[asterisk-users] Recording from a script
Greetings, I have been asked to provide a one off solution for someone. They would like to take a message left on a remote voicemail system (with their mobile phone provider) and get it to a wav/mp3 file. There is a number I can call from my Asterisk system that would allow playback of the message, but it would require sending some DTMF tones to do it (traversal of the remote IVR on the voicemail system) I would then have to record the resulting message (even if I can just use record() and get it to GSM I can the transcode it to what I want). In short I would like to know: a) if this is actually possible b) if anyone can give me some pointers on how I might go about automating this with a script. Thanks in advance. -- Nikolai Lusan # # # Weblog: http://lusan.id.au/~nikolai/blog # Website:http://lusan.id.au/~nikolai # # ___ --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: ZapHFC quadBRI D-Channel going down randomly
asterisk ha scritto: On most traditional pabx's it's possible to set layer 1 to permanent or call. It sounds like your system is configured for permanent and your lines to call. How you would set this on asterisk I have no idea. fadge The question is: is it possible I am the only one with such problems on all asterisk boxes on different sites and different ISDN lines? I've googled around on many forums but no one seems to have this one. The old replaced PBXs had layer 1 set for call, as you say, and they showed no problems at all. With asterisk as a PBX, every 2-3 hours, you cannot dial out for 5 to 15 minutes then everything gets back to normal (no idea about what triggers the return to working state). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alberto Pastore Sent: 16 October 2006 17:26 To: asterisk-users@lists.digium.com Subject: [asterisk-users] ZapHFC quadBRI D-Channel going down randomly Hi. I'm running some asterisk boxes on different sites, some equipped with a couple of ZapHFC cards, others with Junghanns quadBRI cards. All boxes were compiled with Asterisk 1.2.10 (libpri 1.2.3 zaptel 1.2.6) and bristuff 0.3.0 pre 1s, distribution is Fedora Core 4 with kernel 2.6.17.3 The cards are connected to Telecom Italia's NT1/NT1+ S/T lines; some of them are point-to-point, others are point-to-multipoint. I keep getting always the same problem: after some hours of regular working, some boxes report the usual message Primary D-Channel on span n down (where n is different every time, depending on the number of active bri spans) I've read on previous postings that having layer 1 down on ptmp spans is normal. However after getting a down message (on ptp spans too!) I'm no more able to place outgoing calls on that span, until I restart asterisk zaptel drivers. Sometimes, they get back working by themselves (with the related span up notification) after a random time period. During the down period, incoming calls are regularly served. However these calls do not change the status of the span, i.e. as soon as the calls are hung up, the span gets down again. I've tried to capture the dialog between the card and NT1 equipment, and during the down state, I got this repeated over and over: Sending Set Asynchronous Balanced Mode Extended [ 00 8b 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 069EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] == Primary D-Channel on span 1 down In zapata.conf I'm pretty sure I've always set the correct signalling settings (switchtype = euroisdn, signalling = bri_cpe_ptmp or bri_cpe depending on the case) In /etc/zaptel.conf, I've tried many combinations with no difference; my current settings are like this: span=1,1,0,ccs,ami bchan=1-2 dchan=3 span=2,1,0,ccs,ami bchan=4-5 dchan=6 etc Any clue? Thanks, Alberto -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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 -- -- Alberto Pastore B-Press Srl - Gruppo MSoft P.IVA 01697420030 P.le Lombardia, 4 - 28100 Novara - Italy Tel. 0321-499508 Fax 0321-492974 http://www.msoft.it ___ --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] Spandsp and tif
2006/10/5, [EMAIL PROTECTED] [EMAIL PROTECTED]: try the rx_fax and tx_fax below the snapshot-tree within test-apps-asterisk-1.x http://www.soft-switch.org/downloads/snapshots/spandsp/ [EMAIL PROTECTED] schrieb am 04.10.2006 22:11:43: 2006/10/4, Steve Underwood [EMAIL PROTECTED]: Giedrius Augys wrote: Hi, Now I'm testing faxes with spandsp. I have problems that spandsp do not add headers to fax page: LOCALHEADERINFO. Please help me. There is a bug in adding page header with spandsp-0.0.2pre26. I have fixed this in the development code, but I haven't yet put the fix into the 0.0.2prexx series. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I have installed spandsp 0.0.3 , but I couldn't install rx_fax and tx_fax(from 0.0.2pre release) , because I've got error. I also have problem with tiff files, because I get error, if I have created tiff file from MS WORD (printing to tiff file) . Maybe you can say what parameters/atributes and programs I must choose, that avoid these erorrs (there is no problem with tiff fiiles created by rxfax :) ). Can you give me some advices how to solve these problems? Thanks ___ --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 listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-usersHi againI want to ask does spandsp works with t.38 fax protocol. And what about stability ? ___ --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] nat auto detect ?
Hello ppl, This post is to do with the variables 'nat' or 'canreinvite' for sip entities. Idealy users, wont be static, they could be roaming all over the globe. So, setting someone as behind NAT, and disabling canreinvite, etc., restricts the roaming capabilities of a user. Is there any way for Asterisk to auto detect, if a user is behind NAT, also, if two users are behind the same NAT, help in having a peer-to-peer rtp flow between the two users in the call?? cheerz - Ben ___ --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: Is 1.2.12.1 production ready
On 2006-10-16 20:54:09 -0700, Mike Lynchfield [EMAIL PROTECTED] said: reboots are wise No, they are foolish... snip ___ --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: Stopping putgoing calls after working hours
On 2006-10-16 17:10:49 -0700, Lacy Moore - Aspendora [EMAIL PROTECTED] said: So I was wondering is there a way to make this happen in asterisk?? Depending on where you are located, you might want to allow emergency calls to go through. The bloodsuckers, I mean attorneys, here in the US would have a field day if something were to happen to someone at a company that did not allow emergency numbers to be dialed. Translated: If something were to happen to someone outside of business hours (in the US), and the phones did not allow emergency calls, it would cost your company millions of dollars. As well it should. Thanks for the very important reminder. ___ --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] 1.4 Beta and oracle
Morning all, I updated to 1.4 now but it seems the oracle is not working with it? I get error with 1.2 all is fine: Mar 29 08:10:54 WARNING[3876] config.c: Realtime mapping for 'sippeers' found to engine 'oracle', but the engine is not available Mar 29 08:10:54 NOTICE[3876] chan_sip.c: Registration from 'sip:[EMAIL PROTECTED]' failed for xx.xx.xx.x- Username/auth name mismatch Mar 29 08:10:58 WARNING[3876] config.c: Realtime mapping for 'realtime_ext' found to engine 'oracle', but the engine is not available Mar 29 08:10:58 WARNING[3876] config.c: Realtime mapping for 'realtime_ext' found to engine 'oracle', but the engine is not available Mar 29 08:10:58 WARNING[3876] config.c: Realtime mapping for 'realtime_ext' found to engine 'oracle', but the engine is not available Mar 29 08:10:58 WARNING[3876] config.c: Realtime mapping for 'realtime_ext' found to engine 'oracle', but the engine is not available regards rene --RenéEnskatInternet-Administrator Teamware GmbHStahlgruberring 11D-81829 MünchenTel: 089-427005.31Fax: 089-427005.55E-Mail: [EMAIL PROTECTED]http://www.tmwr.de ___ --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-ooh323c Video ?
16 okt 2006 kl. 17.34 skrev Patrick: I know this question has been asked a great deal, but does any1 have a simple way Of getting video to work using this particular channel... Or at least is it possible just using the conf files, or do I Have to have a separate decoder to encode the video Thanks again There is not video support at all in any of the H.323 channels at this point in time. There are work in progress for the H.323 channel based on OpenH323 (which includes video support) but I don't know the state of that project. Regards, /olle --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ * Next training: Stockholm, Sweden, November 13-17 2006 ___ --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] Unable to open Asterisk database
Ciao Andrea, thanks for answering. Indeed you are rightit was some kind of strange problem, infact after setting the correct native music-on-hold (but why??) everything got right. It is such a strange thing I'm still a bit confused about it so didn't post the solution :)) Thank you Giorgio Incantalupo Andrea Spadaccini wrote: Ciao Giorgio, I'm using mysql to store my cdr data. I compiled asterisk-addon module without problems and I see nothing unusual in my cdr_mysql.conf but when I do a reload I get this messages (never seen before): Oct 16 09:43:16 WARNING[8576]: db.c:67 dbinit: Unable to open Asterisk database Oct 16 09:43:16 WARNING[8576]: db.c:423 ast_db_gettree: Database unavailable But If I try to connect from shell it works without any problem. Does anybody know why? I think that the error message refers to the Asterisk internal database (AstDB), and not to MySQL. This doesn't clarify the error, but might explain why you get a working CDR. Try to issue the db get and db put CLI commands, to see if AstDB is working. HTH, ___ --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] Psst... Top secret information: Codename Pineapple
14 okt 2006 kl. 09.44 skrev Brian Candler: On Fri, Oct 13, 2006 at 07:00:54PM -0500, Eric ManxPower Wieling wrote: * Phones = stations, regardless of where they are Asterisk = SIP Server, Phone = SIP Client * Trunks = trunks to other SIP servers, bilateral Asterisk and the other server is peer to peer * Services = services you register for, like BroadVoice, Voop or FWD. (where asterisk acts as a phone) Asterisk = SIP Client, Other End = SIP Server Hmm, but I don't see how these ideas map to formal SIP concepts (RFC 3261). Let's try to clarify then. phones are devices that connect to Asterisk. They register with Asterisk acting as a SIP location server/registrar and use Asterisk as the outbound SIP proxy. They get calls from Asterisk and place calls to Asterisk. The phone use one of the SIP domains that are hosted within your Asterisk server. (this is like the current friend) service is when Asterisk is the UA, acting as a phone towards another SIP server - we register with a SIP location server/registrar to get incoming calls. We place calls, masquerading as a phone (using the registrars domain). Currently, this is a mixture between a peer (matched on IP for incoming calls) and a register= statement. In some cases, two peers and a register= statement. Very confusing. trunk is when we exchange traffic with another server. We send calls to their SIP domain and receive calls to our SIP domain. We may use realm based authentication for the incoming part of the trunk (not based on caller ID/From: header) and a combination of SIP domain and ACLs. This is currently handled by defining sip peers for outbound calls and separate SIP peers for inbound calls - where we match on IP. The problem with the IP matching is when a trunking partner use several SIP servers to connect to us, we need to define one peer per server instead of just matching on domain and then authenticate. In all cases, we're a SIP user agent client/server in SIP terminology. In fact, we're a super-SIP ua called a B2BUA. I am trying to avoid sip client since the whole user/peer client/server concept does not really match SIP. In some cases, we're the SIP registrar/location server and in other we're configured as the outbound proxy, even though we are not a proxy. I hope I did not add to the confusion by this confusing message. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ - Stockholm, Sweden, November 13-17 ___ --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] Page hangs up after 5 seconds
OK... A bit more research done... This problem does not occur in version
1.2.7.1, which was the platform where we developed our dialplan.
Looking at a diff between app_page.c for the two version reveals that
the only change that has been done is the addition of (5) to the w option:
1.2.7.1, line 182:
snprintf(meetmeopts, sizeof(meetmeopts), %ud|%sqxdw, confid,
ast_test_flag(flags, PAGE_DUPLEX) ? : m);
1.2.12.1, line 182:
snprintf(meetmeopts, sizeof(meetmeopts), %ud|%sqxdw(5), confid,
ast_test_flag(flags, PAGE_DUPLEX) ? : m);
Why this change? And I can't imagine that it is the intended behaviour.
