Re: [asterisk-users] Problems sending voicemail emails
You also need to specify the port so telnet mx1.datagrama.net 25 return is the command to use. db From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 10:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Problems sending voicemail emails Hi Michelle, If I try telnet mx1.datagrama.net I have no answer, I get: Trying 212.9.65.110... ¿? 2009/8/24 Michelle Dupuis supp...@ocg.ca Start with simple mail testing (forget asterisk) Does mx1.datagrama.net http://mx1.datagrama.net/ accept messages for testu...@mydomain.com ? Try a telnet session first... _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joan Antoni Terre Sent: Monday, August 24, 2009 9:56 AM To: Asterisk Users List Subject: [asterisk-users] Problems sending voicemail emails Hi everybody, I'm trying my Asterisk to send emails when a new message arribes to a voicemail user but no email arribes. my voicemail configuration is the following: VOICEMAIL.CONF: [general] format=wav serveremail=aster...@mydomain.com attach=yes maxmsg=20 maxsecs=180 minsecs=3 maxsilence=10 silencethreshold=128 maxlogins=3 fromstring=My Asterisk When I look at maillog file, this is what I get: * n7OCivth003603: from=root, size=5340, class=0, nrcpts=1, msgid=asterisk-1-227856683-222-3...@myserver, relay=r...@localhost * n7OCiw9W003604: from=r...@localhost.localdomain, size=5473, class=0, nrcpts=1, msgid=asterisk-1-227856683-222-3...@prosima, proto=ESMTP, daemon=MTA, relay=MYSERVER [127.0.0.1] * n7OCivth003603: to=Test User 1 testu...@mydomain.com, ctladdr=root (0/0), delay=00:00:06, xdelay=00:00:05, mailer=relay, pri=35340, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent (n7OCiw9W003604 Message accepted for delivery) * n7OCiw9W003604: to=testu...@mydomain.com, ctladdr=r...@localhost.localdomain (0/0), delay=00:00:01, xdelay=00:00:01, mailer=esmtp, pri=125473, relay=mx1.datagrama.net http://mx1.datagrama.net/ . [212.9.65.111], dsn=5.1.8, stat=User unknown * n7OCiw9W003604: n7OCj49W003606: DSN: User unknown * n7OCj49W003606: to=r...@localhost.localdomain, delay=00:00:01, xdelay=00:00:00, mailer=local, pri=36710, dsn=2.0.0, stat=Sent I don't understand why it looks as the message is been sent (Message accepted for delivery) but then I get the message dsn=5.1.8, stat=User unknown and fiinally I get the message Sent but I don't receive any email. do I have to change any configuration? Many thanks in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com/ -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net http://www.astricon.net/ 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 -- AstriCon 2009 - October 13 - 15 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] DNS Query Overload
Another item is the sequence for lookups! So have you confirmed that your nsswitch.conf file has been set to look at /etc/hosts first then dns? Dave -Original Message- From: Adam Lovegrove [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DNS Query Overload Date: Sun, 21 Sep 2008 23:24:28 +1000 G'day, I've got this problem too. I've tried altering my /etc/hosts files as per the suggestion, but my DNS server is still being sent an A query - for every call. Please help! I'm using 1.4.21.2. Cheers Adam On Sat, Jun 28, 2008 at 12:31 AM, Andres [EMAIL PROTECTED] wrote: I have seen that before. If I remember correctly, the solution was to put the IP Address of the Box in the /etc/hosts file. Like for example: 192.168.2.1asterisk.localhost If you have multiple interfaces with private IP addresses then put them all in the file. Andres http://www.neuroredes.com Mik Cheez wrote: I'm finding that my Asterisk server is bombarding my DNS servers with lookups like the following: Queries 5060-b7bfce38: type A, class IN Name: 5060-b7bfce38 Type: A (Host address) Class: IN (0x0001) One call alone has a handful of requests to our server, simply looking for an A record for something like '5060-b7bfce38' (listed above). The DNS server immediately responds with No such name. I use only SIP on my box, and even if I just have the call go to hangup it does this. My SIP.conf contains 'srvlookup=no' in the general section. Any thoughts or suggestions? Best regards, Mik ___ -- 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 ___ -- 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] Really WEIRD: can register but can not call!
-Original Message- From: ims.asuser ims.asuser [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Really WEIRD: can register but can not call! Date: Mon, 25 Aug 2008 12:26:45 +0200 Hi all, I have a very weird problem. I have 2 users (103 and 105). They are able to register in Asterisk, but they can not call each other. Hereunder is the outcome: openwrt3*CLI -- Registered SIP '103' at 192.168.3.9 port 6127 expires 3600 -- Saved useragent eyeBeam release 3010n stamp 19039 for peer 103 openwrt3*CLI openwrt3*CLI -- Registered SIP '105' at 192.168.3.6 port 8377 expires 3600 -- Saved useragent eyeBeam release 3010n stamp 19039 for peer 105 openwrt3*CLI openwrt3*CLI -- Executing Dial(SIP/105-0ead, SIP/l03) in new stack Jan 1 00:19:26 WARNING[498]: chan_sip.c:1407 create_addr: No such host: l03 Jan 1 00:19:26 NOTICE[498]: app_dial.c:764 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time openwrt3*CLI openwrt3*CLI -- Timeout on SIP/105-0ead == CDR updated on SIP/105-0ead -- Executing Goto(SIP/105-0ead, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/105-0ead, demo-thanks) in new stack Jan 1 00:19:36 WARNING[498]: file.c:475 ast_openstream: File demo-thanks does n ot exist in any format Jan 1 00:19:36 WARNING[498]: file.c:787 ast_streamfile: Unable to open demo-tha nks (format ulaw): No such file or directory Jan 1 00:19:36 WARNING[498]: app_playback.c:83 playback_exec: ast_streamfile fa iled on SIP/105-0ead for demo-thanks -- Executing Hangup(SIP/105-0ead, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/105-0ead' The show sip registry command shows that no users are registered. That's really WEIRD! Please see the sip.conf and extension.conf files. sip.conf: [general] context=default ; Default context for incoming calls ;recordhistory=yes ; Record SIP history by default ; (see sip history / sip no history) ;realm=mydomain.tld ; Realm for digest authentication ; defaults to asterisk ; Realms MUST be globally unique according to RF ; Set this to your host name or domain name port=5060 ; UDP Port to bind to (SIP standard port is 5060 bindaddr=x.x.x.x ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls ; Note: Asterisk only uses the first host ; in SRV records ; Disabling DNS SRV lookups disables the ; ability to place SIP calls based on domain ; names to some other SIP users on the Internet [103] ; ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! type=friend username=103 ; Authorization User dans X-Lite secret=1234 callerid=Philippe 103 ; nom et numéro affichés dans le X-Lite appelé l context=default host=dynamic nat=no ; X-Lite is behind a NAT router canreinvite=no; Typically set to NO if behind NAT disallow=all; désactive tous les codages sauf ceux spécifiés ci-aprè allow=gsm ; GSM consumes far less bandwidth than ulaw allow=ulaw allow=alaw [105] ; ;Turn off silence suppression in X-Lite (Transmit Silence=YES)! type=friend username=105 ; Authorization User dans X-Lite secret=1234 callerid=Khalid 105 ; nom et numéro affichés dans le X-Lite appelé lor context=default host=dynamic nat=no ; X-Lite is behind a NAT router canreinvite=no; Typically set to NO if behind NAT disallow=all; désactive tous les codages sauf ceux spécifiés ci-aprè allow=ulaw allow=alaw extension.conf: [default] ; context par défaut des utilisateurs SIP répertoriés dans sip.c exten = 103,1,Dial(SIP/l03) exten = 105,1,Dial(SIP/l05) ___ -- 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 Your extensions are listed as SIP/l03 and SIP/l05 and should be SIP/103 and SIP/105. Plus a problem with some recorded files. Regards, Dave ___ -- 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:
Re: [asterisk-users] Really WEIRD: can register but can not call!
You need to reload the configurations, either by reload command or restart asterisk. Dave -Original Message- From: ims.asuser ims.asuser [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED], [EMAIL PROTECTED], [EMAIL PROTECTED] Subject: Re: [asterisk-users] Really WEIRD: can register but can not call! Date: Mon, 25 Aug 2008 16:17:34 +0200 That's right, I used a 'l' instead of '1'! Thank you. I've made the modification on extension.conf (there's nothing to change on the sip.conf) but the call can not go through...is there another file to modify? The new outcome is: -- Executing Dial(SIP/105-6298, SIP/l03) in new stack Jan 1 00:54:38 WARNING[606]: chan_sip.c:1407 create_addr: No such host: l03 Jan 1 00:54:38 NOTICE[606]: app_dial.c:764 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time -- Saved useragent eyeBeam release 3010n stamp 19039 for peer 103 -- Timeout on SIP/105-6298 == CDR updated on SIP/105-6298 -- Executing Goto(SIP/105-6298, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/105-6298, demo-thanks) in new stack Jan 1 00:54:48 WARNING[606]: file.c:475 ast_openstream: File demo-thanks does n ot exist in any format Jan 1 00:54:48 WARNING[606]: file.c:787 ast_streamfile: Unable to open demo-tha nks (format ulaw): No such file or directory Jan 1 00:54:48 WARNING[606]: app_playback.c:83 playback_exec: ast_streamfile fa iled on SIP/105-6298 for demo-thanks -- Executing Hangup(SIP/105-6298, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/105-6298' -- Saved useragent eyeBeam release 3010n stamp 19039 for peer 105 extension.conf [default] exten = 103,1,Dial(SIP/103) exten = 105,1,Dial(SIP/105) Thank you all! Khaldon 2008/8/25 Pavel Jezek [EMAIL PROTECTED] you should issue 'sip show peers' command to see, if your phones are available, put 'qualify=yes' in your sip.conf 'sip show registry' command is usefull to see if your _asterisk_ is registered to some another sip server, eg. voip provider.. PJ David Boyd wrote: -Original Message- From: ims.asuser ims.asuser [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: [asterisk-users] Really WEIRD: can register but can not call! Date: Mon, 25 Aug 2008 12:26:45 +0200 Hi all, I have a very weird problem. I have 2 users (103 and 105). They are able to register in Asterisk, but they can not call each other. Hereunder is the outcome: openwrt3*CLI -- Registered SIP '103' at 192.168.3.9 port 6127 expires 3600 -- Saved useragent eyeBeam release 3010n stamp 19039 for peer 103 openwrt3*CLI openwrt3*CLI -- Registered SIP '105' at 192.168.3.6 port 8377 expires 3600 -- Saved useragent eyeBeam release 3010n stamp 19039 for peer 105 openwrt3*CLI openwrt3*CLI -- Executing Dial(SIP/105-0ead, SIP/l03) in new stack Jan 1 00:19:26 WARNING[498]: chan_sip.c:1407 create_addr: No such host: l03 Jan 1 00:19:26 NOTICE[498]: app_dial.c:764 dial_exec: Unable to create channel of type 'SIP' == Everyone is busy/congested at this time openwrt3*CLI openwrt3*CLI -- Timeout on SIP/105-0ead == CDR updated on SIP/105-0ead -- Executing Goto(SIP/105-0ead, #|1) in new stack -- Goto (default,#,1) -- Executing Playback(SIP/105-0ead, demo-thanks) in new stack Jan 1 00:19:36 WARNING[498]: file.c:475 ast_openstream: File demo-thanks does n ot exist in any format Jan 1 00:19:36 WARNING[498]: file.c:787 ast_streamfile: Unable to open demo-tha nks (format ulaw): No such file or directory Jan 1 00:19:36 WARNING[498]: app_playback.c:83 playback_exec: ast_streamfile fa iled on SIP/105-0ead for demo-thanks -- Executing Hangup(SIP/105-0ead, ) in new stack == Spawn extension (default, #, 2) exited non-zero on 'SIP/105-0ead' The show sip registry command shows that no users are registered. That's really WEIRD! Please see the sip.conf and extension.conf files. sip.conf: [general] context=default ; Default context for incoming calls
[asterisk-users] Control of individual call legs
Hello , is it possible to control multiple legs (channels) of a call individually, ie. call 1 -- incoming call connected to IVR call 2 -- outgoing call to party a made via manager interface call 3 -- outgoing call to party b made by call-script I would like to allow the caller on call1 to be able to decide if they want to be connected to call2, call3, or generate an additional call4 for there use, and I don't want to use a meeting room. Thanks for any tidbits! Dave ___ -- 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] test message please do not reply and clog up the list
