Re: [Asterisk-Users] This mailing list is being spam filtered on my site.
On Thu, June 16, 2005 3:26, Gary Guthary said:
Sorry if this not the right place to post this BUT...
Since May 31st, ALL of these user list messages have been filtered by
spamassassin running on my Linux box. - Claim to be listed in Bayes as
spam. - Have no clue why this is happening.
Luckily, spamassassin sent the messages to the probably-spam folder on
the Linux box I was able to retrieve them.
If anybody else is having this problem **AND** us using procmail (along
with spamassassin) on a *NIX box, put the following three lines in the
top
of your .procmailrc file:
:0
^To:[EMAIL PROTECTED]
${DEFAULT}
Note: - That's a zero in the first line.
This will allow delivery until somebody can figure out why this Bayesian
filtering is happening and can get it stopped.
If anybody wants to contact me off-list to discuss, email:
[EMAIL PROTECTED]
Gary Guthary
Check whether Bayes filter is set for auto-learn. It has somehow aquired
enough keywords from this list to mark the emails from here as SPAM. I do
not know which filter you use, but the SpamAssassin built in Bayesan
allows for 'HAM' (ie NON-SPAM) mails to be learnt... Try collecting a
weeks worth of list mails and then have the filter scan them (look for
sa-learn) as 'HAM'...
Good luck!
--
Francesco Peeters
GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704
If your program doesn't recognize my signature, please visit
http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate.
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Re: [Asterisk-Users] newbie question..
hi Rich,
thanks for ur help..
it works..
i have found another way,
_9XXX,1,Dial(Zap/4/1800XX,5,D(${EXTEN}))
D = will send dtmf
thank a lot Rich..
best regard,
shahdan
--- Rich Adamson [EMAIL PROTECTED] wrote:
the situation here is i want when user make
outgoing
call, asterisk will call 1800XX first then
after 3
or 4 sec asterisk will insert the number that user
want to call..
user don't know that the call is go to 1800XX
first..
means user just insert the number that they want
to
call then asterisk will insert that number after 3
or
4 sec..
can i that in asterisk?
i'll apreciate any help or advise..
Might try something like this:
exten =
_9XXX,1,Dial(Zap/4/1800XXw${EXTEN})
where each w adds a some delay.
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RE: [Asterisk-Users] iax2 can't listen on virtual interface
Yes you can. Just tell iax to bind to that virtual address in iax.conf -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lance Grover Sent: Thursday, 16 June 2005 14:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] iax2 can't listen on virtual interface Can anyone shed some light on this, I have two asterisk boxes using heartbeat for failover. Sip traffic works just fine with the virtual IP but IAX does not. For example on my servers one server has the following: eth0 = 192.168.1.95 eth0:0 = 192.168.1.2 the other server has: eth0 = 192.168.1.220 if the first Master server goes down the second server will take that virtal IP for it's eth0:0 but in either case the IAXY phones cannot connect to this floating virtual IP but can connect to either of the regular interfaces IPs. Please let me know if I am incorrect or if ther is something I can do. -- Thanks, Lance Grover ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cheap Asterisk FXO PCI cards
Hi, Does anybody know a website or company where I can buy cheap Asterisk and SIP compatible PCI cards that have 2, 3 or 4 FXO ports? Digium cards that have 2 or more FXO ports work great, but they are a bit over my budget at the moment. I have found digium compatible clone cards on the internet that are cheap, but haven't found any that have more than 1 port. Any help would be appreciated. Regards, -- Ing CIP Alejandro Celi Maritegui [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nasty little incident ...
On Wed, 15 Jun 2005, Rich Adamson wrote: Just a wild guess When the two meridian links disappeared, the channel numbers probably changed. Instead of channels 1 through 124, you probably have channels 1 through 62 and your supporting dialplan (and other channel specific items) likely don't match. No - just because the span goes doesn't mean its gone. Its simply there and down. Here's my theory as to the problem: In the config, Spans 1 and 3 are to the telco, 2 and 4 to the old pbx. Clocking is being taken from spans 1 and 3. Now the symptom when the meridian was disconnected was like zaptel had no clock. So theory one is that the spans are actually plugged into the board upside down, with telco on 4 and 2, meridian on 3 and 1. so when the meridian was disconnected there was no more clock. Second theory is that the zaptel.conf was changed - maybe moving which spans clock comes from - and the zaptel modules weren't reloaded or ztcfg wasn't run. Port 1 on the TE410P is at the top (away from the mobo), btw. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Integration with an SBC-410 phone system
Hi, I am new to the world of Asterisk PBX. I have been given the task of coming up with a solution to our office phone situation. From what I have been reading, Asterisk sounds like it could be ideal. However, most of the information I am finding is focused on the VoIP aspects, which is something we want to use for our remote employees, but the first hurdle is integrating Asterisk with the SBC-410 phone system. I know the system says that it can be used with a PBX, but I haven't found any info from that side of the street, and the Asterisk info on analog integration is scattered. So, if anyone has any experience meshing asterisk with a SBC-410 or similar type of small office phone setup, I'd be happy to hear from you. Best Regards, J Scott Pitman ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] How to stop Asterisk from changing the SDP?
I'm trying to set up a direct SIP connection and have Asterisk stay
out of the media stream. When I look at the INVITE messages, I see
that Asterisk is changing the Session Description Protocol in the
INVITE message it receives, and send a INVITE message with a different
SDP to the receiver. This is not what I want. Is there any way to make
Asterisk leave the SDP exactly like it is sent from the sender?
I have set canreinvite=yes on both participants and my dialingplan is simply:
exten = _.,1, Dial(SIP/${EXTEN},20)
and NAT is not a problem
Thanks.
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Re: [Asterisk-Users] Cheap Asterisk FXO PCI cards
If you want to go into a serious IP telephony project using Asterisk (as oppoosed to an experiment or toy system), I recommend you to buy a real Digium card; i.e. the TDM400 mainframe series. Believe me, I have 2 of the clones sitting in my shelf gathering dust as the voice quality is simply atrocious, especially in a non-US telco environment (of course YMMV). Furthermore, the act of buying hardware from Digium is a surest means of supporting the coders for such a great GPLed product that Asterisk is. And great is an understatement. On 6/16/05, Ing CIP Alejandro Celi Maritegui [EMAIL PROTECTED] wrote: Hi, Does anybody know a website or company where I can buy cheap Asterisk and SIP compatible PCI cards that have 2, 3 or 4 FXO ports? Digium cards that have 2 or more FXO ports work great, but they are a bit over my budget at the moment. I have found digium compatible clone cards on the internet that are cheap, but haven't found any that have more than 1 port. Any help would be appreciated. Regards, -- Ing CIP Alejandro Celi Maritegui [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_capi-cm-0.5 release announcement
Hi all, I would like to announce the first release of the chan_capi channel driver on sourceforge.net The package is available for download with name chan_capi-cm-0.5 and is the current CVS HEAD. It is derived from the chan_capi-0.4.0PRE1 of kapejod. The main changes are: - complete rework - fix race-conditions - fix call state handling - rework of debug/verbose messages - added capiFax feature (provided by Frank Sautter) - auto-config (compile and work with Asterisk CVS-HEAD and older versions) - use with ELinOS cross-toolbox and project handling For the versioning, I have decided to use the name extention 'cm' to avoid confusion with kapejod's version. This first release is 0.5 (not 0.1) because the base is 0.4.0. Only the major and the minor number will be used. The exception to have a third number (patch-version) will be added for fixup-patches only. Feedback welcome. Armin PS: sorry for cross-posting. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: iax2 can't listen on virtual interface
In article [EMAIL PROTECTED], Boris Bakchiev [EMAIL PROTECTED] wrote: Yes you can. Just tell iax to bind to that virtual address in iax.conf I don't think that will work on the box that doesn't currently own that virtual address. I think the only way is to make sure the bind address is 0.0.0.0 If you've already done that and it still doesn't work, then I don't know, sorry. Cheers Tony -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lance Grover Sent: Thursday, 16 June 2005 14:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] iax2 can't listen on virtual interface Can anyone shed some light on this, I have two asterisk boxes using heartbeat for failover. Sip traffic works just fine with the virtual IP but IAX does not. For example on my servers one server has the following: eth0 = 192.168.1.95 eth0:0 = 192.168.1.2 the other server has: eth0 = 192.168.1.220 if the first Master server goes down the second server will take that virtal IP for it's eth0:0 but in either case the IAXY phones cannot connect to this floating virtual IP but can connect to either of the regular interfaces IPs. Please let me know if I am incorrect or if ther is something I can do. -- Thanks, Lance Grover ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Live! CF
Bob Goddard wrote: Got some proof of that? It's generally regarded as common knowlege in these circles that the via processors claim 686 compatibility but lack some 686-specific instructions (CMPXCHG among them), and this is what causes the trouble. GCC says 686 instructions, ok. and the Via throws a fit (SIGILL) when seeing the ones it doesn't support. The Via C3 processors lack the CMPXCHG8B (CMOV) instructions and I assume others which are listed in the Intel documents as being optional. GCC assumes that they are always there. Look at http://radagast.bglug.ca/epia/epia_howto/x1098.html, section 13.2. This has been well documented. Bob is correct in this. However, the C3 processors comes in different flavours. The ones using the Nehemiah core do have the optional instructions, and runs 686 binaries fine. -- Torgeir Berg ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER with Asterisk Problem
Dear All,
I am trying to make my sip phones register with SER and make use of
Asterisk capabilities such as voicemail and parking calls for example.
on SER side
the ip of the server is 192.168.99.170 and uses port 5060
in my ser.cfg I added the following lines :
if (uri=~"sip:[EMAIL PROTECTED]") {
rewritehostport("10.3.26.2:5090");
t_relay();
break;
}
all my sip phones can register to ser without passwords.
On the Asterisk side:
the ip is 10.3.26.2 and uses port 5090
in my sip.conf I added:
register => 10:[EMAIL PROTECTED]/10
[sip-ser}
type=friend
user=10
userfrom=10
host=192.168.99.170
Now My problem :
1- the asterisk console shows failed messages to register to the ser
(Forbidden - wrong password authentication)
--
Thx
MAG
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[Asterisk-Users] Asterisk on Fedora Cora 3
I have problems monitorino cpu loading about Asterisk process on Fedora Cora 3. If Asterisk starts at stratup it will be recognise with 14 processes and if it start normaly in terminal top dont give me any activity about is work. Can anyone help me? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to stop Asterisk from changing the SDP?
On Thu, 16 Jun 2005, Stian Selnes wrote:
I'm trying to set up a direct SIP connection and have Asterisk stay
out of the media stream. When I look at the INVITE messages, I see
that Asterisk is changing the Session Description Protocol in the
INVITE message it receives, and send a INVITE message with a different
SDP to the receiver. This is not what I want. Is there any way to make
Asterisk leave the SDP exactly like it is sent from the sender?
I have set canreinvite=yes on both participants and my dialingplan is simply:
exten = _.,1, Dial(SIP/${EXTEN},20)
and NAT is not a problem
Hi,
Asterisk isn't a SIP proxy. And here is an example of where the
difference shows.
You should probably look at using SER for this SIP stuff and only send
calls to Asterisk where necessary (treat Asterisk like a pstn gateway or
sip service box).
