Re: [Asterisk-Users] MSN MESSENGER 4.7 with Asterisk -SOMEONE HELP HERE PLEASE!-
On Wed, 2003-12-03 at 05:00, Carlos Arnt wrote: Hi all, I just trying to test MSN 4.7 that has SIP. Because with him i can use a video and voice transmission and * . But when i try to call someone using the DIALPAD of MSN, when i insert any digit into * the numbers appears twice !! like this. channel 456 appears in asterisk 445566 How can i fix this ? dtmfmode=inband; Choices are inband, rfc2833, or info have you played around with this setting in sip.conf? If not, try :) roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] remove me
rm -rf reggie On Tue, 2003-12-02 at 22:35, reginald huey wrote: Please Remove me from the list Reggie Reginald Huey __ Do you Yahoo!? Free Pop-Up Blocker - Get it now ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IAX port numbers?
I thought the origin of outbound connections were random, but the destination was always the port of the service you're attempting to acquire? That's the case with TCP. Not UDP. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk with Voicetronix OpenLine4 card
hi there, i've been able to successfully run asterisk with the Voicetronix OpenLine4 card, it can accept calls and function normally. The only problem I'm experiencing so far is getting the card to outdial to a third party. What I'm trying to achieve is basically call bridging, where the caller dials in to asterisk, some IVR plays and then attempts to perform a transfer to a third party, and once the outbound call is connected both legs are bridged. I've seen some dialplans out there that use the normal Dial application. in my dialplan i've used various different methods: exten = s,5,Dial() exten = s,5,Dial(vpb/) exten = s,5,Dial(vpb/1-3/) (the third one is assuming it means board 1 line 3) in the log file, the following error is recorded each time the outbound dial is attempted: File app_dial.c, Line 499 (dial_exec): Unable to create channel of type 'vpb' As far as the vpb.conf file goes, my attempts include: 1) Setting channels 1 and 2 as FXO, channels 3 and 4 as immediate 2) Setting channels 1 and 2 as FXO, channels 3 and 4 as dialtone 3) Setting channels 1-4 as FXO i may have something mixed up here, has anyone had any success with this? note that i'm not using the OpenSwitch card, it's the OpenLine. Thanks, Faiz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to set the gatekeeper? help me pls.
Hello every one, I have got a H323 gatekeeper for testing. The informations are something like this: account code: test01 gk ip address:192.168.10.12 I don't know how to set it in the h323.conf or oh323.conf, I have tried it for almost one day but I always got the error. Help me please. Regards.
[Asterisk-Users] BOOM! Crash when trying to use SIPDtmfMode on an outgoing call!
All,
Here's a cool one.. I was attempting to call a retarded conferencing
service, and was having problems with it picking up my DTMF.. after trying
all the settings my Sipura SPA2000 offers, I found inband actually works..
unfortunately, I can't get anything else to pick up my inband DTMF
(including asterisk's builtin voicemail! It just times out and says I never
entered a login!). So, I did some digging around, and figured I might try
SIPDtmfMode to change my DTMF mode when I'm calling out.. that resulted in a
prompt crash, and the info included below out of gdb. Is it me? Am I
misunderstanding the appropriate use of SIPDtmfMode? If so, that's fine,
just bonk me on the head with a yellow pages book or something.. Also.. how
can I change the DTMF timing? I think the SIP INFO dtmf I'm sending is too
brief for the conferencing service.. is there any way I can change the
timings? Finally, how come * voicemail won't recognize my inband digits?
I'm using ulaw from my * box to my Sipura on a local 100megabit switched
lan.
Thanks!
Pat
-- extensions.conf --
[toll-trunks]
exten = _1NXXNXX,1,SIPDtmfMode(inband)
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
-- gdb crash --
[New Thread 278546 (LWP 4192)]
-- Executing SIPDtmfMode(SIP/1000-9732, inband) in new stack
-- Executing Dial(SIP/1000-9732, IAX2/[EMAIL PROTECTED]/18882245408)
in new stack
-- Called [EMAIL PROTECTED]/18882245408
-- Call accepted by 66.234.228.132 (format ULAW)
-- Format for call is ULAW
-- IAX2[voicepulse]/3 stopped sounds
Program received signal SIGSEGV, Segmentation fault.
[Switching to Thread 278546 (LWP 4192)]
0x0808c75d in __ast_dsp_silence (dsp=0x0, s=0xbd7fe774, len=160,
totalsilence=0x0) at dsp.c:969
969 if (accum dsp-threshold) {
(gdb) Quit
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Re: [Asterisk-Users] Proper use of echotraining=yes
Brian West wrote: If you have echo on the X100P's Mark setup chan_zap to pretrain the echo can, but it had a few issues until today which Mark nailed down the bug that caused the DTMF to be unreliable. Ok here is how you would do it: Thank you! http://www.voip-info.org/tiki-index.php?page=Asterisk+x100p+echotraining /O ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Unable to check my voice mails
Hi All,
I am a newb to *. Just configured and lucklily it
worked on the first attempt. My setup is on Rh 7.2 and i d/led the build on Dec
1st. i hv installed X-Lite on two of my laptops. i am unable to check my
voicemails. when ever i enter my password * prompts me again and again to enter
the password. Log shows password error. No clue on whats the the
issue.
the following are my config files.
[general]port = 5060bindaddr =
0.0.0.0context = bogon-calls
[2000]
type=friendusername=2000secret=qweqweauth=md5host=dynamiccontext=from-sipdtmfmode=inbandmailbox=2000
[2001]
type=friendusername=2001secret=asdasdauth=md5host=dynamiccontext=from-sipdtmfmode=inbandmailbox=2001
extension.conf
[general]
static=yeswriteprotect=yes
[bogon-calls]
exten = _.,1,Congestion
[from-sip];2000exten = 2000,1,Dial(SIP/2000,20)exten =
2000,2,Voicemail(u2000)exten = 2000,102,Voicemail(b2000)exten =
2000,103,Hangup
;2001exten = 2001,1,Dial(SIP/2001,20)exten =
2001,2,Voicemail(u2001)exten = 2001,102,Voicemail(b2001)exten =
2001,103,Hangup
;2999 Voice mailexten = 2999,1,VoicemailMain(${CALLERIDNUM})
voicemail.conf
[general]
;format=wavformat=gsm
[local]
2000 = 1234,Balaji NJL,[EMAIL PROTECTED]
2001 = 5678,Ojasvi Sinha,[EMAIL PROTECTED]
Any idea whats the issue. any help appreciated.
thanks a lot,
-Balaji
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Re: [Asterisk-Users] SMS over PRI/E1?
On Wed, Dec 03, 2003 at 08:30:34AM +0100, Roy Sigurd Karlsbakk wrote: hi all I spoke to this guy the other day, working with Cisco's VoIP system. He told me they were using a PRI/E1 to transport SMS, and could even do so from their phones. May this be possible with asterisk? I have an E100P in my primary asterisk server connected to a E1/PRI. This is a carrier service. I seem to remember there are some ETSI standards for SMS over ISDN transport/gatewaying. As far as I understand (which is not very far, I admit, right now!), Asterisk doesn't support Q931 user-to-user info transmission (although there is support in libpri). It should be possible, however, to modify chan_zap and add a SendText application (and/or modify the Dial app) to handle this. -- Nicolas Bougues Axialys Interactive ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unable to make it work with MSN Messenger
Hi All,
I am a newb to *. My setup is on Rh 7.2 and i
d/led the build on Dec 1st. i hv installed X-Lite on two of my laptops. i am
able to make calls between X-Lite (Ext 2000 and 2001) . i configured MSN as ext
2002. When ever i am trying to log on using MSN it rejects my password.
the following are my config files.
[general]port = 5060bindaddr =
0.0.0.0context = bogon-calls
[2002]
type=friendhost=dynamicinsecure=yesdtmfmode=inbandcontext=from-sipmailbox=2002
[2000]
type=friendusername=2000secret=qweqweauth=md5host=dynamiccontext=from-sipdtmfmode=inbandmailbox=2000
[2001]
type=friendusername=2001secret=asdasdauth=md5host=dynamiccontext=from-sipdtmfmode=inbandmailbox=2001
extension.conf
[general]
static=yeswriteprotect=yes
[bogon-calls]
exten = _.,1,Congestion
[from-sip];2000exten = 2000,1,Dial(SIP/2000,20)exten =
2000,2,Voicemail(u2000)exten = 2000,102,Voicemail(b2000)exten =
2000,103,Hangup
;2001exten = 2001,1,Dial(SIP/2001,20)exten =
2001,2,Voicemail(u2001)exten = 2001,102,Voicemail(b2001)exten =
2001,103,Hangup
;2002exten = 2002,1,Dial(SIP/2002,20)exten =
2002,2,Voicemail(u2002)exten = 2002,102,Voicemail(b2002)exten =
2002,103,Hangup
;2999 Voice mailexten = 2999,1,VoicemailMain(${CALLERIDNUM})
voicemail.conf
[general]
;format=wavformat=gsm
[local]
2000 = 1234,Balaji NJL,[EMAIL PROTECTED]2001 = 5678,Ojasvi
Sinha,[EMAIL PROTECTED]
2002 = 1234, Balaji NJL,[EMAIL PROTECTED]
Any idea whats the issue. any help appreciated.
thanks,
-Balaji
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Re: [Asterisk-Users] How to set the gatekeeper? help me pls.
:) h323.conf is just a bit strange (there is no simple/clear alias options as in the oh323.conf) But it's a good idea to read Readme and h323.conf.sample ... here is one h323.conf [general] port = 1720 bindaddr = 0.0.0.0 tos=lowdelay dtmfmode=rfc2833 context = your-unautorized-context noFastStart = yes noH245Tunneling = yes gatekeeper = 192.168.10.12 AllowGKRouted = yes disallow = all allow=gsm allow=ulaw [test01] type=h323 host = 192.168.10.12 context = your-incomming-context Lubo [EMAIL PROTECTED] wrote: Hello every one, I have got a H323 gatekeeper for testing. The informations are something like this: account code: test01 gk ip address:192.168.10.12 I don't know how to set it in the h323.conf or oh323.conf, I have tried it for almost one day but I always got the error. Help me please. Regards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk behind NAT How to do it.
On Tue, 2003-12-02 at 15:55, Arnold Ligtvoet wrote: Hi Leif, I tried the patch. Installed it exactly as described per your email. Thought that you might be interested that it works for me as well. Like a charm, I can finally call FWD numbers like 10001 and 612 (speaking clock demo). BTW: For anybody wanting to install this, if your version of chan_sip.c is older than the one described, first use 'cvs update -C asterisk/channels/chan_sip.c'. Awesome! Have you tried the newer patch / diff for 1.259 (which as of right now is the newest chan_sip file). If you goto bugs.digium.com and login anonymously and jump to bug 104, then you can get the newest patch. Same instructions as before. I just updated it to test the new sip.conf structure which is externip= localnet= localmask= Still working great for me here! BTW! Can you login to the bug tracker and post a comment ? Thanks! -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unable to make it work with MSN Messenger
Prolly change the auth= to plaintext...
