Re: [asterisk-users] Changing the messages that are played when a user is unavailable/busy
Thnx for your quick replies. I will try all of the above methods :-) On Fri, 2007-06-08 at 13:02 -0400, Justin Moore wrote: > On 6/8/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote: > > Log into your mailbox. Press "0", then press the option listed to > > record your unavail and busy greetings. > > I'm no expert, so someone feel free to correct me if I'm wrong, but > you should be able to make one or two recordings and then either copy > or symlink that file in > /var/spool/asterisk/voicemail/[context]/[user]/busy.[gsm|wav] and > /var/spool/asterisk/voicemail/[context]/[user]/unavail.[gsm|wav] > > If those files exist, Asterisk will play them instead of the default > "The person at extension XXX is not available right now, please" > > The only problem I forsee with this scenario is if the end-user goes > in to their voicemail and creates a new unavailable or busy message. > Surely you should be able to block them from over-writing those files > though my making them read-only. > > Best of luck. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing the messages that are played when a user is unavailable/busy
Hi, I have my custom sounds which should be played instead of the default ones when a user is busy or unavailable: "The person at extension XXX is not available right now, please" Of course I can simply replace the files, but the problem is my implementation shouldn't (MUST NOT) mention the extension. My files say something like "The person you are trying to reach is not available right now. If you want to contact this person on his cellphone, press 1, if you want to leave a voice message, press 2 or wait." As you can see the extension is not mentioned, so simply replacing the files would probably cause something weird. Where do I define what message are/aren't played in this case? Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can be called on FreeWorldDialup/IAX channel, but can't make calls
That extension is a mobile number, a number I called ealier that day, but does not seem to be related to my problem. On Sat, 2007-03-24 at 17:35 -0400, Chris Nighswonger wrote: > On 3/24/07, Timothy Parez <[EMAIL PROTECTED]> wrote: > > Mar 24 15:28:55 ERROR[2873]: chan_sip.c:11076 handle_request_subscribe: > > Got SUBSCRIBE for extension [EMAIL PROTECTED] from 172.17.249.253, > > but there is no hint for that extension > > I believe the subscribe error comes from not having a 'hint' in the > context of the extension for the sip @ 172.17.249.253 indicating the > sip at extension 00032498043823 (what an extension!). > > I am new myself to * so someone may need to correct me on this one. > > Chris > ___ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > - > > WARNING: Computer viruses can be transmitted via email. The recipient should > check this email and any attachments for the presence of viruses. The company > accepts no liability for any damage caused by any virus transmitted by this > email. E-mail transmission cannot be guaranteed to be secure or error-free as > information could be intercepted, corrupted, lost, destroyed, arrive late or > incomplete, or contain viruses. The sender therefore does not accept > liability for any errors or omissions in the contents of this message, which > arise as a result of e-mail transmission. > > Warning: Although the company has taken reasonable precautions to ensure no > viruses are present in this email, the company cannot accept responsibility > for any loss or damage arising from the use of this email or attachments > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Can be called on FreeWorldDialup/IAX channel, but can't make calls
Hi,
I have an FWD account and it's configured in asterisk.
I can be called by people using FWD, but I cannot make FWD calls myself.
Every number dialed with a 8 prefix goes to FWD,
if for example I call the echo servie I get this:
Connected to Asterisk 1.2.13 currently running on asterisk (pid = 2865)
Verbosity is at least 35
-- Executing SetCallerID("SIP/timothy-08224f08", ""Het Bos"") in new
stack
-- Executing Dial("SIP/timothy-08224f08",
"IAX2/814179:[EMAIL PROTECTED]/613|60|r") in new
stack
-- Called 814179:[EMAIL PROTECTED]/613
Mar 24 15:26:28 NOTICE[2875]: chan_iax2.c:2869 auto_congest:
Auto-congesting call due to slow response
-- IAX2/192.246.69.186:4569-5 is circuit-busy
-- Hungup 'IAX2/192.246.69.186:4569-5'
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Congestion("SIP/timothy-08224f08", "") in new stack
== Spawn extension (internal, 8613, 3) exited non-zero on
'SIP/timothy-08224f08'
Mar 24 15:28:55 ERROR[2873]: chan_sip.c:11076 handle_request_subscribe:
Got SUBSCRIBE for extension [EMAIL PROTECTED] from 172.17.249.253,
but there is no hint for that extension
Who can help me with this?
