[asterisk-users] Escape characters or replace function
Hello,
I need to use the ${DATETIME} macro inside the filename saved by Record,
but the colons (':') used in the time interfere with the command
(everything after the colon is interpreted as the format I wish to save to):
My command is:
Record(/path/to/voicemail/${EXTEN}-${DATETIME}-${UNIQUEID}:wav)
I need some function to escape the colons inside DATETIME... I'm sure
it's something really basic, but I couldn't find it
Daniel
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Re: [asterisk-users] Discover connected Zap lines
2008/5/2 Tzafrir Cohen [EMAIL PROTECTED]: On Fri, May 02, 2008 at 09:06:01AM +0200, Vinz486 wrote: 2008/4/30 Tzafrir Cohen [EMAIL PROTECTED]: On Wed, Apr 30, 2008 at 09:07:48PM +0200, Vinz486 wrote: - [May 2 08:51:00] WARNING[5119]: chan_zap.c:6685 handle_init_event: Detected alarm on channel 3: No Alarm [May 2 08:51:03] NOTICE[5119]: chan_zap.c:6678 handle_init_event: Alarm cleared on channel 4 - This means that you should be able to see it in the InAlarm: field in 'zap show channel 3' Ok. Made some experiments. InAlarm field show 1 if cable unplugged *BUT* only if in previouos time cable was plugged. In few words, at boot, InAlarm is 0, Cable plugged: 0, Cable unplugged: 1 If i use this field, after a boot without cable, my software will think that the cable is plugged. I found another useful field: Hookstate (FXS only). It tell me if the cable is plugged ever after a boot without cable. Hookstate (FXS only): Offhook --Cable plugged Hookstate (FXS only): Onhook --Cable unplugged I hope this can help other people (and make to think at Zaptel developing to insert a field exactly for this purpose, eg: Cable: plugged or Cable: unplugged). Bye. -- PicoStreamer - the real WEB live streaming software vinz486.com ___ -- 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] Escape characters or replace function
On Monday 12 May 2008 01:36:04 Daniel Grad wrote:
I need to use the ${DATETIME} macro inside the filename saved by Record,
but the colons (':') used in the time interfere with the command
(everything after the colon is interpreted as the format I wish to save
to):
My command is:
Record(/path/to/voicemail/${EXTEN}-${DATETIME}-${UNIQUEID}:wav)
I need some function to escape the colons inside DATETIME... I'm sure
it's something really basic, but I couldn't find it
Use ${STRFTIME()} instead to get the format that you want and do not
include colons in the format. core show function STRFTIME should give
you the details on this dialplan function.
--
Tilghman
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Re: [asterisk-users] Dialplan Visualization (Extensions.conf orDialplan Show)
Hello list, I've done some work with basic parsing of extensions.conf in order to generate some visualizations of the dialplan. I've just posted it this past weekend over on the Asterisk-Java blog at asterisk-java.org. There's a Java web start demo if you have your extensions.conf handy. Cheers, =Martin Smith, Systems Developer [EMAIL PROTECTED] Bureau of Economic and Business Research University of Florida (352) 392-0171 Ext. 221 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Gibson Sent: Friday, April 18, 2008 4:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Dialplan Visualization (Extensions.conf orDialplan Show) Hello, About 4 years ago there used to be a script floating around to generate dynamic graphs/diagrams of extensions.conf (the asterisk dialplan). It was using GraphViz to perform the graphing. Does anyone have a copy of this script, or a better solution to generate a flowchart of my dialplan? Thanks, Matt ___ -- 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] Escape characters or replace function
Tilghman Lesher wrote:
On Monday 12 May 2008 01:36:04 Daniel Grad wrote:
I need to use the ${DATETIME} macro inside the filename saved by Record,
but the colons (':') used in the time interfere with the command
(everything after the colon is interpreted as the format I wish to save
to):
My command is:
Record(/path/to/voicemail/${EXTEN}-${DATETIME}-${UNIQUEID}:wav)
I need some function to escape the colons inside DATETIME... I'm sure
it's something really basic, but I couldn't find it
Use ${STRFTIME()} instead to get the format that you want and do not
include colons in the format. core show function STRFTIME should give
you the details on this dialplan function.
Found STRFTIME just after I posted the message.
I tried ${STRFTIME(${EPOCH}, %d%m%Y-%H\:%M\:%S)} but it returned an
empty string (I wanted to get the same output as DATETIME but with
escaped colons. What would have been the correct syntax? I finally used
just ${EPOCH} instead, but I'm curious how I could have escaped the colons.
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[asterisk-users] test message please do not reply and clog up the list
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Re: [asterisk-users] G.722 for polycom
zhao_x_q wrote: I have test G.722 for many phones. I have try calls between sip G.722, sip G.722 to sip G.711, G.722 to RRI cards, PRIcards to G.722. I also test meetme conference. Other phones such as grandstream and fanwei have no problems. The sounds is good, grandstreams have little difference between G.711 and G.722. But Polycom's IP 550 have many problems. Polycom's G.722 to TE210E1 have problems the sound is choppy. Polycom's G.722 to conference also have problems, I even cannot heard the sounds. Has any friend knows the reasons for that? What version of Asterisk are you using? I have made a lot of G.722 related fixes over the last few months. -- Russell Bryant Senior Software Engineer Open Source Team Lead Digium, Inc. ___ -- 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] module reload question
Dear all, I have installed asterisk 1.4.13 and configured all the /etc/asterisk files very well. Always I enter the CLI (with asterisk -r) and when I make a change after that I execute module reload and everything is OK. But a few days ago, without make any change, I execute module reload from within CLI and the terminal turn into black color and the color of the letters was white (exactly the opposite to the normal colors). I think because I get some warning and notice message: [May 12 10:19:10] NOTICE[6265]: cdr.c:1362 do_reload: CDR simple logging enabled. [May 12 10:19:10] NOTICE[6265]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'us' [May 12 10:19:10] WARNING[6265]: res_smdi.c:746 reload: No SMDI interfaces were specified to listen on, not starting SDMI listener. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4090 pbx_load_module: Starting AEL load process. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4097 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 112, Cols: 34-34: Warning! The empty context ael-dundi-e164-canonical will be IGNORED! -- Reloading module 'codec_gsm.so' (GSM Coder/Decoder) [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 120, Cols: 34-34: Warning! The empty context ael-dundi-e164-customers will be IGNORED! [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 128, Cols: 33-33: Warning! The empty context ael-dundi-e164-via-pstn will be IGNORED! [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4105 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-canonical' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-customers' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-via-pstn' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 276-283: The included context 'ael-parkedcalls' cannot be found. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4108 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4110 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4113 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-customers' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-via-pstn' [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4116 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. After that I test the system and it work OK. What can be the problem ??? Is it a normal situation ??? Thanks a lot. Alejandro ___ -- 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] Lone worker system
Has anybody got any scripts for a lone worker system using Asterisk before I write them? Something along the lines of a regular phonecall with some kind of random question (e.g. press 1 then 5) to provide monitoring of lone workers with alerts? Steve The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk___ -- 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] Sangoma and Voicetronix cards