On Wed, 2003-12-03 at 10:07, Balaji NJL wrote:
Hi All,
I am a newb to *. My setup is on Rh 7.2 and i d/led the build on Dec
1st. i hv installed X-Lite on two of my laptops. i am able to make
calls between X-Lite (Ext 2000 and 2001) . i configured MSN as ext
2002. When ever i am trying to log on using MSN it rejects my
password.
the following are my config files.
[general]
port = 5060
bindaddr = 0.0.0.0
context = bogon-calls
[2002]
type=friend
host=dynamic
insecure=yes
dtmfmode=inband
context=from-sip
mailbox=2002
[2000]
type=friend
username=2000
secret=qweqwe
auth=md5
host=dynamic
context=from-sip
dtmfmode=inband
mailbox=2000
[2001]
type=friend
username=2001
secret=asdasd
auth=md5
host=dynamic
context=from-sip
dtmfmode=inband
mailbox=2001
extension.conf
[general]
static=yes
writeprotect=yes
[bogon-calls]
exten = _.,1,Congestion
[from-sip]
;2000
exten = 2000,1,Dial(SIP/2000,20)
exten = 2000,2,Voicemail(u2000)
exten = 2000,102,Voicemail(b2000)
exten = 2000,103,Hangup
;2001
exten = 2001,1,Dial(SIP/2001,20)
exten = 2001,2,Voicemail(u2001)
exten = 2001,102,Voicemail(b2001)
exten = 2001,103,Hangup
;2002
exten = 2002,1,Dial(SIP/2002,20)
exten = 2002,2,Voicemail(u2002)
exten = 2002,102,Voicemail(b2002)
exten = 2002,103,Hangup
;2999 Voice mail
exten = 2999,1,VoicemailMain(${CALLERIDNUM})
voicemail.conf
[general]
;format=wav
format=gsm
[local]
2000 = 1234,Balaji NJL,[EMAIL PROTECTED]
2001 = 5678,Ojasvi Sinha,[EMAIL PROTECTED]
2002 = 1234, Balaji NJL,[EMAIL PROTECTED]
Any idea whats the issue. any help appreciated.
thanks,
-Balaji
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Re: [Asterisk-Users] How to set the gatekeeper? help me pls.
Hi,Lubo, Thank you very much for your reply. I want to use the gatekeeper for outbound call, but I really don't know how to use it in the extensions.conf ,I think there are something diffrence between the chan_h323 channel and the chan_oh323 channel. A little example of extensions.conf would be appreciated. (Sorry for my poor English). Regards. frank - Original Message - From: Lubomir Christov [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 1:18 AM Subject: Re: [Asterisk-Users] How to set the gatekeeper? help me pls. :) h323.conf is just a bit strange (there is no simple/clear alias options as in the oh323.conf) But it's a good idea to read Readme and h323.conf.sample ... here is one h323.conf [general] port = 1720 bindaddr = 0.0.0.0 tos=lowdelay dtmfmode=rfc2833 context = your-unautorized-context noFastStart = yes noH245Tunneling = yes gatekeeper = 192.168.10.12 AllowGKRouted = yes disallow = all allow=gsm allow=ulaw [test01] type=h323 host = 192.168.10.12 context = your-incomming-context Lubo [EMAIL PROTECTED] wrote: Hello every one, I have got a H323 gatekeeper for testing. The informations are something like this: account code: test01 gk ip address:192.168.10.12 I don't know how to set it in the h323.conf or oh323.conf, I have tried it for almost one day but I always got the error. Help me please. Regards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to set the gatekeeper? help me pls.
Take a look here, I hope it will help you :) http://www.voip-info.org/tiki-index.php?page=Asterisk http://sprackett.com/asterisk/conf/ http://www.loligo.com/asterisk/current/ http://www.fnords.org/~eric/asterisk/ Lubo [EMAIL PROTECTED] wrote: Hi,Lubo, Thank you very much for your reply. I want to use the gatekeeper for outbound call, but I really don't know how to use it in the extensions.conf ,I think there are something diffrence between the chan_h323 channel and the chan_oh323 channel. A little example of extensions.conf would be appreciated. (Sorry for my poor English). Regards. frank - Original Message - From: Lubomir Christov [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 1:18 AM Subject: Re: [Asterisk-Users] How to set the gatekeeper? help me pls. :) h323.conf is just a bit strange (there is no simple/clear alias options as in the oh323.conf) But it's a good idea to read Readme and h323.conf.sample ... here is one h323.conf [general] port = 1720 bindaddr = 0.0.0.0 tos=lowdelay dtmfmode=rfc2833 context = your-unautorized-context noFastStart = yes noH245Tunneling = yes gatekeeper = 192.168.10.12 AllowGKRouted = yes disallow = all allow=gsm allow=ulaw [test01] type=h323 host = 192.168.10.12 context = your-incomming-context Lubo [EMAIL PROTECTED] wrote: Hello every one, I have got a H323 gatekeeper for testing. The informations are something like this: account code: test01 gk ip address:192.168.10.12 I don't know how to set it in the h323.conf or oh323.conf, I have tried it for almost one day but I always got the error. Help me please. Regards. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_queue different behaviour
Hello, is there a way to make app queue to first try to ring the agents and start music on hold only when they are all talking to other callers? So when the caller calls, and there are free operators he hears ringing, and * is not picking up until call is answere, or specified timeout. And if the caller calls , and there are no free operators , some message like please wait for next avalable operator and them the music on hold start. thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfer via # on Grandstream not always working
Hello, After a while the transfer on grandstream stops working, only the reboot fixes the problem. It also seems that it may be the phone I`m trying to transfer _to_ also sometimes requires a reboot. After that it starts working. I`m using RFC2833 signlaing between phones and *. Does anybody see this happening also? Thanks -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Double 3's Problem - H323 . Very weird
Hi Folks, I have a X100P with Asterisk running connection to a non-asterisk device in the other side. It was working perfectly with H323(chan_h323)+ G.729 in the last weeks. Suddenly, I am getting double 3's in the other side's POTS. Any number is not repeated, only the 3 is being repeated. I guess I'm sending it correctly and no change was done in the dial plans recently. Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bug in MGCP using host=dynamic
Hi, there is a bug in chan_mgcp.c which shows up if you have more than one MGCP gateway configured with host=dynamic. The problem is in the routine find_subchannel when a MGCP response is received. When the response is handled find_subchannel is called with name = NULL and sin = address. This cause the find_subchannel routine to alter the address of all gateways up to the one the response is addressed to. This will cause asterisk to send MGCP messages to the wrong address for the altered gateways. I have changed my call of find_subchannel so that the sin parameter is also set to NULL. This works for me but I'm not shure this is the correct solution. /Bertil -- Bertil Engelholm [EMAIL PROTECTED] i3 micro technology ab ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue different behaviour
Anton, Take a look at the latest version of the patch in: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=214 Good luck! Michiel Anton Yurchenko wrote: Hello, is there a way to make app queue to first try to ring the agents and start music on hold only when they are all talking to other callers? So when the caller calls, and there are free operators he hears ringing, and * is not picking up until call is answere, or specified timeout. And if the caller calls , and there are no free operators , some message like please wait for next avalable operator and them the music on hold start. thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More voicemodem
Hi, I got this setup. analog phone (ext7) --- analog pbx - (ext 6 analog) voicemodem (ext 3 asterisk) ttyS0/asterisk sipphones q1: I got the voicemodem to work, but oneway only. I can talk from my analog phone, to my sipphone, but not the other way ? I know it only suppose to works in half duplex, but nothing come TO the phone. q2: From SIPphone I dial 3+ext on my analog pbx - it works :) From analog phone I dial my voicemodem (ext 6) asterisk answer and it automatic forward to one specific sipphone, how do I get a new 'dialtone' from asterisk so I can dial ANY number in asterisk ? Hope to get some hints. (I'm really new to asterisk so an exsample would be good) /HHA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] app_queue different behaviour
Michiel Betel wrote: Anton, Take a look at the latest version of the patch in: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=214 It does adds an abiliti to make an announcment to a user once they are in queue, but no this behaviour with cheking if all operators are busy or not. Thank you Good luck! Michiel Anton Yurchenko wrote: Hello, is there a way to make app queue to first try to ring the agents and start music on hold only when they are all talking to other callers? So when the caller calls, and there are free operators he hears ringing, and * is not picking up until call is answere, or specified timeout. And if the caller calls , and there are no free operators , some message like please wait for next avalable operator and them the music on hold start. thanks ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Anton Yurchenko[EMAIL PROTECTED] Digital Generation ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] oh323 calling party number
How do I get asterisk to populate the Calling Party Number field in an H.323 call? I have asterisk configured to accept a SIP call and connect it to an H.323 IVR system. The call goes through, but the caller id is put in the Display field rather than the Calling Party Number field. -Original Message- From: Skuse, Phil [mailto:[EMAIL PROTECTED] Sent: 01 December 2003 17:23 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] How do I get caller's number in oh323 ? We have an h.323 based IVR platform. When we make a call to it using an h.323 phone, it can see the callers number (ANI), but when we make a call to it via asterisk, the call goes through OK, but we don't get the number. How can I make this work? h323.conf === [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=alaw dtmfmode=inband [ivr] type=h323 context=default extensions.conf === exten = 602,1,Dial,h323/[EMAIL PROTECTED] exten = 602,2,HangUp Phil Skuse [EMAIL PROTECTED] *** UNIX System Administrator. NIC Handle: MBJEJPIEUI Vicorp UK Limited: The Telephony Engine Company. Tel +44 (0)1753 660523 http://www.vicorp.com *** ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Kerio SIPPS problems -please help!!!
Please help!!! Anyone have tried * with kerio SIPPS softphone? It registers ok with *, but I get missing sdp body message when dialing any extension. Thanks. Hector-. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] New Multilingual DIAX (0.9.5) available for download
Hi all, The new multilingual version of DIAX (0.9.5) is now available for: - English - Romanian - German - Dutch - Italian - French - Spanish - Portuguese at the following locations: http://www.laser.com/dante http://www.geocities.com/tdanro What's new in 0.9.5 : - double support(IAX(1)/IAX2) - Multilanguage support: English, Romanian, German, Dutch, Italian, French, Spanish, Portuguese(for the moment). - support for IAXTEL direct number dial (if registered at iaxtel.com) - automatically load default audio device(Sound Mapper) if the one from the config file is invalid (do not ask) - the fullmesage in the Statusbar is now in tooltip too(when dragging the mouse over), even if it is erased from the statusbar - can use just the executable with the old config/calls/phonebook files (from 0.9.4), the new config parameters are generated automatically using default values - wider buttons for functions and memories, to accommodate different languages - enable/disable statusbar display - automatically scroll long status messages in the statusbar - color codes for status messages (red-error, green-status/notice, blue-iaxmsg, black - others) I want to take this opportunity to thanks to the following people for the translation effort: Peer Oliver Schmidt (German) - a lot of help to discover hidden bugs too..;-) Florian Overkamp (Dutch) Nicolas Bougues (French) Nicolas Gudino Rafael Gonzalez Lomeña (Spanish) Emanuele Pucciarelli (Italian) Nuno Cruz Isamar Maia (Portuguese) The included help file is the old 0.9.4 English version, but as they are no major changes in the operation mode, it still can be used till the new multilingual version will be available. The specific help file for each language will pe posted as a separate file, to decrease the quantity of data during download. Please send me your feedback in order to help me improove the application I am open to integrate more languages if they are interested people in translation. Before asking for more features, please check the Wish list from my web page to see if it something new. Thank you and best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk with Voicetronix OpenLine4 card
Try the following:
vpb.conf:
[interfaces]
echocancel = on
board = 1
context = default
mode = fxo
channel = 3
extensions.conf:
exten = _9.,1,Dial(vpb/1-3/${EXTEN:1})
exten = _9.,2,Congestion
Hope help you
Jorge
Ahmad Faiz wrote:
hi there,
i've been able to successfully run asterisk with the Voicetronix OpenLine4
card, it can accept calls and function normally. The only problem I'm
experiencing so far is getting the card to outdial to a third party.
