[asterisk-users] Asterisk console suddenly extremely verbose...
Hi there. I started the console today to reload the extensions.conf file ; only to be greeted with extremely verbose console. Seems related to the zaptel card: Example: Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 020 P/F: 1 0 bytes of data voip*CLI [ 00 01 01 2f ] voip*CLI Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 023 P/F: 1 0 bytes of data Handling message for SAPI/TEI=0/0 -- ACKing all packets from 22 to (but not including) 23 -- Since there was nothing left, stopping T200 counter -- Stopping T203 counter since we got an ACK -- Nothing left, starting T203 counter -- Got RR response to our frame -- Restarting T203 timer This is repeating every 10s or so... Any ideas what this message means and is there a way to prevent it from happening. No changes has been made on this asterisk box in years (running old 1.4.25 if it ain't broken version) Thanks in advance JY -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk console suddenly extremely verbose...
Hi On 16 December 2011 13:24, Richard Mudgett rmudg...@digium.com wrote: You have pri intense debug span x enabled. Disable with pri no debug span x. Thanks... I couldn't find any configuration file showing this ; but ran the command in the CLI... Seems to have done it. I really wonder how it could have been turned on ... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel - DAHDI: now echo
Hello I have upgraded our asterisk box from zaptel to dhadi two weeks ago... Since, there has been quite a significant amount of echo when making a call. Only for the local outgoing call, the person on the other side doesn't hear any echo. This is with a TE-110P ISDN PRI card .. I've pretty much took the original zaptel configuration and used it as-is with the dahdi one ; to no available.. Any help would be greatly appreciated. Here is what zaptel.conf and zapata.conf used to be: /etc/zaptel.conf: loadzone = au defaultzone=au #TE110P span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=16 /etc/asterisk/zapata.conf [channels] language=en usecallerid=yes hidecallerid=no callerid=asreceived restrictcid=no usecallingpres=yes ; ISDN Exchange Lines (Fractional E1 PRA10) switchtype=euroisdn signalling=pri_cpe immediate=no pridialplan=unknown prilocaldialplan=unknown overlapdial=yes echocancel=yes echocancelwhenbridged=yes echotraining=256 rxgain=1.0 txgain=8.0 context=incoming faxdetect=incoming group=1 channel=1-10 now for the dahdi configuration: /etc/dahdi/system.conf: loadzone = au defaultzone=au #TE110P span=1,1,0,ccs,hdb3,crc4 bchan=1-10 dchan=16 /etc/asterisk/chan_dahdi.conf [channels] language=en usecallerid=yes hidecallerid=no callerid=asreceived restrictcid=no usecallingpres=yes switchtype=euroisdn signalling=pri_cpe immediate=no pridialplan=unknown prilocaldialplan=unknown echocancel=yes echocancelwhenbridged=yes echotraining=256 rxgain=1.0 txgain=8.0 context=incoming faxdetect=incoming group=1 channel=1-10 --- From reading the various documentation, I was convinced that moving from zaptel to dahdi was almost just a matter of renaming the configuration file... Am I mistaken ? Thank you in advance for any help. Jean-Yves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel - DAHDI: now echo
Hi That was a fast answer, impressive ! 2009/8/18 Kevin P. Fleming kpflem...@digium.com: Did you read the upgrade documentation that comes with DAHDI, specifically from UPGRADE.txt: I did, but I guess I did not pay enough attention... * It is no longer possible to select a software echo canceler at compile time to build into dahdi.ko; all four included echo cancelers (MG2, KB1, SEC and SEC2) are built as loadable modules, and if the Digium HPEC binary object file has been placed into the proper directory the HPEC module will be built as well. Any or all of these modules can be loaded at the same time, and the echo canceler to be used on the system's channels can be configured using the dahdi_cfg tool from the dahdi-tools package. Note: It is *mandatory* to configure an echo canceler for the system's channels using dahdi_cfg unless the interface cards in use have echo canceler modules available and enabled. There is *no* default software echo canceler with DAHDI. So, knowing my card (a Digium TE-110P, which AFAIK doesn't have any hardware echo cancellation module)... Which software echo canceller should I be using ? Is see that there are particular software configuration available , but I haven't had a clue on what they are for, nor did I find documentation about it... I'm not building asterisk nor dahdi myself, but instead rely on packaged from ATrpms.conf Thank you Jean-Yves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zaptel - DAHDI: now echo
Hi 2009/8/18 Tzafrir Cohen tzafrir.co...@xorcom.com: Something is missing here... http://docs.tzafrir.org.il/dahdi-tools/#_echo_canceller_modules Thanks .. I added to /etc/dahdi/system.conf the following: echocanceller=mg2,1-10 However, I have no clue about the various echo canceller, between mg2, kb1, sec2, and sec which one will provide the best performance ? (knowing that I was happy with whatever zaptel was doing before) JY ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to prevent logging of some entries in CDR
Hi On Jan 21, 2008 11:05 PM, Jean-Yves Avenard [EMAIL PROTECTED] wrote: This works great. However in the CDR, than seeing one entry for each call, I see several entries in the CDR Worse, if I do something like: Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]) 40. 2008-01-21 13:59:34 Local/1... 04 MOB. 04 04DialSIP/ipp100SIP/ipp100.1 100 NO ANSWER 00:11 incoming-zap 41. 2008-01-21 13:59:34 SIP/ipp... 04 04 s NO ANSWER 00:11 42. 2008-01-21 13:59:34 SIP/ipp... 04 04 s NO ANSWER 00:11 43. 2008-01-21 13:59:33 Zap/7-1... 04 MOB. 04 04DialLocal/[EMAIL PROTECTED]|10|tr286 ANSWERED00:12 incoming-zap 44. 2008-01-21 13:49:39 Local/1... 04 MOB. 04 04DialSIP/ipp100SIP/ipp100.1 100 NO ANSWER 00:05 102 NO ANSWER 00:05 52. 2008-01-21 13:49:39 Local/1... 04 MOB. 04 04DialSIP/ipp119SIP/ipp119.1 119 NO ANSWER 00:05 No one else is seeing this issue ? Jean-Yves ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to prevent logging of some entries in CDR
Hi On Jan 25, 2008 4:58 AM, John Faubion [EMAIL PROTECTED] wrote: I have the same issue but I haven't put much effort into solving it yet. Too many other issues seem to get in the way. If you do, please post your results ! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to prevent logging of some entries in CDR
Hi
In order to add several phones to a single extension number, I have
replaced entries like:
exten = 100,1,Dial(SIP/sipphone100,20,Tr)
into:
exten = 100,1,Dial(Local/[EMAIL PROTECTED],20,Tr)
[phones]
exten = XXX,1,Dial(SIP.sipphone${EXTEN})
etc
This works great. However in the CDR, than seeing one entry for each
call, I see several entries in the CDR
Worse, if I do something like:
Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED])
If each Local entry calls two phones, I will have here 6 entries in
the CDR , one for each Local (here 3) and one for each phone, all with
the same date, example, here it's one incoming call, making a few
extensions ringing with one or two phones per extension
As shown below... You'll see that now one incoming call will create
over 30 entries in CDR...
When I just add one line doing Dial(SIP/ippxx SIP/ippxx .) which
would only create one entry.
Any ideas on how to prevent this? it's screwing up all my records
Thank you
Jean-Yves
CDR for one incoming call
40. 2008-01-21 13:59:34 Local/1... 04 MOB.
04 04DialSIP/ipp100SIP/ipp100.1 100
NO
ANSWER 00:11 incoming-zap
41. 2008-01-21 13:59:34 SIP/ipp... 04 04
s
NO ANSWER 00:11
42. 2008-01-21 13:59:34 SIP/ipp... 04 04
s
NO ANSWER 00:11
43. 2008-01-21 13:59:33 Zap/7-1... 04 MOB.
04
04DialLocal/[EMAIL PROTECTED]|10|tr 286
ANSWERED00:12
incoming-zap
44. 2008-01-21 13:49:39 Local/1... 04 MOB.
04
04DialSIP/ipp100SIP/ipp100.1 100 NO
ANSWER 00:05
incoming-zap
45. 2008-01-21 13:49:39 SIP/ipp... 04 04
s
NO ANSWER 00:05
46. 2008-01-21 13:49:39 Local/1... 04 MOB.
04
04DialSIP/ipp102SIP/ipp102.1 102 NO
ANSWER 00:05
incoming-zap
47. 2008-01-21 13:49:39 SIP/ipp... 04 04
s
NO ANSWER 00:05
48. 2008-01-21 13:49:39 Local/1... 04 04
100
NO ANSWER 00:05
49. 2008-01-21 13:49:39 Local/1... 04 MOB.
04
04DialSIP/ipp103SIP/ipp103.1 103 NO
ANSWER 00:05
incoming-zap
50. 2008-01-21 13:49:39 SIP/ipp... 04 04
s
NO ANSWER 00:05
51. 2008-01-21 13:49:39 Local/1... 04 04
102 NO ANSWER 00:05
52. 2008-01-21 13:49:39 Local/1... 04 MOB.
04
04DialSIP/ipp119SIP/ipp119.1 119 NO
ANSWER 00:05
incoming-zap
53. 2008-01-21 13:49:39 Local/1... 04 MOB.
04
04DialLocal/[EMAIL PROTECTED]|20|Ttr) 219
NO ANSWER 00:05
incoming-zap
54. 2008-01-21 13:49:39 SIP/ipp... 04 04
s
NO ANSWER 00:05
55. 2008-01-21 13:49:39 Local/1... 04 04
103
NO ANSWER 00:05
56. 2008-01-21 13:49:39 SIP/ipp... 04 04
s
NO ANSWER 00:05
57. 2008-01-21 13:49:39 Local/1... 04 04
119
NO ANSWER 00:05
58. 2008-01-21 13:49:39 Local/1... 04 04
119
NO ANSWER 00:05
59. 2008-01-21 13:49:39 Local/1... Dial
SIP/ipp111SIP/ipp111.1
111 FAILED 00:00
60. 2008-01-21 13:49:39 Local/1... 04 04
111
NO ANSWER 00:00
61. 2008-01-21 13:49:20 Zap/6-1... 04 MOB.
04
04DialLocal/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]
0 ANSWERED00:24
incoming-zap
62. 2008-01-21 13:48:38 Local/1... 04 MOB.
04
04DialSIP/ipp100SIP/ipp100.1 100 NO
ANSWER 00:10
incoming-zap
63. 2008-01-21 13:48:38 SIP/ipp... 04 04
s
NO ANSWER 00:10
64. 2008-01-21 13:48:38 SIP/ipp...
[asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately
Dear all I am trying to upgrade our asterisk from 1.2 to 1.4.x There is something that now fails to work, reading the various documentations, I can not explain why. Here is an extract of my extensions.conf [welcome] exten = 299,1,Answer ; Answer the line exten = 299,2,Set(TIMEOUT(digit)=5); Set Digit Timeout to 5 seconds exten = 299,3,Set(TIMEOUT(response)=10); Set Response Timeout to 10 seconds exten = 299,4,BackGround(welcome) ; Play Welcome to ... include = internal-sip exten = t,1,Goto(0,1) ; If they take too long, go back to default exten = i,1,Playback(invalid) ; That's not valid, try again exten = i,2,Goto(s,2) ; try again However, right after playing welcome, it will hangup ! this used to wait 10s then fall on the extension t This is what I see on the console: -- Executing [EMAIL PROTECTED]:1] Answer(Zap/5-1, ) in new stack -- Executing [EMAIL PROTECTED]:2] Goto(Zap/5-1, |s|1) in new stack -- Goto (welcome,s,1) -- Executing [EMAIL PROTECTED]:1] Set(Zap/5-1, TIMEOUT(digit)=5) in new stack -- Digit timeout set to 5 -- Executing [EMAIL PROTECTED]:2] Set(Zap/5-1, TIMEOUT(response)=10) in new stack -- Response timeout set to 10 -- Executing [EMAIL PROTECTED]:3] BackGround(Zap/5-1, hello) in new stack -- Zap/5-1 Playing 'transfer' (language 'en') == Auto fallthrough, channel 'Zap/5-1' status is 'UNKNOWN' -- Hungup 'Zap/5-1' Any ideas? Thank you in advance Jean-Yves ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately
Hi On 11/5/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote: Look at the section starting on line 100 in /path/to/src/asterisk-1.4.13/UPGRADE.txt You should have read this file before upgrading to 1.4. Excellent. Thank you! I've added a WaitExten() just after and now everything works fine. I didin't install asterisk 1.4 from the source code but from a RPM package. I also seem to be missing something here, not sure if it's a bug. in 1.4.13: even though in asterisk.conf I've set astvarlibdir = /data/asterisk/var/lib/asterisk it still goes and fetch sound files in /var/lib/asterisk/sounds Jean-Yves ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Issues after upgrading from 1.2 to 1.4: hangup immediately
Hi On 11/5/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: astdatadir ? What is the default location in asterisk? Why have this hen you have astvarlibdir ? Jean-Yves ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Question about outgoing callerid
Hi
I have an ISDN connection with 100 DIDs assigned to it...
What I'm trying to achieve is set the proper outgoing callerID while
showing the local caller's extension in the CDR.
There is a behaviour that I just can't explain.
the callerid field in sip.conf is set as :
callerid=Jean-Yves/E 300
the callerid in iax.conf is set a:
callerid=Jean-Yves/E 300
(just the same)
Prior to making the call using the zap interface, I do:
[macro-zaptel]
;ARG1=Number to call
; set default outgoing caller ID if FROMNUMBER is empty
exten = s,1,GotoIf($[${FROMNUMBER} = ]?2:4)
exten = s,2,Set(CALLERID(number)=03)
exten = s,3,Goto(s,5)
exten = s,4,Set(CALLERID(number)=${FROMNUMBER})
exten = s,5,SetMusicOnHold(random)
exten = s,6,Dial,Zap/g1/${ARG1}
Now, after making a call using SIP, in the CDR I have:
channel = SIP/ipp...
source = 03
clid = Jean-Yves/E 03
last data = Dial Zap/g1/0123456789
after making a call using IAX I get:
channel = IAX2/ia...
source = 300
clid = Jean-Yves/E 300
last data = Dial Zap/g1/0123456789
So my questions are:
why are the source and clid different between when a call was made
through IAX or SIP?
Ultimately, I want the clid to show up like it does for IAX:
that is:
outgoing caller ID is set to the public DID (03)
but in the CDR, I see clid = 300 (which is the local extension/account)
Is this possible?
I am using asterisk 1.2.24
Thank you
Jean-Yves
___
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Re: [asterisk-users] Question about MWI in Asterisk 1.4.0
Hi On 12/27/06, Douglas Garstang [EMAIL PROTECTED] wrote: Sounds great. What's the mechanism by which Asterisk servers communicate the mwi status between them? With new IAX commands. The client can ask the server how many messages are waiting. I've started to port the modification on 1.4, but there's been a lot of changes between 1.2 and 1.4 with the introduction of context etc.. I need to understand how 1.4 is working now which may take a while. JY ___ --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] Question about MWI in Asterisk 1.4.0
Hello I am running the following setup in order to make VoIP calls at home. Home Phone - SPA3000 - Asterisk Home - IAX2 over Internet - Asterisk Office The voice mail for Home Phone is hosted on the Asterisk Office machine. I wanted to have a way to check the status of my voicemail on my home phone directly. In order to do so, I've patched both Asterisk Home and Asterisk Office to add new sets of command allowing to check the MWI on the remote asterisk (so I see a nice voicemail icon on my home phone). This adds the possibility to configure which remote voicemail is going to be checked on a regular interval. So on the Asterisk Home machine, in the sip.conf file, for the entry used with the SPA3000, I have something like: mailbox=IAX2/hydrix/500 Which means that voicemail will be checked on mailbox 500 over the IAX2 channel named hydrix. Patching Asterisk so somewhat of a pain as I have to recreate a new patch quite regularly. Does Asterisk 1.4 introduce new capabilities allowing to remotely check the MWI on a remote machine like what my patches are doing? In the list of changes for 1.4.0 I read SIP MWI subscription support, what exactly is this ? Thanks JY ___ --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] Question about MWI in Asterisk 1.4.0
Hi On 12/26/06, Kevin P. Fleming [EMAIL PROTECTED] wrote: No, Asterisk 1.4 does not include any functionality for multi-server MWI. The SIP functionality improvements are just better support for the 'pull' model of SIP MWI, in addition to the 'push' model Asterisk has used in the past. If I adapt the patch for multi-server WMI for Asterisk 1.4, is there any chances it would be committed to trunk? Would be ace if it became a standard feature... JY ___ --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 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
Hi On 12/20/06, Lee Howard [EMAIL PROTECTED] wrote: Sure, I guess. The fax detection part comes from Asterisk or OpenPBX or whatever. Same as with rxfax/txfax, etc. Well, I know have Hylafax and iaxmodem running on my machine. Works really well so far and with spandsp 0.0.3 Will see how it goes with international fax transfer. Now when can we expect txfax and rxfax to work again? ___ --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 0. 0.3 RxFax fax =?ISO-8859-1?Q?_re ception crashes bristuffed_aster isk_1=2E2=2E13_[?= Virusgeprüft]
Hi On 12/21/06, Colin Anderson [EMAIL PROTECTED] wrote: I second that. After struggling with rxfax (which was total cake to set up, but reception reliability in my specific installation was poor) I bit the bullet and put in a separate Hylafax server connected to my Asterisk box with a crossover cable, rolled up my sleeves, and stated making IAXmodems - 1 per user. I am at over 200 IAXmodem's, and my failure rate on faxes plummeted to about .8 % - more than comparable to a regular fax machine. I don't see why rxfax would be less reliable than iaxmodem/hylafax as it's using the same spandsp to receive fax. The only difference I can think of is iaxmodem disable V17 reception within spandsp which is known to be a bit unreliable in reception. JY ___ --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 0. 0.3 RxFax fax =?ISO-8859-1?Q?_re ception crashes bristuffed_aster isk_1=2E2=2E13_[?= Virusgeprüft]
Hi On 12/21/06, Lee Howard [EMAIL PROTECTED] wrote: spandsp is a dsp library with lots of pieces to it. IAXmodem uses the T.31 portion (the Class 1 modem) which uses the actual DSP parts of V.21, V.29, and V.27ter (and potentially V.17). However, the bulk of the fax protocol is actually performed by the HylaFAX Class 1 driver. With txfax/rxfax it does not use the T.31 portion, using the T.30 portion of spandsp, instead, and thus also does not use the same fax protocol driver as does the HylaFAX+IAXmodem combination. So the only thing the same are the DSPs... V.21, V.29, V.27ter (and potentially V.17). thank you for this. What are the advantages of T31 over T30 ? In any case, congratulations on a great piece of software, it works really for well me so far. One thing I'm missing right now (but I haven't looked at what can be done to fix it), is that rxfax add useful bits of information at the top of the page (faxid, time of call etc), while hylafax leave it empty. yesterday, while doing my test, i found out that iaxmodem would unregister from asterisk once in a while (running on the local interface), what i've done around that is set up more iaxmodem than I need, however at one stage I had to restart the iaxmodem for it to reconnect as after a few minutes there were still non-registered modem. JY ___ --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] [Fwd: Re: spandsp 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13]
Hi On 12/19/06, Danny [EMAIL PROTECTED] wrote: Hi Hermann ! I am using this script [ check the commented line ] Can we please stay within the topic of this thread? Thanks JY ___ --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 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
Hi On 12/20/06, Lee Howard [EMAIL PROTECTED] wrote: This thread seems like an awfully crazy amount of work to get fax working when using IAXmodem and HylaFAX would do it without the headache, most likely. Does IAXmodem allows you to receive faxes with any extensions (auto-detecting incoming faxes). JY ___ --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 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
Hi Last month, people reported a crash with Asterisk 1.2.13 and spandsp-0.0.3 when receiving a fax using fax detection. Today I've tried 1.2.14 with the latest spandsp 0.0.3pre27 and with the snapshots for app_rxfax.c and app_txfax.c. The problem still happens. Has anyone found how to resolve this issue? I tried emailing Steve Underwood (with crash backtrace) but he hasn't answered... Otherwise, have you found spandsp 0.0.3 to provide better fax reception quality than 0.0.2? While I've had no problem with 0.0.2 locally, it usually fails when we receive faxes from overseas :( Thanks Regards Jean-Yves ___ --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 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
Hi On 12/18/06, Danny [EMAIL PROTECTED] wrote: I am using CentOS 4.4 [ asterisk-1.2.12.1 ] I too had problems with RxFax application. I tried spandsp-0.0.2pre26 spandsp-0.0.3pre23 .0.2 could install, but it crashed .0.3 doesnt install I never had any problems installing spandsp 0.0.2 with any of the version of Asterisk, and this for a few years and without a crash ever. The reason I'm looking at spandsp0.0.3 is that it's supposed to support T38 and I was also hoping it would work better with fax coming from overseas... JY ___ --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 0.0.3 RxFax fax reception crashes bristuffed asterisk 1.2.13
Hi On 12/18/06, Tzafrir Cohen [EMAIL PROTECTED] wrote: Can you provide a backtrace of the crash? Sure. I've attached a backtrace for both 1.2.13 and 1.2.14 running the same version of spandsp and all other libraries. This is on a Fedora Core 6 machine (I can not attach the message as it makes the message over 40kB) http://www.avenard.org/asterisk/trace1-2-13.txt http://www.avenard.org/asterisk/trace1-2-14.txt Just saying it crashed doesn't really help. Well, the full backtrace was reported here last month, I was just pointing out that it was still happening with 1.2.14. Also: what libraries are involved? ldd /usr/lib/asterisk/modules/app_rxfax.so linked with spandsp 0.0.2 I get: linux-gate.so.1 = (0x00c6c000) libspandsp.so.0 = /usr/lib/libspandsp.so.0 (0x0071) libtiff.so.3 = /usr/lib/libtiff.so.3 (0x006b4000) libc.so.6 = /lib/libc.so.6 (0x001e7000) libm.so.6 = /lib/libm.so.6 (0x00e43000) libjpeg.so.62 = /usr/lib/libjpeg.so.62 (0x009c5000) libz.so.1 = /usr/lib/libz.so.1 (0x00d9e000) /lib/ld-linux.so.2 (0x00534000) I unfortunately can't try with spandsp 0.0.3 right now as I need a working asterisk ... linux-gate: spandsp: 0.0.3pre27 libtiff: 3.8.2 glibc: 2.5-3 libjpeg: 6b-37 and report what is the version and package of each library mentioned there. Any more automated way of doing this? This is standard Fedora Core 6. You can find last month, on this distribution list For the archive: http://lists.digium.com/pipermail/asterisk-users/2006-November/172652.html People mentioned this issue as well as where it was crashing. Hope that helps. Jean-Yves ___ --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 0.0.3 RxFax fax recepti on crashes bristuffed asterisk 1.2.13 [Virusgeprüft]
On 12/19/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: a few weeks ago I encountered the same problem. I found out that asterisk is crashing when app_rxfax.so is calling line 327 of app_rxfax.c 'ast_frfree(int);' out of the testing tree running with actual spandsp-0.0.3 commenting this line out it doesn't crash *, but that's no solution it do work with asterisk-1.2.9 but not with 1.2.13 - not tested 1.2.14 yet I tried with all version of Asterisk since 1.2.9, all crashes at the same spot as you mentioned. I guess commenting the line is one solution, provided you restart asterisk so it doesn't leak memory too much. We don't receive that many faxes anyway... JY ___ --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] Digium TE405P with French E1 = Red Alert
Hi On 12/18/06, Noc Phibee [EMAIL PROTECTED] wrote: Hi it's Colt-Telecom. you have a TE405P ? you don't mention what's wrong with it though... ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and spandsp 0.3
Hi As there been any progress regarding the use of spandsp 0.3 with Asterisk 1.2.13? Last month there was a thread about how spandsp 0.3 and rxfax from http://www.soft-switch.org/downloads/snapshots/spandsp made asterisk crash. Is there any resources on how to get spandsp 0.3 work with Asterisk otherwise? Thank you JY ___ --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] Permission for files generated by voicemail
Hi
There is a problem in Asterisk 1.2.10 (at least). Even though in
theorie the source code of app_voicemail.c can be modifier to set up
the proper permission on the directories and file created for the
voicemail, this code can not work.