I've been having a look at some of the information on the website of Voicetronix in Australia, and see that their cards make use of wanpipe and wanrouter. I already knew that Sangoma cards also make use of those, so my question is: what is the relationship, if any, between Voicetronix and Sangoma cards? Or are wanpipe and wanrouter generic software components that both companies just happen to use? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ -- 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] Polycom causes conference to fail
Sorry to be a pest, but does anyone have any ideas on this? I've opened a bug, but I was hoping someone else on the list has encountered this issue before. Thanks, Jason On May 9, 2008, at 12:36 PM, Jason Dixon wrote: We have a remote office that's having problems with their Polycom. Sometime after they start a conference, the audio will halt and the Polycom will become unresponsive. The only recourse is to kill the Polycom meetme. Symptoms include a flood of RTP packets from the Asterisk server to the Polycom, a loss of audio for all participants, and the Polycom console becomes frozen. It appears to be isolated to this particular device; we routinely have conference bridges with other offices and Polycoms without issue. Considering we have other Polycoms (same model) operating successfully in bridges, I'm hesitant to put all of the blame on an Asterisk bug. But I guess it couldn't hurt, worst case is they smack me down and tell me what we fudged up. :) For the sake of curiosity (if anyone is), here is the channel information for the Polycom while it's in the frozen state. Just below that is the output from kicking it. pbx*CLI core show channel SIP/seattleconference-08a1fc68 -- General -- Name: SIP/seattleconference-08a1fc68 Type: SIP UniqueID: 1210346914.429 Caller ID: 293 Caller ID Name: Conference DNID Digits: 7000 State: Up (6) Rings: 0 NativeFormats: 0x4 (ulaw) WriteFormat: 0x40 (slin) ReadFormat: 0x40 (slin) WriteTranscode: Yes ReadTranscode: Yes 1st File Descriptor: 62 Frames in: 12330 Frames out: 21899 Time to Hangup: 0 Elapsed Time: 0h7m23s Direct Bridge: none Indirect Bridge: none -- PBX -- Context: internal Extension: 7000 Priority: 1 Call Group: 0 Pickup Group: 0 Application: MeetMe Data: 642696|aciAsdpr| Blocking in: ast_waitfor_nandfds Variables: MEETME_RECORDINGFILE=conf-recordings/642696-160 AstVar=0 [EMAIL PROTECTED] SIPUSERAGENT=PolycomSoundStationIP-SSIP_4000-UA/2.0.3.0127 SIPDOMAIN=192.168.100.1 SIPURI=sip:[EMAIL PROTECTED] CDR Variables: level 1: clid=Conference 293 level 1: src=293 level 1: dst=7000 level 1: dcontext=internal level 1: channel=SIP/seattleconference-08a1fc68 level 1: lastapp=MeetMe level 1: lastdata=642696|aciAsdpr| level 1: start=2008-05-09 11:28:34 level 1: answer=2008-05-09 11:28:39 level 1: end=2008-05-09 11:28:39 level 1: duration=0 level 1: billsec=0 level 1: disposition=ANSWERED level 1: amaflags=DOCUMENTATION level 1: uniqueid=1210346914.429 pbx*CLI meetme kick 642696 all 1 pbx*CLI meetme kick 642696 1 -- SIP/seattleconference-08a1fc68 Playing 'conf- kicked' (language 'en') -- Hungup 'Zap/pseudo-1440941539' -- Hungup 'Zap/pseudo-47320381' == Spawn extension (internal, 7000, 1) exited non-zero on 'SIP/ seattleconference-08a1fc68' --- Jason Dixon OmniTI Computer Consulting, Inc. [EMAIL PROTECTED] 443.325.1357 x.241 ___ -- 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] Lone worker system
On Mon, May 12, 2008 at 10:28 AM, Steve Hanselman [EMAIL PROTECTED] wrote: Has anybody got any scripts for a lone worker system using Asterisk before I write them? Something along the lines of a regular phonecall with some kind of random question (e.g. press 1 then 5) to provide monitoring of lone workers with alerts? Steve I think a little more elaboration would get you more helpful advice. I have read your message a couple of times and still don't really understand what you need. Thanks, Steve Totaro ___ -- 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] x100p card or similar in India
Hello All, Anyone purchased a asterisk card, x100p or similar in India, if yes from where and what model ? I am interested in setting up a Asterisk Server at home, for single line at the moment and if things work out great, I would like to migrate that to my business and replace the aging pbx solution. Thankx, Amit. ___ -- 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] x100p card or similar in India
We have been using Sangoma A200 for about an year now with BSNL connection. I don't know if you can get it in India directly as in our case it was brought from US directly. Regards, Sanjay Rajdev - Original Message - From: Amit Patel [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, May 12, 2008 8:12:37 PM GMT +05:30 Chennai, Kolkata, Mumbai, New Delhi Subject: Re: [asterisk-users] x100p card or similar in India Hello All, Anyone purchased a asterisk card, x100p or similar in India, if yes from where and what model ? I am interested in setting up a Asterisk Server at home, for single line at the moment and if things work out great, I would like to migrate that to my business and replace the aging pbx solution. Thankx, Amit. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sangoma and Voicetronix cards
On Mon, May 12, 2008 at 10:36 AM, Tony Mountifield [EMAIL PROTECTED] wrote: I've been having a look at some of the information on the website of Voicetronix in Australia, and see that their cards make use of wanpipe and wanrouter. I already knew that Sangoma cards also make use of those, so my question is: what is the relationship, if any, between Voicetronix and Sangoma cards? Or are wanpipe and wanrouter generic software components that both companies just happen to use? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org Disclaimer: This is just what I have come to believe through deduction. It may not be factual. Wanpipe and wanrouter are from Sangoma I believe (99.9% sure). I think this is similar to BRIstuff working with many different vendor's hardware. Xorcom makes use of it as well as Junghanns (who handle the original BRIstuff). Xorcom has a version with more tools, I am not aware of any code differences) I believe the bottom line is a compatible chipset that allows drivers and other software to work with different vendors. Thanks, Steve Totaro ___ -- 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] Lone worker system
He wants a randomly generated phone call to be generated to a specific extension. Eg once an hour for the midnight to dawn shift at a random time per hour. When the person picks up they are asked a question using an audio file. (or text to speech). Then the person has to enter the correct dtmf answering the question (eg 1 - 5) If the person fails to answer the phone (I'm guessing here but a second call will be placed 2 mins later). If this call is also 'fail to answer' an escalation call to a supervisor or something similar will occur indicating that the 'lone worker' failed to respond and is either - dead from a stabbing, or 2 jerking off in the bathroom and not at his post. Cheers, Dean P.S. Hope your lone worker is paid a lot to be working for a shitty company checking up on them like that :) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, May 12, 2008 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lone worker system On Mon, May 12, 2008 at 10:28 AM, Steve Hanselman [EMAIL PROTECTED] wrote: Has anybody got any scripts for a lone worker system using Asterisk before I write them? Something along the lines of a regular phonecall with some kind of random question (e.g. press 1 then 5) to provide monitoring of lone workers with alerts? Steve I think a little more elaboration would get you more helpful advice. I have read your message a couple of times and still don't really understand what you need. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] exten = pattern match query
Hi, I read the WiKi, which implied there was a way of working around this, but the HTML nature of the WiKi seems to have destroyed some of the output so I cannot see the correct answer... I would like to match a special case of a number dialled 0x, now normally I would simply do: exten = _0x.,1,NoOp(Got Hex dialling) But the X pattern match is case-insensitive, so the above pattern will match any 3 or more digit number starting with a zero. I suspect that the answer may be: exten = _0[x].,1,NoOp(Got Hex dialling) Could someone confirm this for me? Thanks, Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Zap Channels Collide (Incoming Outgoing)