Hasn't anyone else noticed this? Or are we doing something fundamentally
wrong?
I still do not understand what the usage and result of the w option are,
could someone elaborate?
// Torbjörn
Torbjörn Abrahamsson wrote:
Hi asterisk-users,
We are using Asterisk 1.2.12.1, and are trying to use the Page
application. It seems to work but after approx 4-5 seconds the call is
hung up.
The dialplan code look like this:
exten = _*2XX,1,AGI(get-paging-devices.agi,${EXTEN:2})
exten = _*2XX,n,GotoIf($[ ${PAGING_DEVICES} = invalid ]?i,1)
exten = _*2XX,n,SIPAddHeader(Call-Info: sip:192.168.20.1\; answer-after=0)
exten = _*2XX,n,Page(${PAGING_DEVICES},dq)
The CLI outputs the following:
-- Executing AGI(SIP/snom1-b7d0c328, get-paging-devices.agi|01)
in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/get-paging-devices.agi
-- AGI Script get-paging-devices.agi completed, returning 0
-- Executing GotoIf(SIP/snom1-b7d0c328, 0?i|1) in new stack
-- Executing SIPAddHeader(SIP/snom1-b7d0c328, Call-Info:
sip:192.168.20.1; answer-after=0) in new stack
-- Executing Page(SIP/snom1-b7d0c328, SIP/snom1SIP/snom3|dq) in
new stack
-- Created MeetMe conference 1023 for conference '2028709590d'
-- Launching MeetMe(2028709590d|qxdw(5)) on SIP/snom3-08984140
-- Hungup 'Zap/pseudo-1436409106'
== Spawn extension (wx3trunk2, *201, 4) exited non-zero on
'SIP/snom1-b7d0c328'
-- Executing Hangup(SIP/snom1-b7d0c328, ) in new stack
The 'full' log has this contents:
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'Goto'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing
Goto(SIP/snom1-b7d0c328, wx3trunk2|*201|1) in new stack
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Goto (wx3trunk2,*201,1)
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'AGI'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing
AGI(SIP/snom1-b7d0c328, get-paging-devices.agi|01) in new stack
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Launched AGI Script
/var/lib/asterisk/agi-bin/get-paging-devices.agi
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- AGI Script
get-paging-devices.agi completed, returning 0
Oct 16 11:01:12 DEBUG[6767] pbx.c: Expression result is '0'
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'GotoIf'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing
GotoIf(SIP/snom1-b7d0c328, 0?i|1) in new stack
Oct 16 11:01:12 DEBUG[6767] pbx.c: Not taking any branch
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'SIPAddHeader'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing
SIPAddHeader(SIP/snom1-b7d0c328, Call-Info: sip:192.168.20.1;
answer-after=0) in new stack
Oct 16 11:01:12 DEBUG[6767] pbx.c: Launching 'Page'
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Executing
Page(SIP/snom1-b7d0c328, SIP/snom1SIP/snom3|dq) in new stack
Oct 16 11:01:12 DEBUG[6767] chan_sip.c: sip_answer(SIP/snom1-b7d0c328)
Oct 16 11:01:12 DEBUG[6767] app_meetme.c: Building dynamic conference
'2028709590d'
Oct 16 11:01:12 DEBUG[6767] chan_zap.c: Using channel -2
Oct 16 11:01:12 VERBOSE[6767] logger.c: -- Created MeetMe conference
1023 for conference '2028709590d'
Oct 16 11:01:12 DEBUG[6767] channel.c: Set channel SIP/snom1-b7d0c328 to
write format slin
Oct 16 11:01:12 DEBUG[6767] channel.c: Set channel SIP/snom1-b7d0c328 to
read format slin
Oct 16 11:01:12 DEBUG[6767] app_meetme.c: Placed channel
SIP/snom1-b7d0c328 in ZAP conf 1023
Oct 16 11:01:12 DEBUG[6772] app_queue.c: Device 'SIP/snom1' changed to
state '2' (In use) but we don't care because they're not a member of any
queue.
Oct 16 11:01:12 DEBUG[6773] app_queue.c: Device 'Zap/pseudo' changed to
state '2' (In use) but we don't care because they're not a member of any
queue.
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Allocating new SIP dialog for
(No Call-ID) - INVITE (With RTP)
Oct 16 11:01:12 DEBUG[6771] res_config_mysql.c: MySQL RealTime:
Everything is fine.
Oct 16 11:01:12 DEBUG[6771] res_config_mysql.c: MySQL RealTime: Retrieve
SQL: SELECT * FROM sipusers WHERE name = 'snom3'
Oct 16 11:01:12 VERBOSE[6771] logger.c: -- SIP Seeding peer from
astdb: 'snom3' at [EMAIL PROTECTED]:59283 for 60
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Allocating new SIP dialog for
(No Call-ID) - OPTIONS (No RTP)
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Setting NAT on RTP to 524288
Oct 16 11:01:12 DEBUG[6771] chan_sip.c: Outgoing
[asterisk-users] Re: 1.4 beta2 on intel mac
On 2006-10-16 03:22:47 -0700, Tim Panton [EMAIL PROTECTED] said: On 16 Oct 2006, at 09:09, Martin Joseph wrote: On 2006-10-15 23:50:34 -0700, Tim Panton [EMAIL PROTECTED] said: On 16 Oct 2006, at 07:15, Martin Joseph wrote: On 2006-10-12 02:40:35 -0700, Tim Panton [EMAIL PROTECTED] said: On 11 Oct 2006, at 19:35, Dean Collins wrote: Lol - use a real PC maybe :P Nah, that would be dull. In some ways the mac intel is nearer to a 'normal PC' (whatever that is) than the systems I normally run asterisk on - a NatSemi Nemiah and an arm5 :-) Asterisk 1.2.X runs fine on the intel macs, so I guess there must be a bug in 1.4beta2 that stops it running. Did you need to update the version of Make? My PowerPC mac seems to be complaining about version 3.80. I don't have any Intel mac's to test with (yet). Yes. I had to install a new make from source (with configure -- prefix=/usr) I've got some stuff to get ready for Astricon Dallas next week, where we will be launching Corraleta SDK - our zero install web- based Java softphone. Once that's done and I get back I'll look into what the problem is (unless someone solves it for me while I'm there - drop by our stand in the exhibition If you have got 1.4beta2 working on an intel mac - or if you want to see Corraleta in action! ) Tim Panton www.mexuar.com I just built 1.4b2 on a powerpc mac system, and although it seems to build ok, and starts up, the command line is completely non- reponsive (although exit works). I have head people describe this kind of dead CLI in the past, but never saw it before. 1.4b2 doesn't accept registrations or do anything. If this sounds like what you saw, then I guess it's an OSX issue and not an Intel OSX issue specifically. Yep exactly that, I'll grab an SVN head when I get back from Astricon and try that (unless beta3 comes out first) SVN Trunk doesn't currently build on OSX (10.4.8). Although the 1.40b2 tarball builds it doesn't work. Trunk appears to work until you do a make install, and then it quits trying to wget some sound stuff. Oh well, hopefully beta 3 helps? ___ --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] Call Forwarding Using Asterisk
Can I do this with Asterisk, Call comes to Asterisk Server (Master), Then master just forwards calls to other slave asterisk servers one by one. Like this Master forward 1st call to Slave 1, Second call to Slave 2, Third call to slave 1 Fourth call to slave 2. Is it possible? I will appreciate if some one help me with this. Thank you, -Jai ___ --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: 1.4 beta2 on intel mac
On Tue, Oct 17, 2006 at 12:57:46AM -0700, Martin Joseph wrote: SVN Trunk doesn't currently build on OSX (10.4.8). If you're in for stability now, try branches/1.4 and *not* trunk. This will eventually become beta3, rc or 1.4.0. -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] Why is this happening?
On Mon, Oct 16, 2006 at 05:25:22PM -0400, Time Bandit wrote: Thanks for the answer, but I don't buy it. There are currently 0 calls up on that bridge, while another connection which has calls up on it is on Port 4569.. please try again. IAX2 is suppose to run on ONLY one port.. this is why it is so nice for use in firewall situations. It doesn't change a thing ! Same thing happens with a webserver. It listen for connections on port 80 (default port) and when a connection comes in, it is handed to another free port on the server so the main server can continue listening on port 80. Same thing with FTP, etc. All TCP servers that accept more than one connection For the benefit of the archives, I'd just like to point out that this description is entirely wrong. Each TCP connection has four parameters associated with it: - local IP address - local port - remote IP address - remote port It is all four together which uniquely identifies a TCP connection. A webserver uses local port 80 for *all* inbound connections, and that is for the *entire* duration of each connection. It does not somehow magically change the local port number after accepting the connection. Additional connections can be accepted because they have a different remote IP address (if they are coming from a different machine) or a different remote port (if they are coming from another socket on the same machine) Check on your machine while you're surfing the web, your browser doesn't use port 80 as the originating port. Now, that is correct; the browser (the client) picks a port 1024 for its end of the connection. However check your netstat output and you'll also see the far side (server) is port 80. Connect to an FTP server and check your netstats, you'll see that you're not connected to port 21 on the remote server No, you'll see that you *are* connected to port 21 on the remote server. However your local port number will be something else. Active Internet connections Proto Recv-Q Send-Q Local Address Foreign Address(state) tcp4 0 0 172.31.131.189.62505 69.16.138.164.21 ESTABLISHED ^ ^^ Regards, Brian. ___ --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] How to get Linksys-Sipura error codes ?
Hi,Through our reseller, we couldn't get any clue to Linkys-Sipura products error codes.This keeps us from analysing Syslogs.Has anyone a clue to get this error codes and their meaning ?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
Re: [asterisk-users] Monitor stops recording midstream?
On Mon, 2006-10-16 at 08:00 -0500, Tim Connolly wrote: Asterisk SVN-trunk-r7230 built by root @ pbx01.timsnet.com on a i686 running Linux on 2006-06-17 When I used monitor, I seem to get most calls cut off if they run very long. Sometimes two minutes, sometimes 5 or 15.. Seems random. Any ideas what might kill the recording process? I'm beginning to wonder if soxmix is truncating the file when it blends the in/outbound streams together due to bad data or something. Same here. I have kept the original wav files from asterisk to check wether it is sox that truncates it and can confirm it is not sox but asterisk cutting them off. There also appears to be a timing problem, because left+right channel become out-of-sync. This is even in the wav files created by asterisk - before I run them through sox. I didn't spend more time on it because it's only a nuisance not a big problem for us, but now at least you know you're not alone ;) Conrad ___ --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] Recording from a script
On Tue, 2006-10-17 at 16:00 +1000, Nikolai Lusan wrote:
Greetings,
I have been asked to provide a one off solution for someone. They would
like to take a message left on a remote voicemail system (with their
mobile phone provider) and get it to a wav/mp3 file. There is a number I
can call from my Asterisk system that would allow playback of the
message, but it would require sending some DTMF tones to do it
(traversal of the remote IVR on the voicemail system) I would then have
to record the resulting message (even if I can just use record() and get
it to GSM I can the transcode it to what I want).
In short I would like to know:
a) if this is actually possible
b) if anyone can give me some pointers on how I might go about
automating this with a script.
Thanks in advance.
My first shot at it would be
exten = getmsg,1,Monitor(wav,/tmp/msgdir/${UNIQUEID},m)
exten = getmsg,2,Dial(${VOICEMAILNUMBER}www${VOICEMAILDTMF})
I think there was an option for Dial() to hangup after some time
elapsed, but ideally you would detect end-of-voicemail tone if there is
one.