Steve
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RE: [Asterisk-Users] phantom answer
Title: Message
All,
Got it
working. Turned out to the cable between the out port on the tdm400 and the
telephone wall socket. It appears that it requires a cable that you would
ordinarily get with a modem. e.g. two wires (red green) with the red wire
on the right if you look at the rj11 with the lever at the top (or the red cable
on the left if look from the bottom of the rj 11 plug , with the copper pins
exposed)
Hope
this helps someone.
D.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]Sent: 15 June 2005 20:08To:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] phantom
answer
People,
My goal is to get
asterisk dialing out via my landline (POTS) from a sip softphone. Ive got the
phone, The TDM400p is installed and working. (See below) When ever I dial a
number that is directed to the outgoing port on my card (fxs/fxo?) I get no
ringing, then it claims its been answered. the CLI reports the
following:
Executing
Dial("SIP/301-f97a", "Zap/4/01614299100|20") in new
stack -- Called 4/01614299100 --
Zap/4-1 answered SIP/301-f97aJun 15 17:57:38 NOTICE[11121]: rtp.c:277
process_rfc3389: Comfort noise support incomplete in Asterisk (RFC
3389). Please turn off on client if possible. Client IP:
192.168.0.7 -- Hungup 'Zap/4-1'
Anyone Any Ideas?
BTW Apologies for the disclaimer at the bottom, but the mail server adds it on
by default and there's nothing I can do about it.
*CLI zap show
channels Chan Extension
Context Language
MusicOnHoldpseudo
default
1
default
default
4
incoming
default*CLI
This is the
important bit from zapata.conf
; DYLAN ADDED FROM DIGIUM.COM
echocancel=yes ; You can set this to 32,
64, or 128, tweak to your
needs.echocancelwhenbridged=yesechotraining=yes ; Asterisk trains to
the beginning of the call, number is in
millisecondscallerid=01614830073signalling=fxo_ksgroup=1context=default
; Points to the default context of your extensions.confchannel =
1
signalling=fxs_ks;callerid=asreceivedgroup=2context=incomingchannel=
4; END OF DYLAN ADDED FROM DIGIUM.COM *
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Re: [Asterisk-Users] How to stop Asterisk from changing the SDP?
Hi.
Tanks for your answer. So I understand you correct if you mean that
there is no way to let asterisk leave the SDP untouched? I have tried
SER, and I just wanted to look at the possibilities that Asterisk
offered. A bit dissapointing that it doesn't satisfy my needs :-)
- Stian
On 6/16/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
On Thu, 16 Jun 2005, Stian Selnes wrote:
I'm trying to set up a direct SIP connection and have Asterisk stay
out of the media stream. When I look at the INVITE messages, I see
that Asterisk is changing the Session Description Protocol in the
INVITE message it receives, and send a INVITE message with a different
SDP to the receiver. This is not what I want. Is there any way to make
Asterisk leave the SDP exactly like it is sent from the sender?
I have set canreinvite=yes on both participants and my dialingplan is
simply:
exten = _.,1, Dial(SIP/${EXTEN},20)
and NAT is not a problem
Hi,
Asterisk isn't a SIP proxy. And here is an example of where the
difference shows.
You should probably look at using SER for this SIP stuff and only send
calls to Asterisk where necessary (treat Asterisk like a pstn gateway or
sip service box).
Steve
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[Asterisk-Users] Asterisk Problems with FXO Ground Start Trunks and DID Wink Start Trunks
I have the following configuration: Stable Asterisk running on a Dell PowerEdge 800 with Enterprise 3 Redhat: Digium TE110P card, connected to a Adtran TA 750 Telco IF: 4 analog DID loop start wink lines, connected to the Adtran FXS card in DPO mode 4 combo analog ground start trunks, connected to the Adtran FXO card in Ground Start Mode. The telco lines and the Adtran channel bank are working. The Digium TE110P card seems to be working also. I can see the bits on the zttool based on changes on the telco lines. All the bits from the Adtran all are correct. However Asterisk does not seem to be setting the correct bits from the software. Problems: 1. Asterisk is not recognizing the incoming DID calls. The CAS bits showing on zttool are correct for the incoming calls, however, Asterisk does not come back with a wink acknowledge. 2. We can receive calls on the FXO ground start channels. However, outbound calls are not working. From the Zttool the idle bits are set fine. However, Asterisk is not setting the correct CAS bits for ground start signaling on a outgoing call on the FXO channel. Has anyone experienced this problem? I have zaptel.conf configured for: span=1,0,0,esf,b8zs fxsgs=1-4 em=5-8 loadzone=us defaultzone=us Zapata.conf is configured as: [trunkgroups] [channels] context=default switchtype=national wink=300 rxwink=300 usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 callgroup=1 pickupgroup=1 immediate=no group=1 signaling=fx_gs context=external channel=1-4 group=2 signaling=em_w context=directindial channel=5-8 Syed Akbar Alico Systems Inc www.alicosystems.com Tel: 562-436-1510 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Help with Cron and Reload
Another note regarding a related issue: On Wed, Jun 15, 2005 at 02:47:00PM -0400, Federico Alves wrote: This will sound weird but the command 'asterisk -r -x reload' fails to work when issued by Cron. But it works when I issue it from a bash session. What is not configured correctly? I need to refresh the configuration every a short amount of time. rom [EMAIL PROTECTED] Wed Jun 15 18:42:00 2005 Date: Wed, 15 Jun 2005 18:42:00 -0400 From: [EMAIL PROTECTED] (Cron Daemon) To: [EMAIL PROTECTED] Subject: Cron [EMAIL PROTECTED] asterisk -r -x reload X-Cron-Env: SHELL=/bin/sh X-Cron-Env: HOME=/root X-Cron-Env: PATH=/usr/bin:/bin X-Cron-Env: LOGNAME=root /bin/sh: line 1: asterisk: command not found /path/to/asterisk , as others have noted (or manually set PATH) However, if you leave asterisk in debug/verbose (verbose =2? ) mode, cron jobs will start sending emails, because the command has generated an output. You could do something like: /usr/sbin/asterisk -r -x reload | egrep -v '^(Core debug|Verbosity) is at least [0-9]+' (Which is basically what I currentlly use in the wrapper for asterisk -rx on Rapid) But then the return status will be the return status of grep, which is not exactly what you want. cron will always send an email: if there is output: to give you the output. If there is no output: grep returns 1. For the record, the current init.d script on Debian simply runs $DAEMON -rx 'reload' on the command 'reload' Any ideas? -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Grandstream phones losing registration with server.
Hi Everyone, Im using Asterisk, actually [EMAIL PROTECTED] 1.1 with all Grandstream 102 phones. NAT is not an issue as all including the server have public IPs The problem is that the phones keep losing registration with the server. I have not timed this exactly to see if they drop off with exactly the same frequency. The SIP TRUNK connection to my provider SIPGATE does not lose registration, and neither does a Grandstream 2000 connecting in as an external extension on a public IP on a different network; lose registration. The problem is that I am not always on site where the server is located so have to keep rebooting the phones remotely so it does not affect the users. They panic when asked to do anything remotely technical even like unplugging the phone and re-plugging it to reboot, lol Is there anything that you could recommend I do to stop the phones from losing registration with the server? Cheers Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to stop Asterisk from changing the SDP?
On Thu, 16 Jun 2005, Stian Selnes wrote: Hi. Tanks for your answer. So I understand you correct if you mean that there is no way to let asterisk leave the SDP untouched? I have tried SER, and I just wanted to look at the possibilities that Asterisk offered. A bit dissapointing that it doesn't satisfy my needs :-) - Stian No - for a particular application you can probably get the SDP stuff going out of Asterisk similar enough to the incoming for what you need. This will be by adjusting the codec allows and disallows. But Asterisk's core does not think in terms of sdp descriptors and so forth. An incoming SIP invite becomes an incoming call into the Asterisk core, which via the dialplan may become an outgoing call to some other sip peer. There's no connection between the incoming sip channel and the outgoing one from the point of view of the SIP channel driver. Now I can't say whether Asterisk can satisfy your needs, seeing you didn't say what the needs are beyond don't touch the SDP. If that is really what you need, Asterisk as it is can't satisfy them. But perhaps if you write the actual problem you are trying to solve you'll find that Asterisk can meet that! Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] MeetMe ERROR Unable to dup channel
I would us Meetme for conferance SIP--SIP fist. my Meetme.conf: [rooms] conf = my extensions.conf: exten = ,1,MeetMe() But : == Parsing '/etc/asterisk/meetme.conf': Found Jun 16 10:33:22 WARNING[12100]: chan_zap.c:916 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Jun 16 10:33:22 ERROR[12100]: chan_zap.c:6969 chandup: Unable to dup channel: No such file or directory Jun 16 10:33:22 WARNING[12100]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Jun 16 10:33:22 WARNING[12100]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') I don't unederstand because i don't use zap channel. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: iax2 can't listen on virtual interface
He is using HA so I'm assuming he is running Master-Slave combo. That means HA will start asterisk on slave after taking over the IP and becoming a master. Until that time, asterisk does not need to be running on a slave so there should be no problems whatsoever. If he wants to run asterisk in Master-Master that is a different story but probably not what you want. Even then it is possible. When becoming a master just script HA to unload chan_iax, assume the virtual IP, substitute the bindip in iax.conf (sed will do just fine) and then load chan_iax backup again. All that can be done while asterisk is still running. Regards -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Thursday, June 16, 2005 5:19 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Re: iax2 can't listen on virtual interface In article [EMAIL PROTECTED], Boris Bakchiev [EMAIL PROTECTED] wrote: Yes you can. Just tell iax to bind to that virtual address in iax.conf I don't think that will work on the box that doesn't currently own that virtual address. I think the only way is to make sure the bind address is 0.0.0.0 If you've already done that and it still doesn't work, then I don't know, sorry. Cheers Tony -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Lance Grover Sent: Thursday, 16 June 2005 14:53 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] iax2 can't listen on virtual interface Can anyone shed some light on this, I have two asterisk boxes using heartbeat for failover. Sip traffic works just fine with the virtual IP but IAX does not. For example on my servers one server has the following: eth0 = 192.168.1.95 eth0:0 = 192.168.1.2 the other server has: eth0 = 192.168.1.220 if the first Master server goes down the second server will take that virtal IP for it's eth0:0 but in either case the IAXY phones cannot connect to this floating virtual IP but can connect to either of the regular interfaces IPs. Please let me know if I am incorrect or if ther is something I can do. -- Thanks, Lance Grover ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This message (and any associated files) is intended only for the use of the individual or entity to which it is addressed and may contain information that is confidential, subject to copyright or constitutes a trade secret. If you are not the intended recipient you are hereby notified that any dissemination, copying or distribution of this message, or files associated with this message, is strictly prohibited. If you have received this message in error, please notify us immediately by replying to the message and deleting it from your computer. Messages sent to and from us may be monitored... Internet communications cannot be guaranteed to be secured or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Therefore, we do not accept responsibility for any errors or omissions that are present in this message, or any attachment, that have arisen as a result of e-mail transmission. If verification is required, please request a hard-copy version. Any views or opinions presented are solely those of the author and do not necessarily represent those of the company. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] How to stop Asterisk from changing the SDP?