It doesn't take into account that the umask needs to be set properly
for the argument given to open to act as intended. As a result,
changing the value of VOICEMAIL_FILE_MODE will have no effect in most
cases.
I've adapted a patch that I found earlier which also set-up the group
owner. I've only extracted setting up the permissions as that's all I
needed and starting asterisk with the right group permission does the
job just as well.
Is there a centralized way to post all those patches? I have a few
more in the pipeline ...
Thanks
JY
diff -r -u asterisk-1.2.10/apps/app_voicemail.c asterisk-1.2.10-umask/apps/app_voicemail.c
--- asterisk-1.2.10/apps/app_voicemail.c 2006-07-14 07:22:11.0 +1000
+++ asterisk-1.2.10-umask/apps/app_voicemail.c 2006-08-01 18:24:08.0 +1000
@@ -74,9 +74,12 @@
#include asterisk/res_odbc.h
#endif
+#include pwd.h
+#include grp.h
+
#define COMMAND_TIMEOUT 5000
-#define VOICEMAIL_DIR_MODE 0700
-#define VOICEMAIL_FILE_MODE 0600
+#define VOICEMAIL_DIR_MODE 0770
+#define VOICEMAIL_FILE_MODE 0660
#define VOICEMAIL_CONFIG voicemail.conf
#define ASTERISK_USERNAME asterisk
@@ -421,6 +424,36 @@
LOCAL_USER_DECL;
+static void set_owner_and_group_all(const char* dir, int msgnum)
+{
+ DIR *vmdir = NULL;
+ struct dirent *vment = NULL;
+char fn[32];
+ char pn[1024];
+ snprintf(fn, sizeof(fn), msg%04d, msgnum);
+
+ if (sizeof(dir) + 11 = sizeof(pn)) {
+ ast_log(LOG_WARNING, directory name too long to set owner and group, skipping\n);
+ return;
+ }
+ if ((vmdir = opendir(dir))) {
+ while ((vment = readdir(vmdir))) {
+ if (!strncmp(vment-d_name, fn, 7)) {
+strcpy(pn, dir);
+pn[strlen(dir)] = '/';
+pn[strlen(dir)+1] = 0;
+strcat(pn, vment-d_name);
+if (chmod(pn, VOICEMAIL_FILE_MODE)) {
+ast_log(LOG_WARNING, chmod '%s' failed: %s\n,
+ pn, strerror(errno));
+}
+ }
+ }
+ closedir(vmdir);
+ }
+}
+
+
static void populate_defaults(struct ast_vm_user *vmu)
{
ast_copy_flags(vmu, (globalflags), AST_FLAGS_ALL);
@@ -2635,6 +2668,7 @@
rename(tmptxtfile, txtfile);
ast_unlock_path(dir);
+ set_owner_and_group_all(dir, msgnum);
/* Are there to be more recipients of this message? */
while (tmpptr) {
___
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Re: [asterisk-users] Message waiting question...
Hi
For those interested.
I've ported the following solution:
http://bugs.digium.com/view.php?id=4236
for asterisk 1.2.10
it works well.
To activate MWI over IAX2 add the following line in sip.conf or
zaptel.conf etc...
mailbox=IAX2/{iax2_context}/[EMAIL PROTECTED]
For some weird reason though, on my panasonic while I can hear the
message waiting tone, the little enveloppe icon doesn't appear on the
phone
The patch against 1.2.10 is attached to this email.
I have also ported this solution:
http://bugs.digium.com/view.php?id=4371
to asterisk 1.2.10 if anyone is interested.. It's more powerful but
much more complicated to configure ...
Jean-Yves
On 7/31/06, Jean-Yves Avenard [EMAIL PROTECTED] wrote:
Hi
On 7/27/06, Luki [EMAIL PROTECTED] wrote:
There is this old patch that does remote MWI over IAX (among other
things). I used it on earlier versions and it worked quite nicely.
This was before 1.2 so it may no longer work at all. At the very least
it will likely required some updating. Doable, just depends how much
time you want to put into it :).
I think I got it working with Asterisk 1.2.10. I can see the mailbox
information being sent between the two servers.
However, my SPA3000 is still not getting the MWI properly. I believe
that the problem is with the configuration file. I'm not sure I fully
understand the information provided: in the bug report there are
several conflicting setup and I'm not sure which one is good.
Setup being:
SPA3000 -(SIP) Asterisk1 ---(IAX2) Asterisk2
and I want SPA3000 to check if there's voicemail waiting on Asterisk2.
On Asterisk2:
in iax.conf I have:
[iax_peer_name_to_asterisk1]
type=friend
mailbox=500
host=dynamic
...
in voicemail.conf
voicemail_server=iax_peer_name_to_asterisk1
[default]
[EMAIL PROTECTED] = password,email etc..
On Asterisk1:
iax.conf:
[iax_peername_to_asterisk2]
type=friend
username=iax_peer_name_to_asterisk1
secret=..
host=asterisk2_hostname
sip.conf:
[spa3000]
mailbox=500:[EMAIL PROTECTED]
Am I correct?
I haven't digged too much in the source code, I'm starting to have a
good understanding but any help would be appreciated.
JY
diff -r -u asterisk-1.2.10/app.c asterisk-wmi.patch/app.c
--- asterisk-1.2.10/app.c 2006-07-13 01:46:56.0 +1000
+++ asterisk-wmi.patch/app.c 2006-08-01 03:02:08.0 +1000
@@ -232,6 +232,8 @@
static int (*ast_has_voicemail_func)(const char *mailbox, const char *folder) = NULL;
static int (*ast_messagecount_func)(const char *mailbox, int *newmsgs, int *oldmsgs) = NULL;
+static int (*ast_iax_has_voicemail_func)(const char *mailbox, const char *folder) = NULL;
+static int (*ast_iax_messagecount_func)(const char *mailbox, int *newmsgs, int *oldmsgs) = NULL;
void ast_install_vm_functions(int (*has_voicemail_func)(const char *mailbox, const char *folder),
int (*messagecount_func)(const char *mailbox, int *newmsgs, int *oldmsgs))
@@ -246,35 +248,135 @@
ast_messagecount_func = NULL;
}
+void ast_install_iax_vm_functions(int (*iax_has_voicemail_func)(const char *mailbox, const char *folder),
+ int (*iax_messagecount_func)(const char *mailbox, int *newmsgs, int *oldmsgs))
+{
+ ast_iax_has_voicemail_func = iax_has_voicemail_func;
+ ast_iax_messagecount_func = iax_messagecount_func;
+}
+
+void ast_uninstall_iax_vm_functions(void)
+{
+ ast_iax_has_voicemail_func = NULL;
+ ast_iax_messagecount_func = NULL;
+}
+
int ast_app_has_voicemail(const char *mailbox, const char *folder)
{
- static int warned = 0;
- if (ast_has_voicemail_func)
- return ast_has_voicemail_func(mailbox, folder);
-
- if ((option_verbose 2) !warned) {
- ast_verbose(VERBOSE_PREFIX_3 Message check requested for mailbox %s/folder %s but voicemail not loaded.\n, mailbox, folder ? folder : INBOX);
- warned++;
+ static int loc_warned = 0;
+ static int iax_warned = 0;
+ char *workspace, *mbp, *comma;
+ int ret = 0;
+
+ if (!mailbox)
+ return 0;
+
+ workspace = mbp = strdup(mailbox);
+ if (!workspace)
+ return 0;
+
+ while (*mbp) {
+ if ((comma = strchr(mbp, ',')))
+ *comma++ = '\0';
+
+ if (strncasecmp(iax2/, mbp, 5) == 0) {
+ if (!ast_iax_has_voicemail_func) {
+if (!iax_warned (option_verbose 2)) {
+ ast_verbose(VERBOSE_PREFIX_3 Message check requested for %s/folder %s but IAX voicemail checks not loaded.\n, mailbox, folder ? folder : INBOX);
+ iax_warned++;
+}
+ret = 0;
+ } else
+ret = ast_iax_has_voicemail_func(mbp + 5, folder);
+ } else { /* local voicemail system check */
+ if (!ast_has_voicemail_func) {
+if (!loc_warned (option_verbose 2)) {
+ ast_verbose(VERBOSE_PREFIX_3 Message check requested for mailbox %s/folder %s but voicemail checks not loaded.\n, mailbox, folder ? folder : INBOX);
+ loc_warned++;
+}
+ret = 0;
+ } else
+ret = ast_has_voicemail_func(mbp, folder);
+ }
+
+ if (ret)
+ break;
+
+ if (comma)
+ mbp = comma;
+ else
+ break;
}
- return 0;
-}
+ free(workspace);
+ return ret;
+}
int ast_app_messagecount
Re: [Asterisk-Users] How to configure NOKIA N70 with Asterisk?
Hi On 8/1/06, FaberK [EMAIL PROTECTED] wrote: Hi folks, I got an N70. Any lynks for the voip/sip configuration? Thanks .:FaberK:. they aren't hard to find ! this one works for me: http://www.newlc.com/Using-SIP-with-Nokia-Series60-and.html One note of warning : the Nokia will not work if behing NAT ... I've tried everything but I've never managed to get it to work unless the Nokia had a public IP address or was on the same subnet as the asterisk server. Be interested to know if you can find a way around this JY ___ --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] Message waiting question...
