[asterisk-users] how to improve sound file quality?
We have recorded wav files with 44k, 22k, 16k, 11k and 8k Asterisk does not accept these wav files. I used sox input.wav output.gsm to get them to work. However, the only the 8k file did convert and the quality is poor. How can I improve the quality? bye Ronald ___ -- 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] Can asterisk work with a dynamic IP?
I know I can setup asterisk without Internet at all and it works as local pbx. Would an asterisk box work with a dynamic IP, with a dyndns name? What must I take care if I try that? bye Ronald ___ -- 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] Wellgate Asterisk
I got a Wellgate 3804A and need some hints:
Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate
Wellgate 3804A settings (Line1~Line4):
1. Sip Config
Mode: Proxy
Primary Proxy IP Address: *.131
Primary Proxy port: 5060
Line1 Number: 1002
2. Security Config
Line1 Account: 1002
Line1 Password: **
3. Line Configuration
Line1: Type=FXO, Hunting Group=2, Hot Line = 88621002
Asterisk settings:
users.conf:
[1002]
context = DID_1002
host = *.133
username = 1002
secret = **
trunkname = WellGate-1002 ; GUI metadata
hasiax = no
registeriax = no
hassip = yes
registersip = yes
trunkstyle = voip
hasexten = no
host = dynamic
disallow = all
allow = ulaw,alaw,gsm,g726,g729
extensions.conf
1002 = SIP/1002
...
[DID_1002]
exten = _88621002,1,NoOp(${CALLERID(num)})
exten = _88621002,n,Wait(1)
exten = _88621002,n,SayUnixTime
include = DID_1001_timeinterval_working day|${timeinterval_working day}
include = DID_1001_default
[DID_1001_default]
exten = s,1,NoOp,${CALLERID(num)}-${CALLERID(name)}
exten = s,n,Answer
exten = s,n,zapateller(nocallerid) ; torture telemarketers
exten = s,n,DigitTimeout,5 ; Set Digit Timeout to 5 seconds
exten = s,n,ResponseTimeout,10 ; Set Response Timeout to 10 seconds
exten = s,n,Hangup
include = default
[DID_1001_timeinterval_working day]
exten = _6888,1,Goto(default|6888|1)
If I call in at line2, then I can hear the Time announcement and I can
dial during that announcement an extension number.
BTW, where can I find the additional sounds I had at an previous setup
(If you know the extension, ...), which should replace the SayUnixTime
I have no idea how to get dial out to work. Can anybody give me a hint,
please?
In Asterisk I see:
[Nov 27 20:58:00] NOTICE[5095]: chan_sip.c:9227 sip_reg_timeout:--
Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #102)
-- Got SIP response 486 Busy Here back from *.133
*CLI sip show peers
1002/1002 *.133D 5060 Unmonitored
*CLI sip show users
1002 **
DID_1002 No RFC3581
*CLI sip show registry
*.133:5060 1002 120 Request Sent
bye
Ronald
___
-- 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] [Solved] Wellgate Asterisk
Guillermo Salas M. wrote: El jue, 27-11-2008 a las 21:05 +0800, Ronald Wiplinger (Lists) escribió: I got a Wellgate 3804A and need some hints: Both have public IP *.131=asterisk (1.6.0.1) *.133= Wellgate Wellgate 3804A settings (Line1~Line4): I've one wellgate 3804 (old version) with 4 fxo ports integrated with asterisk 1.4. Regards, I could solve it! I had to add routing in the 3804A. Now both, dialin and dialout is working. bye Ronald ___ -- 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] [SOLVED] Re: Upgrade 1.4.19 to 1.6 = segementation fault
Ronald Wiplinger (Lists) wrote: During compiling I have not seen an error, however, when I start asterisk again it ends with: app_morsecode.so = (Morse code) == Registered custom function 'SYSINFO' func_sysinfo.so = (System information related functions) Segmentation fault (core dumped) How can I figure out what is wrong? I removed all modules, which were left from the 1.4 installation and now it works! ___ -- 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] Upgrade 1.4.19 to 1.6 = segementation fault
During compiling I have not seen an error, however, when I start asterisk again it ends with: app_morsecode.so = (Morse code) == Registered custom function 'SYSINFO' func_sysinfo.so = (System information related functions) Segmentation fault (core dumped) How can I figure out what is wrong? bye Ronald ___ -- 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] Snom - we are puzzled
we have installed asterisk and snom with PUBLIC IPs (IP/25) on one DSL line we have for our office a different ADSL with one IP shared. Two identical setup snom 360 (except the user name) with two public IP addresses are connected at the hub to the server / DSL line phone A can call B, B cannot call A, because A is not registered!!! We disconnect A and setup a softphone (on the ADSL line with stun) and it works. How can I track down this problem. bye R. ___ -- 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] Maybe a crazy idea, but are there Asterisk hoster outside there?
I used to run an Asterisk server in the office, ... was looking for a small replacement. I am not sure if that one is a good idea yet either. How about this one: I have VoIP phones, I have a Welgate 3804 (=2 FXO), all what I need is an Asterisk server. Is there a Asterisk hoster out there? Maybe as a virtual machine? The mini solution does not have all features, but maybe this would still allow me to turn off another machine here. bye Ronald ___ -- 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
[asterisk-users] I used to use an Asterisk server, but now it is overkill, ...
I had installed in the office an Asterisk server, but the company is gone and I could keep the server. However, for my family with three members and two phone lines this server is overkill. I am looking for a compact solution, which is more suitable for me. I want a small silent box, which can connect two phone lines and 6 internal VoIP phones and about 6 external VoIP phones. I would like to have: 1. Announcements for callers (dial the extension number) 2. voice mail with mail forwarding 3. wakeup call 4. pickup group 5. call forwarding after 20 seconds, ... 6. ISN support, Sipbroker support 7. remote gateway support I guess that is all what I would need at home. What is your suggestion for that? bye Ronald ___ -- 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
[asterisk-users] remote server with Snom 190
I have a local asterisk 1.2 and a remote asterisk 1.4. Snom 190 can be used with the local asterisk but not with the remote one. I need some hints where to track down this issue. Some information: Snom 190: Line 1: Account: 615 Password: OnlyIknowit Registrar: ast.mydomain.com Status: OK Line 2: Account: 6888 Password: Otherside Registrar: 22.33.44.55 (only IP address!) Status: Not found Function keys: P1 Line Number sip:[EMAIL PROTECTED];user=phone P2 Line Number sip:[EMAIL PROTECTED];user=phone Remote server is a fresh installed Ubuntu 8.04 server. What do I miss? bye Ronald ___ -- 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] rxfax does not work (anymore)
Below is my extensions.conf for the fax part
[incoming_28345474]
;
;
; BEGIN - Inbound call handlers
;
;
exten = 8862100,1,NoOp(${CALLERID(num)})
exten = 8862100,2,Background(if-u-know-ext-dial)
exten = 8862100,3,Set(CALLERID(num)=${CALLERID(num)})
exten = h,1,hangup()
include = fax2emailstart
include = local
[fax2emailstart]
exten = 3000,1,SetVar(CALLEDFAX=${EXTEN}) ; [EMAIL PROTECTED]
exten = 3000,2,Answer
exten = 3000,3,Macro(fax2emailservice)
exten = h,1,System(/var/lib/asterisk/scripts/fax2emailservice
${CALLERIDNUM} ${CALLEDFAX} ${EXTNAME} ${EXTEMAIL} ${FAXFILE}
${EXTCOMPANY})
[macro-fax2emailservice]
exten =
s,1,SetVar(FAXFILE=/var/spool/asterisk-fax/${CALLEDFAX}/${UNIQUEID})
exten = s,2,Set(EXTEMAIL=${DB(${MACRO_EXTEN}/xEmail)})
exten = s,3,NoOP()
exten = s,4,Set(EXTNAME=${DB(${MACRO_EXTEN}/xName)})
exten = s,5,NoOP()
exten = s,6,Set(EXTCOMPANY=${DB(${MACRO_EXTEN}/xCompany)})
exten = s,7,rxfax(${FAXFILE}.tif)
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(7)
exten = s,105,SetVar(EXTNAME=Ronald)
exten = s,106,Goto(7)
exten = s,107,SetVar(EXTCOMPANY=Elmit)
exten = s,108,Goto(7)
When I call this PSTN number and dial the extension number 3000, then I
see that:
*CLI
[Jan 27 16:03:21] -- Zap/3-1 answered SIP/601-006a2970
[Jan 27 16:03:24] -- Executing NoOp(SIP/88621001-00728610,
88621001) in new stack
[Jan 27 16:03:24] -- Executing BackGround(SIP/88621001-00728610,
if-u-know-ext-dial) in new stack
[Jan 27 16:03:24] -- Playing 'if-u-know-ext-dial' (language 'en')
[Jan 27 16:03:28] -- Executing Set(SIP/88621001-00728610,
CALLERID(num)=88621001) in new stack
[Jan 27 16:03:32] == CDR updated on SIP/88621001-00728610
[Jan 27 16:03:32] -- Executing SetVar(SIP/88621001-00728610,
CALLEDFAX=3000) in new stack
[Jan 27 16:03:32] -- Executing Answer(SIP/88621001-00728610, )
in new stack
[Jan 27 16:03:32] -- Executing Macro(SIP/88621001-00728610,
fax2emailservice) in new stack
[Jan 27 16:03:32] -- Executing SetVar(SIP/88621001-00728610,
FAXFILE=/var/spool/asterisk-fax/3000/1201421004.8) in new stack
[Jan 27 16:03:32] -- Executing Set(SIP/88621001-00728610,
[EMAIL PROTECTED]) in new stack
[Jan 27 16:03:32] -- Executing NoOp(SIP/88621001-00728610, ) in
new stack
[Jan 27 16:03:32] -- Executing Set(SIP/88621001-00728610,
EXTNAME=Ronald Wiplinger) in new stack
[Jan 27 16:03:32] -- Executing NoOp(SIP/88621001-00728610, ) in
new stack
[Jan 27 16:03:32] -- Executing Set(SIP/88621001-00728610,
EXTCOMPANY=Elmit.com) in new stack
[Jan 27 16:03:32] -- Executing RxFAX(SIP/88621001-00728610,
/var/spool/asterisk-fax/3000/1201421004.8.tif) in new stack
vpbx*CLI
Disconnected from Asterisk server
I have no idea why it disconnects and hope somebody can help me to get
to work.
bye
Ronald
___
-- 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] Upgrade fails, need system upgrade advice
I have a AMD64 CPU and use SuSE 9.2 with kernel 2.6.8-18 I tried to upgrade svn version 1.4.x but it fails at each part and mainly because the system is with 1100 days getting to old. I have to make a decision and need your advice. CPU AMD64 3200+ 1 GB RAM Digium card with 2 FXS and 2 FXO external Wellgate box 3804 I want to keep my current settings (backup /etc/asterisk and /var/lib/asterisk and /var/spool/asterisk) I use festiva I need multiple fax on different extensions I would like to run also OpenSer on the same machine I would like to re-install a new system with svn asterisk 1.4.x and the above settings. Would you suggest me to install a. OpenSuse 10.x b. Ubuntu desktop c. Ubuntu server Any other hints? to backup directories? or just use a new hard disk. With LVM? bye Ronald ___ -- 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] dial extension number
Ronald Wiplinger wrote:
Can anybody give me a hint, please.
