Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration
Great tip Duncan :)
On Thu, Jul 12, 2012 at 10:29 AM, Duncan Turnbull dun...@e-simple.co.nzwrote:
You can also specify routes with an callerid qualifier as 09XX/20X
This would only have it apply to extensions in the 200-209 range
That route can then point to a trunk going nowhere if you want to block
them
In freepbx there is a field in outbound route page to select callerid that
the route applies to
Cheers Duncan
On 12/07/2012, at 4:52 PM, SamyGo govoi...@gmail.com wrote:
See
Route-Permissions module,
It lets you restrict certain phones/extensions to follow a dial-plan
pattern and dial out to the defined trunk etc meanwhile not breaking any
other functionality or features of FPBX- though you can restrict the
features from this too.
http://www.freepbx.org/support/documentation/howtos/how-to-give-a-particular-extension-different-or-restricted-trunk-access
http://www.freepbx.org/support/documentation/module-documentation/third-party-unsupported-modules/outbound-route-permission
http://mirror.freepbx.org/modules/release/contributed_modules/
OR
Custom Context
http://www.freepbx.org/support/documentation/module-documentation/third-party-unsupported-modules/customcontexts
See w/e fits your requirements. What I suggest suits your need is the
Route-permission module. Though it'll be bit complicated but worth giving a
try.
Regards,
Sammy
On Thu, Jul 12, 2012 at 4:01 AM, Warren Selby wcse...@selbytech.comwrote:
On Wed, Jul 11, 2012 at 4:56 PM, bilal ghayyad bilmar...@yahoo.comwrote:
Fine, did you read the question well and understand about what I am
asking?
Perhaps I did not understand what you were asking. I thought you were
wanting to do something custom per extension (in the case of my example,
the something custom was control outbound call access to either local
only or local and long distance, etc. You can figure out you're own
something custom), but still have all the calls have all the standard
FreePBX features that you only get when using the [from-internal] context.
In my example, the extensions are in the 2XXX range, and they would
either have a context of [custom-local-only] or [custom-long-distance],
depending on what you wanted to allow that extension to dial.
To break down my example:
[custom-local-only] -- The name of our custom context. It could be
anything you want, as long as it's in square brackets
exten = _281NXX,1,Verbose(Outbound call from local-only context) --
This step is purely informational, it has no bearing on CDRs or anything
else...it's just a useful step for debugging. I tend to do this for
everything, it's the same as some people use the NoOp() command to have
debugging information in their CLI output.
same = n,Goto(${EXTEN},from-internal,1) -- This step sends the call
to the [from-internal] context and handles it exactly as if you weren't
using any custom call controls. In my example, however, it will only go
there if it meets the criteria of matching the pattern (in other words, the
call would have to be placed to a number that matches the _281NXX
pattern). same = n is a shorthand way of writing exten =
_281NXX,n. It was added in around 1.6 I think, I'm not entirely
sure.
exten = _2XXX,1,Verbose(Internal extension-to-extension call) --
Again, this is purely an informational step, useful for debugging. It can
be skipped or expanded as you see fit, it has no bearing on CDR records or
anything else, other than CLI output.
same = n,Goto(${EXTEN},from-internal,1) -- This does the same as the
previous example, however it will only go to the [from-internal] context if
the pattern that was dialed matches _2XXX. This is assuming you're using
internal extensions in the range of _2XXX. You can change this to whatever
works for you.
[custom-long-distance] -- another custom context, this time it allows
long distance NANPA calling as well as local and internal calls
exten = _1NXXNXX,1,Verbose(Outbound call from local and
long-distance context) -- I hope you're seeing the pattern by now. This
is simply a useful debugging step, with no bearing on anything else.
same = n,Goto(${EXTEN},from-internal,1) -- The call passes into the
[from-internal context if it matches the pattern of _1NXXNXX, a typical
NANPA long distance call.
include = custom-local-only -- include the local dialing context that
way we don't have to duplicate any code that we've previously written,
mostly useful for the internal extension dialing.