Eric Wieling wrote: People that try to wing it and install Asterisk when they don't know telecom just gives people a bad impression of Asterisk and VoIP in general. This helps nobody except the pocketbook of the consultant. I agree. But I think that comment is incredibly funny. I'd like to re-write it for about 20 years ago... (and some even today) People that try to wing it and install Networks when they don't know networking just gives people a bad impression of Servers and Computers in general. People = Telco Guys Oh, yes. I saw an entire Cat 5 network on punch blocks one time! Everybody needs to learn the other side before getting involved. Bill ___ -- 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] Lone worker system
Spot on (except for the shitty way, it's pretty standard, in building there are paging systems that start an escalating tone, beyond the building these don't work, so if you're away then we'd be dialling the mobile). Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: 12 May 2008 16:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lone worker system He wants a randomly generated phone call to be generated to a specific extension. Eg once an hour for the midnight to dawn shift at a random time per hour. When the person picks up they are asked a question using an audio file. (or text to speech). Then the person has to enter the correct dtmf answering the question (eg 1 - 5) If the person fails to answer the phone (I'm guessing here but a second call will be placed 2 mins later). If this call is also 'fail to answer' an escalation call to a supervisor or something similar will occur indicating that the 'lone worker' failed to respond and is either - dead from a stabbing, or 2 jerking off in the bathroom and not at his post. Cheers, Dean P.S. Hope your lone worker is paid a lot to be working for a shitty company checking up on them like that :) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, May 12, 2008 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lone worker system On Mon, May 12, 2008 at 10:28 AM, Steve Hanselman [EMAIL PROTECTED] wrote: Has anybody got any scripts for a lone worker system using Asterisk before I write them? Something along the lines of a regular phonecall with some kind of random question (e.g. press 1 then 5) to provide monitoring of lone workers with alerts? Steve I think a little more elaboration would get you more helpful advice. I have read your message a couple of times and still don't really understand what you need. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ -- 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] 3U server chassis Digium TE405P?
Gentlemen, First let me say it's great to be back on the Asterisk mailing lists. Those of you who have been around for a while will remember me as Rushowr. I look forward to answering questions and whatnot in the future, but for the moment I have a minor question that I cannot find a definitive answer for online. I am in possession of a Digium TE405P card which I _know_ will fit in a 4U chassis, but we are building a new server and cannot get a 4U from the supplier that my current client wants to use. However, we can get a 3U chassis. My question is, will this card fit? Does anyone out there have a 405 out there that they have installed in a 3U? Thanks in advance for any help that can be offered, Sherwood McGowan VoIP / Telecom Solutions Consultant ___ -- 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] Polycom causes conference to fail
Sorry to be a pest, but does anyone have any ideas on this? I've opened a bug, but I was hoping someone else on the list has encountered this issue before. Jason Does the Polycom have the Buddy List turned on? We had an IP601 that would reboot (or lock up) about 60% of the time when IT started a Page All (which is nothing but a meetme). All of the other phones would send their status for the Buddy List to the IP601 and would simply overwhelm it - causing a reboot. Just a shot in the dark... Bill ___ -- 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] Discover connected Zap lines
So, do you mean that if : 1. Asterisk server boots, 2. A cable from telco analog line is plugged in and out in every FXO port 3. Analog lines (from Telco) are plugged into definitive FXO ports 4. Then, any query to InAlarm field would tell if a cable is plugged or not ? ___ -- 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] Using multiple variables in SIP.CONF setvar
Hi, What is the syntax to set more than one variable in the SIP.conf file for a particular sip peer? (using the setvar line) Regards, Mick ___ -- 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] Discover connected Zap lines
2008/5/12 Olivier [EMAIL PROTECTED]: So, do you mean that if : 1. Asterisk server boots, 2. A cable from telco analog line is plugged in and out in every FXO port 3. Analog lines (from Telco) are plugged into definitive FXO ports 4. Then, any query to InAlarm field would tell if a cable is plugged or not ? Yes. In a simple manner: InAlarm is right only when a cable is plugged at least once. Otherwise will tell you that cable is plugged (0 alarm). -- PicoStreamer - the real WEB live streaming software vinz486.com ___ -- 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] module reload CLI Asterisk question
Dear all, I have installed asterisk 1.4.13 and configured all the /etc/asterisk files very well. Always I enter the CLI (with asterisk -r) and when I make a change after that I execute module reload and everything is OK. But a few days ago, without make any change, I execute module reload from within CLI and the terminal turn into black color and the color of the letters was white (exactly the opposite to the normal colors). I think because I get some warning and notice message like these: [[May 12 10:19:10] NOTICE[6265]: cdr.c:1362 do_reload: CDR simple logging enabled. [May 12 10:19:10] NOTICE[6265]: indications.c:505 ast_unregister_indication_country: Removed default indication country 'us' [May 12 10:19:10] WARNING[6265]: res_smdi.c:746 reload: No SMDI interfaces were specified to listen on, not starting SDMI listener. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4090 pbx_load_module: Starting AEL load process. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4097 pbx_load_module: AEL load process: calculated config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 112, Cols: 34-34: Warning! The empty context ael-dundi-e164-canonical will be IGNORED! -- Reloading module 'codec_gsm.so' (GSM Coder/Decoder) [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 120, Cols: 34-34: Warning! The empty context ael-dundi-e164-customers will be IGNORED! [May 12 10:19:10] WARNING[6265]: ael.y:205 ael_yyparse: File: /etc/asterisk/extensions.ael, Line 128, Cols: 33-33: Warning! The empty context ael-dundi-e164-via-pstn will be IGNORED! [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4105 pbx_load_module: AEL load process: parsed config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-canonical' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-customers' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 141-145: The included context 'ael-dundi-e164-via-pstn' cannot be found. [May 12 10:19:10] WARNING[6265]: pbx_ael.c:838 check_includes: Warning: file /etc/asterisk/extensions.ael, line 276-283: The included context 'ael-parkedcalls' cannot be found. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4108 pbx_load_module: AEL load process: checked config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4110 pbx_load_module: AEL load process: compiled config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4113 pbx_load_module: AEL load process: merged config file name '/etc/asterisk/extensions.ael'. [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-local' tries includes nonexistent context 'ael-parkedcalls' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-canonical' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-customers' [May 12 10:19:10] WARNING[6265]: pbx.c:6246 ast_context_verify_includes: Context 'ael-dundi-e164-local' tries includes nonexistent context 'ael-dundi-e164-via-pstn' [May 12 10:19:10] NOTICE[6265]: pbx_ael.c:4116 pbx_load_module: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. After that I test the system and it work OK. What can be the problem ??? Is it a normal situation ??? Thanks a lot. Alejandro ___ -- 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] externip not working...