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Re: [asterisk-users] lots of warnings from translate.c
On Tue, 2008-04-22 at 12:28 +0200, Francesco Castellano wrote: We have a couple of servers with asterisk 1.4.19 and zaptel 1.4.10, acting as gateways from SIP to ISDN PRI interfaces. Each has one Digium TE420 (with hardware echo cancellation) and one TC400B for transcoding, in order to handle 60/90 contemporary calls in peak hours. In my logs there are hundreds of thousand warnigs per day like these: transcode.c: no samples for lintoulaw transcode.c: zapg729toalaw did not update samples ### Could you suggest me what are the possible causes for that? Are they signs of bad audio quality? Any ideas for resolving these issues? In addition I can say that we are using a quite large jitter buffer in zapata.conf: jitterbuffers=16 (= 0.32s) Moreover, it uses the fixed implementation, because when I tried the adaptive one I experienced one-way audio. Finally I have to note that, using a Siemens IP phone (G.729 no AnnexB) in conditions of no load on servers, I could replicate non-deterministically (sigh!) each of these problems, with a very noisy audio, and a annoying period of silence during the first seconds of call. Regards, Francesco PS. Previous versions of asterisk and zaptel presented an identical situation. Have you tried additional types of phones and if so can you produce the same non-deterministic problems? Dave ___ -- 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] UK FXO hangup detection with a twist
On Mon, 2008-03-31 at 16:25 +0100, Steve Davies wrote: On 31/03/2008, Mike Dent [EMAIL PROTECTED] wrote: On 31/03/2008, Steve Davies [EMAIL PROTECTED] wrote: The twist? We actually have far-end hangup detection working fine, and that seems to be where the problem lies for most people. Our problem seems to be with requesting a hangup from our end reliably. If we originate the call, we can hang it up. This suggests to me that the Sangoma A200D is sending the correct hangup signaling. This way round, it is 100% reliable. If we accept a call originated elsewhere, then we cannot hang it up. Only the call originator seems to be able to do that. The upshot is that if asterisk hangs-up a line, and then tries to re-use it for an outbound call before the remote has disconnected, we are simply re-connected to the original caller, and start to play DTMF at them! Has anyone experienced this before? Anyone found a solution? People regularly use this feature to answer a call, then decide they need to run upstairs to speak etc, so they put the receiver down, go upstairs (or wherever) and pick up the handset to speak. It dates back many years and I should think is designed in to the system in the UK. Not sure on modern exchanges how long it would take for the line to clear. I do the same myself, but for PABX use, that feature must be fatal! The line clearing time is long... I waited a couple of minutes at least. Is it possible to turn it off (call BT and ask for a certain feature to be enabled/disabled) or to shorten the line-clearing time to zero? Or perhaps Asterisk is able to detect the line state or the dialtone and act correctly to avoid re-using an open channel. In fact, the obvious way to do this might be if Asterisk could set the channel state to hanging up and wait for the far end signal (loop disconnect) that the line has actually been disconnected. This is a bit of Zaptel that I've never looked at, so I have no clue if what I am suggesting is even slightly possible. Steve ___ -- 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 You should ask for ground start signaling. This will resolve your issues. Dave ___ -- 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] Sangoma FXO EC vs Rhino FXO EC
On Wed, 2008-02-20 at 17:34 -0600, Tilghman Lesher wrote: On Wednesday 20 February 2008 16:42:59 Steve Totaro wrote: On Tue, Feb 19, 2008 at 8:23 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Tuesday 19 February 2008 18:27:32 Michael J. Liberatore wrote: Hi all, I am a huge fan of Sangoma cards after having many problems with digium cards and then switching to sangoma cards and them giving me excellent support with excellent results. I would recommend that you give the Digium cards another shot. There is zero risk now, as Digium cards are now backed with a 5 year warranty and a 100% money-back guarantee: Digium will make it work, or you get your money back. http://www.digium.com/en/company/riskfree-facts.php Is Digium's money back guarantee and five year warranty retroactive? It's a new guarantee on cards sold after February 1st, 2008. What had to happen first was to address the legitimate concerns that customers had about our boards, and that's been done. All of the new boards should not have any of the problems that previously plagued customers, and we'd really like to see customers who had problems in the past come back and give it another try. And we're confident enough about our boards that we're willing to provide a full 5 years of warranty on these new boards AND back them up with a 100% money back guarantee. Unfortunately, the old boards that are already out of warranty are just that -- out of warranty. The boards that have given our customers trouble have been discontinued, so going forward, everybody should be good. I realize that Michael's customer is a bit miffed about old boards that aren't working, but the current boards should now work for everybody. The only investment that you'd really be risking is time, and unfortunately, we can't manufacture that for you. ;-) A trade in policy would be a great incentive for people to try digium again. In particular if you are already happy with a current vendor, and have no reason to spend more money just to prove to yourself that Digium now has good product, it could be construed as potentially throwing good money after bad. just my 2cents worth. Dave ___ -- 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] txfax not working with spandsp
On Fri, 2007-12-21 at 18:41 +0800, Dinesh Nair wrote: the attached log with verbose=6 and debug=6 refers. we've got a sangoma A104 (no hwec) with PRI ports 1 3 loopbacked to each other. we're trying to have txfax send out on one of those pri ports with rxfax listening on the other side, hence having asterisk send a fax to itself. we however have bad, and i mean really bad (10%) success rates. we're currently using asterisk 1.2.24 with spandsp 0.0.4-20071214 (snapshot of 14/12/07) and we keep getting Fax send not successful - result (25) No response after sending a page. errors. ECM is turned on in both app_txfax.c and app_rxfax.c. from what we gather just reading the code, time T4 expires in txfax because apparently rxfax is not sending a response back out, and thus after the maximum message retries (3) txfax just drops the call, leading rxfax to say that the call was dropped prematurely. does anyone know what's going on here, and if there is a version of spandsp which could work in this scenario ? ___ --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 Do you have any timing issues such as slips or bi-polar violations taking place. It sounds like there are dropped packets or something. dave ___ --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] Upgrade to Asterisk 1.4 - it's one year's old!
Thanks for your thoughtful response. Dave On Sun, 2007-12-16 at 10:43 -0600, Tilghman Lesher wrote: On Saturday 15 December 2007 12:14:29 David Boyd wrote: On Sat, 2007-12-15 at 10:51 -0600, Tilghman Lesher wrote: Of course, all of these deprecations should be covered in UPGRADE.txt, so please read that file every time you upgrade to a new version. It will contain everything that has changed in a possibly incompatible way. And if you find something that broke that wasn't in this file, please let us know, so we can revise that file. We may not have gotten everything, but we do try. So if I read you correctly, all of the pain of the upgrade is due to lack of effort on the participants part! I wouldn't say all of it, but it would be a lot easier if people paid attention to the deprecation notices and resolved them. The whole point of deprecating methods is to allow people a transitional period in which they stop using said method and move to its replacement. This seems a whole lot like the attitude of proprietary vendors when they don't want to support a feature that is outside the scope of what they want to maintain. I thought this was an open source project that would allow participants to have a voice in what is or isn't included in a new release. Even an non developing end user provides valuable benefit to the project in QA and bug information to improve the project as a whole. Most (With exceptions) projects have a bit more interest in what the user community wants or needs in a package. The attitude of this project seems to be If you want it code it yourself, however if it something that doesn't map to the ideas of what Digium wants then it will never make it into the official release. Digium is a company; it does not want anything. The developers of the project, of which Digium has sponsored a great many, most of whom were developers prior to being employed by Digium, get to make those types of calls. Do you see the distinction? One of the nice things about working for Digium is that I maintain my individual perspective as a developer; we do not engage in groupthink. I don't understand why so much community support is placed into the project considering that the typical end user is treated like a second class citizen. I can't think of a single software project where the typical end user is anything but. Every open source project is not a democracy; they are meritocracies. That is, the degree to which your opinion matters is the degree to which you are able to contribute. And this isn't just code writers, either. People who put forth the effort to document the code also get a kudos and karma, as do people who report bugs, suggest fixes, and give feedback on candidate patches. To a lesser extent, knowledgable users who help on the various forums and business leaders who sponsor developers to work on Asterisk also have a greater voice than the typical end user. And that's true for closed source, as well. When was the last time that an end user asked for and received a new feature from Microsoft? So Digium, (I address the company since Tilghman now works for you) do you have any plans to query the user community and determine what a typical end user of the product needs? With the knowledge and skill that exists in your organization it would seem trivial to put something in place to allow user feedback not only developer feedback for release direction. It is extremely insulting for you to try to address my employer, when we're discussing code practice. For one thing, the company (though legally a person) does not generally respond on these lists. And secondly, as I mentioned before, all developers maintain their individual perspective, so when I make points on here, I do so as an individual contributor. If you have an issue with the way that I have approached something, then please talk to me. Trying to go over my head is rude and unlikely to produce better results. As far as user feedback, there are multiple forums that exist that will influence individual developers, to a certain extent, which are the -dev list (please discuss code or policy, NOT user-level assistance; that's what this list is for), the #asterisk-dev channel on Freenode (same condition applies; use #asterisk for user-level questions), and the bugtracker (which is for reporting bugs, inconsistencies, and other things that relate to execution, not policy, which should be discussed on the mailing lists). Of course, if you want your voice heard more loudly, then contribute some of your efforts towards furthering the project. Complaints are always heard more critically when they come from somebody who has made the effort to give back in some way. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list
Re: [asterisk-users] Upgrade to Asterisk 1.4 - it's one year's old!