So you can see, the Verbose() statement has no bearing on CDR's what so
ever. I wasn't aware that FreePBX used any kind of custom CDR database, I
assumed it was simply using the asterisk CDR database, where any call
through the system generates a CDR. Since someone else had mentioned that
they did not get any CDR logging or any of the other FreePBX features
without making the extension have a context of [from-internal], I
Re: [asterisk-users] FreePBX: using context other than the default context and the generation for the configuration
Thanks Samy
I am figuring you may know but with freepbx if you want to make it a bit more
tailored then send it to a custom trunk
In freepbx add a custom trunk with the custom dial string
Local/$OUTNUM$@blocked-number-custom
/etc/asterisk/extensions_custom.conf
[blocked-number-custom]
exten = _X.,1,Answer()
exten = _X.,n,Verbose(Blocked an 0900 trunk call)
exten = _X.,n,Playback(custom/0900-block)
exten = _X.,n,Hangup
And you can record a message or send it somewhere else or whatever you feel like
Cheers Duncan
On 12/07/2012, at 6:11 PM, SamyGo wrote:
Great tip Duncan :)
On Thu, Jul 12, 2012 at 10:29 AM, Duncan Turnbull dun...@e-simple.co.nz
wrote:
You can also specify routes with an callerid qualifier as 09XX/20X
This would only have it apply to extensions in the 200-209 range
That route can then point to a trunk going nowhere if you want to block them
In freepbx there is a field in outbound route page to select callerid that
the route applies to
Cheers Duncan
On 12/07/2012, at 4:52 PM, SamyGo govoi...@gmail.com wrote:
See
Route-Permissions module,
It lets you restrict certain phones/extensions to follow a dial-plan pattern
and dial out to the defined trunk etc meanwhile not breaking any other
functionality or features of FPBX- though you can restrict the features from
this too.
http://www.freepbx.org/support/documentation/howtos/how-to-give-a-particular-extension-different-or-restricted-trunk-access
http://www.freepbx.org/support/documentation/module-documentation/third-party-unsupported-modules/outbound-route-permission
http://mirror.freepbx.org/modules/release/contributed_modules/
OR
Custom Context
http://www.freepbx.org/support/documentation/module-documentation/third-party-unsupported-modules/customcontexts
See w/e fits your requirements. What I suggest suits your need is the
Route-permission module. Though it'll be bit complicated but worth giving a
try.
Regards,
Sammy
On Thu, Jul 12, 2012 at 4:01 AM, Warren Selby wcse...@selbytech.com wrote:
On Wed, Jul 11, 2012 at 4:56 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Fine, did you read the question well and understand about what I am asking?
Perhaps I did not understand what you were asking. I thought you were
wanting to do something custom per extension (in the case of my example, the
something custom was control outbound call access to either local only or
local and long distance, etc. You can figure out you're own something
custom), but still have all the calls have all the standard FreePBX
features that you only get when using the [from-internal] context.
In my example, the extensions are in the 2XXX range, and they would either
have a context of [custom-local-only] or [custom-long-distance], depending
on what you wanted to allow that extension to dial.
To break down my example:
[custom-local-only] -- The name of our custom context. It could be
anything you want, as long as it's in square brackets
exten = _281NXX,1,Verbose(Outbound call from local-only context) --
This step is purely informational, it has no bearing on CDRs or anything
else...it's just a useful step for debugging. I tend to do this for
everything, it's the same as some people use the NoOp() command to have
debugging information in their CLI output.
same = n,Goto(${EXTEN},from-internal,1) -- This step sends the call to
the [from-internal] context and handles it exactly as if you weren't using
any custom call controls. In my example, however, it will only go there if
it meets the criteria of matching the pattern (in other words, the call
would have to be placed to a number that matches the _281NXX pattern).
same = n is a shorthand way of writing exten = _281NXX,n. It was
added in around 1.6 I think, I'm not entirely sure.
exten = _2XXX,1,Verbose(Internal extension-to-extension call) -- Again,
this is purely an informational step, useful for debugging. It can be
skipped or expanded as you see fit, it has no bearing on CDR records or
anything else, other than CLI output.
same = n,Goto(${EXTEN},from-internal,1) -- This does the same as the
previous example, however it will only go to the [from-internal] context if
the pattern that was dialed matches _2XXX. This is assuming you're using
internal extensions in the range of _2XXX. You can change this to whatever
works for you.
[custom-long-distance] -- another custom context, this time it allows long
distance NANPA calling as well as local and internal calls
exten = _1NXXNXX,1,Verbose(Outbound call from local and long-distance
context) -- I hope you're seeing the pattern by now. This is simply a
useful debugging step, with no bearing on anything else.
same = n,Goto(${EXTEN},from-internal,1) -- The call passes into the
[from-internal context if it matches the pattern of _1NXXNXX, a typical
NANPA long
Re: [asterisk-users] Regrading Speech Recognition.