I have a Welltech FXO device and from PSTN coming calls will be
transfered to the extension number 1001.
I want that the caller can reach the extension number by dialing
said number.
My 1st try was:
exten = 1001,1,NoOp(${CALLERID(num)})
exten = 1001,2,Wait(1)
exten = 1001,3,Set(CALLERID(num)=${CALLERID(num)})
;
include = local; all extensions inhouse (including )
Above any dialed number will be ignored.
Replaceing the second line (Wait) with:
exten = 8862100,2,Background(if-u-know-ext-dial)
the extension will be dialed.
I do not want to have an announcement to ask for the dialing the
extension number. What can I use instead?
I tried now WaitExten(10), but that is not recognizing dialing as well.
Thanks!
bye
Ronald
___
-- 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] Help needed for Fax2Email with Welltech FXO 3804
I have this in my extension.conf:
[incoming_28345474]
; 8862100 is the hotline number of the Welltech 3804
;
exten = 8862100,1,NoOp(${CALLERID(num)})
exten = 8862100,2,Wait(1)
exten = 8862100,3,Set(CALLERID(num)=${CALLERID(num)})
include = fax2emailstart
[fax2emailstart]
exten = 3000,1,SetVar(CALLEDFAX=${EXTEN}); me
exten = 3000,2,Answer
exten = 3000,3,Macro(fax2emailservice)
exten = 3001,1,SetVar(CALLEDFAX=${EXTEN}); dave
exten = 3001,2,Answer
exten = 3001,3,Macro(fax2emailservice)
exten = h,1,System(/var/lib/asterisk/scripts/fax2emailservice
${CALLERIDNUM} ${CALLEDFAX} ${EXTNAME} ${EXTEMAIL} ${FAXFILE}
${EXTCOMPANY})
[macro-fax2emailservice]
exten =
s,1,SetVar(FAXFILE=/var/spool/asterisk/fax/${CALLEDFAX}/${UNIQUEID})
; exten = s,2,DBGet(EXTEMAIL=${MACRO_EXTEN}/xEmail)
exten = s,2,Set(EXTEMAIL=${DB(MACRO_EXTEN/xEmail)})
exten = s,3,NoOP()
exten = s,4,Set(EXTNAME=${DB(MACRO_EXTEN/xName)})
exten = s,5,NoOP()
exten = s,6,Set(EXTCOMPANY=${DB(MACRO_EXTEN/xCompany)})
exten = s,7,rxfax(${FAXFILE}.tif)
exten = s,103,SetVar([EMAIL PROTECTED])
exten = s,104,Goto(7)
exten = s,105,SetVar(EXTNAME=Ronald)
exten = s,106,Goto(7)
exten = s,107,SetVar(EXTCOMPANY=Boss)
exten = s,108,Goto(7)
CLI shows:
[Jan 14 22:58:51] -- Zap/3-1 answered SIP/601-006c3610
[Jan 14 22:58:54] -- Executing NoOp(SIP/88621001-007263d0,
88621001) in new stack
[Jan 14 22:58:54] -- Executing Wait(SIP/88621001-007263d0, 1) in
new stack
[Jan 14 22:58:55] -- Executing Set(SIP/88621001-007263d0,
CALLERID(num)=88621001) in new stack
[Jan 14 22:59:05] WARNING[20366]: pbx.c:2415 __ast_pbx_run: Timeout, but
no rule 't' in context 'incoming_28345474'
[Jan 14 22:59:05] -- Executing System(SIP/88621001-007263d0,
/var/lib/asterisk/scripts/fax2emailservice 88621001 )
in new stack
[Jan 14 22:59:09] -- Executing NoOp(SIP/88621001-006f8ea0,
88621001) in new stack
[Jan 14 22:59:09] -- Executing Wait(SIP/88621001-006f8ea0, 1) in
new stack
[Jan 14 22:59:10] -- Executing Set(SIP/88621001-006f8ea0,
CALLERID(num)=88621001) in new stack
[Jan 14 22:59:20] WARNING[20389]: pbx.c:2415 __ast_pbx_run: Timeout, but
no rule 't' in context 'incoming_28345474'
[Jan 14 22:59:20] -- Executing System(SIP/88621001-006f8ea0,
/var/lib/asterisk/scripts/fax2emailservice 88621001 )
in new stack
[Jan 14 22:59:20] -- Got SIP response 486 Busy Here back from
192.168.250.244
[Jan 14 22:59:21] -- Got SIP response 486 Busy Here back from
192.168.250.244
[Jan 14 22:59:21] -- Got SIP response 486 Busy Here back from
192.168.250.244
[Jan 14 22:59:22] -- Got SIP response 486 Busy Here back from
192.168.250.244
[Jan 14 22:59:23] -- Hungup 'Zap/3-1'
[Jan 14 22:59:24] -- Executing NoOp(SIP/88621001-006f3160,
88621001) in new stack
[Jan 14 22:59:24] -- Executing Wait(SIP/88621001-006f3160, 1) in
new stack
[Jan 14 22:59:25] -- Executing Set(SIP/88621001-006f3160,
CALLERID(num)=88621001) in new stack
[Jan 14 22:59:30] -- Executing System(SIP/88621001-006f3160,
/var/lib/asterisk/scripts/fax2emailservice 88621001 )
in new stack
I dial the number 28345474 and as soon the dialtone is to hear I dial
3000, but that is not shown in CLI. What am I missing?
bye
Ronald
___
-- 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] Multiple fax extensions
I need to setup multiple fax extension numbers. What is the best way to do that? It should send the fax as pdf to the assigned email address (or addresses) of that extension number. It should also move the fax to a web site for online view. It should - if possible - try to make OCR text file as email body. Thanks for your hints. bye Ronald ___ -- 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] I want to record each phone call
1. Instead of using *1 (automon) I need to record each phone call at a certain * box. 2. While already talking about this. I want to autodelete with cron at 2 am in the morning all recordings which are older than 50 hours! How can I do that? bye Ronald ___ --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] SVN update
I haven't updated for a while and when I looked on the web site how to do a SVN update, I cannot find it anymore. CLI show version Asterisk SVN-branch-1.2-r42600M built by root @ asterisk on a x86_64 running Linux on 2006-09-10 22:52:42 UTC 1. Where is the description for the SVN update now? 2. Is there anything I have to take care of when updating from such an old version? Thanks! bye Ronald ___ --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] Bandwidth shapping device
I have a link to a building (e.g. 10Mb/s) and want to split up the bandwidth to different users. Each user should get e.g., 512kB/s plus 256kB/s dedicated for VoIP. What kind of device can I use for that ? (managing switch ??? which one?) bye Ronald Wiplinger ___ --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] MRTG with 4 graphs
How can I set-up a MRTG with 4 graphs, whereby: 1 data in 2 data out 3 ONLY voice(/video) data in 4 ONLY voice(/video) data out bye Ronald Wiplinger ___ --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] moving WiFi phone
Can anybody tell me how I can set-up multiple access points with overlapping coverage, so that a moving WiFi phone user can continuesly use the phone. bye Ronald Wiplinger ___ --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] SMS via VoIP and web
Where can I get a starting point for setting up sms via VoIP and via web. I want to send SMS from VoIP or web to VoIP phones and GSM phones. 1. how to set-up? 2. which smsc should I use? (what is the price?) 3. which phones can be used? bye Ronald ___ --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] Soundfiles adding during phone calls
bails wrote: Ronald Wiplinger wrote: Ronald Wiplinger wrote: Tom Lynn wrote: Ron, The guy is trying to help you. Go to the link and read it. There is a feature that you can use to play a recording into the voice channel. Mine is set so when you press #9, the caller hears the lots of monkeys recording. The best part of it is that you can hang up and the recording will continue to play to the caller. When it expires, so does the call I tried this: features.conf [featuremap] blindxfer = ##; Blind transfer was #1 - now press # twice disconnect = *0; Disconnect automon = *1; One Touch Record atxfer = *2; Attended transfer [applicationmap] tortore= *9,callee,Playback,tt-monkeys Yap, that magic word helped! I got still some problems with it. I understand that I do not hear the sound, but wonder if I should get the call back after the playback or not anymore. In my experience the caller hang up and my phone remains on the status connected I have only the choice to power cycle the phone. Anything I can do ? bye Ronald Wiplinger ___ --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] Soundfiles adding during phone calls
Ronald Wiplinger wrote:
Tom Lynn wrote:
Ron,
The guy is trying to help you. Go to the link and read it. There is
a feature that you can use to play a recording into the voice
channel. Mine is set so when you press #9, the caller hears the
lots of monkeys recording.
The best part of it is that you can hang up and the recording will
continue to play to the caller. When it expires, so does the call
I tried this:
features.conf
[featuremap]
blindxfer = ##; Blind transfer was #1 - now press # twice
disconnect = *0; Disconnect
automon = *1; One Touch Record
atxfer = *2; Attended transfer
tortore= *9,callee,Playback,tt-monkeys
extensions.conf
exten = 601,1,Set(DYNAMIC_FEATURES=hangup#play#tortore#automon) ;
enable One-touch
exten = 601,2,Dial(${PHONE_601},30,tTwWr)
I make a call from 615 to 601
601 hits *9 but nothing happens!
when 601 hits *1 it records the conversion.
vpbx*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# ##
Attended Transfer *2
One Touch Monitor *1
Disconnect Call * *0
Dynamic Feature Default Current
--- --- ---
(none)
Call parking
Parking extension : 750
Parking context : parkedcalls
Parked call extensions: 751-770
I added already in extenions.conf:
include = featuremap
bye
Ronald Wiplinger
What do I miss?
bye
Ronald Wiplinger
On 11/11/06, * Ronald Wiplinger* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Andrew Joakimsen wrote:
http://www.voip-info.org/wiki-Asterisk+config+features.conf
... and where exactly did you see this feature
bye
Ronald Wiplinger
On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
I want to add some sound filed on demand during a phone call
only
possible on some extension numbers.
I get many phone calls from local companies, but don't
understand
Chinese! I would like to record the call, but also ask the
caller some
questions, which should be added into the call with some
keys on the
phone, ... e.g. *66554 should add into the call: How are
you? or What
is your phone number?
But I do have another application for that too.
I get many fake phone calls, where Chinese people tell you
that your
phone bill is not paid, your court fee is not paid, and
ask the
caller to go to the ATM machine and key in a series of key
strokes,
most likely it will clear out your account.
For such fake callers I would like to add a terrible noise
to the
call
and make scare them as much as possible.
Such fake calls I get now for each of my phone lines at
least 10
each!!!
Either the caller-id is not set, is 0 or is a tollfree number.
bye
Ronald Wiplinger
___
--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] Soundfiles adding during phone calls
Tom Lynn wrote:
Ron,
The guy is trying to help you. Go to the link and read it. There is
a feature that you can use to play a recording into the voice
channel. Mine is set so when you press #9, the caller hears the lots
of monkeys recording.