Conrad
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[asterisk-users] how to config chanspy
hi all, please any one help me ,how to configure chanspy application . and also send me if u have any sample configure file. -thiru ___ --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] Remote UNIX connection, Remote UNIX disconnected displayed every second
Hi Brian, yes, I have more copies of safe_asterisk running, I know this is the underline problem but I do not how to solve it because I do not know how to reproduce it. I'm still looking the safe_asterisk for some strange but found nothing till now. Have you got the same problem? Why is it happening? TIA Giorgio Incantalupo ___ --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] sending sip style messages in response
Hello ppl, Is it possible to send SIP messages as response to the calling UA on failure, for e.g. if a number is blacklisted I would want to send Forbidden to the caller, not just for user comfort but also for testing purposes? I see only Congestion available which sends Service Unavailable. cheerz - Ben ___ --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] chan_bluetooth, mobile handset as VoIP terminal?
I have been looking at chan_bluetooth, so far being unable to make it compile with Asterisk SVN trunk. I was wondering about the different ways it can be used. What I have read so far implies two possibilities: 1. Asterisk pretends to be a handsfree unit, and can use the cell phone for placing calls over the mobile network, or answer inbound calls from the mobile network. 2. Asterisk pretends to be a phone, and you can use a headset as a VoIP terminal (unfortunately only useful for receiving calls, as headsets don't have keypads) However, the possibility which really interests me is: 3. Can I use my mobile phone as a terminal, originating calls over bluetooth via Asterisk, using the phone's keypad to dial? And answering inbound calls from Asterisk? This makes my mobile phone into a cordless phone replacement - avoiding mobile charges while at home, and being able to receive PSTN and VoIP calls via Asterisk. I notice BT's Fusion service appears to work in this way - http://www.bt.com/btfusion/ - as it looks like you get a normal mobile phone which can route VoIP calls via Bluetooth and DSL when in range of the base station. So the question is: - has anyone got Asterisk working this way? - what bluetooth profile would the phone need to support to do this? - does that limit me to particular models of mobile phone? If this is possible, it's not clear to me how the phone would know that I wanted to set up a call over bluetooth rather than over the mobile network. Would I need to load some sort of app onto the phone? Thanks, Brian. ___ --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] Why the MusicOnHold sound so soft?
My MusicOnHold sound is very soft, but when I hear it directly from mp3 playe on desktop, the loudness is quite ok. Wonder whether there is any configuration to change the loudness of MusicOnHold. Regards, Liangliang ___ --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] Why the MusicOnHold sound so soft?
On Tue, 2006-10-17 at 17:18 +0800, Xue Liangliang wrote: My MusicOnHold sound is very soft, but when I hear it directly from mp3 playe on desktop, the loudness is quite ok. Wonder whether there is any configuration to change the loudness of MusicOnHold. If you play it with mpg123 you can try the -g option. Alternatively you can change the volume of the file(s) itself with sox. I'm not aware of any volume setting to musiconhold. Conrad ___ --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] Inaccurate CDRs
Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls ( i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my call disposition colume. I have also noticed that both the call duration and the billsec are always equal. here is my setup below PSTN va E1 (Primary Asterisk) =Sip and IAX trunks (Secondary PBX) Clients are connected to the Secondary PBX. this pbx handles registration of all clents. The billing irregularities happen on the Secondary PBX. When a call is maked from the Secondary and it is routed across the trunks,call dispositionalways registeres 'AWNSERED', unless the Primary PBX sends back a busy signal. the duration and billsecs are always equla. this means that the user gets billed for ring time, and calls disconnected from the Secondary PBX Can someone help me out here ? Thanks ___ --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] Stopping putgoing calls after working hours
Dear Lacy Thx Lacy for this important reminder we engineers do tend sometimes to forget about all the law part, indeed while I was putting down the implementation we do have exceptions we have a 24x7 call center and ofcourse the emergency number. Thx MAG Lacy Moore - Aspendora wrote: So I was wondering is there a way to make this happen in asterisk?? Depending on where you are located, you might want to allow emergency calls to go through. The bloodsuckers, I mean attorneys, here in the US would have a field day if something were to happen to someone at a company that did not allow emergency numbers to be dialed. Translated: If something were to happen to someone outside of business hours (in the US), and the phones did not allow emergency calls, it would cost your company millions of dollars. ___ --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 -- Thx MAG ___ --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] sending sip style messages in response
Benjamin Jacob wrote: Hello ppl, Is it possible to send SIP messages as response to the calling UA on failure, for e.g. if a number is blacklisted I would want to send Forbidden to the caller, not just for user comfort but also for testing purposes? I see only Congestion available which sends Service Unavailable. Hangup(CALL_REJECTED) or Hangup(21) should work, I think. Mikael ___ --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] how to activate recording (automon)
Hi all, If I activate recording for an extension everything is OK. but If I activate call recording on demand i am non able to start recording In principle I should have to press *1, as indictaed in features.conf (I am using almost last asterisk code, updated 2 days ago from svn, version SVN-branch-1.2-r39379M ) Actually it produce no effect at all I am using FreePBX interface, and I saw under General Setting two fields, denoted Asterisk Dial command options and Asterisk Outbound Dial command options Here the help says something about w and W options, but every combination of this options does not produce anything Anyway, apart from FreePBX, what I have to check ? And moreover, what are the correct actions to do to record a call ? Let's say extension 555 calls extension 567, 567 answers the call and then press *1 and no other key ? I am trying with at320 sip phones and snom 320 sip phones thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --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] sending sip style messages in response
Nope Mikael, * always seems to send a Decline, n also plays an unavailable file. I tried another scenario, A calls B, B rejects the call. In the tcpdump I see B sending a Forbidden to *, but * sends a Service Unavailable to A. hmm... not too sure, why this decision was made. Mikael Magnusson wrote: Benjamin Jacob wrote: Hello ppl, Is it possible to send SIP messages as response to the calling UA on failure, for e.g. if a number is blacklisted I would want to send Forbidden to the caller, not just for user comfort but also for testing purposes? I see only Congestion available which sends Service Unavailable. Hangup(CALL_REJECTED) or Hangup(21) should work, I think. Mikael ___ --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] TIMEOUT() function missing
Hello everybody, I want to use the TIMEOUT() function, but in the CLI the show functions command only shows 7 custom functions: QUEUEAGENTCOUNT SORT CUT CHECKSIPDOMAIN SIPCHANINFO SIPPEER SIPHEADER In addition, sometimes I get the debug message function LANGUAGE not registered. How can I install those functions? I'm using Asterisk 1.2.10. Thanks in advance, -- Andrea Spadaccini Multimedia Technologies Institute s.r.l. ___ --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] Inaccurate CDRs
Dumpolid Exeplish wrote: Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls ( i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my call disposition colume. I have also noticed that both the call duration and the billsec are always equal. here is my setup below PSTN va E1 (Primary Asterisk) =Sip and IAX trunks (Secondary PBX) Clients are connected to the Secondary PBX. this pbx handles registration of all clents. The billing irregularities happen on the Secondary PBX. When a call is maked from the Secondary and it is routed across the trunks, call disposition always registeres 'AWNSERED', unless the Primary PBX sends back a busy signal. the duration and billsecs are always equla. this means that the user gets billed for ring time, and calls disconnected from the Secondary PBX Can someone help me out here ? Hi, I have not had this particular problem, but I had it where my billsec were wrong for some other reason. Try callprogress=yes in zapata.conf, although I dont even think this will help, but you can try. -- thanks, yusuf -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ --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] Warning translate.c:88 powerof
Hi, When I dial out the calls get complete and the audio is ok but I'm getting some warnings (The complete output is here: http://pastebin.ca/206342) Oct 17 13:08:10 WARNING[84615]: translate.c:88 powerof: Powerof 0: No power?? Oct 17 13:08:10 WARNING[84615]: translate.c:88 powerof: Powerof 0: No power?? I've searched on google for this error but I could not find any good answer... Could somebody give me a little help understanding this? Thank you in advance _ MSN Messenger: instale grátis e converse com seus amigos. http://messenger.msn.com.br ___ --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: Reception Console
[EMAIL PROTECTED] is believed to have said: We are currently writing a reception console for Asterisk - if anyone is interested in beta testing it, feel free to ask. Paul Hales Does it run on *nix (Linux/MacOSX)? Is there a place we can see some information without cluttering the list? TIA Aldo ___ --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] Inaccurate CDRs
On 10/17/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls ( i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my call disposition colume. I have also noticed that both the call duration and the billsec are always equal. here is my setup below PSTN va E1 (Primary Asterisk) =Sip and IAX trunks (Secondary PBX) Clients are connected to the Secondary PBX. this pbx handles registration of all clents. The billing irregularities happen on the Secondary PBX. When a call is maked from the Secondary and it is routed across the trunks, call disposition always registeres 'AWNSERED', unless the Primary PBX sends back a busy signal. the duration and billsecs are always equla. this means that the user gets billed for ring time, and calls disconnected from the Secondary PBX Could you provide a snippet of the dialplan used on each of the primary and secondary boxes to complete a call? For example, is the primary executing an Answer() before it does the onward Dial() on behalf of the secondary? Cheers, Steve ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inaccurate CDRs
thanks for your response yusuf, but the problem is actually on the secondary PBX. The CDR beign generated by the Primary (i.e. the asterisk box that carries the E1s) is very accurate. The reason i can use this CDR for bill is because it does not containg user extensions and account codes. Thanks On 10/17/06, yusuf [EMAIL PROTECTED] wrote: Dumpolid Exeplish wrote: Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls ( i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my call disposition colume. I have also noticed that both the call duration and the billsec are always equal. here is my setup below PSTN va E1 (Primary Asterisk) =Sip and IAX trunks (Secondary PBX) Clients are connected to the Secondary PBX. this pbx handles registration of all clents. The billing irregularities happen on the Secondary PBX. When a call is maked from the Secondary and it is routed across the trunks, call disposition always registeres 'AWNSERED', unless the Primary PBX sends back a busy signal. the duration and billsecs are always equla. this means that the user gets billed for ring time, and calls disconnected from the Secondary PBX Can someone help me out here ?Hi,I have not had this particular problem, but I had it where my billsec were wrong for some otherreason.Try callprogress=yes in zapata.conf, although I dont even think this will help, but you can try.--thanks,yusuf--This message has been scanned for viruses anddangerous content by MailScanner, and isbelieved to be clean.___ --Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo 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] how to activate recording (automon)
Hi Andrea, Try the following: featuredigittimeout=1500 ; Slow down digits for the record [featuremap] automon = *0 ; One Touch Record Use both option switches(wW) Check that the dial plan on your SIP phones doesn't preclude this feature code. Henry L.Coleman CEO *VoIP-PBX* 1-866-415-5355 Toronto Ontario Canada Hi all, If I activate recording for an extension everything is OK. but If I activate call recording on demand i am non able to start recording In principle I should have to press *1, as indictaed in features.conf (I am using almost last asterisk code, updated 2 days ago from svn, version SVN-branch-1.2-r39379M ) Actually it produce no effect at all I am using FreePBX interface, and I saw under General Setting two fields, denoted Asterisk Dial command options and Asterisk Outbound Dial command options Here the help says something about w and W options, but every combination of this options does not produce anything Anyway, apart from FreePBX, what I have to check ? And moreover, what are the correct actions to do to record a call ? Let's say extension 555 calls extension 567, 567 answers the call and then press *1 and no other key ? I am trying with at320 sip phones and snom 320 sip phones thanks in advance, Andrea Chi ricevesse questa mail per errore e' gentilmente pregato di cancellarla. Visitate il sito http://www.frameweb.it ___ --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] Re: Centos kernel 34 vs. 42? [was: asterisk-users Digest, Vol 27, Issue 72]
