Re: [asterisk-users] Strange beep during calls
Hi Felippe, in the past we had some trouble with a specific SNOM Firmware, which did not handle dtmf tones correctly. As a workarround, we tried to set relaxdtmf=yes in sip.conf. As a result we had these beep-tones generated randomly. Not shure, if this is your problem too... Friendly Regards, Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de/ http://www.gwsnettech.de mail mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] _ Von: Felippe Silvestre [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 6. August 2008 19:46 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Strange beep during calls Hi all, Our users are complaining about beeps that happen in the middle of some calls. They are similar to the sound heard you are in a call and press any button in your phone. Please find bellow some examples of these beeps(the recordings are in Portuguese, but the beeps are easy to identify): http://www.katizak.locaweb.com.br/asterisk/beep.mp3 http://www.katizak.locaweb.com.br/asterisk/beep.mp3 http://www.katizak.locaweb.com.br/asterisk/beep2.mp3 http://www.katizak.locaweb.com.br/asterisk/beep2.mp3 http://www.katizak.locaweb.com.br/asterisk/beep3.mp3 http://www.katizak.locaweb.com.br/asterisk/beep3.mp3 http://www.katizak.locaweb.com.br/asterisk/beep4.mp3 http://www.katizak.locaweb.com.br/asterisk/beep4.mp3 We are sure that our users are not pressing any button in the softphones during the conversations. Do you guys are able to identify where these beeps are coming from? Maybe an * functionality that we need to turn off... We are using Asterisk 1.4.21.2. Thanks. Felippe Silvestre ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Issues
Not shure if it helps here, but we had nearly the same dtmf problem in an asterisk 1.4 install with SNOM 320/360 phones. After hours of fiddling arround, we used relaxdtmf=yes in sip.conf and the problem went away. Guido _ Von: Ian [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 30. April 2008 11:34 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] DTMF Issues Hi all I am getting the feeling you are going to be hearing alot more from me in the near future. I have yet another issue here. This time its with DTMF (again). Ok the setup is as follows: We have a few Grandstream BT 200 phones and a few X-Lite phones. When dialing out to, say a bank, with the Grandstream the bank complains that it is unable to detect our DTMF tones. However when I do the same with the X-Lite it detects it correctly. Now I am in the process of implementing authentication into the system. This on the other hand detects Grandstream DTMF but not X-Lite DTMF. The Grandstream phones have been set up to send their DTMF via sip info (I set it up that way in order to use the *2 transfer feature). I have tried Zoiper and when I set it up to also send DTMF via sip info, the asterisk detects the DTMF correctly. Can anyone tell me what I can do to enable the DTMF to work? Thanks in advance. Ian -- www.vddi.co.za http://www.vddi.co.za/ I Coetzee IT Tegnikus Telefoon: 012 664 2300 Selfoon : 079 522 6519 Faks: 012 644 2902 E-pos : [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] Skype : vddb_igcoetzee ___ -- 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] Asterisk SNOM and DTMF
For information purposes: We had problems with proper dtmf recognition. Asterisk Version: SVN-branch-1.4-r114083 SNOM FW: 7.1.30 With the above constellation dtmf tones sometimes worked and sometimes not. Already a few hours and much testings later, we found and used the configuration parm relaxdtmf. In the past we diddn't use this parameter at all without any dtmf problems. Perhaps a silent change in asterisk's way of handling dtmf? With relaxdtmf=yes in sip.conf's general context, dtmf tones seem to work reliable. Not shure about other magic side-effects, but it helped. In the hope of saving someone else a few hours... Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mail[EMAIL PROTECTED] ___ -- 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] Queues +Exiting
-Ursprüngliche Nachricht-
Von: Rob Schall [mailto:[EMAIL PROTECTED]
Gesendet: Mittwoch, 9. April 2008 15:50
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: [asterisk-users] Queues +Exiting
I'm having a problem getting my queue to function as it should.
After 20 seconds or so, it should prompt the user with a
message thanks
for holding. press # to leave a message or stay on the line to
continue holding. I set up the context in the queues.conf
file, so if
a user presses a digit, they should be able to leave. But I get a SIP
BUSY message.
Here are my confs:
queues.conf
[custserv]
music=default
strategy=ringall
;timeout=10
retry=20
wrapuptime=0
maxlen=0
context = queue-out
periodic-announce=cont_holding
periodic-announce-frequency=15
;announce-frequency=15
;announce-holdtime=yes
member = SIP/2001
member = SIP/2002
member = SIP/1004
extensions.conf
[queue-out]
exten = s,1,Voicemail(u${vmbox})
exten = s,2,Hangup
Perhaps it's the s extension, did you try with
exten = 1,1,Voicemail(u${vmbox})
exten = 1,2,Hangup
Regards,
Guido
gwsNetTech
Guido Hecken
Quirrenbacher Str. 36
53639 Königswinter
Germany
fon +49(2244) 870663
fax +49(2244) 870664
mobil +49(179) 1267353
web http://www.gwsnettech.de
mailto:[EMAIL PROTECTED]
___
-- 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] Parked calls and callerid
Hi list,
sorry if this has been discussed in the past and is also posted twice to the
list,
but I couldn't find anything wise about it.
Since we had some trouble with the builtin hold function of some (all?) SNOM
320/360
phones, we decided to use the call parking feature in asterisk instead.
Assume, a call comes in with CALLERID(num) 1234567 for extension 10.
Extension 10 parks this call into 801, dials extension 11 and asks if she/he
could
fetch the call on parkposition 801.
Extension 11 dials 801 and get's the call and can only see 801 in the phones
display.
So, how could we get the original CALLERID(num) 1234567 back in the phones
display?
Using a channel variable or use astdb comes in mind, but what is the best
way to achieve
this?
In traditional pbx systems this seems to be a standard function.
Any ideas on how do handle these callerids?
Here some of the involved configs
---
in extensions.conf:
[parkedcalls]
exten = 80[1-5],1,NoCDR()
exten = 80[1-5],2,ParkedCall(${EXTEN})
in features.conf:
[general]
parkext = 800
parkpos = 801-805
context = parkedcalls
parkingtime = 60
-
We are using SVN-branch-1.4-r96449 and other, older versions of asterisk
Kind regards,
Guido
gwsNetTech
Guido Hecken
Quirrenbacher Str. 36
53639 Königswinter
Germany
fon +49(2244) 870663
fax +49(2244) 870664
mobil +49(179) 1267353
web http://www.gwsnettech.de/ http://www.gwsnettech.de
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk parked calls and callerid
Hi list,
sorry if this has been discussed in the past, but I couldn't find anything
wise about it.
Since we had some trouble with the builtin hold function of some (all?) SNOM
320/360
phones, we decided to use the call parking feature in asterisk instead.
Assume, a call comes in with CALLERID(num) 1234567 for extension 10.
Extension 10 parks this call into 801, dials extension 11 and asks if she/he
could
fetch the call on parkposition 801.
Extension 11 dials 801 and get's the call and can only see 801 in the phones
display.
So, how could we get the original CALLERID(num) 1234567 back in the phones
display?
Using a channel variable or use astdb comes in mind, but what is the best
way to achieve
this?
In regular pbx systems this seems to be a standard function.
Any ideas on this?
Here some of the involved configs
---
in extensions.conf:
[parkedcalls]
exten = 80[1-5],1,NoCDR()
exten = 80[1-5],2,ParkedCall(${EXTEN})
in features.conf:
[general]
parkext = 800
parkpos = 801-805
context = parkedcalls
parkingtime = 60
-
We are using SVN-branch-1.4-r96449 and other, older versions of asterisk
Regards
Guido
gwsNetTech
Guido Hecken
Quirrenbacher Str. 36
53639 Königswinter
Germany
fon +49(2244) 870663
fax +49(2244) 870664
mobil +49(179) 1267353
web http://www.gwsnettech.de/ http://www.gwsnettech.de
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk parking hold and transferdigittimeo ut
-Ursprüngliche Nachricht-
Von: Mojo with Horan Company, LLC [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 25. März 2008 23:23
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Asterisk parking hold and
transferdigittimeout
It seems that the dialplan comes into play. If your parking
lot is 700,
and you have any extension patterns that COULD begin with that, then
asterisk will wait to make SURE you're not typing 700:
Let's say that 700 is my parking lot extension.
exten = _NXXNXX,1,blahblahblah
This could match 7005551212, so asterisk waits around to make
sure I'm
not trying to find any more buttons before it accepts that I
meant 700.