The best part of it is that you can hang up and the recording will
continue to play to the caller. When it expires, so does the call
I tried this:
features.conf
[featuremap]
blindxfer = ##; Blind transfer was #1 - now press # twice
disconnect = *0; Disconnect
automon = *1; One Touch Record
atxfer = *2; Attended transfer
tortore= *9,callee,Playback,tt-monkeys
extensions.conf
exten = 601,1,Set(DYNAMIC_FEATURES=hangup#play#tortore#automon) ;
enable One-touch
exten = 601,2,Dial(${PHONE_601},30,tTwWr)
I make a call from 615 to 601
601 hits *9 but nothing happens!
when 601 hits *1 it records the conversion.
What do I miss?
bye
Ronald Wiplinger
On 11/11/06, * Ronald Wiplinger* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
Andrew Joakimsen wrote:
http://www.voip-info.org/wiki-Asterisk+config+features.conf
... and where exactly did you see this feature
bye
Ronald Wiplinger
On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
I want to add some sound filed on demand during a phone call
only
possible on some extension numbers.
I get many phone calls from local companies, but don't
understand
Chinese! I would like to record the call, but also ask the
caller some
questions, which should be added into the call with some
keys on the
phone, ... e.g. *66554 should add into the call: How are
you? or What
is your phone number?
But I do have another application for that too.
I get many fake phone calls, where Chinese people tell you
that your
phone bill is not paid, your court fee is not paid, and
ask the
caller to go to the ATM machine and key in a series of key
strokes,
most likely it will clear out your account.
For such fake callers I would like to add a terrible noise
to the
call
and make scare them as much as possible.
Such fake calls I get now for each of my phone lines at least 10
each!!!
Either the caller-id is not set, is 0 or is a tollfree number.
bye
Ronald Wiplinger
___
--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] Soundfiles adding during phone calls
Tom Lynn wrote: Ron, The guy is trying to help you. Tom, I believe it! Go to the link and read it. There is a feature that you can use to play a recording into the voice channel. Mine is set so when you press #9, the caller hears the lots of monkeys recording. I am not sure if that is correct: feature.conf: [applicationmap] shout2caller = *911,callee,Playback,shout-100dB ;Shout to caller if *911 was pressed - use 'callee' or 'caller' ask4name-Chinese = *910,callee,Playback,ask4name-Chinese; Ask caller for her/his name in Chinese and in extensions.conf and where should Set(DYNAMIC_FEATURES=hangup#play#testfeature) be and I want that only 601 and 621 can use this feature. bye Ronald Wiplinger The best part of it is that you can hang up and the recording will continue to play to the caller. When it expires, so does the call On 11/11/06, * Ronald Wiplinger* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Andrew Joakimsen wrote: http://www.voip-info.org/wiki-Asterisk+config+features.conf ... and where exactly did you see this feature bye Ronald Wiplinger On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I want to add some sound filed on demand during a phone call only possible on some extension numbers. I get many phone calls from local companies, but don't understand Chinese! I would like to record the call, but also ask the caller some questions, which should be added into the call with some keys on the phone, ... e.g. *66554 should add into the call: How are you? or What is your phone number? But I do have another application for that too. I get many fake phone calls, where Chinese people tell you that your phone bill is not paid, your court fee is not paid, and ask the caller to go to the ATM machine and key in a series of key strokes, most likely it will clear out your account. For such fake callers I would like to add a terrible noise to the call and make scare them as much as possible. Such fake calls I get now for each of my phone lines at least 10 each!!! Either the caller-id is not set, is 0 or is a tollfree number. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0647-0, 2006/11/09 Tested on: 2006/11/11 �U�� 11:07:21 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus
Re: [asterisk-users] Soundfiles adding during phone calls
Andrew Joakimsen wrote: http://www.voip-info.org/wiki-Asterisk+config+features.conf ... and where exactly did you see this feature bye Ronald Wiplinger On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I want to add some sound filed on demand during a phone call only possible on some extension numbers. I get many phone calls from local companies, but don't understand Chinese! I would like to record the call, but also ask the caller some questions, which should be added into the call with some keys on the phone, ... e.g. *66554 should add into the call: How are you? or What is your phone number? But I do have another application for that too. I get many fake phone calls, where Chinese people tell you that your phone bill is not paid, your court fee is not paid, and ask the caller to go to the ATM machine and key in a series of key strokes, most likely it will clear out your account. For such fake callers I would like to add a terrible noise to the call and make scare them as much as possible. Such fake calls I get now for each of my phone lines at least 10 each!!! Either the caller-id is not set, is 0 or is a tollfree number. bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0647-0, 2006/11/09 Tested on: 2006/11/11 �U�� 11:07:21 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Soundfiles adding during phone calls
I want to add some sound filed on demand during a phone call only possible on some extension numbers. I get many phone calls from local companies, but don't understand Chinese! I would like to record the call, but also ask the caller some questions, which should be added into the call with some keys on the phone, ... e.g. *66554 should add into the call: How are you? or What is your phone number? But I do have another application for that too. I get many fake phone calls, where Chinese people tell you that your phone bill is not paid, your court fee is not paid, and ask the caller to go to the ATM machine and key in a series of key strokes, most likely it will clear out your account. For such fake callers I would like to add a terrible noise to the call and make scare them as much as possible. Such fake calls I get now for each of my phone lines at least 10 each!!! Either the caller-id is not set, is 0 or is a tollfree number. bye Ronald Wiplinger ___ --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: Real-time and priority n
Brian Capouch wrote: Tony Mountifield wrote: In article [EMAIL PROTECTED], Ronald Wiplinger [EMAIL PROTECTED] wrote: Is it exclusive? Either Realtime or priority n ??? If so, what is the better way? I believe 'n' is just a shorthand way of writing previous line + 1, and gets converted into an actual number as the dialplan is compiled. After compilation, the information about whether a line had been given as 'n' or as a specific number has been lost, as far as I know. Rows can be added to a database table at any time. Imagine a series of priorities added to a table using nothing more than n as a priority number beyond the first one. Now imagine wanting to add a new priority in between any two arbitrary entries in the table. How would you even specify which two lines should surround it, when they have no identifying serial number associated with them? Unless you were to add a new field, e.g. priority location identifier, or somesuch. Which does nothing more than move back to the present situation. The extensions.conf parser adds a real priority to each line, but in Realtime that responsibility falls on the DB maintainer. B. Short: EXCLUSIVE thanks! bye Ronald -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Real-time and priority n
Is it exclusive? Either Realtime or priority n ??? If so, what is the better way? bye Ronald ___ --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] Context default incoming ENUM
I want to make the context [default] as an alarm, for not having set-up correct. I am looking for a way to get incoming calls via ENUM or via names (e.g. sip:[EMAIL PROTECTED]) into a defined context. How can I do that? bye Ronald ___ --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] Priority n
How do I use priority n correct?
Here is the current example:
exten = 615,1,Dial(${PHONE_615},60,tr)
exten = 615,2,Voicemail,[EMAIL PROTECTED]
exten = 615,103,Voicemail,[EMAIL PROTECTED]
and:
exten = 617,109,GotoIf($[${DIALSTATUS} :
(CHANUNAVAIL|CONGESTION)]?110:999)
exten = 617,110, .
exten = 617,999,hangup
That would greatly help me to throw out the NoOp statements I have
inserted over the time if I tested some parts, ..
bye
Ronald
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING: chan_sip.c add_realm_authentication: ???
When I reloaded my asterisk I saw these lines, which I have noticed before: [Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 797 [Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 822 [Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 847 [Sep 27 11:46:09] WARNING[27468]: chan_sip.c:12039 add_realm_authentication: Format for authentication entry is user[:[EMAIL PROTECTED] at line 872 [Sep 27 11:46:11] == Parsing '/etc/asterisk/sip_notify.conf': [Sep 27 11:46:11] Found What does it mean? Should I care? bye Ronald ___ --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] Accounting and re-invite
I am thinking if re-invite will interfere accounting. Please help me to figure it out: Phone A is registered at asterisk and calls a gateway. If the gateway allows re-invite than the rtp would go directly from phone A to the gateway, while the sip messages are still going through Asterisk. Asterisk will be informed when the call ended. If it is a postpaid accounting, just bill the customer, however, how is it for a pre-paid (calling card user)? I think Asterisk will have no power to turn off the call from A to the gateway. Even more, if the gateway would allow to end a call and continue with a new call, the new call would not be billed (or would it)? I guess the solution must be re-invite=no However, re-invite=no means that each call is going with rtp also through my server, what means for a remote phone, I have to provide for both legs the bandwidth. Would here a rtpproxy or mediaproxy help? If how and why? bye Ronald ___ --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] pickupgroup 1
I have problems with pickupgroup. While 621 can pickup a call to 601 with *8, no phone can pickup a call to 621. Below are the settings for two phones. 601 is static in the sip.conf, while 621 is in the Real-time database. What could be the problem? I have an extension 601: [601] type=friend context=ELMIT username=hotline secret=shhshh canreinvite=no host=dynamic ;defaultip=61.220.121.19 dtmfmode=rfc2833 [EMAIL PROTECTED] nat=yes callgroup=1 pickupgroup=1 callerid=Ronald Hotline,601 qualify=1000 and and extension 621: CREATE TABLE `sip_buddies` ( `id` int(11) NOT NULL auto_increment, `name` varchar(80) NOT NULL default '', `accountcode` varchar(20) default NULL, `amaflags` varchar(13) default NULL, `callgroup` varchar(30) default NULL, `callerid` varchar(80) default NULL, `restrictcid` char(3) default 'NO', `canreinvite` char(3) default 'yes', `context` varchar(80) default NULL, `defaultip` varchar(15) default NULL, `dtmfmode` varchar(7) default NULL, `fromuser` varchar(80) default NULL, `fromdomain` varchar(80) default NULL, `host` varchar(31) NOT NULL default '', `incominglimit` int(2) default NULL, `outgoinglimit` int(2) default NULL, `insecure` varchar(4) default NULL, `language` char(2) default NULL, `mailbox` varchar(50) default NULL, `md5secret` varchar(80) default NULL, `nat` varchar(5) NOT NULL default 'yes', `permit` varchar(95) default NULL, `deny` varchar(95) default NULL, `mask` varchar(95) default NULL, `pickupgroup` varchar(10) default NULL, `port` varchar(5) NOT NULL default '', `qualify` varchar(4) default NULL, `rtptimeout` char(3) default NULL, `rtpholdtimeout` char(3) default NULL, `secret` varchar(80) default NULL, `type` varchar(6) NOT NULL default 'friend', `username` varchar(80) NOT NULL default '', `disallow` varchar(100) default 'all', `allow` varchar(100) default 'g729;ilbc;gsm;ulaw;alaw', `musiconhold` varchar(100) default NULL, `regseconds` int(11) NOT NULL default '0', `ipaddr` varchar(15) NOT NULL default '', `regexten` varchar(80) NOT NULL default '', `cancallforward` char(3) default 'yes', `fullcontact` varchar(80) default NULL, `setvar` varchar(100) NOT NULL default '', PRIMARY KEY (`id`), UNIQUE KEY `name` (`name`), KEY `name_2` (`name`) ) TYPE=MyISAM ROW_FORMAT=DYNAMIC AUTO_INCREMENT=102 ; -- -- Dumping data for table `sip_buddies` -- INSERT INTO `sip_buddies` VALUES (1, '621', NULL, NULL, NULL, 'Ronald private,621', '', 'no', 'ELMIT', NULL, 'rfc2833', NULL, NULL, 'dynamic', NULL, NULL, NULL, 'en', '[EMAIL PROTECTED]', NULL, 'yes', NULL, NULL, NULL, '1', '5060', '1000', NULL, NULL, 'shhshh', 'friend', '621', 'all', 'ulaw;alaw;g729;gsm', NULL, 1158361126, '192.168.250.76', '', 'yes', 'sip:[EMAIL PROTECTED]:5060', ''); ___ --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] ASTCC: change from no pin to pin request?