On Sun, 15 Oct 2006, Les Bell wrote: Cutting to the chase: I'm not aware of any audio problems, but our system doesn't get heavy use (only two lines and eight phones). OK, thanks for the reply. The anouncement at trixbox.org is not very clear on this. There is reference to 'distorted voice prompts' but not about general voice quality. From trixbox.org : quote There are some strange audio problems with the 42 kernel. This is most apparent with vmware. When the 42 kernel is used the audio prompts are jittery and broken. This was not a problem with the 34 kernel. I am making the 34 kernel the standard for trixbox until further notice. If somebody has a good reason to move to a higher release kernel please post to the forum. But for now this should resolve all the problems with Zaptel. This update has a new yum configuration file that will keep yum from updating the kernel. If you want to update the kernel you can do it manually unquote I just went back to kernel 35 just in case. ___ --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] Is 1.2.12.1 production ready
We have several sites in this configuration with no nightly reboots. All sites except one are problem free. One site still has dropped calls. None of the sites crashes and some of them have been up for a few weeks. Tom Vile wrote: fine for me here since it came out. We are running 15 extension all day long. On 10/16/06, *shadowym* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am getting ready to image a production system. Right now I am planning on using Centos 4.4, Asterisk 1.2.12.1 http://1.2.12.1, Freepbx 2.1.3. I will be using a Sangoma A200D card. I read of some people having problems with Asterisk 1.2.12.1 http://1.2.12.1 crashing. Is this across the board or is there anyone out there with no problems. If you have 24/7 uptime and no nightly reboot crons I would definitely appreciate hearing about it. Cheers ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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] Inaccurate CDRs
Well I am using APM on the two boxes i have modified the srripts extensievely and i am sure that there is no Awnser befor a dial when Dialing through the PBX trunks On 10/17/06, Steve Davies [EMAIL PROTECTED] wrote: On 10/17/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls ( i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my call disposition colume. I have also noticed that both the call duration and the billsec are always equal. here is my setup below PSTN va E1 (Primary Asterisk) =Sip and IAX trunks (Secondary PBX) Clients are connected to the Secondary PBX. this pbx handles registration of all clents. The billing irregularities happen on the Secondary PBX. When a call is maked from the Secondary and it is routed across the trunks, call disposition always registeres 'AWNSERED', unless the Primary PBX sends back a busy signal. the duration and billsecs are always equla. this means that the user gets billed for ring time, and calls disconnected from the Secondary PBXCould you provide a snippet of the dialplan used on each of theprimary and secondary boxes to complete a call?For example, is the primary executing an Answer() before it does the onward Dial() on behalf of the secondary?Cheers,Steve___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inaccurate CDRs
this Cdr Record if from the Primary PBX '2006-10-17 07:11:37', 'Admin', 'XXX, 'aa', 'from-internal', 'IAX2/[EMAIL PROTECTED]', 'Zap/1-1', 'ResetCDR', 'w', 10, 0, 'BUSY', 3, '', '', '' this is the CDR record from the secondsry for the same call '2006-10-17 13:31:57', 'Admin X', 'X', 'aa', 'from-internal', 'SIP/401-8f0c', 'IAX2/TRUNK1-2', 'Dial', 'IAX2/TRUNK1/aaa|120', 15, 15, 'ANSWERED', 3, '4147', '', '' in this setup, the caller dropped the call after allowing it to ring for 15 seconds On 10/17/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: Well I am using APM on the two boxes i have modified the srripts extensievely and i am sure that there is no Awnser befor a dial when Dialing through the PBX trunks On 10/17/06, Steve Davies [EMAIL PROTECTED] wrote: On 10/17/06, Dumpolid Exeplish [EMAIL PROTECTED] wrote: Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls ( i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my call disposition colume. I have also noticed that both the call duration and the billsec are always equal. here is my setup below PSTN va E1 (Primary Asterisk) =Sip and IAX trunks (Secondary PBX) Clients are connected to the Secondary PBX. this pbx handles registration of all clents. The billing irregularities happen on the Secondary PBX. When a call is maked from the Secondary and it is routed across the trunks, call disposition always registeres 'AWNSERED', unless the Primary PBX sends back a busy signal. the duration and billsecs are always equla. this means that the user gets billed for ring time, and calls disconnected from the Secondary PBXCould you provide a snippet of the dialplan used on each of theprimary and secondary boxes to complete a call?For example, is the primary executing an Answer() before it does the onward Dial() on behalf of the secondary?Cheers,Steve___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why is this happening?
On Monday 16 October 2006 17:26, Matt wrote: My question is... if both machiens are set to listen on 4569, will the fact that that router is mangeling the port cause any issues? Nope. The router should have udp/4569 forwarded to the correct server on the inside, so that when *it* gets a request on that port it sends it off to the correct server/port. From my sip list peers output: 2206/2206 216.xxx.yyy.96D N 15061Unmonitored 2201/2201 216.xxx.yyy.96D N 5060 Unmonitored 2200/2200 216.xxx.yyy.96D N 15060Unmonitored As you can see, the first one that registered (2201) didn't have its source port mangled. However, 2200 and 2206, both behind the same NATing router, had their source port mangled. Asterisk works just fine like this, and IAX2 is even better since the audio path is multiplexed on the same port. Olle's awesome RTP patches which get symmetric RTP into Asterisk (part of Asterisk for quite some time now) make SIP and NAT almost stupidly easy. I've got installations with a dozen IP501s behind a totally-standard (and probably factory default configuration!) WRT54G router with *no* issues. -A. ___ --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] One way audio on chan_gtalk
Hi! I´m trying with 1.4b2, chan_jabber and chan_gtalk. Jabber client register fine on talk.google.com, and when I start a call from gtalk to asterisk, I can see the incoming call and I see that asterisk play prompts (ie: demo and thank-you), but i can´t hear audio. If I redirect incoming call to a sip client, at sip I can hear but I can't in google talk. Asterisk is at public no firewalled network. Google Talk are behind a nat. Could anybody help me? Thanks in advance, Gustavo Hernandez ___ --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] One way audio on chan_gtalk
From our experience, chan_jabber doesnt work behind nat. We tried to patch it (in a similar way as nat=yes in chan_sip) but quickly bumped into other problems. (problems explained on mantis). Zoa. Gustavo Hernandez Baratta wrote: Hi! I´m trying with 1.4b2, chan_jabber and chan_gtalk. Jabber client register fine on talk.google.com, and when I start a call from gtalk to asterisk, I can see the incoming call and I see that asterisk play prompts (ie: demo and thank-you), but i can´t hear audio. If I redirect incoming call to a sip client, at sip I can hear but I can't in google talk. Asterisk is at public no firewalled network. Google Talk are behind a nat. Could anybody help me? Thanks in advance, Gustavo Hernandez ___ --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] Remote UNIX connection, Remote UNIX disconnected displayed every second
On Tue, Oct 17, 2006 at 10:59:33AM +0200, Giorgio Incantalupo wrote: Hi Brian, yes, I have more copies of safe_asterisk running, I know this is the underline problem but I do not how to solve it because I do not know how to reproduce it. I'm still looking the safe_asterisk for some strange but found nothing till now. Have you got the same problem? Why is it happening? Which brings up the obvious question: why do you need safe_asterisk in the first place? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] what hardware and is it possible
Imagine i want to create application like SMS Alert, however it's a call alert when something happened, for example server is crashed, i want to call 100 of my staff (administrator, manager, and others) using asterix, when they pick up their phone, my asterix will play an audio file Is it possible? what is the correct hardware for this? ___ --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] Dial - i parametar
There was patch for 1.0.x version of Asterisk that is quite useful. Is there patch for 1.2.x version and will this i parameter be in 1.4.x version of Asterisk? Have a nice day! -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr ___ --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] Why is this happening?
On Monday 16 October 2006 17:25, Time Bandit wrote: Same thing happens with a webserver. It listen for connections on port 80 (default port) and when a connection comes in, it is handed to another free port on the server so the main server can continue You've got a very poor grasp on how things work. Please don't pretend to know what you're talking about. # netstat -apn | grep :80 tcp0 0 0.0.0.0:80 0.0.0.0:* LISTEN 782/httpd tcp0 0 204.xxx.yyy.188:8080.xxx.yyy.167:58620 ESTABLISHED 814/httpd tcp0 0 204.xxx.yyy.188:8062.xxx.yyy.15:55384 ESTABLISHED 1068/httpd tcp0 0 204.xxx.yyy.188:80165.xxx.yyy.230:4392 ESTABLISHED 1084/httpd tcp0 0 204.xxx.yyy.188:8065.xxx.yyy.111:6982 TIME_WAIT - tcp0 0 204.xxx.yyy.188:80200.xxx.yyy.43:8198 ESTABLISHED 817/httpd tcp0 0 204.xxx.yyy.188:80165.xxx.yyy.230:4304 ESTABLISHED 815/httpd As you can see, I am *still* listening on port 80 and have numerous connections from different systems, even numerous connections from the same system. -A. ___ --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] Please help me!!
Hi to all, I've a segmentation fault while using asterisk relatime conf with mysql db. I've cretate sip_buddies and extensions tables into db and edit res_mysql.conf, extconf.conf without any issues. So when I start asterisk and my phone try to register using sip user configured in my db, asterisk stops with Segmentation fault error. Follow post gdb backtrace 0 0x400337c0 in pthread_setcanceltype () from /lib/libpthread.so.0 #21 0x0805d8de in ast_load_realtime (family=0x666d7464 Address 0x666d7464 out of bounds) at config.c:994 #22 0x4047cdad in realtime_peer (peername=0xbe7f8891 101, sin=0x730) at chan_sip.c:1696 #23 0x4046cf67 in find_peer (peer=0xbe7f8891 101, sin=0x0, realtime=1) at chan_sip.c:1776 #24 0x40485dfd in register_verify (p=0x81944d8, sin=0xbe7fe79c, req=0xbe7fe7ac, uri=0xbe7fe9cd sip:192.168.1.2, ignore=1718449252) at chan_sip.c:6514 #25 0x404839b7 in handle_request (p=0x81944d8, req=0xbe7fe7ac, sin=0xbe7fe79c, recount=0x666d7464, nounlock=0x666d7464) at chan_sip.c:11083 #26 0x4048150d in sipsock_read (id=0x813ed80, fd=15, events=1, ignore=0x0) at chan_sip.c:11377 #27 0x080558dd in ast_io_wait (ioc=0x8162320, howlong=1718449252) at io.c:284 #28 0x404776a9 in do_monitor (data=0x0) at chan_sip.c:11536 #29 0x40034cc4 in pthread_detach () from /lib/libpthread.so.0 #30 0x40201037 in clone () from /lib/libc.so.6 Please anyone can help me with a suggestion? I can also post asterisk debug trace anyway. Thaks for all, flavio -- * (o ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ___ --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] what hardware and is it possible
On Tue, 2006-10-17 at 20:38 +0700, Ady Wicaksono wrote: Imagine i want to create application like SMS Alert, however it's a call alert when something happened, for example server is crashed, i want to call 100 of my staff (administrator, manager, and others) using asterix, when they pick up their phone, my asterix will play an audio file Is it possible? did you look at http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out ? what is the correct hardware for this? if you want to do just that, then pretty much any odd box that runs asterisk will do. There is very little required in terms of cpupower or memory. I'd probably choose an embedded system like soekris for that sort of stuff. Conrad ___ --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] duplicate ghost calls with long duration
Hello everybody, I am running 1.2.10-BRIstuffed-0.3.0-PRE-1s with florz-patches on Linux 2.6.16 with 4 HFC-Cards in TE-mode connected to 4 PtP ISDN-Anlagenanschluesse. There are about 40 SIP-clients connected (mostly Sipura/Linksys PAP2, and some SNOMs and softclients) to this server. Everything works fine, except that my CDR reports some very long _concurrent_ calls from one sip client to (an expensive) pstn destination. The CDR from my telco tells the same! First case: Sep. 05 2006 11:46:40 20 secs call from X to 0900xxx (valid call, micropayment) Sep. 05 2006 11:52:18 3573 secs call from same X to 0900xxx () Sep. 05 2006 11:53:24 3466 secs call from same X to 0900xxx () X is a Sipura-connected hardphone. The guy swears he has only done the first call. Strangely the others are concurrent and to the same micropayment number, which itself disconnects callers after the micropayment value has been reached! Second case: Sep. 21 2006 14:30:22 35981(!) secs call from Y to 01805xxx () Sep. 21 2006 14:31:00 1823 secs call from Z to 01805xxx (valid, conference provider call) Sep. 21 2006 14:36:09 35634(!) secs call from Y to 01805xxx () You see the valid call is made from a different caller, Z instead of Y. Y is also a sipura phone as well as Z. Y did never call this number itself. Since my telco's (Deutsche Telekom) CDR tells me the same, this problem seems to be an asterisk internal one. Any hints? Regards, Bjoern ___ --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] what hardware and is it possible
I have such a setup here myself, although not for 100 people. Any recent server will do, but make sure you don't call 100 people the same second, spread them a little over time. Google for .call files Zoa. Ady Wicaksono wrote: Imagine i want to create application like SMS Alert, however it's a call alert when something happened, for example server is crashed, i want to call 100 of my staff (administrator, manager, and others) using asterix, when they pick up their phone, my asterix will play an audio file Is it possible? what is the correct hardware for this? ___ --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] Help with Dialplan Rules Please!
