[Asterisk-Users] X100P in Switzerland
Hi Does anybody if the X100P works in Switzerland? We can't get a line to PSTN. When I run zttool it shows me always a red alert. I can make and receive calls with an anlog phone plugged in the phone connector. I've compiled and configured the card according to the wiki. Everything seemed to be ok. Is there a way to debug this? Regards Reto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CDR not always correct / IAX clients unmonitored
We are using IAX clients and CDR logging to MySQL. It seems that sometimes calls are
not
correctly logged in the CDR db in terms of the duration of the call.
It maybe related to the fact that all peers show up as "Unmonitored" in CLI ("iax2 show
peers").
Most (or all) our clients run behind NAT if that matters. So far I did not set any
"qualify=XXX" value in iax.conf. When I set it to "qualify=2000" for instance then the
status goes to "UNREACHABLE" but I can still reach my voicemail.
Any hints?
Reto
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Re: [Asterisk-Users] CALLERIDNUM not passed over?
I usually search here: http://www.mail-archive.com/[EMAIL PROTECTED]/ Reto > How do you search? I don't see a search interface to the mailing list > archives. Did I miss something? > > - > Robert Withrow, [EMAIL PROTECTED], +1 978 288 8256, ESN 248 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CALLERIDNUM not passed over?
> [Synopsis]: > Enter voicemail system > > [Description]: > VoiceMailMain([EMAIL PROTECTED]): Enters the main voicemail system > for the checking of voicemail. The mailbox can be passed as the option, > which will stop the voicemail system from prompting the user for the > mailbox. > ->If the mailbox is preceded by 's' then the password check will be > skipped. <--If > a context is specified, logins are considered in that voicemail context > only. > Returns -1 if the user hangs up or 0 otherwise. > > Is that what you want?? Not exactly, sorry if my explanation was capable of being misunderstood. It was not primarily about the password but the extension/mailbox number. It is solved now via callerid in iax.conf like this: callerid="Some Name" <999> I had before only callerid="Some Name" which forces the user to enter his own mailbox number first (and afterwards the password). Thanks Reto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CALLERIDNUM not passed over?
> This one came up a week or so ago on list... please check the archives before
> posting.
Sorry, did a search but obviously not for the right key words ...
> use 's' before the CALLERIDNUM
> ie
> exten => 999,2,VoicemailMain(s${CALLERIDNUM})
The "s" is as I read now for suppressing the password . My problem was that I did not
specify the extension in the callerid:
only
callerid="Some Name"
instead of
callerid="Some Name" <9999>
Thanks anyway!
Reto
> On 03/06/2004 at 14:41 Reto Stauss wrote:
>
> >When a user dials 999 he is always asked for the mailbox and has to enter
> >his mailbox
> >number and password. As I understand this shouldn't happen because the
> >CALLERIDNUM is
> >passed over to VoicemailMain. It's annoying to have to enter the number
> >everytime ...
> >
> >The voice mail configuration is read from MySQL. We are using the CVS
> >version from a few
> >days ago.
> >
> >Extract from extensions.conf:
> >
> >exten => 999,1,wait(1)
> >exten => 999,2,VoicemailMain(${CALLERIDNUM})
> >exten => 999,3,Hangup
> >
> >Any hints?
> >
> >Thanks in advance
> >Reto
> >
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[Asterisk-Users] CALLERIDNUM not passed over?
When a user dials 999 he is always asked for the mailbox and has to enter his mailbox
number and password. As I understand this shouldn't happen because the CALLERIDNUM is
passed over to VoicemailMain. It's annoying to have to enter the number everytime ...
The voice mail configuration is read from MySQL. We are using the CVS version from a
few
days ago.
Extract from extensions.conf:
exten => 999,1,wait(1)
exten => 999,2,VoicemailMain(${CALLERIDNUM})
exten => 999,3,Hangup
Any hints?