I want to change that ASTCC will ask for pin. 1. Where to set it? Pin length and number? 2. Can I set the pin only for a few people? E.g. Would deleting the pin number not ask for the pin or needs than still the # 3. How to change the pin? Can the user change the pin? bye Ronald ___ --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] Makefile.moddir_rules: No such file or directory
I need h.264 and tried therefore svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk (currently I have branches 1.2 installed) make clean; make update; make install . make[1]: Entering directory `/usr/local/src/svn-versions/asterisk' rm -f .depend rm -f .depend rm -f .depend Makefile:60: /usr/local/src/svn-versions/asterisk/Makefile.moddir_rules: No such file or directory make[2]: *** No rule to make target `/usr/local/src/svn-versions/asterisk/Makefile.moddir_rules'. Stop. make[1]: *** [channels-clean-depend] Error 2 make[1]: Leaving directory `/usr/local/src/svn-versions/asterisk' make: *** [update] Error 2 Why is Makefile.moddir_rules missing, or what have I forgotten to do? bye Ronald ___ --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] ast_parse_allow_disallow: Cannot allow unknown format 'h264'
I see in CLI: ast_parse_allow_disallow: Cannot allow unknown format 'h264' What can I do ? I see on Asterisk home page, that h264 is not listed. When does Asterisk need h264 at all? If one phone calls another phone, than it is only passed through and does not need it, or am I wrong here? BTW, if I use SER, would this be solved? bye Ronald ___ --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] svn trunk or branches ???
My last update was a while back and as I remember svn trunk did not compile and I was advised to use branches 1.2 till further notice. Have I missed the further notice and can we use now svn trunk or is the advice still to use branches 1.2 ??? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to check which rtp ports my firewall let through?
I thought with iptable -L |grep udp I will find out which ports are open for the rtp stream, but I cannot get this info from here, or at least I cannot interpret it: # iptables -L |grep udp ACCEPT udp -- anywhere anywherestate RELATED,ESTABLISHED LOGudp -- anywhere anywherelimit: avg 3/min burst 5 LOG level warning tcp-options ip-options prefix `SFW2-FWDdmz-DROP-DEFLT ' LOGudp -- anywhere anywherelimit: avg 3/min burst 5 LOG level warning tcp-options ip-options prefix `SFW2-FWDext-DROP-DEFLT ' LOGudp -- anywhere anywherelimit: avg 3/min burst 5 LOG level warning tcp-options ip-options prefix `SFW2-FWDint-DROP-DEFLT ' LOGudp -- anywhere anywherelimit: avg 3/min burst 5 LOG level warning tcp-options ip-options prefix `SFW2-INdmz-DROP-DEFLT ' ACCEPT udp -- anywhere anywhereudp dpts:ndmp:dnp ACCEPT udp -- anywhere anywhereudp dpt:mgcp-callagent ACCEPT udp -- anywhere anywhereudp dpt:4569 ACCEPT udp -- anywhere anywhereudp dpt:5036 ACCEPT udp -- anywhere anywhereudp dpt:sip LOGudp -- anywhere anywherelimit: avg 3/min burst 5 LOG level warning tcp-options ip-options prefix `SFW2-INext-DROP-DEFLT ' LOGudp -- anywhere anywherelimit: avg 3/min burst 5 LOG level warning tcp-options ip-options prefix `SFW2-INint-DROP-DEFLT ' REJECT udp -- anywhere anywherereject-with icmp-port-unreachable However, /etc/rc.d/SuSEfirewall2_final status includes the line: 0 0 ACCEPT udp * * ::/0 ::/0 udp dpts:1:2 Why I am looking for that? My voice connection to phones is usually working, however, we have now also video phones and they do not receive any Video packages, bye Ronald ___ --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] Need somebody for video phone testing
I need somebody who can test with me video phone settings. I use Eyebeam! Please contact me via MSN first: [EMAIL PROTECTED] bye Ronald ___ --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] Blind transfer 3/4 digits
Koopmann, Jan-Peter wrote: On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? Because the transfer button on the SNOM is using a totally different mechanism than sending # to Asterisk. On your snom configuration (like ours) the phone does not start to create/send a SIP message until you hit OK. At that time the entire number is there and a complete SIP transfer is created. Cool down a bit. The problem you are having is most probably just a dialplan problem. It takes some time and experience to get those things right. No need to yell here... What's happen to you guys? I am not yelling, just asking. It is sure not a dialplan question! If it would be a dialplan question, than it would be for each dialing, but it isn't. You mentioned SIP message and that makes me wonder! Are we not using here dtmf ?? that is in my opinion not a sip message, isn't it? If it is a sequence of tones, than why is it different if it is in a string (like snom) or another phone, with single tones? If we understand this part, than is the question, where can I turn on the system to take a longer break between tones still as a string? Back to the dialplan: A Voip number can have different length of digits. Each number is seen as a complete picture, and so a three digit and a four digit number is something different. While in the legacy telephony the digits are worked down one by one and if there is no more use of the digits, they are just garbage and will be not used. Unlike in VoIP, where you can have a three digit number and if you dial four digit, than it is a WRONG number I just verified that: I dialed from 601 to 61522, however, 61522 does not exist, but 615 exists. Guess what? I get a busy tone! That should proof my thoughts (and that without yelling, ... hehehehe) bye Ronald ___ --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] Blind transfer 3/4 digits
David Gagnon wrote: Ronald, Like someone already told you, you should explain more clearly the way you try to transfer, we need more details on the procedure, using which button on which phone. We need every detail to help you. This as nothing to do with the way the dial plan is loaded, this is totally false. I'm sure most of the people here don't understand how you try to transfer. David David, I am not sure how the explanation how to punch the keys changes something, ;-.) Ok, here we go: Snom: pick up the phone and hit ##6014 followed by [ok] Noname: pick up the phone and hit ##6014 must be pushed very fast!!! No end # needed, since the phone 601 starts to ring as soon I reach 1. In my opinion Asterisk remembers all numbers and therefore it does not wait for the 4, since it found a match. This is in VoIP (in my opinion) wrong, since overlapping numbers are allowed. Sip message / dtmf, this is something different! How is the transfer made? Maybe snom does send a sip message, while the noname only send dtmf tones. bye Ronald -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 4 septembre 2006 09:22 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Koopmann, Jan-Peter wrote: On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? Because the transfer button on the SNOM is using a totally different mechanism than sending # to Asterisk. On your snom configuration (like ours) the phone does not start to create/send a SIP message until you hit OK. At that time the entire number is there and a complete SIP transfer is created. Cool down a bit. The problem you are having is most probably just a dialplan problem. It takes some time and experience to get those things right. No need to yell here... What's happen to you guys? I am not yelling, just asking. It is sure not a dialplan question! If it would be a dialplan question, than it would be for each dialing, but it isn't. You mentioned SIP message and that makes me wonder! Are we not using here dtmf ?? that is in my opinion not a sip message, isn't it? If it is a sequence of tones, than why is it different if it is in a string (like snom) or another phone, with single tones? If we understand this part, than is the question, where can I turn on the system to take a longer break between tones still as a string? Back to the dialplan: A Voip number can have different length of digits. Each number is seen as a complete picture, and so a three digit and a four digit number is something different. While in the legacy telephony the digits are worked down one by one and if there is no more use of the digits, they are just garbage and will be not used. Unlike in VoIP, where you can have a three digit number and if you dial four digit, than it is a WRONG number I just verified that: I dialed from 601 to 61522, however, 61522 does not exist, but 615 exists. Guess what? I get a busy tone! That should proof my thoughts (and that without yelling, ... hehehehe) bye Ronald ___ --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 --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0635-4, 2006/09/01 Tested on: 2006/9/4 ¤U¤È 11:40:24 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfer 3/4 digits
wendell hamilton wrote: Please excuse the top-posting. ... so we are faster at the solution, ... ;-) In features.conf, uncomment transferdigittimeout and adjust its timing as desired. You may also want to uncomment and adjust featuredigittimeout to a higher value as well. That was it!!! Now it works!!! Also, since the dialplan does first match, you can eliminate the problem by putting the 4 digit extensions before the 3 digit extensions in the dialplan. See the match as you go section at http://www.voip-info.org/wiki/index.php?page=Asterisk+Extension+Matching Thank you for the link, btw. your comment above does not match the link. Copy of the important part of your provided link: Example FooBar Incorporated wants their incoming telephone calls to be answered with a voice message welcoming the caller and inviting them to choose which extension they want. FooBar has six telephone extensions. Their extension numbers are 1, 2, 21, 22, 31, 32. So this is the context created for incoming calls for FooBar Incorporated: [incoming] exten = s,1,Background(welcome-to-foobar-incorporated) exten = 1,1,Dial(Zap/1) exten = 2,1,Dial(Zap/2) exten = 21,1,Dial(Zap/3) exten = 22,1,Dial(Zap/4 exten = 31,1,Dial(Zap/5) exten = 32,1,Dial(Zap/6) When you call FooBar, Asterisk plays the welcome-to-foobar-incorporated.gsm sound file. After that, having run out of commands to execute, it waits for you to dial something. This is what Asterisk would do if you dialed various options: Number DialedAsterisk's Action 1 Immediately performs Dial (Zap/1) 2 Waits for timeout, then performs Dial(Zap/2) 21 Immediately performs Dial (Zap/3) 22 Immediately performs Dial (Zap/4) 3 Waits for timeout, then hangs up. 31 Immediately performs Dial (Zap/5) 32 Immediately performs Dial (Zap/6) 4 Immediately hangs up. Note that when a caller tries to dial extension 2, they are not connected immediately. Asterisk waits to see if the caller dials more digits, to determine whether the caller wants extension 2 or 21 or 22. As callers would like to be connected immediately if possible, it would be more user-friendly to avoid using ambiguous extension numbers. Thanks for the solution, bye Ronald HTH routerguy -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Monday, September 04, 2006 5:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Blind transfer 3/4 digits David Gagnon wrote: Ronald, Like someone already told you, you should explain more clearly the way you try to transfer, we need more details on the procedure, using which button on which phone. We need every detail to help you. This as nothing to do with the way the dial plan is loaded, this is totally false. I'm sure most of the people here don't understand how you try to transfer. David David, I am not sure how the explanation how to punch the keys changes something, ;-.) Ok, here we go: Snom: pick up the phone and hit ##6014 followed by [ok] Noname: pick up the phone and hit ##6014 must be pushed very fast!!! No end # needed, since the phone 601 starts to ring as soon I reach 1. In my opinion Asterisk remembers all numbers and therefore it does not wait for the 4, since it found a match. This is in VoIP (in my opinion) wrong, since overlapping numbers are allowed. Sip message / dtmf, this is something different! How is the transfer made? Maybe snom does send a sip message, while the noname only send dtmf tones. bye Ronald -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 4 septembre 2006 09:22 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Koopmann, Jan-Peter wrote: On Sunday, September 03, 2006 3:40 AM Ronald Wiplinger wrote: try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? Because the transfer button on the SNOM is using a totally different mechanism than sending # to Asterisk. On your snom configuration (like ours) the phone does not start to create/send a SIP message until you hit OK. At that time the entire number is there and a complete SIP transfer is created. Cool down a bit. The problem you are having is most probably just a dialplan problem. It takes some time and experience to get those things right. No need to yell here... What's happen to you guys? I am not yelling, just asking. It is sure not a dialplan question! If it would be a dialplan question
Re: [asterisk-users] Blind transfer 3/4 digits
Anthony Rodgers wrote: With respect, the problem is with your numbering plan.. WHERE do you see a problem in the numbering plan? I see the problem in ASTERISK, because it does not wait for the last digit!!! Where can I set that it waits for it? The beauty on voip IS that you can have different length and overlapping, bye Ronald CP On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote: I found a problem in blind transfer: I have an extension number 601 and I have an extension 6014 If I get a call on 615 (snom) and transfer to 6014 it works, since snom requires me to hit ok If I get a call on 601 and transfer to 6014, than 601 will get the busy signal and I hang up as usually with transfer. Howerver the caller get the announcements: I could not get that, What could be the problem ? bye Ronald ___ --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 --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0635-4, 2006/09/01 Tested on: 2006/9/2 ¤U¤È 03:52:00 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Blind transfer 3/4 digits
David Gagnon wrote: Ronald, You seem to be a little bit angry about VoIP. If so, I could give you my old Nortel system. Does this would make you happy? David David, I am not angry about VoIP, but please send my your old Nortel system ! I just do not understand why I can DIAL 601 and 6014, but not use blind transfer. Is the question too difficult? I am sure there is somewhere a switch to say, wait two seconds (as for dialing) before you assume it is a complete number. It is also strange that snom phone can do it correct, because it uses the ok key. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 2 septembre 2006 04:20 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Anthony Rodgers wrote: With respect, the problem is with your numbering plan.. This answer is therefore totally nonsense !!! (With all respect!!!) Both answers have actually not lead to any step further, but to more messages. I use to refer to such answers as NON-ANSWERS. Please only reply if and really only if you know a solution for the problem! Thanks for your understanding. bye Ronald - again, I am not angry at all. WHERE do you see a problem in the numbering plan? I see the problem in ASTERISK, because it does not wait for the last digit!!! Where can I set that it waits for it? The beauty on voip IS that you can have different length and overlapping, bye Ronald CP On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote: I found a problem in blind transfer: I have an extension number 601 and I have an extension 6014 If I get a call on 615 (snom) and transfer to 6014 it works, since snom requires me to hit ok If I get a call on 601 and transfer to 6014, than 601 will get the busy signal and I hang up as usually with transfer. Howerver the caller get the announcements: I could not get that, What could be the problem ? bye Ronald ___ --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 --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0635-4, 2006/09/01 Tested on: 2006/9/2 ¤U¤È 03:52:00 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Keys pressed not registering ...