Hi On 7/27/06, Luki [EMAIL PROTECTED] wrote: There is this old patch that does remote MWI over IAX (among other things). I used it on earlier versions and it worked quite nicely. This was before 1.2 so it may no longer work at all. At the very least it will likely required some updating. Doable, just depends how much time you want to put into it :). I think I got it working with Asterisk 1.2.10. I can see the mailbox information being sent between the two servers. However, my SPA3000 is still not getting the MWI properly. I believe that the problem is with the configuration file. I'm not sure I fully understand the information provided: in the bug report there are several conflicting setup and I'm not sure which one is good. Setup being: SPA3000 -(SIP) Asterisk1 ---(IAX2) Asterisk2 and I want SPA3000 to check if there's voicemail waiting on Asterisk2. On Asterisk2: in iax.conf I have: [iax_peer_name_to_asterisk1] type=friend mailbox=500 host=dynamic ... in voicemail.conf voicemail_server=iax_peer_name_to_asterisk1 [default] [EMAIL PROTECTED] = password,email etc.. On Asterisk1: iax.conf: [iax_peername_to_asterisk2] type=friend username=iax_peer_name_to_asterisk1 secret=.. host=asterisk2_hostname sip.conf: [spa3000] mailbox=500:[EMAIL PROTECTED] Am I correct? I haven't digged too much in the source code, I'm starting to have a good understanding but any help would be appreciated. JY ___ --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] Message waiting question...
Hi On 7/27/06, Luki [EMAIL PROTECTED] wrote: There is this old patch that does remote MWI over IAX (among other things). I used it on earlier versions and it worked quite nicely. This was before 1.2 so it may no longer work at all. At the very least it will likely required some updating. Doable, just depends how much time you want to put into it :). Thank you for this link, very interesting. I've started porting it on Asterisk 1.2. For it to work, do you need to use two patched Asterisk on either side? or only on the machine wanting to retrieve the MWI status ? Thanks JY ___ --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] Message waiting question...
Hi On 7/27/06, Joshua Colp [EMAIL PROTECTED] wrote: I don't believe there's anything configurable but if you open app_voicemail.c there's two declarations, VOICEMAIL_DIR_MODE and VOICEMAIL_FILE_MODE which set the permissions. DIR mode is at 0770 right now and FILE mode is at 0660. Hum.. Weird then, on my maching the file mode is definitely 0600 .. I used the ATrpm package for Fedora Core 5... JY ___ --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] Message waiting question...
Hi I have the following setup: SPA3000 (at home) -- Asterisk1 server (at home) --- Asterisk2 server (at work). The reason the SPA3000 isn't connected directly to Asterisk server 2 is because the SPA3000 can't register to more than one SIP account at a time, plus it was more fun that way :) Anyhow, Asterisk1 and Asterisk2 are connected using IAX2. What I would like is to have the SPA3000 Message Waiting indicator based on the voicemail message hosted on the Asterisk2 server. Is this possible? Thanks JY ___ --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] Message waiting question...
Hi. Thank you so much for answering. I guess I couldn't get a better qualified answer ! On 7/27/06, Joshua Colp [EMAIL PROTECTED] wrote: Anything is possible, it's just to what extreme do you want to go to make it happen. Right now we have no way of transporting arbitrary information (like MWI status) between servers. In the future however I'm hoping we'll have something. For now there's two er I mean three ways off the top of my head you could approach this. Hum... I'm afraid that what I was expecting 1. Using ODBC storage to store your voicemail in a database and have each server setup against that database. The MWI will just query the database to see if there are messages, and since there will be... MWI will be sent to the phone. This may be a disturbing solution, I have over 30 voicemail on server2 and I guess I would have to convert all of them first. This may be the easiest solution if you can set up a database voicemail for one user only... 2. Using the ability to execute an outside application that exists right now and using your own method to communicate back to turn on MWI (maybe generating a SIP NOTIFY to poke the phone with?). That sounds quite complicated... 3. Share the voicemail directory over something like NFS. How often does asterisk check the content of the voicemail directory? the two machines connect over a 512kbit/s link, I'm afraid there could be a bandwidth problem. JY ___ --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] Message waiting question...
Hi On 7/27/06, Joshua Colp [EMAIL PROTECTED] wrote: chan_sip requests the count fairly frequently, dunno how much traffic it would actually generate though. Well I took the very easy route. Every minute I do a rsync between server2 and server1 of the INBOX directory I want to check. I also only transfer the .txt file so it never needs to transfer more than 500 bytes max every minute. Having just the .txt file is sufficient for Asterisk to tell the SPA3000 that there's a message waiting. And best of all: it works :) As a side question, is there a way to force asterisk to set specific group permission on the file generated for the voicemail? I found some patches for earlier version of Asterisk and at one stage that it made its way into asterisk trunk I can't find any documentation about how to configure it though. Thanks JY ___ --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] Uniden UIP200 and Asterisk v1.2.4: problem notregistering
On 2/7/06, Nabeel Jafferali [EMAIL PROTECTED] wrote: Removing this line will likely fix the problem. Since you don't have a NAT,the qualify= setting doesn't help keep the port(s) open. At the same time,most SIP devices have a NAT Keep Alive option, if that is an issue. HelloIt did fix my problem, thank you for this.Wonder why this use to work with Asterisk earlier than 1.2.x ___ --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] Uniden UIP200 and Asterisk v1.2.4: problem not registering
HelloWe recently moved to Asterisk 1.2.4 (from 1.0.x) and our 10 Uniden UIP200 have stopped working ever since.We can make a call with the UIP200 to any other extensions, but it can not receive a call. In fact the UIP200 always appears offline: It does show up in asterisk a few seconds after the UIP200 reboot:-- Saved useragent Uniden SIP Phone p2 Ver BS4.70 for peer uip200but after about 5s I will get something like:UIP200 is now unreachable. htpc*CLI sip show peersName/username Host Dyn Nat ACL Port Status uip200/uip200 192.168.10.104 D 5061 UNREACHABLE I have tried the latest firmware (v4.70) and the previous one we've been running for over 18 months (v4.59) without any luckHere is the sip.conf I've created on a test server where Asterisk is using the port 5061 , same for the UIP200 using port 5061. There is no NAT, the UIP200 is on the same subnet as the asterisk server: (I'm trying to isolate the issue without affecting our main asterisk server)[uip200]type=friendport=5061secret=uip200 ; password for registrationnat=never ; phone may be behing nat host=dynamicreinvite=nocanreinvite=noqualify=3000 ; send udp every 2 seconds, to keep nat opencallerid=Jean-Yves 200dtmfmode=rfc2833 ; DTMF mode context = jya-in ; Default context for incoming callsdisallow=allallow=ulawallow=alawallow=g729If I unable: sip debug ip 192.168.10.104 (the UIP200 IP address), I get every 3-4 seconds on the console: ---Retransmitting #2 (no NAT) to 192.168.10.104:5061:OPTIONS sip:[EMAIL PROTECTED]:5061 SIP/2.0Via: SIP/2.0/UDP 192.168.10.11:5061 ;branch=z9hG4bK08dcd4a8;rportFrom: asterisk sip:[EMAIL PROTECTED]:5061;tag=as67a81892To: sip:[EMAIL PROTECTED]:5061Contact: sip:[EMAIL PROTECTED]:5061Call-ID: [EMAIL PROTECTED]CSeq: 102 OPTIONSUser-Agent: Asterisk PBXMax-Forwards: 70Date: Mon, 06 Feb 2006 15:40:37 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFYContent-Length: 0Any help would be greatly appreciated.Thank you in advance.RegardsJY ___ --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] Problem: Can't make outgoing call
HelloI'm having trouble since I recently upgraded to Asterisk 1.2.x.I have a Sipura SPA3000 which is registered to my asterisk server.It can receive VoIP call perfectly, but can't make call.In the Asterisk SIP debug I see things like:"SIP/2.0 407 Proxy Authentication Required"Googling gave me some clues, and I found that by removing the "secret=" in sip.conf and leaving blank the password field on the sipura configuration page, actually allowed me to make calls just fine.Obviously, I do not want to leave in place a SIP client that connects without any password !Any idea on why I would be able to make calls if their no password needed, but as soon as I put a password then it fails ?Is this an issue with Asterisk 1.2 ?RegardsJean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 8573 5200 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem Asterisk: can't make call but can receive calls
Dear all.Okay, I'm about to give up now as this issue has been driving me insane for a whileI have an asterisk server (which I have just updated to 1.2beta1).Several phones are connected on LAN, no problemPeople on the go using either x-lite or eyebeam without a problemI have a Sipura 3000 configured at my parents in France used to work fine until about 1 month ago. Since it can receive calls but can't make call. I keep hearing the busy signalToday, one of our guy is travelling to Spain. He connected with x-lite just fine to the asterisk server, he can receive calls but he can't make calls.It keeps getting the error:"Call failed: 408 Timeout" when dialling a numberI don't see any attempts for connection on the Asterisk server logs...There's no firewall issue on the server side as just for the sake of it I've opened all ports... the firewall on his laptop is also turned offAny ideas?RegardsJean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 8573 5200 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TE405P V2 - Fantastic!