As an example, if your parking lot extension was **, then
asterisk could
be pretty darn sure that that won't match anything else, and
will accept
it directly as a number to transfer too.
SOLUTION ###
Thanks for the tip, it was really the dialplan. In our * installations we
have an
outgoing context, named capi-out starting with this:
[capi-out]
exten = _XXX.,1,DoSomethingReallyImpressive()
...
After I changed it to:
[capi-out]
include = notfall ; special context for 3-digit emergency numbers
exten = _.,1,DoSomethingReallyImpressive()
...
[notfall]
exten = _11X,1,Dial(CAPI/ISDN3/${EXTEN}/b,60,tT)
...
BTW these includes are really magic, cause sometimes they don't do what you
(especially I) expext.
Please take a look at this:
EXAMPLE ###
;DIALPLAN
...
[capi-in]
include = capi-in-sub
exten = _955623XX,1,DoSomethingReallyImpressive()
...
[capi-in-sub]
exten = 9556230,1,DoSomethingReallyImpressive()
exten = 95562315,1,DoSomethingAnybodyWouldExpect()
...
Now, what happens:
Call for 9556230 reaches capi-in, is redirected through include statement to
capi-in-sub and executed.
So far so fine, expected behaviour.
Call for 95562315 reaches capi-in and is executed direct, the include
directive isn't executed at all!
Why?
Through the include statement, asterisk has to look first in capi-in-sub,
there it should
find this extension:
exten = 95562315,1,DoSomethingAnybodyWouldExpect()
...
and follow the dialplan under capi-in-sub since a valid extension was found.
What's wrong, any ideas?
Regards,
Guido Hecken
gwsNetTech
Guido Hecken
Quirrenbacher Str. 36
53639 Königswinter
Germany
fon +49(2244) 870663
fax +49(2244) 870664
mobil +49(179) 1267353
web http://www.gwsnettech.de
mailto:[EMAIL PROTECTED]
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk parking hold and transferdigittimeo ut
-Ursprüngliche Nachricht- Von: Jared Smith [mailto:[EMAIL PROTECTED] Gesendet: Mittwoch, 26. März 2008 13:01 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Asterisk parking hold and transferdigittimeout On Wed, 2008-03-26 at 12:30 +0100, Guido Hecken wrote: Now, what happens: Call for 9556230 reaches capi-in, is redirected through include statement to capi-in-sub and executed. So far so fine, expected behaviour. Call for 95562315 reaches capi-in and is executed direct, the include directive isn't executed at all! Why? Through the include statement, asterisk has to look first in capi-in-sub, there it should find this extension: exten = 95562315,1,DoSomethingAnybodyWouldExpect() ... and follow the dialplan under capi-in-sub since a valid extension was found. What's wrong, any ideas? This is a very popular misconception regarding include statements, so let me try to reiterate how includes work. When a call comes into a particular context, Asterisk looks for the following items: 1) Exact matches in the current context. If a match is found, Asterisk will not continue searching. 2) Pattern matches in the current context. If at least one pattern match is found that matches the dialed extension, Asterisk will not continue searching (even if a better match is included . 3) Any switch = statements. These can be used for remote dialplan lookups, realtime dialplan lookups, DUNDi, etc. If Asterisk finds at least one match, it will not continue searching. 4) Any included contexts. These will be followed in order, and for each included context, this same list will be applied. In your example above, you somehow assumed that the included context would be searched, even though Asterisk already found a match in the current context. Hi Jared, thanks for your general explanations on using contexts and includes. Since I have your book in my rack, I really should have done some better reading and obviously understanding ;-) Homework done, lesson learned! Regards, Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk parking hold and transferdigittimeo ut
According to this post: http://lists.digium.com/pipermail/asterisk-dev/2007-April/027281.html Includes are tacked on to the end of the dialplan they are mentioned in, not where they stand. So, since your exten = _955623XX,1,DoSomethingReallyImpressive() matches, asterisk doesn't need to even bother checking the included context. Moj Hi Moj, thanks for pointing this out. In the meantime I've learned how to use contexts and includes ;-) Jared's post, some hours ago, was very helpful. Regards, Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] ___ -- 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] Asterisk parking hold and transferdigittimeout
Hi, anyone out there with the same problems and a possible solution to the following? The functions callparking and hold use the same transferdigittimeout in features.conf. While I think 3 to 5 seconds are enough to let the user find their keys on the phone, the double ammount of time ( 2 x 5 secs) you have to wait before a call is parked and the parkposition is announced, is really too long. Did I miss something in the documentation? We are using SVN-branch-1.4-r96449. Regards, Guido Hecken gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] [OT] Wifi SIP phon es - LinkSys WIP330
-Ursprüngliche Nachricht- Von: Wayne [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 28. Dezember 2006 22:20 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] [OT] Wifi SIP phones - LinkSys WIP330 Hi List, Hope everyone is recovering from the festive season :) (ok we still have new years i guess!) Anyways, I was wondering if anyone has had any successful dealings with WiFi phones and operation with '*' at all? I've been keeping my eye on the LinkSys WIP330 ( http://preview.tinyurl.com/nccxn ) and wondered your collective thoughts? Would I be correct in thinking that (as long as the relevant ports were open on the firewall) it would be possible to still be an extension to * if you could access the internet from, say, a wifi hot spot that was not a part of the lan? Thanks Wayne We tried the Siemens Gigaset SL75 W-LAN in a customer's asterisk installation. Voice quality is superb, standbytime and range are ok, looks really convincing. Pricing about 169.- € Regards Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Patton 1400
Hi Kevin, you have to create a gateway in the Smart Node: gateway sip sip bind interface eth1 router service default domain gwsnettech.local realm gwsnettech.local authentication isdngw2 password huffvtzddzdjkhuztztufuz== encrypted default default-server hallinux2.gwsnettech.local 5060 loose-router registrar hallinux2.gwsnettech.local 5060 In sip.conf, something like this: [isdngw2] type=friend username=isdngw2 secret=the_unencrypted_password_from _above host=192.168.161.135 ;host=dynamic fromuser=gwsnettech fromdomain=gwsnettech.local nat=no context=isdngw2-in canreinvite=no If you need a complete running config, I can send it to you offlist. Hope, it helps... Guido Von: Kevin Withnall [mailto:[EMAIL PROTECTED] Gesendet: Samstag, 4. November 2006 05:58 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Patton 1400 I have a patton 1400 setup to handle the bri interface. As a trixbox user, I wanted a sip trunk rather than having to re-compile bri support into trixbix. Anyway, I have it working now so that asterisk can make calls and they are passed properly to the telephone network. Incoming calls however are another matter. I have (after turning on cli debug in the 1400) determined that its getting stuck in the routing system. I don't know what destination to make to get it to sip connect to the asterisk box. Ive tried making an interface that has a remote address of the asterisk box but that doesn't work. Can someone send me a config for anything they have done that is similar to this ? I believe the Patton 1200 is also the same unit apart from extra ports. Any help would be greatly appreciated. Regards Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: ASTTAPI
-Ursprüngliche Nachricht- Von: Tomislav Parcina [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 28. September 2006 09:10 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Re: ASTTAPI In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Has anyone actually gotten ASTTAPI to work? I can't seem to get it to work, yet I have other TAPI setups (SNAP and xtelsio) working fine. I have noticed that SNAP and Xtelsio act differently. Etelescript is the application that will be calling TAPI. Hi Mike! I have been using ASTTAPI, but it takes time to configure it and I'm not sure it's developing any more. Now I'm using SNAP for several days but it seams that it has some bugs. I'm using Snap's forum to check with developer about this, but it's going slowly. I don't think that Snap is for business production yet. If developer doesn't solve those problems with Snap, I'll try Etelescript. Is Etelescript free? Is it open source? After spending many many hours on asttapi and other tapisolutions, we found Tapi for Asterisk here: http://www.phonesuite.de/de/produkte/ast_tsp/phonesuite_tapi_for_asterisk.ht m It works like a charm and the licensefee with 25.-€/10 Clients is really fair. We couldn't find any bugs in the software and in combination with tapicall www.tapicall.de it's our preferred link-up to Outlook/Exchange in all of our asterisk installations. Since it's language is in german, you might have a closer look on some german dictionaries, but after configuration is done (5 minutes) you can forget about ever installed it. ;-) Hope, these informations saved you some time, money and nerves Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0