I have an Asterisk 1.4.19.1 server that is behind a Fortinet firewall. Localnet is 192.168.2.0/255.255.255.0 and all external sip devices look as if they are on the same local network because the Fortinet rewrites the incoming IP as its own address. The problem I have is that when I set externip=148.XXX.XXX.XXX it is being ignored and I can see SDP packets that have the internal network number of the server (192.168.2.24). Since the SPA3102 has symmetric RTP it simply ignores the internal IP address and sends the rtp back to the same address it received it from they work fine. The problem is the PAP2T that receives the internal address and tries to send rtp there and obviously fails. I have nat=yes and canreinvite=no on all devices, both on the global sip.conf and on each device. If I set externhost and remove the localnet I can get the PAP2T to work but the SPA3102 goes crazy. This is a NAT issue. This is the first time I deal with a firewall that rewrites the incoming address so I am guessing that may be the problem. Is there a work around for this situation? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ -- 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] Anyone Know How to Have Asterisk Work Like GranCentral and Require a Touch-Tone to Connect?
srv04*CLI show application Dial srv04*CLI -= Info about application 'Dial' =- [Synopsis] Place a call and connect to the current channel *SNIP* p- This option enables screening mode. This is basically Privacy mode without memory. P([x]) - Enable privacy mode. Use 'x' as the family/key in the database if it is provided. The current extension is used if a database family/key is not specified. n- This option is a modifier for the screen/privacy mode. It specifies that no introductions are to be saved in the priv-callerintros directory. N- This option is a modifier for the screen/privacy mode. It specifies that if callerID is present, do not screen the call. On Sun, May 11, 2008 at 12:24 PM, Robert DeVries [EMAIL PROTECTED] wrote: GrandCentral has a feature where when you call the GrandCentral number it can ring multiple phones. However, it's not the first phone to answer that gets connected, but the first phone to answer AND play a touch-tone after hearing a recording. The advantage of this is that if one of the called phones has voicemail, it won't get connected to the calling party because the VM won't send a touch tone in response to the recording, unlike a live person. I have always resisted implementing a multiple ring scenario with Asterisk that included a cellphone because of the voicemail answering problem, but this seems to be a solution. Anyone know how to implement it with Asterisk? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Problem with SIP Subscription Status
Hello All, I've been having some intermittent trouble with an Asterisk 1.2.10 installation that is supporting roughly 50 SIP clients on a LAN, mostly soft phones and about 10 snom VoIP phones. We have a custom soft phone client which displays presence information for various extensions. Unfortunately, this information regularly gets out of sync with the actual status of the various extensions. An extension will show up as 'InUse' or 'Unavailable' when the individual is in fact not on the line, i.e. the status should be 'Idle'. I can verify this by issuing a 'sip show subscriptions' and typically for every client subscribed to the problem extension the status column displays 'InUse'/'Unavailable'. However I have also noticed that in some instances, half the subscribed clients will get an 'Idle' status and the other half will have 'InUse' or 'Unavailable'. Often this behavior will follow some other SNAFU, e.g. a rogue mpg123 process for MoH consuming abnormal amounts of CPU and creating high loads. However, this is not always the case. The lack of a pattern, and more then anything, a simple solution has forced my hand and I'm appealing to the list for help. I've been over voip-info countless times and have searched around more then I care to remember. Restarting the soft phones does nothing to alleviate the problem (which doesn't entirely surprise me as I'm starting to think it's a server side issue). Restarting asterisk itself generally resolves things however this is not an option in the middle of the day. Thanks in advance for all your help. -- Zach Segal ___ -- 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] Best Linux distribution to use in Asterisk server
Tzafrir Cohen wrote: On Fri, May 09, 2008 at 11:50:59AM -0400, Drew Gibson wrote: equis software wrote: Hi, I allways use Gentoo y my Asterisk servers and work well, but what do you think about to use Ubuntu or another distibution?? Thanks I have run Asterisk on several Fedora versions, Debian, Unslung (on the NSLU2 or Slug) and recently Ubuntu. My most critical servers are on Debian and new ones will be on Ubuntu LTS. I have had very few OS specific issues. I have always built from source except on the Slug but I noticed that Ubuntu has it in the apt repository which would be a great convenience if you are new to Linux or manage a lot of servers. Now that you mention it, are packages of that distribution really maintained? The recent volnurability of AST-2008-006 is a good test case for that. If affects both 1.2 and 1.4 . The annoncement by Digium: http://downloads.digium.com/pub/asa/AST-2008-006.html As with the previous ones, the text is quite clear about the fix. backporting that patch to a slightly older version is not that tricky (and it is something that a distribution package maintainer is used to doing anyway). So what about updates? The optware feed for Unslung (on the NSLU2) is up-to-date on Asterisk 1.4 with 1.4.19.1 but a rev or two behind on 1.2 with 1.2.24 The LWN page for this advisory only lists Fedora and Debian: http://lwn.net/Articles/280318/ Response ime in both was quite reasonable. LWN also tracks adsisories from various other distributions. You can see the list in http://lwn.net/Alerts/ . The following other distributions have 'asterisk' packages: * Gentoo * Mandriva (??? - probably only in contrib and is unsupported) * rPath (Not sure. See below about AstriskNOW) * SUSE * Ubuntu (the package is in 'universe', and not officially supported) The issue is listed as corrected in AsteriskNOW 1.0.3, but the latest version available for download is 1.0.2 . If I read rpath's repository page correctly, then the most recently released version of Asterisk is 1.4.17-2 , from Feb-2008 and thus does not contain this fix. To see the versions of packages i Ubuntu: http://packages.ubuntu.com/asterisk As you can see, both Hardy and the development distribution (Interpid) include the same version of the package. As you can see from following the changelog link: http://changelogs.ubuntu.com/changelogs/pool/universe/a/asterisk/asterisk_1.4.17~dfsg-2ubuntu1/changelog The security issues of 1.4.18.1 were backported to that 1.4.17 package. But nothing about the recent advisory. The Gentoo port is basically where the Ubuntu package is: missing only the last one: http://packages.gentoo.org/package/asterisk The FreeBSD port has not been updated yet. It is still at 1.4.18, and no sign of backported fixes: http://www.freebsd.org/cgi/cvsweb.cgi/ports/net/asterisk/ OpenBSD port was updated pretty fast (by upgrading to asterisk 1.4.19.1) http://www.openbsd.org/cgi-bin/cvsweb/ports/telephony/asterisk/ I don't know where to look for in other distributions. -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ -- 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] 3U server chassis Digium TE405P?