What I'm actually trying to do is to send video over SIP. The video codecs I would like to use is H.261, H.262 or H.264. I can see from typing show codecs in the CLI that Asterisk supports H.261 and H.263. I guess this means that if I set Asterisk to disallow all but these two codecs, I'm able to get the video through? But the video stream then would go via Asterisk and not directly between the two video applications? On 6/16/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: On Thu, 16 Jun 2005, Stian Selnes wrote: Hi. Tanks for your answer. So I understand you correct if you mean that there is no way to let asterisk leave the SDP untouched? I have tried SER, and I just wanted to look at the possibilities that Asterisk offered. A bit dissapointing that it doesn't satisfy my needs :-) - Stian No - for a particular application you can probably get the SDP stuff going out of Asterisk similar enough to the incoming for what you need. This will be by adjusting the codec allows and disallows. But Asterisk's core does not think in terms of sdp descriptors and so forth. An incoming SIP invite becomes an incoming call into the Asterisk core, which via the dialplan may become an outgoing call to some other sip peer. There's no connection between the incoming sip channel and the outgoing one from the point of view of the SIP channel driver. Now I can't say whether Asterisk can satisfy your needs, seeing you didn't say what the needs are beyond don't touch the SDP. If that is really what you need, Asterisk as it is can't satisfy them. But perhaps if you write the actual problem you are trying to solve you'll find that Asterisk can meet that! Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Busy, differences between SIP and Zaptel(bristuff)
Hi all,
a lot of my snoms are being called with this macro:
[macro-ohne-AB]
exten = s,1,DBget(temp=UML/${ARG1})
exten = s,2,Goto(default|${temp}|1)
exten = s,3,Dial(${ARG2},600,g)
exten = s,4,SetVar(PRI_CAUSE=17)
exten = s,5,Hangup
[default]
...
exten = 77,1, Macro(ohne-AB,77,SIP/snom8556)
...
When a call comes over QuadBRI in and the called phone is Busy the caller gets a Busy.
That is fine. When another snom is calling a busy snom, then it gets an
forbidden.
When i change Hangup to Busy the call snom to busy snom is OK. Incoming
ISDN calls get silence, then after 10 seconds an congestion. That is ugly.
What is the right way to make a busy for Incoming calls (QuadBRI) and internal
calls (SIP to SIP).
Best reagards
Thomas
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[Asterisk-Users] Error on incoming calls
I've a standard debian asterisk installation, with a Juhngans quadBRI ISDN board. Sometimes, at unpredictable moments, incoming calls are not answered by the asterisk server, with the caller hearing only silence, and the only message I can find about this in the logs is the following: PRI: received SETUP message for call that is not a new call, wicked!!! I've done researchs on the net and I've found other people having the same problems, but nobody ever answered them... somebody can suggest me, if not where the problem lies, at least where to start to search where the problem is? Thanks to everybody in advance... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Features.conf Set Language
I use features.conf in order to park call, but I would like use french speaker. how set langage in features.conf? Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] MeetMe ERROR Unable to dup channel
On Thu, Jun 16, 2005 at 10:46:30AM +0200, sylvain garcia wrote: I would us Meetme for conferance SIP--SIP fist. my Meetme.conf: [rooms] conf = my extensions.conf: exten = ,1,MeetMe() But : == Parsing '/etc/asterisk/meetme.conf': Found Jun 16 10:33:22 WARNING[12100]: chan_zap.c:916 zt_open: Unable to open '/dev/zap/pseudo': No such file or directory Jun 16 10:33:22 ERROR[12100]: chan_zap.c:6969 chandup: Unable to dup channel: No such file or directory Jun 16 10:33:22 WARNING[12100]: app_meetme.c:227 build_conf: Unable to open pseudo channel - trying device Jun 16 10:33:22 WARNING[12100]: app_meetme.c:230 build_conf: Unable to open pseudo device -- Playing 'conf-invalid' (language 'en') I don't unederstand because i don't use zap channel. http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+MeetMe The MeetMe application needs a timer to work. There are different ways to get the timer to work, but it won't work by default if you haven't got a Digium Zaptel hardware interface card installed. At this time only zaptel devices may be used. If you do not have a Zaptel device see the ztdummy instructions for timing. i guess you don't have digium card nor using ztdummy or such module for timing. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Grandstream phones losing registration with server.
On Thursday 16 Jun 2005 09:25, Mark Brown wrote: Hi Everyone, I'm using Asterisk, actually [EMAIL PROTECTED] 1.1 with all Grandstream 102 phones. NAT is not an issue as all including the server have public IP's The problem is that the phones keep losing registration with the server. I have not timed this exactly to see if they drop off with exactly the same frequency. The SIP TRUNK connection to my provider SIPGATE does not lose registration, and neither does a Grandstream 2000 connecting in as an external extension on a public IP on a different network; lose registration. The problem is that I am not always on site where the server is located so have to keep rebooting the phones remotely so it does not affect the users. They panic when asked to do anything remotely technical even like unplugging the phone and re-plugging it to reboot, .. lol Is there anything that you could recommend I do to stop the phones from losing registration with the server? Is this what is happening? 1. Phone rings but does not answer 2. Phone reregisters as normal (after an hour?) 3. Phone rings but does not answer 4. Phone fails to register If so, then it is a known problem. Grandstream know about it and should have put the fix into 1.0.6.7 but they mucked up. It should be fixed in the next release. This only seems to happen with the BT10x phones. What I have done to alleviate it is to set the Register Expiration on the phone to some large value. It's not perfect, but it helps. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_h323 context
Hi all, All incoming H.323 calls on chan_h323 were forwarded to default context but not detroit. It seems context=detroit is not effective. Any helps??? [det-gw] type=h323 prefix=1248,1313 context=detroit Thanks. IM ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This mailing list is being spam filtered on my site.
On Thursday 16 June 2005 02:01, Francesco Peeters wrote: Check whether Bayes filter is set for auto-learn. It has somehow aquired enough keywords from this list to mark the emails from here as SPAM. I do not know which filter you use, but the SpamAssassin built in Bayesan allows for 'HAM' (ie NON-SPAM) mails to be learnt... Try collecting a weeks worth of list mails and then have the filter scan them (look for sa-learn) as 'HAM'... I too am seeing this and I've been using SA for YEARS. I've been trying to train it but some of the messages to this list just do not want to be classified as non-spam. Im trying to get them to come out clean without resorting to a whitelist. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Grandstream phones losing registration withserver.
On Thursday 16 Jun 2005 09:25, Mark Brown wrote: Hi Everyone, I'm using Asterisk, actually [EMAIL PROTECTED] 1.1 with all Grandstream 102 phones. NAT is not an issue as all including the server have public IP's The problem is that the phones keep losing registration with the server. I have not timed this exactly to see if they drop off with exactly the same frequency. The SIP TRUNK connection to my provider SIPGATE does not lose registration, and neither does a Grandstream 2000 connecting in as an external extension on a public IP on a different network; lose registration. The problem is that I am not always on site where the server is located so have to keep rebooting the phones remotely so it does not affect the users. They panic when asked to do anything remotely technical even like unplugging the phone and re-plugging it to reboot, .. lol Is there anything that you could recommend I do to stop the phones from losing registration with the server? Is this what is happening? 1. Phone rings but does not answer 2. Phone reregisters as normal (after an hour?) 3. Phone rings but does not answer 4. Phone fails to register If so, then it is a known problem. Grandstream know about it and should have put the fix into 1.0.6.7 but they mucked up. It should be fixed in the next release. This only seems to happen with the BT10x phones. What I have done to alleviate it is to set the Register Expiration on the phone to some large value. It's not perfect, but it helps. B ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users They are all BT102 phones, they seem to register fine and work fine for a while then just drop off the server. Have now tried setting the Register Expiration on the phone and tweaked a few server settings as well. Will keep you updated. Thanks for that though.. Any idea when the new Firmware will be released? Already got the 1.0.6.7 Mark ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zaphfc unable to dial out
Im using bristuff-0.2.0-RC8g with two HFC-PCI controllers. Inbound calls work just fine, but when im dialing out asterisk shows: -- Executing Macro(SIP/8010-20a1, dial_out|xxx) in new stack -- Executing Answer(SIP/8010-20a1, ) in new stack -- Executing SetCallerID(SIP/8010-20a1, xx) in new stack -- Executing Dial(SIP/8010-20a1, ZAP/g1/xx) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/xx -- Zap/1-1 is ringing The operator then tells me The number your calling is not used. I've tried different numbers and different prefixes. Outdialing works fine with my old chan_capi config with a fritz pci card. Frits van Tiel___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi IP Phones
Dean. I think We are starting the other way around on this. First We need to find if there are such SIP Phones out there and look at specs then I can go to my client and ask all these questions you just listed. No use asking all of these questions if there aren't any such phones out there to start with. The client specifically requested wifi sip phones. |-Original Message- |From: Dean Collins [mailto:[EMAIL PROTECTED] |Sent: Mircoles, 15 de Junio de 2005 04:10 p.m. |To: Asterisk Users Mailing List - Non-Commercial Discussion |Cc: [EMAIL PROTECTED] |Subject: RE: [Asterisk-Users] WiFi IP Phones | |Anton, did you even read my post? | |If you don't know what you are doing then you don't deserve to |be a distributor in this space. | |How are you going to isolate the access points? |What rating zones will you need to meet? |What existing arrestors are in place? (gas, fibre, etc) | |In addition, what ranges are they expecting from the access points? |What physical environments are the signals going to be |operating in? (steel etc) What 2.4 ghz interference can be |expected from other emitting industrial equipment. |What physical environment issues need to be considered |condensation, salination, operating temps etc. |Will these wifi handsets be a workers only form of |communication or is there a backup dead mans pager/radio alert |systems (occupational health and safety issues etc) | |There's a reason why intrinsic costs money and it's got a lot |more to do with safety than will the satellite signal work well. | |Cheers, |Dean | | | | | -Original Message- | From: [EMAIL PROTECTED] |[mailto:asterisk-users- | [EMAIL PROTECTED] On Behalf Of Anton Krall | Sent: Wednesday, 15 June 2005 4:53 PM | To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - |Non-Commercial | Discussion' | Subject: RE: [Asterisk-Users] WiFi IP Phones | | I know... The term anti explosive is new to me.. I never |heard of it | but a possible client is asking for that exactly since the |phones are | going to be used in oil refinary and rd platforms using voip over | satelite connections... | | What do you think? | | BTW, how is voip over satelite? I know you have the usual 500 ms lag | for up and down stream delays but hows quality? | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On Behalf Of Cory | |Andrews | |Sent: Mircoles, 15 de Junio de 2005 02:16 p.m. | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] WiFi IP Phones | | | |Anton - if you had a large opportunity and wanted a manufacturer to | |certify the phones as anti-explosive, I know a few that would | |probably attest to their phones being anti explosive as long as | |there was no major liability involved. | | | |I do not see anti explosive listed in any of the technical | |specifications of WLAN phones made by | | | |Zyxel | |Hitachi | |UTStarCom | |Uniden | |Cisco | |Net2Com | | | |Cory Andrews | |Purchasing / EVP | |VOIPSupply.com | |v - 716.630.1555 X22 | |e - [EMAIL PROTECTED] | | | | | | | |Anton Krall wrote: | | | |Guys. | | | |I know there are wifi sip phones out there but I have a question, | |are any of these phones anti explosive? By that I mean, there | |are certain | |regulations about phones or cel phones that are not recommended to | |operate in environments like gas stations due to sparks and | |the chance | |of ingiting gas fumes. | | | |Are there any wifi sip phones out here that have complaince with | |regulations to operate in hazardous environments like Oil |Platforms, | |etc? phones denominated anti explosive or something? | | | |___ | |Asterisk-Users mailing list | |Asterisk-Users@lists.digium.com | |http://lists.digium.com/mailman/listinfo/asterisk-users | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | | | | | | |___ | |Asterisk-Users mailing list | |Asterisk-Users@lists.digium.com | |http://lists.digium.com/mailman/listinfo/asterisk-users | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | ___ | Asterisk-Users mailing list | Asterisk-Users@lists.digium.com | http://lists.digium.com/mailman/listinfo/asterisk-users | To UNSUBSCRIBE or update options visit: |http://lists.digium.com/mailman/listinfo/asterisk-users | | | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi IP Phones
Guys.. We havent even started taking about costs here. Velieve me, this customer is not concerned about costs ... At least not yet... He just wants to know if there are any IS wifi phones compatible with asterisk out there. You are thinking too fast too soon.. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Terry H. Gilsenan |Sent: Mircoles, 15 de Junio de 2005 06:58 p.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] WiFi IP Phones | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] On Behalf Of Dean | Collins | Sent: Thursday, 16 June 2005 7:10 AM | To: Asterisk Users Mailing List - Non-Commercial Discussion | Subject: RE: [Asterisk-Users] WiFi IP Phones | | Anton, did you even read my post? | | If you don't know what you are doing then you don't deserve to be a | distributor in this space. | | How are you going to isolate the access points? | What rating zones will you need to meet? | What existing arrestors are in place? (gas, fibre, etc) | | In addition, what ranges are they expecting from the access points? | What physical environments are the signals going to be operating in? | (steel etc) What 2.4 ghz interference can be expected from other | emitting industrial equipment. | What physical environment issues need to be considered condensation, | salination, operating temps etc. | Will these wifi handsets be a workers only form of |communication or is | there a backup dead mans pager/radio alert systems (occupational | health and safety issues etc) | | There's a reason why intrinsic costs money and it's got a |lot more to | do with safety than will the satellite signal work well. | | Cheers, | Dean | | |Thanks Dean, I was trying to communicate this to him also. I |manage IT/IS/Comms for an Oil company[1] and we have equipment |installed at our refinery and at our drillsites. | |IS is a fundamental building block of all our work, and yes it |does increse the cost by a factor of at least 3 and more |likely 10 when all the other costs are taken in to account. | |[1]http://www.interoil.com/ | | | | | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:asterisk-users- | [EMAIL PROTECTED] On Behalf Of Anton Krall | Sent: Wednesday, 15 June 2005 4:53 PM | To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - | Non-Commercial | Discussion' | Subject: RE: [Asterisk-Users] WiFi IP Phones | | I know... The term anti explosive is new to me.. I never | heard of it | but a possible client is asking for that exactly since the | phones are | going to be used in oil refinary and rd platforms using voip over | satelite connections... | | What do you think? | | BTW, how is voip over satelite? I know you have the usual | 500 ms lag | for up and down stream delays but hows quality? | | |-Original Message- | |From: [EMAIL PROTECTED] | |[mailto:[EMAIL PROTECTED] On |Behalf Of Cory | |Andrews | |Sent: Mircoles, 15 de Junio de 2005 02:16 p.m. | |To: Asterisk Users Mailing List - Non-Commercial Discussion | |Subject: Re: [Asterisk-Users] WiFi IP Phones | | | |Anton - if you had a large opportunity and wanted a | manufacturer to | |certify the phones as anti-explosive, I know a few that would | |probably attest to their phones being anti explosive as long as | |there was no major liability involved. | | | |I do not see anti explosive listed in any of the technical | |specifications of WLAN phones made by | | | |Zyxel | |Hitachi | |UTStarCom | |Uniden | |Cisco | |Net2Com | | | |Cory Andrews | |Purchasing / EVP | |VOIPSupply.com | |v - 716.630.1555 X22 | |e - [EMAIL PROTECTED] | | | | | | | |Anton Krall wrote: | | | |Guys. | | | |I know there are wifi sip phones out there but I have a |question, | |are any of these phones anti explosive? By that I mean, there | |are certain | |regulations about phones or cel phones that are not | recommended to | |operate in environments like gas stations due to sparks and | |the chance | |of ingiting gas fumes. | | | |Are there any wifi sip phones out here that have complaince with | |regulations to operate in hazardous environments like Oil | Platforms, | |etc? phones denominated anti explosive or something? | | | |___ | |Asterisk-Users mailing list | |Asterisk-Users@lists.digium.com | |http://lists.digium.com/mailman/listinfo/asterisk-users | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | | | | | | | |___ | |Asterisk-Users mailing list | |Asterisk-Users@lists.digium.com | |http://lists.digium.com/mailman/listinfo/asterisk-users | |To UNSUBSCRIBE or update options visit: | | http://lists.digium.com/mailman/listinfo/asterisk-users | | | |
[Asterisk-Users] Do includes include the includes
I am grouping my extensions by building like so: 1XX is Building 1 2XX is Building 2 7XX is Office [Office] extensions has the following includes 7xx Include = Local Include = International Include = Building1 Include = Building2 [Building1] has 1xx Include = Office Include = Building2 Include = Local I don't want building1 to access international, but does it inherit that include through including the office context? If it does, how can I structure a dialplan so that each building can call each other but building1 does not have international? Chris Mason ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi IP Phones
Thank you very much for the explanation Terry. I knnew some of this stuff (point 1 and 2) and this client is requesting IS equip specifically so you couldn?t be more right. Im dealing with this thru a 3rd party (the client is a client his). Im just in charge of finding out the viability of using asterisk with IS wifi phones if there are any. |-Original Message- |From: [EMAIL PROTECTED] |[mailto:[EMAIL PROTECTED] On Behalf Of |Terry H. Gilsenan |Sent: Mircoles, 15 de Junio de 2005 11:29 p.m. |To: 'Asterisk Users Mailing List - Non-Commercial Discussion' |Subject: RE: [Asterisk-Users] WiFi IP Phones | | | | -Original Message- | From: [EMAIL PROTECTED] | [mailto:[EMAIL PROTECTED] On Behalf Of Anton | Krall | Sent: Thursday, 16 June 2005 6:53 AM | To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - |Non-Commercial | Discussion' | Subject: RE: [Asterisk-Users] WiFi IP Phones | | I know... The term anti explosive is new to me.. I never |heard of it | but a possible client is asking for that exactly since the |phones are | going to be used in oil refinary and rd platforms using voip over | satelite connections... | | What do you think? | | BTW, how is voip over satelite? I know you have the usual 500 ms lag | for up and down stream delays but hows quality? | |Anton, | |1, Intrinsically Safe is a _Requirement_, not an option on an |oil rig, refinery, etc... There are a few Zone Categorys |within the IS framework, and different equipment is certified |by an independent body to pass different tests, and so get |different ratings. | |2, Each and every IS certified device is shipped with a |Certificate explaining the _exact_ rating and zone category |that the item has been certified for. This certificate will be |tied to the device by serial number. | |3, Each and every IS certified device and its powersupply are |tagged with a green Ex or Fm sticker that displays the Zone |category for which they have been certified, this _Must_ match |the details of the accompanying Certificate. | |4, Using non IS equipment in an IS zone will likely void any |and all private, public, and other insurance for the entire |site, and most likely result in the On the spot dissmissal |of the person that brought the non IS device on-site, and will |at the very least cause a muster and possibly a shutdown that |in the case of an offshore oil platform could cost about |$10Million in lost production before the plant could re-start. | |5, How do I know this? I work for an oil company and it is my |job to make sure that IS regulations are followed. Ie: my job |is on the line. | |6, VoIP works fine over vsat, there is a slight delay, but it |is easy to get accustomed to it. In fact the quality can be |better than some intercontinental voice calls using telco's as |the pipe. | |Further, there are sections of a refinery or oil rig that are |not IS zones, |eg: Sleeping quarters, mess(canteen) An asterisk box could be |located there and added to the existing OF infrastructure, or |added to existing IS certified equipment using TDM cards. | |On the platform itself they will need both an IS Intercom and |handheld IS VHF or UHF radios (Both for HSEQ reasons) | |I hope this helps | |T | |___ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WiFi IP Phones
On Thursday 16 June 2005 06:59, Anton Krall wrote: Guys.. We havent even started taking about costs here. Velieve me, this customer is not concerned about costs ... At least not yet... He just wants to know if there are any IS wifi phones compatible with asterisk out there. Any IS SIP phone should work just fine with Asterisk, but I have never seen an IS SIP phone. It might be far better to use an IS cordless phone and put the base station and ATA in an explosion-proof box. We do that with our industrial drives. VERY cool looking (1/2 or 3/4 bolt every 1 around the entire enclosure door, 1 thick steel) but as you can imagine, unbelievably expensive. The IS crowd is a lot like the food agency crowd -- stainless steel enclosures ain't cheap but you're talking about health and safety -- they realize and understand that they're not looking for cheap. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] This mailing list is being spam filtered on my site.