On Sat, 2007-12-15 at 10:51 -0600, Tilghman Lesher wrote: On Saturday 15 December 2007 10:02:23 Rob Hillis wrote: One of the biggest barriers to upgrading are the number of little gotchas in syntax changes that can make an upgrade from 1.2 to 1.4 quite painful. After the pain I went through upgrading to 1.4, I've always been recommending to people to think twice about upgrading if 1.2 does what they require. Many of the changes may have been seen as minor - one or two changes are to be expected, but I ran into at least half a dozen - mostly variable changes if I recall correctly - things such as deprecating CALLERIDNUM in favour of CALLERID(num). Some of the breakage was minor (e.g. loss of caller ID processing) but some of them resulted in calls being dropped in unpredictable places. All I can say is with 1.6, if a change is made that causes something that worked in 1.4 not to work in 1.6, please think twice, three times or four times before making the change, or making the change in such a way that it won't break dialplan stuff from 1.4. If anything broke from the transition from 1.2 to 1.4, it is because you were using something that was deprecated in 1.2. What we had attempted to do in deprecation modes was to print the warning ONCE for each deprecated operation, per Asterisk startup. I think that this was much too conservative. It is very easy to miss that deprecation warning, since it occurs so few times. Of course, the opposite side is that we don't want deprecation warnings to fill up your logs, so there's a balancing act here. But we could probably do with making the deprecation warnings a bit more prominent and print them multiple times (for example, every 10th usage). That should make it more clear that there's something to change. Of course, all of these deprecations should be covered in UPGRADE.txt, so please read that file every time you upgrade to a new version. It will contain everything that has changed in a possibly incompatible way. And if you find something that broke that wasn't in this file, please let us know, so we can revise that file. We may not have gotten everything, but we do try. Hello Tilghman, So if I read you correctly, all of the pain of the upgrade is due to lack of effort on the participants part! This seems a whole lot like the attitude of proprietary vendors when they don't want to support a feature that is outside the scope of what they want to maintain. I thought this was an open source project that would allow participants to have a voice in what is or isn't included in a new release. Even an non developing end user provides valuable benefit to the project in QA and bug information to improve the project as a whole. Most (With exceptions) projects have a bit more interest in what the user community wants or needs in a package. The attitude of this project seems to be If you want it code it yourself, however if it something that doesn't map to the ideas of what Digium wants then it will never make it into the official release. I don't understand why so much community support is placed into the project considering that the typical end user is treated like a second class citizen. So Digium, (I address the company since Tilghman now works for you) do you have any plans to query the user community and determine what a typical end user of the product needs? With the knowledge and skill that exists in your organization it would seem trivial to put something in place to allow user feedback not only developer feedback for release direction. My 2 cents, ok 25 cents, Dave ___ --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] OT: Best firmware for Linksys Router that is SIP AWARE
On Mon, 2007-11-26 at 08:08 -0600, Erik Anderson wrote: On Nov 26, 2007 7:51 AM, Dovid B [EMAIL PROTECTED] wrote: Hi, I am currently playing with DD-WRT and I like it. I am looking for something that is more SIP Aware. Anyone know one those that are out there ? Dovid - what exactly are you hoping this sip aware firmware will do that dd-wrt doesn't? I've been using dd-wrt in combination with various SIP ITSPs for several years and have had no problems - just add the necessary port forwards and a few traffic shaping rules and it works just fine. I do know that they (the dd-wrt people) have a voip edition of dd-wrt available. I'm not sure what additional functionality it has over the standard version, though. -erik Erik, I struggle with the traffic shaping rules, would you be willing to provide additional details as to what you have done in past? Any additional information would be greatly appreciated. Thanks, Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk and installing chan_h323.so rpm
- Original Message - From: Bincy K. Philip [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, November 08, 2007 2:13 PM Subject: [asterisk-users] asterisk and installing chan_h323.so rpm Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same files? I had this issue in the past. I do not remember what I did to resolve it. In the end I went with the h323 channel driver located in the asterisk add-ons. It was a lot easier to work with and worked with out any issues. It seems to me that you need to run ldconfig so as to pick up the location of the specified libraries. Do a google on it to see syntax of man ldconfig. You could also hack things by linking to the libraries from the expected directories (What the rpm is expecting) if executing ldconfig doesn't work. db ___ --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] SIP response time in Asterisk
On Fri, 2007-10-26 at 11:12 -0700, John Riek wrote: I need to know how fast a sip device needs to respond to an INVITE sip message from asterisk before asterisk retransmits the INVITE message again. Thanks Snip --- 7.2.1 INVITE received When an INVITE request is received by the gateway, a 100 Trying response MAY be sent back to the SIP network indicating that the gateway is handling the call. The necessary hardware resources for the media stream MUST be reserved in the gateway when the INVITE is received, since an IAM message cannot be sent before the resource reservation (especially TCIC selection) takes place. Typically the resources consist of a time slot in an E1/T1 and an RTP/UDP port on the IP side. Resources might also include any quality-of-service provisions (although no such practices are recommended in this document). ** After sending the IAM the timer T7 is started. The default value of T7 is between 20 and 30 seconds. The gateway goes to the 'Trying' state. ** ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Polycom Provisioning Tool
Michael, way cool. Works in WINE also :) db On Wed, 2007-10-24 at 23:09 -0400, Michael Munger wrote: Not sure if one exists, but someone had asked me for this a while ago. Here it is! My Polycom Provisioning Tool. Notice the version is 0.0.1. Just a concept program (but it works well). I am open for suggestions, feature additions, and bug fixes. Email me with any requests. I want to improve this to make it really useful for the community, so let me know what you think. http://www.wintrisk.com/ppt.html Michael Munger, dCAP High Powered Help, Inc [EMAIL PROTECTED] 404-438-2128 x 101 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --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] DS3 Interface
What a waste of time... dave On Fri, 2007-10-12 at 09:35 -0600, Stephen Bosch wrote: Brian West wrote: And what was the purpose of this? So that we would realize who we were talking to. :) -Stephen- ___ --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] swift.conf - cepstral voice quality adjustment options
On Tue, 2007-09-25 at 10:57 -0400, Larry Costigan wrote: Hi all, I hope that I'm not breaking protocol too much by posting a message in this group about a problem that I'm having with an Asterisk Business Edition installation, but the reason that I'm posting here is because the problem that I'm having isn't really with the Business Edition, it is with the Cepstral text to speech product that I'm using with it, and also because this group has so much more activity that I'm really hoping that somewhere in this great Asterisk community there are some clever people who might have some good suggestions to help me improve the voice quality on this system. SNIP Here is a link that provides a snippet of info for you. http://www.mezzo.net/asterisk/app_swift.html I would think that the buffer setting might be of importance! dave ___ 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] Digium Appliance
Hi Mat, i have been working with the aa50 for a couple of weeks now. They are slick looking devices that still have a few bugs. I tried to use the device like an end user without previous knowledge of Asterisk or the asteriskGUI, and can say right off that a typical person will not be able to use the device by gui only. The interface does not create all entries required to configure either outbound routing or DID, outbound caller id for either sip or IAX looks to the fullname field in the users.conf file rather than CID entry They are working to correct the issues, however as of yet no known release date for firmware fixes. Having said that if you want to edit files via the gui by hand and make appropriate changes then the device seems to work ok. Did have an issue where after reboot the system would register an IAX trunk with the provider but outbound calls would fail until you kicked the system to force a new registration. A couple of times changes that were saved at the home page failed to commit to the flash card, replaced the flash and have not seen that issue again, but Little things that make me oogee about putting into a customer location right now. db On Wed, 2007-09-12 at 12:52 -0400, Matt wrote: Hi, Has anyone actually gotten their hands on an appliance yet? If so, how robust and working are they? Any issues? ___ 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 ___ 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] 99 bottles of beer
On Thu, 2007-08-16 at 19:38 -0600, Steve Murphy wrote:
On Thu, 2007-08-16 at 07:56 -0400, Russell Bryant wrote:
Gordon Henderson wrote:
; *99:
; 99 bottles of beer on the wall.
exten = *99,1,Noop(99 Bottles of beer on the wall)
exten = *99,n,Answer()
exten = *99,n,Set(bottles=99)
exten = *99,n(loop),Noop(There are ${bottles} bottles of beer on the
wall)
exten = *99,n,SayNumber(${bottles})
exten = *99,n,Noop(Take one done and pass it round and there's)
exten = *99,n,Set(bottles=$[${bottles}-1])
exten = *99,n,Noop(${bottles} bottles of beer on the wall)
exten = *99,n,SayNumber(${bottles})
exten = *99,n,GotoIf($[${bottles} 0]?loop)
exten = *99,n,Noop(We're out of beer!)
exten = *99,n,Hangup()
Too much dial plan mashing this morning and I rememberd this site:
http://99-bottles-of-beer.net/
And now, in AEL! (This is untested, I just wanted to see how it would
look.)
context silly {
*99 = {
NoOp(99 Bottles of beer on the wall);
Answer();
bottles=99;
while (${bottles} 0) {
NoOp(${bottles} bottles of beer on the wall, ${bottles} bottles of
beer);
SayNumber(${bottles});
NoOp(Take one down, pass it around);
bottles=${bottles} - 1;
NoOp(${bottles} bottles of beer on the wall);
}
NoOp(We're out of beer!);
Hangup();
}
}
Lol, Well done, Russell!
How about this one: from an extensions.conf that someone posted on the
internet, I think, and I converted to AEL; I'm sorry, but I can't find
the original author.
(If anybody can find his post, I'd love to give him credit.) I did test
this out,
and it works; just put a call to the macro ( guessgame(); ) in an
extension in your dialplan
macro guessgame()
{
startpoint:
while (1)
{
Playback(guessit/intro);
set(GUESS=);
GUESS=${EPOCH}%9;
Set(TIMEOUT(digit)=3);
Set(TIMEOUT(response)=5);
while (1)
{
Read(NUMBER,guessit/input_number,1);
Verbose(Got ${NUMBER} from Read);
if( ${NUMBER} = * || ${NUMBER} = # ||
${NUMBER} = )
{
Playback(guessit/thatsnotanumber);
}
else if (${NUMBER} = ${GUESS})
{
Playback(guessit/win);
break; // the only way out of this loop!
}
else if (${NUMBER} ${GUESS})
{
Playback(guessit/less);
}
else if (${NUMBER} ${GUESS})
{
Playback(guessit/more);
}
else /* what other stuff can the user enter than a
number, #,
nothing, or * ? */
{
Playback(guessit/thatsnotanumber);
}
}
/* You get here after a successful guess */
Wait(.5);
Read(AGAIN,guessit/playagain,1);
if (${AGAIN} != 1)
break;
}
Playback(guessit/goodbye);
return;
catch t
{
playback(guessit/goodbye);
return;
}
catch i
{
playblack(invalid);
}
}
murf
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Hey murf,
here is the link for the credit,
http://www.voipphreak.ca/archives/358-Asterisk-Howto-Numbers-Guessing-Game.html
its also in the wiki examples.
http://www.voip-info.org/wiki/view/AEL+Example+Snippets
db
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Re: [asterisk-users] IAX2 INBAND DTMF?
On Wed, 2007-07-25 at 13:02 -0500, Eric ManxPower Wieling wrote: Short Answer: No. Long Answer: Maybe. If you can get your device to send inband DTMF and tell Asterisk you are using INFO or RFC2833 DTMF, then Asterisk should just pass the DTMF as audio. Then if the call goes via IAX2 it should be inband. This is an ungly hack, should not be supported in any way and if it works just count your blessings. I can think of no reason to ever need to do this. Matt wrote: Is it possible to make Asterisk do inband DTMF over IAX? Snip--- Ok, I am confused. Are you saying that if I use an IAX2 inter machine trunk from one asterisk box to another, and terminate a call over the pstn to a voicemail system or other type of IVR, IAX2 will regenerate the DTMF tones that were originated from the original callers phone? I thought the original posting said that the IAXy device was failing to pass DTMF through to the termination side of the call. What have I missed? Dave ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] [asterisk-biz] Testers needed for VoIP router solution
Hi Robert, which of the distros are you using as your base, dd-wrt , open-wrt ? Dave On Tue, 2007-07-24 at 17:19 -0400, Robert Augustyn wrote: Hi all, We have put together a firmware for a range of inexpensive routers. It has been configured to provide optimum VoIP performance. We have internally tested it for number of months and it looks very good. You should be able to run it easily with 20+ phones on local network ( we still did not hit the upper limit ) assuming that you have bandwidth. Your VoIP will get prioritized over other types of traffic. You should be able to talk, download files and run torrents at the same time with no visible degradation of the VoIP voice quality. It will be delivered ready to upload with all your configurations, which you will have to provide to us. We will custom build firmware for your configuration. We just ask you to upload it, test it and provide feedback. If you are interested ( sorry only first 10 will be accepted ) please contact me at firmware at linqone dot com and we will send you the set of questions we need you to answer before we can build a solution for you. Thanks, This firmware will work on: * Linksys WRT54G v1-v4, WRT54GS v1-v4, WRT54GL v1.x, WRTSL54GS (no USB support) * Buffalo WHR-G54S, WHR-HP-G54, WZR-G54, WBR2-G54 * Asus WL500G Premium (no USB support) This will not work on Linksys WRT54G/GS v5-v7 or newer WRT54G/GS routers. If you do not have any of the above routers you can get one for UNDER $40 shipped at: http://www.circuitcity.com/ccd/Search.do?c=1context=keyword=Buffalo +WHR-G54SsearchSection=Allgo.x=11go.y=10 How do I find my Linksys WRT54G/WRT54GS/WRT54GL's version? Look at the bottom side of the router to check for the version number, or compare the first 4 characters of the serial number with the following list: CDF0/CDF1 = WRT54G v1.0 CDF2/CDF3 = WRT54G v1.1 CDF5 = WRT54G v2.0 CDF7 = WRT54G v2.2 CDF8 = WRT54G v3.0 CDF9 = WRT54G v3.1 CDFA = WRT54G v4.0 CGN0/CGN1 = WRT54GS v1.0 CGN2 = WRT54GS v1.1 CGN3 = WRT54GS v2.0 CGN4 = WRT54GS v2.1 CGN5 = WRT54GS v3.0 CGN6 = WRT54GS v4.0 CL7A = WRT54GL v1.0 CL7B = WRT54GL v1.1 If it's not listed above, and it's not a WRT54GL, it's not supported. Sincerely, Robert Augustyn This firmware is provided as-is without any warranty. I will NOT be responsible for damages that occur due to the use of this firmware. USE AT YOUR OWN RISK. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-biz mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-biz ___ --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] Can Asterisk hear on two IP addresses? And can I do
Noah, or anyone actually, question, can the IP address receiving the incoming call be used in extension logic to determine call handling procedures, or maybe a better way to ask is can asterisk provide information as to the IP address on which a request was received? Dave On Mon, 2007-07-23 at 10:10 -0400, Noah Miller wrote: Hi Bilal - The question here is: how asterisk will be able to receive calls at two network cards where each network card has a different IP address. Maybe we need to know if asterisk is doing a hear on the ports only without caring for IP or it is doing a hear only on the IP:port? If you look in the sample configuration files, you'll see that iax.conf, sip.conf, mgcp.conf, and skinny.conf all have a line that looks like this: bindaddr= If you set it to an IP address like 192.168.1.150, Asterisk will listen on that address only. If you set it to 0.0.0.0, asterisk will listen on all available ethernet interfaces. You can configure this individually for each different VoIP protocol (sip, iax, mgcp, skinny, etc). So, say you have an asterisk server that has two network cards, one configured to 192.168.1.150 and another configured to 222.6.7.8, and in sip.conf, you set bindaddr=0.0.0.0. In this case, your asterisk server will be listening on 192.168.1.150:5060 and 222.6.7.8:5060. Another sip device could call your asterisk server at either 192.168.1.150 or 222.6.7.8 (provided you don't have any firewalls blocking sip traffic). Does this make sense? - Noah ___ --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] G729 copy protection