Lenny wrote: Hello all, For some reason when dialing in I get the IVR or if I forward to my conference line... any keys pressed seem like they aren’t received .. Like I’m pressing them, but they aren’t being registered with the server .. Any ideas? I’m using the vmware nerdvittles build, the latest trixbox v1.1 .. FreePBX 2.1.1. Everything else works just fine. I’m using VoIPDiscount for outgoing and Stana-in/Stanaphone to receive calls. Any help is appreciated.. Have a look at the dtmfmode settings, inband, rfc2833, ... and try different settings. bye Ronald Regards, LB ___ --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 --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0635-4, 2006/09/01 Tested on: 2006/9/2 ¤U¤È 04:42:02 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Keys pressed not registering ...
Lenny wrote: Hello Ronald .. This is what I'm trying to learn of now .. Where in freepbx do I place these settings? sip.conf ;-) that was easy, ... do you have another question? bye Ronald Trunk settings? If I could just get that bit of info.. Thanks LB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ronald Wiplinger Sent: Saturday, September 02, 2006 11:52 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Keys pressed not registering ... Lenny wrote: Hello all, For some reason when dialing in I get the IVR or if I forward to my conference line... any keys pressed seem like they aren’t received .. Like I’m pressing them, but they aren’t being registered with the server .. Any ideas? I’m using the vmware nerdvittles build, the latest trixbox v1.1 .. FreePBX 2.1.1. Everything else works just fine. I’m using VoIPDiscount for outgoing and Stana-in/Stanaphone to receive calls. Any help is appreciated.. Have a look at the dtmfmode settings, inband, rfc2833, ... and try different settings. bye Ronald Regards, LB ___ --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] Blind transfer 3/4 digits
Kevin Smith wrote: Dialing a number and transferring a number are two different things. And no offense, you are not really providing a lot of details along with your problem. So you can dial the numbers but not transfer from one to the other. I was not thinking that it would be too much difference. Therefore I also do not know what more info could help to distinguish the problem. I hardly can post my entire configuration. What does the CLI say when you try the transfer? That would provide a lot of information that could clue you in to what is going on. You hit another problem with that. I hardly see here anything anymore. The messages fly by so fast, Especially annoying messages: chan_sip.c:10888 handle_request_register: Registration from 'sip:192.168.250.20' failed for '192.168.250.244' - Username/auth name mismatch -- Got SIP response 486 Busy Here back from 192.168.250.244 -- Got SIP response 400 Bad Request back from xx.xx.xx.126 NOTICE[5936]: chan_sip.c:9600 handle_response_register: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3) . It would be nice to filter the CLI for such investigation for a moment. What type of phones are you using? Some phones have the ability to pattern match and wait for a certain number of seconds before sending the number to asterisk. For example. On our Polycom phones a user has 3 seconds (between digits) to enter in 10 digits. This could be where most of your problem is. That is a very good point and I will contact the manufacturer of these no-name phones. My guess the problem lies with the Phones, not Asterisk form the information you provided. I disagree with that! Why Asterisk treats dialing and transfer different. That makes not really sense, does it? bye Ronald Kevin Ronald Wiplinger wrote: David Gagnon wrote: Ronald, You seem to be a little bit angry about VoIP. If so, I could give you my old Nortel system. Does this would make you happy? David David, I am not angry about VoIP, but please send my your old Nortel system ! I just do not understand why I can DIAL 601 and 6014, but not use blind transfer. Is the question too difficult? I am sure there is somewhere a switch to say, wait two seconds (as for dialing) before you assume it is a complete number. It is also strange that snom phone can do it correct, because it uses the ok key. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 2 septembre 2006 04:20 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Anthony Rodgers wrote: With respect, the problem is with your numbering plan.. This answer is therefore totally nonsense !!! (With all respect!!!) Both answers have actually not lead to any step further, but to more messages. I use to refer to such answers as NON-ANSWERS. Please only reply if and really only if you know a solution for the problem! Thanks for your understanding. bye Ronald - again, I am not angry at all. WHERE do you see a problem in the numbering plan? I see the problem in ASTERISK, because it does not wait for the last digit!!! Where can I set that it waits for it? The beauty on voip IS that you can have different length and overlapping, bye Ronald CP On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote: I found a problem in blind transfer: I have an extension number 601 and I have an extension 6014 If I get a call on 615 (snom) and transfer to 6014 it works, since snom requires me to hit ok If I get a call on 601 and transfer to 6014, than 601 will get the busy signal and I hang up as usually with transfer. Howerver the caller get the announcements: I could not get that, What could be the problem ? bye Ronald ___ --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] Blind transfer 3/4 digits
Tim St. Pierre wrote: Are you using # to transfer? If so, it's not sending it as a new call, it's just sending asterisk digits using whatever DTMF mode. Asterisk parses these based on a first match in the dialplan. Make sure that the longer extension numbers are loaded first in the dialplan. That is a good thought. I can remember that the docs said that you cannot force the order of the dialplan, except with includes. I will try that way. However, I have doubts as well. If you are right, than why snom phone does not have this problem? Would not here also the first match count? bye Ronald -Tim On September 2, 2006 20:12, Ronald Wiplinger wrote: Kevin Smith wrote: Dialing a number and transferring a number are two different things. And no offense, you are not really providing a lot of details along with your problem. So you can dial the numbers but not transfer from one to the other. I was not thinking that it would be too much difference. Therefore I also do not know what more info could help to distinguish the problem. I hardly can post my entire configuration. What does the CLI say when you try the transfer? That would provide a lot of information that could clue you in to what is going on. You hit another problem with that. I hardly see here anything anymore. The messages fly by so fast, Especially annoying messages: chan_sip.c:10888 handle_request_register: Registration from 'sip:192.168.250.20' failed for '192.168.250.244' - Username/auth name mismatch -- Got SIP response 486 Busy Here back from 192.168.250.244 -- Got SIP response 400 Bad Request back from xx.xx.xx.126 NOTICE[5936]: chan_sip.c:9600 handle_response_register: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3) . It would be nice to filter the CLI for such investigation for a moment. What type of phones are you using? Some phones have the ability to pattern match and wait for a certain number of seconds before sending the number to asterisk. For example. On our Polycom phones a user has 3 seconds (between digits) to enter in 10 digits. This could be where most of your problem is. That is a very good point and I will contact the manufacturer of these no-name phones. My guess the problem lies with the Phones, not Asterisk form the information you provided. I disagree with that! Why Asterisk treats dialing and transfer different. That makes not really sense, does it? bye Ronald Kevin Ronald Wiplinger wrote: David Gagnon wrote: Ronald, You seem to be a little bit angry about VoIP. If so, I could give you my old Nortel system. Does this would make you happy? David David, I am not angry about VoIP, but please send my your old Nortel system ! I just do not understand why I can DIAL 601 and 6014, but not use blind transfer. Is the question too difficult? I am sure there is somewhere a switch to say, wait two seconds (as for dialing) before you assume it is a complete number. It is also strange that snom phone can do it correct, because it uses the ok key. -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Ronald Wiplinger Envoyé : 2 septembre 2006 04:20 À : Asterisk Users Mailing List - Non-Commercial Discussion Objet : Re: [asterisk-users] Blind transfer 3/4 digits Anthony Rodgers wrote: With respect, the problem is with your numbering plan.. This answer is therefore totally nonsense !!! (With all respect!!!) Both answers have actually not lead to any step further, but to more messages. I use to refer to such answers as NON-ANSWERS. Please only reply if and really only if you know a solution for the problem! Thanks for your understanding. bye Ronald - again, I am not angry at all. WHERE do you see a problem in the numbering plan? I see the problem in ASTERISK, because it does not wait for the last digit!!! Where can I set that it waits for it? The beauty on voip IS that you can have different length and overlapping, bye Ronald CP On 1-Sep-06, at 10:37 PM, Ronald Wiplinger wrote: I found a problem in blind transfer: I have an extension number 601 and I have an extension 6014 If I get a call on 615 (snom) and transfer to 6014 it works, since snom requires me to hit ok If I get a call on 601 and transfer to 6014, than 601 will get the busy signal and I hang up as usually with transfer. Howerver the caller get the announcements: I could not get that, What could be the problem ? bye ___ --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] Blind transfer 3/4 digits
I found a problem in blind transfer: I have an extension number 601 and I have an extension 6014 If I get a call on 615 (snom) and transfer to 6014 it works, since snom requires me to hit ok If I get a call on 601 and transfer to 6014, than 601 will get the busy signal and I hang up as usually with transfer. Howerver the caller get the announcements: I could not get that, What could be the problem ? bye Ronald ___ --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] GIZMO and Asterisk, Failed to authenticate
[Aug 31 04:32:22] NOTICE[20241]: chan_sip.c:5291 sip_reg_timeout:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #984) [Aug 31 04:32:23] NOTICE[20241]: chan_sip.c:9600 handle_response_register: Failed to authenticate on REGISTER to '[EMAIL PROTECTED]' (Tries 3) sip.conf: register = 1747mynumber:[EMAIL PROTECTED] ; Gizmoproject [proxy01.sipphone.com] type=friend context=default disallow=all allow=ulaw allow=alaw allow=ilbc dtmfmode=rfc2833 host=proxy01.sipphone.com insecure=very secret=mypassword username=1747mynumber canreinvite=yes What did I wrong? bye Ronald ___ --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] Wellgate 3804a: Got SIP response 486 Busy Here
I cannot explain why I get all the time: Got SIP response 486 Busy Here back from 192.168.250.244 I have a Wellgate 3804a there. How can I solve it? bye Ronald ___ --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] Am I looking for automon?