On Thu, June 16, 2005 12:34, Andrew Kohlsmith said: On Thursday 16 June 2005 02:01, Francesco Peeters wrote: Check whether Bayes filter is set for auto-learn. It has somehow aquired enough keywords from this list to mark the emails from here as SPAM. I do not know which filter you use, but the SpamAssassin built in Bayesan allows for 'HAM' (ie NON-SPAM) mails to be learnt... Try collecting a weeks worth of list mails and then have the filter scan them (look for sa-learn) as 'HAM'... I too am seeing this and I've been using SA for YEARS. I've been trying to train it but some of the messages to this list just do not want to be classified as non-spam. Im trying to get them to come out clean without resorting to a whitelist. -A. I too have SA running on my FC3/Postfix server, and it only picks out the occasional post, so I'm not complaining (yet!) First thing I did though was make sure it did not autolearn, and set up a HAM and SPAM alias to send identified e-mails to for SA to learn from... I have 31 mails in the HAM box and 1100+ in the SPAM box... (I also have SPF and Grey-listing on, which catches a good amount of spam, as do the sorbs and monkey lists) -- Francesco Peeters GPG Key = AA69 E7C6 1D8A F148 160C D5C4 9943 6E38 D5E3 7704 If your program doesn't recognize my signature, please visit http://www.CAcert.org/index.php?id=3 to retrieve the Root CA certificate. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] unamble to dialout to mobiles and others special numbers
Asterisk 1.0.7-BRIstuffed-0.2.0-RC8a on a Debian 3.1 The system is connected with an HFC card directly to the telco line card is in TE mode and signalling used is bri_cpe_ptmp I am able to dial out some numbers and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... If i use a normal hardware isdn phone i am able to do such calls. This is a call that works: -- Executing NoOp(SIP/11-1ecc, Call to 756756756) in new stack -- Executing GotoIf(SIP/11-1ecc, 0?3:5) in new stack -- Goto (default,059305698,5) -- Executing GotoIf(SIP/11-1ecc, 0?6:8) in new stack -- Goto (default,059305698,8) -- Executing NoOp(SIP/11-1ecc, External call) in new stack -- Executing Goto(SIP/11-1ecc, esterni|756756756|1) in new stack -- Goto (esterni,059305698,1) -- Executing Dial(SIP/11-1ecc, Zap/g1/756756756) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/756756756 -- Zap/1-1 is ringing [now i hangup] -- Hungup 'Zap/1-1' == Spawn extension (esterni, 756756756, 1) exited non-zero on 'SIP/11-1ecc' -- Executing Goto(SIP/11-1ecc, default|h|1) in new stack -- Goto (default,h,1) -- Executing Hangup(SIP/11-1ecc, ) in new stack == Spawn extension (default, h, 1) exited non-zero on 'SIP/11-1ecc' == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up This is a call that does NOT work (ir. i'm calling my mobile phone): == Primary D-Channel on span 1 down == Primary D-Channel on span 1 up -- Executing NoOp(SIP/11-9d74, Call to 3777) in new stack -- Executing GotoIf(SIP/11-9d74, 0?3:5) in new stack -- Goto (default,3777,5) -- Executing GotoIf(SIP/11-9d74, 0?6:8) in new stack -- Goto (default,3473042866,8) -- Executing NoOp(SIP/11-9d74, External call) in new stack -- Executing Goto(SIP/11-9d74, esterni|3777|1) in new stack -- Goto (esterni,3777,1) -- Executing Dial(SIP/11-9d74, Zap/g1/3777) in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/3777 -- Channel 0/1, span 1 got hangup Jun 16 13:07:17 WARNING[17330]: app_dial.c:412 wait_for_answer: Unable to forward voice Jun 16 13:07:17 WARNING[17330]: app_dial.c:412 wait_for_answer: Unable to forward voice -- Hungup 'Zap/1-1' == No one is available to answer at this time -- Executing Answer(SIP/11-9d74, ) in new stack -- Executing Playtones(SIP/11-9d74, congestion) in new stack -- Executing Congestion(SIP/11-9d74, ) in new stack Some configuration files: http://marcopar.altervista.org/extensions.conf http://marcopar.altervista.org/zapata.conf http://marcopar.altervista.org/zaptel.conf in the system messages i'm getting this: Zapata Telephony Interface Registered on major 196 PCI: Enabling device :00:06.0 ( - 0003) ACPI: PCI interrupt :00:06.0[A] - GSI 17 (level, low) - IRQ 185 zaphfc: CCD/Billion/Asuscom 2BD0 configured at mem 0xd08eaf00 fifo 0xcf338000(0xf338000) IRQ 185 HZ 1000 zaphfc: Card 0 configured for TE mode zaphfc: 1 hfc-pci card(s) in this box. Zaptel Configuration == SPAN 1: CCS/ AMI Build-out: 399-533 feet (DSX-1) Channel map: Channel 01: Individual Clear channel (Default) (Slaves: 01) Channel 02: Individual Clear channel (Default) (Slaves: 02) Channel 03: D-channel (Default) (Slaves: 03) 3 channels configured. frequently i get: zaphfc: empty HDLC frame or bad CRC received (framelen = 6, stat = 0xff, card = 0). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unamble to dialout to mobiles and others special numbers
Hi, I am able to dial out some numbers and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... try with: pridialplan=unknown prilocaldialplan=unknown matteo -- Matteo Brancaleoni System Administrator Tel +39.02.70633354 Sip [EMAIL PROTECTED] Iax2 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Nasty little incident ...
Doubtful its a clocking issue; the clock is actually on the E1 card and it obtains sync from whichever card you specify. The total lack of sync will not cause a total failure of the card as described. The OP did not mention whether the asterisk system was rebooted after disconnecting the meridian, so I don't believe one can _assume_ the channel numbers didn't change. Exactly what I was about to say Steve. The numbers won't change. They are configured when the driver actually detects the E1 card and it's spans. If a span goes down it doesn't disappear. Turning off the meridian would be the same as an E1 that's connected to a carrier going down. If the channel numbers changed and everything stopped working every time that happened, no one would be using asterisk. Our carrier friends are hardly 100% reliable. I'm going with clock source. I have a feeling that it was using span 4 for clocking and when it lost that, it broke everything... Jamie On Wed, 2005-06-15 at 21:32 +0100, Steve Hanselman wrote: I doubt they do, if they are marked as being there, but happen to be down then the numbers would stay the same. Sounds more likely that something happened with the clock source. You'd need to reproduce it out of hours and look at the output of pri show span x and cat /proc/zaptel/* From: [EMAIL PROTECTED] on behalf of Rich Adamson Sent: Wed 15/06/2005 5:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Nasty little incident ... We have a te410p, with the following connections: span 1 connected to a 32 Channel EuroISDN span 2 connected to a card in a legacy pbx (Meridian) span 3 connected to a 10 Channel EuroISDN span 4 connected to a card in a legacy pbx (Meridian) We have no need for the meridian now, and decided to turn it off. I did not change the zaptel.conf settings, nor the zapata.conf settings. When the meridian was turned off, * would no longer allow any outbound or inbound calls through spans 1 and 3 (although these are connected to the pstn). When I turned the meridian back on - in a hurry I might add ;) (had no time to play with configurations) and restarted *, then everything was ok again ... Should I comment out span 2 and 4, run a ztcfg, unplug the cables in 2 and 4, and then turn off the meridian ? Julian. /* zaptel.conf */ span=1,1,0,ccs,hdb3,crc4 bchan=1-15,17-31 dchan=16 span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 dchan=47 span=3,2,0,ccs,hdb3,crc4 bchan=63-77,79-93 dchan=78 span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 dchan=109 loadzone=uk defaultzone=uk Just a wild guess When the two meridian links disappeared, the channel numbers probably changed. Instead of channels 1 through 124, you probably have channels 1 through 62 and your supporting dialplan (and other channel specific items) likely don't match. I thought that the definitions in the zaptel.conf and zapata.conf (see below) defined the channel numbers, not the physical channels themselves ? I use Dial(zap/g3) to call on the zap channels. /* zapata.conf */ context=isdn32-b prilocaldialplan=national internationalprefix = 00 nationalprefix = 0 localprefix = 01702 group=1 signalling=pri_cpe switchtype=euroisdn channel=1-15,17-31 context=meridian-b group=2 signalling=pri_net switchtype=euroisdn channel=32-46,48-62 context=isdn32-a pridialplan=unknown group=3 signalling=pri_cpe switchtype=euroisdn channel=63-77,79-93 context=meridian-a group=4 signalling=pri_net switchtype=euroisdn channel=94-108,110-124 I'm sure there are others on this list that can add to this, but when the card drivers are loaded and ztfg run, the channels that are discovered have to be mapped to what's in zaptel.conf one way or another. (Moving card driver load around changes the discovered order and one must manually modify zaptel.conf to match.) Then each zap channel is defined in zapata.conf, and those definitions have to match the channel numbers resulting from the above zaptel.conf stuff. So, what happens when two E1s disappear? Do the avaiable channel numbers change at the zaptel.conf level? My best guess is they do, but I don't have E1s around to play with to prove it. So, that's my best guess and it certainly can be an incorrect guess on my part.
[Asterisk-Users] How to dimension Asterisk - that is used solely as callback server - only sending untranscoded voice between two ISDN channels on PRI ?
Hi, I wonder how I could dimension Asterisk system that will be used solely as callback server : - when user calls it registers ring, hangup and calls back - it gives him a dial signal and calls dialed number on another ISDN channel out that means plain transfer between two ISDN channels - no transcoding or any other stuff... I guess using Asterisk in this way I could dimension for higher number of parallel calls - but how many ? General rule is to put 1 octo PRI card per PC, but could I add another one or more if used in described way ? Any similar examples of dimensioning ? Do I get any better with cluster for such purpose ? Any other advice ? Thanks in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SER and Asterisk question
Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and which has a uri rule that says forward 4 digit starting with 1 to the asterisk sip port the asterisk extensions.conf has an entry for 1999 and dials [EMAIL PROTECTED], if not answered voicemail runs and so on. ain't there a way to make 666 directly call 999 without using 1999. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Fall back dialing
We have ServerA that connects to ServerB to dial long distance via an IAX2
trunk. I have setup an international dialing plan so that there is a backup
route via pstn if the IAX channel is down.
exten = _1NXXNXX,1,Dial,IAX2/${SERVERB}/${EXTEN},60)
exten = _1NXXNXX,2,Dial(Zap/g2/${EXTEN},70)
exten = _1NXXNXX,3,Macro(fastbusy)
exten = _1NXXNXX,4,hangup
exten = _1NXXNXX,102,Dial(Zap/g2/${EXTEN},70)
exten = _1NXXNXX,103,Macro(fastbusy)
exten = _1NXXNXX,104,hangup
Would it be better to use
exten = _1NXXNXX,102,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?i,1:103)
Any other improvments? We want to make is transparent to the users.
Chris Mason
NetConcepts
Int: (305) 704-7249 Fax: (815)301-9759
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Re: [Asterisk-Users] SER and Asterisk question
yes, there is. run everything through asterisk, no matter how long the extensions are. for example, 666 calls 999 goes to asterisk, sees a dial sip:[EMAIL PROTECTED], goes back to SER. bounces back to ser. If everything is working well asterisk will set up the call and get out of the way. I don't see why you need to prepend digits in order to make this work, if i'm missing something let me know. -yair On 6/16/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote: Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and which has a uri rule that says forward 4 digit starting with 1 to the asterisk sip port the asterisk extensions.conf has an entry for 1999 and dials [EMAIL PROTECTED], if not answered voicemail runs and so on. ain't there a way to make 666 directly call 999 without using 1999. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER and Asterisk question
If these are the only calling rules you could try
if (!lookup(location))
{
t_relay to your asterisk box
break
}
Mohamed A. Gombolaty wrote:
Dear All,
I am trying to make the phones always talk to each other (peer to peer)
using SER as a sip proxy, and incase the call is not answered we will
use the voicemail of asterisk and other feautures, I have done that
already, but in order to do so I found that I have to make the users
dial different exten numbers, here is an example:
user with exten 666 wants to call 999 .
666 dials 1999 and which has a uri rule that says forward 4 digit
starting with 1 to the asterisk sip port
the asterisk extensions.conf has an entry for 1999 and dials
[EMAIL PROTECTED], if not answered voicemail runs and so on.
ain't there a way to make 666 directly call 999 without using 1999.
--
Thx
MAG
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--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:[EMAIL PROTECTED]
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Re: [Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - whatsettings work ?