What I'm trying to achieve is basically call bridging, where the caller
dials in to asterisk, some IVR plays and then attempts to perform a
transfer to a third party, and once the outbound call is connected both
legs are bridged.
I've seen some dialplans out there that use the normal Dial application. in
my dialplan i've used various different methods:
exten = s,5,Dial()
exten = s,5,Dial(vpb/)
exten = s,5,Dial(vpb/1-3/)
(the third one is assuming it means board 1 line 3)
in the log file, the following error is recorded each time the outbound dial
is attempted:
File app_dial.c, Line 499 (dial_exec): Unable to create channel of type
'vpb'
As far as the vpb.conf file goes, my attempts include:
1) Setting channels 1 and 2 as FXO, channels 3 and 4 as immediate
2) Setting channels 1 and 2 as FXO, channels 3 and 4 as dialtone
3) Setting channels 1-4 as FXO
i may have something mixed up here, has anyone had any success with this?
note that i'm not using the OpenSwitch card, it's the OpenLine.
Thanks,
Faiz
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Re: [Asterisk-Users] New Multilingual DIAX (0.9.5) available for download
- Original Message - From: Dan [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 9:04 AM Subject: [Asterisk-Users] New Multilingual DIAX (0.9.5) available for download Hi all, The new multilingual version of DIAX (0.9.5) is now available for: - English - Romanian - German - Dutch - Italian - French - Spanish - Portuguese at the following locations: http://www.laser.com/dante http://www.geocities.com/tdanro I don't know if this was intentional or not, but my newest download defaulted to Romanian? - Andrew Thompson Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] un-subscribe
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This e-mail and attachments, if any, may contain confidential and/or
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-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jorge Mendoza
Sent: Wednesday, December 03, 2003 9:09 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk with Voicetronix OpenLine4 card
Try the following:
vpb.conf:
[interfaces]
echocancel = on
board = 1
context = default
mode = fxo
channel = 3
extensions.conf:
exten = _9.,1,Dial(vpb/1-3/${EXTEN:1})
exten = _9.,2,Congestion
Hope help you
Jorge
Ahmad Faiz wrote:
hi there,
i've been able to successfully run asterisk with the Voicetronix OpenLine4
card, it can accept calls and function normally. The only problem I'm
experiencing so far is getting the card to outdial to a third party.
What I'm trying to achieve is basically call bridging, where the caller
dials in to asterisk, some IVR plays and then attempts to perform a
transfer to a third party, and once the outbound call is connected both
legs are bridged.
I've seen some dialplans out there that use the normal Dial application. in
my dialplan i've used various different methods:
exten = s,5,Dial()
exten = s,5,Dial(vpb/)
exten = s,5,Dial(vpb/1-3/)
(the third one is assuming it means board 1 line 3)
in the log file, the following error is recorded each time the outbound
dial
is attempted:
File app_dial.c, Line 499 (dial_exec): Unable to create channel of type
'vpb'
As far as the vpb.conf file goes, my attempts include:
1) Setting channels 1 and 2 as FXO, channels 3 and 4 as immediate
2) Setting channels 1 and 2 as FXO, channels 3 and 4 as dialtone
3) Setting channels 1-4 as FXO
i may have something mixed up here, has anyone had any success with this?
note that i'm not using the OpenSwitch card, it's the OpenLine.
Thanks,
Faiz
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Re: [Asterisk-Users] BUG in New Multilingual DIAX (0.9.5) available for download
Bug: In the Phonebook, (I've only tried it this way, so far) if you delete an item, then choose a different item from the dropdown, the delete button doesn't work anymore. I was deleting the default entries and decided to not delte the DIGI entry. When I chose the next one after that, it wouldn't delete without closing and reopening the window. Can someone else test/confirm? - Andrew Thompson Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Multilingual DIAX (0.9.5) available for download
Hi, - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 4:29 PM Subject: Re: [Asterisk-Users] New Multilingual DIAX (0.9.5) available for download - Original Message - From: Dan [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 9:04 AM Subject: [Asterisk-Users] New Multilingual DIAX (0.9.5) available for download Hi all, The new multilingual version of DIAX (0.9.5) is now available for: - English - Romanian - German - Dutch - Italian - French - Spanish - Portuguese at the following locations: http://www.laser.com/dante http://www.geocities.com/tdanro I don't know if this was intentional or not, but my newest download defaulted to Romanian? Sorry for that. It was not intended to be like that. You can switch to any language you want using CTRL+ first letter of the language. For example, use CTRL+e to switch to English. Then it will be saved in the config file. If you use the old config file from 0.9.4 then the default at first start will be English Sorry for the inconvenience. Best regards, Dan - Andrew Thompson Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] BUG in New Multilingual DIAX (0.9.5) available for download
Hi, - Original Message - From: Andrew Thompson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 4:33 PM Subject: Re: [Asterisk-Users] BUG in New Multilingual DIAX (0.9.5) available for download Bug: In the Phonebook, (I've only tried it this way, so far) if you delete an item, then choose a different item from the dropdown, the delete button doesn't work anymore. I was deleting the default entries and decided to not delte the DIGI entry. When I chose the next one after that, it wouldn't delete without closing and reopening the window. Can someone else test/confirm? Yup!.. You're right... I'll solve that... BR, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Multilingual DIAX (0.9.5) available for download
I don't know if this was intentional or not, but my newest download defaulted to Romanian? Sorry for that. It was not intended to be like that. You can switch to any language you want using CTRL+ first letter of the language. For example, use CTRL+e to switch to English. Then it will be saved in the config file. If you use the old config file from 0.9.4 then the default at first start will be English Sorry for the inconvenience. I now see that your email is from a .ro domain. That kicks me back into global world mode and reminds me that not everyone speaks English as a first language. It's not a problem, I just was being snobby about my preference, I guess. No hard feelings, just a suprise that I couldn't read the words in the menu bar! Best regards, Dan Enjoy... - Andrew Thompson Your eyes are weary from staring at the CRT. You feel sleepy. Notice how restful it is to watch the cursor blink. Close your eyes. The opinions stated above are yours. You cannot imagine why you ever felt otherwise. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm, etc.)
On Tue, 2 Dec 2003 21:11:31 -0500 (EST), firedude wrote The new versions of iaxcomm and DIAX are both now using the iax2 protocol. So in order to receive incoming calls on either of them in your extensions.conf file change IAX/clientname to IAX2clientname. Then you should be able to receive incoming calls on either iaxcomm or DIAX. Also there is a mailing list for the iaxclient library. It's [EMAIL PROTECTED] Hope this helps. AJ or: exten=1500,1,Dial(IAX/clientIAX2/client,30) my 0.02pln grzegorz nosek ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel i2004
I have 40 of these phones. they dont run SIP or any usable protocol they can hook up to a Nortel box and proxy SIP out of that box, but they wont run SIP native if im wrong please let me know... I'd relly like to use my 40 phones that are collecting dust Dave [EMAIL PROTECTED] 12/2/2003 7:00:05 PM Is anyone successfully using this phone with Asterisk? There is a lot mentioned about CISCO but nothing about Nortel... Alex. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco IAD with MGCP
I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently noted the following message at * logs: Nov 21 13:06:58 NOTICE[8201]: File chan_mgcp.c, Line 1099 (find_subchannel): Gateway '192.168.65.200' (and thus its endpoint '*') does not exist But as you can see at mgcp.conf the gateway is defined ! Attached are the configs: Cisco -- ! version 12.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname 192.168.65.200 ! logging queue-limit 100 enable secret enable password ! ip subnet-zero ! ! no ip domain lookup ! isdn switch-type primary-net5 ! ! voice call carrier capacity active ! voice service pots ! voice service voip ! voice class codec 10 codec preference 1 gsmfr codec preference 2 g711alaw ! ! ! ! ! ! ! no voice hpi capture buffer no voice hpi capture destination ! ! mta receive maximum-recipients 0 ! ! controller T1 1/0 shutdown framing esf linecode b8zs ! ! ! interface FastEthernet0/0 ip address 192.168.65.200 255.255.255.0 duplex auto speed auto ! interface FastEthernet0/1 no ip address shutdown duplex auto speed auto ! ip http server ip classless ip route 0.0.0.0 0.0.0.0 192.168.65.1 ! ! ! dialer-list 1 protocol ip permit ! ! call rsvp-sync ! voice-port 2/0 no vad timing hookflash-in 750 ! voice-port 2/1 no vad timing hookflash-in 750 ! voice-port 2/2 no vad timing hookflash-in 750 ! voice-port 2/3 no vad timing hookflash-in 750 ! voice-port 2/4 ! voice-port 2/5 ! voice-port 2/6 ! voice-port 2/7 ! voice-port 2/8 ! voice-port 2/9 ! voice-port 2/10 ! voice-port 2/11 ! voice-port 2/12 ! voice-port 2/13 ! voice-port 2/14 ! voice-port 2/15 ! mgcp mgcp call-agent 192.168.65.100 service-type mgcp version 1.0 mgcp package-capability rtp-package mgcp default-package dtmf-package no mgcp timer receive-rtcp no mgcp validate domain-name mgcp bind control source-interface FastEthernet0/0 ! mgcp profile default ! dial-peer cor custom ! ! ! dial-peer voice 2 pots application mgcpapp port 2/1 ! dial-peer voice 3 pots application mgcpapp port 2/2 ! dial-peer voice 4 pots application mgcpapp port 2/3 ! dial-peer voice 1 pots application mgcpapp port 2/0 ! ! line con 0 line aux 0 line vty 0 4 password login ! end mgcp.conf -- ; ; MGCP Configuration for Asterisk ; [general] port = 2727 bindaddr = 0.0.0.0 [192.168.65.200] host = 192.168.65.200 context = local line = aaln/S2/0 line = aaln/S2/1 line = aaln/S2/2 line = aaln/S2/3 -- Juanjo sin .sig ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PREPAID APPLECATION
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of PJ Welsh Sent: Tuesday, December 02, 2003 9:39 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] PREPAID APPLECATION It is a shame that within a couple of hours they can tell you to remove helpfull documentation, but not (seemingly) help answer questions regarding there Cisco stuff on this list. I think Cisco must have their priorities mixed up! Just my opinion... which also means I won't support a company like that... so I won't buy their products... No, Cisco has their priorities just fine. Companies are in business to make money. Not to give things away. When I have a problem with a piece of Cisco equipment, it is answered promptly and accurately, nearly 100% of the time. I have SmartNet on all of the devices for which I expect this service. Cisco documents are property of Cisco. Many of them require a CCO account to access, and there are varying levels of CCO access. Many newer technologies are initially available to all with a CCO login until their maturity and complexity reaches a point where Cisco makes a specialty for them, at which time those documents and new ones on the subject are sometimes no longer available to just anyone with a CCO login. Cisco also maintains and updates their documents on an as-needed basis. Storing copes of their documents on your own web site for public use defeats their ability to do all of these things. And if you want to argue that much of this is done just to charge you more money, you are correct. Cisco is an enterprise infrastructure company. Not a home user/home office/small office outfit where you can call up and talk to a 17 year old with a script for help when you have a problem. To get real support costs money. Example: Digium. Daryl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] COnfiguring an * system for a non-profit organization