Is there similar changes for the TE110 card? This new firmware only works on new hardware I guess.. Jean-Yves On 26/09/2005, at 2:36 PM, Kevin P. Fleming wrote: Bridged calls with 2nd gen firmware result in the audio never leaving the card; that's why you are seeing such an improvement. Essentially, the Zaptel 'native bridge' is pushed all the way down into the card, so the audio stream is never passed across the PCI bus (it's not even packetized, just directly connected between the two channels). --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 8573 5200 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax detection: Problem with extension number
Hi.On 26/05/2005, at 4:31 PM, Jean-Christophe Heger wrote:For what I'm seeing in your log, the fax is detected, but you're missingthe fax extension. Here is how it works on my asterisk:uh??did you really read my email?Jean-Yves Avenard a écrit :[answer-extension]exten = 1,1,Answerexten = 1,2,Macro(setcallerid)exten = 1,3,Ringingexten = 1,4,Wait(3)exten =1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},${EXTENSION})exten = fax,1,Goto(faxreceive,1,1)Here.. I have a fax extension..My point was that if a macro is called, and it detects the fax while in this macro, then it tries to jump to the fax extension but can't find it.Even if I put a fax extension in the Macro it makes no difference.Looks like a bug to me in AsteriskJean-Yves___
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[Asterisk-Users] Fax detection: Problem with extension number
HelloI've been having the following problem today :I have a quite simple dialplan made to receive a fax:[answer-extension]exten = 1,1,Answerexten = 1,2,Macro(setcallerid)exten = 1,3,Ringingexten = 1,4,Wait(3)exten = 1,5,Macro(stdfwd3iax-notransfer,${EXTENSION},${EXTENSION},${EXTENSION})exten = fax,1,Goto(faxreceive,1,1)The Wait(3) is there simply to let the system a bit of time to detect if it's a fax calling, this has worked so far in all cases except today.I received a fax from overseas and it seems that Asterisk has been unable to detect that it was a fax in the 3 seconds wait. Changing to 5s was sufficient to receive the fax. But obviously this is not a solution I want to adopt all the time as 1-it's a tool long wait, 2-what happens if 5s is still not enough another time.So Asterisk recognized that a fax was being received while executing the macro stdfwd3iax-notransfer (this extension simply checkAnd I saw the following message in the console: -- Executing VoiceMail("Zap/10-1", "u200") in new stack -- Playing '/data/asterisk/var/spool/asterisk/voicemail/default/200/unavail' (language 'en') -- Redirecting Zap/10-1 to fax extensionMay 25 01:19:34 WARNING[17629]: pbx.c:2412 ast_pbx_run: Timeout, but no rule 't' in context 'answer-extension' -- Hungup 'Zap/10-1'It seems that Asterisk once entered in a Macro is unable to jump to the fax extension and gave me a timeout (which I do not handle in my dialplan). If I change the Wait(3) into Wait(0) the problem can be easily reproduced at all times.I then added a fax extension in the Macro just in case, but it made no difference whatsoever.Any ideas on what I'm doing wrong or is this a problem with Asterisk?The fully log is below.Thank you in advanceJean-YvesWhen I looked in the console on what what happening I say this: -- Accepting call from '' to '85735200' on channel 0/10, span 1 -- Executing AGI("Zap/10-1", "getnumber.agi|200") in new stack -- Launched AGI Script /data/asterisk/var/lib/asterisk/agi-bin/getnumber.agi -- AGI Script getnumber.agi completed, returning 0 -- Executing Set("Zap/10-1", "EXTENSION=00") in new stack -- Executing Goto("Zap/10-1", "answer-extension|1|1") in new stack -- Goto (answer-extension,1,1) -- Executing Answer("Zap/10-1", "") in new stack -- Executing Macro("Zap/10-1", "setcallerid") in new stack -- Executing GotoIf("Zap/10-1", "11?10:11") in new stack -- Goto (macro-setcallerid,s,10) -- Executing Set("Zap/10-1", "CALLERID(number)=''") in new stack -- Executing GotoIf("Zap/10-1", "0?20") in new stack -- Executing Ringing("Zap/10-1", "") in new stack -- Executing Wait("Zap/10-1", "0") in new stack -- Executing Macro("Zap/10-1", "stdfwd3iax-notransfer|200|200|100") in new stack -- Executing DBget("Zap/10-1", "temp=CFIM/200") in new stack -- DBget: varname=temp, family=CFIM, key=200 -- DBget: Value not found in database. -- Executing Goto("Zap/10-1", "3") in new stack -- Goto (macro-stdfwd3iax-notransfer,s,3) -- Executing Dial("Zap/10-1", "IAX2/iax100|20|tr") in new stackMay 25 01:19:21 NOTICE[17629]: app_dial.c:972 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) == Everyone is busy/congested at this time (1:0/0/1) -- Executing NoOp("Zap/10-1", "CHANUNAVAIL") in new stack -- Executing Goto("Zap/10-1", "s-CHANUNAVAIL|1") in new stack -- Goto (macro-stdfwd3iax-notransfer,s-CHANUNAVAIL,1) -- Executing Goto("Zap/10-1", "s|400") in new stack -- Goto (macro-stdfwd3iax-notransfer,s,400) -- Executing Dial("Zap/10-1", "SIP/ipp100|20|tr") in new stackMay 25 01:19:21 NOTICE[17629]: app_dial.c:972 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/0/1) -- Executing Goto("Zap/10-1", "s2-CHANUNAVAIL|1") in new stack -- Goto (macro-stdfwd3iax-notransfer,s2-CHANUNAVAIL,1) -- Executing Goto("Zap/10-1", "s|200") in new stack -- Goto (macro-stdfwd3iax-notransfer,s,200) -- Executing DBget("Zap/10-1", "temp=CFBS/200") in new stack -- DBget: varname=temp, family=CFBS, key=200 -- DBget: Value not found in database. -- Executing Goto("Zap/10-1", "202") in new stack -- Goto (macro-stdfwd3iax-notransfer,s,202) -- Executing VoiceMail("Zap/10-1", "u200") in new stack -- Playing '/data/asterisk/var/spool/asterisk/voicemail/default/200/unavail' (language 'en') -- Redirecting Zap/10-1 to fax extensionMay 25 01:19:34 WARNING[17629]: pbx.c:2412 ast_pbx_run: Timeout, but no rule 't'
Re: [Asterisk-Users] Asterisk - Spandsp: fax header
HelloOn 20/05/2005, at 6:41 AM, Andrew Kohlsmith wrote:That would be the ${REMOTESTATIONID} extension variable. RxFax sets it, among other variables, upon fax reception completion. "Show Application RxFax" for more information. Great.. Exactly what I was looking for... Thank you for that.Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___
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Re: [Asterisk-Users] Asterisk - Spandsp: fax header
HelloOn 18/05/2005, at 4:09 PM, Peter Svensson wrote:I think he is refering to the remote fax id to be presented, not the header. I.e. the 20 digit user selectable number on the remote fax. The one often seen on the lcd of the receiving fax and so on. Yes that's exactly what I'm referring to.Most fax machines I've used print this information on the top left corner or top right corner on any fax received.Is it possible to do this with SpanDSP?Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Spandsp: fax header
HiOn 18/05/2005, at 9:35 PM, Andrew Kohlsmith wrote:You can get the info and stamp it into the image yourself with some third party TIFF manipulation tools, I bet. I wouldn't mind doing so if I knew where this Fax ID information is stored or how to retrieve it, or if it's even possible.JY --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Spandsp: fax header
Hi PeterOn 18/05/2005, at 10:05 PM, Steve Underwood wrote:It is only there because the sending machine put it there in the image. Spandsp is not different from how any FAX machine I have ever used behaves. As well as sending the 20 digit number as text, the sending machine puts in the header. This is not what I'm referring to... I know what is being put by the remote fax !On my Brother's fax machine (MFC-8820D) today, I've received 3 faxes: all of them at the top showed the caller Fax identity.I received 2 faxes on Asterisk with spandsp, one from the same sender as earlier on the brother: there's nothing at the top.I wouldn't ask if it was obvious the data was inside the image, give me some credits for God's sake !Typically, when somebody is sending a fax on the Brother unit, once the connection has been established the identity of the fax caller is then displayed on the Brother's LCD (and this has nothing to do with PSTN CallerID), what is displayed on the LCD will be printed at the top of each pages. This is this behavior I'm trying to reproduce with Asterisk/Spandsp.JY --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - Spandsp: fax header
HiOn 19/05/2005, at 1:41 AM, Steve Underwood wrote:So you get the calling machine's number shown twice at the top of each page? Once in this extra header, and once in the normal header sent as part of the image? Weird. FAX machines don't normally do that. Does this extra header overlay a part of the page, or does it make the page one line longer? It is printed in very faint characters very close to the top edge of a page. Spandsp puts the calling machine's number in one of the tag fields in the TIFF headers. It puts several things in those tags - the name of the software which generated the file (spandsp), the hostname of the receiving computer, the far machine's ident, the far machine's maker and model (if they can be identified). Programs like tiffdump will show that information. Some image viewers also allow you to see it (don't ask me which ones off hand). I just ran tiffdump on some of the tifff files received, and I can't see the Fax ID in those :(In the fax I sent to myself there is a "ImageDescription (270) ASCII (2) 13" which then contain the entry i entered in the fax settings .Thanks for the hint.Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - Spandsp: fax header
Hello Steve.On 17/05/2005, at 10:54 PM, Steve Underwood wrote:When a fax is received the header you see is part of the image. As such, it ends up in the TIFF file as part of the image. It is not available as text anywhere. The only way to make it available as text would be to implement OCR. That is complex and messy, and I have no intention to try. The only information sent between the FAX machines as text is the "identifier" - a 20 character string, which the standard says should be digits, and which is usually set to the telephone number of the FAX machine.That's exactly what I was referring to. Is there any way to display that in the image received?As I wrote previously I usually don't get the sender's callerID, but I do get the phone number printed when receiving with a usual fax machine. It would be nice to get this information with spandsp tooTo send a fax I'm still using our Brother fax unit, works well...Sorry for not using a new threads before... I thought about it, but the title was correct :)JY --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Dial plan - does not stop after first match
HelloOn 17/05/2005, at 8:13 AM, Michael Stahl wrote: My dial plan seems to work great - in that when I call extensions 1234 it connects to 1234. Strangely, after the call terminates (the other side hangs up first), Asterisk continues in the same context and then matches to extensions _. which causes an invalid extension error! Why does asterisk not leave the context (called internalmenu) after the remote hangup? Instead, it continues to the InternalInvalid context (included later in the InternalMenu context). I'm confused! Here is a snippet of the relevant context and macro. Thanks, Mike But that's exactly what your dialplan ask asterisk to do: keep doing stuff after the call has hanged up.If you do not want to continue after Dial, then make sure there a Hangup and nothing else after that. ;; Macros; [macro-stdexten];; Standard extension macro:; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well; ${ARG2} - Device(s) to ring;exten = s,1,Playback(transfer,skip) ; "Please hold while..." but skip if channel is not up;exten = s,2,SendText('Calling extension '${ARG1}) ; Tell the user what extension being calledexten = s,2,SetVar(LastStatus=CallDone) ; Ensure script knows that a Dial was completedexten = s,3,Dial(${ARG2},${RINGTIME},r) ; Ring the interface, 20 seconds maximumexten = s,4,NoOp(${DIALSTATUS}) ; Show status after hangupexten = s,5,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) Here , you tell asterisk to Goto s-. after the Dial operation... exten = s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten = s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce exten = s-ANSWER,1,NoOp ; If call answered, then do nothing after hangup exten = _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer exten = a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___
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Re: [Asterisk-Users] Asterisk - fax - spandsp
On 16/05/2005, at 11:48 PM, Steve Underwood wrote:How can that work? You can measure the error, but you have no ability to tweak the clock from software. Two cards could only be synced by hardware.Side questions about spandsp... Is it possible to print the fax header like what most faxes do (that is: who is sending the fax, how many pages are included etc...) I'm not talking about printing callerid, often I receive fax from the US (and there's no CallerID being displayed then) but my fax machine can print the fax header very well.Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
HelloOn 15/05/2005, at 1:55 PM, Steve Underwood wrote:The right thing to do is to sync to the PSTN. The E1s connected to the PSTN should be listed as the lowest numbered clock sources, starting from 1. Places you never want to sync to should be set to zero. Your mail was very interesting, but to be honest I don't think I understood a single word of what you said :)This is my zaptel.conf configuration:loadzone = audefaultzone=au#TE110Pspan=1,0,0,ccs,hdb3,crc4bchan=1-10dchan=16#bchan=17-31#TDM400fxsls=32-35Do you mean that I should change the span=1,0,0,ccs,hdb3,crc4intospan=1,1,0,ccs,hdb3,crc4I only have one E1 connected (on TE110P card) and a TDM440 (4 FXO ports)Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