Hi Akhilesh, Probably this link would give you some idea on ASR. With the help of it, add some logic in dialplan to develop an application of your choice. (Courtesy Lefteris Zafiris) Goto https://github.com/zaf/asterisk-speech-recog/ and read README --Satish Barot On Thu, Jul 5, 2012 at 12:46 PM, akhilesh chand omakhileshch...@gmail.comwrote: ok,how can i develop with short vocab like sales,support,etc. I have read many article but I'm not able to pick the right point, how can i develop or configure speech reorganization with asterisk. Is there any article or link please share and guide me. Regards Akhilesh On Thu, Jul 5, 2012 at 12:29 PM, Mitul Limbani mi...@enterux.in wrote: Things that look simple r quite complex to build :-) Indian Accent ASR on proper names is herculean task. No speech recognition known to mankind as of date can handle so many dialects being spoken in India, so in short what you want is nice to have, but nearly impossible to develop. Better try with short vocab on generic words (sales, support, etc.) Mitul On Jul 5, 2012 12:23 PM, akhilesh chand omakhileshch...@gmail.com wrote: Hi, I want to develop a IVR application that repond to speech input from the caller in asterisk. For example, imagine a caller who wants to speak with Ram Kumar. On a traditional IVR/auto attendant, the caller may be entering “76484” to spell “Kumar” and the system may respond with: “Press 1 for Radhe Kumar, 2 for Shyam Kumar, 3 for Krishan Kumar, ..., 5 for Ram Kumar.” The caller can simply say “Ram Kumar” and conversation can be established much more quickly. Is there any article or link regrading the same please guide me. Regrads Thanks Akhilesh -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] chan_ss7 quick patch to enable RBT
Hello everyone, I am trying to apply thishttp://www.voip-info.org/storage/users/496/27496/images/499/rbt.patch.diffpatch on chan_ss7-2.1.0 for RingBack tone but its not accepting and throwing errors: Hunk #1 FAILED at 704. Hunk #2 FAILED at 715. I have done the patch modifications manually in l4isup.c There is just one question, how do I pass the RB file-to-play on an SS7 channel via asterisk? -- Thanks. P.S. here is the source of the patch: http://www.voip-info.org/wiki/view/chan_ss7+quick+patch+to+enable+RBT -- _ -- 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] AMR - Segmentation Fault
Hi Chandrakant, for what I can see the patch you're trying to use ( http://sourceforge.net/projects/aterisk-amr/files/) is going to work with 1.8.13 as well, however the issue behind the crash is not in Asterisk's code but in 3GPP's original AMR implementation when running on 64-bit hosts. I've found that applying the patch suggested here: http://lists.digium.com/pipermail/asterisk-video/2008-October/002229.html to codecs/amr/typedef.h fixed the crash. I hope this helps and I'd be interested to know if this solves your issue. Regards, Giacomo Hi All, OS : Cent OS 5 64Bit Asterisk : 1.8.0-rc2 AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/ When I tried to call or start asterisk, I found Segmentation Fault. Below I paste same for AMR Loaded symbols for /usr/lib/asterisk/modules/app_db.so Core was generated by `asterisk -qg'. Program terminated with signal 11, Segmentation fault. #0 D_plsf_3 (st=value optimized out, mode=value optimized out, bfi=value optimized out, indice=value optimized out, lsp1_q=0x7fff11d05df0) at sp_dec.c:567 567 tmp = ( ( cos_table[ind+1]-cos_table[ind] )*offset ) 1; (gdb) br Breakpoint 1 at 0x2aaab57093f1: file sp_dec.c, line 567. (gdb) bt #0 D_plsf_3 (st=value optimized out, mode=value optimized out, bfi=value optimized out, indice=value optimized out, lsp1_q=0x7fff11d05df0) at sp_dec.c:567 #1 0x2aaab570df95 in Decoder_amr (st=0x2aaad6147d00, mode=MR515, parm=0x7fff11d06a40, frame_type=value optimized out, synth=0x7fff11d060a0, A_t=0x7fff11d06730) at sp_dec.c:4717 #2 0x2aaab5712e6a in Speech_Decode_Frame (st=0x2aaad613e200, mode=80, parm=0x2aaab5725400, frame_type=4294949091, synth=0x2aaad6142ba0) at sp_dec.c:5676 #3 0x2aaab56efb25 in Decoder_Interface_Decode (st=0x2aaad613e1e0, bits=value optimized out, synth=0x2aaad6142ba0, bfi=value optimized out) at interf_dec.c:816 #4 0x2aaab56ee6f9 in amrtolin_framein (pvt=0x2aaad613e5c0, f=value optimized out) at codec_amr.c:263 #5 0x00528244 in framein (pvt=0x2aaad613e5c0, f=0x2aaab5942e40) at translate.c:178 #6 0x00529538 in calc_cost (t=0x2aaab593ff40, seconds=1) at translate.c:397 #7 0x0052a00c in __ast_register_translator (t=0x2aaab593ff40, mod=value optimized out) at translate.c:835 #8 0x2aaab56ee37b in load_module () at codec_amr.c:490 #9 0x004c29e3 in start_resource (mod=0xdf) at loader.c:785 #10 0x004c3308 in load_resource_list (load_order=0x7fff11d07000, global_symbols=0, mod_count=0x7fff11d0701c) at loader.c:973 #11 0x004c3727 in load_modules (preload_only=0) at loader.c:1126 #12 0x0043c2c4 in main (argc=value optimized out, argv=0x7fff11d095e8) at asterisk.c:3794 (gdb) quit -- Regards, Chandrakant Solanki -- Giacomo Vacca +44 7408866340 +39 3928454855 giacomo.va...@gmail.com skype: giacomo.vacca -- _ -- 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] Regrading Speech Recognition.