Hi List, is there a known problem compiling chan-capi-0.7.0 against asterisk branch 1.4? System: Fedora Core 4 with Kernel 2.6.17-1.2142_FC4 AVM Fritz Card is present and fcpci running and up isdn4k-utils and isdn4k-utils-devel installed capi4hylafax installed make in chan_capi source said: gcc -pipe -fPIC -Wall -Wmissing-prototypes -Wmissing-declarations -g -I/usr/include -D_REENTRANT -D_GNU_SOURCE -O6 -march=i686 -Wno-missing-prototypes -Wno-missing-declarations -DCRYPTO -c -o chan_capi.o chan_capi.c chan_capi.c:146: warning: type defaults to 'int' in declaration of 'STANDARD_LOCAL_USER' chan_capi.c:146: warning: data definition has no type or storage class chan_capi.c:147: warning: type defaults to 'int' in declaration of 'LOCAL_USER_DECL' chan_capi.c:147: warning: data definition has no type or storage class chan_capi.c: In function 'capi_new': chan_capi.c:2078: error: 'struct ast_channel' has no member named 'type' chan_capi.c: In function 'pbx_capicommand_exec': chan_capi.c:4582: warning: implicit declaration of function 'LOCAL_USER_ADD' chan_capi.c:4597: warning: implicit declaration of function 'LOCAL_USER_REMOVE' chan_capi.c: At top level: chan_capi.c:5244: error: unknown field 'send_digit' specified in initializer chan_capi.c:5244: warning: initialization from incompatible pointer type make: *** [chan_capi.o] Error 1 Thanks for any hints and ideas Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0
-Ursprüngliche Nachricht- Von: Armin Schindler [mailto:[EMAIL PROTECTED] Gesendet: Dienstag, 26. September 2006 13:37 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] asterisk 1.4 branch and chan-capi-0.7.0 On Tue, 26 Sep 2006, Guido Hecken wrote: Hi List, is there a known problem compiling chan-capi-0.7.0 against asterisk branch 1.4? chan-capi was not ported to Asterisk 1.4 yet. See bug http://bugs.melware.net/mantis/view.php?id=20 Armin Armin, thanks for the info. Are there any plans on porting it to 1.4 and if yes, is there an approximate release date? Regards, Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] fast SIP failover (outgoing sIP requests) wi th 1.2
-Ursprüngliche Nachricht-
Von: Klaus Darilion [mailto:[EMAIL PROTECTED]
Gesendet: Dienstag, 19. September 2006 16:03
An: asterisk-users@lists.digium.com
Betreff: [asterisk-users] fast SIP failover (outgoing sIP requests) with
1.2
Hi!
I have the following problem: I route calls from one office to the other
office via SIP, but if for any reason this SIP call fails, I want a
backup route via the PSTN.
Thus, I use:
exten = _[1-9].,4,Dial(SIP/${enumresult},90)
exten = _[1-9].,5,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?103:6)
exten = _[1-9].,6,GotoIf($[${DIALSTATUS} = CONGESTION]?103:7)
exten = _[1-9].,7,Hangup
exten = _[1-9].,103,Dial(ZAP/g1/${EXTEN},90)
The problem is, if the SIP server at the remote office is down, thus no
responses to the INVITE, it takes 64 seconds to timeout.
Is there a method to shorten this interval - e.g. if there is no
response within 10 seconds give up - without changing the hardcoded
retransmission value (6) in chan_sip ?
regards
klaus
Hi,
maybe I'm wrong, but what about using the ChanisAvail function?
We did something like this in a customer installation:
exten = _XXX.,1,Set(LANGUAGE()=de)
exten = _XXX.,2,ChanisAvail(CAPI/ISDN3/${EXTEN},s)
exten = _XXX.,3,Dial(CAPI/ISDN3/${EXTEN}/b,60,tT)
exten = _XXX.,4,Congestion
exten = _XXX.,103,ChanisAvail(CAPI/ISDN2/${EXTEN},s)
exten = _XXX.,104,Dial(CAPI/ISDN2/${EXTEN}/b,60,tT)
exten = _XXX.,105,Congestion
Hope, it helps ...
Guido
gwsNetTech
Guido Hecken
Quirrenbacher Str. 36
53639 Königswinter
Germany
fon +49(2244) 870663
fax +49(2244) 870664
mobil +49(179) 1267353
web http://www.gwsnettech.de
mailto:[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Fedora
Hi, we're using Fedora Core 3-5 on all of our customer asterisk installations, as well as on other projects like mythtv, mailservers and general servers without any problems. We like Fedora's bleeding edge state. Guido Hi list; Does asterisk work with fedora because redhat enterprise is licensed and costly. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Server Down
Hi, obviously asterisk doesn't start with the installed(?) start script. Try to start it manually and watch the cli for informations with asterisk -vvvc AFAIK a make config in the asterisk source should install the start script for your system. Hope it helps... Guido Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Gesendet: Sonntag, 17. September 2006 15:27 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Asterisk Server Down I rebooted the server on which the Asterisk is hosted on. The * did not come back up and I get this message when I attempt to use CLI [EMAIL PROTECTED] ~]# asterisk -r Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk Server Down
-Ursprüngliche Nachricht- Von: Tzafrir Cohen [mailto:[EMAIL PROTECTED] Gesendet: Sonntag, 17. September 2006 15:56 An: asterisk-users@lists.digium.com Betreff: Re: [asterisk-users] Asterisk Server Down On Sun, Sep 17, 2006 at 03:54:46PM +0200, Guido Hecken wrote: Hi, obviously asterisk doesn't start with the installed(?) start script. Try to start it manually and watch the cli for informations with asterisk -vvvc One warning: if your system is normally configured to run as non-root, this may cause it to write some fiels as root, and not start properly next time you start it with the standard script. With the Debian packages, use: /etc/init.d/asterisk debug Which is normally just a glorified: asterisk -U asterisk -vv Tzafrir, you're right, one should proof, under which user asterisk runs... Besides security reasons, running asterisk as root, doesn't it allow a higher prioritization of asterisk processes? Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk 1.2 and SATA drives
-Ursprüngliche Nachricht- Von: Tharanga [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 8. September 2006 07:35 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Asterisk 1.2 and SATA drives Greeitngs !, I am haivng asterisk 1.0.x verison and going to upgrade it to version 1.2.4. with zaptel 1.2.8. i have PATA drives. asterisk 1.0.x verison didnt support SATA (IRQ problems). so..this new relaeasr 1.2.x can support SATA drives on dual core processor ?? iam using TDM04B card. hope u guys can comment on this.. thxs in advance Tharanga Hi, AFAIK, this isn't an asterisk related problem. We used SATA Disks in early 0.9 and 1.0 Asterisk versions without any problems. I think it has to do with OS and Kernelversions and the hardware you have chosen. Sometimes ago we set up an asterisk server with sata2 (3 Gbit/s) and had permanent kernel panics. After setting the jumpers on the disk to 1,5 Gbit/s transfer rate, every thing worked great. Hope, it helps... Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Incoming call problem-calling part is busy(I PKall)
Von: Crazy Boy [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 7. September 2006 14:25 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Incoming call problem-calling part is busy(IPKall) Hi, I have registered with IPKall ang got the number i.e., 206XXX. When I call to this number, It is telling that The party you are calling is currently busy. Here I am giving my config details. When I registered with IPKall, I entered these below values: SIP Phone number: 7312567 SIP Proxy: voip-co1.teliax.com Contents in sip.conf file: [7312567] type=peer dtmfmode=rfc2833 context=inbound insecure=very host=voiper.ipkall.com Contents in extensions.conf file: [inbound] exten = 7312567,1,Dial(SIP/250,20) include = internal Here, 250 is the SIP account. I have given my total configuration. Please tell me the solution. Looking forward to your response. Thank you. Hi, might be I'm wrong, but you need a at least a register statement in the general section in your sip.conf register = USER-ID:[EMAIL PROTECTED]/USER-ID Hope, it helps... Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Experience Patton BRI gateways and Asterisk?