Thank you.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] windows mobile 5 softphone for square screen devices
Hi, I've been looking for a good SIP application for Windows Mobile for ages. I found speaQ, but it has the same problem as any other softphone for Windows Mobile. You see, it uses the speaker to output the conversation instead of the phone speaker, you know the one that is used when you make a normal phone call with your WM Mobile PDA/Smartphone. At first I was asking myself if every SIP client developer out there is down right stupid but in the end I found out this is actually Microsoft blocking access to that phone speaker. The claim that allowing the developers to access it would allow for invasion of privacy (like recording phone calls). So unless someone can work around this, softphones for WM will remain quite useless. Timothy. Anton Krall wrote: Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Wildcard B410P
You'll need mISDN A small tutorial (in dutch): http://www.blicbox.be/node/22 You should be able to translate using: http://babelfish.altavista.com/ Timothy. Henrik Woffinden wrote: Hi list, Is Digium Wildcard B410P compatible with Asterisk 1.2 / 1.4 (zaptel of course) directly out of the box, or do I need things like bristuff? http://www.digium.com/en/products/hardware/b410p.php Best regards, Henrik Woffinden ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX connection to FWD not working
Indeed, they can call me, I can call 613 but not them Their phone rings for like 1 second. I get callended. Alex Robar wrote: You mean that you can't call other FWD users? Alex On 12/20/06, *Timothy Parez* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: However I can call 613 and it works I can be called and it works but when I call any other number I get call ended right away :p Alex Robar wrote: > Hi Timothy, > > Mine seems to be working OK as of a few minutes ago: > > unlimited*CLI> iax2 show registry > Host UsernamePerceived Refresh State > 192.246.69.186:4569 <http://192.246.69.186:4569> <http://192.246.69.186:4569> 727044 > 216.58.41.183:4569 <http://216.58.41.183:4569> <http://216.58.41.183:4569> 60 Registered > > Do you have any other IAX trunks? Are they working for you? > > Alex > > > On 12/20/06, *Timothy Parez* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> > <mailto:[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>>> wrote: > > Ever since a few weeks ago the connection to FreeWorldDialup stopped > working on our Asterisk server: > > This is all we can get out of it: > > asterisk*CLI> iax2 show registry > Host UsernamePerceived > Refresh State > 192.246.69.186:4569 <http://192.246.69.186:4569> <http://192.246.69.186:4569 <http://192.246.69.186:4569>> > 814179 60 Timeout > 192.246.69.186:4569 <http://192.246.69.186:4569> <http://192.246.69.186:4569 > > 805208 60 Timeout > > Any ideas? > > > > > > > > - > > WARNING: Computer viruses can be transmitted via email. The > recipient should check this email and any attachments for the > presence of viruses. The company accepts no liability for any > damage caused by any virus transmitted by this email. E-mail > transmission cannot be guaranteed to be secure or error-free as > information could be intercepted, corrupted, lost, destroyed, > arrive late or incomplete, or contain viruses. The sender > therefore does not accept liability for any errors or omissions in > the contents of this message, which arise as a result of e-mail > transmission. > > Warning: Although the company has taken reasonable precautions to > ensure no viruses are present in this email, the company cannot > accept responsibility for any loss or damage arising from the use > of this email or attachments > ___ > --Bandwidth and Colocation provided by Easynews.com <http://Easynews.com> > <http://Easynews.com> -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > -- > Alex Robar > [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> <mailto:[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> > > > ___ > --Bandwidth and Colocation provided by Easynews.com <http://Easynews.com> -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments
Re: [asterisk-users] IAX connection to FWD not working