Hi, Maybe someone has seen this before... I've installed a new T100P, but it doesn't seem to work. I've attached the T100P to an Adtran 750 using a crossover cable. The Adtran shows a red alarm on the T1 interface. The Adtran has been set to factory defaults with FXS cards in 1-3 and an FXO card in 5. The T100P shows no signs of life--the leds are not lit. Should I be seeing lights on the T100P? Here is zaptel.conf: loadzone = us defaultzone=us span=1,0,0,esf,b8zs fxsls=1-12 fxols=17-20 This is the ztcfg (following the 'modprobe zaptel' command): [EMAIL PROTECTED] etc]# ztcfg -vv Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXS Loopstart (Default) (Slaves: 01) snip Channel 20: FXO Loopstart (Default) (Slaves: 20) 16 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) I've found two different T1 crossovers specifications on the web. The first is: 1-5 2-4 4-2 5-1 The second is: 1-4 2-5 4-1 5-2 Neither work--the Adtran always shows a red alarm on the T1 interface. Can you help me with this? Is there some way to determine if the T100P is working or if it's DOA? Thanks for your help, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] COnfiguring an * system for a non-profit organization
Hi Mike- Not sure why your card seems dead, but your second crossover spec seems to be the correct one. Here's a link to a good diagram of the crossover cable (see the bottom of the linked page). Only 1, 2, 4, and 5 are used. It is not necessary to connect the others. http://www.nmscommunications.com/NMS/nms_technotes.nsf/0/91d49c8785b2aab0852 566fa0050740a?OpenDocument Scott M. Stingel Emerging Voice Technology Inc. URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Welter Sent: Wednesday, December 03, 2003 3:43 PM To: [EMAIL PROTECTED]; Howard White Subject: [Asterisk-Users] COnfiguring an * system for a non-profit organization Hi, Maybe someone has seen this before... I've installed a new T100P, but it doesn't seem to work. I've attached the T100P to an Adtran 750 using a crossover cable. The Adtran shows a red alarm on the T1 interface. The Adtran has been set to factory defaults with FXS cards in 1-3 and an FXO card in 5. The T100P shows no signs of life--the leds are not lit. Should I be seeing lights on the T100P? Here is zaptel.conf: loadzone = us defaultzone=us span=1,0,0,esf,b8zs fxsls=1-12 fxols=17-20 This is the ztcfg (following the 'modprobe zaptel' command): [EMAIL PROTECTED] etc]# ztcfg -vv Zaptel Configuration == SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: FXS Loopstart (Default) (Slaves: 01) snip Channel 20: FXO Loopstart (Default) (Slaves: 20) 16 channels configured. ZT_SPANCONFIG failed on span 1: No such device or address (6) I've found two different T1 crossovers specifications on the web. The first is: 1-5 2-4 4-2 5-1 The second is: 1-4 2-5 4-1 5-2 Neither work--the Adtran always shows a red alarm on the T1 interface. Can you help me with this? Is there some way to determine if the T100P is working or if it's DOA? Thanks for your help, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Any updates on the Cisco 7920 and SIP?
I know this isn't the Cisco list, but enough people here are wired into the VoIP world that perhaps someone has heard if Cisco has released a SIP image for the 7920 yet... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Issues with Privacy Manager and Zapateller
I am still having these same problems. Anyone with experience with
these apps that could point me in the right direction?
I am having issues with Privacy Manager and Zapateller.
If I set callerid= on a sip user zapateller sends the tones
If I set callerid=Anonymous 8475551212 zapateller doesn't send
the
tones
If I call from a phone after dialing *67 zapateller doesn't send the
tones
In the last 2 cases, the display on the phone shows -Blocked Call-
PrivacyManager always gives the following messages:
-- Executing PrivacyManager(SIP/8475551212-9ec4, ) in new
stack
-- CallerID Present: Skipping
Even when the phone shows -Blocked Call- and even when zapateller
sends
tones.
Here is the Dial-Plan for the extension
exten = _NXXNXX/,1,Zapateller
exten = _NXXNXX,1,NoOp
exten = 847666,2,PrivacyManager
exten = 847666,3,Dial(SIP/${EXTEN},,r)
exten = 847666,4,Hangup
Stephen
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Re: [Asterisk-Users] VoiceGlo
I bet they are going to use SIP at some point.. just not yet. On Wed, 3 Dec 2003, Gary wrote: which would make their Multimedia Terminal Adapter an interesting device ?? On Wed, 3 Dec 2003 10:41:15 +1100, Adam Hart wrote: did you even read what I said? but if you look, it's actually using iaxcomm - Original Message - From: Brian West [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 5:21 AM Subject: Re: [Asterisk-Users] VoiceGlo WROOGGG Voiceglo's webphone is IAX and they use GSM. I have my Asterisk server registered with voiceglo right now.. so I know for a fact its IAX :P s you didn't hear that from me. bkw On Tue, 2 Dec 2003, Adam Hart wrote: from their site: What technology does voiceglo use? voiceglo uses a standard voice-over-IP protocol called SIP with patent-pending software that allows voiceglo endpoints to work on IP networks that employ address translations (NAT) and firewalls. voiceglo also uses advanced voice compression protocols to maximize voice quality and minimize latency over IP networks. In many instances, voiceglo's voice quality exceeds that available on PSTN or cell phone networks. but if you look, it's actually using iaxcomm - i'd like to see them patent that. (side note: I can't get the source code from anywhere on the site but iaxcomm is LGPL) - Original Message - From: Chris HARIGA To: [EMAIL PROTECTED] Sent: Tuesday, December 02, 2003 9:34 AM Subject: [Asterisk-Users] VoiceGlo Hi, VoiceGlo is comercial version of Asterisk? :))) loo Take a loock on http://www.voiceglo.com/ The softphone is IAX :) Best regards, Chris HARIGA Techselesta Inc. http://www.techselesta.com/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Implement missing features in Meetme application
Hi all ( dev user list ),
I'm starting to implement the missing features in Meetme application :
's' -- send user to admin/user menu if '*' is received
Line 438
app_meetme.c
-
else if ((f-frametype == AST_FRAME_DTMF) (f-subclass == '*') (confflags
CONFFLAG_STARMENU)) {
if ((confflags CONFFLAG_ADMIN)) {
/* Do admin stuff here */
} else {
/* Do user menu here */
}
I guess to use the pbx_builtin_background ( that already implement a loop playing
waitting a digit )
to play the menu , and allow Admin/User choose a option sending DTFM.
And i would like to know the better way to implement that ... any hint about ?
Thanks a lot,
P.S : Future plans will be more complex as be able to join a new caller to conference
room using
outgoing call , implement the options of Admin/User menu as :
- Give/Remove Talk only / Monitor only to an user .
- Kick a User from the conference room
- etc ...
Something as IRC channel management ;-)
--
Angel Carpintero - [EMAIL PROTECTED]
_
ELECTRONIC GROUP INTERACTIVE - www.electronic-group.com
World Trade Center, Moll de BARCELONA
Edificio Norte 4 Planta
08039 BARCELONA SPAIN
Tel :+34 93 600 23 23
Direct : +34 93 600 23 19
Fax :+34 93 600 23 10
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--
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World Trade Center, Moll de BARCELONA
Edificio Norte 4 Planta
08039 BARCELONA SPAIN
Tel :+34 93 600 23 23
Direct : +34 93 600 23 19
Fax :+34 93 600 23 10
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[Asterisk-Users] Cisco and Asterisk 2621
Ok here is a question that has gotten me stumped. I have an Asterisk system up and running. I need toconnect it via the Internet to a Sip Cisco system. This is what they have. I have there IP address's and login. X-lite is able to connect to them and make a call! So I have the name right! CISCO router model: 2621 VoIP module: NM-HDA-4FXS I have done Google lookup and at the Wiki about this. WhatI didgetis the following from them.Following in the SIP.CONF file. register = [EMAIL PROTECTED]:5060 This does not seem to work. I have also tried at the extensions.conf a setting of. exten = 380,1,Dial(SIP/[EMAIL PROTECTED]) I feel I have missed something some place or I just don't understand what to do!
Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm, etc.)
On 03/12/03 16:43, Steven Sokol wrote: Thanks, but I already have the clients configured as IAX2 rather than IAX. The failure is not universal (not ALL calls are missed). Rather the client seems to go to sleep for some reason -- almost always after handling a call. I have been monitoring the process from both the Asterisk CLI (with IAX2 debug and IAX debug turned on), from Gastman (monitoring call activity), and from a packet sniffer (unfortunately not Ethereal with the new plugin). Trust me on this one - you *really* want to take the time to install Etheral with the plugin. It makes debugging problems like this much easier - you'll be able to see whether the client sees the packet, whether it sends a response, if there's version skew causing INVALID packets to be sent for certain challenge/responses, etc. I'd only stick trace code in the iax-client library when you've sniffed what's going on so you know where to add it. :) I can, I suppose, add some trace code to the iaxClient library, but I don't really know where to go in the code to get it to trace/log. I would like to place it as low as possible -- in the listener function, then perhaps in the parser. If anybody knows how to do this, please let me know. My C coding skills are fairly rusty. Just point out the proper file and function(s) and I will be on my way. iaxclient/lib/libiax2/src/iax.c is probably where you'd want to look. Which functions depends on what's happening. iax_do_event() might be relevant for outbound packets, for example. You'll have to delve. Alastair ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re-routing of existing calls
Does Asterisk have the capability to re-route calls that have already been connected? By this, I mean: 1. A call comes in to Asterisk. 2. It is routed to an extension as normal. 3. This extension answers, and the conversation starts. 4. After a few minutes, a plugin that I am writing decides that it wants to transfer the call to somewhere else. 5. It signals this to the core of Asterisk (this is the part I am unsure how to do, if it can be done at all). 6. Asterisk hangs up on the extension. 7. (optional) Asterisk plays a 'please hold' message to the caller. 8. The call is routed to the new extension. Is this possible? Can anyone point me to documentation on how to do step 5? -- Alistair Cunningham, Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Transfer via # on Grandstream not always working
Anton Yurchenko wrote: Hello, After a while the transfer on grandstream stops working, only the reboot fixes the problem. It also seems that it may be the phone I`m trying to transfer _to_ also sometimes requires a reboot. After that it starts working. I`m using RFC2833 signlaing between phones and *. Does anybody see this happening also? Thanks When I first started using GS phones with *, I tried RTP signaling and had a problem with bouncy keys. I switched to SIP signaling and all is well. From what I can remember looking at the sniff traces, it appeared to be an * bug, not a GS bug. But SIP works well.. -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Any updates on the Cisco 7920 and SIP?