On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote: Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. Hi Bruce, the question is meaningful, when you realize that each of your messages/posts to the list come in twice that's (2) times :) db ___ --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] G729 copy protection
On Fri, 2007-07-20 at 08:55 -0400, Martin Smith wrote: I'd bet the emails are addressed to the list and the original sender, both, so for the original person they appear twice, but everyone on the list gets them a single time. I haven't seen any duplicates. Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bruce McAlister Sent: Friday, July 20, 2007 8:38 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] G729 copy protection David Boyd wrote: On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote: Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. Hi Bruce, the question is meaningful, when you realize that each of your messages/posts to the list come in twice that's (2) times :) In that case, then, no i dont double-click. I'm posting via gmane if that means anything (gmane.comp.telephony.pbx.asterisk.user). Thunderbird only shows my messages once, so I'm not sure why you're seeing it twice. db Nope, the mails from Bruce are being delivered twice. Yours however only came in once, as do everyone else. So something is strange about the way his emails are encoded I suppose. It isn't really that important to me, but it appeared that Bruce thought he was being slammed for something he wasn't and I wanted to try and let him know he wasn't getting doo. db ___ --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] G729 copy protection
Bruce sorry for the top post, but your last two messages have not come in twice Go figure... db On Fri, 2007-07-20 at 13:37 +0100, Bruce McAlister wrote: David Boyd wrote: On Fri, 2007-07-20 at 08:46 +0100, Bruce McAlister wrote: Mojo with Horan Company, LLC wrote: Sorry that this is unrelated but, Bruce, do you double-click to send your messages? Just curious. Sorry that this is unrelated but, Mojo with Horan, do you wake up each morning and think of a meaningful question to ask someone, such as the above, every day?, Just curious. Hi Bruce, the question is meaningful, when you realize that each of your messages/posts to the list come in twice that's (2) times :) In that case, then, no i dont double-click. I'm posting via gmane if that means anything (gmane.comp.telephony.pbx.asterisk.user). Thunderbird only shows my messages once, so I'm not sure why you're seeing it twice. db ___ --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 ___ --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] List delays
Walter, I know you just said that list users can not help with this problem, however I must beg to differ with you. The information that you just provided is a big help, if people take your advice about the configuration of their internal systems. So in one way it is off the topic of Asterisk, but is on target for helping people solve their issues if they interact more efficiently with the list and the resources that it provides. IMHO, Dave On Wed, 2007-07-11 at 08:47 -0400, Walt Reed wrote: No, as I explained before with the reasons why, please don't post them here. Send them DIRECTLY to the list admins. It is 100% off topic to keep discussing a list administration / mail delivery problem here. List USERS can not help you. Considering that the vast majority of users do not experience such delays, and that it's HIGHLY unlikely that Digium maintains a list of who to delay mail for, the problem is 99% likely to be something wrong with the recipient's system. It could be DNS, routing problems, anti-spam mechanisms (greylisting, active sender verification, dspam, SA, etc.) or timeouts caused by slow responses due to said anti-spam mechanisms, etc. Many people fail to realize that high-volume mail servers (especially for large mailing lists) don't have long timeouts and therefore can't tolerate slow recipient servers. It takes too many resources. Make sure that you whitelist list mail at all phases of your protection systems. Make sure you are NOT doing sender callouts, running every message through spamassassin, greylistging, etc. for list mail. Lastly, there is nothing Digium is going to be able to do if your DNS servers are flakey, or route path is. Headers just tell you that there is a delay. We already know this. Only the sending AND receiving server logs can tell you WHY, and then you may only know if the session was run in debugging mode. On Wed, Jul 11, 2007 at 07:33:43AM -0400, Jared Smith said: On 7/11/07, Bill Maidment [EMAIL PROTECTED] wrote: email delays here are about 8 days. I don't expect to see this until 19th July When you do get the message, please reply with the email headers, so we have some chance of tracking down the problem. For example, below ___ --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] North American voice BRI - Informal survey
On Thu, 2007-07-05 at 15:08 -0400, Jon Pounder wrote: Quoting Jeff Davis [EMAIL PROTECTED]: Jon Pounder wrote: I have a bunch of old cisco stuff with BRI ports on it but it was never meant for voice, just purely data, so I don't think its very useful for this purpose, but some of the basic signalling could probably be tested with it. is exploring some sort of back to back card setup worth looking into without committing to a line ? Then at least if a pair of cards can talk in the new format there is a better chance of them working on a real line. This would also have the advantage of being able to see both ends of the line for debugging purposes, or put the line in the euro mode which should work out of the box just to make sure the hardware setup is all valid before making changes. I'm seeing ISDN phones on ebay for US $15-$40. Does anyone know if the line simulator and a phone would work. Then get a BRI line when there's a driver that looks like it works. the signalling on the line simulator and phone would have to be compatible - phone and card are going to have the same issues with supporting the northamerican signalling. probably depend mainly what country the hardware is being sold from what the odds of working are. SNIP I seem to remember that the wan Pipeline units supported BRI, and also provided a couple of analog phone jacks. I will dig around in the basement and try to find the one that I had, if I find it, who wants it for play? Dave ___ --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] North American voice BRI - Informal survey
On Thu, 2007-07-05 at 16:42 -0600, Stephen Bosch wrote: David Boyd wrote: I seem to remember that the wan Pipeline units supported BRI, and also provided a couple of analog phone jacks. I will dig around in the basement and try to find the one that I had, if I find it, who wants it for play? Well, whoever ends up with the simulator should get it. I'm not familiar with the Pipeline stuff. Got a link you can share? -Stephen- No link, it was something I used 8+ years ago, so I am surprised i pulled it out of my memory :) I will dig around this weekend and see if I can find it. Pretty easy to setup, used it for an ISP connection for centrex purposes. Hopefully I am not mis-remembering it capabilities. Dave ___ --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] Play dial tone withou answer
Hi Arjan, As I see it, the issue at hand is as follows: You are attempting to provide a tandem service, meaning as you say no charge to the originator unless the called party answers. However under this circumstance you want to also provide a non-standard call treatment to the line without an answer occurring. Standard treatment is to allow the originating Switch/device to continue to provide the ringing condition to the originators phone while the outbound attempt is being completed. Very few carriers that utilize digital services (non-analog) do not propagate audio back to the originating caller until such time as an answer has been accomplished. SO, this leads me to asking the following, how are the callers originating calls into your system, what are they using for authentication as well as indication of desired outbound calling data? Dave On Fri, 2007-06-22 at 08:22 +0200, Arjan Kroon wrote: Yes Dave, We want to use to principle for the following reason. If the outbound call is not picked up, the inbound caller won't be charged for the call, because there was no answer. Arjan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Boyd Sent: dinsdag 19 juni 2007 17:03 To: Lee Jenkins Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Play dial tone withou answer Yes Lee, he could, however he doesn't want to answer the call until the call has been completed on the outbound leg. Dave On Tue, 2007-06-19 at 10:26 -0400, Lee Jenkins wrote: David Boyd wrote: Two points, first (I believe from many previous threads, and viewing source code ) you must answer a call to place audio on the channel. second, depending on the type of access being used by the originator of the call, the carrier will not pass audio on the channel back to the originator unless they receive an answer indication from asterisk, so even if you could place audio on the channel without an answer, there is no guarantee still it would propagate back to the originator of the call. Can't he just setup an extension to Answer() the call, play message or Ringing() and then transfer the call? ___ --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] asterisk 0 dial outgoing call
On Fri, 2007-06-22 at 05:59 -0700, satish patel wrote: can u give me example how do i create plan for this task or job ram [EMAIL PROTECTED] wrote: On 6/22/07, satish patel [EMAIL PROTECTED] wrote: Dear all i have one confusion about how to dial outgoing call through asterisk like when i press 0 i got dial ton of exchange for outgoing call my setup is [sip_phone]-[*][mediant2k]-[Avaya_PBX]--e1-[Exchange_PSTN] now i want to setup whn i press 0 in my sip phone i got dialton of PSTN so i can call outside people is there any special configuration to give dialtone from pstn how to setup extention.conf for outside call create dialplan for the same ram What digit do you dial on the avaya to get PSTN dialtone? Setup a dial plan entry for dial digit 0 to access the avaya and dial the access code for the PSTN . dave ___ --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] Play dial tone withou answer
Two points, first (I believe from many previous threads, and viewing source code ) you must answer a call to place audio on the channel. second, depending on the type of access being used by the originator of the call, the carrier will not pass audio on the channel back to the originator unless they receive an answer indication from asterisk, so even if you could place audio on the channel without an answer, there is no guarantee still it would propagate back to the originator of the call. dave On Tue, 2007-06-19 at 11:58 +0200, Arjan Kroon wrote: Hi, I’m looking fore a way to play a dial tone before our IVR platform answered the phone line. I want to use for the following reason: When a caller calls our Voice Platform, the call will direct dial out to a number. I want to dial out before the inbound call is answered. But now the inbound call here’s nothing. When the outdial call is picked the inbound call will here something. This is confusing voor in inbound call, therefore I want to let the inbound call to here a dial tone. I already tried the function Ringing() and Playtones(). (before the inbound call is answered) Can anybody give me a hint or solution to this problem? Kind Regards, ___ --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] Play dial tone withou answer
Yes Lee, he could, however he doesn't want to answer the call until the call has been completed on the outbound leg. Dave On Tue, 2007-06-19 at 10:26 -0400, Lee Jenkins wrote: David Boyd wrote: Two points, first (I believe from many previous threads, and viewing source code ) you must answer a call to place audio on the channel. second, depending on the type of access being used by the originator of the call, the carrier will not pass audio on the channel back to the originator unless they receive an answer indication from asterisk, so even if you could place audio on the channel without an answer, there is no guarantee still it would propagate back to the originator of the call. Can't he just setup an extension to Answer() the call, play message or Ringing() and then transfer the call? ___ --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] Re: [asterisk-dev] CDR changes in Trunk -- Transfers, CDRs, Life, and Everything
Murf, you crack me up, but I totally agree with the vote or don't complain model. Thanks, Dave On Tue, 2007-06-12 at 13:05 -0600, Steve Murphy wrote: I have created an asterisk.org blog entry: http://www.asterisk.org/node/48358 to describe what I will shortly be committing to trunk to correct the weaknesses of CDRs, that asterisk users and developers have been complaining about for quite some time. Highlights: Restructuring the code and philosophy of CDRs. Plans to eliminate the ForkCDR() application Plans to create the CDRstart(), CDRanswer(handle), and CDRclose(handle) functions to provide dialplan ability to create CDR records. (I am considering restructuring the CDR function, also, to allow mods to be made to not only Channel-attached CDR's, but also the fields in CDRs created by CDRstart(), BTW). I seek feedback from folks who have battled with CDRs to develop billing applications, and those who plan doing so in the future. Participate or be happy with the senseless mess that will surely result from your non-participation! murf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev ___ --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] CDR on transfers of called ZAP channel
On Mon, 2007-06-11 at 09:11 -0600, Steve Murphy wrote:
Gunnar--
CDR generation that covers transfers is an umimplemented feature in
Asterisk, in any version.
I have been working on a solution, but unfortunately, my solution is
radical enough that I dare not apply it to 1.2 or even 1.4. It will most
likely break every working implementation of billing that has been built
on Asterisk by end users/developers. Unpleasant visions of angry mobs of
developers armed with baseball bats, who want nothing more than to drag
me out of my home and share their pain and frustration over my
fixes. you get the idea.
Actually, I have TWO solutions! One, is to modify the current CDR
engine, the other is to provide an entirely different solution that is
single-event driven, kinda along the lines of manager events, but more
streamlined for CDR billing purposes.
The first solution somewhat reorganizes CDRS by no longer posting them
to the backend db's when a hangup occurs. Rather, it will post them when
a bridge between channels is finished, or ends. Since a Local
channel acts as a sort of bridge, I think I will have to do the same
thing there. I'm in the middle of it now. I spent/wasted a good amount
of time generating extra CDR's that would describe time in different
parts of a transfer, but as I traveled further down that road, I see
that this will only make things unnecessarily complex. So, I'm not going
to do it. What this means is that a CDR will get generated for each
chunk of a conversation involved in a transfer, but these pieces will
not tell you much about how the chunks relate to each other. The channel
originating the conversation will be the source, and the channel
originally connected to will be the destination. Time spent in 3-way
conferences, music on hold, etc. etc. will most likely not be available.
My theory is that, in most cases, it won't matter. All you REALLY want
to know is who to bill, and for how much time. If a transfer occurs, it
involves someone internally dialing another party. This second
conversation, will generate another CDR, and the guy who dialed it
will be assigned that call, even if he hung up before the call was
answered (blind xfer). For example, picture this: a switch in Modesto
gets a call from Sacramento, and extension 151 gets this call, and dials
Shanghai, and blind transfers the Sacramento call to Shanghai, and then
Sacramento and Shanghai talk for an hour. Two CDR's will be generated.
One will cover the incoming call from Sacramento, and will be little
over an hour. The other CDR that will come out will say 151 dialed
Shanghai and talked an hour. That's it.
The second solution, the event-based one, will generate an event record
for each significant event in the life of each channel. So, START
events when a channel is born; ANSWER events when someone answers a
call; END events when somebody hangs up. There will also be Park,
and Transfer, and MOH, and 3-WAY, Conference-Join, and several
others. Just enough information will be included with each event to
thread together billable sequences. Along with each event record will be
the time the event happened, and channel info. This approach will be
very much more fine-grained, and allow you to do fancy things like
figure out that Sacramento was the only person talking to Shanghai, and
allow you to bill the call to the guy/gal in Sacramento. Trouble with
this approach is that threading together the event records is a
non-trivial operation! But I hope to provide some tools that will make
this easier to do.