However I can call 613 and it works I can be called and it works but when I call any other number I get call ended right away :p Alex Robar wrote: Hi Timothy, Mine seems to be working OK as of a few minutes ago: unlimited*CLI> iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 <http://192.246.69.186:4569> 727044 216.58.41.183:4569 <http://216.58.41.183:4569> 60 Registered Do you have any other IAX trunks? Are they working for you? Alex On 12/20/06, *Timothy Parez* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI> iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 <http://192.246.69.186:4569> 814179 60 Timeout 192.246.69.186:4569 <http://192.246.69.186:4569> 805208 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --Bandwidth and Colocation provided by Easynews.com <http://Easynews.com> -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX connection to FWD not working
That's odd :) It's been like this for days I post a message and it's up ? :) They are now registered :) Cool. Alex Robar wrote: Hi Timothy, Mine seems to be working OK as of a few minutes ago: unlimited*CLI> iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 <http://192.246.69.186:4569> 727044 216.58.41.183:4569 <http://216.58.41.183:4569> 60 Registered Do you have any other IAX trunks? Are they working for you? Alex On 12/20/06, *Timothy Parez* <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote: Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI> iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 <http://192.246.69.186:4569> 814179 60 Timeout 192.246.69.186:4569 <http://192.246.69.186:4569> 805208 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --Bandwidth and Colocation provided by Easynews.com <http://Easynews.com> -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar [EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]> ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX connection to FWD not working
Ever since a few weeks ago the connection to FreeWorldDialup stopped working on our Asterisk server: This is all we can get out of it: asterisk*CLI> iax2 show registry Host UsernamePerceived Refresh State 192.246.69.186:4569 814179 60 Timeout 192.246.69.186:4569 805208 60 Timeout Any ideas? - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Settings CallerId for outgoing calls based on the sip account making them
Hi, I have 10 DID numbers. Calls coming from the PSTN network are routed correctly to the SIP users based on the number that was called. But when sip users call the PSTN network, the CallerID should be set to correspondent with their DID number. At the moment I can set the CallerID to a global number, but I have no idea how to check who's making the call. All sip users start in the context [internal] Any ideas? Thank you. - WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. Warning: Although the company has taken reasonable precautions to ensure no viruses are present in this email, the company cannot accept responsibility for any loss or damage arising from the use of this email or attachments ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN
Hi,
Sorry, uncomenting that actually worked.
Now I need to filter on the last two numbers, that shoulnd't be to hard
I guess.
Tim.
Giorgio Incantalupo schreef:
Hi Patrick,
it seems like Asteirsk cannot match any number, try uncomment the two
following lines:
;exten => _X.,1,Dial(SIP/timothy,30,r)
;exten => _X.,2,Hangup()
Giorgio Incantalupo
On Wed, 2006-11-29 at 13:26 +0100, Timothy Parez wrote:
Hi,
I'm able to place outgoing calls using mISDN,
but I cannot get incoming calls to work.
Whenever someone calls one the incoming numbers I get this:
Nov 29 13:21:48 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
The caller is then informed by our telco company that the number is
unavailable.
In misdn.conf I have
[myoutsidelines]
msns=*
ports=1,2,3,4
context=inisdn
I then have a context in extensions.conf
[inisdn]
;exten => _.,1,NoOp(Incoming Call from telco ${CALLERID} for
[EMAIL PROTECTED])
;exten => _.,2,LookupCIDName
;exten => _NXXNXX,3,Dial(sip/sammy,30,r)
;exten => h,1,HangUp()
;exten => s,1,Dial(SIP/timothy)
;exten => s,2,Hangup()
;exten => _X.,1,Dial(SIP/timothy,30,r)
;exten => _X.,2,Hangup()
exten s,1,NoOp(Incoming call from ${CALLERID} for ${EXTEN})
exten s,2,Answer()
exten s,3,Echo()
exten s,4,Hangup()
exten i,1,NoOp(Invalid call from ${CALLERID} for ${EXTEN})
exten i,2,Answer()
exten i,3,Echo()
exten i,4,Hangup()
I had a similar issue. If you turn on the chan_misdn debug messages than
you can see what chan_misdn sees as the incoming number. My problem was
that my dialplan had for example 881234567 while chan_misdn was seeing
0881234567 so there was no match. A quick change from 881234567 to
0881234567 in my dialplan fixed it.