Hi, thanks for response I have following in zapata.conf, so I guess point to multipoint setting is right ? Is framing and coding (ami,ccs) right for Italy ? Thanks in advance, regards, Rob. zapata.conf: [channels] switchtype = euroisdn ;pridialplan = dynamic je delalo pridialplan = unknown ;prilocaldialplan = local nationalprefix = 0 internationalprefix = 00 usecallingpres=yes echocancel = yes echocancelwhenbridged = yes echotraining = 100 callerid=asreceived overlapdial=yes ; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode) signalling = bri_cpe_ptmp context=from-isdn group = 1 ; S/T port 1-3 (first quadBRI, or lower ports of an octoBRI) channel = 1-2 channel = 4-5 channel = 7-8 ;--- - Original Message - From: Matteo Brancaleoni [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, June 09, 2005 10:45 AM Subject: Re: [Asterisk-Users] EuroISDN Italy - quadbri - zaptel.conf - whatsettings work ? You're connected to a p2mp bri, switch to bri_cpe_p2mp Matteo. Il giorno mer, 08-06-2005 alle 19:54 +0200, Robert Rozman ha scritto: Hi, I'm pulling my hair out, cause cannot connect to EuroISDN BRI in Italy with octobri card from Beronet. I use bristuff and have following zaptel.conf... # # This file is parsed by the Zaptel Configurator, ztcfg # # # First come the span definitions, in the format # span=span num,timing,line build out (LBO),framing,coding[,yellow] # # The timing parameter determines the selection of primary, secondary, and # so on sync sources. If this span should be considered a primary sync # source, then give it a value of 1. For a secondary, use 2, and so on. # To not use this as a sync source, just use 0 # loadzone=it defaultzone=it span=1,1,3,ccs,ami span=2,0,3,ccs,ami span=3,0,3,ccs,ami span=4,0,3,ccs,ami span=5,0,3,ccs,ami span=6,0,3,ccs,ami span=7,0,3,ccs,ami span=8,0,3,ccs,ami bchan=1,2 dchan=3 bchan=4,5 dchan=6 bchan=7,8 dchan=9 bchan=10,11 dchan=12 bchan=13,14 dchan=15 bchan=16,17 dchan=18 bchan=19,20 dchan=21 bchan=22,23 dchan=24 I get this on bri intense debug... Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=64864 [ fc ff 03 0f fd 60 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=39384 [ fc ff 03 0f 99 d8 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Sending TEI Request ri=38343 [ fc ff 03 0f 95 c7 01 ff ] Unnumbered frame: SAPI: 63 C/R: 0 EA: 0 TEI: 127EA: 1 M3: 0 P/F: 0 M2: 0 11: 3 [ UI (unnumbered information) ] 5 bytes of data Thanks very much in advance, regards, Rob. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nasty little incident ...
Just a wild guess When the two meridian links disappeared, the channel numbers probably changed. Instead of channels 1 through 124, you probably have channels 1 through 62 and your supporting dialplan (and other channel specific items) likely don't match. No - just because the span goes doesn't mean its gone. Its simply there and down. Here's my theory as to the problem: In the config, Spans 1 and 3 are to the telco, 2 and 4 to the old pbx. Clocking is being taken from spans 1 and 3. Now the symptom when the meridian was disconnected was like zaptel had no clock. So theory one is that the spans are actually plugged into the board upside down, with telco on 4 and 2, meridian on 3 and 1. so when the meridian was disconnected there was no more clock. Second theory is that the zaptel.conf was changed - maybe moving which spans clock comes from - and the zaptel modules weren't reloaded or ztcfg wasn't run. Port 1 on the TE410P is at the top (away from the mobo), btw. The E1 card does not receive clocking from any span. It sync's the on-board clock to whatever span you choose. If you watch what others have posted on the list over many months, you'll notice many have never specified a clock sync source. The problem they have is typically associated with clicking and other audio distortion; not a total failure. So, highly unlikely to have anything to do with clock sync. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] reload from dialplan
Is there any way of reloading * from the dialplan (short of executing a system asterisk -rx) ? I was thinking of allowing someone to dial a special extension, enter a password and then have an ivr to 1) Reload SIP 2) Reload VM 3) Reload Agents 4) Reload Queues 5) Reload All We are running with static .conf files and have not yet ventured into the realms of realtime ... Julian. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SER and Asterisk question
Dear Yair, Actually what happens is that from SER debug I can see the call is looping between Asterisk and SER. but adding a number makes no loops. Thx MAG Yair Hakak wrote: yes, there is. run everything through asterisk, no matter how long the extensions are. for example, 666 calls 999 goes to asterisk, sees a dial sip:[EMAIL PROTECTED], goes back to SER. bounces back to ser. If everything is working well asterisk will set up the call and get out of the way. I don't see why you need to prepend digits in order to make this work, if i'm missing something let me know. -yair On 6/16/05, Mohamed A. Gombolaty [EMAIL PROTECTED] wrote: Dear All, I am trying to make the phones always talk to each other (peer to peer) using SER as a sip proxy, and incase the call is not answered we will use the voicemail of asterisk and other feautures, I have done that already, but in order to do so I found that I have to make the users dial different exten numbers, here is an example: user with exten 666 wants to call 999 . 666 dials 1999 and which has a uri rule that says forward 4 digit starting with 1 to the asterisk sip port the asterisk extensions.conf has an entry for 1999 and dials [EMAIL PROTECTED], if not answered voicemail runs and so on. ain't there a way to make 666 directly call 999 without using 1999. -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thx MAG ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] unamble to dialout to mobiles and others special numbers
Matteo Brancaleoni ha scritto: I am able to dial out some numbers and some not. In particular it seems that i can't call mobiles and special telco numbers like the information call center, emergency numbers,... try with: pridialplan=unknown prilocaldialplan=unknown it works. thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] terminating DID to FWD
I have an 800 number over IAX2 from Clearpath in Detroit. I bet they'd do it. Michael On Wed, 15 Jun 2005 22:08:33 -0600, Darren Wiebe wrote: This would not be a problem if you could find a provider willing. You would probably have better luck with a smaller provider as I'm not aware of any of the big ones that would do it. Forwarding a tollfree number to a FWD number through an asterisk box would be trivial. Darren Wiebe [EMAIL PROTECTED] Joseph wrote: Is it possible to terminate (or forward) lets say 800 DID number to FWD number. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] o713-861-4005 o800-905-6412 c713-201-1262 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # to work during a call
Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby 2001 host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 incominglimit=2 [EMAIL PROTECTED] disallow=all allow=alaw allow=ulaw callgroup=2 pickupgroup=2 and in the SIPDefault.cnf for the phones I have: # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 DTMF works for voicemail and for remote services over both analogue Zap channels and digital (ISDN) channels. Asterisk doesn't appear to be 'monitoring' the audio so I can't get to Asterisk features like Asterisk's transfer, parked calls and one-tuch-record... Am I missing something? Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reload from dialplan
Is there any way of reloading * from the dialplan (short of executing a system asterisk -rx) ? I was thinking of allowing someone to dial a special extension, enter a password and then have an ivr to 1) Reload SIP 2) Reload VM 3) Reload Agents 4) Reload Queues 5) Reload All We are running with static .conf files and have not yet ventured into the realms of realtime ... Sure. Try something like exten=1234 1,reload with the proper syntax and construction. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bill seconds
Your customers are not going to like this. You have to change the way you bill for calls. For $1 your customer gets 60 seconds worth of phone time. However you have to also charge, like the Bells used to, for setup and teardown time. Remember the operator used to say Deposit $1.85 for the first three minutes and then it would be 30 cents per minute after that. Buy a phone card from a competitor and look at the fine print on the card. You charge buy seconds they are connected to your system, not for the time they are actually talking to the remote party. Example: To set up the call you charge 10 seconds, and to stop the call you charge 5 seconds. So the customer only gets 45 seconds of call time. You get a 15 second cushion. Does not seem fair does it. But if they buy an hour 3600 seconds worth of calls the missing 15 seconds won't be noticed. You can go further. Say they buy a 3600 second card. When they call to check their time the first time on the card you tell them they have 60 minutes, but you charge them 30 seconds for asking. Set up the code so that every time they call you have too fields to track call time. The time they think they have and the time you know they have. You tell them they have 45 minutes, but the other field knows they only have 30 minutes. If they ask then your script says 45 minutes left but you cut them off when the use 30. Then you chip away each time the call. 10 seconds for making a call, and 5 seconds when they hang up. This way you are always in credit and can cut them off without loosing money. Some card vendors go even further. They sell 3600 seconds, but each time a call is made they whack a random percentage of the time. Worse yet their card system will randomly or systematically hang up on callers. This will cause the user to redial the call and get hit with connection charges that vary. Customers eventually figure out which cards do this type of chicanery and they stop buying them, but only if there is a competitor for the route they want to call. Such is the world of unregulated phone calls. Not pretty is it. Charging time for each call is part of the business. If you don?t want to charge time to setup and teardown then you have to charge more per minute. Your customers get all the time the pay for down to the second, but you are going to have to charge more per minute or you will be in the boat you are in now. Race the tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Thursday, June 16, 2005 1:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bill seconds I've done a little thinking on this one If you are using ASTCC, it would be fairly straightforward to edit it and have it make a 2 second adjustment. If your using another solution it probably would be fairly easy also... Darren Wiebe [EMAIL PROTECTED] Americo Sanchez C. wrote: Hi all, We've installed Asterisk on a rural development project and we're testing a prepaid phone service. As far as now we're having terrific service results but there's a problem with the calls billing at our local telecom. For instance, a farmer buys a 1 dollar phone card and use it to dial a USA number, the call should lasts for 60 seconds. Asterisk is doing a great job finishing the call exactly at 60 seconds. The problem is that the telecom company billing system adds a two second delay for each call, so the bill is not for 1 but 2 minutes (they round fractions up). We're loosing money and the local telecom doesn't seem to have a solution for this matter. Have you experienced something similar? Do you have any idea of how can we solve this? Is it possible to configure Asterisk so that the system thinks that a minute has 58 seconds instead of 60? _ MSN Amor: busca tu naranja http://latam.msn.com/amor/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cisco 7960 (SIP) with Asterisk: how to get # to work during a call
# and * are mapped later in the SIP(Default/MAC).cnf it has a section in the manual if you want to see why. On 6/16/05, Michael J. Tubby B.Sc (Hons) G8TIC [EMAIL PROTECTED] wrote: Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby 2001 host=dynamic nat=no canreinvite=no dtmfmode=rfc2833 incominglimit=2 [EMAIL PROTECTED] disallow=all allow=alaw allow=ulaw callgroup=2 pickupgroup=2 and in the SIPDefault.cnf for the phones I have: # Inband DTMF Settings (0-disable, 1-enable (default)) dtmf_inband: 1 # Out of band DTMF Settings (none-disable, avt-avt enable (default), avt_always - always avt ) dtmf_outofband: avt # DTMF dB Level Settings (1-6dB down, 2-3db down, 3-nominal (default), 4-3db up, 5-6dB up) dtmf_db_level: 3 DTMF works for voicemail and for remote services over both analogue Zap channels and digital (ISDN) channels. Asterisk doesn't appear to be 'monitoring' the audio so I can't get to Asterisk features like Asterisk's transfer, parked calls and one-tuch-record... Am I missing something? Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- sig Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) WWW: http://lathama.com Email: [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! /sig ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] WiFi IP Phones
Ahmm Andrew, are you sure they are steel? It's been a long time since I did any work in this space but we used to install them in plastic not metal.plastic works better with the radio waves. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith Sent: Thursday, 16 June 2005 7:12 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] WiFi IP Phones On Thursday 16 June 2005 06:59, Anton Krall wrote: Guys.. We havent even started taking about costs here. Velieve me, this customer is not concerned about costs ... At least not yet... He just wants to know if there are any IS wifi phones compatible with asterisk out there. Any IS SIP phone should work just fine with Asterisk, but I have never seen an IS SIP phone. It might be far better to use an IS cordless phone and put the base station and ATA in an explosion-proof box. We do that with our industrial drives. VERY cool looking (1/2 or 3/4 bolt every 1 around the entire enclosure door, 1 thick steel) but as you can imagine, unbelievably expensive. The IS crowd is a lot like the food agency crowd -- stainless steel enclosures ain't cheap but you're talking about health and safety -- they realize and understand that they're not looking for cheap. -A. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reload from dialplan