I'm not sure if a full-height card would fit (vertically) in a 3U chassis... but I would probably also assume that if it would not, that the chassis/mobo would have a PCI/PCI-Express riser card that would mount the cards horizontally. Might want to check that out with the manufacturer of the chassis. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Monday, May 12, 2008 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 3U server chassis Digium TE405P? Gentlemen, First let me say it's great to be back on the Asterisk mailing lists. Those of you who have been around for a while will remember me as Rushowr. I look forward to answering questions and whatnot in the future, but for the moment I have a minor question that I cannot find a definitive answer for online. I am in possession of a Digium TE405P card which I _know_ will fit in a 4U chassis, but we are building a new server and cannot get a 4U from the supplier that my current client wants to use. However, we can get a 3U chassis. My question is, will this card fit? Does anyone out there have a 405 out there that they have installed in a 3U? Thanks in advance for any help that can be offered, Sherwood McGowan VoIP / Telecom Solutions Consultant ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Crappy sound on Console (chan_oss)
Hi all, on my debian box i configured chan_oss to work with /dev/audio device. CLI console command and Dial(CONSOLE/dsp) work perfectly but i notice 2 problems: 1. audio is very low in volume, even if i set 100 the mixer volume (via cmd line setmixer utility) 2. the sound is very crappy: the voice is vibrant, words sounds like 'ttthhhiiisss iiisss aaa ttteeessstt. Seems like the audio frames are pumped in soundcard with a little gap beetween one and the next chunk. I'm speaking of a Debian 4 and Asterisk 1.4.18 (compiled) Someone have experience? -- PicoStreamer - the real WEB live streaming software vinz486.com ___ -- 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] 3U server chassis Digium TE405P?
Matt Watson wrote: I'm not sure if a full-height card would fit (vertically) in a 3U chassis... but I would probably also assume that if it would not, that the chassis/mobo would have a PCI/PCI-Express riser card that would mount the cards horizontally. Might want to check that out with the manufacturer of the chassis. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Monday, May 12, 2008 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 3U server chassis Digium TE405P? Gentlemen, First let me say it's great to be back on the Asterisk mailing lists. Those of you who have been around for a while will remember me as Rushowr. I look forward to answering questions and whatnot in the future, but for the moment I have a minor question that I cannot find a definitive answer for online. I am in possession of a Digium TE405P card which I _know_ will fit in a 4U chassis, but we are building a new server and cannot get a 4U from the supplier that my current client wants to use. However, we can get a 3U chassis. My question is, will this card fit? Does anyone out there have a 405 out there that they have installed in a 3U? Thanks in advance for any help that can be offered, Sherwood McGowan VoIP / Telecom Solutions Consultant ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks for the heads up, I've found full height capable 3U chassis. The worst thing about this whole ordeal was that I assumed (very bad idea, of course, stupid stupid stupid) that the 2u server had a riser card, which it did not :( Ah well, live and learn... Sherwood McGowan ___ -- 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] Is there a way to have Manager Bridge Channel without being connected
Hello All, Is there a way to have Manager Bridge Channel to the specified extension without the channel being connected. In the current scenario the channel only bridges once the call get connected, it does not bridge when any service provider (telco) message is played. I want to record all call originated by manager even if a telco message is played. Regards, Sanjay Rajdev ___ -- 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] 3U server chassis Digium TE405P?
Getting the RIGHT card for the RIGHT bus type and the RIGHT Chassis is NOT as simple as everyone will lead you to believe. My suggestion, worth exactly what you paid for it :) Get Exact Spec for the card your are considering and FAX / Email to PC vendor and have him send you In Writing that the card WILL fit in the box and in the bus. Then I would get the Exact Spec for the BOX and BUS in the box and send to DIGIUM or their OEM and get THEM to tell you it should all work. Overkill - some will say yes. But THEY won't be sitting there with you if your expensive Server comes in and your expensive card come in an they no-workie together Sherwood McGowan wrote: Matt Watson wrote: I'm not sure if a full-height card would fit (vertically) in a 3U chassis... but I would probably also assume that if it would not, that the chassis/mobo would have a PCI/PCI-Express riser card that would mount the cards horizontally. Might want to check that out with the manufacturer of the chassis. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Monday, May 12, 2008 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 3U server chassis Digium TE405P? Gentlemen, First let me say it's great to be back on the Asterisk mailing lists. Those of you who have been around for a while will remember me as Rushowr. I look forward to answering questions and whatnot in the future, but for the moment I have a minor question that I cannot find a definitive answer for online. I am in possession of a Digium TE405P card which I _know_ will fit in a 4U chassis, but we are building a new server and cannot get a 4U from the supplier that my current client wants to use. However, we can get a 3U chassis. My question is, will this card fit? Does anyone out there have a 405 out there that they have installed in a 3U? Thanks in advance for any help that can be offered, Sherwood McGowan VoIP / Telecom Solutions Consultant ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks for the heads up, I've found full height capable 3U chassis. The worst thing about this whole ordeal was that I assumed (very bad idea, of course, stupid stupid stupid) that the 2u server had a riser card, which it did not :( Ah well, live and learn... Sherwood McGowan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Which sound file formats?