Regards,
Patrick
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN
I get the following with debug on:
P[ 3] I IND :SETUP oad:497978546 dad:50556010 pid:10 state:none
P[ 3] --> channel:1 mode:TE cause:16 ocause:16 rad: cad:
P[ 3] --> info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0
P[ 3] --> Bearer: Speech
P[ 3] --> Codec: Alaw
P[ 0] --> * NEW CHANNEL dad:50556010 oad:497978546
P[ 3] --> CTON: Unknown
P[ 3] EXPORT_PID: pid:10
P[ 3] --> PRES: Restricted (0)
P[ 3] --> SCREEN: Unscreened (0)
Nov 29 16:39:40 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
P[ 3] I SEND:RELEASE oad:0497978546 dad:050556010 pid:10
P[ 3] --> bc_state:BCHAN_CLEANED
P[ 3] --> channel:1 mode:TE cause:16 ocause:1 rad: cad:
P[ 3] --> info_dad: onumplan:2 dnumplan:2 rnumplan: cpnnumplan:0
P[ 3] I IND :RELEASE_COMPLETE oad: dad: pid:10 state:EXTCANTMATCH
P[ 3] --> channel:0 mode:TE cause:16 ocause:16 rad: cad:
P[ 3] --> info_dad: onumplan:0 dnumplan:0 rnumplan:0 cpnnumplan:0
P[ 3] hangup_chan
P[ 3] -> hangup
P[ 3] * IND : HANGUPpid:10 ctx:inisdn dad:050556010 oad:0497978546
State:EXTCANTMATCH
P[ 3] --> l3id:6000b
P[ 3] --> cause:16
P[ 3] --> out_cause:16
P[ 3] --> state:EXTCANTMATCH
P[ 3] Channel: mISDN/3-1 hanguped new state:CLEANING
P[ 3] release_chan: bc with l3id: 6000b
so I change extensions.conf:
exten 50556010,1,NoOp(Incoming call from ${CALLERID} for ${EXTEN})
exten 50556010,2,Answer()
exten 50556010,3,Echo()
exten 50556010,4,Hangup()
And the debug message above is what I get
Timothy.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] mISDN
Hi,
I did, that was my first try,
but it didn't work.
Giorgio Incantalupo schreef:
Hi Patrick,
it seems like Asteirsk cannot match any number, try uncomment the two
following lines:
;exten => _X.,1,Dial(SIP/timothy,30,r)
;exten => _X.,2,Hangup()
Giorgio Incantalupo
On Wed, 2006-11-29 at 13:26 +0100, Timothy Parez wrote:
Hi,
I'm able to place outgoing calls using mISDN,
but I cannot get incoming calls to work.
Whenever someone calls one the incoming numbers I get this:
Nov 29 13:21:48 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
The caller is then informed by our telco company that the number is
unavailable.
In misdn.conf I have
[myoutsidelines]
msns=*
ports=1,2,3,4
context=inisdn
I then have a context in extensions.conf
[inisdn]
;exten => _.,1,NoOp(Incoming Call from telco ${CALLERID} for
[EMAIL PROTECTED])
;exten => _.,2,LookupCIDName
;exten => _NXXNXX,3,Dial(sip/sammy,30,r)
;exten => h,1,HangUp()
;exten => s,1,Dial(SIP/timothy)
;exten => s,2,Hangup()
;exten => _X.,1,Dial(SIP/timothy,30,r)
;exten => _X.,2,Hangup()
exten s,1,NoOp(Incoming call from ${CALLERID} for ${EXTEN})
exten s,2,Answer()
exten s,3,Echo()
exten s,4,Hangup()
exten i,1,NoOp(Invalid call from ${CALLERID} for ${EXTEN})
exten i,2,Answer()
exten i,3,Echo()
exten i,4,Hangup()
I had a similar issue. If you turn on the chan_misdn debug messages than
you can see what chan_misdn sees as the incoming number. My problem was
that my dialplan had for example 881234567 while chan_misdn was seeing
0881234567 so there was no match. A quick change from 881234567 to
0881234567 in my dialplan fixed it.