Oh Nuts, I tried looking for that in the applications - it did not show .. I know it's available on the command line. I've just tried that, Jun 16 15:01:33 WARNING[5491]: pbx.c:1648 pbx_extension_helper: No application 'Reload' for extension ... Julian Rich Adamson wrote: Is there any way of reloading * from the dialplan (short of executing a system asterisk -rx) ? I was thinking of allowing someone to dial a special extension, enter a password and then have an ivr to 1) Reload SIP 2) Reload VM 3) Reload Agents 4) Reload Queues 5) Reload All We are running with static .conf files and have not yet ventured into the realms of realtime ... Sure. Try something like exten=1234 1,reload with the proper syntax and construction. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Subject: [Asterisk-Users] asterisk gsm gateway hardware
Hello, I would like to implement a home GSM gateway using asterisk. What would you recommend me as a low-cost hardware for creating a gsm channel? I found 2n gsm gateway, that supports sip and chan_blue for bluetooth connections. Any recommendations? Basically, I want to end calls to some GSM number in my sip telephone and for some prefixes dial out using that same sip telephone. Also sending and receiving SMS will be a plus. I have a friend living in luxembourg, which would like a slovak phone number to communicate with friends. It would end on my server at home and all calls to his sim card will be routed to his ip telephone in luxembourg (and vice versa). Support for more than one sim card is a plus. Since it's a home/hobby use, I would prefer a low-cost solution. Any ideas (may be off-list) are welcome). The solution I use works very well if you need to be able to take the mobile phone away with you when you leave the house. I use a phonelabs.com dock'n'talk with a bluetooth module connected to a standard digium one-port FXO card (XP100). I can make and receive GSM calls via my mobile from asterisk, treating it as just another channel. The phone automatically connects to the dock'n'talk when it comes into bluetooth range. If you are happy with a fixed solution, where you leave the SIM permanently installed, you might want to look for a Nokia Premicell or equivalent on e-bay. This would also connect to a standard FXO port. HTH Patrick ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] 1-800 DID in Alberta
Try Thinktel http://www.thinktel.ca asterisk friendly -Original Message- From: Leon Sun [mailto:[EMAIL PROTECTED] Sent: Wednesday, June 15, 2005 7:14 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] 1-800 DID in Alberta Group Telecom and Telus. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: June 15, 2005 5:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 1-800 DID in Alberta Are there any 800 DID number providers for Alberta? -- #Joseph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Case studies for Asterisk Voicemail
I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the existing PBXs for MWI, etc). The customer would like some case studies of people who've done this before, even if it's just Yes, we've done it and are happy with the results. Now, I've implemented systems of this size with IBM Websphere Voice Response, but not with Asterisk, so don't have case studies to offer, despite being pretty confident it will work. Does anyone have a production Asterisk Voicemail system in this range using either of these models and would be willing to put their hand up and say Yes, we're pleased with our system? -- Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] VoiceXML? question
Read the voip-info post on Tellme but unfortunately only if you have a large minute application. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of El Flynn Sent: Thursday, 16 June 2005 12:15 AM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non- Commercial Discussion Subject: Re: [Asterisk-Users] VoiceXML? question dave cantera wrote: hi, is there anything going with VoiceXML in asterisk??? is this the list to query regarding this or should I put this on the dev list? thanks, dave cantera I don't think there's anything built-in to support VoiceXML, but you _can_ do something like this: 1. get a developer account on Voxeo (http://community.voxeo.com/account/register.jsp) or some other VoiceXML provider 2. create your VXML app, and point to it appropriately on the developer account pages 3. connect via SIP from Asterisk to your VoiceXML app. 4. Fini Voxeo provides facilities to call in via Free world dialup, and your hosted applications can be accessed via a FWD number. I've got a simple demo running on our pbx and it works. Flynn ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem with 2 digium cards
Hi all, I have a problem with 2 digium cards(t100p and te110p), I can load each card but not in same time. FATAL: Error inserting wct1xxp ... : No such device FATAL: Error running install command for wct1xxp My two cards don't have same IRQ. Someone have an idea? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Max TNT
What signalling are you using, PRI, RBS, What model of TNT are you using??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Baird Sent: Wednesday, June 15, 2005 10:55 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk and Max TNT Hello, I'm currently testing Asterisk over a T1 cross connect to a MaxTNT chassis that we have. It is working fine switching the calls through, but there is about a 10 second delay from the time Asterisk initiates the call until the TNT accepts it. It appears to be a ANI issue, I've changed several settings and formatting options on the T1 between the two, as well as turning on/off the callerid options in Zapata.conf, it's very strange. I'm pretty sure this is an interoperability issue between the two devices, I'm looking for a magic setting. The TNT doesn't have this problem via SIP. Regards Michael Baird ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reload from dialplan
A google search with reload dialplan yields something like: system(asterisk -rx reload) Oh Nuts, I tried looking for that in the applications - it did not show .. I know it's available on the command line. I've just tried that, Jun 16 15:01:33 WARNING[5491]: pbx.c:1648 pbx_extension_helper: No application 'Reload' for extension ... Julian Rich Adamson wrote: Is there any way of reloading * from the dialplan (short of executing a system asterisk -rx) ? I was thinking of allowing someone to dial a special extension, enter a password and then have an ivr to 1) Reload SIP 2) Reload VM 3) Reload Agents 4) Reload Queues 5) Reload All We are running with static .conf files and have not yet ventured into the realms of realtime ... Sure. Try something like exten=1234 1,reload with the proper syntax and construction. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] AgentCallBacklogin (logout continued...)
Thanks for the info alan unfortunately I am trying to logout an agent that has a password. Example did give me ideas on how to do some other stuff though. I agree completely it is kind of silly to require a password to logout. Anyone know if: there is a way to execute something like the below but from the dialplan CLI agent logoff AGENT/2000 or is there some variable I can use in the dialplan that refers to the agent password or can i use realtime for agents.conf and do a db lookup to get the password so I can log them out Thanks J. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Nobody picked up in 30000 ms
Hi all, again, with another question ( may be the final one) I have come up to this point, means when I dial a number in my analogue (panasonic) phone I hear the ring at the end through my asterisk box (via TDM20B card) that uses IAX2 over teliax and after time-out, it gives this message. Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1,IAX2/[EMAIL PROTECTED]/10094472239112|30|tr) in new stack -- Called [EMAIL PROTECTED]/10094472239112 -- Nobody picked up in 3 ms -- Hungup 'IAX2/teliax-4' the total digits that I should dial are more than the instructions given by teliax, then in my extensions.conf file I increased the digits up to the suitable no as such _1X,1,...and informed teliax about the problem. Do you also think this is the reason for this situation? or something else? Please help me. I think I can make a call soon with your genourus help. Thank you Kumara ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] reload from dialplan
I did google :) I did mention quote (short of executing a system asterisk -rx) unquote. However, that's what I've ended up doing. 1) Reload SIP : asterisk -rx SIP reload 2) Reload VM : asterisk -rx reload app_voicemail.so 3) Reload Agents : asterisk -rx reload chan_agent.so 4) Reload Queues : asterisk -rx reload app_queue.so 5) Reload All : asterisk -rx reload Julian. Rich Adamson wrote: A google search with reload dialplan yields something like: system(asterisk -rx reload) Oh Nuts, I tried looking for that in the applications - it did not show .. I know it's available on the command line. I've just tried that, Jun 16 15:01:33 WARNING[5491]: pbx.c:1648 pbx_extension_helper: No application 'Reload' for extension ... Julian Rich Adamson wrote: Is there any way of reloading * from the dialplan (short of executing a system asterisk -rx) ? I was thinking of allowing someone to dial a special extension, enter a password and then have an ivr to 1) Reload SIP 2) Reload VM 3) Reload Agents 4) Reload Queues 5) Reload All We are running with static .conf files and have not yet ventured into the realms of realtime ... Sure. Try something like exten=1234 1,reload with the proper syntax and construction. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Error when compiling in freeTDS support
I'm trying to use freetds/odbc to write CDR records to a MSSQL database but when I installed them and tried to compile asterisk again I get: _tds.c cdr_tds.c: In function `mssql_connect': cdr_tds.c:415: `TDSCONNECTINFO' undeclared (first use in this function) cdr_tds.c:415: (Each undeclared identifier is reported only once cdr_tds.c:415: for each function it appears in.) cdr_tds.c:415: `connection' undeclared (first use in this function) cdr_tds.c:460: warning: implicit declaration of function `tds_free_connect' /usr/include/ctype.h: At top level: cdr_tds.c:71: warning: `connect_time' defined but not used make[1]: *** [cdr_tds.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/cdr' make: *** [subdirs] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bridged-appearances
I soo want this feature. This would be the last hurdle in getting off my Lucent/Avaya Definity G3. Mark Tim Connolly wrote: Has anyone figured out how to mimick a traditional bridged-appearance? My guys like the ability to put a call on hold on line 3 and it's the same call on line 3 on everyone else's phone. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SER and Asterisk question
[EMAIL PROTECTED] wrote: Actually what happens is that from SER debug I can see the call is looping between Asterisk and SER. but adding a number makes no loops. Check what the origin (IP/DNS name) of the incoming SIP message is. If it's from asterisk, send it to the user, if it is not from asterisk, it must be meant to go to asterisk. Add a couple of other tests (known user, etc) to it and then I think you'll have what you're looking for. -- Andreas SikkemaRits tele.com Van Vollenhovenstraat 33016 BE Rotterdam t: +31 (0)10 2245544 f: +31 (0)10 413 65 45 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Error when compiling in freeTDS support
Scrap this question.. found the answer later... so I'm using ODBC... but for some reason varchar(80) is coming in as 80 characters.. if say CLID is only 10 characters it will appear as 5703332121 [80 characters] any ideas? On 6/16/05, Matt [EMAIL PROTECTED] wrote: I'm trying to use freetds/odbc to write CDR records to a MSSQL database but when I installed them and tried to compile asterisk again I get: _tds.c cdr_tds.c: In function `mssql_connect': cdr_tds.c:415: `TDSCONNECTINFO' undeclared (first use in this function) cdr_tds.c:415: (Each undeclared identifier is reported only once cdr_tds.c:415: for each function it appears in.) cdr_tds.c:415: `connection' undeclared (first use in this function) cdr_tds.c:460: warning: implicit declaration of function `tds_free_connect' /usr/include/ctype.h: At top level: cdr_tds.c:71: warning: `connect_time' defined but not used make[1]: *** [cdr_tds.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk/cdr' make: *** [subdirs] Error 1 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk and 2 line MGCP phone
HI, Anybody here know or using Asterisk with 2 lines MGCP phone? I am trying to figure out if there are such device available and if so, how does it differenciate between the lines that is associated with extention number. Thanks ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk-biz] Case studies for Asterisk Voicemail
On Thu, Jun 16, 2005 at 03:27:49PM +0100, Alistair Cunningham wrote: I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). This will work. - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the existing PBXs for MWI, etc). I really don't think that you want to run NFS over the wide area. Not only do you have to be very very careful security-wise (i.e. do it over IPSec or something and make sure your NFS is not visible from the Internet itself) but do you really want to deal with the local VM server wedging when something funny happens on the network between the remote and central sites? It's not impossible but IMO you're asking for trouble doing it like this. -w ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] misdn and call hangup problem
Hi,
we test the misdn module together with beronet BN8S0 card.