Is there a tool integrated with asterisk which can give us the pitch of the utterance? On Thu, Jul 12, 2012 at 3:09 PM, Satish Barot satish4aster...@gmail.comwrote: Hi Akhilesh, Probably this link would give you some idea on ASR. With the help of it, add some logic in dialplan to develop an application of your choice. (Courtesy Lefteris Zafiris) Goto https://github.com/zaf/asterisk-speech-recog/ and read README --Satish Barot On Thu, Jul 5, 2012 at 12:46 PM, akhilesh chand omakhileshch...@gmail.com wrote: ok,how can i develop with short vocab like sales,support,etc. I have read many article but I'm not able to pick the right point, how can i develop or configure speech reorganization with asterisk. Is there any article or link please share and guide me. Regards Akhilesh On Thu, Jul 5, 2012 at 12:29 PM, Mitul Limbani mi...@enterux.in wrote: Things that look simple r quite complex to build :-) Indian Accent ASR on proper names is herculean task. No speech recognition known to mankind as of date can handle so many dialects being spoken in India, so in short what you want is nice to have, but nearly impossible to develop. Better try with short vocab on generic words (sales, support, etc.) Mitul On Jul 5, 2012 12:23 PM, akhilesh chand omakhileshch...@gmail.com wrote: Hi, I want to develop a IVR application that repond to speech input from the caller in asterisk. For example, imagine a caller who wants to speak with Ram Kumar. On a traditional IVR/auto attendant, the caller may be entering “76484” to spell “Kumar” and the system may respond with: “Press 1 for Radhe Kumar, 2 for Shyam Kumar, 3 for Krishan Kumar, ..., 5 for Ram Kumar.” The caller can simply say “Ram Kumar” and conversation can be established much more quickly. Is there any article or link regrading the same please guide me. Regrads Thanks Akhilesh -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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 -- _ -- 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] weird dect beheaviour multiple handsets
I have this very specific problem with two dect sets. Problem that I have is one-way audio, in this very rare situation. I am calling with a Siemens N510 with C610 handset to Panasonic KX-TGP500 with KX-TPA50 handset. This gives me problems when I am calling to a SIP account that is configured to ring all handsets. Then when one handset answers, I only hear the panasonic, but they don't hear me. When I call to an extension that is configured to ring only one handset, I don't have this problem. When I use panasonic on both sides, or Tiptel - Panasonic, I don't have this problem as well. I am breaking my head what this could be. Any idea's? -- _ -- 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] Asterisk with OpenBTS and mobile phone
Hello mailinglist,
I want to connect Asterisk with OpenBTS and make a call with a mobile
phone.
I use:
Ubuntu 11.10 + Kernel 3.0.22
GnuRadio 3.3.0
Asterisk 1.8.13
OpenBTS 2.8
Nokia Mobile Phone
OpenBTS works and I can send sms from the OpenBTS server to the
mobile phone. What I also need is a call between Asterisk and OpenBTS.
I have also two soft phones which works with Asterisk. And also OpenBSC
is working with Asterisk successfully (OpenBSC is another project).