-Ursprüngliche Nachricht- Von: Koopmann, Jan-Peter [mailto:[EMAIL PROTECTED] Gesendet: Dienstag, 5. September 2006 13:54 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [asterisk-users] Experience Patton BRI gateways and Asterisk? Hi, can anybody comment on patton inalp voice gateways and Asterisk? How good is there echo cancellation? How good is the interop with Asterisk? I am especially looking for reports on 4630 and 45xx series with BRI. Hi JP, we used the Smartnodes 1400 and the Smartnodes 2300 in the past. Echo canceler is great and they work really rock stable. Good Support from Patton/Inalp was included. You get many functions for your money, (integrated DSL-Router,QOS,SIP/H323 Support etc.). But eventuallly you pay for functions, you don't really need. I found the BRI Cards from Gerdes Primux2S0/Te/NT and Primux4S0/Te/NT work great and you have to configure only one device, your Asterisk. Another benefit of ISDN cards I see in handling the ISDN-Ports direct in Asterisk, in your dialplan. This gives you a more flexible way of call routing. BTW, multiple Primux Cards in one system are supported! On the other hand, you have some kind of backup by using the Smartnodes, if your asterisk dies. Hope, the informations are usefull Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Queue timeout problems
-Ursprüngliche Nachricht- Von: Mr. Jones [mailto:[EMAIL PROTECTED] Gesendet: Sonntag, 3. September 2006 06:10 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [asterisk-users] Queue timeout problems Thanks Guido - I tried that and still have the same problem. The call never seems to leave the queue. Any other ideas? Hmm, to have a closer look on the problem, one could do the following Activate debugging, error and verbose logging in logger.conf by having a line like this: console = notice,warning,error,debug,verbose Open the cli and do a logger reload set verbose to 5 or even 255 Initiate a call to the queue and watch for errors/informations. Perhaps, define a context named test and put a really simple command in it. Something like this [test] exten = 120,1,Answer() exten = 120,2,Playback(some-sound-file) exten = 120,3,Hangup Change your queue to call this context in the second priority. Also have a closer look on your include commands in the dialplan... Normally an extensions reload on the cli should activate the changes to the dialplan, but with a restart now you should be save. Good luck Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Queue timeout problems
-Ursprüngliche Nachricht- Von: Mr. Jones [mailto:[EMAIL PROTECTED] Gesendet: Sonntag, 3. September 2006 01:12 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] Queue timeout problems Hi Folks, I'm trying to use the Queue feature to essentially implement a multiple call appearance situation for some of our executives. Essentially I have a queue defined per executive like: exten=9495551212,1, Queue(stever|tTr|||25) exten=9495551212,2, Goto(druid-users,1212,1) Give these settings a try: exten=9495551212,1,Wait(2) exten=9495551212,2,(Playback(some-announce) ; could be an empty sound file exten=9495551212,3,Queue(stever|tT|||60) ; try without option r exten=9495551212,4,Goto(druid-users,1212,1) [stever] strategy=ringall context=druid-default joinempty=yes member= SIP/1200 member= SIP/1201 member= SIP/1212 timeout=15 So the user hits the queue ok, but they never fallout to the 2nd priorty, which has macros for follow-me, and handles the voice mail. What happens to the call instead? Dropped, endless in the queue? in queues.conf I have [stever] strategy=ringall context=druid-default joinempty=yes member= SIP/1200 member= SIP/1201 member= SIP/1212 timeoutreset = no Hope, it helps Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Snom Function keys
Having an issue with Snom function keys. I've tried it all doesn't work...
Extension has a voicemail message as the MWI is flashing. Hits Retrieve
button, automatically goes to Comedian mail and is prompted for his
username...
What I'd like to do is have Comedian stop being a Comedian, recognize
that extension and simply ask for the password. (and thanks to those who
responded before but nothing seems to work...)
Updated all my phones to 6.2.3 and tried the following unsuccessfully:
You should have something like this in your default section of
extensions.conf
exten = asterisk,1,VoiceMailMain(s${CALLERIDNUM})
leave the Retrieve button on it's default:
RETRIEVE:Key EventF_RETRIEVE
RETRIEVE:Speed Dialsip:[EMAIL PROTECTED];user=extension_number
RETRIEVE:Speed Dialsip:[EMAIL PROTECTED];username=extension_number
RETRIEVE:Speed Dialsip:[EMAIL PROTECTED];user=extension_number
RETRIEVE:Speed Dialsip:[EMAIL PROTECTED];username=extension_number
RETRIEVE:Speed Dialsip:[EMAIL PROTECTED];user=extension_number
RETRIEVE:Speed Dialsip:[EMAIL PROTECTED];username=extension_number
RETRIEVE:Key EventF_RETRIEVE
Nothing seems to work. Any suggestions?
And for the sidecar, same applies... I have a sidecar on one phone, and
I wanted that sidecar to show me who is on the phone...
Do you have something like this in the context of the snoms in
extensions.conf?
exten = 10,hint,SIP/10,name
This works for us in all * installs.
Hope, it will help
Guido
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changes in handling anonymous calls entering ast erisk
Hi list,
after too much time of googling and trial and error, I need some help.
In older Asterisk Versions 0.9 - 1.0 (Asterisk CVS-HEAD-02/13/05-15:26:28)
we used this setup:
extensions.conf
exten = 876779,1,AGI,reverse.agi| ${CALLERIDNUM}
exten = 876779,2,SetCIDName(Privat ${LONGNAME})
exten = 876779,3,SetCIDNum(${CALLERIDNUM})
exten = 876779,4,Dial(SIP/6303SIP/6301SIP/6302SIP/6304,120,tTo)
exten = 876779,5,SetLanguage(de)
exten = 876779,6,Voicemail(6301)
exten = 876779,7,Hangup
Everything works great, calls coming in with an existing CIDNUM are shown on
the display (SNOM 360) with Privat somename somenumber.
Calls without CIDNUM are displayed Privat anonymous.
The Problem in a newer Asterisk 1.2 Version (SVN-branch-1.2-r30874 built by
root @ x on a i686 running Linux on 2006-05-31 12:22:14 UTC):
With the above config, all Clients (SNOM 190, SNOM 360, SJPhone) only show
unknown for both, name and number, if the CIDNUM sent from the ISDN-Card is
empty. Although the CDRs seem to be correct after tweaking with some
GotoIf($[${CALLERID(num)} = ...and do some SetCALLERID(num)=anonymous,
it only works, if a CIDNUM is sent from the ISDN.
CDR With CIDNUM and SNOM Display ok:
,02244872000,2507017,capi-in,Privat-gwsNetTech Guido Hecken
02244872000,CAPI/ISDN1/2507017-44,SIP/19-1edc,Dial,SIP/19|65|tT,
2006-08-28 16:05:22,,2006-08-28 16:05:26,4,0,NO ANSWER,DOCUMENTATION
CDR Without CIDNUM and SNOM Display unknown:
,anonymous,2507017,capi-in,Privat-anonymous
anonymous,CAPI/ISDN1/2507017-4e,SIP/19-9ee0,Dial,SIP/19|65|tTo,2
006-08-28 16:34:59,,2006-08-28 16:35:03,4,0,NO ANSWER,DOCUMENTATION
I also noticed, that the Dial String displayed in Sjphone changed between
the older and newer * versions from sip:[EMAIL PROTECTED] to
sip:[EMAIL PROTECTED] .
Any help would be really great
Guido
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Changes in handling anonymous calls entering asterisk
FYI and to answer my own question,
SetCallerPres(allowed) did the trick.
exten = msn,n,SetCallerPres(allowed)
Background:
If a call arrives with the flag CallerPres() set to prohib, you can change
the CALLERID(num) or CALLERID(name) with the corresponding set function and
the cdrs look good, but the sip channel still has something like
sip:[EMAIL PROTECTED] in the from field. So the Phone/Softphone only
reports about Unknown number and Unknown name.