This was posted at The Asterisk Blog ForumsClick here for the original post. I need someone to explain how the dialplan rules work? I'm having a hard time getting it. I know that to dial out you need a 9 and to ignore that 9 once your out... requires a switch of sorts that tells asterisk to ignore the first digit on the left. In freePBX it's this: 9|NXX For Long distance it is 9|1NXXNXX Here is my problem using Free PBX: I want to be able to dial long distance and local at will while using free PBX to set it up. Right now we have 1 line for testing purposes and soon to be expanded into 2. When the rules are arranged like this in FreePBX 9|1NXXNXX 9|NXX the long distance portion works but the local one does not. When its arranged like this 9|NXX 9|1NXXNXX They both work! But the above is only done when it's hard coded into the configuration file (additional_extensions.conf) and free PBX always puts it in this order... wether I like it or not. 9|1NXXNXX 9|NXX And causes problems in the configuration file when and I change the settings. This isn't going to help me much! Im just a tad bit confused. A Little help here?-- www.AsteriskBlog.comYour home for easy to learn Asterisk 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
[asterisk-users] Setting the H323 Callerid sent by asterisk (using chan_h323)
Successfully hooked up a Cisco CallManager to a local * installation via H323. Things are working fine when dialling from CCM to *, but I'm seeing a callerid of 'root' coming up on the CCM Phone when dialling into *. Is there any way of setting this to be something else? I'm guessing the name is coming from the ID that * is running as. I'm using the chan_h323 driver and have searched for a way of setting it in the h323.conf file without luck. Any ideas?? TIA ___ --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] Call Forwarding Using Asterisk
Thank you Ram, Can you give me some example, how can I do that. -Jk ram wrote: Hi its possible you need mention in the config Ram On 10/17/06, *jk* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Can I do this with Asterisk, Call comes to Asterisk Server (Master), Then master just forwards calls to other slave asterisk servers one by one. Like this Master forward 1st call to Slave 1, Second call to Slave 2, Third call to slave 1 Fourth call to slave 2. Is it possible? I will appreciate if some one help me with this. Thank you, -Jai ___ --Bandwidth and Colocation provided by Easynews.com http://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] Remote UNIX connection, Remote UNIX disconnected displayed every second
Hi Tzafrir, 1) it is provided with asterisk 2) it is called by asterisk init script by default (and the asterisk init script is provided by default too) so I think/hope it is good enough 3) actually I have got nothing else and time is very short 4) I need something to restart asterisk in case of failure 5) many people on internet say to use it I used to launch safe_asterisk directly...maybe this was my error...now I use the init script inside contrib/init.d...maybe I'll be more lucky. Giorgio Incantalupo Tzafrir Cohen wrote: On Tue, Oct 17, 2006 at 10:59:33AM +0200, Giorgio Incantalupo wrote: Hi Brian, yes, I have more copies of safe_asterisk running, I know this is the underline problem but I do not how to solve it because I do not know how to reproduce it. I'm still looking the safe_asterisk for some strange but found nothing till now. Have you got the same problem? Why is it happening? Which brings up the obvious question: why do you need safe_asterisk in the first place? ___ --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] what hardware and is it possible
Hi Ady - Imagine i want to create application like SMS Alert, however it's a call alert when something happened, for example server is crashed, i want to call 100 of my staff (administrator, manager, and others) using asterix, when they pick up their phone, my asterix will play an audio file Is it possible? Yes. For more information: http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out what is the correct hardware for this? Any modern Linux, BSD (including OS X), or Solaris compatible computer to run asterisk. If you are using an ITSP (VoIP provider) you don't need any other hardware than your network card. If you have a PSTN phone connection, at the very least you'll need a card (like Digium, Sangoma, Rhino, etc), or an external gateway (like linksys, dlink, mediatrix, etc). - Noah ___ --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] One way audio on chan_gtalk
Hi Zoa: Thanks for your answer. Let me explain: Asterisk are not behind a NAT, google talk user are. Do you think that is the same problem? Thanks a lot! gus At 10:28 a.m. 17/10/2006, you wrote: From our experience, chan_jabber doesnt work behind nat. We tried to patch it (in a similar way as nat=yes in chan_sip) but quickly bumped into other problems. (problems explained on mantis). Zoa. Gustavo Hernandez Baratta wrote: Hi! I´m trying with 1.4b2, chan_jabber and chan_gtalk. Jabber client register fine on talk.google.com, and when I start a call from gtalk to asterisk, I can see the incoming call and I see that asterisk play prompts (ie: demo and thank-you), but i can´t hear audio. If I redirect incoming call to a sip client, at sip I can hear but I can't in google talk. Asterisk is at public no firewalled network. Google Talk are behind a nat. Could anybody help me? Thanks in advance, Gustavo Hernandez ___ --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 crashes with ealtime mysql configuration [was: Re: [asterisk-users] Please help me!!]
On Tue, Oct 17, 2006 at 03:42:55PM +0200, flavio wrote: Hi to all, I've a segmentation fault while using asterisk relatime conf with mysql db. I've cretate sip_buddies and extensions tables into db and edit res_mysql.conf, extconf.conf without any issues. So when I start asterisk and my phone try to register using sip user configured in my db, asterisk stops with Segmentation fault error. Follow post gdb backtrace 0 0x400337c0 in pthread_setcanceltype () from /lib/libpthread.so.0 #21 0x0805d8de in ast_load_realtime (family=0x666d7464 Address 0x666d7464 out of bounds) at config.c:994 #22 0x4047cdad in realtime_peer (peername=0xbe7f8891 101, sin=0x730) at chan_sip.c:1696 #23 0x4046cf67 in find_peer (peer=0xbe7f8891 101, sin=0x0, realtime=1) at chan_sip.c:1776 #24 0x40485dfd in register_verify (p=0x81944d8, sin=0xbe7fe79c, req=0xbe7fe7ac, uri=0xbe7fe9cd sip:192.168.1.2, ignore=1718449252) at chan_sip.c:6514 #25 0x404839b7 in handle_request (p=0x81944d8, req=0xbe7fe7ac, sin=0xbe7fe79c, recount=0x666d7464, nounlock=0x666d7464) at chan_sip.c:11083 #26 0x4048150d in sipsock_read (id=0x813ed80, fd=15, events=1, ignore=0x0) at chan_sip.c:11377 #27 0x080558dd in ast_io_wait (ioc=0x8162320, howlong=1718449252) at io.c:284 #28 0x404776a9 in do_monitor (data=0x0) at chan_sip.c:11536 #29 0x40034cc4 in pthread_detach () from /lib/libpthread.so.0 #30 0x40201037 in clone () from /lib/libc.so.6 Please anyone can help me with a suggestion? Three: 1. Use a proper subject line. Something like Asterisk crashes with realtime mysql configuration 2. Don't post such messages on asterisk-dev . Cross-posting them is even worse. 3. Provide more information about the system: - Version of Asterisk - Operating system and special libraried used - Did this work before? - Is the crash reproducable? What is the minimal configuration required to reproduce it? -- Tzafrir Cohen sip:[EMAIL PROTECTED] icq#16849755 iax:[EMAIL PROTECTED] +972-50-7952406 jabber:[EMAIL PROTECTED] [EMAIL PROTECTED] http://www.xorcom.com ___ --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] what hardware and is it possible
Perhaps your Network Management system could email a text file to the Asterisk server, which could then use Festival to read the text file to each person it calls... Zoa wrote: I have such a setup here myself, although not for 100 people. Any recent server will do, but make sure you don't call 100 people the same second, spread them a little over time. Google for .call files Zoa. Ady Wicaksono wrote: Imagine i want to create application like SMS Alert, however it's a call alert when something happened, for example server is crashed, i want to call 100 of my staff (administrator, manager, and others) using asterix, when they pick up their phone, my asterix will play an audio file Is it possible? what is the correct hardware for this? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Why is this happening?
You've got a very poor grasp on how things work. Please don't pretend to know what you're talking about. # netstat -apn | grep :80 tcp0 0 0.0.0.0:80 0.0.0.0:* LISTEN 782/httpd tcp0 0 204.xxx.yyy.188:8080.xxx.yyy.167:58620 ESTABLISHED 814/httpd tcp0 0 204.xxx.yyy.188:8062.xxx.yyy.15:55384 ESTABLISHED 1068/httpd tcp0 0 204.xxx.yyy.188:80165.xxx.yyy.230:4392 ESTABLISHED 1084/httpd tcp0 0 204.xxx.yyy.188:8065.xxx.yyy.111:6982 TIME_WAIT - tcp0 0 204.xxx.yyy.188:80200.xxx.yyy.43:8198 ESTABLISHED 817/httpd tcp0 0 204.xxx.yyy.188:80165.xxx.yyy.230:4304 ESTABLISHED 815/httpd As you can see, I am *still* listening on port 80 and have numerous connections from different systems, even numerous connections from the same system. I am really sorry, I've read that explanation somewhere and it made sense. Now that I've been corrected, I won't make that same mistake again. Please excuse me. The one that never did a mistake, never did anything ___ --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] acami
Hi list, Im searching for a web configuration front-end for Asterisk, and found ACaMI: http://sourceforge.net/projects/acami/ Anyone here try it? any feedback will be great. Greg ___ --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] duplicate ghost calls with long duration
Bjoern Metzdorf wrote:
Everything works fine, except that my CDR reports some very long
_concurrent_ calls from one sip client to (an expensive) pstn
destination. The CDR from my telco tells the same!
Any hints?
I run into that from time to time for this business account we have
where channels were staying open for a long time so I made a script run
from cron to hang up any extension over X amount of time:
/usr/sbin/asterisk -rx show channels concise |awk -F : '($11 5400)
{print /usr/sbin/asterisk -rx \soft hangup $1 \} '|sh
This looks at any calls over 90 minutes then hangs it up. You can modify
it for your issue say something like:
/usr/sbin/asterisk -rx show channels concise |awk -F :
'/YOUR_X_SIPURA_NUMBER/'|awk -F : '($11 5400) {print
/usr/sbin/asterisk -rx \soft hangup $1 \} '|sh
Not practical though for saving money... If someone is on for say 1
minute and there is an issue with the channel not hanging up, 5399
minutes would still be billed.