I have not heard and I was just looking myself. I would say no at this time, possible 1st QTR 2004 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Todd Sent: Wednesday, December 03, 2003 11:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Any updates on the Cisco 7920 and SIP? I know this isn't the Cisco list, but enough people here are wired into the VoIP world that perhaps someone has heard if Cisco has released a SIP image for the 7920 yet... JT ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco and Asterisk 2621
I have a 2621 working with asterisk. See below: sip.conf == [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls [cisco] ; Cisco 2621 Router type=friend canreinvite=no insecure=yes host=192.168.62.1 ; address of the cisco router dtmfmode=inband context=default extensions.conf === ; My asterisk numbers are 600-699 (omitted from example) ; Send all calls prefixed with 9 to the cisco exten = _9.,1,Dial,sip/[EMAIL PROTECTED] relevant part of cisco configuration [c2600-is-mz.122-13.T.bin] ! dial-peer voice 6 pots description Incoming Call from PSTN to number 6xx application session incoming called-number 6.. destination-pattern 6.. no digit-strip direct-inward-dial port 1/0:15 ! dial-peer voice 600 voip description Outgoing call to Asterisk Server for numbers 6xx application session destination-pattern 6.. session protocol sipv2 session target ipv4:192.168.62.60 session transport udp dtmf-relay rtp-nte codec g711ulaw ! dial-peer voice 9 voip description Incoming Call from Asterisk Server to number beginning with 9 application session incoming called-number 9T dtmf-relay rtp-nte codec g711ulaw ! dial-peer voice 900 pots description Outgoing call to PSTN for numbers beginning with 9 application session destination-pattern 9T no digit-strip port 1/0:15 ! -Original Message- From: Ariel Batista [mailto:[EMAIL PROTECTED] Sent: 03 December 2003 17:06 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco and Asterisk 2621 Ok here is a question that has gotten me stumped. I have an Asterisk system up and running. I need to connect it via the Internet to a Sip Cisco system. This is what they have. I have there IP address's and login. X-lite is able to connect to them and make a call! So I have the name right! CISCO router model: 2621 VoIP module: NM-HDA-4FXS I have done Google lookup and at the Wiki about this. What I did get is the following from them. Following in the SIP.CONF file. register = [EMAIL PROTECTED]:5060 This does not seem to work. I have also tried at the extensions.conf a setting of. exten = 380,1,Dial(SIP/[EMAIL PROTECTED]) I feel I have missed something some place or I just don't understand what to do! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re-routing of existing calls
check the manager interface ... you can transfer the active call to some other extension. (redirect). If these are zap channels there is zaptransfer command and zapdialoffhook via the manager. regards Martin On Wed, 3 Dec 2003, Alistair Cunningham wrote: Does Asterisk have the capability to re-route calls that have already been connected? By this, I mean: 1. A call comes in to Asterisk. 2. It is routed to an extension as normal. 3. This extension answers, and the conversation starts. 4. After a few minutes, a plugin that I am writing decides that it wants to transfer the call to somewhere else. 5. It signals this to the core of Asterisk (this is the part I am unsure how to do, if it can be done at all). 6. Asterisk hangs up on the extension. 7. (optional) Asterisk plays a 'please hold' message to the caller. 8. The call is routed to the new extension. Is this possible? Can anyone point me to documentation on how to do step 5? -- Alistair Cunningham, Email: [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Options for 3rd party call control
Mark Johnston wrote:
Alistair Cunningham [EMAIL PROTECTED] wrote:
I am working on a project on 3rd party call control for a call center, for
which I think Asterisk may be useful. What I would like to do is:
This is something I've given some thought to lately, with the goal of
writing a queueing engine to replace the basic Asterisk one. I'll
describe how I envision it inline.
- Have a call come in to Asterisk.
- Asterisk asks another machine, over a slow IP link, such as a modem, how
it
should route the call. Asterisk passes the called and calling numbers.
- This other machine looks up the destination, based on called and calling
numbers, in an SQL database, and responds to Asterisk.
- When Asterisk gets a reply, it routes the call.
[ answer, etc. ]
exten = s,5,AGI(router|getCallDestination)
exten = s,6,Dial(Something/${CallDestination})
which essentially treats your AGI script as a library. Your script
communicates with the remote machine and uses SET VARIABLE to set
CallDestination to whatever you like, and logic is handled in the
dialplan.
Mark,
This sounds ideal, and will be the approach that I will take. Thank you very
much!
--
Alistair Cunningham,
Email: [EMAIL PROTECTED]
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Re: [Asterisk-Users] How to set the gatekeeper? help me pls.
Lubomir Christov wrote: :) h323.conf is just a bit strange (there is no simple/clear alias options as in the oh323.conf) But it's a good idea to read Readme and h323.conf.sample ... ; H.323 Alias definitions ; ; Type 'h323' will register aliases to the endpoint ; and Gatekeeper, if there is one. ; ; Example: if someone calls [EMAIL PROTECTED] ; Asterisk will send the call to the extension 'time' ; in the context default ; ; [default] ; exten = time,1,Answer ; exten = time,2,Playback,current-time ; ; Keyword's 'prefix' and 'e164' are only make sense when ; used with a gatekeeper. You can specify either a prefix ; or E.164 this endpoint is responsible for terminating. ; ; Example: The H.323 alias 'det-gw' will tell the gatekeeper ; to route any call with the prefix 1248 to this alias. Keyword ; e164 is used when you want to specifiy a full telephone ; number. So a call to the number 18102341212 would be ; routed to the H.323 alias 'time'. ; ;[time] ;type=h323 ;e164=18102341212 ;context=default ; WHAT IS NOT CLEAR ABOUT THAT? Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re-routing of existing calls
Hi! Does Asterisk have the capability to re-route calls that have already been connected? Look at astman and its redirect button, I guess that is more or less what you want. So: Use the manager interface. Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm, etc.)
Hi, - Original Message - From: Grzegorz Nosek [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 5:08 PM Subject: Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm, etc.) On Tue, 2 Dec 2003 21:11:31 -0500 (EST), firedude wrote The new versions of iaxcomm and DIAX are both now using the iax2 protocol. So in order to receive incoming calls on either of them in your extensions.conf file change IAX/clientname to IAX2clientname. Then you should be able to receive incoming calls on either iaxcomm or DIAX. Also there is a mailing list for the iaxclient library. It's [EMAIL PROTECTED] Hope this helps. AJ or: exten=1500,1,Dial(IAX/clientIAX2/client,30) This is the best option in the mean time, till the IAX2 library bug will be solved. DIAX can use both IAX and IAX2, so please check if with IAX this is not an issue and send me your feedback. Thank you and best regards, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DIAX 0.9.5 and some resolutions for the displaty
Hi, I need to know if someone encounters display errors (like the window displayed partially) when some 'strage' resolutions are used for the display in Windows XP native theme mode. Thanks, Dan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How do you differentiate Busy and Congestion on Dialing PRI
OK, an answer is in README.variables causes.h...
[7-digit-PRI-Machine-2] ; The machine connected to PRI 2 (on its g1)
exten = _9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _9NXX,2,gotoif,$[${HANGUPCAUSE} = 2]?9|1
exten = 9,1,Busy
John
original message *
I have asterisk boxes in 2 different buildings each connected to the telco
with a PRI. I am now setting up asterisk machines in remote buildings -
dialing out via one of the other 2 machines. These are a snip from each
extension.conf on 1 remote and the 2 machines connected to the PRIs, to
illustrate what I want to do...
[remote_bldg_7_digit_out] ; The remote machine connected through IAX
exten = _9NXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
; if PRI-Machine-1 is congested or off-line, try PRI-Machine-2.
exten = _9NXX,2,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
exten = _9NXX,3,Congestion
[7-digit-PRI-Machine-1]; The machine connected to PRI 1
exten = _9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _9NXX,102,Busy
[7-digit-PRI-Machine-2] ; The machine connected to PRI 2 (on its g1)
exten = _9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _9NXX,102,Busy
...however Dial does not increment the priority by an extra 100 when it
encounters a busy on PRI. How can I best get this functionality?
John
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Re: [Asterisk-Users] CallerId in Voicemail message announcement??
--- Gary Mart [EMAIL PROTECTED] wrote: Is there a way to make the voicemail message announcement include the callerid. It would be handy to know who called (well, at least where the call was from) especially if they just hung up. I know I can get it from msg.txt but for the lay user it would be much more handy if it was included in the announcement. Gary Check: http://bugs.digium.com/bug_view_page.php?bug_id=156 Kevin _ Are you a Techie? Get Your Free Tech Email Address Now! Visit http://www.TechEmail.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Forwarding a call to another FXO port
Greetings, I'm trying to setup an option in my greetingmenu that would allow the caller to select this particular option for emergency calls. That option would dial out on an available PSTN line to a cell phone number. Currently it is setup as such exten = 9,1,Dial(Zap/g1/CELLPHONENUMBER where CELLPHONENUMBER is the number it is calling out to. When option 9 is selected, a horrible feedback noise is heard and caller cannot hear anything else. The cell phone that the call is going to does ring and can be answered and hears the same noise. Hardware on this is Asterisk Box - T100P - Adtran750 FXO channels are 1 and 2 set in group = 1 Both channels otherwise operate normally echocancel = 64 echocancelwhenbridged = no Any ideas? Raymond McKay ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unsuscribe
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RE: [Asterisk-Users] Forwarding a call to another FXO port
I would change the option number to something else because 9 is often picked up in another context as 9NXXNX You might have to make a sub menu in order to get there, but try using 2-8 for the menu options. Tim Thompson Commercial Sales Engineer http://www.amatechtel.com (806) 722-2227 -Original Message- From: Raymond McKay [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 12:59 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Forwarding a call to another FXO port Greetings, I'm trying to setup an option in my greetingmenu that would allow the caller to select this particular option for emergency calls. That option would dial out on an available PSTN line to a cell phone number. Currently it is setup as such exten = 9,1,Dial(Zap/g1/CELLPHONENUMBER where CELLPHONENUMBER is the number it is calling out to. When option 9 is selected, a horrible feedback noise is heard and caller cannot hear anything else. The cell phone that the call is going to does ring and can be answered and hears the same noise. Hardware on this is Asterisk Box - T100P - Adtran750 FXO channels are 1 and 2 set in group = 1 Both channels otherwise operate normally echocancel = 64 echocancelwhenbridged = no Any ideas? Raymond McKay ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to set the gatekeeper? help me pls.
Lubomir Christov wrote: [test01] type=h323 host = 192.168.10.12 context = your-incomming-context The keyword host in a type=h323 makes absolutely no sense. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How do you differentiate Busy and Congestion on Dialing PRI
John Harragin wrote:
OK, an answer is in README.variables causes.h...