Very annoying, but solved :-)
Hope, the above will help someone in the future
Guido
Betreff: [asterisk-users] Changes in handling anonymous calls entering ast
erisk
Hi list,
after too much time of googling and trial and error, I need some help.
In older Asterisk Versions 0.9 - 1.0 (Asterisk CVS-HEAD-02/13/05-15:26:28)
we used this setup:
extensions.conf
exten = 876779,1,AGI,reverse.agi| ${CALLERIDNUM}
exten = 876779,2,SetCIDName(Privat ${LONGNAME})
exten = 876779,3,SetCIDNum(${CALLERIDNUM})
exten = 876779,4,Dial(SIP/6303SIP/6301SIP/6302SIP/6304,120,tTo)
exten = 876779,5,SetLanguage(de)
exten = 876779,6,Voicemail(6301)
exten = 876779,7,Hangup
Everything works great, calls coming in with an existing CIDNUM are shown
on
the display (SNOM 360) with Privat somename somenumber.
Calls without CIDNUM are displayed Privat anonymous.
The Problem in a newer Asterisk 1.2 Version (SVN-branch-1.2-r30874 built
by
root @ x on a i686 running Linux on 2006-05-31 12:22:14 UTC):
With the above config, all Clients (SNOM 190, SNOM 360, SJPhone) only show
unknown for both, name and number, if the CIDNUM sent from the ISDN-Card
is
empty. Although the CDRs seem to be correct after tweaking with some
GotoIf($[${CALLERID(num)} = ...and do some SetCALLERID(num)=anonymous,
it only works, if a CIDNUM is sent from the ISDN.
CDR With CIDNUM and SNOM Display ok:
,02244872000,2507017,capi-in,Privat-gwsNetTech Guido Hecken
02244872000,CAPI/ISDN1/2507017-44,SIP/19-1edc,Dial,SIP/19|65|tT,
2006-08-28 16:05:22,,2006-08-28 16:05:26,4,0,NO
ANSWER,DOCUMENTATION
CDR Without CIDNUM and SNOM Display unknown:
,anonymous,2507017,capi-in,Privat-anonymous
anonymous,CAPI/ISDN1/2507017-4e,SIP/19-9ee0,Dial,SIP/19|65|tTo,2
006-08-28 16:34:59,,2006-08-28 16:35:03,4,0,NO ANSWER,DOCUMENTATION
I also noticed, that the Dial String displayed in Sjphone changed between
the older and newer * versions from sip:[EMAIL PROTECTED] to
sip:[EMAIL PROTECTED] .
Any help would be really great
Guido
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] IP phone with 2 ethernet jacks
We like the SNOM 360 Phones. They have really good features. Guido -Ursprüngliche Nachricht- Von: Mindaugas Kuprys [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 25. August 2006 09:40 An: asterisk-users Betreff: [asterisk-users] IP phone with 2 ethernet jacks Hi, Can anyone suggest good quality IP phone with 2 Ethernet jacks. I wanted Sipura but they don't have such product. Thanks, Mindaugas ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Unable to receive Incoming calls to my DID. Please tell me the solution
Hi, you have in your sip.conf: register = xyz.abc:[EMAIL PROTECTED] This register command doesn't tell asterisk what to do with it. Take for example this register command and other definitions in sip.conf: register = sipgate-id:[EMAIL PROTECTED]/sipgate-id and this peer definition [sipgate] type=friend username=sipgate-id secret=password host=sipgate.de fromuser=sigate-id fromdomain=sipgate.de nat=yes context=incomingsipgate qualify=yes defaultexpirey=no canreinvite=no insecure=very and in extensions.conf a context named incomingsipgate [incomingsipgate] exten = sipgate-id,1,DIAL(SIP/1234,60,tT) exten = sipgate-id,2,SetLanguage(de) exten = sipgate-id,3,Voicemail(1234) exten = sipgate-id,4,Hangup With the above, incoming calls from sipgate are routed to the context incomingsipgate and are processed by a simple dial command, which forwards the call to internal extension 1234. Hope, it helps... Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] How to connect Snom softphone from my home?
Open Firewallports in Office: 5060, 1-2 Edit sip.conf: ;nat externip=xxx.xxx.xxx.xxx externrefresh=10 localnet=192.168.0.0/255.255.0.0 ; fit to your net nat=no [19] type=friend username=19 secret= ;canreinvite=no host=dynamic disallow=all ;allow=g723 allow=alaw allow=ulaw allow=gsm dtmfmode=rfc2833 mailbox=20 context=your main context callgroup=1 pickupgroup=1 group=2 nat=yes callerid = Someone 19 Don't know much about SNOM Softphones, I like the real ;-) but things from sjphone should be straight equal: Proxydomain: Extern IP from Asterisk Userdomain: Extern IP from Asterisk Port:5060 Initialize: Account: 19 Passwort: You need a natted account in sip.conf, like the above 19 Hope it helps... Guido Von: Crazy Boy [mailto:[EMAIL PROTECTED] Gesendet: Freitag, 4. August 2006 12:17 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] How to connect Snom softphone from my home? Hi Friends, We have installed Asterisk in our office and using it successfully. I have given public IP to our Asterisk server. We are using Snom360 5.3 softphone for communication. I tried to connect to our Asterisk server with my Snom360 5.3 softphone from my house. But, it is not connecting. How can I connect from my house to my Asterisk server through Snom softphone? This is very urgent. Looking forward to your kind response. Thank you. Regards, Chandra. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] AgentCallBackLogin+Queue
this asks only for a password
exten = 123,1,AgentCallbackLogin(${CALLERIDNUM},${EXTEN})
hope it helps...
Guido
-Ursprüngliche Nachricht-
Von: Gleidson Antonio Henriques [mailto:[EMAIL PROTECTED]
Gesendet: Samstag, 8. April 2006 18:38
An: Asterisk-Digium
Betreff: [asterisk-users] AgentCallBackLogin+Queue
Hi all,
I´m begginner with asterisk and i need to setup one Support Call
Center.
First of all,
I want to authenticate my users in call center with AgentCallBackLogin
or something similar and the tranfer the Logged Agent to main queue.
I play with some setups from www.voip-info.org.
But in all of them I have the same problem.
Call center user dial 123
AgentCallBackLogin prompt for User and Password
I enter the same user and password that i have in my agents.conf
AgentCallBackLogin prompt for a new extension That´s my
problem...
Is there some way to cancel that prompt for a new extension ? I don´t
wanna give a valid extension to attendant, i only wanna put they on the
main
queue.
I´ll put my configs below just for checking.
Any suggestions are pretty welcome.