So, the bad news is: you will not see any solutions for this problem, in
1.2, or 1.4. the CDR fix (first solution) will most likely end up in
1.6, the event-based solution will probably not be available until 1.8
or 1.10; we shall see.
murf
On Mon, 2007-06-11 at 14:26 +0200, Gunnar Schaller wrote:
Hello list,
I have a problem with called ZAP channels making an attended-transfer
or blind-xfer. Signalling at the phones is ok, but the CDR of Asterisk
is wrong.
At the moment there is a bristuffed Asterisk 1.2.18 running with
bristuff-0.3.0-PRE-1y-g. Here is my dialplan, I simplified it a bit:
[default]
exten = 0123456789,1,Macro(dialpstn,${EXTEN})
[macro-dialpstn]
exten = s,1,Set(TRANSFER_CONTEXT=transfer)
exten = s,2,Set(FORWARD_CONTEXT=transfer)
exten = s,3,Set(CIDNUMORIG=${CALLERID(num)}) ; save callerid-num
exten = s,4,Set(CALLERID(num)=98765) ; dial out using msn 98765
exten = s,5,Dial(Zap/g1/${ARG1}|30|t)
exten = h,1,Set(CALLERID(num)=${CIDNUMORIG}) restore callerid-num for
CDR
[transfer]
exten = _X.,1,Set(CALLERID(all)=External 0123456789)
exten = _X.,1,Dial(SIP/${EXTEN})
So I call 0123456789 with SIP phone 10. The callee dials *1 20 for
attended transfer and SIP phone 20 (I have *1 for attended transfer in
features.conf). The called SIP-phone shows the caller-information I
set in context
Re: [asterisk-users] FX Dialing Odd
What happens if you connect the fxo to the fxs and try several attempts at completing a call? This should at least tell you if the issue is outdialed digit issues or telco receipt issues. Dave On Mon, 2007-06-04 at 10:30 -0500, Rob Schall wrote: But if this was the case, then why would the message playback (from the provider) read back the digits from the start. I mean, I dialed 630-XXX-XXX and it reads back 312 (my local area code)-630-XXX-X I would think if it wasn't waiting, then it would do like a 312-30X-XX-XX Rob John Novack wrote: Rob Schall wrote: Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However,every 4 or 5 times, we get an error back from the provider that says The number you have dialed. blah blah blah. Common defect in the Zaptel driver. It does NOT listen for dial tone, so if you have not inserted a w or three into the dial string, it will dial before the Central Office is ready, and it may miss a digit, causing misdials. Curious that cheap modems years ago learned to listen for dial tone, but the Zaptel driver doesn't, and of course this is considered a feature request rather than a bug, and no one seems to want to fix it. John Novack ___ --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
[asterisk-users] Delay in posting of messages to list
Can anyone enlighten me as to why it takes 40 minutes or more for a posting to the list to appear. This seems excessive, as other forums do not take this long. Dave ___ --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] *End Of Life ASTERISK 1.2.X Was: INSTRUCTIONSFORTHE ASTERISK COMMUNITY - PLEASEREAD NOW *
On Wed, 2007-05-30 at 21:54 +0200, Olivier wrote: 2007/5/30, Matt [EMAIL PROTECTED]: The problem with this is that if 1.2 has a bug that is making it unstable, it should be fixed to make a stable project, rather then steam rolling ahead to the next release. Further, I have seen on several occassions a security patch cause stability issues in Asterisk. I'm not aware of any easy way to turn an unstable server into a stable one nor aware of any bug-free application software. And if such software did exist, what happens with security patches from Operating System, or hardphone upgrades or devices you don't manage ? The real questions are : - Which open bugs are keeping you from proving given telephony services ? - Do you then have a way to lower your service level or to investigate ? - How many open serious bugs are still affecting 1.4 ? Regards ___ --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 Not trying to start a flame war, however the issues that I see with 1.2 and 1.4 are very similar to the issues relating to Redhat and Fedora. Redhat didn't want to continue supporting the open source model and convinced? the end user community to support all of the old releases based on the number of deployed systems. If the user community really doesn't want the versions to go away, then they won't allow it to happen. My question is this: Will digium provide the needed support to the community to allow them to continue supporting the 1.2 release, or will this prove to be related to business issues that the user community is not aware of, which will result in a much broader support of callweaver? my $.02 which probably isn't worth $.02! dave ___ --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] multiple host= in sip.conf
On Wed, 2007-05-30 at 15:29 -0500, Eric ManxPower Wieling wrote: Bryan Laird wrote: for inbound connections how does asterisk manage host=host-name returning multiple A records... will it allow authentication for any of the IP's returned? I would assume that in the case of 'inbound' if you specify a host-name that you have PTR records for you could do it in one entry again I'm making a blind assumption. As I understand it, Asterisk does a DNS lookup on load/reload and uses whatever the first IP address returned. allow= and deny= is what should be used for access control. Not the host= line. The host= line is normally used for Asterisk - Device stuff. ___ --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 Does that mean that even when dynamic dns entries exist and the time to live is set to 15 minutes asterisk will continue to try using the old expired results? Dave ___ --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] Re: asterisk-users Digest, Vol 33, Issue 33
Could someone please remove this person from the list. It seems that the person is saying they will be away for (9) nine months, with their auto-reply set. dave On Mon, 2007-04-09 at 18:26 +0200, [EMAIL PROTECTED] wrote: Je suis absent du 2/04/2007 au 11/04/2007. Je répondrai à votre message dès mon retour. Pour toute urgence, contacter Emmanuelle Parache Moga ou Cédric Buzay. ___ --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] Configuration assistance needed.
On Fri, 2007-04-06 at 09:30 -0600, David Thomas wrote: A start would be to get the contact information and actually CONTACT the person about it. Come on now. Maybe I'm confused... Isn't that what Dovid did when he replied to Tim's post? Regards, David ___ --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 My Two (2) cents, Exactly why would you want to force someone to perform another look up to offer you assistance? When I need assistance, I actually provide enough information for a person to make a decision with the info at hand not make them waste time looking up something you should have provided anyway. Dave ___ --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] Re: Problem converting a Cisco 7960 to SIP
On Thu, 2007-03-29 at 12:16 -0400, Brad Stockdale wrote:
Hello all,
I've got myself into a bizzare situation that I can't seem to get myself
out of... Was wondering if anyone had some advice that might get me 'over the
hill' on this...
Some background: PBX consists of an Asterisk box (running TrixBox), 4
Cisco
7960's, 2 Polycom IP500's, and now an additional Cisco 7960. The phones are
all on a separate LAN. There is no VLAN configuration. The Asterisk box also
is running a TFTP server and DHCP server. The 4 original Cisco's work fine
still. The Polycom IP500's work fine.
The problem is with trying to get this new Cisco 7960 online... It came
pre-loaded with the SCCP image and I cannot get it to convert to SIP.
Currently it is running the following versions:
App Load ID: P0030301MFG2
Boot Load ID: PC0303010200
Version: 3.1(MF.G2)
The phone contacts the DHCP server and gets an IP successfully. The
dhcpd.conf file:
##
# dhcpd.conf - dhcp config file for eth1 / sip phones
##
authoritative;
ddns-update-style interim;
ignore client-updates;
local-address 192.168.1.1;
option tftp-boot-server code 150 = ip-address;
option tftp-boot-server 192.168.1.1;
subnet 192.168.1.0 netmask 255.255.255.0 {
option routers 192.168.1.1;
option subnet-mask 255.255.255.0;
option domain-name-servers 192.168.1.1;
option time-offset -18000; # Eastern Standard Time
option ntp-servers 192.168.1.1;
option tftp-server-name 192.168.1.1;
default-lease-time 43200;
max-lease-time 86400;
pool {
range 192.168.1.100 192.168.1.150;
}
}
Then the phone contacts the TFTP server. Below are the logs:
Mar 29 12:09:15 asterisk1.local atftpd[32276.-1208575056]: Serving OS79XX.TXT
to 192.168.1.144:49427
Mar 29 12:09:16 asterisk1.local atftpd[32276.-1208575056]: Serving
SEP001795B05B1D.cnf.xml to 192.168.1.144:49428
Mar 29 12:09:16 asterisk1.local atftpd[32276.-1208575056]: Serving
XMLDefault.cnf.xml to 192.168.1.144:49429
Mar 29 12:09:16 asterisk1.local atftpd[32276.-1208575056]: Serving
SEP001795B05B1D.cnf to 192.168.1.144:49430
OS79XX.TXT contains:
P003-08-6-00
Originally the SEP001795B05B1D.cnf file didn't exist. Since it was for
CallManager, I didn't bother to configure it and just setup the SIPmac.cnf
file instead. The phone never requested the SIPmac.cnf file...
I found a trick via google that uses the SEPmac.cnf file to change
firmware. The SEP file now contains:
Default
callManagerGroup
members
member priority=0
callManager
ports
ethernetPhonePort2000/ethernetPhonePort
/ports
processNodeName192.168.1.1/processNodeName
/callManager
/member
/members
/callManagerGroup
loadInformation6 model=IP Phone 7910/loadInformation6
loadInformation124 model=Addon 7914/loadInformation124
loadInformation9 model=IP Phone 7935/loadInformation9
loadInformation8 model=IP Phone 7940/loadInformation8
loadInformation7 model=IP Phone 7960P003-08-6-00/loadInformation7
loadInformation2 model=IP Phone 7905/loadInformation2
loadInformation30008 model=IP Phone 7902/loadInformation30008
loadInformation30007 model=IP Phone 7912/loadInformation30007
/Default
The TFTP directory contains:
0004f20049bc-app.log
0004f20049bc-boot.log
SEP001795B05B1D.cnf
polycom_brad.cfg
sip.cfg
WORKING_POLYCOM_sip.cfg
WORKING_POLYCOM.cfg
phone1.cfg
0004f20049bc.cfg
0004f20049bc-phone.cfg
0004f20049bc-appFlash.log
SoundPointIPLocalization
.cfg
-directory~.xml
SoundPointIPWelcome.wav
sip.ld
sip.ver
bootrom.ld
SIP001795B05B1D.cnf
snom.cnf
SIP0012DABF2AAA.cnf
SIP0012D9B94C72.cnf
SIP001280B9D6E1.cnf
SIP001280F3AFC7.cnf
SIPDefault.cnf
DSM2ColorLogo_3.bmp
OS79XX.TXT
P003-08-6-00.bin
P003-08-6-00.sbn
P0S3-08-6-00.loads
P0S3-08-6-00.sb2
797x_template.cnf.xml
cisco_util
Desktops
dialplan.xml
merlin2.pcm
RINGLIST.DAT
syncinfo.xml
All other phones work fine. Therefore, I assume all the firmware is in the
right place... They all converted to SIP firmware fine...
When I try to do the **# unlocking, it does nothing... Everything still
shows locked. The phone doesn't have an Unlock Settings function (assuming
firmware is too old)
The phone, when it boots, goes through an endless loop consisting of:
Configuring VLAN
Configuring IP
Then it starts over.