[7-digit-PRI-Machine-2] ; The machine connected to PRI 2 (on its g1)
exten = _9NXX,1,Dial(Zap/g1/${EXTEN:1})
exten = _9NXX,2,gotoif,$[${HANGUPCAUSE} = 2]?9|1
exten = 9,1,Busy
Added to
http://www.voip-info.org/tiki-index.php?page=Asterisk%20variable%20hangupcause
/Olle
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Re: [Asterisk-Users] COnfiguring an * system for a non-profit organization
On Wednesday 03 December 2003 09:42, Michael Welter wrote: I've installed a new T100P, but it doesn't seem to work. I've attached the T100P to an Adtran 750 using a crossover cable. The Adtran shows a red alarm on the T1 interface. The Adtran has been set to factory defaults with FXS cards in 1-3 and an FXO card in 5. The T100P shows no signs of life--the leds are not lit. Should I be seeing lights on the T100P? Yes. You probably have not loaded the drivers. Try 'modprobe wct1xxp', then re-run ztcfg. -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT How to do it.
Hi! I need help to undestand the options: externip= static/ dynamic ip? can be a domain? localnet= internal ip of * machine? localmask= 255.255.255.0 ? Thanks - Original Message - From: Leif Madsen [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 7:25 AM Subject: RE: [Asterisk-Users] Asterisk behind NAT How to do it. On Tue, 2003-12-02 at 15:55, Arnold Ligtvoet wrote: Hi Leif, I tried the patch. Installed it exactly as described per your email. Thought that you might be interested that it works for me as well. Like a charm, I can finally call FWD numbers like 10001 and 612 (speaking clock demo). BTW: For anybody wanting to install this, if your version of chan_sip.c is older than the one described, first use 'cvs update -C asterisk/channels/chan_sip.c'. Awesome! Have you tried the newer patch / diff for 1.259 (which as of right now is the newest chan_sip file). If you goto bugs.digium.com and login anonymously and jump to bug 104, then you can get the newest patch. Same instructions as before. I just updated it to test the new sip.conf structure which is externip= localnet= localmask= Still working great for me here! BTW! Can you login to the bug tracker and post a comment ? Thanks! -- Leif Madsen [EMAIL PROTECTED] http://www.hacklocalhost.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk behind NAT How to do it.
On Wed, Dec 03, 2003 at 05:47:59PM -0200, listas iPfone wrote: Hi! I need help to undestand the options: hello. externip= static/ dynamic ip? can be a domain? externip can by an IP address or a domain. it uses gethostbyname(3) in the code. localnet= internal ip of * machine? localnet should be the internal network address not the internal ip address. i.e. if your asterisk server is 192.168.0.245, localnet should be 192.168.0.0 localmask= 255.255.255.0 ? that is correct. (unless you have a different netmasks of course) cheers, -w -- /~\ The ASCII Ribbon Campaign \ /No HTML/RTF in email X No Word docs in email / \ Respect for open standards ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] phone port on the x100p
Can the phone port on the x100p be an addressable extension on asterisk? I want to plug our conference phone into that phone jack as it is an analog phone. Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco IAD with MGCP
Message: 11 From: Juan J. Sierralta P. [EMAIL PROTECTED] To: Asterisk Users [EMAIL PROTECTED] Organization: Telefonica CTC Chile Date: 03 Dec 2003 12:23:26 -0300 Subject: [Asterisk-Users] Cisco IAD with MGCP Reply-To: [EMAIL PROTECTED] snip hostname 192.168.65.200 [192.168.65.200] host = 192.168.65.200 I seem to recall a similar issue with a different IAD. Try changing the hostname and endpoint name to something else (like cisco2430) darren ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: How to restart * thru phone when convenient
Hi! for the record: Put an behind the line? It does help to get a proper hang up for the client, but there is no restart initiated at all... looks like now the system calls gets cancelled due to the fact that the client is gone. Ah. Then put a 'nohup' in front of it: System(nohup /usr/sbin/asterisk -rx restart when convenient /dev/null ) Should do it. This works ok, but I tested it twice, and once * stopped but did not restart (could have other reasons, though). Thus I prefer the method below: System(echo '/usr/sbin/asterisk -rx restart when convenient /dev/null' | at now + 1 minute) Thanks guys! Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] phone port on the x100p
that's only a pass-through, no extension (fxs) is provided. Matteo. Il mer, 2003-12-03 alle 22:12, Todd Wallace ha scritto: Can the phone port on the x100p be an addressable extension on asterisk? I want to plug our conference phone into that phone jack as it is an analog phone. Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Dedicated * voicemail server
On Tue, 2003-12-02 at 08:27, Richard Alexander wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Low, Adam Sent: Tuesday, December 02, 2003 7:58 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Dedicated * voicemail server Hey All, I've started to try and distribute the functionality of my single * server amongst a few varying servers. The issue I have is that when splitting out the voicemail portion onto a dedicated server I am no longer able to inform the voicemail application (when call originated from a different box) if the call hitting the voicemail server was sent there because it was unanswered or if the phone was busy. I'm not sure if there is something within IAX that can pass this information on from one * server to another or if there is another solution ? Rgds, Adam * DISCLAIMER * This message and any attachment are confidential and may be privileged or otherwise protected from disclosure and may include proprietary information. If you are not the intended recipient, please telephone or email the sender and delete this message and any attachment from your system. If you are not the intended recipient you must not copy this message or attachment or disclose the contents to any other person ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users You could add an initial digit based on whether it was a busy or no answer forward, use the extra digit to determine the message played on the VM server and just strip it back off to get the mailbox number. Wouldn't it be easier to make a busy context and a unavailable context? Then the extension could be passed right on into voicemail without modifying it. You can route calls with iax to specific contexts right? -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] phone port on the x100p
-Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Todd Wallace Sent: Wednesday, December 03, 2003 4:12 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] phone port on the x100p Can the phone port on the x100p be an addressable extension on asterisk? I want to plug our conference phone into that phone jack as it is an analog phone. Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users In a word No. You can use it to check the POTS line and not much else. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] More voicemodem
On Wed, 2003-12-03 at 06:34, Hans-Henrik Andresen wrote: Hi, I got this setup. analog phone (ext7) --- analog pbx - (ext 6 analog) voicemodem (ext 3 asterisk) ttyS0/asterisk sipphones q1: I got the voicemodem to work, but oneway only. I can talk from my analog phone, to my sipphone, but not the other way ? I know it only suppose to works in half duplex, but nothing come TO the phone. by the X100P and stop whining. This is known behavior and expected to continue. q2: From SIPphone I dial 3+ext on my analog pbx - it works :) From analog phone I dial my voicemodem (ext 6) asterisk answer and it automatic forward to one specific sipphone, how do I get a new 'dialtone' from asterisk so I can dial ANY number in asterisk ? DISA. Hope to get some hints. (I'm really new to asterisk so an exsample would be good) /HHA ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users -- Steven Critchfield [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF
What DTMF options are available to me. My carrier is using DTMF relay H245 Alpha Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo cancel in MeetMe?
I'm trying to put multiple Linphones and Snom 200's into a Meetme room. With two devices, echo is quite noticeable. With 3 or more it degenerates into white noise. Which part of the software is responsible for echo cancellation in a MeetMe room? Is it a setting on the phones themselves, or within Asterisk? And is this related to echo cancellation on the POTS lines? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm, etc.)
On Wed, 3 Dec 2003 20:45:21 +0200, Dan wrote Hi, - Original Message - From: Grzegorz Nosek [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 5:08 PM Subject: Re: [Asterisk-Users] Iax Client Library Issues? (DIAX, iaxComm, etc.) On Tue, 2 Dec 2003 21:11:31 -0500 (EST), firedude wrote The new versions of iaxcomm and DIAX are both now using the iax2 protocol. So in order to receive incoming calls on either of them in your extensions.conf file change IAX/clientname to IAX2clientname. Then you should be able to receive incoming calls on either iaxcomm or DIAX. Also there is a mailing list for the iaxclient library. It's [EMAIL PROTECTED] Hope this helps. AJ or: exten=1500,1,Dial(IAX/clientIAX2/client,30) This is the best option in the mean time, till the IAX2 library bug will be solved. DIAX can use both IAX and IAX2, so please check if with IAX this is not an issue and send me your feedback. Thank you and best regards, Dan is this the bug that you mean? filed it today, patch included, works for me (tm). even if it isn't, take a look, it was a big showstopper for me as it essentially blocked any iax2 - iax2 call if any client used libiax2 (asterisk itself doesn't). http://bugs.digium.com/bug_view_page.php?bug_id=621 regards, greg ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] John Brown from Chagres!
I got an email from him this morning, and I quote: Hi Aaron, We are expecting a large container of GS product at the end of this week or Monday next week. This will clear all backorders that are currently in the system. BT-101, BT-102 and HT-286 products are in this container. Thank you for your understanding and patience. We will be including a little extra bonus with the order. John Brown Chagres Technologies, Inc. - Original Message - From: mattf [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 2:56 PM Subject: RE: [Asterisk-Users] John Brown from Chagres! Hello, We ordered 100 grandstream 102's from them over a month ago, we got the first shipment of 40 within a week and a half which was great. We got the next 40 a few weeks later. And we still have had no communications as to when the last 20 of the phones that we ordered(and paid for) over a month ago are going to be shipped to us. If you get a hold of him let me know, I'm still out 20 phones. MATT--- -Original Message- From: Aaron Martin [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 02, 2003 4:26 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] John Brown from Chagres! Sorry to everyone on the list, but for some reason this is the only reliable way to get hold of John. John Brown of Chagres Technologies, please contact me! I have been trying for weeks now to get hold of you via email and phone after wire transfering money into your account for the Grandstream phones we ordered, but so far I have not had a single response, nor have the phones arrived! Please contact me ASAP Aaron Martin Comtek Computing Solutions Ltd. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk behind NAT How to do it.
Leif wrote: Awesome! Have you tried the newer patch / diff for 1.259 (which as of right now is the newest chan_sip file). If you goto bugs.digium.com and login anonymously and jump to bug 104, then you can get the newest patch. Same instructions as before. Installed the new patch, no errors here. Ran make and copied chan_sip.o. All went fine. I just updated it to test the new sip.conf structure which is externip= localnet= localmask= Updated my sip.conf to match these settings. The result seems to be better, yesterday I noticed a slight delay in the setup of the audio channel, the speaking clock would only start at the second word, this is now gone. Still working great for me here! BTW! Can you login to the bug tracker and post a comment ? Thanks! I do have one strange issue. I have a test setup here which is very simple. * server and one windows machine. * is connected via ISDN (chan_i4l) to my home pbx. On my windows machine I installed Diax, SjPhone and SIPPS. The strange thing I now have is that both skinny clients are able to receive audio but not send any when I call an extension on my pbx (so external for *). I first thought it was my mic, but diax is working fine. I have already been looking at my sip.conf for the extensions but I'm not sure if this is the problem. Anyway my sip.conf now is : [general] disallow=all; Disallow all codecs allow=ulaw ; Allow codecs in order of preference allow=alaw allow=ilbc allow=gsm ; for fix 1.259 externip=212.238.144.173 localnet=192.168.0.100 localmask=255.255.255.0 [phone1] type=friend host=dynamic defaultip=192.168.0.2 dtmfmode=inband mailbox=1000 ; Mailbox for message waiting indicator context=default callerid=Me 2124 ;reinvite=no ;canreinvite=no ;nat=yes ;insecure=yes I'll wait your reply for the one-way sound 'issue' (probably me!) before posting to the bugtracker. Hopefully someone has some clue as to why my sip clients are not able to send sound. Thanks, Arnold Ligtvoet. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] John Brown from Chagres!