Hello SteveOn 15/05/2005, at 4:40 PM, Steve Underwood wrote:The second parameter now says "treat this E1 as the first priority as the clock source". Your box should lock itself to the PSTN's clock. If that makes no sense, the bottom line is "this is good". :-) I do not know if it was from upgrading the linux kernel to 2.6.10, turning off hyperthreading and changing the zaptel configuration line.But the bottom line is this: "i now receive faxes perfectly!"As a side note regarding clock source, the documentation for zaptel.conf states that 1 is for primary timing source, 2 is for secondary and 0 is for not using this connecting as a timer source...Should it be that any number 0 is in fact the priority for using it as an external source?I'm glad it works now.Is there any future support for fax in colours? just wondering, I don't think I've ever used it before anyway :)CheersJean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
HelloOn 15/05/2005, at 4:40 PM, Steve Underwood wrote:span=1,1,0,ccs,hdb3,crc4 The second parameter now says "treat this E1 as the first priority as the clock source". Your box should lock itself to the PSTN's clock. If that makes no sense, the bottom line is "this is good". :-) Hum.. an interesting side effect to using the E1 as a primary clock source, is that I can't remove the wcte11xp kernel module anymore...it hangs the machine if I do so. Any ideas on how to do that?Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
HiOn 15/05/2005, at 5:51 PM, Bryce Chidester wrote:I have the same trouble with wct1xxp. I'd just chalked it up to a PCI bug or other low-level hardware problem with the Digium card. I've simply learned not to, though it would be nice to not have to learn work-arounds.This only happen if I have something like:span=1,1,0,ccs,hdb3,crc4If I have:span=1,1,0,ccs,hdb3,crc4then it doesn't hang... Doubt it's a PCI bug Luckily I have an electronic switch I can control remotely to turn the machine off/on !JY --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
HelloOn 15/05/2005, at 5:49 PM, Steve Underwood wrote:I don't get that with a TE410P. What happens if you disconnect the cable first, so your box cannot sync to the port, and no longer treats the far end as the clock master? Can you remove the module like that? Whatever, it is a bug, and should be reported. Actually.. I seem to have jumped to improper conclusion..It seems to happen everytime I unload the zaptel module in the 2.6.10 kernel ; it doesn't happen in 2.6.8.I will revert to 2.6.8 and do some more testings: it seems that changing the zaptel configuration was the only factor as to why I couldn't receive fax properly: I changed the line back to "span=1,0,0,ccs,hdb3,crc4" and the fax were corrupted just as beforeThank you very much for spending the time to answer my questions, I feel honoured: you've done an amazing job with spandspJean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
On 15/05/2005, at 5:58 PM, Steve Underwood wrote:Well, it could be worse. You could be using Dialogic card, which randomly crash the machine when you try to restart the drivers, and have continued to do so through 10 years of updates. :-\ Seriously, this really shouldn't happen. It must be a silly bug or procedural thing somewhere. Are you sure you stopped Asterisk completely, so there was nothing left holding on to the driver? No matter on how well I remove the drivers (including simply rebooting the machine), the zaptel drivers will hang with the 2.6.10 kernel.I have reverted to 2.6.8 now and removing the zaptel modules is working well (as it has done so for the past year).I received a fax with the 2.6.8 kernel (and asterisk 1.0.7) and it seems to have some errors in it (very small glitches) while it was perfect with 2.6.10.Correction: I have just received another fax again with 2.6.8 and it's perfect... So I guess there are just occasional glitches. So I'll stick with 2.6.8. I don't want to bother with a kernel that can hung when unloading a moduleJY --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wcte11xp / wcfxs kernel module removal
Ok.On 15/05/2005, at 6:10 PM, Bryce Chidester wrote:Seems rather fishy that no matter medium or timing source, rmmod'ing the module could hang the machine. Maybe I'll look into this more when I get the logic analyzer hooked up. I've identified how to reproduce the problem.If I do:modprobe zaptelmodprobe wcte11xpmodprobe wcfxsthen if I do:modprobe -r wcfxs : OKmodprobe -r wcte11xp : linux 2.6.10 will handIf I charge the TDM driver first:modprobe zaptelmodprobe wcfxsmodprobe wcte11xpthen I can unload all the modules just fine...And as I can't make the TE110P work if the driver isn't loaded first.. I guess there's little hope for me to use the 2.6.10 kernel..Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with extensions and when channel is unavailable
HelloI used to have an extension like this which worked fine with asterisk 1.0.7I first dial to see if an IAX phone is present, if not I would try on SIP insteadexten=s,1,Dial(IAX2/iax${ARG3},20,tr) ; 20sec timeoutexten=s,2,Goto(s-${DIALSTATUS},1); Default actionexten=s,200,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing, goto 301exten=s,201,Dial(Local/[EMAIL PROTECTED]/n) ; Forward on unavailableexten=s,202,VoiceMail,u${ARG2}exten=s,203,hangup()exten=s,301,Goto(202) ; entry doesn't exist - Voicemailexten=s-CHANUNAVAIL,1,Goto(s,400)exten=s-CONGESTION,1,Goto(s,200)exten=s-NOANSWER,1,Goto(s,200)exten=s-.,1,Goto(s,200);Now try SIPexten=s,400,Dial(SIP/ipp${ARG3},20,tr) ; 20sec timeoutexten=s,401,Goto(s2-${DIALSTATUS},1)exten=s2-CHANUNAVAIL,1,Goto(s,200)exten=s2-CONGESTION,1,Goto(s,200)exten=s2-NOANSWER,1,Goto(s,200)exten=s2-.,1,Goto(s,200)If the IAX channel didn't exist or wasn't connected, it would jump to s-CHANUNAVAIL which jumps to priority 400 and try SIPIn CVS-HEAD it now goes to s-CONGESTION which is definitely not what I want as it goes in this case to voicemails-DIALSTATUS has changed now? is there a documentation on how it should work now?Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___
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Re: [Asterisk-Users] Problem with extensions and when channel is unavailable
A bit more details:Calling an IAX entry which doesn't exist:May 15 21:09:49 WARNING[608]: chan_iax2.c:2727 create_addr: No such host: iax107May 15 21:09:49 NOTICE[608]: app_dial.c:968 dial_exec_full: Unable to create channel of type 'IAX2' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) -- Executing NoOp("Zap/5-1", "CONGESTION") in new stackShould have been CHANUNAVAILSame for SIP -- Executing Dial("SIP/ipp100-e5aa", "SIP/ipp107|20|Ttr") in new stackMay 15 21:12:55 NOTICE[608]: app_dial.c:968 dial_exec_full: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time (1:0/1/0) -- Executing NoOp("SIP/ipp100-e5aa", "CONGESTION") in new stackJean-YvesOn 15/05/2005, at 8:36 PM, Jean-Yves Avenard wrote:If the IAX channel didn't exist or wasn't connected, it would jump to s-CHANUNAVAIL which jumps to priority 400 and try SIPIn CVS-HEAD it now goes to s-CONGESTION which is definitely not what I want as it goes in this case to voicemails-DIALSTATUS has changed now? is there a documentation on how it should work now? --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___
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Re: [Asterisk-Users] FXO/FXS suggestions:
I don't think you have many choices for this oneOn 16/05/2005, at 9:53 AM, Tim Connolly wrote: Im looking for a zaptel type device with one (or more) FXO and one (or more) FXS port. Basically this guy would sit in-line of your phone line (PCI card). Any suggestions? TDM400 would be overkill.Hum TDM400 ?JY --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Old DBGet/DBPut vs. new Set(var=${DB(...
HelloI upgraded to CVS head yesterday (due to the lack of zaptel drivers working with 2.6.10)And noticed that now DBGet and DBPut have been deprecated in favour of the new Set/DB one.In the UPGRADING.txt in Asterisk it says:* The applications DBGet and DBPut have been deprecated in favor of functions. Here is a table of their replacements: DBGet(foo=family/key) Set(foo=${DB(family/key)}) DBPut(family/key=${foo}) Set(${DB(family/key)}=${foo})I fail to see how DBGet and DBPut can be replaced by those two commandsIf I want to create a new database entry:DBPut(CFIM/200=300)I will create the entry if it doesn't existWith the new Set(${DB(CFIM/200)}=300) I get:May 16 12:39:39 WARNING[1]: func_db.c:54 function_db_read: DB: CFIM/200 not found in database. -- Executing Set("SIP/ipp100-1d45", "=300") in new stack -- Executing Playback("SIP/ipp100-1d45", "auth-thankyou") in new stackas abviously DB(CFIM/200) always get replaced by its value which in this instance doesn't exist yetthe other serious problem is that DBGet used to automatically jump to prioriy n+101 if the entry didn't exist. Now I will do things like:Set(temp=${DB(CFIM/200)})which will set temp to "" instead of jumping to an error.I wish DBGet and DBPut weren't removed their replacements are no good and can't be made to behave the same without serious re-work (like testing the returned entry is not null etc...)Any ideas on work-around or did I miss anything?Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___
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Re: [Asterisk-Users] FXO/FXS suggestions:
Does it work about as well as a PCI FXO interface? I have places where I have already linked facilities with ptp T1 or frame relay. I don't want to build a * server just to tie 1 or 2 pots lines to a * PBX back at the main location. I also have customers who buy dells with service contracts but leave os and all to me. I don't want to add PCI cards to those dells.I have a few SPA-3000 and they work perfectly. I only noticed one issue in France where the SPA-3000 wouldn't notice that the user hanged-up and it would wait for about 5 minutes for the FXO line to be available again..Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FXO/FXS suggestions:
On 16/05/2005, at 1:15 PM, Paul wrote:That would be a major issue if it happened often. My primary use would be to get incoming calls handled by the * pbx. If the ata keeps the line offhook or ignores ring in those 5 minutes we have a big problem. As I said, this only happened on the French telephone network. I do not have this problem here in Australia. Seem to be an issue with the busy tone detection.Note, that I would have hang-up detection failure in many cases, including with the Digium TDM400 card !Going digital / ISDN is the only way to make sure you'll never have any of those problems.Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
HelloOn 15/05/2005, at 12:00 AM, Colin Anderson wrote:wierd. Im running fc2 2.6.8 smp no problems. Could be timing slips on your PRI, happened to me until I looked hard at the PRI So you turned off Hyperthreading on your linux box or not??What could I do to check if my PRI has timing slips? I have to say that my knowledge in this area is close to zero. The extent of my knowledge was to connect a TE110P card, configure it and run it. That's itJean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk - fax - spandsp
Dear allI have a Asterisk server (version 1.0.7) running with a TE110P card on a 3Ghz P4 machine (running nothing else but asterisk)So far so good.Today I was trying to receive faxes using spandsp fax client.I used spandsp v0.0.2pre18 (which compiled against libtiff 3.5.7)It compiled fine, installed fine, run fine.But when sending a fax (from a Brother fax machine) the image received is always corrupted: lines are missing, segments of the faxed page aren't there etc...I could attach an example (it's 40KB) but I wasn't sure that the rules of the asterisk-user DL forbid attachment.In your experience, is this something to expect or should the fax received be complete?Is there any other fax package out there to receive a fax and send it through email?Thank you for your helpJean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk - fax - spandsp
On 14/05/2005, at 3:59 AM, Colin Anderson wrote: Run ZTTEST in /usr/src/zaptel. Your output should be 99.98XXX% or higher. If it isn't the corruption is most likely because of IRQ misses by the card. Run it for a couple of minutes. Your "worst" should be 99.98XXX%. Subtle timing errors and IRQ misses that are not audible will screw up a fax. Also timing errors on your PRI are another cause. You have to ensure your card is recieving timing correctly by the telco. and your span is set up appropriately to reflect that. Ensure the usual things are true: turn off any service you do not absolutely need, disable Hyperthreading in the CPU, turn off any device that you do not need, such as sound card, USB interface, serial ports, parallell ports, everything you can think of is off. I ran zttest:and got:--- Results after 347 passes ---Best: 100.00 -- Worst: 99.987793I would say that 2/3rd of the tests were 100%.I tried using the old stable spandsp library: 0.0.1k, but the results were even worse ; only the first 3 or 4 cm of the received fax were correct.I will try upgrading to a newer kernel (I'm using Fedora Core 2 with 2.6.8 kernel), turn off hyper-threading and see if it makes any difference...Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95722686 | office +61 3 8573 5299 | direct +61 3 8573 5200 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Regex in number dialed
On 26/12/2004, at 12:42 PM, Jean-Yves Avenard wrote:
Hello
Didn't find any information in the wiki. Regex only refers to the
dialing syntax
Thank you all for your answers, it seems to work in most cases...