What in the heck am I doing wrong? I thought that the OS79XX.TXT file
should have taken care of pushing out the new image. And the phone is
grabbing the file via TFTP, but it's like it ignores the idea of changing
Re: [asterisk-users] Re: queue information into db
On Wed, 2007-02-28 at 13:05 +0100, nik600 wrote: On 2/28/07, Tomislav Parcina [EMAIL PROTECTED] wrote: nik600 wrote: In the last months i've developed a web application for the use of an asterisk call center. Yuo can - make calls from a web interface - login/logout in queue - view members logged in a queue - view callers queued in a queue - pickup a callers from a queue What is license of this application? Can it be downloaded from somewhere? -- Tomislav Parcina [EMAIL PROTECTED] ___ --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 actually it isnìt released under any type of licence. if you want i can put the code on my web site (but no earlier than the next week) ___ --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 That would be great, can you provide a URL when it is available. This would greatly assist us in our trouble handing scenario. db ___ --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] Automatic Dial, Play message
On Thu, 2007-02-08 at 16:48 -0800, Yuan LIU wrote: From: Stefan Wintermeyer [EMAIL PROTECTED] Date: Thu, 8 Feb 2007 21:56:11 +0100 Am 08.02.2007 um 18:39 schrieb Forrest Beck: Does anyone have some method, or AGI scripts that will automatically call a list of numbers from a database and play a pre-recorded message? Just for example, you have a database of FirstName, LastName, PhoneNumber Jon, Beck, 9194713175 So it would pull each record with phone number, dial the number, when answered play a pre-recorded message. Have a look at an e-mail which I send yesterday to this list. It contains a simple example for a call file. That is the way you want to go. With that you can create a script which solves your problem. Stefan I looked this and http://voip-info.org/wiki/view/Asterisk+auto-dial+out+deliver+message, both using call files. Can the same commands be used from inside extensions.conf to do same? Yuan Liu ___ --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 The issue is not can you execute the same commands from within extensions.conf, but how are you going to trigger the action without external input. We process calls using the following methodology: 1.Cron starts a job at preset times 2.script log into postgresql and determines if any call are to be made at this time 3.Script then determines how many calls can be made based on codecs, time of day, and service provider to be used 4.Script generates call file/s into temporary directory based on above criteria and moves them to /var/spool/asterisk/outgoing 5.Asterisk places calls, and using cdr_pgsql writes cdr to database 6.upon insert a trigger fires to update list of called numbers and indicate success or failure 7.goto 1 Simple process, extensions.conf is used for all call flow, and no external processes used for updates to database. We used AGi in past and found that this process was actually easier to maintain as the only code written was a simple php script for db access and call file generation. Don't know if this helps with ideas but if you are interested in additional details contact me off list. ___ --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] Billing pulses
Hi Stefano, I have a question, how would you go about using the billing pulses to generate an invoice/bill. Also can you provide an ascii drawing of the layout of the equipment as you intend to use it, they say a picture is worth a thousand words:) db On Thu, 2007-02-08 at 15:13 +0100, Stefano Corsi wrote: I must clarify my original message. Maybe confusion is due to my poor english. So I'll make a list of statements: - Each ISDN line in Italy can be splitted in two analog lines - You can use those analog lines as normal analog lines - I have already invested in analog hardware (my fault of course) for both FSX and FXO - ISDN hardware installed by the telco can, in Italy, be programmed to send a billing pulse. - I guess this billing pulse is sent on each of the two analog lines in which a single ISDN line can be splitted (so there's no risk, I guess, for double billing). - I'm considering if there's a small chance for me to avoid buying additional hardware (ISDN cards or gateways) and have an accurate billing using those analog lines resulting from splitting an ISDN line. - To get an accurate billing, I'm wandering if it's possibile to use billing pulse provided by those analog lines. - I have full specifications of the billing pulse provided: frequency 12 kHz ± 1% level .. 200 mVrms on 200 distortion... 5% pulse duration .125 ± 25 ms pause duration 180 ms period ... 300 ms Do you think it's worth considering it? Rgds Stefano Bill them both. We are talking about mere BRI's, right:-) Good catch, David. As others noted, billing pulse really applies to analogue lines only, and ISDN providers should always send status. Yuan Liu Thanks, Yuan But my confusion came from the original post stating the use of ISDN circuits for this implementation. Id ISDN is in fact the circuit of choice for this app, I agree why wouldn't he simply use the cause codes for billing purposes. We have a lot of experience in telecommunications billing, and have always found cause codes to be more than sufficient even for weird tiers, and bizarre rounding functions. ___ --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] Billing pulses
On Wed, 2007-02-07 at 10:14 -0800, Yuan LIU wrote: From: Jorge Mendoza [EMAIL PROTECTED] Funny that a digital line have a analogue pulse. Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system. Jorge Mendoza Stefano Corsi wrote: Hello, I've discovered that in Italy ISDN lines can be programmed to generate a billing pulse every n seconds (it dipends from the pricebook). The pulse has these figures: Whatever reason, if telco provides them, there's a good chance that some ISDN interface cards can use them. (Just googled to confirm that some non-Digium cards can be used in Asterisk.) This doesn't mean that Asterisk can use them. So you may need significant programming to get going. If they are truly analogue pulses, it could be cheaper to produce a little dedicated circuit to feed an AGI or something. Yuan Liu frequency 12 kHz ?1% level .. 200 mVrms on 200 distortion... 5% pulse duration .125 ? 25 ms pause duration 180 ms period ... 300 ms Does someone know if these values can be used somehow to get an accurate billing using asterisk with these lines? Could be a matter of configuration or programming? Thanks Stefano ___ --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 How would you be able to determine which call was being billed for if the pulse is sent down the wire on an ISDN circuit with multiple channels in use? db ___ --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] Billing pulses
On Wed, 2007-02-07 at 14:49 -0800, Yuan LIU wrote: From: David Boyd [EMAIL PROTECTED] Date: Wed, 07 Feb 2007 15:24:04 -0500 On Wed, 2007-02-07 at 10:14 -0800, Yuan LIU wrote: From: Jorge Mendoza [EMAIL PROTECTED] Funny that a digital line have a analogue pulse. Normally the billing pulse is used on payphones. IMO you only need the answer supervision to trigger your own billing system. Jorge Mendoza Stefano Corsi wrote: Hello, I've discovered that in Italy ISDN lines can be programmed to generate a billing pulse every n seconds (it dipends from the pricebook). The pulse has these figures: Whatever reason, if telco provides them, there's a good chance that some ISDN interface cards can use them. (Just googled to confirm that some non-Digium cards can be used in Asterisk.) This doesn't mean that Asterisk can use them. So you may need significant programming to get going. If they are truly analogue pulses, it could be cheaper to produce a little dedicated circuit to feed an AGI or something. Yuan Liu ... How would you be able to determine which call was being billed for if the pulse is sent down the wire on an ISDN circuit with multiple channels in use? db Bill them both. We are talking about mere BRI's, right:-) Good catch, David. As others noted, billing pulse really applies to analogue lines only, and ISDN providers should always send status. Yuan Liu ___ --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 Thanks, Yuan But my confusion came from the original post stating the use of ISDN circuits for this implementation. Id ISDN is in fact the circuit of choice for this app, I agree why wouldn't he simply use the cause codes for billing purposes. We have a lot of experience in telecommunications billing, and have always found cause codes to be more than sufficient even for weird tiers, and bizarre rounding functions. db ___ --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] Using Local Channels with Originate
On Mon, 2007-02-05 at 13:21 -0800, Michael Collins wrote:
I haven’t quite gotten this working yet but I am going to update the
thread with what I have learned. Maybe this will help the next guy who
tries to figure this out…
The trick to using the DIALSTATUS seems to be to put it in the handler
for the h (hang-up extension).
[outdialer]
exten = 100, 1, Dial(${numberToDial})
exten = h, 1, Goto(s-${DIALSTATUS},1)
exten = s-ANSWER,1,NoOp(Answered)
exten = s-BUSY,1,NoOp(Busy)
exten = s-NOANSWER,1,NoOp(Not answered)
exten = s-CANCEL,1,NoOp(Cancelled)
exten = s-CONGESTION,1,NoOp(Fast busy)
exten = s-CHANUNAVAIL,1,NoOp(Channel unavailable)
[dialerplan]
exten = s,1,Background(demo-congrats)
exten = s,n,WaitExten
so on ...
Here are the manager commands I am using:
Action: login
Username: test
Secret: nottelling
Action: originate
Channel: Local/[EMAIL PROTECTED]/n
Context: dialerplan
Extension: s
Priority: 1
Variable: numberToDial=ZAP/4/1234567890
Action: logoff
I am always getting ANSWERED for ${DIALSTAUS} so something is not
quite right. Hopefully I am getting closer.
Brian,
What kind of Zap hardware/telco lines are you using? I am using PRI
and I am able to get a dial status in the hangup extension. The
problem I run into is that I get “NO ANSWER” as the hangup cause even
for invalid phone numbers… I also get cluttered CDR’s. In the
meantime I’m working on a solution that I hope will give the best of
both worlds. I’m relying on the API events instead of local channels.
I’ll post more information when I’ve made more progress. However,
I’ve made 2500 test calls and I haven’t lost a single
‘OriginateSuccess’ or ‘OriginateFailure’ event. (I’m keying on these,
specifically the ‘OriginateFailure’ event because it has a ‘Reason’
value that gets populated: 0=Invalid, 3=No Ans, 5=Busy.)
Hope to have more info posted this week.
-MC
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Very cool, thanks for the info.
db
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Re: [Asterisk-Users] DMS-500 CID name not in CDR
What code set is the 500 PRI configured for? Dave On Fri, 2005-06-10 at 17:05 -0400, [EMAIL PROTECTED] wrote: Hi Guys, I have several * servers connected to T1 PRI's from various service providers in multiple locations the US. All the * servers use the same hardware with the same OS and * version. When connected to 5ESS Switches, using the NI2 (national) PRI protocol, the CID name and number come across fine and populate into the * CDR fine. I connected to a DMS-500, NI2 (national) protocol and the CID name doesn't get populated in the * CDR. The only variance in the PRI debug outputs is this, from the DMS switch: Jun 9 16:41:15 WARNING[30369]: chan_zap.c:7133 zt_pri_error: PRI: !! Facility message shorter than 14 bytes The interesting part is that the CID name does come into the * server and is forwarded to the destination phone, the CID name and number does come across from the service provider. Im thinking there is something particular with the DMS-500 and * causing this. Is there a special terminating setting I can throw into Zapata.conf to help this situation. I can send full debug outputs from both working and non-working servers if needed. Any help would be greatly appreciated. Thanks in advance. JR ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIPURA SPA-2000 webserver dead after firmware upgrade
Run nmap against the ip address and see what ports are active for tcp service. Maybe you can connect via a different port (I know it should be 80), and see if the configuration is different between voice and web dave On Tue, 2005-05-10 at 21:50 -0400, Steve Prior wrote: I just got a refurb Sipura SPA-2000 and was able to assign it an IP address with DHCP and ping the device, but then I ran the firmware upgrade utility to bring it up to spa2k-2.0.13g which seemed to work just fine, but after it rebooted I cannot connect to its webserver for configuration. I can still ping the unit. When I use the built in voice menu it reads back the right IP address, webserver port, and claims the webserver is enabled, but I can't connect to port 80 on the device and running the firmware upgrade utility says that it cannot connect to the unit either. Has anyone seen something like that and is there a fix? A google search didn't turn up any apparent hits. Thanks Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys/Cisco buys Sipura
On Wed, 2005-04-27 at 23:22 -0400, Steven Kalcevich wrote: I think its a win win situation. Cisco has tons of money to throw at them to get a better product with more features. I dont believe they would aquire them and not put money in them to make a better product. I guess the prices will go up like a rocket Not necessarily, When Cisco acquired linksys the prices of the linksys equipment went down. Guess you never know until it happens. ___ 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 In 1998 Cisco purchased a company called Summa Four for $116 Million, and left them to die on the vine. It all depends on what they (Cisco ) want from the transaction. If Sipura has a part in causing a drop in Cisco revenue due to adoption by the Open source community, then they may well buy the company to shut it down. Time will tell! ___ 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] CDR
Why not simply delete the cdr via AGI script (ie delete cdr from table name where number dialed ) for those calls that don't adhere to the dialed number that you want to capture, or am I missing something ? This would allow you to remove the cdr at the completion of the call, and preserve storage space on the system. The downside would be the use of system resources for the deletion after every call. In addition why do you want to only capture certain calls? Is it storage issues, not understanding SQL.. Dave On Thu, 2005-03-03 at 08:59, R A wrote: I need that my records cdr only get the calls that begin with 9 or any other rule is this possible?? thanks in advance wert __ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web __ ___ 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 -- They that would give up essential liberty for temporary safety deserve neither liberty nor safety. Benjamin Franklin ___ 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] Voice mail
How would you deliver calls to the voicemail system without the PBX functions? db On Thu, 2005-01-27 at 08:22, [EMAIL PROTECTED] wrote: HI I would like know if it's possible to use the VoiceMail only of the Asterisk Sytem without use the PBX part ? Thank. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP + NAT = horrible mess