I've got the text files created -- thanks to Russell Bryant -- for re-building the core and extra sounds using another voice but I'm not sure which formats to actually build. This will be a small/personal system using Vitelity.net so will only have SIP connections. The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729, .gsm, .ulaw, and .wav. What are the minimal formats I need or can get by with. Possibly even an ordered preference list. Thanks, Rod -- ___ -- 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] 3U server chassis Digium TE405P?
On Mon, May 12, 2008 at 3:15 PM, Sherwood McGowan [EMAIL PROTECTED] wrote: Matt Watson wrote: I'm not sure if a full-height card would fit (vertically) in a 3U chassis... but I would probably also assume that if it would not, that the chassis/mobo would have a PCI/PCI-Express riser card that would mount the cards horizontally. Might want to check that out with the manufacturer of the chassis. -- Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sherwood McGowan Sent: Monday, May 12, 2008 11:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] 3U server chassis Digium TE405P? Gentlemen, First let me say it's great to be back on the Asterisk mailing lists. Those of you who have been around for a while will remember me as Rushowr. I look forward to answering questions and whatnot in the future, but for the moment I have a minor question that I cannot find a definitive answer for online. I am in possession of a Digium TE405P card which I _know_ will fit in a 4U chassis, but we are building a new server and cannot get a 4U from the supplier that my current client wants to use. However, we can get a 3U chassis. My question is, will this card fit? Does anyone out there have a 405 out there that they have installed in a 3U? Thanks in advance for any help that can be offered, Sherwood McGowan VoIP / Telecom Solutions Consultant ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Thanks for the heads up, I've found full height capable 3U chassis. The worst thing about this whole ordeal was that I assumed (very bad idea, of course, stupid stupid stupid) that the 2u server had a riser card, which it did not :( Ah well, live and learn... Sherwood McGowan Riser cards are dirt cheap. http://search.ebay.com/search/search.dll?from=R40_trksid=m37satitle=riser+cardcategory0= You can put a TE405 in a 1 server (horizontally of course). Thanks, Steve Totaro ___ -- 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] Which sound file formats?
Asterisk will automatically chose the best format - per ATFOT Roderick A. Anderson wrote: I've got the text files created -- thanks to Russell Bryant -- for re-building the core and extra sounds using another voice but I'm not sure which formats to actually build. This will be a small/personal system using Vitelity.net so will only have SIP connections. The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729, .gsm, .ulaw, and .wav. What are the minimal formats I need or can get by with. Possibly even an ordered preference list. Thanks, Rod ___ -- 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] 3U server chassis Digium TE405P?
A quality 3U chassis will mount the cards parallel to the mainboard with the use of a riser card, just as a 1U chassis does. If you are intent on sourcing the components yourself may I suggest a Tyan or Supermicro barebones server? I think that is the best solution for integration in these sort of specialized systems. I know they've saved me many headaches. On Mon, May 12, 2008 at 11:29 AM, Sherwood McGowan [EMAIL PROTECTED] wrote: Gentlemen, First let me say it's great to be back on the Asterisk mailing lists. Those of you who have been around for a while will remember me as Rushowr. I look forward to answering questions and whatnot in the future, but for the moment I have a minor question that I cannot find a definitive answer for online. I am in possession of a Digium TE405P card which I _know_ will fit in a 4U chassis, but we are building a new server and cannot get a 4U from the supplier that my current client wants to use. However, we can get a 3U chassis. My question is, will this card fit? Does anyone out there have a 405 out there that they have installed in a 3U? Thanks in advance for any help that can be offered, Sherwood McGowan VoIP / Telecom Solutions Consultant ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] chan_mobile install with Asterisk 1.4.19
Does anybody know if i can make (chan_mobile) module to be installed and work with Asterisk 1.4.19 ?___ -- 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] chan_mobile install with Asterisk 1.4.19
On Tue, May 13, 2008 at 3:44 AM, gres [EMAIL PROTECTED] wrote: Does anybody know if i can make (chan_mobile) module to be installed and work with Asterisk 1.4.19 ? Make sure you have the dependencies (bluez*) and do a make menuselect when compiling Asterisk. Good info and links http://www.voip-info.org/wiki/view/chan_mobile Thanks, Steve Totaro ___ -- 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] Lone worker system
P.S. Hope your lone worker is paid a lot to be working for a shitty company checking up on them like that :) Actually, in the UK, a company has a duty under the Health and Safety at Work etc Act of 1974, and the Management of Health and Safety at Work Regulations of 1999. 'Employers have responsibilities for the health, safety and welfare at work of their employees and the health and safety of those affected by the work, eg visitors, such as contractors and self-employed people who employers may engage. These responsibilities cannot be transferred to people who work alone. It is the employer's duty to assess risks to lone workers and take steps to avoid or control risk where necessary. Employees have responsibilities to take reasonable care of themselves and other people affected by their work and to co-operate with their employers in meeting their legal obligations.' -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Hanselman Sent: Monday, May 12, 2008 11:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lone worker system Spot on (except for the shitty way, it's pretty standard, in building there are paging systems that start an escalating tone, beyond the building these don't work, so if you're away then we'd be dialling the mobile). Steve -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins Sent: 12 May 2008 16:02 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lone worker system He wants a randomly generated phone call to be generated to a specific extension. Eg once an hour for the midnight to dawn shift at a random time per hour. When the person picks up they are asked a question using an audio file. (or text to speech). Then the person has to enter the correct dtmf answering the question (eg 1 - 5) If the person fails to answer the phone (I'm guessing here but a second call will be placed 2 mins later). If this call is also 'fail to answer' an escalation call to a supervisor or something similar will occur indicating that the 'lone worker' failed to respond and is either - dead from a stabbing, or 2 jerking off in the bathroom and not at his post. Cheers, Dean P.S. Hope your lone worker is paid a lot to be working for a shitty company checking up on them like that :) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Monday, May 12, 2008 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Lone worker system On Mon, May 12, 2008 at 10:28 AM, Steve Hanselman [EMAIL PROTECTED] wrote: Has anybody got any scripts for a lone worker system using Asterisk before I write them? Something along the lines of a regular phonecall with some kind of random question (e.g. press 1 then 5) to provide monitoring of lone workers with alerts? Steve I think a little more elaboration would get you more helpful advice. I have read your message a couple of times and still don't really understand what you need. Thanks, Steve Totaro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] Problem with SIP Subscription Status
Hach Segal a écrit : Hello All, I've been having some intermittent trouble with an Asterisk 1.2.10 Before anything else did you tried an updated asterisk 1.2 The last one is 1.2.28 or something like that, and there has been a lot of security patches, and fixes since your version. Did you look through the changelog / bugs tracker to see if your problem has already been reported ? -- Benoit Plessis ___ -- 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] business class sip provider with a SIP proxy server in India ?