I want to record a call, either it is an incoming call or an outgoing call. I have in features.conf: automon = *1 However, I am not sure if that is what I need, and how to use it. bye Ronald ___ --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 crashes after two years
Michael Welter wrote: My Asterisk colo server has been up for almost two years. Today it crashed. When I gave the reboot command, it crashed so hard that it had to be power cycled. I wasn't in attendance, but I can speculate that it had a kernel panic during the shutdown. Yesterday I added a PHP agi script, and it had been user over 1000 times before the crash. I don't think the Linux/Asterisk crash is coincidental. Can someone give me things to look for? I'm watching memory, and it has 750MB free (out of 1GB). When I restart Asterisk, I see 19 processes--is this normal? What else should I be doing to narrow down on this problem. Thanks for your help. Have you checked the log files? Do you use Real-time? Is your database ok? Have you checked the hard disk space? 2 years Asterisk sounds strange, since I can remember there was a bug with the date a year ago. If you have not upgraded, than this bug is still in your code. Maybe you just meant no reboot for two years. bye Ronald ___ --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] Problems with recording
Tim St. Pierre wrote: Try creating an extension with a lower priority that answers the channel first. If you don't, the application will run, but the call will timeout as no answer, since it was never actually answered. It sounds weird, but this is how you get messages like please check the number and try your call again without getting billed for the call. - Asterisk doesn't indicate answer until you tell it to, or until it bridges a call. -Tim On August 31, 2006 06:55, Giedrius Augys wrote: Hi, I am trying to record a speech with this command: exten = 205,3,Record(speech:wav). But it records aproximately about 10 seconds and asterisk hangs up. Does somebody know how to solve this problem, I also tried with max duration, but it didn't help.. Are you sure it is recording? Do you get a partial file or nothing? Do you have permission to write to your defined recording directory? Maybe you just record into the RAM and you cannot access the recording directory and so, after the recording time in the RAM is full, it has no other choice than to hang up. bye Ronald ___ --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] Wellgate 3804a
I want that each call from PSTN goes to Asterisk to the context for this line. Within this context can be a menu or a dial command, ... As more I read, as more I get confused, ... and each try is not working! My sip.conf: [WG88621001] type=friend defaultip=192.168.250.244 insecure=very context=incoming_WG dtmfmode=rfc2833 [EMAIL PROTECTED] language=en nat=yes auth=md5 host=dynamic canreinvite=no disallow=all allow=ulaw allow=alaw allow=g726 allow=g729 username=88621001 fromuser=88621001 secret= qualify=yes canreinvite=no extensions.conf [incoming_WG] exten = s,1,NoOp(*** I am here now ***) Wellgate settings: Network Interface: IP address of the device 192.168.250.244 Sip Config: Mode Proxy Primary Proxy IP address: 192.168.250.20 Line 1 Number:88621001 Security Config Line1 Account: WG8862001 Line 1 password: (secret from the asterisk setting) Line configuration Line 1 (LINE) Type: FXO Hunting Group: 1 HotLine: 601 Registration: Not Registered Status: Ready System Configuration Keypad type: rfc2833 Routing Table Index: IP Default Destination: FXO E.164: x Index: FXO Destination: IP Default E.164: x *CLI sip show peers like ^WG Name/username HostDyn Nat ACL Port Status WG88621001/88621001(Unspecified)D N 0UNKNOWN 1 sip peers [0 online , 1 offline] Calls from PSTN comes to the IVR asking for the extension number and than nothing happens. Asterisk shows nothing either. Can somebody enlighten me: 1. Do I need to have a register statement in sip.conf? (I tried register = 88621001:secret-from-above ; Wellgate GW.3801-Line-1) 2. where to turn off the IVR? 3. Do I use the right name, user name, line account, line 4. Hotline. Why, how, which number?? bye Ronald ___ --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] if command for or missing callerid?
I am looking for a way to make a decission in the dialplan if I have a caller id or not. What I want to do with it: Call on the PSTN line should either use astcc.agi with the caller-id in place as card number, or asking for the calling card number. How can I make this gotoif ??? bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] re-writing the dial plan - some hints please
My dialplan grew over the last months and I want to restructure it. What
hints do you have for me?
There are some points I want to do, but none of my tests worked.
I use realtime, and have there a field called key, which can have
several flags.
E.g. a flag if the user is allowed to use a conference room, can call
long distance, can call overseas, can call local pstn, different
tariffs,
I tried something like:
[test-key]
exten = _.,1,NoOp(variable key is ${key})
exten = _.,2,Set(flag_int =${CUT(key,,1)})
exten = _.,3,Set(tarif=${CUT(key,,2)})
exten = _.,4,NoOP(flag_int is ${flag_int} and tarif is ${tarif})
and wanted to use this variables in the next context, by using include
statments, but it did not work.
[caller]
include = test-key
include = A
include = B
...
The idea was to set at each entrance point first all flags and
variables. Than I can use a common dialplan.
If a flag is set, than I could include another context. Unfortunately
there is no IF()include. I might be able to set a jump in each context
to the end if the flag is not set.
Any idea how I can do that?
Any ideas of structuring the dialplan more efficiently?
bye
Ronald
___
--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] sox gsm
sox needs for gsm an optional library. I was not able to locate this one. Can anybody point me to this place? bye Ronald ___ --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] Zip code, city and area codes
Is there a table available, which tells me if a zip code, city and area code matches? For now I did it with google, type each info in and found out if it matches, but it would be easier if there is a table available. bye Ronald ___ --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] NuFone, please send the log file
Kevin P. Fleming wrote: Can we please keep the discussions about carriers, money, jobs, work, etc. off of this list? This is not the place to discuss your experiences with _any_ company, it's a place to talk about Asterisk and using Asterisk. Please move flamewars and similar discussions to some other forum. I agree with you! Which place is in your opinion the right place? As long there is no other place, such messages will always pop up. bye Ronald ___ --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] NuFone, please send the log file
trixter aka Bret McDanel wrote: On Tue, 2006-07-11 at 20:51 -0400, C F wrote: While I don't disagree with you, look at what my point was, just accusing them for such without any documentation doesn't make sens. He only brought that up after people started questioning it. So I dunno. And lets face it, this is the internet there is really no proof of anything. Screen captures of a webpage? That is easy enough to forge. Invoices? They too are easy enough to forge. I don't think so!!! I guess you never lost your web site (accidentally) a have been than very happy that at least a big portion you could retrieve from the Internet archive!!! It is even funny to see how some web pages have been developed and changed. Even if someone states they had horrible call quality you have no proof, but that is generally accepted that that one person experienced that. And where does that leave you? You have to either take a chance on your own or go with those that you trust and/or whatever is said the most. Call quality changes often and in my experience depends not so often from the VoIP provider, but from the users Internet connection. bye Ronald So since its hard to get any sort of proof you kinda just have to accept that it happened or not and move on. ___ --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 --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0628-2, 2006/07/11 Tested on: 2006/7/12 ¤W¤È 09:27:28 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] [EMAIL PROTECTED] founded
To keep the Asterisk mailing list free of Voip provider complaints: VoIP is a growing business area. We all find days of problems. Some companies can handle problems. Some VoIP providers create problems. In this group we can discuss and learn how to handle conflicts. What to do and what not to do in this group: 1. Report cases and your impression. 2. Try to word it polite, even it is sometimes hard to do so. 3. Do not use any words you would not say also to your own 12 year old child. 4. Accept advices. Caution and remember, this is a Google list. All messages are UNREMOVABLE in the Search engine, Good luck! ___ --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] NuFone, please send the log file
Dear NuFone, Without misunderstanding I ask you again, please send the log file and pay back my money! Not following this request results in the assumption that NuFone is cheating and I will post this info every hour on more Internet places. This should help that other people will not trap into a cheating company. bye Ronald ___ --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] NuFone, please send the log file
Andrew D Kirch wrote: Ronald Wiplinger wrote: Dear NuFone, Without misunderstanding I ask you again, please send the log file and pay back my money! Not following this request results in the assumption that NuFone is cheating and I will post this info every hour on more Internet places. This should help that other people will not trap into a cheating company. bye Ronald ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I'm going to note two more issues I've just found with this post. 1. this is a specifically NON-Commercial list (your post is commercial) Yes, I did not write too much, but one part of the issue is, that NuFone does not answer to technical questions either, but asks for set-up help in IRC. So to see, it is a hint for technical people to take care if they suddenly get an offer for consulting, just when you ask a technical question. 2. you have threatened to post it to further such lists and forums where it is not desired (your post is being made in bulk) I therefore must determine you have posted UCE/UBE and you are a spammer. I strongly disagree with that! places are not only lists! Maybe you are too new on the net to figure out, that there are still other places. Have you tried to Google for Nufone? Than you might find other places too. Again, I just want to have the log files. I do not get answer and that is a fact. If you have good contacts to Jeremy, maybe you can convince him to send the log file. It is that simple. I have set-up a filter for NuFone, and when I have time and catch a message with that trigger word, I will post my thoughts. Thanks for pointing out not to send too many messages. However, to answer to another ones message, . have a nice day! bye Ronald ___ --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 --- avast! Antivirus: Inbound message clean. Virus Database (VPS): 0628-2, 2006/07/11 Tested on: 2006/7/12 ¤W¤È 07:29:36 avast! - copyright (c) 1988-2006 ALWIL Software. http://www.avast.com -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] NuFone, please send the log file
Andrew D Kirch wrote: Ronald Wiplinger wrote: Dear NuFone, Without misunderstanding I ask you again, please send the log file and pay back my money! Not following this request results in the assumption that NuFone is cheating and I will post this info every hour on more Internet places. This should help that other people will not trap into a cheating company. bye Ronald ___ --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 Wow, that was productive, either never do that again or I'm invoicing you for the time it took me to read it write a response telling you what a moron you were for posting it in the first place, and then deleting it and making sure that the poor hard drive it was stored on was shot humanely and put out of its misery. In other words take it off-list. This is not the people-who-bitch-about-nufone (for values of nufone that equate to any provider BroadVoice anyone?), Broadvoice was the other one. They sent me a message, sorry our service does not work and charged without a refund my credit card. or #nufone-sucks on some IRC channel. Wow, I did not know that there is such a channel. Than it seems more likely that they do suck !!! hehehehehe Quite honestly (and I've noted before) that NuFone seems to have a business model of catering only to clued customers. I am still curious as to the eventual outcome (their long-term survival), but you have aptly demonstrated above why you yourself aren't a customer. Get a clue, grow one, buy one EBay but quit spouting this crap on a help list, for I tell you that there is no help for you and not because NuFone screwed you. I find it strange, that I was with them a long time, kept silent when they lost all their connectivity and got shot from them afterwards still. Anyway, there a tones of other providers available and many of them who try to help and not just to take your money. I personally believe that the market is over in 3 years and that we should now look for other business. The big players are taking over soon. bye Ronald ___ --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] NuFone, please send the log file
Andrew D Kirch wrote: Michael Workman wrote: So Nufone Screwed ya I feel Sorry... W Take your Lumps... Cut Your Losses and Get on with Life Your not the only one Nufone Screwed They Screwed me Out of $3,000.00 How do you figure this at 2.9c/min? Andrew That is easy to calculate: 3,000 US$ times your zip code times the phone number you are calling times 2.9cents/5 seconds divided by the Social Security number of the called party ... Or how does NuFone calculate that? But hey, just look at the log file, hmm, didn't we start here? WHERE ARE THE LOG FILES Thanks for all the encouraging funny answers. I go now to 7-eleven to buy some candies, ... bye Ronald ___ --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] NuFone, please send the log file
Michael Workman wrote: I am not talk about Call Time.. They Screwed me by Me Hiring them to consult on setting up server and they took the money and never did the work This they tried also with me, but I only answered, that I would like to learn it by myself, bye Ronald -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew D Kirch Sent: Tuesday, July 11, 2006 8:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] NuFone, please send the log file Michael Workman wrote: So Nufone Screwed ya I feel Sorry... W Take your Lumps... Cut Your Losses and Get on with Life Your not the only one Nufone Screwed They Screwed me Out of $3,000.00 How do you figure this at 2.9c/min? Andrew ___ --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] NuFone suggests to use Vonage!!!!
