ZT_CHANCONFIG failed on channel 1: No such device or address (6) and as I said, it doesn't appear that the card has been recognised by the kernel. could you try the X100P in anther system to rule out issues with the Via board you're using ? best I can manage is a very old dell optiplex gxi, and it's not recognised in that either. Time to assume a dodgy card? -- Chris Hastie ___ --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] Music on Hold restart at beginning for each call
On Fri, 12 May 2006, Tim Sharp [EMAIL PROTECTED] wrote
I am using the m option on the dial command to play a message instead
of ringing. The message is something like please wait while I try to
locate your party so I need it to start at the beginning for each
call. I think there might be a way in 1.2.x be we are not ready to
upgrade yet so a solution for 1.0.9 is what I am after. Thanks.
The 'm' option is for music on hold, not really announcements. Wouldn't
it be simpler to play the announcement first, then dial? eg
exten = s,1,Playback(local/please_hold_locate)
exten = s,2,Dial(${FOO},20,tm)
--
Chris Hastie
___
--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] Dial Command Reference for SIP channel
On Fri, 12 May 2006, Dave Morrow [EMAIL PROTECTED] wrote Hi all. I was reading a sample config someone had posted relating to call forwarding, and in it, they use a Dial command with components that I cannot find any reference to. Can someone point me to a reference which could explain the difference between Dial(SIP/100|20|Ttr,,wW) and Dial(SIP/100,,wW) Specifically, what is the |20|Ttr ? I cannot seem to find any reference which would indicate this is even a valid format for the SIP channel. Well to my inexpert eye it looks wrong. Are you sure about the those stray commas in the first example? Arguments can be separated by either a comma (,) or a pipe (|). So in your first you are dialling the SIP/100 with a 20 seconds timeout. What's left I think should be TtrwW, not Ttr,,wW. These are the 'options' and are as listed at http://www.voip-info.org/wiki-Asterisk+cmd+Dial. So if you loose those rogue commas, the answer to your what's the difference question is a specific time out value, the ability for both parties to perform transfers and a ringing tone. -- Chris Hastie ___ --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
[Asterisk-Users] Sip domains, contexts and CHECKSIPDOMAIN
Hi
I'm struggling with setting up SIP domains.
If I specify a domain and a context in [general], that context overrides
any set in type=peer blocks elsewhere. This results in incoming calls
from PSTN gateways I use arriving in the wrong context.
If I don't specify a context (which the docs I've found suggest is
valid), then I get:
2006-05-12 07:36:16 WARNING[95290]: chan_sip.c:12539 reload_config:
Empty context specified at line 43 for domain 'domain.com'
and the domain does not appear when I do a sip show domains. It isn't
recognised as local, CHECKSIPDOMAIN doesn't do what I want and calls I
want are rejected.
If I specify autodomain=yes, then the IP address and canonical hostname
of the box are added to the domain list, and sip show domains shows them
with a context of (default). It would appear that for incoming calls
from PSTN gateways at least this does what I want, in that the context
specified in the type=peer block is the one used. However, I can find no
way to add other domains to the list with this '(default)' context. I
particularly want to add the domain name, rather than the host's FQDN,
because my internal SIP clients are all configured to use this. At the
moment, specifying any domain but not that means the clients can't
register, specifying that domain with a context of 'incoming' means the
internal clients can't make out bound calls, and using the context
'outbound' has huge security implications.
I'd like to get sip domains working if only because I'd like to change
the difficult to maintain
exten = s,3,GoToIf($[${SIPDOMAIN} : ${LOCALREGEX}]?4:20)
in my dial plan to something like
exten = s,3,GoToIf($[${CHECKSIPDOMAIN(${SIPDOMAIN})} = ]?4:20)
I'm using Asterisk 1.2.7.1 on FreeBSD 5.4.
Thanks
--
Chris Hastie
___
--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
[Asterisk-Users] Limit on number of SIP channels?
Can anyone tell me what limits the total number of SIP channels available? I'm just setting up an Asterisk system and have found that I seem to be limited to about 5 - is there a configuration option somewhere? call-limit is not set in sip.conf. The symptoms: If I have two incoming SIP connections, each connected to a local SIP phone, an attempt to make a further inbound SIP call gets me to asterisk, but trying to put the call through to a local SIP phone results in a channel unavailable error. Similar things happen when I try to call out. An inbound IAX call can be put through to the SIP phone. But having done that, further inbound SIP calls do nothing at all - no apparent response from *. All the inbound calls are using GSM and there is plenty of bandwidth left. It just seems like I can get to five SIP connections and no more. -- Chris Hastie ___ --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] Test numbers for ENUM (e164.arpa, e164.org, etc.)