I need SIP trunking from a high quality business service provider for 25,000 SIP minutes growing at approximately 10% each month. Currently we are using exgn.net to provide inbound 800 (Costs $200 for approx 10,000 minutes a month) and we are using broadvoice.com for outgoing calls (Costs $100 for approx 15,000 minutes a month. We have two unlimited business plans with them.) The reason I want to change over to a different sip trunk provider is: 1. Broadvoice.com will only allow 2 channels to be active at one time. To have more then two concurrent calls we need to have more then one line and each line will have its own caller id. I would like to have one caller id and broadvoice will not allow 4 channels with 1 caller id. Even If I am willing to pay for the 2 lines. 2. I would prefer to have a single provider for both the inbound 800 calls and the outgoing calls. 3. Possibly save some money. My requirements are: 1. A lot of the customer support is done from India. So if there is a SIP provider with a SIP proxy server in India that would definitely be a plus. 2. The offices in US are both on each coast and west coast with their own local asterisk servers. So I defnitely want a SIP provider with multiple SIP proxy servers distributed geographically. I have considered the following vendors till now: 1. Bandwidth.com - Left them a voice mail and filled out the inquiry form on their website. Still waiting to hear back from them. Any recommendations would be appreciated. Thanks for your time, Sysadmin http://www.debtconsolidationcare.com Internets First get out of debt community ___ -- 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] Which sound file formats?
Andreas van dem Helge wrote: Best is to keep native format files for all codecs you intend to use. Where are these rebuilt files with another voice? And any chance they'll ever be done by Pat Fleet or Ann, the Cisco voice? FWIW I'm in contact with someone who's been in contact with Pat who says she's willing to do it but we don't think its worth the effort unless these were distributed officially. I'm doing this for my own system. I picked Callie-8kHz, from Cepstral, and want all the core and extra sounds done in her voice. I'm not going to, nor do I think I can, distribute these. Rod -- On Mon, May 12, 2008 at 4:57 PM, Roderick A. Anderson [EMAIL PROTECTED] wrote: I've got the text files created -- thanks to Russell Bryant -- for re-building the core and extra sounds using another voice but I'm not sure which formats to actually build. This will be a small/personal system using Vitelity.net so will only have SIP connections. The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729, .gsm, .ulaw, and .wav. What are the minimal formats I need or can get by with. Possibly even an ordered preference list. Thanks, Rod -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which sound file formats?
Al Baker wrote: Asterisk will automatically chose the best format - per ATFOT I guess I'm not getting my head wrapped around this concept. I understand the choosing but not how I might influence it. Probably best to just build them all and let Asterisk sort it out. I'll research this some more. Thanks Al. Rod -- Roderick A. Anderson wrote: I've got the text files created -- thanks to Russell Bryant -- for re-building the core and extra sounds using another voice but I'm not sure which formats to actually build. This will be a small/personal system using Vitelity.net so will only have SIP connections. The /var/lib/asterisk/sounds/ directory contains .alaw, .g722, .g729, .gsm, .ulaw, and .wav. What are the minimal formats I need or can get by with. Possibly even an ordered preference list. Thanks, Rod ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Require a Touch-Tone to Connect? proof of concept with meetme()
I have read the post about the touch tone before to connect so
transfered calls don't end up in voicemail boxes of mobile phones. I
have done some work last year on transfering an inbound call to
different extensions by using meetme() and local channels so a whole
group can start talking. I end up with a remarkably low number of
lines and it is actually working . It is just a proof of concept that
can be complemented with voiceprompts and a mechanism to make sure
that just one extra line enters the conference room. If you have
improvements please share them on the mailing list. I hope someone
will find this usefull.
Below are the actual Asterisk lines. It is pure old fashioned Asterisk
without any additional AGI scripts or whatever.
With friendly regards,
Erik de Wild
Tripple-o
Your Asterisk migration partner
; this is where te inbound call is routed to with exten =
whatever,n,Goto(inbound_forking,s,1)
;;
[inbound_forking]
; this are the three local channels used for dialing the external or
local numbers. In this
; example all the numbers are external
exten = s,1,Dial(local/[EMAIL PROTECTED]local/[EMAIL PROTECTED]local/
[EMAIL PROTECTED],20)
; this is where the inbound call is routed to the conference room,
notice the /n
;;
exten = s,n,Dial(local/[EMAIL PROTECTED]/n,10)
[meetme]
;; extension for the inbound call
;;;
exten = inbound,1,MeetMe(9000,qM1)
exten = inbound,n,Hangup()
;
; extensions for the three (or more) different outbound lines
; that are routed into a macro
exten = intern1,1,Dial(SIP/3120/0031621xx,
20,M(meetme_test))
exten = intern2,1,Dial(SIP/3120/0031642xx,
20,M(meetme_test))
exten = intern3,1,Dial(SIP/3120/0031556xx,
20,M(meetme_test))
; this is the macro for joining the conference room
; first it read the number of
[macro-meetme_test]
exten = s,1,Set(ROOMNUMBER=9000)
exten = s,n,MeetMeCount(${ROOMNUMBER}|COUNT)
exten = s,n,Wait(2)
exten = s,n,SayNumber(${COUNT}) ; as long as the number is 0 or 1
it makes sense to join
exten = s,n,Authenticate(1) ; here is the one
touch needed before you can join. A voicemail box of a mobile can't do
that ;-)
exten = s,n,Meetme(9000)
exten = s,n,Hangup()
___
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Re: [asterisk-users] Zap Channels Collide (Incoming Outgoing)
There are krone blocks designed for CAT5, and I've seen them in use as well. However, there's no way I'd be using them for today's networks. /Especially/ having seen one of these krone blocks used to double-punch two network ports together. Bill Andersen wrote: Oh, yes. I saw an entire Cat 5 network on punch blocks one time! Everybody needs to learn the other side before getting involved. ___ -- 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] Which sound file formats?