--
J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net
The happiness of society is the end of government.
John Adams
smime.p7s
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RE: [asterisk-users] VoipSupply? [Semi-Urgent]
VoIPSupply operations are completely back online as of this morning. Grid power and carrier services have been restored to our area. My apologies to any customers who were inconvenienced in the last several days. We experienced an odd weather event with results quite similar to the aftermath of a hurricane. If anyone reading this has unresolved customer service issues which need immediate assistance, feel free to contact me directly and I will gladly intervene on your behalf. Regards, Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice direct- 716.250.3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - AspendoraSent: Monday, October 16, 2006 10:45 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] VoipSupply? [Semi-Urgent] Why don't you want to speak to the janitor at this point? Where doestech support come in? I think the janitor is a lot more knowledgeable than most tech support departments these days. Maybe we should think about asking for the janitor. ___ --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] Help with Dialplan Rules Please!
If the order is giving you problems, create two separate outbound routes, one for local calls and one for long distance. Make sure the local route is before the LD route, and it should work for you. Both outbound routes can use the same trunk without issue. AlexOn 10/17/06, Chris Ramsey [EMAIL PROTECTED] wrote: This was posted at The Asterisk Blog Forums Click here for the original post. I need someone to explain how the dialplan rules work? I'm having a hard time getting it. I know that to dial out you need a 9 and to ignore that 9 once your out... requires a switch of sorts that tells asterisk to ignore the first digit on the left. In freePBX it's this: 9|NXX For Long distance it is 9|1NXXNXX Here is my problem using Free PBX: I want to be able to dial long distance and local at will while using free PBX to set it up. Right now we have 1 line for testing purposes and soon to be expanded into 2. When the rules are arranged like this in FreePBX 9|1NXXNXX 9|NXX the long distance portion works but the local one does not. When its arranged like this 9|NXX 9|1NXXNXX They both work! But the above is only done when it's hard coded into the configuration file (additional_extensions.conf) and free PBX always puts it in this order... wether I like it or not. 9|1NXXNXX 9|NXX And causes problems in the configuration file when and I change the settings. This isn't going to help me much! Im just a tad bit confused. A Little help here?-- www.AsteriskBlog.comYour home for easy to learn Asterisk stuff. ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Alex Robar[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
Re: [asterisk-users] One way audio on chan_gtalk
Yes, its the same as what we tried. Gustavo Hernandez Baratta wrote: Hi Zoa: Thanks for your answer. Let me explain: Asterisk are not behind a NAT, google talk user are. Do you think that is the same problem? Thanks a lot! gus At 10:28 a.m. 17/10/2006, you wrote: From our experience, chan_jabber doesnt work behind nat. We tried to patch it (in a similar way as nat=yes in chan_sip) but quickly bumped into other problems. (problems explained on mantis). Zoa. Gustavo Hernandez Baratta wrote: Hi! I´m trying with 1.4b2, chan_jabber and chan_gtalk. Jabber client register fine on talk.google.com, and when I start a call from gtalk to asterisk, I can see the incoming call and I see that asterisk play prompts (ie: demo and thank-you), but i can´t hear audio. If I redirect incoming call to a sip client, at sip I can hear but I can't in google talk. Asterisk is at public no firewalled network. Google Talk are behind a nat. Could anybody help me? Thanks in advance, Gustavo Hernandez ___ --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 ___ --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] Why is this happening?
Matt wrote: On 10/16/06, Time Bandit [EMAIL PROTECTED] wrote: Why is it running on port 1207? because Asterisk is listening on port 4569 and when a connection comes in, it as handed to another port so it can continue listening on port 4569. Otherwise you would only be handling 1 connection at a time. Pretty basic networking stuff I think :c) Thanks for the answer, but I don't buy it. There are currently 0 calls up on that bridge, while another connection which has calls up on it is on Port 4569.. please try again. IAX2 is suppose to run on ONLY one port.. this is why it is so nice for use in firewall situations. The source port on the REMOTE side is 1207. It seems like Asterisk is many people's first introduction to networking. Asterisk 4569 - 1027 SIP Device ___ --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] Why is this happening?
Time Bandit wrote: Thanks for the answer, but I don't buy it. There are currently 0 calls up on that bridge, while another connection which has calls up on it is on Port 4569.. please try again. IAX2 is suppose to run on ONLY one port.. this is why it is so nice for use in firewall situations. It doesn't change a thing ! Same thing happens with a webserver. It listen for connections on port 80 (default port) and when a connection comes in, it is handed to another free port on the server so the main server can continue listening on port 80. Same thing with FTP, etc. All TCP servers that accept more than one connection This is totally and completely wrong. An IP connection is uniquely identified by the information of Source IP + Source Port AND Destination IP and Destination Port. In the case of you example the IAX2 registration came in from the source port on the far device of 1207. Connections don't just move between ports. When you do an iax2 show peers you are seeing the REMOTE IP address and the REMOTE port. It does not show anything about the local ports or local IP addresses. ___ --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] nat auto detect ?
Benjamin Jacob wrote: Hello ppl, This post is to do with the variables 'nat' or 'canreinvite' for sip entities. Idealy users, wont be static, they could be roaming all over the globe. So, setting someone as behind NAT, and disabling canreinvite, etc., restricts the roaming capabilities of a user. No. Almost all devices work fine with nat=yes, even if they are not behind NAT. ___ --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] Is 1.2.12.1 production ready
On 10/17/06, Mike Clark [EMAIL PROTECTED] wrote: We have several sites in this configuration with no nightly reboots. Allsites except one are problem free. One site still has dropped calls.None of the sites crashes and some of them have been up for a few weeks. Tom Vile wrote: fine for me here since it came out.We are running 15 extension all day long. On 10/16/06, *shadowym* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am getting ready to image a production system.Right now I am planning on using Centos 4.4, Asterisk 1.2.12.1 http://1.2.12.1, Freepbx 2.1.3.I will be using a Sangoma A200D card. I read of some people having problems with Asterisk 1.2.12.1 http://1.2.12.1 crashing.Is this across the board or is there anyone out there with no problems.If you have 24/7 uptime and no nightly reboot crons I would definitely appreciate hearingabout it. Cheers ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856Hi guys,I'm having a problem with chanspy.When I'm hearing the calls on third or forth change asterisk gives me: Asterisk ended with exit status 139 Asterisk exited on signal 11. And restart.I'm using postgres for CDR, asterisk 1.2.12.1, addons-1.2.4 and zaptel-1.2.If would could test it, it will be very nice.-- Ralph LiebessohnICQ: 74835911 Skype: liebessohn ___ --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] Locking phones at night...
I have a customer that wants to lock his phone when he goes home at night so no one else can use it. What would be the easiest way to do this? -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 signature.asc Description: This is a digitally signed message part ___ --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] Authenticate application
Plesae, doesthe 'authenticate' application awnser a channel when requesting for password ? ___ --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] sipXezphone
Can someone tell me where I can DL a windows binary for sipXezphone. Everything I find ultimately points me back to http://www.sipfoundry.org/sipXezPhone/ which is broken. Tx M -- Opportunity is missed by most people because it is dressed in overalls and looks like work. Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures. ___ --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] Why is this happening?
On Tuesday 17 October 2006 10:31, Time Bandit wrote: The one that never did a mistake, never did anything *amen* to that! -A. ___ --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] duplicate ghost calls with long duration
I run into that from time to time for this business account we have
where channels were staying open for a long time so I made a script run
from cron to hang up any extension over X amount of time:
/usr/sbin/asterisk -rx show channels concise |awk -F : '($11 5400)
{print /usr/sbin/asterisk -rx \soft hangup $1 \} '|sh
This looks at any calls over 90 minutes then hangs it up. You can modify
it for your issue say something like:
/usr/sbin/asterisk -rx show channels concise |awk -F :
'/YOUR_X_SIPURA_NUMBER/'|awk -F : '($11 5400) {print
/usr/sbin/asterisk -rx \soft hangup $1 \} '|sh
Not practical though for saving money... If someone is on for say 1
minute and there is an issue with the channel not hanging up, 5399
minutes would still be billed.
What version are you using?
I never had these issues with asterisk 1.0.x in 15 months. That leads me
to a problematic 1.2.x or to faulty bristuff-patches.
I will upgrade asterisk asap to latest 1.2.x and add an absolute timeout
to those destinations.
But: Are we the only ones experiencing this?
Regards,
Bjoern
___
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Re: [asterisk-users] Re: SIP stuck channel soft hangup?
14 okt 2006 kl. 22.15 skrev Benny Amorsen: MJ == Martin Joseph [EMAIL PROTECTED] writes: MJ I added the rtptimeout=60 to my general section in sip.conf, and MJ now when the e60 goes out of wifi range, 61 seconds later, my MJ channels are clear! Sweet. Does this work with canreinvite=yes? (I can't see how it could, but I'd like to be surprised) I would be *very* surprised if that worked!!! I guess we have to implement the sip timer extension to be able to solve that issue. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] VoipSupply? [Semi-Urgent]
Cory, You may wish to search the archives of this list (and more appropriately the commercial list). There seem to be a number of open support issues, lack of follow-through, and unprofessional behavior on the part of VoIPSupply support. It's always hard to separate fact from fiction on internet lists, but the number and nature of comments would certainly cause concern on the part of a potential customer. You now have contact info for dissatisfied customers - the best testament would be for them to post successful resolution to their issues (not an announcement from VoIPSupply that you're back). Good luck, MD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory AndrewsSent: Tuesday, October 17, 2006 10:40 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] VoipSupply? [Semi-Urgent] VoIPSupply operations are completely back online as of this morning. Grid power and carrier services have been restored to our area. My apologies to any customers who were inconvenienced in the last several days. We experienced an odd weather event with results quite similar to the aftermath of a hurricane. If anyone reading this has unresolved customer service issues which need immediate assistance, feel free to contact me directly and I will gladly intervene on your behalf. Regards, Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice direct- 716.250.3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - AspendoraSent: Monday, October 16, 2006 10:45 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] VoipSupply? [Semi-Urgent] Why don't you want to speak to the janitor at this point? Where doestech support come in? I think the janitor is a lot more knowledgeable than most tech support departments these days. Maybe we should think about asking for the janitor. ___ --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] Stopping putgoing calls after working hours
I am trying to find a way to stop people who use phones after business hours (a policy the company wants to implement), we have cisco 7940 and 7910 phones and sadly they don't have a phone lock password system (on these ciscos it locks config menu changes but not the calls but the cisco 7920 has this feauture). So I was wondering is there a way to make this happen in asterisk?? You need to better describe your objectives. If you really mean stop all calls (including emergency calls), that's easy. If you mean stop all calls that cleaning folks initiate (usually not employees), that just requires some extensions.conf changes to force the user to enter an access code before a call can be placed. (Just don't advertise that access code anyone that you don't want making calls. If your talking about a fairly major security issue (such as your users call forwarding their phones to the brother-in-law after normal hours, you'll probably need to disable call forwarding on the phone itself. If your talking about primarily managing expenses, use the CDR detail to generate a personalized report for each employee show this calls make between 5pm and 7am, and forward that report to each employee (and cc: the manager). That's usually enough to significantly cut those calls. If you don't have a policy relative to use of company assets (phones PC's) for personal use, you might put one together and reference that policy in the morning CDR detail report. (I'm sure at lease some of those calls are likely legitimate calls, so cutting all calls is not likely a workable solution. ___ --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] sending sip style messages in response
17 okt 2006 kl. 12.13 skrev Benjamin Jacob: Nope Mikael, * always seems to send a Decline, n also plays an unavailable file. I tried another scenario, A calls B, B rejects the call. In the tcpdump I see B sending a Forbidden to *, but * sends a Service Unavailable to A. hmm... not too sure, why this decision was made. Asterisk is a multiprotocol PBX, every error that arrives to the SIP channel is translated (like all other signalling) then sent to the core. The core sends it out again on the other channel that translates back to SIP. In this translation, which follows the details specs for ISDN to SIP translation, some granularity is lost. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * Asterisk Training http://edvina.net/training/ ___ --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] install MAGI
Hi, can somebody point me where to get MAGI patch to run AGI commands through asterisk manager. What i need to do is play a sound after originating a call on a zap channel. Or if its another way for doing this can somebody tell me. Is MAGI patch included in a recently asterisk version?Thanks.Alvaro. ___ --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] Why is this happening?