However I have something like this:
exten = _8[89]XXX,1,Dial,Zap/4/1414${EXTEN:1}
exten = _8[89]XXX,2,Dial,Zap/3/1414${EXTEN:1}
exten = _8[89]XXX,3,Dial(SIP/jya-home-out/${EXTEN:1},60,r)
exten = _8[89]XXX,4,Dial(IAX2/freshtel/${EXTEN:1},60,r)
exten = _8X.,1,Dial(IAX2/freshtel/${EXTEN:1},60,r)
exten = _8X.,2,Congestion
When I dial:
8
It still dial using the IAX2/freshtel
If I replace the whole lot with:
exten = _88XXX,1,Dial,Zap/4/1414${EXTEN:1}
exten = _88XXX,2,Dial,Zap/3/1414${EXTEN:1}
exten = _88XXX,3,Dial(SIP/jya-home-out/${EXTEN:1},60,r)
exten = _88XXX,4,Dial(IAX2/freshtel/${EXTEN:1},60,r)
exten = _89XXX,1,Dial,Zap/4/1414${EXTEN:1}
exten = _89XXX,2,Dial,Zap/3/1414${EXTEN:1}
exten = _89XXX,3,Dial(SIP/jya-home-out/${EXTEN:1},60,r)
exten = _89XXX,4,Dial(IAX2/freshtel/${EXTEN:1},60,r)
exten = _8X.,1,Dial(IAX2/freshtel/${EXTEN:1},60,r)
exten = _8X.,2,Congestion
Then if dialing it will use the Zap interface first...
How could that be?
Jean-Yves
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[Asterisk-Users] Regex in number dialed
Hello Didn't find any information in the wiki. Regex only refers to the dialing syntax I'd like to do something like: exten = _8001133[1-5,7-9]XX.,1,Dial(SIP/france-gateway,60,tr) is there a possibility? right now I've had to enter all possible choice like: exten = _80011331XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011332XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011333XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011334XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011335XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011337XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011338XX.,1,Dial(SIP/france-gateway,60,tr) exten = _80011339XX.,1,Dial(SIP/france-gateway,60,tr) Thank you in advance Cheers Jean-Yves ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Inviting somebody in a conference call
Hello Is there a way once you're inside a conference call to invite an external party to join? Of course I could tell the party what the extension number and password is, but unfortunately, often people are unable to dial the password especially if calling from overseas. I've looked a lot and didn't find a way having to rely on the conference feature of the phone itself which uses way more bandwidth. Thank you Cheers Jean-Yves ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Where to purchase ISDN (BRI) cards in Australia (preferably)
On 20/08/2004, at 5:05 PM, David MacKinnon wrote: I appreciate approval costs, but in this case the card can be had 65EUR new (about $110). The only place I've seen the Fritz advertised in Australia had it at $360. Approval costs money, sure, and it's not a high volume item, but still... There seems a reasonable interest in using them in Australia, but I'd say very few will fork out for it when the price differential is so great. Check Australia Technology Partnership: http://www.austechpartnerships.com/ They are quite cheap and very knowledgeable. Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Alternative SIP phone
Dear all. I've placed an order for several Uniden UIP200 SIP phone to connect to our Asterisk server but it seems that they're not going to be available for another while. The seller recommended the Ipdialog Siptone 2 instead which is a little bit dearer (around $185 vs $145 for the Uniden). The minimal feature I'm looking at are: 1-Good interactivity with Asterisk obviously 2-Dual 100mbit/s switch 3-PoE (Power over Ethernet) 4-Support monitored transfer (e.g. two lines) 5-Cheap ! I liked the Uniden with its 8 programmable buttons, the Ipdialog doesn't seem to have that. Comments are welcome Regards Jean-Yves ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] 2.4.x-SMP vs. 2.6.x-SMP
Hello 2.6 scheduler performs in O(1), it will perform much better in multi-processor environment than the 2.4 series Jean-Yves On 11/08/2004, at 8:00 PM, Bastian Schern wrote: Which Kernel is better for my constellation (Asterisk with SMP, CAPI and ZAPHFC)? Kernel 2.6.x or Kernel 2.4.x? --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)
I posted a comment on how I did it on the exact page you referred to (click on the *comments* link) So yes, after I got the 2.6.7 kernel installed (which is very trouble free once you copied the original fedora .config file) compile, install , reboot the machine then compile asterisk normally. You don't even need to to make linux26 anymore, the makefile just pick it up that you're running on a linux26 kernel No need to creat files, run make with special commands etc... Jean-Yves On 11/08/2004, at 7:23 AM, Oliver wrote: Interesting ... so just the kernel 2.6.7 and make linux26; make install made it work for you? Or did you have to create the files as described at http://voip-info.org/tiki-index.php?page=Linux%20Fedora#comments as well? Thanks, Oliver --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Kernel 2.6 and zaptel data
I'm using linux 2.6.7 (FC2 distribution) with zaptel and a TDM403 card without any problems Jean-Yves On 11/08/2004, at 8:41 AM, [EMAIL PROTECTED] wrote: I saw somewhere that the last kernel to work properly with the zaptel drivers when using data over it was 2.4.20. Has this been since fixed to work with newer kernels? --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fedora FC2 and Zaptel (Torisa)
Or you can install the kernel 2.6.7 and all those little worries disappeared. I don't know what they did in FC2 to get it so wrong with their kernel... Jean-Yves On 10/08/2004, at 5:37 AM, Oliver wrote: I had the same problem ... Changing the linux-2.6 symlink in /usr/src to /lib/modules/2.6.5-1.358 (with kernel-2.6.5-1.358) and building it again with make clean; make linux26 made it work (so the symlink is /usr/src/linux-2.6 - /lib/modules/2.6.5-1.35). Cheers, Oliver --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] G729 or GSM
Dear all. I recently subscribed to a VoIP provider through IAX. The require to connect with either G729 or GSM, I chose G729 based on their recommendation. The service works very well, however ... people mentions how distorted our voice sounds. We have plenty of bandwidth available so I don't think it comes from our side. What it means is that it comes from two things: 1-G729 gives bad voice quality. So will GSM provides a better voice quality than G729? 2-VoIP provider has a screwy setup. I guess I will have to sort out 1 first. Maybe I can convince them to use iLbc instead. Regards Jean-Yves --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Zaptel doesn't see remote hangup ?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 - -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Here is what you can try: Add in zapata.conf: busydetect=yes busycount=6 The maximum it will take for asterisk to see the person hanged-up is after 6 busy dial-tones. Worked for me Jean-Yves On 30/07/2004, at 12:09 PM, Walter Klomp wrote: Thanks Peter, Yes, indeed the problem seems to be exactly what you describe. It's overhere the same. If I dial a mobile number it disconnects immediately when I hangup the mobile. But for analog numbers it takes around 10 seconds or so... Well, at least now I know how to debug pri :-) Walter. - - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 - -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCkUfXeDVKqIr3GURAoUVAKCJ6NMIMR8MFQTXJVD0vvx7foo5twCeNJBR jZCbw9plXMnAafrgp1Lmqs0= =idgL - -END PGP SIGNATURE- -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCk/jXeDVKqIr3GURArbMAJ9542GqnGEDcOsYaXKoVd89HmdMPgCfXzUa wmvaNg7sYJxwOzjF3UrbBzk= =pNum -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP connections do not hang up
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 If you just bothered to search this list in the past 12 hours, you would have found a solution around that: to summarize: Add in zapata.conf: busydetect=yes busycount=6 The maximum it will take for asterisk to see the person hanged-up is after 6 busy dial-tones. On 31/07/2004, at 6:58 AM, Florian Rau wrote: I'm calling from inside (either X-Lite using SIP channel or a ISDN telephone using Zap Channel) using sipgate to a number in public network. When I'm hanging up before the other person picked up the phone, the line is not closed correctly. The phone keeps on ringing until timeout (of Sipgate I assume) and it even costs my money, if the other person picks up the ringing phone, even if I already hung up. - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCsHLXeDVKqIr3GURArjyAJ9p97F/wWIiIesaYo85QfHut8zbzQCgj2l2 uuKZxyJoaSmpI9V9I+ojnJc= =Y8jQ -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VoiceMail Not releasing
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Here we go again On 31/07/2004, at 11:59 AM, [EMAIL PROTECTED] wrote: About twice a week we have a caller that comes in and hangs up on voicemail. We have 2 x100ps each with their own irq. When the caller hangs up asterisk does not release the line. The line rings busy, sometimes I can do a soft hangup Zap/1 and release the line sometimes I have stop asterisk and remove and re-insert the modules. 3 twice the same question/answer in 24 hours! Add in zapata.conf: busydetect=yes busycount=6 - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCxWBXeDVKqIr3GURAt7AAJ0b3f2kizebKQvA41hTvyD4Zi3KFwCffp1T 4fXSMiB0Ypukjsctc4IP3FU= =aibw -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Experience with this online seller?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I'm about to order some few phones from this place: www.thevoipconnection.com Do you guys have any experience with this store? Thank you Regards Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCH3+XeDVKqIr3GURAs4EAJ4zHpqfAWj5ZmHkg6g/prg5ljAkBQCeIxE1 JqYQcuraeBkWICAFnNwvP4k= =DuVi -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some questions on uniden uip200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello For those who currently own an Uniden UIP200 do you know if the power adaptor that come with it works also for 220V ? Thank you Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCH5YXeDVKqIr3GURAn0wAJ9GPy6eHC5grKBfO239RM+u6lfORACfZPmQ y0zqyfFh3TKpQ14Rlqnum4A= =idRm -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some questions on uniden uip200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Great Thank you I'm going to order 10 of those babies.. On 29/07/2004, at 8:46 AM, Ryan Courtnage wrote: Yes, the adapter says Input: 100 - 240V ~, 50/60Hz, 0.26A - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBCIgNXeDVKqIr3GURAnXcAJ9OaKyy6IgGPRGk9hpejx1YKLAWnwCfUmIW iqTGXgVi1LbiEu4ZBnwtJWk= =J6Cx -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Beta version of Grandstream Firmware 1.0.5.9
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 People have mentioned a lot of crashes making the phone unusable with this totally *unofficial* firmware Check the distribution list for more information Jean-Yves On 27/07/2004, at 4:23 PM, Dave Cotton wrote: I just pulled it from http://www.hellofone.com/downloads.html - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBBff2XeDVKqIr3GURAtndAJ4rLXpCjUxT3x9ujFf9n/s4IOtldQCghqoP TwReRmYfTsQc/XlzY6i9QL4= =bFot -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Digium G729 codecs
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dear all Last week I purchased 10 G729 codec licenses from Digium. The only thing I got from them was the invoice. No license file nothing. After chasing them up, I got an other email giving me something that looks like this: asteriskpbx-600x:G729-xx they told me a README attachment file was provided but there was nothing. Is that all I need? i looked on google, wiki to no available.. If yes, how can I install this? Regards Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBBaEAXeDVKqIr3GURAsIMAJwOKKvFmGwiwWFshECracsBYTIS3QCfSMQe 7MxmGKgCyIJlNYXQ5jXsT5U= =G5mX -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium G729 codecs
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Great. All that I needed. Thank you very much, got it up and running without any problems. Regards Jean-Yves On 27/07/2004, at 10:31 AM, Adam Hart wrote: just send me your key and I'll help :p just kidding try ftp://ftp.digium.com/pub/asterisk/g729/ the README and the needed files are in there - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFBBcXWXeDVKqIr3GURAhZ4AJ9ZQlxISADHLVce4KCxCMFSOugPHwCfdBYz kHwP9Z0ZHOu15rO/GiJomc4= =bJnE -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Adding voice mail box