Thanks in Advance,
Gleidson Antonio Henriques
- extensions.conf
[trunk-group1]
exten = 0400,1,Answer
exten = 0400,2,Queue(queue-group1|t|||45)
exten = 0400,3,Hangup
[nulled]
exten = 123,1,Answer
exten = 123,2,AgentCallBackLogin(|,@nulled)
exten = 205,1,Answer
exten = 205,2,Wait(2)
exten = 205,3,Record(asterisk-recording%d:gsm)
exten = 205,4,Wait(2)
exten = 205,5,Playback(${RECORDED_FILE})
exten = 205,6,Wait(2)
exten = 205,7,Hangup
-
--- agents.conf
agent = 01,123,Name1
agent = 02,1234,Name2
-
--- queues.conf
persistentmembers = no
[queue-group1]
musiconhold = default
strategy = ringall
member = Agent/01
member = Agent/02
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] NAT and externip problem or bug
posting the relevant parts of your config (sip.conf, extensions.conf) could help to solve your problem. Guido I've recently got asterisk running on it's own pc inside my firwewall. Mostly it's working fine, but there is one silly problem I can't figure out. (For reference, Asterisk is the latest stable version as of last weekend 14th July. All connectivity is SIP or IAX). I initially had 'externip' set to my public IP. I have the appropriate 5000 range ports forwarded to the asterisk PC and external calls seem OK. The 'local' phones are a mixture of Sipura boxes and softphones. Problem: No or one-way audio in internal calls. Reason: Asterisk appears to be using the 'externip' address for all SIP devices, regardless of their NAT setting. Once a call starts, some softphones change the address they are responding to use the external IP rather than the asterisk PCs local IP on the same subnet... I have tried all NAT options and spent quite a while reading everything I can find about sip.conf, but I can't so far find any way of changing this behaviour. All the internal phones work fine if I comment out the externip line, but then the connections outside the firewall are likely to have problems. Is there any way of configuring externip on a per-device basis, or should it only have effect on NATed devices? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Redundant Ethernet
LOL found this: http://fcp.homelinux.org/modules/smartfaq/faq.php?faqid=549 hope it helps... Guido We're using OSPF... Is That? Oh Shit PBX Failed? SNIP ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] CDRTools please help
Von: ravi reddy [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 13. Juli 2006 10:03 An: asterisk-users@lists.digium.com Betreff: [asterisk-users] CDRTools please help ... but when i gave command #mysqladmin -uroot -px -hlocalhost ./setup_mysql.sh create cdrtool it is creating cdrtool database in mysql server but with no tables and nothing just creating databse and then i tried to run the commands like #mysqladmin -uroot -px -hlocalhost ./create_tables.mysql I don't know much about CDRTool, but I think you've a typo in your mysqladmin statement. Perhaps try this: mysqladmin -u root -px -h localhost ./setup_mysql.sh create cdrtool With your command you try to export something out of your database: mysqladmin -uroot -px -hlocalhost ./setup_mysql.sh create cdrtool Also, have a look at your setup_mysql.sh script, it is broken after the above command. This happens normaly when it's too late at night ;-) Hope it helps... Guido ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
On Fri, 26 May 2006, Guido Hecken wrote: We had the same problems with some cheap LevelOne Switches. The Snoms rebooted during a call, calls dropped etc. Replacing the switches was the solution. A switch should NEVER cause ANY device to lockup, ever. Period. If a phone locks up / reboots due to something a switch sends, then the phone is faulty. Okay, it shouldn't reboot if not told to do so but a switch with e.g. corrupt mac tables can bring your whole network down and the phone has no chance too. However, if the SNOMS still reboot with the follwing settings attached, I would also think of a possible bug in the firmware. Setup/Advanced Detect Ethernet Cable Unplug: off Action on Ethernet cable replug: ignore Guido ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom firmwares suck --additional datapoint to consider
I looked long and hard at the LAN and it was basically narrowed down to the switches. In this smaller install, several cheapo Dlink ($30) switches de-aggregate a Cisco Catalyst switch. What I noticed was that any phone plugged direcly into the Catalyst did *not* lock up or reboot. Any phone plugged into the crap switches experienced the lockup. So now we are down to the cheap switches themselves. We are nuking the Dlink switches and replacing them with 3com workgroup switches, same as what we use in the large install to good effect, and I fully expect the problem to dissapear. We had the same problems with some cheap LevelOne Switches. The Snoms rebooted during a call, calls dropped etc. Replacing the switches was the solution. Guido ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP TAPI
Clint, thanks for your comments and documentation on asttapi, great work! Some weeks ago after hours of reverseengineering we gave up on asttapi :( Provided with your informations, things seem to become clearer now and we'll try again. Guido ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Quad BRI card
On Thursday 18 May 2006 03:35, Mark Coccimiglio wrote: Otherwise the Diva server cards are a good option (extensive, but come highly recomended from most that I hear). Good luck and happy hunting. Ouch, you weren't joking. 1453 Euro! What about the Gerdes Primux Cards. They can be used in NT and TE Mode. Price ~ 670.- EURO We have a 2S0 card running on a customer site with chan_capi-cm and all looks good. Have a look at http://www.primuxisdn.de Perhaps it helps... Regards Guido ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Beeps and noises during calls
First, you could use a softphone like sjphone or X-Lite. If the problem is still there, pull any card from your server and try from internal (sip) extension to another internal extension. Also have a closer look on your nic. only some ideas to isolate the problem... hope it helps a little Regards Guido The beeps are not DTMF tones (at least they don't sound like it). It sounds more like the system is trying to compensate for something or adjusting something. There is a beep, sometimes several, or maybe one or 2 in a row, and it can be faint, or loud, or whatever, but is always the same pitch and tone. Sometimes it is accompanied with loud talkback to the earpiece. I'm going nuts, and cannot in good conscience, install or recommend this to anyone till I can resolve this. It has happened with 2 separate installs of *, with different hardware, different packages installed (one is * 1.2.4 with freepbx, the other was * 1.0 with nothing), and different digium hardware. The only thing that was the same is the Polycom phones, and SBC as a provider for the POTS lines... HELP ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Snom 360 problems
Anyone have a Snom they're happy with? How did you manage that? :) I have a system of: Asterisk 1.2.3 2 Wildcard TDM400P Rev I and E/F 1 Snom 360 + sidecar ~15 Sipura/Linsys SPA-841 ~15 Grandstream 101 Everything (currently) is on the same network, not a router to be seen between any two. Also everything, except the snom, is working sweetly. The main problem is ECHO.. awful echo and only on the Snom. When using a Zap line or to another sip phone. I've tweaked the * for echo and managed to only create echo and piss everyone else off, pounded the settings in the Snom trying to find anything, and updated the firmware to Application-Version:snom360-SIP 5.2 Rootfs-Version:snom360 jffs2 v3.36 after noticing a changelog that sounded like it may have related to echo. Not even a slight reduction in echo so far. A second serious problem is Call join. Even with Call join on Xfer (2 calls) OFF if the user is doing a transfer of one call when a second starts ringing the 2 callers get bridged, no transfer. Really nice, now I have two customers talking to each other with no clue what's going on and neither gets who they were trying to reach. Any ideas on what I can try next? This firmware works well for us: snom360-SIP 4.1 available here: http://snom.com/download/share/snom360-4.1-SIP-j.bin No echo and overall voice quality is excellent. Did you check the codecs on the snom and on asterisk (sip.conf)? Is Silence Suppression off on the snom? If you would post your config (under settings on the snom) we could have a closer look in the problem. Regards, Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for Windows based Asterisk
Well I disagree on the untraedit, since vi does a far better job. :) you're right, if you know the secrets of this operating system ;-) Regards and nice weekend to all out there Guido Hecken ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Looking for Windows based Asterisk
also add winscp and ultraedit to your windows system, it works great. http://winscp.net/eng/index.php http://www.ultraedit.com/ Regards Guido Hecken Without putty, my windows would be meaningless. PaulH Subtle Paul! but nice! :) Mike UK ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] is there any free pocket pc softphone??
I was able to register Portrait with our Asterisk box, but no audio, no signaling at all. Played a while with different codecs but no success. Did anybody make it really work with asterisk? Any hints, configs etc. Regards Guido Hecken I've also had some luck with Microsoft Portrait Guido Hecken wrote: is there any free pocket pc softphone Try sjphone from http://www.sjlabs.com/sjp.html Regards ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] is there any free pocket pc softphone??
is there any free pocket pc softphone Try sjphone from http://www.sjlabs.com/sjp.html Regards Guido Hecken ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Suggestions for tunning SJphone with Asteris k?