The moving to another port discussion is actually what happens with sockets. A socket listens on a designated port (ex: port 80) and when a connection is made to that socket, another socket begins to listen to port 80 for NEW connections. Sockets and Ports often gets confused with each other. -- Mitch Eric ManxPower Wieling wrote: Time Bandit wrote: Thanks for the answer, but I don't buy it. There are currently 0 calls up on that bridge, while another connection which has calls up on it is on Port 4569.. please try again. IAX2 is suppose to run on ONLY one port.. this is why it is so nice for use in firewall situations. It doesn't change a thing ! Same thing happens with a webserver. It listen for connections on port 80 (default port) and when a connection comes in, it is handed to another free port on the server so the main server can continue listening on port 80. Same thing with FTP, etc. All TCP servers that accept more than one connection This is totally and completely wrong. An IP connection is uniquely identified by the information of Source IP + Source Port AND Destination IP and Destination Port. In the case of you example the IAX2 registration came in from the source port on the far device of 1207. Connections don't just move between ports. When you do an iax2 show peers you are seeing the REMOTE IP address and the REMOTE port. It does not show anything about the local ports or local IP addresses. ___ --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] Sipura 901? Any experiences
I am interested in using the Sipura 901 as a home phone. Does anyone have experience with this unit? Positives, negatives, opinions welcomed. Thanks in advance, Marty ___ --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] Locking phones at night...
On Tue, 2006-10-17 at 10:25 -0500, Carlos Chavez wrote: I have a customer that wants to lock his phone when he goes home at night so no one else can use it. What would be the easiest way to do this? To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. ___ --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] Locking phones at night...
Use AstDB to keep State from Lock and Unlock and create a simple menu where user with a pincode may unlock his phone (also stored in Astdb, or other database) So 2 Menus one to lock the phone with a *XXX combination call, and another to unlock requesting pin code. Seems to me the simple way On 10/17/06, Carlos Chavez [EMAIL PROTECTED] wrote: I have a customer that wants to lock his phone when he goes home at night so no one else can use it. What would be the easiest way to do this? -- Carlos Chavez Prats Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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 -- Com os melhores cumprimentos, Marco Mouta ___ --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] Electric usage of a tdm400p
Hi people, When you use a TDM400p with 4FXS i know i need to connect a 12V connector to power the FXS lines. Im not good at electric stuff so I ask...If I have a 60W DC to DC adapter (80W peak) then, how much power will the TDM 400P consume? can it be powered? -- Erick Perez ___ --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] Gtalk on Asterisk 1.4
Hello All I have been trying to get Google Talk to work with Asterisk 1.4 SVN for a month now and am getting nowhere fast. Does anybody know how to load chan_gtalk.so and chan_jingle.so? I have tried to do it from with the modules.conf file as well as issuing from the CLI modules (load chan_gtalk.so) but when I do a module list it does not show up as loaded. Before installing asterisk I made sure that Gtalk and Jingle were selected in the make menuselect and I can see the modules in the /usr/lib/asterisk/modules directory. Any help would be most appreciated. ___ --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] lots of registrations, sip problem
Hello, I've got a problem with connection to my SIP provider. In general, everything works, but I get lots of these messages: Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's odd... Got a response on a call we dont know about. Cseq 42710 Cmd SIP/2.0 Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's odd... Got a response on a call we dont know about. Cseq 42686 Cmd SIP/2.0 Oct 17 19:10:06 DEBUG[29707]: chan_sip.c:11148 handle_request: That's odd... Got a response on a call we dont know about. Cseq 42686 Cmd SIP/2.0 That itself would not be a problem, but my provider is complaining about lots of faulty registrations. I ran ngrep on that traffic. This is ngrep output after fresh start of asterisk: U 200.100.100.123:5060 - 195.195.12.223:5060 REGISTER sip:sip1.provider.com SIP/2.0..Via: SIP/2.0/UDP 200.100.100.123:5060;branch=z9hG4bK6cb7aba7;rport..From: sip:[EMAIL PROTECTED] xtra.sk;tag=as04cc5746..To: sip:[EMAIL PROTECTED]..Call-ID: [EMAIL PROTECTED]: 102 REGIST ER..User-Agent: Asterisk PBX..Max-Forwards: 70..Expires: 1200..Contact: sip:[EMAIL PROTECTED]..Event: registration..Content-Lengt h: 0 U 195.195.12.223:5060 - 200.100.100.123:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 200.100.100.123:5060;branch=z9hG4bK6cb7aba7;received=200.100.100.123;rport=5060..From: sip:[EMAIL PROTECTED] 1.provider.sk;tag=as04cc5746..To: sip:[EMAIL PROTECTED]..Call-ID: [EMAIL PROTECTED]: 102 REGISTER..User-Agent: SoftSwitch v1.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:4212326 [EMAIL PROTECTED]..Content-Length: 0 U 195.195.12.223:5060 - 200.100.100.123:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 200.100.100.123:5060;branch=z9hG4bK6cb7aba7;received=200.100.100.123;rport=5060..From: sip:4212326601 [EMAIL PROTECTED];tag=as04cc5746..To: sip:[EMAIL PROTECTED];tag=as17802cd7..Call-ID: [EMAIL PROTECTED] 7.67.16.43..CSeq: 102 REGISTER..User-Agent: SoftSwitch v1.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.. Contact: sip:[EMAIL PROTECTED]..WWW-Authenticate: Digest realm=provider, nonce=70495064..Content-Length: 0 U 200.100.100.123:5060 - 195.195.12.223:5060 REGISTER sip:sip1.provider.com SIP/2.0..Via: SIP/2.0/UDP 200.100.100.123:5060;branch=z9hG4bK1362d770;rport..From: sip:[EMAIL PROTECTED] xtra.sk;tag=as51ddb092..To: sip:[EMAIL PROTECTED]..Call-ID: [EMAIL PROTECTED]: 103 REGIST ER..User-Agent: Asterisk PBX..Max-Forwards: 70..Authorization: Digest username=4221917293125, realm=provider, algorithm=MD5, uri=sip:s ip1.provider.sk, nonce=70495064, response=1c6b0193890a8e1c9c6b3a9287d1d30b, opaque=..Expires: 1200..Contact: sip:[EMAIL PROTECTED] 67.16.43..Event: registration..Content-Length: 0 U 195.195.12.223:5060 - 200.100.100.123:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 200.100.100.123:5060;branch=z9hG4bK1362d770;received=200.100.100.123;rport=5060..From: sip:[EMAIL PROTECTED] 1.provider.sk;tag=as51ddb092..To: sip:[EMAIL PROTECTED]..Call-ID: [EMAIL PROTECTED]: 103 REGISTER..User-Agent: SoftSwitch v1.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:4212326 [EMAIL PROTECTED]..Content-Length: 0 U 195.195.12.223:5060 - 200.100.100.123:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 200.100.100.123:5060;branch=z9hG4bK1362d770;received=200.100.100.123;rport=5060..From: sip:[EMAIL PROTECTED] sprovider.sk;tag=as51ddb092..To: sip:[EMAIL PROTECTED];tag=as17802cd7..Call-ID: [EMAIL PROTECTED] ..CSeq: 103 REGISTER..User-Agent: SoftSwitch v1.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Expires: 1 200..Contact: sip:[EMAIL PROTECTED];expires=1200..Date: Tue, 17 Oct 2006 16:38:19 GMT..Content-Length: 0 (**) U 200.100.100.123:5060 - 195.195.12.223:5060 REGISTER sip:my-provider-link SIP/2.0..Via: SIP/2.0/UDP 200.100.100.123:5060;branch=z9hG4bK102b43da;rport..From: sip:[EMAIL PROTECTED] sprovider.sk;tag=as6550d352..To: sip:[EMAIL PROTECTED]..Call-ID: [EMAIL PROTECTED]: 42620 R EGISTER..User-Agent: Asterisk PBX..Max-Forwards: 70..Authorization: Digest username=4221917293125, realm=provider, algorithm=MD5, uri= sip:my-provider-link, nonce=58c9238d, response=9d6439e3894f6a7e70fd09d683df073e, opaque=..Expires: 120..Contact: sip:4212326601 [EMAIL PROTECTED]..Event: registration..Content-Length: 0 U 195.195.12.223:5060 - 200.100.100.123:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 200.100.100.123:5060;branch=z9hG4bK102b43da;received=200.100.100.123;rport=5060..From: sip:[EMAIL PROTECTED] 1.provider.sk;tag=as6550d352..To: sip:[EMAIL PROTECTED]..Call-ID: [EMAIL PROTECTED]: 426 20 REGISTER..User-Agent: SoftSwitch v1.0..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Contact: sip:42123 [EMAIL PROTECTED]..Content-Length: 0 U 195.195.12.223:5060 - 200.100.100.123:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP
RE: [asterisk-users] VoipSupply? [Semi-Urgent]
My recollection is that VOIP supply is in upstate NY. If that's the case then they really did have a weather event that knocked out power to hundreds of thousands of people (and businesses). Such is the risk of the internet. Weather elsewhere may not be as nice and may delay your shipment or response. Brian GreulTexas Shirt CompanySolutions To Promote Youwww.txshirts.com713-802-0369 / 713-861-6261 (fax)ASI/343253 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michelle DupuisSent: Tuesday, October 17, 2006 11:07 AMTo: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] VoipSupply? [Semi-Urgent] Cory, You may wish to search the archives of this list (and more appropriately the commercial list). There seem to be a number of open support issues, lack of follow-through, and unprofessional behavior on the part of VoIPSupply support. It's always hard to separate fact from fiction on internet lists, but the number and nature of comments would certainly cause concern on the part of a potential customer. You now have contact info for dissatisfied customers - the best testament would be for them to post successful resolution to their issues (not an announcement from VoIPSupply that you're back). Good luck, MD From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory AndrewsSent: Tuesday, October 17, 2006 10:40 AMTo: 'Asterisk Users Mailing List - Non-Commercial Discussion'Subject: RE: [asterisk-users] VoipSupply? [Semi-Urgent] VoIPSupply operations are completely back online as of this morning. Grid power and carrier services have been restored to our area. My apologies to any customers who were inconvenienced in the last several days. We experienced an odd weather event with results quite similar to the aftermath of a hurricane. If anyone reading this has unresolved customer service issues which need immediate assistance, feel free to contact me directly and I will gladly intervene on your behalf. Regards, Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice direct- 716.250.3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Lacy Moore - AspendoraSent: Monday, October 16, 2006 10:45 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [asterisk-users] VoipSupply? [Semi-Urgent] Why don't you want to speak to the janitor at this point? Where doestech support come in? I think the janitor is a lot more knowledgeable than most tech support departments these days. Maybe we should think about asking for the janitor. ___ --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] duplicate ghost calls with long duration
Bjoern Metzdorf wrote:
I run into that from time to time for this business account we have
where channels were staying open for a long time so I made a script
run from cron to hang up any extension over X amount of time:
/usr/sbin/asterisk -rx show channels concise |awk -F : '($11
5400) {print /usr/sbin/asterisk -rx \soft hangup $1 \} '|sh
This looks at any calls over 90 minutes then hangs it up. You can
modify it for your issue say something like:
/usr/sbin/asterisk -rx show channels concise |awk -F :
'/YOUR_X_SIPURA_NUMBER/'|awk -F : '($11 5400) {print
/usr/sbin/asterisk -rx \soft hangup $1 \} '|sh
Not practical though for saving money... If someone is on for say 1
minute and there is an issue with the channel not hanging up, 5399
minutes would still be billed.