Tilghman Lesher wrote: On Monday 12 May 2008 17:27, Roderick A. Anderson wrote: Al Baker wrote: Asterisk will automatically chose the best format - per ATFOT I guess I'm not getting my head wrapped around this concept. I understand the choosing but not how I might influence it. Probably best to just build them all and let Asterisk sort it out. I'll research this some more. If you want to produce just one, I'd recommend producing slin or wav, as they are essentially the same (slin is just wav without the header). The thing to note about this format is that as it is uncompressed, you will reduce the amount of work needed for transcoding (other than if you have native files), because transcoding from, for example, gsm to speex requires two transcoding operations: first, uncompress from gsm, and second, compress to speex. By adapting the most common intermediate format, you only need one transcoding operation per packet. Native files require no transcoding at all. Transcoding tends to be the most CPU intensive task on a typical Asterisk machine. Tilghman, Once again you come through clear. But just to make sure I transcode this correctly I can remove all the other formats and only have .wav and/or .slin files. Simple and clean. Plus swift can generate the .wav files optimized for VoIP. I can go sox-less. :-) Thanks, Rod -- ___ -- 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] Newbie Dialplan: Best Practice in using Context - Do not use Default??
In The future of Telephony, it says ... We should also note for security's sake you should always make sure that your [incoming] context never allows outbound dialing. (If by chance it did, people could dial into your system and make outbound toll calls that would be charged to you!) The book was demonstrating using a PSTN environment and the zapata.conf was something like: context=internal signaling=fxo_ks channel=1 context=incoming signaling=fxs_ks channel=2 In PRI environment, does it mean that we have to purposely separate the say ISDN 20 channels into [internal] and [incoming] as well? This would not make sense to me as ISDN uses a one port card to contain multiple channels while the ports of a say TDM400P refer to each channel. If I just define a [default] context for a PRI environment, is this insecure? Can someone please enlighten me on this? ___ -- 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] Newbie Dialplan: Best Practice in using Context - Do not use Default??
At 9:43 AM on 13 May 2008, Lee, John (Sydney) wrote: In The future of Telephony, it says ... We should also note for security's sake you should always make sure that your [incoming] context never allows outbound dialing. (If by chance it did, people could dial into your system and make outbound toll calls that would be charged to you!) The book was demonstrating using a PSTN environment and the zapata.conf was something like: context=internal signaling=fxo_ks channel=1 context=incoming signaling=fxs_ks channel=2 In PRI environment, does it mean that we have to purposely separate the say ISDN 20 channels into [internal] and [incoming] as well? This would not make sense to me as ISDN uses a one port card to contain multiple channels while the ports of a say TDM400P refer to each channel. If I just define a [default] context for a PRI environment, is this insecure? Can someone please enlighten me on this? In the example you quoted, channel 1 is an FXS port, which would be an internal extension--a phone--from which someone would be allowed to make an outbound call. Channel 2 is an FXO port, which is connected to the PSTN, and would take incoming calls from the wild. So in that example, you wouldn't want the incoming context to be allowed to make outbound calls. In your case, I'm guessing all your Zap channels come from the PRI, which is connected to the PSTN. If so, then you're right--you just need one context for your zapata.conf which you would use on all your ISDN channels. Just don't let that context dial out. I don't know if you'd want to call that context default... because that one seems to be special in Asterisk. But maybe I'm just being superstitious. :-) -- C. Chad Wallace, B.Sc. The Lodging Company http://www.skihills.com/ OpenPGP Public Key ID: 0x262208A0 Debian Hint #14: If you would like to follow things happening to a package (for example, if you want to see bug reports, release notices, and other similar things), consider subscribing to it on the Package Tracking System. You can find out more about the PTS at: http://www.debian.org/doc/manuals/developers-reference/ch-resources.en.html (Section 4.10) ___ -- 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] Crappy sound on Console (chan_oss)
Additional info. I tried to disable chan_oss and enable chan_alsa (seems like kernel 2.6 should use alsa and not oss). Well, no dial CLI command, no Dial(console/dsp) channel available. So, how to use chan_alsa? -- PicoStreamer - the real WEB live streaming software vinz486.com ___ -- 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] Which sound file formats?
On Monday 12 May 2008 18:44, Roderick A. Anderson wrote: Tilghman Lesher wrote: On Monday 12 May 2008 17:27, Roderick A. Anderson wrote: Al Baker wrote: Asterisk will automatically chose the best format - per ATFOT I guess I'm not getting my head wrapped around this concept. I understand the choosing but not how I might influence it. Probably best to just build them all and let Asterisk sort it out. I'll research this some more. If you want to produce just one, I'd recommend producing slin or wav, as they are essentially the same (slin is just wav without the header). The thing to note about this format is that as it is uncompressed, you will reduce the amount of work needed for transcoding (other than if you have native files), because transcoding from, for example, gsm to speex requires two transcoding operations: first, uncompress from gsm, and second, compress to speex. By adapting the most common intermediate format, you only need one transcoding operation per packet. Native files require no transcoding at all. Transcoding tends to be the most CPU intensive task on a typical Asterisk machine. Tilghman, Once again you come through clear. But just to make sure I transcode this correctly I can remove all the other formats and only have .wav and/or .slin files. Correct. Just make sure that the wav files are 8000Hz, 16 bit signed linear samples, single channel (mono) only. Anything else will not give the desired results. Simple and clean. Plus swift can generate the .wav files optimized for VoIP. I can go sox-less. :-) If you have the 8kHz voices, the default swift output will be exactly what Asterisk requires. The 16kHz voices will need to be down-sampled, however. -- Tilghman ___ -- 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] Newbie Dialplan: Best Practice in using Context - Do not use Default??
With an ISDN10/20/30/etc, I would just put all the lines into an 'incoming' context - and make sure that incoming context doesn't have any includes (unless you really need them...) PaulH On Tue, 2008-05-13 at 09:43 +1000, Lee, John (Sydney) wrote: In The future of Telephony, it says ... We should also note for security's sake you should always make sure that your [incoming] context never allows outbound dialing. (If by chance it did, people could dial into your system and make outbound toll calls that would be charged to you!) The book was demonstrating using a PSTN environment and the zapata.conf was something like: context=internal signaling=fxo_ks channel=1 context=incoming signaling=fxs_ks channel=2 In PRI environment, does it mean that we have to purposely separate the say ISDN 20 channels into [internal] and [incoming] as well? This would not make sense to me as ISDN uses a one port card to contain multiple channels while the ports of a say TDM400P refer to each channel. If I just define a [default] context for a PRI environment, is this insecure? Can someone please enlighten me on this? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