Did you try using Sjphone with allow=alaw in sip.conf instead of g729? In our Asterisk Installations we used: ... allow=alaw allow=ulaw allow=gsm ... and didn't have any echo or scratchy problems. Hope it helps... Regards Guido Hecken -Ursprüngliche Nachricht- Von: Chuck Bunn [mailto:[EMAIL PROTECTED] Gesendet: Donnerstag, 17. November 2005 15:15 An: Asterisk - Users Betreff: [Asterisk-Users] Suggestions for tunning SJphone with Asterisk? Hi, I am having voice quality problems with SJPhone under certain conditions. Setup is Fedora 4 with Asterisk 1.2rc2 and Digium TDM 400P (2FXO's and 2 FXS's). SJPhone = outside line = echo's, scratchy ZyXel P2000WV2 = outside line = clear as a bell SJPhone = recording voice mail = clear as a bell ZyXel P2000WV2 = recording voice mail = clear as a bell SJPhone = picking up voice mail = clear as a bell ZyXel P2000WV2 = picking up voice mail = clear as a bell SJPhone = to any voice mail (connected remotely via VPN to network containing Asterisk) = clear as a bell SJPhone = 3 way conference with Meet me = echo's, scratchy ZyXel P2000WV2 = 3 way conference with Meet me = unsure since conference was with 2 SJPhones SJphone = internal extension = echo's, scratchy ZyXel P2000WV2 = internal extension = unsure since talking with SJPhone ** Here is my 'sip.conf' general] context=default srvlookup=yes ;Zyxel - P2000WV2 [300] context=internal-home type=friend username=300 secret=xx callerid=300 nat=no host=dynamic mailbox=300 careinvite=no disallow=all allow=alaw dtmfmode=rfc2833 ;SJphone [301] context=internal-home type=friend username=301 secret=xx callerid=301 qualify=yes nat=no host=dynamic mailbox=301 ;careinvite=no allow=g729 dtmfmode=rfc2833 ;SJphone [302] context=internal-home type=friend username=302 secret=xx callerid=302 qualify=yes nat=no host=dynamic mailbox=302 ;careinvite=no allow=g729 dtmfmode=rfc2833 ;SJphone [303] context=internal-home type=friend username=303 secret=xx callerid=303 qualify=yes nat=no host=dynamic mailbox=303 ;careinvite=no allow=g729 dtmfmode=rfc2833 ;SJphone [304] context=internal-home type=friend username=304 secret=xx callerid=304 qualify=yes nat=no host=dynamic mailbox=304 ;careinvite=no allow=g729 dtmfmode=rfc2833 ;SJphone [305] context=internal-home type=friend username=305 secret=xx callerid=305 qualify=yes nat=no host=dynamic mailbox=305 ;careinvite=no allow=alaw allow=g729 dtmfmode=rfc2833 ;Zyxel - P2000WV2 [306] context=internal-home type=friend username=306 secret=xx callerid=306 nat=no host=dynamic mailbox=306 careinvite=no disallow=all allow=alaw dtmfmode=rfc2833 ;SJphone [307] context=internal-home type=friend username=307 secret=xx callerid=307 qualify=yes nat=yes host=dynamic mailbox=307 ;careinvite=no allow=g729 dtmfmode=rfc2833 ;Zyxel - P2000WV2 [500] context=internal-rest type=friend username=500 secret=xx callerid=500 nat=no host=dynamic mailbox=500 careinvite=no disallow=all allow=alaw dtmfmode=rfc2833 HERE IS MY 'zapata.conf' [trunkgroups] [channels] usecallerid=yes hidecallerid=no callwaiting=no threewaycalling=no transfer=yes echocancel=yes echotraining=yes context=incoming-home signalling=fxs_ks group=1 channel = 1,2 context=trunkdial signalling=fxo_ks group=2 channel = 3,4 Any suggestions for what to tune would be greatly appreciated. Thanks ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where is the text of the voicemail email ??
I was looking for the text in the /etc/asterisk directory, but it must be somewhere else. Can anybody tell me where? And can it include Chinese as well? Isn't it in /etc/asterisk/voicemail.conf ? In our installations we change the voicemail text in this file. Maybe you could include another file in this file, so different charsets could be possible. Hope it helps a bit... Regards Guido Hecken ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Where is the text of the voicemail email ??
I was looking for the text in the /etc/asterisk directory, but it must be somewhere else. Can anybody tell me where? And can it include Chinese as well? Check voicemail.conf in /etc/asterisk or voicemail.conf.sample in the /configs directory of your source code tree. I have never tried with Chinese, but it can handle Swedish :-) BTW wouldn't it be helpfull if the voicemailtext could depend on the language, the user has choosen in extensions.conf? Example: User has language set to de, include language file de in voicemail.conf . Regards Guido Hecken ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Patton SmartNode
Actually we 're running the sip protocol but in the past we did also use h323 in combination with tedas phoneware server (german voip solution). Both ran on SmartNode side very stable. Caller ID Name with sip/h323 should not be a problem, but here in Germany I'm not really shure, if the telco (T-COM) does support this feature on the PSTN side. I guess the SmartNodes should do the job anyway. Regards Guido Hecken Are you running SIP, or H323, or MGCP? Also, do you get callerid name passed through? Guido Hecken wrote: We use the SmartNodes SN1400 and SN2300 as ISDN Gateways in our customer Asterisk installations and are really happy with them. They run very stable and you can configure nearly everything. Support from INALP is also great. With the interface cards for the SmartNode 2300 you should be able to connect nearly everything to VOIP. Regards Guido Hecken Does anybody have any experience using a Patton SmartNode as a SIP/Telco gateway with Asterisk? They seem really inexpensive and appear to support all of the necessary features, but I don't have any experience with their products, so I don't know if they are any good. We are currently using a Cisco 2600 w/ PRI card and it works fine, but I was looking for someone else as a possible alternative. Thanks. Peder ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Patton SmartNode
We use the SmartNodes SN1400 and SN2300 as ISDN Gateways in our customer Asterisk installations and are really happy with them. They run very stable and you can configure nearly everything. Support from INALP is also great. With the interface cards for the SmartNode 2300 you should be able to connect nearly everything to VOIP. Regards Guido Hecken Does anybody have any experience using a Patton SmartNode as a SIP/Telco gateway with Asterisk? They seem really inexpensive and appear to support all of the necessary features, but I don't have any experience with their products, so I don't know if they are any good. We are currently using a Cisco 2600 w/ PRI card and it works fine, but I was looking for someone else as a possible alternative. Thanks. Peder ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] *8 and group pickup not working
Shouldn't it be pickupexten = *8 instead of pickupextn = *8 ? Regards Guido Hecken Hello I have a Junghanns ISDN BRI card for incoming calls and use SIP Polycom IP300 phones. My config files look like this: features.conf pickupextn = *8 zapata.conf context=frompstnisdn group=1 callgroup=1 pickupgroup=1 I also edited sip.conf like this: group=1 callgroup=1 pickupgroup=1 But on internal and incoming calls if I dial *8 from any phone I cannot pickup. Do I need to add anything to extensions.conf? do something else. I also tested with a Snom 190 and that cannot pickup using *8 either! Angus ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk won't listen on different port
AFAIK you have to add port=5062 in the context general. Stop and restart asterisk, and everything should be fine.. [general] port=5062 Regards Guido gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil+49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RE:Asterisk-Users] PhoneCALL v2.6.1 - Released
Really great job, it looks like exactly what we were searching for, since get started with asterisk. Keep on going with this excellent work. Regards Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Transcoding
AFAIK you need a license from Digium if you want to transcode to/from G729a... Hope this information is correct and it helps Regards Guido Hecken I though that Asterisk would transcode between codecs! All my SIP devices support G729a 711a/u but my ISDN PRI/BRI lines use 711a/u. Asterisk is quite happy to accept a call from a SIP device using G729a and then complains that it can't translate into G711 to go onto the ISDN network. Does anybody know if there is some setting somewhere or if this is how it is supposed to work ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange dropped calls
Hello List, has anybody seen (and perhaps decoded) this messages in Asterisk debug: ... Jun 30 17:46:23 DEBUG[11033]: Didn't get a frame from channel: SIP/192.168.1.30-ed86 Jun 30 17:46:23 DEBUG[11033]: Bridge stops bridging channels SIP/5602-c545 and SIP/192.168.1.30-ed86 ... This happens always, when a call get's dropped and today we had more than 10 calls which ended this way. The Problem was only with one phone, the others are ok. My first thought, some kind of network error?! Hope, someone can help Regards, Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk does not function without a DNS ser ver
We have our Asterisk server running smoothly with a SIP BRI gateway for inbound calls. However if the Internet connection goes down and a DNS server becomes unreachable Asterisk basically does not function. By this I mean it does not answer call coming in from the gateway (which is on the local LAN) and you can't even reload it - just hangs there. If I change the DNS setting in resolv.conf to something else which is reachable all is well again. I have tried setting srvlookup=no in sip.conf but it made no difference. Does anyone know how I to make Asterisk continue working for local LAN users/gateways when a DNS server is not reachable? Try to use bind on the * Machine and configure it as a caching only nameserver. Hope, this helps Regards, Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Voicemail and MS Exchange Synchronization