Perhaps you can help me because I think it is an issue with Asterisk.
sip.conf:
;SIP-Phones (Twinkle)
[user1]
callerid = 6000
username = 6000
secret = 6000
canreinvite = no
type = friend
context = phones
allow = all
host = dynamic
dtmfmode = info
[user2]
callerid = 6001
username = 6001
secret = 6001
canreinvite = no
type = friend
context = phones
allow = all
host = dynamic
dtmfmode = info
; Mobile phone
[123456789101112]
callerid = 6201
username = 6201
secret = 6201
canreinvite = no
type = friend
context = sip_external
;context = open-bts
disallow = all
allow = gsm
host = 192.168.0.102
domain = 192.168.0.102
dtmfmode = info
extensions.conf
[internal]
exten = s,1,Verbose(1|Echo test application)
exten = s,n,Echo()
exten = s,n,Hangup()
exten = 6000,1,Verbose(1|Extension 6000)
exten = 6000,n,Dial(SIP/user1,30)
exten = 6000,n,Hangup()
exten = 6001,1,Verbose(1|Extension 6001)
exten = 6001,n,Dial(SIP/user2,30)
exten = 6001,n,Hangup()
[phones]
include = internal
include = default
[open-bts]
exten = 6002,1,Playback(demo-echotest)
exten = 6002,n,Echo
exten = 6002,n,Playback(demo-echodone)
exten = 6002,n,HangUp
[sip_external]
exten = 6201,1,Macro(dialGSM,123456789101112)
[macro-dialGSM]
exten = s,1,Dial(SIP/${ARG1},20)
exten = s,n,Goto(s-${DIALSTATUS},1)
exten = s-CANCEL,1,Hangup
exten = s-NOANSWER,1,Hangup
exten = s-BUSY,1,Busy(30)
exten = s-CONGESTION,1,Congestion (30)
exten = s-CHANUNAVAIL,1,Read(extension_digits,pbx-invalid)
exten = s-CHANUNAVAIL,n,GoTo(open-bts,${extension_digits},1)
I have tried both contexts, [open-bts] and [sip_external] and both don't
work
If I want to call the mobile phone (6201) with a Twinkle soft phone (6000)
I get following message in the CLI-window from Asterisk:
== Using SIP RTP CoS mark 5
-- Executing [6201@DLPN_DialPlan1:1] Macro(SIP/6000-0013,
stdexten,6201,SIP/6201) in new stack
-- Executing [s@macro-stdexten:1] Set(SIP/6000-0013,
__DYNAMIC_FEATURES=) in new stack
*[Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:468 ast_yyerror:
ast_yyerror(): syntax error: syntax error, unexpected '=', expecting $end;
Input:
= 1
^
[Jul 12 12:14:29] WARNING[7092]: ast_expr2.fl:472 ast_yyerror: If you
have questions, please refer to
https://wiki.asterisk.org/wiki/display/AST/Channel+Variables
-- Executing [s@macro-stdexten:2] GotoIf(SIP/6000-0013,
?5:3) in new stack
-- Goto (macro-stdexten,s,3)
-- Executing [s@macro-stdexten:3] Dial(SIP/6000-0013,
SIP/6201,20,) in new stack
[Jul 12 12:14:29] WARNING[7092]: app_dial.c:2274 dial_exec_full:
Unable to create channel of type 'SIP' (cause 20 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)*
-- Executing [s@macro-stdexten:4] Goto(SIP/6000-0013,
s-CHANUNAVAIL,1) in new stack
-- Goto (macro-stdexten,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-stdexten:1]
Goto(SIP/6000-0013, s-NOANSWER,1) in new stack
-- Goto (macro-stdexten,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-stdexten:1]
VoiceMail(SIP/6000-0013, 6201,u) in new stack
-- SIP/6000-0013 Playing 'vm-theperson.gsm' (language 'en')
-- SIP/6000-0013 Playing 'digits/6.gsm' (language 'en')
-- SIP/6000-0013 Playing 'digits/2.gsm' (language 'en')
-- SIP/6000-0013 Playing 'digits/0.gsm' (language 'en')
-- SIP/6000-0013 Playing 'digits/1.gsm' (language 'en')
-- SIP/6000-0013 Playing 'vm-isunavail.gsm' (language 'en')
-- SIP/6000-0013 Playing 'vm-intro.gsm' (language 'en')
== Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero
on 'SIP/6000-0013' in macro 'stdexten'
== Spawn extension (DLPN_DialPlan1, 6201, 1) exited non-zero on
'SIP/6000-0013'
*CLI sip show peers
Name/username HostDyn
Forcerport ACL Port Status
*123456789101112/6201
192.168.0.102N 5060
Unmonitored*
6000/6000 192.168.0.102
D N 5061 Unmonitored
6001/6001 192.168.0.102
D N 5061 Unmonitored
(...)
