[Asterisk-Users] Asterisk with Cisco
Hi, Does anyone have any real world examples of setting up Asterisk to break out to the PSTN via a Cisco router. I have a 2801 with a PVDM2-8 and -1MFT-E1 connected to a ISDN30 PRI circuit. Is it possible to get Asterisk to talk to the Cisco Router, and what is the best protocol to use. I understand the Cisco talks h232 or SIP, but am unsure as the best way to do this. If anyone has any pointers Id be grateful :) Cheers, Jo ___ --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] Asterisk as a gateway to Index PBX
Steve, Thanks for that, now I know it can be done, do you have any references as to how it is accomplished. Pointers are fine, Im more than happy to RTFM and see what I can work out, but im having trouble locating it :) Many thanks, Jo Steve Rawlings wrote: - Original Message - From: Jo Knight [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 01, 2005 10:32 PM Subject: [Asterisk-Users] Asterisk as a gateway to Index PBX Hi, Is it possible to have an Asterisk act as a gateway to an Index PBX. I would like to migrate users from Index to Asterisk, but need to have some kind of mechanism for the 2 systems to communicate during the migration. I have read that this can be done by installing a dual port PRI card into the Asterisk server, to which the ISDN30 will be connected, and then run a PRI crossover cable into the Index. I have seen this mentioned, but have not been able to find any configuration examples or tips. Can it be done, and if anyone knows how then please point me in the right direction... Thanks, Jo ___ --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 Yes this is quit easy to do. We did it with a TE405P configured as two spans CPE and two spans NET, moved the BT connections (Euro ISDN not DASS) to the Asterisk CPE spans and two E1 crossover cables to connect the two NET spans back to our two INDeX PRI cards. Programmed all inbound on CPE spans to call out on the NET spans so all our DDI just went straight through to the INDeX, it's easy enough to tweak if you want any of the DDI to terminate on the Asterisk. Likewise we programmed anything coming in from the INDeX to go straight out to the CPE spans. We didn't programme anything critical in the INDeX, this way if the Asterisk crashed we simply moved the BT PRI cables back to the INDeX. We did programme string analysis, translation and a route list so INDeX users could dial an access code (something other than 9) followed by an extension number to call extensions on the Asterisk, likewise in the Asterisk, we edited the dialplan so extensions could call into the INDeX over the PRI direct to any extension, cool. The main we reason we did this was to enable us to 'monitor' calls only on specific DDI's remotely using chan_spy within Asterisk. It also gave us a VoIP gateway from INDeX. Steve ___ --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
[Asterisk-Users] Asterisk as a gateway to Index PBX
Hi, Is it possible to have an Asterisk act as a gateway to an Index PBX. I would like to migrate users from Index to Asterisk, but need to have some kind of mechanism for the 2 systems to communicate during the migration. I have read that this can be done by installing a dual port PRI card into the Asterisk server, to which the ISDN30 will be connected, and then run a PRI crossover cable into the Index. I have seen this mentioned, but have not been able to find any configuration examples or tips. Can it be done, and if anyone knows how then please point me in the right direction... Thanks, Jo ___ --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] SRV lookup fails on dyndns wildcard domains
Let me add that it is not really a SRV problem but a DNS problem caused bei SRV lookup. Of course usually there are no SRVs on dyndns domains. jo jo wrote: I know that SRVs have been discussed here in different flavours but I couldn't find anything about this: When calling SIP URIs like [EMAIL PROTECTED] * fails if wildcards are enabled on that domain. Error message from * is: Oct 27 22:54:12 WARNING[1753105]: No such host: mydomain.dyndns.org If wildcards or srv lookup is disabled it works as expected. No problems at all when calling with other clients. Anyone else observed this behaviour? Any solutions? jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SRV lookup fails on dyndns wildcard domains
I know that SRVs have been discussed here in different flavours but I couldn't find anything about this: When calling SIP URIs like [EMAIL PROTECTED] * fails if wildcards are enabled on that domain. Error message from * is: Oct 27 22:54:12 WARNING[1753105]: No such host: mydomain.dyndns.org If wildcards or srv lookup is disabled it works as expected. No problems at all when calling with other clients. Anyone else observed this behaviour? Any solutions? jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SRV lookup fails after DNS update
Hi, SRV records have been working fine until my hoster decided to upgrade BIND. working wrong syntax: _sip._udpSRV1010 5050 mydyndns. correct syntax: _sip._udp IN SRV1010 5050 mydyndns. That kicked of one of my domains completly caused by a syntax error that did no harm to the previous version After inserting the missing IN the zonefile loaded but now I can't query for the SRV record: [EMAIL PROTECTED]:~ host -t SRV _sip._udp.mydomain Host _sip._udp.mydomain not found: 3(NXDOMAIN) Any suggestions, jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] German Asterisk Site
Beierlein Moritz wrote: Hello Asterisk Users, is there a good german site for asterisk? Moritz Hi Moritz, there is * dicussion group at the German IP-Phone forum: http://www.ip-phone-forum.de/ http://www.ip-phone-forum.de/forum/viewforum.php?f=24 jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-Users] Asterisk answering only one (dialed-) Number on a PTMP (German Mehrgeräteanschluss)?