I would like to support these plans for exchange/outlook integration with at least $250 as well. Please have a closer look at http://www.click-and-call.com/ . Mediastreams has developed their product e-phone, which we could test a couple of months ago. Their Outlook Integration is really great: - see missed calls in inbox - right click a contact or missed call entry to dial - starting outlook, registers the extension in the system (on asterisk-server ?!) - incoming call pops up, transfer it with one click to voicemail or other extension - Managing Call Groups within outlook - Managing voicemail - Recording of calls ... But if you also have a closer look on their prices... ;-( If the community would be able to develop such a killer-app, Asterisk could really become the leading telephone application, perhaps world-wide! Developers like Thorben Jensen did a realy good job, to get things work on the client side. Perhaps, these guys with the power to code things well, should work - more - together on an Outlook Integration. My experiences with asterisk in short are, that the server-apps are running really stable, many features are developed, tested and made there way to the stable version. But what's really missing, are GUIs that normal users can work with. They have to accept them and should love to work with them. If we can't provide users with these GUIs, the powerfull features within Asterisk are only something for techies like us. Now, this is my 2cts to this discussion. Nice weekend to all and let's make Asterisk a more powerfull application Guido Hecken gwsNetTech Guido Hecken Quirrenbacher Str. 36 53639 Königswinter Germany fon +49(2244) 870663 fax +49(2244) 870664 mobil +49(179) 1267353 web http://www.gwsnettech.de mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] call queues problem
I think you have missed something with your agents.conf and with the member lines in queues.conf. This works for us: In queues.conf: [gws-wartefeld] music = default strategy = ringall context = queue-out timeout = 15 wrapuptime=10 announce-frequency = 0 announce-holdtime = no queue-youarenext = queue-youarenext ; (You are now first in line.) queue-thereare = queue-thereare; (There are) queue-callswaiting = queue-callswaiting ; (calls waiting.) queue-holdtime = queue-holdtime ; (The current est. holdtime is) queue-minutes = queue-minutes ; (minutes.) queue-seconds = queue-seconds ; (seconds.) queue-thankyou = queue-thankyou ; (Thank you for your patience.) queue-lessthan = queue-less-than; (less than) queue-reporthold = queue-reporthold ; (Hold time) joinempty = no member = Agent/6301 member = Agent/6302 member = Agent/6303 member = Agent/6304 member = Agent/6305 In agents.conf: [general] persistentagents=yes [agents] ackcall=no musiconhold = default updatecdr=yes agent = 6301,,Agent 1 agent = 6302,,Agent 2 agent = 6303,,Agent 3 agent = 6304,,Agent 4 agent = 6305,,Agent 5 Hope this helps Regards Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Little Php question
Instead of fputs($socket, Context: mainmenu\r\n\); use fputs($socket, Context: mainmenu\r\n); The trailing backslash was the bug Now every thing works fine... Regards Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Hotel billing in IPSwitchBoard
This seems to be exactly the application I was looking for :-). Since I'm working on a project where accounting and billing (http and voip traffic) is an issue, I'm glad to read that there will be a solution within a reasonable GUI. While dealing with squid and the great Squid2MYSQL script - used for Accounting and Billing - from Eugene V. Chernyshev [EMAIL PROTECTED], I wasn't shure about the udp traffic. A time based Billing seems to be a good solution. Keep on going with your great work on IPSwitchboard. Guido Hecken I am currently working on implementing Hotel Billing in IPSwitchBoard. The idea is that a receptionist in a hotel can just right click an extension button and choose Account; IPS will now calculate the call charges made from that extension and show all calls and charges on a form. The receptionist now has the option to close the account which will reset the account. I will add a table for editing call charges, and there will be a possibility to add a fee for connection charges and also an option to charge calls per xx seconds and to add/subtract a percentage to all calls. I will add a family/key to the asterisk database to indicate if the extension is closed, this way you can stop outgoing calls from being made from a closed extension by checking the dial plan. Please let me know if there are any other features you would like to see in IPSwitchBoard. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Restart after crash
Could you give some more information on where to remove 'daemon' and the effects? Since all our productionservers running FC2 I'm a bit concerned. There is a bug with safe_asterisk and FC2, you must edit the script to remove 'daemon' from the the startup command and then it will auto restart. Thanks a lot Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Restart after crash
In your asterisk in init.d that calls safe_asterisk change this: ... fi $DAEMON $ASTARGS RETVAL=$? [ $RETVAL -eq 0 ] touch /var/lock/subsys/asterisk echo return $RETVAL } ie remove 'daemon' from the command. After applying the change to the init script, it seems to restart the asterisk processes which get killed, but do you have a functional system with this? Our Testsystem spits out some '100% CPU-Loaded mpg123 processes' and asterisk was somehow dead. Did I miss something? Test it by kill -9 asterisk pid and see if it restarts - it is quite aggressive. Yeah, really aggressive ;-) Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Backup for linux/asterisk
What about imaging? We use acronis true image 8.0. You can create an image of your asterisk box within 20 minutes (120 GB HD !) and deploy it to another server in the same time. Even if changing your hardware from VIA to SIS and back to INTEL wasn't a problem for us. Btw we use Fedora Core 2 for our * servers. Regards, Guido Hecken after getting my feet wet with [EMAIL PROTECTED], I want to set up a second asterisk box to add a call shop billing and other add-ons such as LCR. My question is as follows. Is there a backup program that will save to a tape drive or a USB CD Writer so if I mess up an install I don't have to go through a complete reinstall? I saw a few programs out there but they required X windows and from what I read it is suggested that X windows not be installed on an Asterisk box. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Regex howto proof and change a dialed number
this would change +49(2244)870663 to 002244870663 in every line of the file, named number. But how can I achieve this in asterisk dialplan? ... It sounds like an AGI perl script would work well. Thanks for the tip with AGI I'll have a closer look at it. Neveretheless I put an easy number-manipulation tool on my asterisk wishlist. Wouldn't it be nice if something like this could do the job: exten = \+(XX)\((X.)\)(X.),1,Dial/SIP/[EMAIL PROTECTED],60,tTr) Regards Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Call Queues and Transfers
We had the same problems with transferring calls in queues. Sometimes, after pressing the # Key twice !!, we hear Allison say Transferring. Which Phones do you use? What shows up in the cli debug? Are you using t and T options in the dial command? Regards, Guido Hecken Guys.. Why is it that when a call comes to a call queue and in term gets assigned to an agent, if that agent tries to xfer the call using # or any other feature, it doesn't do anything? I just hear the pleeps on the phone but asterisk doesn't intervene with the Transfer prompt. Am I missing something? Thx! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Parked Call
Try using dtmfmode=rfc2833 in your sip.conf. It should work... Hope, this could help. Guido Hecken I have a question, I am unclear on how to park a call. I know that you are supposed to be able to press # and then transfer the call to extension 700. However, * doesn't seem to be graping the dtmf. I am using dtmfmode=inband. Asterisk is in the media path as well. Thanks in advance Justin ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] CVS-HEAD change: queue/agent persistence
For anyone using CVS HEAD, if you are using queue member persistence or agent persistence, your next update will cause the persistence to break. The storage format for these elements has been changed so that it can be more easily extended in the future, but this required breaking compatibility. This should be the last time these features will be broken by an upgrade :-) Thanks for these informations. But what does it mean exactly update will cause the persistence to break. Which actions are required to maintain this feature after updating? Regards, Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Getting phpconfig to work?
When I do click on the phpconfig.php link from http://ip-of-machine/phpconfig/, it returns a page with the actual contents of that file (phpconfig.php) and doesn't load the page. See some of the output below; Try a simple php-script in this directory. Something like this, name it test.php ?PHP print (test) ? If it does not print test, and you see the code instead, check your httpd.conf for a general php-parsing problem. Regards Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] RE: Getting phpconfig to work?
True, I have tried that and all I see is code instead. How do I go about solving the php-parsing problem in my httpd.conf then? Try this in your (actual!!) httpd.conf AddType application/x-httpd-php .php Perhaps (not shure) this too, but the above should work # LoadModule php4_module modules/libphp4.so # DirectoryIndex index.php Regards, Guido Hecken ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