On Thu, 2005-01-27 at 16:06, Kim Lux wrote: I've got Grandstreams (SIP devices) working behind double NATs, none the less. I recommend turning STUN off and make sure that your SIP devices are generating random port numbers. If they generate static port numbers, you'll get port collisions. The other parameter to watch is the keep alive interval. I'm not an expert, but I think this has to be long enough so that the device doesn't disconnect from the router while the various signalling is getting set up. (I've got it set to 20 seconds.) Maybe I'm missing something, but I thought it works quite well without STUN. They've never ever dropped a call. On Fri, 2005-01-28 at 00:18 +0400, Jean-Michel Hiver wrote: Hi Guys, After days of fiddling, I can't really get my SIP device to work communicate with Asterisk behind NAT. Sometimes the STUN server is flaky, sometimes the device isn't reachable if the connection is dropped and then put back on, sometimes it registers OK, sometimes it doesn't, etc. I've come to the same conclusion as the wiki: it's probably better to avoid this horrible mess by either using IAX or doing VPN. Letting the IAX option aside, are you aware of any SIP devices that support some simple, easy to use VPN protocol? Cheers, Jean-Michel. ___ 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 Will you Please share your configuration, I was ready to give up, thinking no one had been successful. TIA db ___ 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] SIP + NAT = horrible mess
On Thu, 2005-01-27 at 17:25, Kim Lux wrote: On Thu, 2005-01-27 at 16:57 -0500, David Boyd wrote: Will you Please share your configuration, I was ready to give up, thinking no one had been successful. I am not using Asterisk, so I can only give you the Grandstream part of things. Maybe some of the Grandstream parameters will twig an idea for an asterisk setting. If I were in your shoes, I'd get my SIP devices working with a simple NATing router and SIP provider before putting asterisk in the mix. Our setup is: Broadband-LinkSys WRT56G-Grandstreams and computers. When operating in this configuration, the Grandstreams are configured with DHCP and as follows: Admin Password: (purposely not displayed for security protection) SIP Server: sip.babytel.ca Outbound Proxy: nat.babytel.ca:5065 SIP User ID: (the user part of an SIP address) Authenticate ID: same as above Authenticate Password: (purposely not displayed for security protection) Name: (optional, e.g., John Doe) Advanced Options: G723 rate: 6.3kbps encoding rate iLBC frame size: 30ms iLBC payload type: 99 Silence Suppression: No Voice Frames per TX: 2 Layer 3 QoS: 48 Layer 2 QoS: 802.1Q/VLAN Tag 0 802.1p priority value 0 Use DNS SRV: Yes User ID is phone number: No SIP Registration: Yes Unregister On Reboot: Yes Register Expiration: 60 minutes Early Dial: No Dial Plan Prefix: nothing No Key Entry Timeout: 4 Use # as Dial Key: No local SIP port: 5060 local RTP port: 5004 Use random port: Yes NAT Traversal: No (Don't set up a STUN server.) keep-alive interval: 20 seconds Use NAT IP nothing Proxy-Require: nothing snip Send DTMF: in-audio When double NATing, the setup is: broadband-WRT54G wireless router- air -laptop wifi-laptop Ethernet port-Grandstream In this config the Grandstream is set up the same, except that I set the laptop Ethernet port to a static IP and the Grandstream to one too. When I am using a static IP, I give it 2 DNS servers so that it can resolve the sip urls. I hope this helps. I've also taken the laptop to other offices and hooked into their wireless networks with no problems. Awesome, I will play with things and let the list know if successful in integrating * in the mix. Thanks for the info. db ___ 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] T1 EM vs PRI question
Responses embedded below! On Mon, 2005-01-24 at 18:49, Keith Burns wrote: Depending on the switch they are using, there are a limited number of D-channels (or D-channel licenses). CAS signaling needs RBS (its the winking in this case). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Beebe Sent: Monday, January 24, 2005 2:47 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] T1 EM vs PRI question Ok, I'm about to take the plunge, and am trying to decide between Channelized T1 EM and PRI. I'm getting an Integrated T1 which will have data and voice capability, all plugged directly into my digium single T1 card. In either case the data piece looks pretty straighforward, just setup the channel properly, hand it off to the linux hdlc layer, and route away the voice side seems a little more complex -- I'm looking for clarification and/or advice: PLease no Flame, just a correction if required. There seemed to be issue using Data/Voice on the digium cards, but I believe it is a setup issue not a technical limitation on the card itself. It seems to me that the major differences between the two different voice delivery mechanisms (other than cost) is caller id functionality and call setup delay. With the PRI, I'll have practically instant call setup and the ability to pass CNAM (caller name) and CID (caller ID) information in BOTH directions. The PRI will give me the ability to have additional directory numbers (typically called DIDs) assigned against my voice trunks and will provide the full ANI (automatic number identification) and DNIS (dialed number identificaton service) over the PRI signalling trunk. Each voice channel will also be 64k clear channel, so I could (theoretically) provide 56k dial-in modem service from the same box (anyone actually doing this?? seems like a neat application for the dsp software guys) I also lose one 64k channel to signalling. Actually DNIS can be provisioned over em trunking also, the separation of digits is done with *'s or KP/ST. So the digiti dump would be something like: DTMF OH - - Wink digit dump *703727131229*8004231212*- -wink -Answer The breakdown of the digits is ani + Info digits then DNIS The *'s would be replaced with KP/ST pulses if MF. KP start sequence, ST stop sequence. Sorry for the crude drawing, and the disclaimer is its been 4 years since I have looked at the digit sequence for an EM t1 :) Sounds like the way to go, but basically the PRI ends up being $100/month more expensive than the Channelized T1 EM. The T1 EM approach will still give me CID (but not CNAM???) over the in-band call setup mechanism (ie: quick DTMF tones during the wink). Each voice channel will actually be 56k because it uses RBS (robbed bit signalling -- not sure what its using this for, as the call setup is delivered via wink???). As a result, this approach would also keep me from implementing a 56k dial-in modem service, but I could still use an ordinary modem or fax dsp to provide 33.6k dial-in. This setup can support DID, but its appended (or prepended, depending on the provider) to the DTMF call setup (which extends the time for calls to actually connect). Not sure if CID or CNAM can be provided for outgoing calls (I think some providers can enable me to be able to wink to them the number to pass as caller id??) I don't know of a way for outbound or inbound CNAM to be provided on a T1 unless you are using SS7 or some like control protocol. The setup time is in milliseconds for PRI and potentially could be 1.2 seconds in EM including wink times, and outpulse dump. This can be decreased if the carrier can accept fast outpulse, and also be decreased if you use MF with KP ST pulses instead of DTMF. Robbed bit allows for the current channel condition to be maintained in the signalling stream. When a channel hangs up the onhook condition has to be able to be passed to the other end of the t1 for disconnect. The wink and digits dump at the start of the call only provides call setup capability. I believe in either case, the normal call features (3-way, forwarding, etc) can be provisioned. Additional features are usually handled within the switching/* system once the call has been setup. There are some features that are available via ISDN, however in my experiences most carriers don't/won't support them. Do I have it about right?? Is it pretty normal for providers to charge a premium for the PRI? Any thoughts/clarifications to my above assumptions?? Are there other pros/cons of each setup? Yes it is normal for increased cost, however IMHO I would spend the additional money (assuming one
Re: [Asterisk-Users] modprobe wcfxo crashes my IBM NetFinity5000 after few seconds
On Sat, 2004-12-18 at 15:50, Steven Critchfield wrote: On Sat, 2004-12-18 at 20:31 +, Antony Stone wrote: On Saturday 18 December 2004 20:27, Rodolfo Grave wrote: Hi and thanks once more. I moved the card around, and it kept the same IRQ. Then I went into setup and changed it. This is the output of lspci -v now: 01:04.0 Communication controller: Tiger Jet Network Inc. Intel 537 Subsystem: Unknown device 8085:0003 Flags: bus master, medium devsel, latency 144, IRQ 5 I/O ports at 4b00 [size=256] Memory at c0fdf000 (32-bit, non-prefetchable) [size=4K] Capabilities: [40] Power Management version 2 That's not a shared IRQ. However, the problem remains. Just after one min or so of executing modprobe wcfxo, the PC reboots. Any other ideas? This card worked great on another PC, so a hardware missfunctioning is not a probable choice. Was the other PC the same architecture (CPU, m/b chipset)? It may be that your motherboard simply doesn't do what Asterisk needs (I've heard that VIA chipsets in particular can be a problem, Intel ones seem okay). Previously it was posted quite a lot of good specs as to what was in this computer. It listed a serverworks chipset. Add to it, IBM wouldn't stoop to using a VIA chipset and I doubt it is the chipset having trouble. In this case, I am just about certain my favorite whipping boy problem is the culpret. RedHat is not a good choice. Fedora Core SHOULD NOT BE USED IN PRODUCTION. For the quick test, nuke the FC3 kernel and comile a fresh kernel from kernel.org. If you problems go away, add Fedora core to the doesn't work well with asterisk in stock config list. Is that Fedora Core 1,2, 3 or 3 only? Dave ___ 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] FX CallerID
On Mon, 2005-01-24 at 16:15, Matthew Boehm wrote: Follow this diagram: Many POTS lines - Many Channel Banks - Mux - DS3 ---cloud--- DS3 - demux into T1s - Many Asterisk's If someone calls into 1 of the asterisk boxes (via PRI or VoIP), and I send the call back down the line above to a POTS phone, who will provide caller id to the POTS line? I think its called FSK signaling. Anyone know anything about this? -Matthew ___ 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 WHo is the service provider for the termination of the call to the POTS line? If there is a switch in the flow, then they would provide it. If not them what type of channel banks are being used and do the FXS line cards within the channel banks support CID? Dave ___ 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: how to manage Digium TDM04B outgoing calls
On Thu, 2005-01-20 at 09:18, [EMAIL PROTECTED] wrote: Last concern about making my channels in a group and add that group in my dial plan. How can I make sure it will start with channel 4 and not pick a random one between the 3 channels as I'm pretty sure if I put in my dial plan a group having channel 2, 3 and 4 it might do the opposite and start with channel 2 then if it's busy switch to 3 and then 4 instead of 4 then 3 then 2 no? I think * start with 1, then 2, ... until it finds an available channel. I you really want it to start with 4, then 3 ... I think just re-managing your lines so that you primary number (line 1) is plugged in port 4, and vice-versa, then put all those lines in the same group, and tell * to dial by this group, it would solve your problem. If I'm wrong, please correct me ___ 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 What about the G vs g setting for hunt criteria when using groups for outdial? d ___ 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] Is it possible to ID payphone calls?
On Mon, 2005-01-17 at 11:11, Jess Coburn wrote: Hello I have a 800 DID setup to dial into my Asterisk server and I'm wondering if it's possible to ID when it's a payphone or not? I suspect it's not since I'm getting calls from someone else's SIP or IAX box. If I had a digium card installed and connected to a couple lines would I be able to get this information and parse it? Thanks, Jess ___ 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 At the risk of not being thorough enough and then being corrected! Jess, information indicator digits are made available when using a t-1 with feature group d like setup or on a PRI also setup like a Feature group D. I say like only because in most circumstances here in the US II's are not part of the original setup and you have to request to have them added by the carrier. In PRI, with an ATT switch they are generally made available through the use of Codeset 6 and are pre-pended to the ANI of the calling party. If MF separated by KP ST or DTMF delivery then separated by asterisk sign. Hope this helps! Dave ___ 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] Anyone interested in a Users-get together in Northern Virginia ?
If so please let me know off list and I will try to coordinate. Dave [EMAIL PROTECTED] 703-727-1312 Mobile ___ 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] Happy Wednesday Morning SMS question, slightly OT
On Wed, 2005-01-05 at 10:23, Jay Milk wrote: We all mostly know that * as well as various SIP phones support SMS. While the final setup is somewhat of a mystery, there are reports of those lucky souls who have it working. We also know that in order to send an SMS to a mobile phone, we need to connect to some SMS message center and get the word out that way. Now, here's the new (?) element: How can I *accept* messages on my voip-based US landline? I know that if I send an SMS from my T-Mobile phone to a friend's Verizon phone, the message goes through, so somewhere there must exist a national message center that knows which carrier to hand the message off to. Technically it should be possible to register a phone number with them to receive messages sent from cell-phones or from other * systems, and then to receive these messages through * and onto a SMS capable IP phone...? Who knows more about this? ___ 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 isn't SMS sent out via SS7? dave ___ 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] X-lIte behind NAT and Asterisk behind NAT
On Wed, 2005-01-05 at 16:03, richard wrote: Hi, I have the following scenario. I have an Asterisk server running on an internal IP address behind a firewall, and I have a remote user trying to connect to my Asterisk box behind his firewall, but he can't seem to get a connection. I have opened up the port (5060) so that he can connect through my firewall, but it still doesn't appear to want to connect. I am pretty sure that the firewall rules are correct, because I have also open up port 21, and he can successfully ssh into my Asterisk box. Any ideas/pointers? Thanks in advance Richard ___ 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 Isn't ssh on port 22? Dave ___ 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] Final call for departments
HOw about : development Dave On Wed, 2004-12-29 at 04:51, Alspach Family wrote: I am getting ready to submit a list of department names to be recorded. This is what I have so far: Accounting Accounts payable Accounts receivable Administration Billing Collections Complaint Customer Service Engineering Facilities Help desk Human Resources Information Technology Inside Sales Investor Relations Legal Mail room Marketing Printing Projects Public Relations Purchasing Receiving Sales Sales Floor Shipping Shop Support Systems Technical Support Travel If any one has additional suggestions, please e-mail them to me ([EMAIL PROTECTED] or [EMAIL PROTECTED]). I am fairly sure that none of the above exist (I was only able to search through the WIKI list, so if there are other prompts in the CVS that are not listed there, I do not know about them.) If I have made a dupe, please let me know so that I can remove it. I was fairly certain that 'Operator' was already available but I was unable to find it by its self. Thanks for your help. I plan on sending these off on Friday the 31st so please try to get them to me by then. Thanks; James ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Horrible BUZZZZ noise when sounds/music play on SIP phone?