Hi Marco, wendys wrote: Hi, please excuse my poor englisch. Is it possible to connect a (privat Test-Asterisk) to my privat ISDN and allow him to only answer one dialed number? We have 3 up to 10 Numbers on each (Euro-)ISDN (2 b-chanels), it cant't be done by the last Digits cause the numbers are completely different. For Example: I have 3 Numbers (641717, 928752) Is it possible to tell Asterisk (in Extensions.conf?) to Answer 641717 an ignore incomming calls on 928752? I need this solution to work with Asterisk without disconnecting my Girlfriend from the rest of the world. ;-) I did this with an AVM Fritz Card and capi_chan from http://www.junghanns.net/asterisk/ You can define incoming and outgoing MSNs in capi.conf so you won't get in conflict with you other MSNs There is some documentation on it in at voip-info.org (seems to be currently down) jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Call generator
GIBERT Frédéric wrote: Hello, Has someone know a good call generator for asterisk including SIP protocol (freeware if possible)? I need to stress a plateform and I don't find any. How about this? http://sipsak.berlios.de/ jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calling the firefly network?
Martijn van Oosterhout wrote: Is there a way to register with or call the firefly network from an Asterisk server. It would be pretty cool if you could gateway calls onto it. Have a nice day, You can register and dial out with * like on other IAX services. You can verify it by changing the network settings from Firefly to IAX Firefly's network tab. On * the connection gets lost if someone sends an IM via Firefly client. I 've added speex and iLBC to the allowed codecs in iax.conf. I can call and receive to and from freshtel numbers, didn't check PSTN gateway yet. jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GSM AUDIOFiles
jeff quade wrote: Hello: I would like to produce some GSM Prompt audio files for a Telephone Directory Project-- and have hired a freelance audio engineer to record, and edit the actual files-- However the GSM files he gives me to upload into asterisk DO NOT work when played back throgh Stream File or Get Data in my agi. It seems that there may be more than one GSM file type (with header and without, linear compressed, quadratically compressed--etc) Files edited in CoolEdit and saved as gsm, even with the proposed settings don't work for me too. Saved as .wav and converted with sox does the job. jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk real life examples and case studies ?