Good to hear... bkw PS when you think about Asterisk do you touch yourself? :P On Thu, 4 Dec 2003, Aaron Martin wrote: I got an email from him this morning, and I quote: Hi Aaron, We are expecting a large container of GS product at the end of this week or Monday next week. This will clear all backorders that are currently in the system. BT-101, BT-102 and HT-286 products are in this container. Thank you for your understanding and patience. We will be including a little extra bonus with the order. John Brown Chagres Technologies, Inc. - Original Message - From: mattf [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 2:56 PM Subject: RE: [Asterisk-Users] John Brown from Chagres! Hello, We ordered 100 grandstream 102's from them over a month ago, we got the first shipment of 40 within a week and a half which was great. We got the next 40 a few weeks later. And we still have had no communications as to when the last 20 of the phones that we ordered(and paid for) over a month ago are going to be shipped to us. If you get a hold of him let me know, I'm still out 20 phones. MATT--- -Original Message- From: Aaron Martin [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 02, 2003 4:26 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] John Brown from Chagres! Sorry to everyone on the list, but for some reason this is the only reliable way to get hold of John. John Brown of Chagres Technologies, please contact me! I have been trying for weeks now to get hold of you via email and phone after wire transfering money into your account for the Grandstream phones we ordered, but so far I have not had a single response, nor have the phones arrived! Please contact me ASAP Aaron Martin Comtek Computing Solutions Ltd. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OpenENUM
Anyone wishing to help build/manage openenum.net please contact me via email [EMAIL PROTECTED] ... I would like to have someone assist in building and management. Thanks, bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fax
Hi I have a second line that we use for a fax server Since we are luck to get 2 faxes a week I want to use this line as a dial out line for * But still need to be able to send and receive faxes on it Has anyone got any ideas how I could accomplish this ?? Regards Mick ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Echo problem on conferencing....no analog interfaces
Okay...here's one for all of you 3 party meet-me conference: Call 1: Comes in to MyAsterisk on an E1 PRI into the system. All TDM, no VoIP at all involved. No echo at all. Call 2: Call comes in via IAX(TDM - Asterisk_1 - IAX/GSM - MyAsterisk. Caller immediately hears his own echo Call 3: Call comes in via IAX(TDM - Asterisk_1 - IAX/GSM - MyAsterisk. Caller hears no echo at all. (Caller 2 and 3 called the same telephone numbercaller 2 is in the same state (NJ) and caller 3 is in California) Caller 2 hung up and called back instill hears echo. Any ideas? Are there any settings that anyone can suggest to try? Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip speaker phone for hands free intercom
Has anyone used the speakers on sip phones as part of an intercom? Are there sip messages you can send a phone to simulate key strokes, like someone hitting the speaker phone button on a GS? -- Bob Knight [-w] the work option [EMAIL PROTECTED] 925-449-9163 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to set the gatekeeper? help me pls.
Thank's Lubomir and Jeremy! It's working now. That's to say,I could dial long distance call from MSN or NetMeeting now. Regards. frank - Original Message - From: Jeremy McNamara [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 11:33 AM Subject: Re: [Asterisk-Users] How to set the gatekeeper? help me pls. Lubomir Christov wrote: [test01] type=h323 host = 192.168.10.12 context = your-incomming-context The keyword host in a type=h323 makes absolutely no sense. Jeremy McNamara ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Echo cancel in MeetMe?
Oops, my bad. Turns out it was just mixer settings, feeding back through the soundcard. Sorry for the noise. Message: 14 Date: Wed, 03 Dec 2003 17:43:16 -0500 From: Matt Lawson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Echo cancel in MeetMe? Reply-To: [EMAIL PROTECTED] I'm trying to put multiple Linphones and Snom 200's into a Meetme room. With two devices, echo is quite noticeable. With 3 or more it degenerates into white noise. Which part of the software is responsible for echo cancellation in a MeetMe room? Is it a setting on the phones themselves, or within Asterisk? And is this related to echo cancellation on the POTS lines? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax
On Thu, Dec 04, 2003 at 10:50:57AM +1030, [EMAIL PROTECTED] wrote: Hi I have a second line that we use for a fax server Since we are luck to get 2 faxes a week I want to use this line as a dial out line for * But still need to be able to send and receive faxes on it Has anyone got any ideas how I could accomplish this ?? Here's a strategy off the top of my head: 1. plug the line into your voicetronix the fax machine 2. configure asterisk to use it as the main dial-out line 3. configure asterisk to not pick up calls incoming on that line 4. configure users to check that the fax machine wasn't in use before dialing out, or to dial a special dial-out code to use the other line, so they don't stuff your faxes. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Fax
I have used this device with good results: http://faxswitch.com/stick_fax_phone_modem.html ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Fax
Thanks for that One question how do I stop * from picking up that line But still allow it to dial Regards Mick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Wood Sent: Thursday, 4 December 2003 11:25 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Fax On Thu, Dec 04, 2003 at 10:50:57AM +1030, [EMAIL PROTECTED] wrote: Hi I have a second line that we use for a fax server Since we are luck to get 2 faxes a week I want to use this line as a dial out line for * But still need to be able to send and receive faxes on it Has anyone got any ideas how I could accomplish this ?? Here's a strategy off the top of my head: 1. plug the line into your voicetronix the fax machine 2. configure asterisk to use it as the main dial-out line 3. configure asterisk to not pick up calls incoming on that line 4. configure users to check that the fax machine wasn't in use before dialing out, or to dial a special dial-out code to use the other line, so they don't stuff your faxes. cheers, Woody ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo problem on conferencing....no analog interfaces
Silly question: what kind of phone was the person in California calling on? Some phones give a local echo while you talk. If that happens, then I could see it causing problems... Just a thought... I hope it helps! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Lowe Sent: Wednesday, December 03, 2003 6:24 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Echo problem on conferencingno analog interfaces Okay...here's one for all of you 3 party meet-me conference: Call 1: Comes in to MyAsterisk on an E1 PRI into the system. All TDM, no VoIP at all involved. No echo at all. Call 2: Call comes in via IAX(TDM - Asterisk_1 - IAX/GSM - MyAsterisk. Caller immediately hears his own echo Call 3: Call comes in via IAX(TDM - Asterisk_1 - IAX/GSM - MyAsterisk. Caller hears no echo at all. (Caller 2 and 3 called the same telephone numbercaller 2 is in the same state (NJ) and caller 3 is in California) Caller 2 hung up and called back instill hears echo. Any ideas? Are there any settings that anyone can suggest to try? Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] More infor on my earlier DTMF question
My phone number is being hosted by a provider and brought inbound on a Cisco 5300. A Nextone softswitch is in the middle passing the inbound call to me as a SIP request to my * box. He shows he is sending me the DTMF's, but I am not picking them up and interpreting them. I have tried info, rfc2833, and inband. No luck. He has tried avail settings in the Nextone. We can't seem to sync up. I do not have this problem when dealing with the X100P, but I really want to have the call handed off SIP via this carrier. Anyone suggestions?? Todd Wallace ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Replicating Legacy Phone Behavior
Hello, I was demonstrating Asterisk capabilities with a SIP Soft phone to a Key system installer yesterday, and we were discussing where Asterisk can fit into that market. He brought up some interesting, user-centric questions which I couldn't answer. I didn't find anything in google that realy addressed some of these questions, so I figured I would post them here to see what yall have to say. Hopefully, someone has already answered this and they can point me to a link or some documentation that I can review. First and foremost, these Key System installers are big believers in VoIP and convergence technologies. While the KSU vendors may see Asterisk as competition, the installers on the ground see it as an excellent addition to help connect remote offices and workers together, but they are driven by the needs of their customers, most of whom want to KISS (Keep It Simple, Stupid). I.E. they want an Asterisk based VoIP solution to work in a similar manner to their existing PBX or Phone System. As a result, these are some of the questions that they threw at me that I am trying to figure out: 1. Legacy KSU and PBX users are used to seeing blinking lights on their phone that indicate outside lines in use, call on hold, voice mail waiting, do not disturb etc.. Is it possible to have these features using SIP phones on the dekstop? I.E. if a user puts a caller on hold at one extension, can it blink a light on all extensions so that user can be picked up at another extension? This gets into issues regarding re-training people with new phones etc.. Kind of like the issue of I don't want to press enter to make a call.. Why can't this phone just work like my old analog phone? 2. How does one go about creating call queues and advanced features such as UCD and ACD using Asterisk? 3. Is it possible to do Phone to Phone paging with SIP phones? This is a feature that I personally use a lot on my Legacy Phone System. I simply hit the extension of the persion I want to chat, and it beeps their phone and we can talk. Sort of like an Intercom system. Thanks in advance for helping me to answer these questions! -- Vice President of N2Net, a New Age Consulting Service, Inc. Company http://www.n2net.net Where everything clicks into place! KP-216-121-ST ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Soundblaster
Hi, I have the VIA chipset, and I'm trying to disable the sound and enable a soundblaster compatible card. Can you tell me what you did in /etc/modules.conf to enable your soundblaster card? Thanks, Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] John Brown from Chagres!
Several things conspired to muck things up the last 3-4 weeks. 1. Surgery (repair of a previous hernia) 2. Travel to work at opening our EU warehouse 3. TSA dropping my laptop, thus breaking my access to our VPN 4. New PRI going to a Asterisk box for our PBX and having the PRI be mucked up. Qwest dorking the number port. 5. 2 employees have a stressed life (they are 20 something and well, life is stressful for them) deciding not to really do the work they where suppose to. ergo cats away, mice play. We fired the kiddies. My other biz partner will be spending more time at Chagres now that he has sold his other company. He will handle operations, I'll handle sales and biz-dev. Two new employees start on Monday that will handle orders and customer calls. Inventory enroute from Grandstream, which will resolve all backorders. We will have stock of BT-101's BT-102's. HT-286s stock levels will be raised next week and we will have those as well. Chagres is alive and going well. We will have inventory of all GS product next week, most Digium product (T100P on 2 week delay from Digium), and maybe SIPURA. Good news is that we now have a Euro warehouse and starting in early January will ship Euro orders from Rotterdam. This will save our Euro customers much in shipping costs and transit time. I want to thank everyone for putting up with the mad couple of weeks, but things are shaped up and I think we can move forward. If anyone needs me urgently, my direct line is 505 998 0567 If I don't answer please leave a short but *clear* voice mail. I do check this voice mailbox several times per day. john brown fwd: 50870 direct: +1 505 998 0567 office: +1 505 830 1200 fax : +1 505 830 1201 On Tue, Dec 02, 2003 at 06:22:27PM -0600, Brian West wrote: I just talked to him lastnight... He was out of the office for a week or so. He got back and had to fire a few people for not doing their jobs.. and that he is slowly but surely getting caught up and that QWest screwed up their number porting. They moved their numbers from QWest to anohter provider and they aren't working... as of lastnight he was about to smack Qwest! :P Just an FYI bkw On Wed, 3 Dec 2003, Aaron Martin wrote: Sorry to everyone on the list, but for some reason this is the only reliable way to get hold of John. John Brown of Chagres Technologies, please contact me! I have been trying for weeks now to get hold of you via email and phone after wire transfering money into your account for the Grandstream phones we ordered, but so far I have not had a single response, nor have the phones arrived! Please contact me ASAP Aaron Martin Comtek Computing Solutions Ltd. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Asterisk overwrite any libraries?