Regards,
Patrick
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: IAX access to FWD broken?
Still doesn't work for me. Still get timeout Michael Graves schreef: I'm travelling today but I was just able to use Firefly to login to FWD via IAX2. I called the echo test with no problems other than the lousy network in this hotel. My Astlinux server Also reports that it's registered with FWD via IAX2. My account is a couple years old. Michael On Wed, 29 Nov 2006 09:49:25 -0500, Al Bochter wrote: >FWD works fine for me. I just set up a trunk in asterisk. > >Best regards, > >Al Bochter >Bochter Services >_http://www.BochterServices.com/?t=Email_ > >(VOIP PBX) 1-866-638-1254 > >(Voip PBX) Free World DialUp: 780-217 >WebSite: _http://www.freeworlddialup.com/_ > >We have Toll Free DID's instock >* * * NO MONTHLY FEE - LIMITED TIME ONLY * * * >_http://www.bochterservices.com/?t=TF(NM)did_ > >BUY Coins, Silver and Gold >_http://www.bochterservices.com/?j=gold&t=email_ > >For new and used security items >_http://www.bochterservices.com/?j=store&t=email_security_ > > > >Jim Lawson wrote: > >> Just as an "it works for me", I created a FWD account a couple of >> weeks ago, which seems to be working fine. I am able to receive calls >> over IAX2 via my IpKall number. >> >> Jim >> >> Timothy Parez wrote: >> >>> I have one account which was created 3 weeks ago and 1 that was >>> created 2 days ago, neither work. jason schreef: >>> >>>> > last I had heard, pretty much all FWD accounts that were created >>>> in > the past year or so no longer work with IAX. Still don't know >>>> why. >>> >> >> ___ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> _http://lists.digium.com/mailman/listinfo/asterisk-users_ >> >> >> >> >> >> Inbound (clean). Database: 0651-2, 11/28/2006 - 11/29/2006 9:27:39 AM >> >> >> >> >___ >--Bandwidth and Colocation provided by Easynews.com -- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > _http://lists.digium.com/mailman/listinfo/asterisk-users_ > > ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX access to FWD broken?
I just sent an e-mail to the FWD support address, I'll let you know where that gets me. jason schreef: last I had heard, pretty much all FWD accounts that were created in the past year or so no longer work with IAX. Still don't know why. Timothy Parez wrote: I've got the same problem here. It can't register anymore --> timeout Brian Capouch schreef: I hadn't used FWD for quite a while. A customer sent me an email last week, "Is FWD broken when one tries to use it with IAX?" I have been playing around, and indeed seems to be the case. Is there anyone out there successfully using the two of them together? Thanks. B. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX access to FWD broken?
I have one account which was created 3 weeks ago and 1 that was created 2 days ago, neither work. jason schreef: last I had heard, pretty much all FWD accounts that were created in the past year or so no longer work with IAX. Still don't know why. Timothy Parez wrote: I've got the same problem here. It can't register anymore --> timeout Brian Capouch schreef: I hadn't used FWD for quite a while. A customer sent me an email last week, "Is FWD broken when one tries to use it with IAX?" I have been playing around, and indeed seems to be the case. Is there anyone out there successfully using the two of them together? Thanks. B. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] mISDN
Hi,
I'm able to place outgoing calls using mISDN,
but I cannot get incoming calls to work.
Whenever someone calls one the incoming numbers I get this:
Nov 29 13:21:48 WARNING[7221]: chan_misdn.c:4735 chan_misdn_log:
Extension can never match, so disconnecting
The caller is then informed by our telco company that the number is
unavailable.