Part of a conversation with NuFone. It is untrue, that they do not answer, but if than: Quote: 3. change your attitude towards customers!! No, if you don't like it, go use Vonage. End of quote! I had always problems with these people. bye Ronald ___ --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] ASTCC: how can I limit to xxx minutes per week?
The big player show us, to limit the free phone calls per week to a certain amount. How can we do that with ASTCC? bye Ronald ___ --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] ASTCC: inuse flag still hangs!
I have patched astcc.agi with the HUP patch, but it still hangs from time to time. Asterisk SVN-branch-1.2-r25165M built by root @ vpbx on a x86_64 running Linux on 2006-05-07 00:31:09 UTC bye Ronald ___ --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] time variable
I want to get a variable, depending on the time. I tried this one, but it does not work: exten = 75,1,Set(guess=SYSTEM(echo $((1 + $(date +%S)*100 % 23))) The idea is that the variable guess will change every 23 times per minute. How would be the right syntax? bye Ronald Wiplinger ___ --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] I am looking for a (graphical) statistic program
I am looking for a graphical statistic program. What I want to see is: a. my bandwidth (MRTG I use now from my upstream, but the time seems to be 20 minutes wrong,...) b. how many phone calls are at the same time (to get the feeling how much bandwidth how many phone calls are using) c. how long phone calls are, separated to different criteria, like prefix number, duration. most of these is in the program from areski, with the exeption that the numbers are wrong, like graphic shows 5 phone call and load shows 4 calls, . What are you using? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to continue after a match in an include
I am looking for a way that after a successfully match in one include the next include is still visited. The first include should just set some variables. I tried to number this extension block either with _. or with s and since it matches, the function (setting some variables) have been done. After that, I want to go to the next include, which has a match for _91NNN. However, since the first match was already successful, the next includes are not visited anymore. How can I overcome this problem? bye Ronald Wiplinger ___ --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] channel shows to be in use
when I try asterisk -rx show channels concise I get an output of: SIP/tf.voipmich.com-8671 ... SIP/1110-78ac The phone 1110 is not anymore on a phone call. How can I remove this zombie channel? bye Ronald Wiplinger ___ --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] multiple includes
I want to set some variables for each phone. For that I use setvar in Real-time. At the beginning of each context should be this include statement before all other include statements. How can I rewrite the dial plan, so that after the include var-key other include statements are still used? bye Ronald Wiplinger [mycontext] include = var-key include = ___ --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] username in Real-time changes all the time
I cannot explain that: One of my users shows up in sip show peers as 654200/Elmit_Unl I can set it back to 654200/654200 but it will change back to 654200/Elmit_Unl Why? bye Ronald Wiplinger ___ --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] Realtime: how to use column setvar?
How can I use the column setvar in my dialplan? I am not sure if it is for that what I need: Many phones have the same jump in place, but need a few variables different, like tariff, silent, need_password, I have for tariff = 4 variations, for silent=2, for need_password=2 ... If I solve it like now, I need 4x4x2 = 32 context variations. If I could use a field in the Real-time database, these info could be there and it is only one context. bye Ronald ___ --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] ASTCC: customer wants 100 accounts
JP Carballo wrote: Ronald Wiplinger wrote: I got a request for one customers to set-up 100 accounts. I use usually the Caller-ID as the card number. Is there a way to make it for 100 accounts easier? To generate 100 cards is not a problem, but if it would work with one account number would be even better I could use a different context for this customer and use only his account code as card number. Any advice would be appreciated. I'm not going to ask why the customer needs 100 cards. If he wants to access them all from 1 account, wouldn't he be happier with a single card that has the credits of 100 cards? In short, an account, not a card. Get my drift? Or, try making another brand with a markup of 100% I guess. Never tried that one though. He want to use 100 phones at the same time!!! bye Ronald -- Ronald Wiplinger (CEO of ELMIT) http://www.elmit.com http://voip.elmit.com http://e-paper.elmit.com Tel. (M) +886.939.775.516 (O) +886.2.2835.7765 (ENUM) or FWD 511208 - I'm a SpamCon Foundation Member, #694, Verify it at http://www.spamcon.org PS: Spam prevention! Our system is protected with a spam prevention program. If you send us an e-mail, our system will send you a confirmation message back. Just reply to this confirmation message please. After receiving this confirmation message, our system will send the hold message (one) and all future messages (after the received confirmation message) to me without asking you again. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ASTCC: customer wants 100 accounts
I got a request for one customers to set-up 100 accounts. I use usually the Caller-ID as the card number. Is there a way to make it for 100 accounts easier? To generate 100 cards is not a problem, but if it would work with one account number would be even better I could use a different context for this customer and use only his account code as card number. Any advice would be appreciated. bye Ronald Wiplinger ___ --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] ASTCC: How to reset periodically all card in use flag back?
Nicolás Gudiño wrote: Hi Ronald, If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. You should install php-pcntl (or compile php to add support for process control functions). The inuse problem will be fixed then. Regards, Can you please give us more info about that? What is php-pcntl? What should it do? How can it be used to be a solution? bye Ronald Wiplinger ___ --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] ASTCC: How to reset periodically all card in use flag back?
JP Carballo wrote: Ronald Wiplinger wrote: If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. I have created a number the user can dial to reset this flag. However, that is written in the manual!!! Who reads a manual anyway I want to make to reset all in use flag with a program. Has anybody done it, or has a better idea? My idea is to check every 5 minutes, the database, which cards are set in use and check if this is true, if not reset it. Q: How do I know if a card is in use? Still banging your head over this one? Get the card's uniqueid and use it to check if that particular channel is up. ok, How do I check if a particular channel is up? (Wasn't that what I asked above anyway) bye Ronald Wiplinger ___ --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] ASTCC: How to reset periodically all card in use flag back?
If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. I have created a number the user can dial to reset this flag. However, that is written in the manual!!! Who reads a manual anyway I want to make to reset all in use flag with a program. Has anybody done it, or has a better idea? My idea is to check every 5 minutes, the database, which cards are set in use and check if this is true, if not reset it. Q: How do I know if a card is in use? bye Ronald Wiplinger ___ --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] Is anybody using XEN in conjunction with Asterisk and/or Openser?
Is anybody using XEN in conjunction with Asterisk and/or Openser? I would like to get some info about such an environment and experience reports. bye Ronald Wiplinger ___ --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] Voip* 300 minutes limit, credit expires
Betamax makes our life more and more difficulty, hehehehehehe.
I found (today) that the free calls are limited to 300 minutes per week.
It is good to know what excess use means!
That gives now also a challenge in the dialplan
Let's assume we have 5 accounts, each one has 300 minutes.
We use a variable as provider and get the right value of the not
outmaxed provider into this variable.
How can we do that?
exten = _9011Z.,103,Dial(SIP/00${EXTEN:[EMAIL PROTECTED],30)
should be replaced with ${voipdiscount}
and we need before a statement that finds the content of this variable.
Q: Does anybody know how to download the recently used statement?
I am interested how Voip* will react to the recently law change in
Germany, where for mobile phone operator (and I assume that this law can
be used for Voip* as well) a prepaid value may not expire anymore
Now lets look at Voip* pricing:
12 US$ per month for 300 x 4 minutes, with the expiration within 3
months (13 weeks) == 1200/(13*300) = 0.3 cents in the BEST case!!! If
you use more than 300, than you have to pay at least whatever that
means in real numbers 1.2 cents.
It is getting more and more complicated, and that for pennies!!!
Unfortunately you cannot reach anybody there. I would like to have only
ONE account and pay more for multiple use than this kind of tricks!
just some thoughts!
Ronald
___
--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] voipbuster dtmf tones?
I failed to transmit dtmf via voipbuster to the destination. Does anybody have success, if how to set it up? bye Ronald Wiplinger ___ --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] transfer other features
*CLI show features Builtin Feature Default Current --- --- --- Pickup*8 *8 Blind Transfer# ## Attended Transfer *2 One Touch Monitor *1 Disconnect Call * *0 Dial option is tTwWr I tried to call from 601 to 615 601 keys in *0nothing happens 603 keys *8 I get the phone call # 621 nothing happens What do I miss ??? bye Ronald Wiplinger ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to make this into a Macro?
I have for each phone such a paragraph in my dialplan.