On Tue, 22 Nov 2005, John Todd [EMAIL PROTECTED] wrote: I'm looking to build a decent list of test numbers which have ENUM resolution. The numbers I'm looking for should go to a recording, an echo test, or some other feature which does NOT lead to a human. These will be for manual or semi-automatic testing (i.e.: we'll test 10 times in a day, but we won't test continuously.) Any public ENUM-ish tree is fine, but I'm really shooting for e164.arpa. Not mine, so make your own mind up whether its ethical to use them, but I inadvertantly came across these recently. All go to major UK Telcos, and as anyone who has ever been an NTL customer will attest, the chance of them going straight to a human is extremely slight :) +44 800 100 152(BT business customer services) +44 800 052 2000 (NTL residential customer services) +44 800 052 8000 (NTL business customer services) +44 800 052 9000 (NTL business sales) -- Chris Hastie ___ --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
[Asterisk-Users] Death at 2am
Hello I've just installed Asterisk 1.2 onto a FreeBSD system and it is mostly working well. But it dies at 2am every morning. Not quite a complete death, but it seems to loose any ability to communicate with the rest of the world. In /var/log/messages I just see endless entries like this: Nov 21 02:00:13 WARNING[18841] chan_sip.c: No such host: voipfone.co.uk Nov 21 02:00:13 WARNING[18841] chan_sip.c: Probably a DNS error for registration to [EMAIL PROTECTED], trying REGISTER again (after 20 seconds) An attempt to connect to the console leaves asterisk eating up CPU cycles and this in /var/log/messages Nov 20 11:51:25 WARNING[94218] asterisk.c: Accept returned -1: Too many open files A message which reoccurs several hundred times a second. Can anyone either solve this problem for me completely, or at least give me a hint as to the significance of 02:00? Is this an Asterisk thing (most of my configurations are as per install samples), or an underlying OS thing? Thanks -- Chris Hastie ___ --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] Re: Death at 2am
On Mon, 21 Nov 2005, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Chris Hastie [EMAIL PROTECTED] wrote: Hello I've just installed Asterisk 1.2 onto a FreeBSD system and it is mostly working well. But it dies at 2am every morning. Not quite a complete death, but it seems to loose any ability to communicate with the rest of the world. Firstly, look and see what the too many open files are: # lsof -p94218 (or whatever the complaining PID is). I'm assuming FreeBSD has lsof. I don't know, as I use Linux. Thanks. I'll give this ago tomorrow morning Next, examine the cron jobs that happen at 2am, to see if any of them could explain anything. I did look at that. Nothing seems to run at 2am that doesn't run on every other hour. My first inclination was that maybe newsyslog was trying to rotate Asterisk's logs at 2am, but that doesn't look to be the case. I take it that Asterisk (with mostly the standard sample config files) doesn't try to do anything at 02:00 then? Failing that, it could be that something ishappening at your provider everyday at 2am and Asterisk is not coping with it gracefully. I hadn't considered that. Connectivity provider, or VOIP provider (of the latter I have more than one)? I'll experiment with some debug output overnight tonight to see if it gives me any more clues. You could also try specifying 212.187.162.178 temporarily instead of voipfone.co.uk - that would tell you whether the problem is DNS related. I'm pretty sure it is not DNS related. Asterisk seems to loose the ability to connect to anything, irrespective of the direction of the connection. Phones can not connect to Asterisk either. I think the inability of Asterisk to connect to a DNS server is merely one of the symptons of a total inability to talk to anything else at all. -- Chris Hastie ___ --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] Re: Death at 2am
On Mon, 21 Nov 2005, Tony Mountifield [EMAIL PROTECTED] wrote: In article [EMAIL PROTECTED], Chris Hastie [EMAIL PROTECTED] wrote: Hello I've just installed Asterisk 1.2 onto a FreeBSD system and it is mostly working well. But it dies at 2am every morning. Not quite a complete death, but it seems to loose any ability to communicate with the rest of the world. Firstly, look and see what the too many open files are: # lsof -p94218 (or whatever the complaining PID is). I'm assuming FreeBSD has lsof. I don't know, as I use Linux. Thanks. I'll give this ago tomorrow morning Next, examine the cron jobs that happen at 2am, to see if any of them could explain anything. I did look at that. Nothing seems to run at 2am that doesn't run on every other hour. My first inclination was that maybe newsyslog was trying to rotate Asterisk's logs at 2am, but that doesn't look to be the case. I take it that Asterisk (with mostly the standard sample config files) doesn't try to do anything at 02:00 then? Failing that, it could be that something ishappening at your provider everyday at 2am and Asterisk is not coping with it gracefully. I hadn't considered that. Connectivity provider, or VOIP provider (of the latter I have more than one)? I'll experiment with some debug output overnight tonight to see if it gives me any more clues. You could also try specifying 212.187.162.178 temporarily instead of voipfone.co.uk - that would tell you whether the problem is DNS related. I'm pretty sure it is not DNS related. Asterisk seems to loose the ability to connect to anything, irrespective of the direction of the connection. Phones can not connect to Asterisk either. I think the inability of Asterisk to connect to a DNS server is merely one of the symptons of a total inability to talk to anything else at all. -- Chris Hastie ___ --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