user1/6000 (Unspecified)
D N 0Unmonitored
user2/6001 (Unspecified)
D N 0Unmonitored
*CLI sip show peer
Re: [asterisk-users] weird dect beheaviour multiple handsets
Hello Is your server behind nat? This problems sounds me nat problems. Regards On Thu, Jul 12, 2012 at 7:53 AM, Roland o/d Akker aster...@rolandow.com wrote: I have this very specific problem with two dect sets. Problem that I have is one-way audio, in this very rare situation. I am calling with a Siemens N510 with C610 handset to Panasonic KX-TGP500 with KX-TPA50 handset. This gives me problems when I am calling to a SIP account that is configured to ring all handsets. Then when one handset answers, I only hear the panasonic, but they don't hear me. When I call to an extension that is configured to ring only one handset, I don't have this problem. When I use panasonic on both sides, or Tiptel - Panasonic, I don't have this problem as well. I am breaking my head what this could be. Any idea's? -- _ -- 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 -- _ -- 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] chan_sip sending from wrong source address when multiple interfaces are used
Kevin P. Fleming kpflem...@digium.com writes: That's quite interesting; can you describe a scenario where this occurs? Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24 and eth1 with 10.0.2.1/24. Further imagine that you wish to be able to move phones between the networks without changing the SIP server address, so you set 192.168.1.1 as the SIP server no matter which network they happen to be on. Now the phones which happen to be connected to eth1 will send a request to 192.168.1.1. If Asterisk is bound to 0.0.0.0, the reply will come from 10.0.2.1. This could be solved if Asterisk did a connect() to the socket and use the same socket for answering. That would tell the system IP stack that this is in fact a connection, and so the system would ensure that the reply source IP would be correct. Alas, few programmers are aware that you can even do connect() for UDP, and I believe it would be a rather large change to the Asterisk SIP stack to pass connected sockets around rather than just remembering IP addresses and port numbers. (Admittedly I haven't looked at that code in ages, so I could easily be wrong). The workaround is to explicitly bind to 192.168.1.1. Since Asterisk can bind to precisely one address, that kills off IPv6. /Benny -- _ -- 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] chan_sip sending from wrong source address when multiple interfaces are used
On 07/12/2012 09:19 AM, Benny Amorsen wrote: Kevin P. Fleming kpflem...@digium.com writes: That's quite interesting; can you describe a scenario where this occurs? Imagine you have a server with two interfaces, eth0 with 192.168.1.1/24 and eth1 with 10.0.2.1/24. Further imagine that you wish to be able to move phones between the networks without changing the SIP server address, so you set 192.168.1.1 as the SIP server no matter which network they happen to be on. Now the phones which happen to be connected to eth1 will send a request to 192.168.1.1. If Asterisk is bound to 0.0.0.0, the reply will come from 10.0.2.1. This could be solved if Asterisk did a connect() to the socket and use the same socket for answering. That would tell the system IP stack that this is in fact a connection, and so the system would ensure that the reply source IP would be correct. I must be missing something. If a phone sends a UDP packet to 192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1 interface on the Asterisk server? The only way I can imagine that happening is if a router in between the phone and the server has been told that 192.168.1.0/24 is reachable *through* 10.0.2.1, which seems like a bizarre way to construct a network. Getting replies from Asterisk *back* to the phone would also require the IP stack on the Asterisk server to route those replies back over the 10.0.2.0/24 interface instead of the 192.168.1.0/24, which doesn't make any sense either. chan_sip does have the ability to use connect()-ed sockets for dialogs now, since that is required for TCP, TLS and WebSocket support. It wouldn't be a huge leap to use them for UDP as well, if that was beneficial. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] chan_sip sending from wrong source address when multiple interfaces are used
On 07/11/2012 11:36 PM, Jeff LaCoursiere wrote:
This does exhibit the problem though. Your OS stack assumes one of
those addresses - the first identified interface? - is the one that all
replies will appear to come from. So phones on the 192.168.2.0/24
network that try to register get replies from 192.168.1.1 and ignore
them.
No, I don't think it does. If the server has four interfaces, on subnets
192.168.{1,2,3,4}.0/24, those are *not* overlapping, and everything will
work as expected. If a UDP packet is received on the third interface,
from an address reachable via routes over that interface, then the reply
to that packet will be sent out over that same interface, with the
source address set to the address assigned to that interface. Servers
are setup this way all the time, and it works as it should.