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 There's a script to create the mailbox with the asterisk source code: contrib/scripts/addmailbox On 19/07/2004, at 8:17 PM, Steve Hanselman wrote: They only get created as they are used and voicemail left, try leaving a message and you should see that the structure etc is created. - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA+6RqXeDVKqIr3GURAvVtAJ4s5Bc658HW6HDC/uehlhB95ZDyjwCbBHrw WlKE8cQzBrqzHY7XXVYEgj4= =BPGt -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and zaptel on Fedora Core 2
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dear all. As I couldn't get to compile and run Asterisk 1.0RC1 on my default RedHat 9 I thought it was about time to upgrade to Fedora Core 2. Well, it was too late to realize the kernel 2.6 wasn't supported by Asterisk *officially* anyway. Here is what I did to get asterisk and zaptel to work under Fedora Core 2: I posted it on the wiki and here is an extract Getting asterisk to work on fedora core 2 is no problem. But getting zaptel to work is another issue. The kernel (2.6.5) source code provided with Fedora Core 2 is missing some auto-generated components. I found that the easiest way to get around all those issues was to download a new kernel source code like 2.6.7 from www.kernel.org. Here is the procedure: 1-Grab the 2.6.7 kernel source code and untar it (do not untar it in /usr/src, this is a very bad practice) 2-Copy the .config file from the default /usr/src/linux-2.6.5-1.358 into the 2.6.7 source code directory. 3-type; make menuconfig and make the necessary change for your hardware configuration. You could just leave it as it is as the default Fedora Core 2 contains everything. But having so much stuff in means much longer compilation time! Quit and save the .config file 4-Compile and install your kernel as describe there: http://www.digitalhermit.com/linux/Kernel-Build-HOWTO.html 5-Create a link linux-2.6 to your 2.6.7 linux kernel directory in /usr/src; something like: ln -s /data/work/src/linux-2.6.7 /usr/src/linux-2.6 6-Reboot with the new kernel 7-Get the latest asterisk, libpri and zaptel source code from the digium CVS directory 8-Go into the zaptel directory and type: make clean make linux26 make install make config 9-Edit the file /etc/init.d/zaptel and replace all: insmod with modprobe and rmmod with modprobe -r That's it. Make sure it works by starting the script /etc/init.d/zaptel start doing lsmod should show the wcfxs and zaptel module being installed. then install and run asterisk as usual. Hope all of this help Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA+o0ZXeDVKqIr3GURAtHeAJsHbLo6Ty6TKNrhoFF7uvkSUuR9XQCeJ9Lr tmITAmxkoVHkVIS/uxOsEPw= =qJZf -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] chan_capi won't compile
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello On 19/07/2004, at 9:08 AM, Thor Atle Rustad wrote: I am trying to compile chan_capi 3.3.4a, but I end up with lots of gibberish. Near the top it states that capi20.h doesn't exist. Searching for the file, several show up: Make sure that you've created a link from /usr/src/linux-2.4.21 to /usr/src/linux ln -s /usr/src/linux-2.4.21 /usr/src/linux then recompile asterisk Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA+w0hXeDVKqIr3GURAoupAJ48PcOTSr+/Sq9a8KPhN06s27PdEQCdEXpr jUh01lQeb6H5v5MVRJVFp7Y= =NlKe -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-1.0 RC1
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello Mark. thank you heaps for this. I also just updated our CVS source copy and it doesn't compile anymore it stops in the pbx directory with hundreds of GTK errors. I'm running RedHat 9.0 Linux (with 2.4.26 kernel) Now, will download the files from digium ftp server Regards Jean-Yves On 17/07/2004, at 4:17 PM, Mark Spencer wrote: ftp://ftp.digium.com/pub/asterisk - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA+MZtXeDVKqIr3GURApOsAKCOkTt5gxAeqPL9W11ia3HG+DgolgCfRoL4 tvsvFzU/X1/g4mkbKCvlqu4= =c025 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk-1.0 RC1
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello On 17/07/2004, at 4:17 PM, Mark Spencer wrote: ftp://ftp.digium.com/pub/asterisk Can someone grab a copy and put it on a mirror server? Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA+MdtXeDVKqIr3GURAjdIAJsEP4bI27MBUL+dnxDNvhRlG0XUGgCfTZXn w+wEs/8DoKnL1jgv6s4cB5Q= =Ks0x -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some questions on uniden uip200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello On 16/07/2004, at 7:02 PM, Jan Goericke wrote: yesterday the uniden uip200 phone was recommended to someone. i am looking for an alternative to grandstream bt-100 because i can not do a supervised tranfer with it. here my questions: With the BT100 you can always use call parking to achieve something similar, sure it's not the most elegant way.. Regarding the UIP200, I couldn't find any distributors for it in Australia, and even in the US it seems to be very hard to find, at least on the net. Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA9578XeDVKqIr3GURArb4AJsHIBcOhXwMkaFwAyS9mCaMlz/uhgCeIDNk FjLhgE0pt5dIBKPQCjB9fT4= =+0P5 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] some questions on uniden uip200
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 16/07/2004, at 7:55 PM, Holger Schurig wrote: What is supervised transfer? Basically you receive a call that you want to transfer ; but before you transfer it you want to speak to the final person once introduced then you transfer. Did I get this right ? :) Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA963MXeDVKqIr3GURArJFAKCKZRH01/YOBrLRQ3FcMk/hBviMvQCfeDb5 IFHYVfbzCgoWvf02Q8l/WMc= =MZ59 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Pressing digits on SNOM phone results in letters on display
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 You should read the manual of your snom then (available from the internal website on the snom phone) Basically you press the a-1 soft-menu button at the top. Jean-Yves On 17/07/2004, at 1:23 PM, Rana Dutt wrote: My SNOM 200 phone got into a funny mode where if I dial any digit, a letter gets displayed and sent, so dialing no longer works. For example, if I dial 9, the letter w gets displayed and sent when I press OK. How do I get it out of this mode? - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA+La9XeDVKqIr3GURAoBUAJ41JCjqT4GYPn7wOx+p3uZWyq1OAwCfSRoe ds7QjBwnDIQmRQe8ptYv+Rs= =bm5i -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP phones recommendations
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dear all. We are currently using either Grandstream BT100 phones or SNOM 200. The BT100 comes with a 10mbit ethernet port and the snom with 2x100mbit port Problem with the SNOM is that they are expensive and I don't really like their design: often the handset slightly move on its base and it makes the whole thing unreliable: poor mechanical design in my opinion. Problem with the BT is that you need either a little switch to connect it or a spare ethernet port: very annoying and it needs far too many cables. Could you recommend a nice SIP phones (which works with * obviously) cheap, well-featured and reliable that comes with 2 x 100mbit port (so no need for a switch or an additional ethernet port). Can be powered over the Ethernet port so it's really cables-free Thank you Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA9jaLXeDVKqIr3GURApC8AJ0XiA1IyIiZlHCrCwtLriUwF9Kb2wCfXS08 0nafjBjZDIB/S8FybYv6YlE= =UsoD -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk on FreeBSD 4.10 dies
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello On 12/07/2004, at 4:24 AM, Arjan wrote: 43676 root63 0 10244K 7628K RUN 2:44 99.05% 99.02% asterisk This is covered in the asterisk FreeBSD section: http://www.voip-info.org/tiki-index.php?page=Asterisk+FreeBSD extract: CPU 99.9 % used by Asterisk? The current version runs amok on a FreeBSD system, occuping all your CPU cycles. To get Asterisk back to a normal level, you have to disable the problemativ module in Asterisk config modules.conf with this statement: noload = pbx_wilcalu.so In any case, I gave up using Asterisk with FreeBSD too many issues that couldn't be explained. Switching to linux fixed all the issues with the exact same configuration file Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA8cVmXeDVKqIr3GURAiDBAJ4yLySDKD8NoozveF8eIHD+jRWtuACeIf1M DyckWYJeN9rpjbfvxGZzQMk= =O3eG -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to differentiate a *busy* call from not available?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello On 10/07/2004, at 5:45 AM, Soren Rathje wrote: Based on extensions.conf.sample from CVS-HEAD... Thank you very much for this information ; for some reason I can't seem to get other version newer than June,29th As a side note, it's pretty amazing that people complain I could ask question on a new feature that was added only a few weeks ago... Anyway.. I got something working now thanks to your information, as well as with call forwarding. There's just what thing I can't figure out. What is the action for s-. Thank you Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA79HUXeDVKqIr3GURAgFSAKCBN2m5lwHOcVyvzFEZJyoz8GPNhACdES1X oYahc18y7RbK/wjeAHENYvc= =XxQB -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to differentiate a *busy* call from not available?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello I'm trying to find a way to differentiate wether a SIP extension is currently busy (e.g. on the phone) or not registered. So i do something like: exten = 100,1,Dial(SIP/foo,20,tr) exten = 100,2,VoiceMail,u100 exten = 100,102,VoiceMail,b100 If the phone doesn't answer I get the message: User is not available if the phone is currently in used i get the message: User is on the phone But if the phone is unplugged, I also get the message: User is on the phone! Any ideas? Thank you Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA7p9gXeDVKqIr3GURAocmAJsFpy4rlj/wsgNxoGgE+WmzfisMFgCfc8z/ WX6zBH+pg2M25phavis0CYY= =ZxKl -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to differentiate incoming calls with grandstream phone
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello
On 06/07/2004, at 10:08 PM, Andrew Yager wrote:
How you go about differentiating - that's a bit harder - but you could
set the caller id in your incoming context, and set it when you
transfer to a specific extension from the local context... or use a
goto statement based on the ${CHANNEL} variable... I haven't got this
working.
Also - try setting callerid = Name number in the sip.conf file for
each of the phones.
That worked perfectly.
Thank you for this.
Jean-Yves
- ---
Jean-Yves Avenard
Hydrix Pty Ltd - Embedding the net
www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724
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Re: [Asterisk-Users] Unreliable dtmf digit generation from tdm400p
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've had this issue too with a TDM4 cards with 3 FXS modules Sometimes I dial a number, and it gets somewhere else. Apologies to the person I called, press redial, hope we go. Quite annoying, it doesn't seem to happen very often though Jean-Yves On 07/07/2004, at 9:39 PM, Andrew Yager wrote: I'm not having this problem on either of my TDM400 cards with a mix of FXO and FXS modules - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA6/+8XeDVKqIr3GURApo6AJ45/sHGSxsu7WZv9kU6B+rXbHhCcgCffRiZ /5bRz3sRlNTdjKFAINbUxQE= =WSH3 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM FXO port remains offhook
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hello On 08/07/2004, at 12:01 AM, Rich Adamson wrote: Not having any such problems here with TDM card (CVS-HEAD-07/01/04). Earlier code had various issues, however the code (and card) have been very stable. I guess I've been unlucky with my cards ; in the past 2 weeks it has happened twice: asterisk thinks that both FXS port are offhook. restarting asterisk fix the problem. I also have the issue of wrong DTMF being sent Jean-Yves - --- Jean-Yves Avenard Hydrix Pty Ltd - Embedding the net www.hydrix.com | fax +61 3 95093717 | phone +61 3 9509 3724 -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA7AAXXeDVKqIr3GURAqN3AJ9oONOPVcQcgQ32EJNFJRosOnbSZwCfS0ir UMFlnWyi1iiTtQxOeOUB8gI= =3ym7 -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