Peter Mitchell wrote: I can't seem to find the link to examples of asterisk installations for different sized sites. I'm not after specific configuration of the conf files, just an overview on what hardware/chassis cards people are running and what channels - phones etc people are using. here is one from my bookmarks: http://graphics.cs.uni-sb.de/VoIP/en/index.html jo ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Adam, works now :-) Just one further question. In my understanding Firefly's RTP Port is the SIP listen port. So there is no chance to influence the RTP/RTCP Portrange for the audio channel. Please correct me if I'm wrong. jo Adam Hart wrote: I just put up another version - fixed that issue and also added to ability to disable registration to a network. Why it's needed? If you will only be making outgoing calls but still need Firefly to use the login info for calling for lazy ppl: http://www.virbiage.com/firefly/download/firefly-thirdparty.exe Quick run down on various ways of calling - 123 - Firefly will find the network marked as internal and dial 123 on that network +123 - Firefly will find the network marked as external and dial 123 (note no plus) on that network. [EMAIL PROTECTED] - Firefly to find the network named blah and dial 123 sip/[EMAIL PROTECTED] (Firefly will try and find the network for this one as well, otherwise make the call as 'guest') (sip:// also works) Otherwise you can use full asterisk urls eg iax/user:[EMAIL PROTECTED]/extension sip/user:[EMAIL PROTECTED]/extension jo wrote: Thanks Adam, no crash after installing over 1.5 B3388. However changing the SIP RTP Port is still not accepted. jo Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Thanks Adam, no crash after installing over 1.5 B3388. However changing the SIP RTP Port is still not accepted. jo Adam Hart wrote: As Promised, I've released a new version of Firefly (ver 1.8) with IAX SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user - software - firefly), then delete tree from your registry. If that fixes it, send me your exported reg file, there's a bug left to do with some wierd reg entry but everyone just deletes it instead of sending it to me :| Transfers will be in the next version - email me any comments, requested features, bugs and I'll see what I can do -Adam ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Changes???
Just updated from latest CVS and works like before :-) jo Julian Pawlowski wrote: The failure has just been fixed as I saw in mantis: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001738 Thanks a lot! ;D Regards Julian Pawlowski ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Changes???
Philipp,
hope that fits your needs:
This is my (now again) working config for sipgate where XXX is your
my CallerID:
register = XXX:[EMAIL PROTECTED]/XXX
(register = XXX:[EMAIL PROTECTED]/internal_extension works too
but sipgate shows me offline)
[sipgate1]
type=peer (friend works too, didn't check user)
secret=my secret
username=XXX
host=sipgate.de
dtmfmode=inband
context=internal
nat=no
reinvite=no
canreinvite=no
fromuser=XXX
fromdomain=sipgate.net (some want .de but sipgate themselves want .net)
in extensions.conf I have:
[internal]
exten = _777.,1,SetCallerID,XXX
exten = _777.,2,SetCIDName,jo via sip
exten = _777.,3,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,r)
exten = _777.,4,Congestion
[fromsipgate]
exten = XXX,1,Dial(sip/2003CAPI/41:32,20,r)
exten = XXX,2,Voicemail,u2003
exten = XXX,102,Voicemail,b2003
works also for other sip servcices.
For fwd I use IAX, there is working examle on their iax page.
jo
Philipp von Klitzing wrote:
Hi!
The failure has just been fixed as I saw in mantis:
http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001738
Unfortunately that didn't solve my problem - however I am not sure
anymore that this is related, and maybe I just have a basic
misunderstanding concerning type=peer and type=user.
Question:
Why do I need type=peer for both cases, e.g. incoming AND outgoing calls?
I am really confused here - or someone/something else is... ;-
1. I want to be able to dial out to FWD with a Dial() statement in
extensions.conf that does not include username or password so that these
do not show up in the CDRs, e.g. using
Dial(SIP/[EMAIL PROTECTED])
2. The above only works if FreeWorld-out-user1 is of type=peer (and not
type=user)
3. On an incoming FWD call * unfortunately always matches the host to the
[FreeWorld-out-user1] section instead of the [FreeWorld-incoming]
section, which is kind of logic becase both are peers. Then
authentication fails because the calling user naturally doesn't have the
correct password for FreeWorld-out-user1.