There must be more to the network configuration than something this
simple in order to cause the IP stack on the Asterisk server to choose
the wrong source IP address for outbound packets.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com www.asterisk.org
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Re: [asterisk-users] chan_sip sending from wrong source, address when multiple interfaces are used
I must be missing something. If a phone sends a UDP packet to 192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1 interface on the Asterisk server? The only way I can imagine that happening is if a router in between the phone and the server has been told that 192.168.1.0/24 is reachable *through* 10.0.2.1, which seems like a bizarre way to construct a network. Getting replies from Asterisk *back* to the phone would also require the IP stack on the Asterisk server to route those replies back over the 10.0.2.0/24 interface instead of the 192.168.1.0/24, which doesn't make any sense either. This can and will happen, if the multi-interface Asterisk server is also being used as the router between these networks. In this case, the packet sent by the client system to 192.168.1.1, is being sent by the client to the client's active (possibly default/only) routing gateway, which will be 10.0.2.1. The packet arrives at the server/router, and because it matches the IP address assigned to one of the server's network adapters (and Asterisk is bound to all of them) it's delivered to Asterisk. When Asterisk replies, it's doing so via a socket which is bound to 0.0.0.0. Since there's no specific IP address bound to this socket, the kernel has to pick one of the host's IP addresses to put into the packet it sends... and the one it picks is the one assigned to the network adapter on which it chooses to transmit the port. In this case, that will be 10.0.2.1. So, the client sends a packet to 192.168.1.1 and gets a response from 10.0.2.1, which is a huge WTF situation for many protocols. This is not just an issue for SIP. I've had exactly this same problem with IAX2 clients and Asterisk, and had to apply the exact same cure - I tell IAX2 to bind itself only to my host's externally-routable public IP address, and not to 0.0.0.0. Then, if I specify the public IP address as the server in each IAX2 client configuration, everything works fine. This is probably a not-unusual configuration if you're setting up a modest-size VoIP-only or VoIP-mostly network on a budget... a Linux or other Unix-ish system running Asterisk will often have enough CPU power to handle the RTP routing between networks, saving you the cost of a dedicated router. I have this very issue on my own system - a modest home network with a couple of internal LANs, some IP ranges set aside for VPNs of various sorts, and one externally routable IP address. chan_sip does have the ability to use connect()-ed sockets for dialogs now, since that is required for TCP, TLS and WebSocket support. It wouldn't be a huge leap to use them for UDP as well, if that was beneficial. Would be well worthwhile... and if you can port similar code over into chan_iax2, it would fix the problem there as well. -- _ -- 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] chan_sip sending from wrong source address when multiple interfaces are used
On 07/12/2012 12:38 PM, Freddi Hansen wrote: We have since Asterisk 1.2 been using a configuration with 6 NIC's bonding to 3 networks, one public internet and 2 private networks. Routing calls between networks and having phones on all 3 networks is no problem. There is one case though where we do fixup with iptables. We have 30 virtuel adresses on one of the private networks and when Asterisk sends a packet to a destination then the first address of the NIC is inserted as source by the OS. example one NIC has ip's 192.168.0.10,192.168.0.20,192.168.30 Telephone (192.168.0.100) sends a packet to Asterisk 192.168.0.30, Asterisk sends response to 192.168.0.100 but with source address 192.168.0.10 as thats the first ip on that NIC. In Iptables OUTPUT q we do a set-mark to an index into our source ip's then in POSTROUTING we insert the source adr using the mark Yes, this is the situation I referred to earlier. In your case, it's all on one interface, but the server has multiple addresses on the *same* network, and thus it cannot know (without help) with address should be used for outbound packets. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] Issue with a ticket system subscribed to asterisk-users
We realize there is an issue with a ticket system subscribed to the list and responding directly to member's posts. We are working on resolving the issue, please bear with us! Thanks, -- Rusty Newton Digium, Inc | Open Source Community Support Manager Check us out at: www.digium.com www.asterisk.org -- _ -- 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] chan_sip sending from wrong source address when multiple interfaces are used