On Wed, 2004-11-24 at 04:14, Mike Dent wrote: Hi, I've recently set Asterisk up, 1.0.2 version. With 1 x X100P card and 1 SIP phone. I've noticed some horrible buzz/rasping type of sounds! These seem to occur when * is trying to play back some audio or sound to me? E.g. If I have an exten rule which plays one of the music on hold files when I dial 800 lets say, I get a really loud buzz for about 2 seconds and then the music plays. E.g. 2. If I dial 500 to connect to Digium, as the call is connecting I get the same loud buzz noise for 0.5 seconds or so. Not sure where this is coming from? I did a search on the wiki for buzz/hum/rasp but could not find anything. Thanks Mike ___ 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 Sound like an IRQ issue. Check to see if you are sharing an interrupt on your X100P card, take a peek with cat /proc/interrupts (on linux at least) :) Dave ___ 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] Get the Caller-ID without Answering
On Sun, 2004-11-21 at 20:53, George Burt wrote:
Thanks, but that does not actually terminate the call. The phone continues
to ring until the caller hangs up.
I have done an application with cellphones that allowed allowed me to send a
signal to the phone company to drop the call. Maybe this is just a cell
phone thing.
George
Put all authorized CallerID into Asterisk database (on cli: database put
allowedcaller 1234567 1) and then do a lookup, whether CallerID is
allowed. (1234567 is CallerID)
exten = s,1,SetVar(allowed=0)
exten = s,2,DBget(allowed=allowedcaller/${CALLERID})
exten = s,3,GotoIf($[${allowed}]?5)
exten = s,4,Hangup ; Hangup if not in allowedcaller list
exten = s,5, do anything for allowed callers
Regards
bt
George Burt schrieb:
I have an application that I want to be able to verify that the call
coming
in on a PSTN 800 number is from an authorized caller.
I want to read the CallerId then terminate the call without answering it.
exten = s,1,Wair(3)
exten = s,2,NoOp(${CALLERID})
exten = s,3,Hangup()
Any ideas would be appreciated.
George
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If you have access to PRI or NET5 (ISDN) then you can send a call reject
for the call, but have not looked at the source to determine if we can
do that or not in * .
Not much assistance, but a bit of additional info.
dave
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Re: [Asterisk-Users] Multiple asterisk process
On Sat, 2004-11-20 at 01:32, Gregory Junker wrote: Add to it, my message wasn't a flame but rather a terse comment. Your Never said it was a flame. I said it was in a tone virutally guaranteed to make the guy consider you and everyone on the list to be a conceited jackass. The difference in your perception of your replies (the one I snipped included) and the way you actually come off in public, is the problem. You think you are being terse. You actually thought your post directed the guy to the answer repository. He probably did end up going to Google, but I'll bet he loses interest in Asterisk before long. I guess your work is done here then, right? If they guy isn't an expert, he has no hope of learning, huh? And they wonder why Linux doesn't catch on... Greg ___ 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 Greg, you need chill; take a deep breath; now say to yourself, let it g!! Critch, has the right to respond, anyway he desires. People need to be responsible for themselves and their actions, and in particular they need to defend themselves if they feel attacked or insulted. I have not seen a response from the individual who posted the original question (Hong) reply at all to the thread; if he isn't concerned then why are you? Why do you think the list as a whole reflects something about you, only your posts say anything about you. I don't wonder at all about Linux catching on, it is, one informed user at a time! Dave P.s. Sorry for bottom posting in my reply;) ___ 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] Starting AGI when handset is picked up?
On Fri, 2004-11-19 at 18:40, Michael Vogel wrote: Hi! Today I played arround with phpagi. I hope I can use it to completely hand over the control about outgoing calls. I don't want to use the extension.conf for that. At now I only found a method to call an agi-script when dialed at least one digit. Is it in any way possible to call an agi script when I pick up the telephone? I found the s extension. But it seems it is not planed for things like that. Bye! Michael ___ 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 Please read wiki pages on the configuration files. This is prominently displayed for your perusal! Immediate=yes in zapata.conf will start the call immediately from the s extension! Dave ___ 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] missing wakeup gsm files
On Sat, 2004-11-06 at 18:42, Ronald Wiplinger wrote: Chris Foster wrote: On Sat, 06 Nov 2004 16:42:30 +0800, Ronald Wiplinger [EMAIL PROTECTED] wrote: Where can I download the missing wakeup gsm files? Ya' kinda have to make them yourself. Do you mean NOBODY has EVER made them? Or NOBODY of them who made them is willing to give them out? Or Nobody ever succeeded to make wakeup to work anyway? bye Ronald Snip- Well Ronald, since no one on the list can speak for EVERYBODY, we don't know for sure that they have EVER been recorded. Although it would seem odd that the feature has made it in to the stable 1.0 release if Nobody has make it work. SOO why don't YOU contribute back to the community and record them for EVERYBODY else? Dave ___ 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] OpenSource Proxies ?.
On Tue, 2004-11-02 at 09:18, Whisker, Peter wrote: I have an * switch at home and one in the office. Both similar new CVS head versions and both with chan_sip2 built in: Asterisk CVS-HEAD-10/12/04-17:43:26 Asterisk CVS-HEAD-10/13/04-12:53:52 One is on a T1 connection and the other is on 576k/288k ADSL. The Ping time is about 30ms between the two servers, 90% of which is the ADSL delay. When I interconnect them with IAX2, I get rather choppy audio - with what sounds like dropped packets and jitter. SNIP- What does this have to do with Open source proxies? dave ___ 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] Digium TheVoice recordings' sound terrible
On Sun, 2004-10-24 at 10:24, Steve Totaro wrote: I know she works at Digium but they probably go down the street to a real sound stage to do the recordings via 3rd party. A sound stage is a facility used to create and process professional recordings. They can be used by anyone employed by an company. SNIP.. Take a look at the Astricon links, I believe that she is in Canada and works for several groups. Dave ___ 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] Geotel integration with Asterisk
On Sun, 2004-10-24 at 17:52, dean collins wrote: From what I read about a year ago was that it was a carrier hosted solution that actually controlled the ss7 switching at the exchange (basically no call costs from tromboning, and was only implemented into an ip-centrex or hosted call centre application. Are you saying that enterprises can buy something similar and control the carriers switching? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Greg Smith Sent: Sunday, October 24, 2004 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Geotel integration with Asterisk Geotel is a company that Cisco bought which provides call control across geographically dispersed locations. The simplest application is being able to query call queue status at another location. For example, a call comes in and can be sent to one of three call center locations. Geotel can query each location to see who is the least busy for this type of call. Traditionally it has been VERY expensive. We provide some primitive Geotel functions in-the-cloud right now. For example, we can know how many live calls are going to a location before we send the call. We can set thresholds (e.g. if a location A has over 100 concurrent calls send them to location B). Geotel can theoretically provide this and carry it further. I think there is some nice enterprise reporting that can come from the Geotel as well. G. SNIP... The GEOtel solution uses a type of interface that was originally designed for tie-in to the MCI network. The MCI network uses something call a DAP(data access point) the DAP performs a database lookup anytime that an 800,888,866,877 or virtual network number is dialed on their network. This lookup is done via SS7 and returns the appropriate routing information ie.. Switch and trunk group with appropriate DNIS to the originating switch which then routes the call to the proper termination location. The GEOtel solution actually works like a wedge into the call routing info. By using an adjunct processor that is in contact with the customers network switches/ACDs the DAP actually queries the adjunct processor for the proper routing data, and returns the appropriate info for call termination. The return data is based on whatever rules that the adjunct uses for the call lookup. The original trial for this service was used by MCI corporate for their own Customer service network Galaxy class ACD's made by Rockwell. The adjunct would poll the ACD's and determine queuing times as well as time of day number of operators etc, and return proper routing information. This was called Intelligent Routing Service (IRS) but the marketing group decided that Intelligent Call Routing was a better name. Hope this was informative in some way :) Dave ___ 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] HELP! With Postresql
Subject: [Asterisk-Users] HELP! With Postresql I am having some real problems with getting CDR records to go to a Postresql database. I think I have followed every post and instruction available and Asterisk still happily writes to a text file. Postresql is installed and working on a Redhat 9.0 box, the same one as Asterisk. I have created the CDR table in a database called Asterisk. Conf files etc are set. I even recompiled Asterisk. Any pointers would be greatly appreciated. Martin Could you provide any details to your configuration and details on the errors that you see? It is a little hard to intuit from a blank page ;) dboyd(at)fullmoonsoft.com ___ 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] VoIP hackers gut Caller ID
See bottom -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Timothy R. McKee Sent: Thursday, July 08, 2004 12:05 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID If he is routing tandem traffic he would be running IMTs and be SS-7 interconnected. Hopefully his switching/prepaid equipment would have authentication capabilities to allow the registered caller id be generated. Note this peeve is against end-users manipulating it, not service providers. This comment is aimed at ISDN BRIs, PRIs, and PBX (trunk-side) DS1s where the end-user currently is able to spoof anything desired to the service provider's switch. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Boyd Sent: Wednesday, July 07, 2004 17:48 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Timothy R. McKee Sent: Wednesday, July 07, 2004 11:58 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID This has always been one of my pet peeves, even as I worked in the industry. A telco switch operating a DS1 on trunk side should enforce caller-id numbers to be within the range of DID numbers assigned to that trunk. There should be a default DID number that is used to replace any *invalid* numbers sent on that trunk. Note that blocked caller ids would still be blocked, but the rest of the data should be corrected. Blocking ID is ok, lying about it is not. Blind trust of a non-SS7 link is a _bad_ thing. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Wednesday, July 07, 2004 10:01 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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 How then should a service provider who is routing tandem traffic place a call through any other network? This would preclude the ability for pre-paid or post paid providers to send out traffic at the originating customers request with correct callerid! Dave ___ 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 No , you don't have to be using SS7 signaling on your IMT's, 4Wire EM configured for DTMF or MF digits will provide the capability to send out ANI/Callerid to the PSTN. When 800 inbound traffic is delivered over FGD circuits the typical pattern received when set for (DTMF) is *npanxxyy
RE: [Asterisk-Users] VoIP hackers gut Caller ID
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Timothy R. McKee Sent: Wednesday, July 07, 2004 11:58 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID This has always been one of my pet peeves, even as I worked in the industry. A telco switch operating a DS1 on trunk side should enforce caller-id numbers to be within the range of DID numbers assigned to that trunk. There should be a default DID number that is used to replace any *invalid* numbers sent on that trunk. Note that blocked caller ids would still be blocked, but the rest of the data should be corrected. Blocking ID is ok, lying about it is not. Blind trust of a non-SS7 link is a _bad_ thing. Timothy R. McKee -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kevin Walsh Sent: Wednesday, July 07, 2004 10:01 To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] VoIP hackers gut Caller ID Adam Hart [EMAIL PROTECTED] wrote: Chris Foster wrote: The Register is carrying a article written by Kevin Poulsen of Securtiy Focus, calling asterisk ..the most powerful tool for manipulating and accessing CPN data.. I hope NuFone doesn't drop asterisk-set-able callerid's after this article; i've been wanting that feature from voicepluse for a long time. These kind of things will be reason (excuse) for Voip to be regulated Perhaps service providers who allow the Caller*ID to be set should insist that customers provide evidence that they own the phone numbers that they want to publish, and then limit the customers' choices to only the numbers in their approved list. Calling the customer on the provided number(s) would be an easy way to check, and a setup fee could be levied to cover the provider's time and expenses, if required. Being able to discover a blocked Caller*ID is another matter. Both are good areas for regulation. -- _/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/ _/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h _/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED] _/ _/ _/_/_/_/ _/_/_/_/ _/_/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ 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 How then should a service provider who is routing tandem traffic place a call through any other network? This would preclude the ability for pre-paid or post paid providers to send out traffic at the originating customers request with correct callerid! Dave ___ 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: [Spam] [SpamSA] [Asterisk-Users] extracting country code from a number
See Below: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of usedcanon Sent: Saturday, May 29, 2004 7:01 PM To: Asterisk users Subject: [Spam] [SpamSA] [Asterisk-Users] extracting country code from a number Hi Does anyone know of an algorythm to extract the country code from a number. I understand that the country codes are of different length and there is no fixed length of local area code or phone numbers. I am sure there is a way, if not how to telephone switches handle them Umar. ___ 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 Umar, you must have a valid list of country codes to match against. This is the only way that I know to parse them out! Dave ___ 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] EM Signaling
Does the system you are connecting to have a digital interface (T1) or is it truly a 2/4 Wire EM system? If it is T-1 then you do not need a channel bank however if it is the analog interface then you will. There are several choices of banks available and there are also several drop and insert muxes that can provide you a smaller number of channels that you can use instead of using all 24 from a channel bank. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher Sent: Monday, March 22, 2004 4:44 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] EM Signalling On Monday 22 March 2004 15:35, David J Carter wrote: I may need to connect to a system with EM connectivity. Am I right in assuming a T1 card and Channel Bank will give me this connectivity? You typically do not need a channel bank when using EM. However, you will probably need a T1 crossover cable, if you are not connecting to the outside world (i.e. cross 12 with 45). -Tilghman ___ 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