Cheers, Philipp
[FreeWorld-incoming]
context=from-FreeWorld
type=peer
host=fwd.pulver.com
[FreeWorld-out-user1]
type=peer
secret=
username=yy
fromuser=yy
host=fwd.pulver.com
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Re: [Asterisk-Users] Freenet iPhone w/Asterisk
Oliver, you should be able to connect * with the same settings required for softphones. http://www.freenet.de/freenetiphone/sip_telefone/index.html Firewall problems depends on your individual situation, a search in this list or browsing fwd's web forum may find a solution. But why would you do that? freenet iPhone is a rather prorietary service without any gateways except PSTN (which is limited to freenet DSL users). They don't even offer DIDs. my1 cent jo [EMAIL PROTECTED] wrote: Has anybody tried to use Freenet's Germany based iPhone Service with Asterisk? Maybe even from behind a NAT? Freenet seems to use SER ... but I can not get a connection to their SIP proxy from Asterisk going through a NATed firewall. Asterisk -SIP- Firewall with NAT -SIP- Freenet iPhone server Thanks, Oliver ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FireFly doesn't work with 3rd party anymore
just keep on using the older version ... jo brian wrote: Just an FYI FireFly no longer works with anything but the FireFly network. No more SIP, No more IAX. It was a damn good IAX client... too bad its crap now. bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to create channel of type 'CAPI'
Since upgrading from stable to latest cvs I can't place CAPI calls (AVM Fritz/chan_capi-0.3.1) Did I miss something that has to be changed in configfiles? Also tried to recompile chan_capi which run into an error. capi info shows me: Contr1: 2 B channels total, 2 B channels free. Any suggestions to these logfile snippets jo * to ISDN -- Accepting AUTHENTICATED call from 192.168.1.8, requested format = 2, actual format = 2 -- Executing Dial([EMAIL PROTECTED]/11, CAPI/41:33:b) in new stack May 25 19:15:25 NOTICE[557071]: app_dial.c:674 dial_exec: Unable to create channel of type 'CAPI' == Everyone is busy at this time ISDN to * May 25 19:18:59 NOTICE[81926]: chan_capi.c:1895 capi_handle_msg: CONNECT_IND ID=003 #0x0038 LEN=0035 Controller/PLCI/NCCI= 0x101 CIPValue= 0x10 CalledPartyNumber = 8141 CallingPartyNumber = A8030 CalledPartySubaddress = default CallingPartySubaddress = default BC = 80 90 a3 LLC = default HLC = 91 81 AdditionalInfo = default == CONNECT_IND (PLCI=0x101,DID=41,CID=30,CIP=0x10,CONTROLLER=0x1) == DISCONNECT_IND PLCI=0x101 REASON=0 --- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Downgrading Asterisk
Sorry, no solution but same problem. Downgrading brings this message on Suse9.0, 2.4.21: [app_txtcidname.so]May 25 23:28:42 WARNING[16384]: loader.c:240 ast_load_resource: /usr/lib/asterisk/modules/app_txtcidname.so: undefined symbol: ast_get_txt May 25 23:28:42 WARNING[16384]: loader.c:408 load_modules: Loading module app_txtcidname.so failed! jo Nik Martin wrote: I upgraded to the latest HEAD version of asterisk, and all IAX calls started sounding choppy. It was suggested on the IRC channel that I go back to asterisk -stable to determine if that fixes it. Is downgrading as simple as upgrading? Because now, -stable builds fine, but I get an error on the asterisk console when starting, something about ast_get_txt not found. Recompiling and installing asterisk HEAD afterwards works just fine. As a side note, I recently upgraded my kernel to 2.4.26 and had an issue with old kernel headers, but have since resolved that prior to trying this downgrade. Any ideas? Nik ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Unable to create channel of type 'CAPI'
Hi Stefan, that's been the solution. Thanks a lot! (I still wonder why I was able to compile against the stable 1 without this patch.) jo Stefan Tichy wrote: On Tue, May 25, 2004 at 07:40:51PM +0200, jo wrote: Since upgrading from stable to latest cvs I can't place CAPI calls (AVM Fritz/chan_capi-0.3.1) Did I miss something that has to be changed in configfiles? Also tried to recompile chan_capi which run into an error. Did you apply the patch from http://lists.digium.com/pipermail/asterisk-users/2004-April/044125.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ISDN WAN ISDN bridge possible?
Patrick, doe a google search for ISDN over IP, maybe that's your solution. jo Patrick Stuckenberger wrote: Hi list, is it possible to create something like a ISDN-WAN-WAN-ISDN bridge? We have to change our location, but our number and the telephone system should shoulb stay the same. kind regards, Patrick Stuckenberger ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