Kevin P. Fleming kpflem...@digium.com writes: I must be missing something. If a phone sends a UDP packet to 192.168.1.1, how does that get routed to (arrive at) the 10.0.2.1 interface on the Asterisk server? The easiest way is that the Asterisk server itself is the router. Phones on 10.0.2.0/24 have 10.0.2.1 as default gateway. Another option is that there are no real routers; the phones are disconnected from the Internet, they still have 10.0.2.1 as default gateway but they only use it to reach the Asterisk server. The only way I can imagine that happening is if a router in between the phone and the server has been told that 192.168.1.0/24 is reachable *through* 10.0.2.1, which seems like a bizarre way to construct a network. I do not feel that the two scenarios above are particularly bizarre. Getting replies from Asterisk *back* to the phone would also require the IP stack on the Asterisk server to route those replies back over the 10.0.2.0/24 interface instead of the 192.168.1.0/24, which doesn't make any sense either. If the phone is on 10.0.2.0/24, the IP stack will route packets to it directly through the 10.0.2.1-interface by default. It actually takes quite serious contortions to make it send the packets elsewhere. Servers with multiple interfaces are a bit out of the ordinary. Right now Asterisk is difficult to work with on such servers, which is not a large problem for Asterisk in general, because they are so rare. You can always work around the problem either by creative routing or by explicitly binding to one address. chan_sip does have the ability to use connect()-ed sockets for dialogs now, since that is required for TCP, TLS and WebSocket support. It wouldn't be a huge leap to use them for UDP as well, if that was beneficial. It would be greatly appreciated :) It is low priority for the Asterisk project, as there are always workarounds. Extra points for making Asterisk support IP addresses appearing and disappearing... That would make VRRP/HSRP failover work. (It works if you bind to 0.0.0.0, but it is difficult to get Asterisk to use the VRRP-address as the source address for outgoing packets). /Benny -- _ -- 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] chan_sip sending from wrong source address when multiple interfaces are used
On 07/12/2012 03:53 PM, Benny Amorsen wrote: chan_sip does have the ability to use connect()-ed sockets for dialogs now, since that is required for TCP, TLS and WebSocket support. It wouldn't be a huge leap to use them for UDP as well, if that was beneficial. It would be greatly appreciated :) It is low priority for the Asterisk project, as there are always workarounds. I've just looked into this a bit, and I don't see how using connect() would actually solve the problem. If we receive a UDP datagram from a SIP endpoint, we could use socket() and connect() to create a socket specifically for sending to (and receiving from) that endpoint in the future, but we can't specify the source address to be used by that socket. The only way I know of to specify the source address for outbound packets is to use a raw socket and compose the IP header ourselves, which would be overkill. Benny, are you aware of some other method to accomplish this? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- 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] chan_sip sending from wrong source address when multiple interfaces are used
On Thursday 12 Jul 2012, Kevin P. Fleming wrote:
On 07/11/2012 11:36 PM, Jeff LaCoursiere wrote:
This does exhibit the problem though. Your OS stack assumes one of
those addresses - the first identified interface? - is the one that
all replies will appear to come from. So phones on the
192.168.2.0/24 network that try to register get replies from
192.168.1.1 and ignore them.
No, I don't think it does. If the server has four interfaces, on
subnets 192.168.{1,2,3,4}.0/24, those are *not* overlapping, and
everything will work as expected. If a UDP packet is received on the
third interface, from an address reachable via routes over that
interface, then the reply to that packet will be sent out over that
same interface, with the source address set to the address assigned
to that interface. Servers are setup this way all the time, and it
works as it should.
Precisely. In fact, if a packet from 192.168.2.n is received on /any/
interface, the response will always go out from the 192.168.2.X
interface. (Barring some weird routing/iptables configuration, of
course.)
There must be more to the network configuration than something this
simple in order to cause the IP stack on the Asterisk server to
choose the wrong source IP address for outbound packets.
I've usually seen this (wrong interface chosen for outbound) happen when
NAT is in the picture. However, the OP doesn't mention any NAT-related
configuration.
Regards,
-- Raj
--
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http://otheronepercent.blogspot.com || http://kandalaya.org || CC68
It is the mind that moves || http://schizoid.in || D17F
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[asterisk-users] How to Auto Answer a sip phone
Hi, i wanted to make dial plan in such a way that the any incoming call to the sip phone should auto answer.(auto pickup) . Help. regards Upendra -- _ -- 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] How to Auto Answer a sip phone
Different phones use different methods. What kind of sip phones do you have? On Jul 13, 2012, at 12:17 AM, upendra uppi...@gmail.com wrote: Hi, i wanted to make dial plan in such a way that the any incoming call to the sip phone should auto answer.(auto pickup) . Help. regards Upendra -- _ -- 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 -- _ -- 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