I would like to save this by using a Macro. How can I do that?
exten = 8863959,1,Dial(SIP/8863959,60,r)
exten = 8863959,2,NoOp(${DIALSTATUS})
exten = 8863959,3,Voicemail,[EMAIL PROTECTED]
exten = 8863959,104,Voicemail,[EMAIL PROTECTED]
exten = 8863959,105,hangup
___
--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] Xlite and # code after call is connected
Can anybody tell me how I can key in # codes after the call is established? All what happens now is that the call will be placed on hold and a new call will initiate!!! bye Ronald Wiplinger ___ --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] Xlite and # code after call is connected
Ronald Wiplinger wrote: Can anybody tell me how I can key in # codes after the call is established? All what happens now is that the call will be placed on hold and a new call will initiate!!! Just enter the required digits, just as if you are accessing voicemail. Don't press the send button. ___ --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] AsteriskOUT
[EMAIL PROTECTED] wrote: Hi all, Has anyone in the group tried the services of www.asteriskout.com. (lunaphone) Just thought of letting you all be aware not to fall on their services as it always seen attractive but to my experience they had always been pulling the the legs of customers with an approach of making money through ways that are thought to be non industrial. Are you talking about BUSINESS If so, there is another mailing list available! If it is a USERS question, and allow me please to quote the name of this email list: Asterisk Users Mailing List - Non-Commercial Discussion than you may specify your problem so that we can try to help you. Just wanted to warn you guys wow! bye Ronald Wiplinger ___ --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] Is there a dialplan emulator available?
I would like to test my extensions.conf before I give it to my users. Is there a dialplan emulator available? bye Ronald Wiplinger ___ --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 -rx 'sip show peers'
I upgraded recently to Asterisk SVN-branch-1.2-r25165M the commandline asterisk -rx 'sip show peers' returns with the first line: on Is that a bug, or how can I omit it? I used: asterisk -rx 'sip show peers'|grep OK|sort | tee /dev/tty |wc -l; echo registered at ELMIT which results (because of on) to: Binary file (standard input) matches 1 registered at ELMIT ___ --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 Make Asterisk-addons
Dan Journo wrote:
The following occurs during make asterisk-addons.
I'm ok with asterisk but debugging things like this isnt my strong point.
Can anyone give me a pointer?
Thanks
Dan Journo
[EMAIL PROTECTED] src]# cd asterisk-addons
[EMAIL PROTECTED] asterisk-addons]# make
make -C format_mp3 all
make[1]: Entering directory `/usr/src/asterisk-addons/format_mp3'
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declara tions
-D_REENTRANT -D_GNU_SOURCE -O6-c -o format_mp3.o format_mp3.c
In file included from /usr/include/asterisk/logger.h:28,
from /usr/include/asterisk/lock.h:83,
from format_mp3.c:20:
/usr/include/asterisk/compat.h:20: error: syntax error before
__extension__
/usr/include/asterisk/compat.h:20: error: syntax error before '' token
In file included from /usr/include/asterisk/utils.h:36,
from /usr/include/asterisk/cdr.h:48,
from /usr/include/asterisk/channel.h:113,
from format_mp3.c:21:
/usr/include/asterisk/strings.h:264: error: syntax error before
__extension__
/usr/include/asterisk/strings.h:264: error: syntax error before ';' token
/usr/include/asterisk/strings.h:264: error: `__len' undeclared here
(not in a fu nction)
/usr/include/asterisk/strings.h:264: error: initializer element is not
constant
/usr/include/asterisk/strings.h:264: error: syntax error before if
/usr/include/asterisk/strings.h:264: error: redefinition of `__retval'
/usr/include/asterisk/strings.h:264: error: `__retval' previously
defined here
/usr/include/asterisk/strings.h:264: error: syntax error before const
/usr/include/asterisk/strings.h:264: error: syntax error before '}' token
/usr/include/asterisk/strings.h:280: error: conflicting types for `strtoq'
/usr/include/stdlib.h:346: error: previous declaration of `strtoq'
format_mp3.c:46: error: redefinition of `struct ast_filestream'
format_mp3.c:325: warning: function declaration isn't a prototype
format_mp3.c: In function `load_module':
format_mp3.c:336: warning: passing arg 1 of `ast_format_register' from
incompati ble pointer type
format_mp3.c:336: error: too many arguments to function
`ast_format_register'
format_mp3.c: At top level:
format_mp3.c:342: warning: function declaration isn't a prototype
format_mp3.c:347: warning: function declaration isn't a prototype
format_mp3.c:359: warning: function declaration isn't a prototype
format_mp3.c:365: warning: function declaration isn't a prototype
{standard input}: Assembler messages:
{standard input}:58: Error: symbol `__retval' is already defined
make[1]: *** [format_mp3.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk-addons/format_mp3'
make: *** [format_mp3/format_mp3.so] Error 2
I have got exactly the same! Could anybody solve it?
bye
Ronald Wiplinger
___
--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] Upgrade SVN failed !!!
I upgraded * via svn and it did not work !!!
1. asterisk-addon did not compile!
pbx:/usr/local/src/svn-versions/asterisk-addons # make
./mkdep -fPIC -I../asterisk -D_GNU_SOURCE -I/usr/include/mysql `ls *.c`
make -C format_mp3 all
make[1]: Entering directory
`/usr/local/src/svn-versions/asterisk-addons/format_mp3'
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o
common.o common.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o
dct64_i386.o dct64_i386.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o
decode_ntom.o decode_ntom.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o
layer3.o layer3.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o
tabinit.o tabinit.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o
interface.o interface.c
gcc -pipe -fPIC -Wall -Wstrict-prototypes -Wmissing-prototypes
-Wmissing-declarations -D_REENTRANT -D_GNU_SOURCE -O6-c -o
format_mp3.o format_mp3.c
In file included from /usr/include/asterisk/logger.h:28,
from /usr/include/asterisk/lock.h:83,
from format_mp3.c:20:
/usr/include/asterisk/compat.h:20: error: parse error before __extension__
/usr/include/asterisk/compat.h:20: error: parse error before '' token
In file included from /usr/include/asterisk/utils.h:36,
from /usr/include/asterisk/cdr.h:48,
from /usr/include/asterisk/channel.h:113,
from format_mp3.c:21:
/usr/include/asterisk/strings.h:264: error: parse error before
__extension__
/usr/include/asterisk/strings.h:264: error: parse error before ';' token
/usr/include/asterisk/strings.h:264: error: `__len' undeclared here (not
in a function)
/usr/include/asterisk/strings.h:264: error: initializer element is not
constant
/usr/include/asterisk/strings.h:264: error: parse error before if
/usr/include/asterisk/strings.h:264: error: redefinition of `__retval'
/usr/include/asterisk/strings.h:264: error: `__retval' previously
defined here
/usr/include/asterisk/strings.h:264: error: parse error before const
/usr/include/asterisk/strings.h:264: error: parse error before '}' token
/usr/include/asterisk/strings.h:280: error: conflicting types for `strtoq'
/usr/include/stdlib.h:346: error: previous declaration of `strtoq'
format_mp3.c:46: error: redefinition of `struct ast_filestream'
format_mp3.c:325: warning: function declaration isn't a prototype
format_mp3.c: In function `load_module':
format_mp3.c:336: warning: passing arg 1 of `ast_format_register' from
incompatible pointer type
format_mp3.c:336: error: too many arguments to function
`ast_format_register'
format_mp3.c: At top level:
format_mp3.c:342: warning: function declaration isn't a prototype
format_mp3.c:347: warning: function declaration isn't a prototype
format_mp3.c:359: warning: function declaration isn't a prototype
format_mp3.c:365: warning: function declaration isn't a prototype
{standard input}: Assembler messages:
{standard input}:49: Error: symbol `__retval' is already defined
make[1]: *** [format_mp3.o] Error 1
make[1]: Leaving directory
`/usr/local/src/svn-versions/asterisk-addons/format_mp3'
make: *** [format_mp3/format_mp3.so] Error 2
2. /usr/sbin/safe_asterisk: line 41: 25091 Segmentation fault (core
dumped) asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}
___
--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] dtmf tones
If I call PSTN number a, than I can call the extension number, while when I call PSTN phone number b the tones are ignored. If I call PSTN PSTN directly the extension number can be dialed. How can I improve that? bye Ronald Wiplinger ___ --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] Softphone ready to go installed on USB flash drive
How can I install a softphone on my USB flash drive like Xlite and have it ready to go when I plug it in at any Windows XP computer? (Same for a Linux softphone, both on one USB flash drive). bye Ronald Wiplinger ___ --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] Softphone ready to go installed on USB flash drive
Bruce Reeves wrote: I do this with the windows version of idefisk from Asteriskguru.com http://Asteriskguru.com. The configuration is stored in the dir with the program and dll. I have actually configured it and emailed it to users. There is no installer and a simple shortcut or autoplay menu should take care of the rest. It is a nice phone, but it is IAX. I would like to use a SIP phone. The reason for that is that there is no IAX server for the mass, but openSER bye Ronald Wiplinger On 5/1/06, *Time Bandit* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: How can I install a softphone on my USB flash drive like Xlite and have it ready to go when I plug it in at any Windows XP computer? (Same for a Linux softphone, both on one USB flash drive). I believe Dan's softphone is suitable for this. See http://www.laser.com/dante/diax/diax.html Actually, I should do that with my softphone instead of using the registry :( hth ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --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] Compare to Skype
[EMAIL PROTECTED] wrote: Original Message Skype uses iLBC codec, which has great jitter compensation. IIRC, the newer SIP channels of * are supposed to have the same capabilities, but I have not tested. I really do not like Skype (prefer FWD), but I must say, over satellite, etc, they provide quality.. All about the codec in this case.. Errr...no...this is wrong. Skype uses ISAC from Global IP Sound. iLBC is something different see http://www.globalipsound.com/solutions/solutions_Codecs.php One of the reasons Skype sounds good is that its a closed system and so can leverage a wideband codec. Instead of the normal 8khz sample rate it uses 16khz. That makes for clearer sound. Since ISAC is a proprietary relative of iLBC its jitter compensation is also very good. My understanding is that Asterisk cannot presently use any wideband codecs as it is hard coded to the 8khz sample rate at its core. Adapting Asterisk to wideband capability has been discussed but will be a huge amount of work. Further, only if you know that the calls will stay wideband end-to-end will the benefits of wideband be apparent. That means no PSTN segments. Michael Graves [EMAIL PROTECTED] Sadly to say, but users do not care about the why, they only care about the quality! and they simple ask to fix it! I hope there is soon a solution, otherwise, we have to skip all our effort and just use skype! And I would hate to see that. I just lost 20 US$ to Ebay - the newly parent company of skype, for a not received parcel, but the rules says, below 25 US$ there is no guarantee that you get anything bye Ronald Wiplinger ___ --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] Compare to Skype
[EMAIL PROTECTED] wrote: What would be ideal is the introduction of an open source wideband codec implementation. Then you could see it adopted into SIP end points and used with SER realtively quickly. Sadly, an Asterisk implmentation would lag a little behind due to the amount of work required in an implementation that processed the streams to bridge into the TDM/PSTN world. It would be greatbut don't hold your breath. For now there are Skype bridges like PSWG and Uplink that interface Skype to SIP. These are simplistic but sometimes workable. Does anyone here have experience with Uplink? I tried PSGW and gave up eventually. I am also a supporter of PSGW although on my AMD it never worked. Now it is getting obsolete at all, since I switch next week finally to a Linux desktop I never heard about Uplink, where is it, does it work? From the uplink web: System Requirements * Windows 98/2000/Me/XP/2003 sigh bye Ronald Wiplinger ___ --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