What version are you using?
I never had these issues with asterisk 1.0.x in 15 months. That leads
me to a problematic 1.2.x or to faulty bristuff-patches.
I will upgrade asterisk asap to latest 1.2.x and add an absolute
timeout to those destinations.
But: Are we the only ones experiencing this?
Regards,
Bjoern
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1.2.9 in this instance.
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[asterisk-users] considering purchasing a t1 card, any recommendations?
I have been looking at the rhino r1t1, and digium single t1, and the sangoma, but from what i read they all sound like good products. Anyone have anything bad about any one of them? I am leaning towards the sangoma as it seems to have a better following here on the lists. Any advice is appreciated.Thanks ___ --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] Locking phones at night...
As mentioned recently on the list in other posts, don't forget to allow emergency calls through no matter what unless you have tremendous lawyers Conrad Wood wrote: On Tue, 2006-10-17 at 10:25 -0500, Carlos Chavez wrote: I have a customer that wants to lock his phone when he goes home at night so no one else can use it. What would be the easiest way to do this? To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. ___ --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 !DSPAM:500,45351a07140271744649862! -- Mojo [EMAIL PROTECTED] Office Manager, Horan Company, LLC (907) 747- x112 ___ --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] IVR problem
All, I'm not able to play background files since this morning. I'm seeing this error message in the logs: [Oct 17 10:23:56] WARNING[4572] file.c: File custom/asterisk-prospectus_IVR-main-day does not exist in any format [Oct 17 10:23:56] WARNING[4572] file.c: Unable to open custom/asterisk-prospectus_IVR-main-day (format 0x4 (ulaw)): Permission denied [Oct 17 10:23:56] WARNING[4572] pbx.c: ast_streamfile failed on IAX2/teliax-2 for custom/asterisk-prospectus_IVR-main-day I know the file is there was working last week. I did update some files on the server over the weekend. I built Asterisk from SVN-trunk-r44731. Any help? Thanks, Jack Morgan ___ --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] Is 1.2.12.1 production ready
Just in case it helps anyone: We had 1.2.12.1 crashing on us on a daily basis, and sometimes several times a day. I found that by disabling all qualify lines in iax.conf and sip.conf the problem went away. Faris. ___ --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 (meetme) and SMP/HT OK? Lots 'o crashes
4 more days, 31 more crashes, no clues. On Mon, 16 Oct 2006, Steve Edwards wrote: More info... All calls come in from a Tekelec-7000/r4.0. The box has 2 te410p's left over from when calls came in from PRI. They were left in for a timing source since I don't have physical access. On Fri, 13 Oct 2006, Steve Edwards wrote: In the past, there have been reports of problems with Asterisk with multiple processors and/or HyperThreading. I'm having a [EMAIL PROTECTED] of a problem with an HPDL380 with 2 3.4gHz Xeon processors, 2 gb RAM -- if I got 24 hours I'd think I had died and gone to heaven :) Am I missing something obvious like Asterisk is single CPU, single core? I can't access the ILO so I can't just try it. I'm currently running Asterisk SVN-branch-1.2-r43977, but Asterisk has never been stable, regardless of the version, release or SVN. I have submitted a bug report, but it's been over 2 months and nobody seems interested in fixing a problem that has crashed 75 times (yes, seventy-five times) in the last 10 days! The vast majority of crashes are in meetme. The bt's look like this: #0 0x005e67a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #0 0x005e67a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 #1 0x006267a5 in raise () from /lib/tls/libc.so.6 #2 0x00628209 in abort () from /lib/tls/libc.so.6 #3 0x0065a71a in __libc_message () from /lib/tls/libc.so.6 #4 0x00660fbf in _int_free () from /lib/tls/libc.so.6 #5 0x0066133a in free () from /lib/tls/libc.so.6 #6 0x080615f3 in ast_channel_free (chan=0xb7904e00) at channel.c:959 #7 0x08062bd7 in ast_hangup (chan=0xb7904e00) at channel.c:1392 #8 0x001aa4fb in conf_free (conf=0xb7901d98) at app_meetme.c:789 #9 0x001acfa3 in conf_run (chan=0x96e94a0, conf=0xb7901d98, confflags=4224, optargs=0xb7ddcd4c) at app_meetme.c:1607 #10 0x001aeb26 in conf_exec (chan=0x96e94a0, data=0xb7de1070) at app_meetme.c:2031 #11 0x08083d43 in pbx_exec (c=0x96e94a0, app=0x9587840, data=0xb7de1070, newstack=1) at pbx.c:553 Any clues leading to the arrest and conviction of this bug will earn you a case of Sierra Nevada at the next west coast Astricon :) Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] Unique ID
Hello guys, We're currently working on asterisk trying to create our own SIP phone, because we need special features. But dunno maybe there's other people who already done that before. Basically, we are a inbound call center. We have serveral customers with different phone numbers, which are redirected to us. When we receive a call coming on a specific phone number, the call gets identified with the number and there's a greeting associated and displayed on the agent soft phone(this technology is still using regular phone with a special computer device). But here's the challenge that we currently face: 1. We need to have the info for the hold time (from agent) and hold time(before the call is actually answered). We currently offer a different pricing for the hold time by an agent than from the other one hold time. 2. We're currently trying to identify calls by unique id for billing, I've found about that the variable $UNIQUEID which I could use, and there's also the cdr table that I can create, but it would be nice to have both in the cdr table ? That way I could probably create a second table in the asterisk db, and store our hold time, sent from the softphone. Anyway, does all that ring a bell to someone ? Something that was already done ? Let me know if I'm unclear. Thanks, ___ --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] IVR problem
On Tuesday 17 October 2006 11:12, Jack Morgan wrote: All, I'm not able to play background files since this morning. I'm seeing this error message in the logs: [Oct 17 10:23:56] WARNING[4572] file.c: File custom/asterisk-prospectus_IVR-main-day does not exist in any format [Oct 17 10:23:56] WARNING[4572] file.c: Unable to open custom/asterisk-prospectus_IVR-main-day (format 0x4 (ulaw)): Permission denied [Oct 17 10:23:56] WARNING[4572] pbx.c: ast_streamfile failed on IAX2/teliax-2 for custom/asterisk-prospectus_IVR-main-day I know the file is there was working last week. I did update some files on the server over the weekend. I built Asterisk from SVN-trunk-r44731. Any help? Nevermind. Looks like it was a local user permission problem just like the error message indicated. Thanks, Jack Morgan ___ --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] Locking phones at night...
On Wednesday 18 October 2006 05:47, Conrad Wood wrote: To do something similar, I created a dialplan extension that - if dialled - creates a file on the server. If dialled again, it removes the file again. Then, in the context of the phone I check for existence of that file and if it exists I play a busy signal and hangup. (Of course, unless the extension to re-enable it is dialled ;) ). Additionally, I ask the user for a password to lock/unlock it. This is a good use for the AstDB hads -- http://nicegear.co.nz New Zealand's VoIP supplier ___ --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] Is 1.2.12.1 production ready
Are you suggesting that periodic reboots are only wise if you're running 1.2.12.1, or does that go for any asterisk installation? - .Dustin On Oct 16, 2006, at 10:54 PM, Mike Lynchfield wrote: reboots are wise On 10/16/06, Tom Vile [EMAIL PROTECTED] wrote: fine for me here since it came out. We are running 15 extension all day long. On 10/16/06, shadowym [EMAIL PROTECTED] wrote: I am getting ready to image a production system. Right now I am planning on using Centos 4.4, Asterisk 1.2.12.1, Freepbx 2.1.3. I will be using a Sangoma A200D card. I read of some people having problems with Asterisk 1.2.12.1 crashing. Is this across the board or is there anyone out there with no problems. If you have 24/7 uptime and no nightly reboot crons I would definitely appreciate hearing about it. Cheers ___ --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 -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --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 -- Mike Sales Manager http://www.theclubvoip.com Making it happen 1.877.807.VOIP (8647) ___ --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] considering purchasing a t1 card, any recommendations?
The Sangoma certainly has better support, a longer warranty, and works with all modern motherboards according to the manufacturer I do know first hand that Sangoma support doesn't say try another motherboard I just got an A200 working remotely. Though their installation instructions leave a little to be desired, You won't regret using either the Sangoma single port T1 card or the A200 with 2 FXO/2FXS I can't speak to other configurations John Novack Greg Kennedy wrote: I have been looking at the rhino r1t1, and digium single t1, and the sangoma, but from what i read they all sound like good products. Anyone have anything bad about any one of them? I am leaning towards the sangoma as it seems to have a better following here on the lists. Any advice is appreciated. Thanks ___ --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] Is 1.2.12.1 production ready
I am getting ready to image a production system. Right now I am planning on using Centos 4.4, Asterisk 1.2.12.1, Freepbx 2.1.3. I will be using a Sangoma A200D card. I read of some people having problems with Asterisk 1.2.12.1 crashing. Is this across the board or is there anyone out there with no problems. If you have 24/7 uptime and no nightly reboot crons I would definitely appreciate hearing about it. I suspect that for every problem you hear about on the list there are probably 100 other happy asterisk administrators. Not to downplay legitimate issues, but many times, instabilities can easily be attributed to the OS, hardware or a million other things not caused by asterisk. I installed 1.2.12.1 for one of my clients about 1 week after it was released. No cron reboots, and no issues. The install is 6 different boxes in 6 different locations. Before that they were running 1.2.11, no cron reboots, no issues. Before that 1.2.10, no cron reboots, no issues. Before that, etc, etc. Last time I've seen any real problem that would cause asterisk to stop working was 1.2.2 which had a major bug. - Noah On 10/17/06, Faris Raouf [EMAIL PROTECTED] wrote: Just in case it helps anyone: We had 1.2.12.1 crashing on us on a daily basis, and sometimes several times a day. I found that by disabling all qualify lines in iax.conf and sip.conf the problem went away. Faris. ___ --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] Why is this happening?
On Tue, Oct 17, 2006 at 11:20:02AM -0500, Mitch Miller wrote: The moving to another port discussion is actually what happens with sockets. A socket listens on a designated port (ex: port 80) and when a connection is made to that socket, another socket begins to listen to port 80 for NEW connections. Actually, the original socket continues to listen on port 80 for new connections, whilst the accept() call creates a new socket for the accepted connection. From the accept(2) manpage: DESCRIPTION The accept() system call is used with connection-based socket types (SOCK_STREAM, SOCK_SEQPACKET and SOCK_RDM). It extracts the first conâ nection request on the queue of pending connections, creates a new conâ nected socket, and returns a new file descriptor referring to that socket. The newly created socket is not in the listening state. The original socket sockfd is unaffected by this call. Regards, Brian. ___ --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