Looks like your box has been compromised. Try ls -l `which ps` You'll probably find an inapropriate date. Whenever I've diagnosed problems like this, I've found badly installed rootkits. To address this on my production machines, I'm going to insruct the router to only allow traffic that is coming from trusted locations to connect to the box anyplace. I really hope I'm wrong about this Costas, but you should probably start verifying your binaries. If your machine has been compromised, a clean install, and patch with all the updated RPMS is a recommended soloution. Paul costas wrote: I am using a brand new RH9.0 installation. I installed Asterisk afterwards so I am not sure if Asterisk caused the problem below. The ps doesn't work. It could also be something else. I also tried installing a some video package. But I thought to ask here first if someone has seen this before. [EMAIL PROTECTED] asterisk]# ps ps: error while loading shared libraries: libproc.so.2.0.6: cannot open shared object file: No such file or directory [EMAIL PROTECTED] asterisk]# which ps /bin/ps Thanks Costas -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Free 20MB Web Site Hosting and Personalized E-mail Service! Get It Now At Doteasy.com http://www.doteasy.com/et/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Echo problem on conferencing....no analog interfaces
Not a silly question. I've given that thought. To be honest, I'm not sure what kind of phone the California or NJ callers were using. However, we've had numerous conferece calls using many other services and have never had echo problems. The problem, in this case, more likely has something to do with the NJ leg of the call (call 2), and possibly my leg (I was caller 1), since the echo happened when it was just the two of us. If anything, the problem phone would be my phone. This is a Lucent Partner phone system and phone. Now, if it was MY phone causing the problem, then both caller's 2 and 3 should be hearing their echo. But they weren't. I'm stumped. For anyone who's trying to figure out how I'm doing an E1 PRI here in the US, it's working like this: Verizon T1/PRI -- Cisco VCO/4K (Programmable switch) -- E1/PRI -- MyAsterisk The other 2 calls are going like this: 888 number -- SS7 -- VCO/4K -- T1/PRI -- Asterisk_1 -- IAX -- MyAsterisk Tom Tom Lowe, President/CTO Compro Technologies, Inc. 512 South Main Street Forked River, NJ 08731 My Phone: +1-212-904-0788 Main Phone: +1-609-242-2211 Fax: +1-609-242-2212 Email: [EMAIL PROTECTED] Web: www.comprotech.com -Original Message- From: David Gomillion [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 8:54 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Echo problem on conferencingno analog interfaces Silly question: what kind of phone was the person in California calling on? Some phones give a local echo while you talk. If that happens, then I could see it causing problems... Just a thought... I hope it helps! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Lowe Sent: Wednesday, December 03, 2003 6:24 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Echo problem on conferencingno analog interfaces Okay...here's one for all of you 3 party meet-me conference: Call 1: Comes in to MyAsterisk on an E1 PRI into the system. All TDM, no VoIP at all involved. No echo at all. Call 2: Call comes in via IAX(TDM - Asterisk_1 - IAX/GSM - MyAsterisk. Caller immediately hears his own echo Call 3: Call comes in via IAX(TDM - Asterisk_1 - IAX/GSM - MyAsterisk. Caller hears no echo at all. (Caller 2 and 3 called the same telephone numbercaller 2 is in the same state (NJ) and caller 3 is in California) Caller 2 hung up and called back instill hears echo. Any ideas? Are there any settings that anyone can suggest to try? Tom ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Does Asterisk overwrite any libraries?
A good rootkit will also modify the date and time of the replaced binaries so they will look the same as the original. Try to replace your ps command with that from a trusted RH9 machine. If it works ok then you must do a clean install to get rid of the rootkit. - Original Message - From: Paul Oster [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, December 03, 2003 10:24 PM Subject: Re: [Asterisk-Users] Does Asterisk overwrite any libraries? Looks like your box has been compromised. Try ls -l `which ps` You'll probably find an inapropriate date. Whenever I've diagnosed problems like this, I've found badly installed rootkits. To address this on my production machines, I'm going to insruct the router to only allow traffic that is coming from trusted locations to connect to the box anyplace. I really hope I'm wrong about this Costas, but you should probably start verifying your binaries. If your machine has been compromised, a clean install, and patch with all the updated RPMS is a recommended soloution. Paul costas wrote: I am using a brand new RH9.0 installation. I installed Asterisk afterwards so I am not sure if Asterisk caused the problem below. The ps doesn't work. It could also be something else. I also tried installing a some video package. But I thought to ask here first if someone has seen this before. [EMAIL PROTECTED] asterisk]# ps ps: error while loading shared libraries: libproc.so.2.0.6: cannot open shared object file: No such file or directory [EMAIL PROTECTED] asterisk]# which ps /bin/ps Thanks Costas -- Costas Menico Meezon Software Corp 201-224-8111 [EMAIL PROTECTED] -- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users Free 20MB Web Site Hosting and Personalized E-mail Service! Get It Now At Doteasy.com http://www.doteasy.com/et/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OpenENUM
http://lists.openenum.net Subscribe to policy if you wish to help with policy and building of OpenENUM. Thanks, Brian On Wed, 3 Dec 2003, Brian West wrote: Anyone wishing to help build/manage openenum.net please contact me via email [EMAIL PROTECTED] ... I would like to have someone assist in building and management. Thanks, bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OpenENUM
Ahh a memory I'd rather forget, unknown to most, John Todd and myself started a free enum service, similar to what you're doing. (it was called freenum.org) Unforunately, the project never really got going, due to lack of time and interest (after thinking it over). I believe it would never have enough numbers to warrant an enum lookup every time you call. Wasting time doing a dns lookup for the 1 in 1000 chance an enum entry will be there isn't worth it. Also, as soon as you get forged numbers (someone taking over other people's numbers), you'll be in big trouble. Our number was taken over and we lost 20% of our business as a result, i want a million dollars Here's a few points: 1) authenicating numbers - JT correctly pointed out, you can't allow people to call you to verify as caller id can be spoofed. He proposed a group of asterisk servers calling for verification. I was going to write into this advertising info so you could get businesses to do the calling for you eg you will be contact by an * server, sponsored by blah insert small banner or link 2) DNS - IMO, bind just won't work - PowerDNS or similar I'd suggest, dumping a zone file from mysql when you reach large numbers of entries doesn't scale 3) You need to work out a good and easy way to verify companies (ranges of numbers). Targetting the single line people I don't think will yield you enough numbers. 4) I think you need to allow users to either point their entry to their DNS or make an easy interface that will generate an entry for them. Don't force them to enter raw E164 entries (but let them if they really want to) 5) make a non profit organisation, or you'll get sued personally. good luck, I'm sure JT will have a few comments (probably cursing my name) Adam Anyone wishing to help build/manage openenum.net please contact me via email [EMAIL PROTECTED] ... I would like to have someone assist in building and management. Thanks, bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Application API
Asterisk Users, Does anyone know the URL for the application API for asterisk? I haven't been able to find any documentation on it. Thanks, Jonathan ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Outbound SIP Call
When I place an outbound call via my Cisco Sip devices 7960 and ATA using iconnect or nikotel as my SIP LD provider, the call connects and then disconnects after a few seconds. When the call is placed from an analog extension via the digium tdm40b it works fine. I have looked at the Debug but an unable to interpret the results. Does anyone have any suggestions? Thanks, Kevin
RE: [Asterisk-Users] Outbound SIP Call
Actually an update here.. there is no audio between any of the sip phones -Original Message- From: Kevin [mailto:[EMAIL PROTECTED] Sent: Thursday, December 04, 2003 12:02 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Outbound SIP Call When I place an outbound call via my Cisco Sip devices 7960 and ATA using iconnect or nikotel as my SIP LD provider, the call connects and then disconnects after a few seconds. When the call is placed from an analog extension via the digium tdm40b it works fine. I have looked at the Debug but an unable to interpret the results. Does anyone have any suggestions? Thanks, Kevin
Re: [Asterisk-Users] OpenENUM
At 3:13 PM +1100 12/4/03, Adam Hart wrote: Ahh a memory I'd rather forget, unknown to most, John Todd and myself started a free enum service, similar to what you're doing. (it was called freenum.org) Unforunately, the project never really got going, due to lack of time and interest (after thinking it over). I believe it would never have A worthy cause, to be sure, but there are many worthy causes out there, all of which take 8 hours a day. :-) enough numbers to warrant an enum lookup every time you call. Wasting time doing a dns lookup for the 1 in 1000 chance an enum entry will be there isn't worth it. Also, as soon as you get forged numbers (someone taking over other people's numbers), you'll be in big trouble. Our number was taken over and we lost 20% of our business as a result, i want a million dollars My goal for the project was to have it housed/resolved from servers located in a country that did not have overly aggressive corporate legal rights. None of the northern Europeans I spoke with had the time to work on such a development from the legal perspective, and I don't have any other contacts in such nations. Here's a few points: 1) authenicating numbers - JT correctly pointed out, you can't allow people to call you to verify as caller id can be spoofed. He proposed a group of asterisk servers calling for verification. I was going to write into this advertising info so you could get businesses to do the calling for you eg you will be contact by an * server, sponsored by blah insert small banner or link I think that this would be a minor cost issue for someone who wanted some good press for their VoIP service, based on the number of calls and the time curves. International might be a struggle, but perhaps gateways in commonly-accessed nations could be obtained. 2) DNS - IMO, bind just won't work - PowerDNS or similar I'd suggest, dumping a zone file from mysql when you reach large numbers of entries doesn't scale 3) You need to work out a good and easy way to verify companies (ranges of numbers). Targetting the single line people I don't think will yield you enough numbers. Yes, this is true. I came up with the (bad) idea that perhaps the first and last number in a range would be targetted, and then some small (1%?) of numbers in the middle of the range would be pre-chosen and the submitter would be told well in advance when those numbers would be tested, such that they would have someone answering on those lines. Limit the blocksize to something reasonable (1000 numbers? 500 numbers?) so that spoofing would be kept to a minimum. 4) I think you need to allow users to either point their entry to their DNS or make an easy interface that will generate an entry for them. Don't force them to enter raw E164 entries (but let them if they really want to) 5) make a non profit organisation, or you'll get sued personally. Indeed, this is critical, or see my point above about neutral nations. good luck, I'm sure JT will have a few comments (probably cursing my name) No, not at all. I too, have many projects to do right now as it stands, and while I think that an open parallel ENUM root is an excellent effort, I also have to keep perspective on the other projects in the works. ENUM is easy to put off, since we can all see real ENUM root service just over the next corner... cough, cough JT Adam Anyone wishing to help build/manage openenum.net please contact me via email [EMAIL PROTECTED] ... I would like to have someone assist in building and management. Thanks, bkw ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