In misdn.conf I have
[myoutsidelines]
msns=*
ports=1,2,3,4
context=inisdn
I then have a context in extensions.conf
[inisdn]
;exten => _.,1,NoOp(Incoming Call from telco ${CALLERID} for
[EMAIL PROTECTED])
;exten => _.,2,LookupCIDName
;exten => _NXXNXX,3,Dial(sip/sammy,30,r)
;exten => h,1,HangUp()
;exten => s,1,Dial(SIP/timothy)
;exten => s,2,Hangup()
;exten => _X.,1,Dial(SIP/timothy,30,r)
;exten => _X.,2,Hangup()
exten s,1,NoOp(Incoming call from ${CALLERID} for ${EXTEN})
exten s,2,Answer()
exten s,3,Echo()
exten s,4,Hangup()
exten i,1,NoOp(Invalid call from ${CALLERID} for ${EXTEN})
exten i,2,Answer()
exten i,3,Echo()
exten i,4,Hangup()
As you can see I tried a few things, but none of them work.
Does anybody know how to solve this ?
Thnx.
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IAX access to FWD broken?
I've got the same problem here. It can't register anymore --> timeout Brian Capouch schreef: I hadn't used FWD for quite a while. A customer sent me an email last week, "Is FWD broken when one tries to use it with IAX?" I have been playing around, and indeed seems to be the case. Is there anyone out there successfully using the two of them together? Thanks. B. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Installing the b410p card, unable to install mISDN
Hi, I'm installing Asterisk on Ubuntu 6.10 When I first compiled the zaptel package I used: make clean make make install So far so good, but the following command failed: make b410p I did some digging on google and found a guide on how to install it manually, but the result was the same. I got these files ftp.digium.com/pub/telephony/zaptel/b410p/misdn-b410p.tar.gz ftp.digium.com/pub/telephony/zaptel/b410p/mISDNuser.tar.gz And did the following: make force make make install All the commands worked just fine. Remark: The guide I was using told me to cd into the mISDNuser directory, but didn't do anything with it. /etc/init.d/misdn-init scan gives me: [OK] found the following devices: card=1,0x4 So I ran /etc/init.d/misdn-init config The output: [OK] /etc/misdn-init.conf created. It's now safe to run "/etc/init.d/misdn-init start" [ii] make your ports (1-4) available in asterisk by editing "/etc/asterisk/misdn.conf" [ii] run "/etc/init.d/misdn-init config" to store this information to /etc/misdn-init.conf I then edited the setting in /etc/misdn-init.conf, guessing I'll be needing nt_ptmp=1,2,3,4 (although I have no idea) Now the problem, if I run /etc/init.d/misdn-init start I get the following: /etc/init.d/misdn-init: line 91: [: 5: unary operator expected - Loading module(s) for your misdn-cards: - modprobe --ignore-install hfcmulti type= protocol=,,, layermask=0x3,0x3,0x3,0x3 poll=128 debug=0xf De output of the lsmod | grep hfcmulti command is hfcmulti 74984 0 mISDN_core 85248 6 mISDN_dsp,mISDN_capi,l3udss1,mISDN_l2,mISDN_l1,hfcmulti The output of dmesg | grep Digium is [42949387.38] HFC-MULTI: Card 'HFC-4S Digium Card' found, but not given by module's options, ignoring... I recompile Asterisk and install it again, but I do not get the misdn command in asterisk. What am I missing ? Thnx. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Zaptel - make b410p fails on Ubuntu 6.10
Hi, I've been able to make make install the Zaptel drivers (1.2). I'm using a b410p so I executed the following command make b410p. I tried this on multiple machines, but it always failes: [EMAIL PROTECTED]:/usr/src/zaptel-1.2.11# make b410p [ -f misdn-b410p.tar.bz ] || wget ftp://ftp.digium.com/pub/zaptel/b410p/misdn-b410p.tar.gz --23:59:54-- ftp://ftp.digium.com/pub/zaptel/b410p/misdn-b410p.tar.gz => `misdn-b410p.tar.gz' Resolving ftp.digium.com... 216.27.40.102, 69.16.138.164 Connecting to ftp.digium.com|216.27.40.102|:21... connected. Logging in as anonymous ... Logged in! ==> SYST ... done.==> PWD ... done. ==> TYPE I ... done. ==> CWD /pub/zaptel/b410p ... done. ==> PASV ... done.==> RETR misdn-b410p.tar.gz ... done. Length: 572,153 (559K) (unauthoritative) 100%[==>] 572,153 61.22K/sETA 00:00 00:00:11 (38.14 KB/s) - `misdn-b410p.tar.gz' saved [572153] tar -zxf misdn-b410p.tar.gz make -C misdn install make[1]: Entering directory `/usr/src/zaptel-1.2.11/misdn' Makeing mISDN = cp /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/Makefile.v2.6 /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/Makefile export MINCLUDES=/usr/src/zaptel-1.2.11/misdn/include ; make -C /lib/modules/2.6.17-10-server/build SUBDIRS=/usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN modules CONFIG_MISDN_DRV=m CONFIG_MISDN_DSP=m CONFIG_MISDN_HFCMULTI=m make[2]: Entering directory `/usr/src/linux-headers-2.6.17-10-server' CC [M] /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.o In file