Thanks in advance
Reto
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Re: [Asterisk-Users] New Firefly version
Thanks, is working now. Reto Zitat von Adam Hart <[EMAIL PROTECTED]>: > fixed > > Reto Stauss wrote: > > > Adam > > > > The link doesn't seems to work. Get back the following: > > > > Parse error: parse error, unexpected T_STRING in > > /usr/virtual/www.virbiage.com/www/firefly/download/firefly-thirdparty.exe on line > 121 > > > > Reto > > > > > >>There's a new version out with some bugs fixed > >> > >>major ones fixed: deadlock on call end, iax thread getting locked out, > >>few contact group list bugs, one on exit crash bug fixed > >> > >>I'd highly recommend upgrading to it > >>http://www.virbiage.com/firefly/download/firefly-thirdparty.exe > >> > >>-Adam > >> > >>Adam Hart wrote: > >> > >> > >>>As Promised, I've released a new version of Firefly (ver 1.8) with IAX & > >>>SIP support back in. > >>> > >>>Get it from Virbiage site or here's the direct link > >>>http://www.virbiage.com/firefly/download/firefly-thirdparty.exe > >>> > >>>If it crashes on startup, export your Firefly tree from the registry > >>>(current user -> software -> firefly), then delete tree from your > >>>registry. If that fixes it, send me your exported reg file, there's a > >>>bug left to do with some wierd reg entry but everyone just deletes it > >>>instead of sending it to me :| > >>> > >>>Transfers will be in the next version - email me any comments, requested > >>>features, bugs and I'll see what I can do > >>> > >>>-Adam > >>>___ > >>>Asterisk-Users mailing list > >>>[EMAIL PROTECTED] > >>>http://lists.digium.com/mailman/listinfo/asterisk-users > >>>To UNSUBSCRIBE or update options visit: > >>> http://lists.digium.com/mailman/listinfo/asterisk-users > >>> > >> > >>___ > >>Asterisk-Users mailing list > >>[EMAIL PROTECTED] > >>http://lists.digium.com/mailman/listinfo/asterisk-users > >>To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] New Firefly version
Adam The link doesn't seems to work. Get back the following: Parse error: parse error, unexpected T_STRING in /usr/virtual/www.virbiage.com/www/firefly/download/firefly-thirdparty.exe on line 121 Reto > There's a new version out with some bugs fixed > > major ones fixed: deadlock on call end, iax thread getting locked out, > few contact group list bugs, one on exit crash bug fixed > > I'd highly recommend upgrading to it > http://www.virbiage.com/firefly/download/firefly-thirdparty.exe > > -Adam > > Adam Hart wrote: > > > As Promised, I've released a new version of Firefly (ver 1.8) with IAX & > > SIP support back in. > > > > Get it from Virbiage site or here's the direct link > > http://www.virbiage.com/firefly/download/firefly-thirdparty.exe > > > > If it crashes on startup, export your Firefly tree from the registry > > (current user -> software -> firefly), then delete tree from your > > registry. If that fixes it, send me your exported reg file, there's a > > bug left to do with some wierd reg entry but everyone just deletes it > > instead of sending it to me :| > > > > Transfers will be in the next version - email me any comments, requested > > features, bugs and I'll see what I can do > > > > -Adam > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Firefly / LibIAX2
Thanks for the clarification. Can you please tell me how to configure those text messages in *? Thanks in advance Reto > It's the standard LibIAX2, the nice features are implemented using text > messages. I'd recommend you use the standard LibIAX2 as it's more upto > date (Something I've been needing to do too) > > Reto Stauss wrote: > > > Hi > > > > Does anybody know how to build the LibIAX2 from Virbiage? It has some nice features > when > > using Firefly (Messaging, Status Indication). > > > > The source can be downloaded here: http://www.virbiage.com/3rdparty/. It does not > contain > > any directions how to compile. > > > > Any hints? > > > > Thanks! > > Reto > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Firefly / LibIAX2
Hi Does anybody know how to build the LibIAX2 from Virbiage? It has some nice features when using Firefly (Messaging, Status Indication). The source can be downloaded here: http://www.virbiage.com/3rdparty/. It does not contain any directions how to compile. Any hints? Thanks! Reto ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Users in MySQL
> In the sip case it is the consequence of the mode MYSQL_FRIENDS is > implemented. Probably the same with iax2. Thanks for the clarification. > When a sip phone registers, the current IP address and other > parameters get updated in the database. This data will be used if > there is a call to this phone. Perfectly true. Because clients are behind NAT we use IAX. Unfortunately we get always "Rejected connect attempt" and "no authority found" back. Any idea? I double checked the name and secret in the db and the client configuration (using latest firefly dev version). Thanks! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Users in MySQL
I've just compilied th latest CVS of * with USE_MYSQL_FRIENDS enabled ("1"). During
startup * tells me that it connects to the db, so this should be fine.
Nevertheless I don't see any users from the db when I run "sip show users" or "iax2
show
users" although I configured some.
It is also not possible to call them.
Any hints?
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[Asterisk-Users] Problems with IAX configuration
We have a running Asterisk on a small server (RH 9.0) und were able to make calls via
SIP.
Since the quality was rather poor (one of us has only a 150/75 kbps DSL connection,
iLBC
did not work(?)) we tried to setup clients (FireFly) and Asterisk for connections via
IAX.
In some point seems to be a mistake. Asterisk says:
-- Executing Dial("[EMAIL PROTECTED]/2", "IAX2/[EMAIL PROTECTED]") in
new stack
May 24 16:56:27 NOTICE[1133718080]: chan_iax2.c:4700 socket_read: Host 217.162.123.197
failed to authenticate as 83022777
May 24 16:56:27 WARNING[1133718080]: chan_iax2.c:4534 socket_read: Call rejected by
217.162.123.197: No authority found
May 24 16:56:27 NOTICE[1133718080]: chan_iax2.c:1038 iax2_destroy: Avoiding IAX destroy
deadlock
-- Called [EMAIL PROTECTED]
-- Hungup 'IAX2[217.162.123.197:4569]/4'
== No one is available to answer at this time
May 24 16:56:37 WARNING[1209214400]: pbx.c:1836 ast_pbx_run: Timeout, but no rule 't'
in
context 'from-iax'
-- Hungup '[EMAIL PROTECTED]/2'
externsions.conf looks like this:
[from-iax]
exten => 83022266,1,Dial(IAX2/[EMAIL PROTECTED])
exten => 83022266,3,Hangup
exten => 83022777,1,Dial(IAX2/[EMAIL PROTECTED])
exten => 83022777,3,Hangup
I guess the problem lies in extensions.conf but was not able to figure out the correct
seeting.
Any hints?
Thanks
Reto
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