included from /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/channel.h:13, from /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:20: include/linux/mISDNif.h:570: warning: âpackedâ attribute ignored for field of type âu_charâ include/linux/mISDNif.h:571: warning: âpackedâ attribute ignored for field of type âu_charâ In file included from /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/channel.h:16, from /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:20: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h: In function âmISDN_queueup_newheadâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:189: warning: implicit declaration of function âmISDN_queue_messageâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:189: error: âFLG_MSG_UPâ undeclared (first use in this function) /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:189: error: (Each undeclared identifier is reported only once /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:189: error: for each function it appears in.) /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h: In function âmISDN_queuedown_newheadâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:199: error: âFLG_MSG_DOWNâ undeclared (first use in this function) /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h: At top level: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/helper.h:280: error: expected â=â, â,â, â;â, âasmâ or â__attribute__â before â*â token In file included from /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:20: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/channel.h: In function âqueue_ch_frameâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/channel.h:108: error: âFLG_MSG_UPâ undeclared (first use in this function) /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c: In function âwrite_ctrlâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:275: error: âmISDNinstance_tâ has no member named âprivatâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c: In function âhdlc_empty_fifoâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:409: error: âmISDNinstance_tâ has no member named âprivatâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c: In function âhdlc_fill_fifoâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:478: error: âmISDNinstance_tâ has no member named âprivatâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c: In function âhdlc_downâ: /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:769: error: âmISDNinstance_tâ has no member named âhwlockâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:775: error: âmISDNinstance_tâ has no member named âhwlockâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:781: error: âmISDNinstance_tâ has no member named âhwlockâ /usr/src/zaptel-1.2.11/misdn/drivers/isdn/hardware/mISDN/avm_fritz.c:784: error: âmISDNinstance_tâ has no member named âhwlockâ
[asterisk-users] Starting out
Hi, I have to decide on hardware to buy real fast (being rocketed into the situation). We have 1 computer, we'll install hardware from digium in there to connect with the ISDN phone lines (2) It's a normal computer, I have no idea what type of card to take and about the 3.3v vs 5v PCI. The idea is the following. We'll have about 10 internal phones. One of the phones should be like a central station, where all other calls can be monitored (if possible) and from that phone the user should be able to press a button to take over a call which is rining on another phone. Then we need less advanced phones for the rest of us, but we should still be able to pick up calls that are rining on a phone in the same room. (if possible) I live in Belgium and we are using ISDN lines. If I had to select phones from this page: http://www.voipsolutions.be/index.php/cPath/54_24 What whould you sugest and why ? Also from this page: http://www.asterisk.org/hardware What would you sugest and why ? Stuff we need - Call forwarding (to another internal phone, to a classic phone number) - Call take over (picking up a phone that is rining somewhere else) - Menu system (got this working) - Voicemail (got this working) - Allowing a employee who's in a hotel somewhere to phone the internal numbers using his softphone over the internet - Allowing that same employee to use his sotphone in order to make phonecalls to normal landlines through our server - Call monitoring/recording (got this working) I know it's a lot to ask and a lot of it is probably documented somewhere (although I couldn't find it in the asterisk manual draft) but like I said I have very little time to decide Thank you for any information you might be able to provide. Tim. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