We connect the pstn ISDN line to Port 1 and an ISDN phone to Port 2.
That works great, the ISDN phone rings an we can make the call.
When the caller hangsup before call is answered by the callee the call
on Port 2 rings until end of day.
This is the extensions.conf part for this:
[incoming]
exten = _., 1, Dial(mISDN/g:ntports/${EXTEN})
exten = _., 2, Congestion
[outgoing]
exten = _., 1, Dial(mISDN/g:teports/${EXTEN})
exten = _., 2, Congestion
This problem does not occur when we call the isdn phone from a sip client.
Can anybody tell what is wrong with this configuration.
Thanks,
Kib
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Re: [Asterisk-Users] Case studies for Asterisk Voicemail
On 6/16/05, Alistair Cunningham [EMAIL PROTECTED] wrote: I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the existing PBXs for MWI, etc). I would suggest using the Realtime Voicemail setup in a MySQL (or other) database. http://voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20Voicemail This works well for us with about 400 users (but still under development so not heavily tested). It makes updates and configurations much easier and could easily be share with many Asterisk machines. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problems with IAX Trunks
I posted this http://lists.digium.com/pipermail/asterisk-users/2005- June/111815.html and never received a response. I just wanted to share with you that I think I fixed the problem. The only thing I changed was my Dial command by removing the 'r' option. Since then, asterisk seems to properly discard all terminated calls. I don't know if it's a bug or expected with the 'r' option. Thanks, Waldo ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nobody picked up in 30000 ms
On Jun 16, 2005, at 8:23 AM, Kumara Jayaweera wrote: Starting simple switch on 'Zap/1-1' -- Executing Dial(Zap/1-1,IAX2/[EMAIL PROTECTED]/10094472239112|30|tr) in new stack -- Called [EMAIL PROTECTED]/10094472239112 What country code is that you're dialing? Robert Goodyear Brand Up LLC http://www.brand-up.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: [Asterisk-biz] Case studies for Asterisk Voicemail
William, I'm happy with the architecture options, as this is a company WAN with dedicated fibre links, and I'll be securing the database and NFS servers comprehensively. All I need are case studies to assure the customer that they're not the first people to do this on Asterisk. Positive reports on either method are fine, as long as they're in the several thousand user range, and are for production systems. Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ William Waites wrote: On Thu, Jun 16, 2005 at 03:27:49PM +0100, Alistair Cunningham wrote: I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). This will work. - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the existing PBXs for MWI, etc). I really don't think that you want to run NFS over the wide area. Not only do you have to be very very careful security-wise (i.e. do it over IPSec or something and make sure your NFS is not visible from the Internet itself) but do you really want to deal with the local VM server wedging when something funny happens on the network between the remote and central sites? It's not impossible but IMO you're asking for trouble doing it like this. -w ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Case studies for Asterisk Voicemail
Michael, Yes, this is exactly what we plan to do. Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ Michael Stearne wrote: On 6/16/05, Alistair Cunningham [EMAIL PROTECTED] wrote: I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the existing PBXs for MWI, etc). I would suggest using the Realtime Voicemail setup in a MySQL (or other) database. http://voip-info.org/tiki-index.php?page=Asterisk%20RealTime%20Voicemail This works well for us with about 400 users (but still under development so not heavily tested). It makes updates and configurations much easier and could easily be share with many Asterisk machines. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Case studies for Asterisk Voicemail
Several people have responded with architecture suggestions. While these are welcome, I'm happy with the architecture options planned, having done many large voicemail implementations on products other than Asterisk. What I had hoped to get from Asterisk-Users and Asterisk-Biz was not a technical discussion (though I don't mind getting this too), but reports of people who already have systems like this so I can put the customer's mind at ease that they're not the first people to use Asterisk voicemail with 3000 users. I should have made this more clear in my first email. Alistair Cunningham, Integrics Ltd, +44 (0)7870 699 479 http://integrics.com/ Alistair Cunningham wrote: I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the existing PBXs for MWI, etc). The customer would like some case studies of people who've done this before, even if it's just Yes, we've done it and are happy with the results. Now, I've implemented systems of this size with IBM Websphere Voice Response, but not with Asterisk, so don't have case studies to offer, despite being pretty confident it will work. Does anyone have a production Asterisk Voicemail system in this range using either of these models and would be willing to put their hand up and say Yes, we're pleased with our system? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Coding a telemarketing call blocker
Hi, I am interested in creating a telemarketing call blocker in my Asterisk dial plan. I am not much of a programmer, and I am wondering if external AGI code would be required to implement this. The logic that I would like to have in place is this: 1. If the incoming call carries proper name and number caller ID, then ring default extension. 2. If the incoming call carries no caller ID information, then send call to recorded message, followed by voice mail. 3. If the incoming call carries number only caller ID (no name info), then check the area code the call is from. If it is my local area code, then ring default extension, but if it is from a different area code, then send call to recorded message, followed by voice mail. Does anyone have any experience with implementing something like this? I could use some pointers to steer me in the right direction. Code samples would also be nice to have. Regards, Tore ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Coding a telemarketing call blocker
On Thu, Jun 16, 2005 at 10:45:04AM -0600, Tore Hansen said: I am interested in creating a telemarketing call blocker in my Asterisk dial plan. I am not much of a programmer, and I am wondering if external AGI code would be required to implement this. The logic that I would like to have in place is this: 1. If the incoming call carries proper name and number caller ID, then ring default extension. 2. If the incoming call carries no caller ID information, then send call to recorded message, followed by voice mail. 3. If the incoming call carries number only caller ID (no name info), then check the area code the call is from. If it is my local area code, then ring default extension, but if it is from a different area code, then send call to recorded message, followed by voice mail. See the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20GotoIf Example 3 has some logic that could easily be extended to do exactly what you want. With modern versions of * (CVS HEAD for example) the dial plan can be simplified a bit. I wrote that example pre 1.0 days... ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] have asterisk box #2 pick up calls.
hi, i am using iax. i am setting up a new asterisk box #2 on my network. It is behind another asterisk box#1. Box#1 acts as a router/firewall/asterisk/nat/dhcp. It has a public IP on ethernetcard1 and a private ip on ethernetcard2. box #2 has a private ip. I have a DID from teliax. When I call the DID number I want box#2 to receive the call. Both boxes are using the same teliax user/pass. How do i set it up so box#2 will pick up the calls ? how do i set things so it will work? __ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Intelligent maximum channels solution?
Hi list! I have an asterisk box connected to an ADSL connection that has 1 Mbit upstream. Is there any way to use max channels intelligently? For example I would like to do some checks on the outgoing calls. When it's quiet I want each and every call to go out to my IAX provider. However when more people start placing calls I would like to leave some room for the real expensive calls and switch chep (local) calls to the PSTN so expensive international calls can still be routed through the IAX provider. I.e. allow 2 local calls and 1 call to a neighbouring country and still leave 3 channels free for calls to Japan or China that would be frightfully expensive from the PSTN. This way I could squeeze the maximum benefit from the IAX / ADSL connection. Thanks! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bill seconds
Hi all, I am using ASTCC From: Darren Wiebe [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Bill seconds Date: Wed, 15 Jun 2005 23:05:54 -0600 I've done a little thinking on this one If you are using ASTCC, it would be fairly straightforward to edit it and have it make a 2 second adjustment. If your using another solution it probably would be fairly easy also... Darren Wiebe [EMAIL PROTECTED] Americo Sanchez C. wrote: Hi all, We've installed Asterisk on a rural development project and we're testing a prepaid phone service. As far as now we're having terrific service results but there's a problem with the calls billing at our local telecom. For instance, a farmer buys a 1 dollar phone card and use it to dial a USA number, the call should lasts for 60 seconds. Asterisk is doing a great job finishing the call exactly at 60 seconds. The problem is that the telecom company billing system adds a two second delay for each call, so the bill is not for 1 but 2 minutes (they round fractions up). We're loosing money and the local telecom doesn't seem to have a solution for this matter. Have you experienced something similar? Do you have any idea of how can we solve this? Is it possible to configure Asterisk so that the system thinks that a minute has 58 seconds instead of 60? _ MSN Amor: busca tu naranja http://latam.msn.com/amor/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ MSN Amor: busca tu naranja http://latam.msn.com/amor/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bill seconds
Another way I have seen this done is to sell units, not pounds and pence credit eg a 2 calling card has 160 units (ratio of 80 units to the pound). If you were to charge 8p per min you make that 8 units per min. This gives you a 20% increase which might help if your on per second billing to your upstream carrier. otherwise you need to make changes to your rating engine with a /60*58 to re-rate all calls back to a second ( /60) and move the minuite charge to be a 58 second minuit (*58) how that is achived needs you to give specific information on which calling card platform you are using. You may have a problem in defining the rates as per minuite if they are not a widely understood minuite legally - it depends on the laws of your country (in the UK the Trades Descriptions Act would apply and you'd be hit hard) David On 16/06/05, Race Vanderdecken [EMAIL PROTECTED] wrote: Your customers are not going to like this. You have to change the way you bill for calls. For $1 your customer gets 60 seconds worth of phone time. However you have to also charge, like the Bells used to, for setup and teardown time. Remember the operator used to say Deposit $1.85 for the first three minutes and then it would be 30 cents per minute after that. Buy a phone card from a competitor and look at the fine print on the card. You charge buy seconds they are connected to your system, not for the time they are actually talking to the remote party. Example: To set up the call you charge 10 seconds, and to stop the call you charge 5 seconds. So the customer only gets 45 seconds of call time. You get a 15 second cushion. Does not seem fair does it. But if they buy an hour 3600 seconds worth of calls the missing 15 seconds won't be noticed. You can go further. Say they buy a 3600 second card. When they call to check their time the first time on the card you tell them they have 60 minutes, but you charge them 30 seconds for asking. Set up the code so that every time they call you have too fields to track call time. The time they think they have and the time you know they have. You tell them they have 45 minutes, but the other field knows they only have 30 minutes. If they ask then your script says 45 minutes left but you cut them off when the use 30. Then you chip away each time the call. 10 seconds for making a call, and 5 seconds when they hang up. This way you are always in credit and can cut them off without loosing money. Some card vendors go even further. They sell 3600 seconds, but each time a call is made they whack a random percentage of the time. Worse yet their card system will randomly or systematically hang up on callers. This will cause the user to redial the call and get hit with connection charges that vary. Customers eventually figure out which cards do this type of chicanery and they stop buying them, but only if there is a competitor for the route they want to call. Such is the world of unregulated phone calls. Not pretty is it. Charging time for each call is part of the business. If you don't want to charge time to setup and teardown then you have to charge more per minute. Your customers get all the time the pay for down to the second, but you are going to have to charge more per minute or you will be in the boat you are in now. Race the tyrant Vanderdecken -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: Thursday, June 16, 2005 1:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bill seconds I've done a little thinking on this one If you are using ASTCC, it would be fairly straightforward to edit it and have it make a 2 second adjustment. If your using another solution it probably would be fairly easy also... Darren Wiebe [EMAIL PROTECTED] Americo Sanchez C. wrote: Hi all, We've installed Asterisk on a rural development project and we're testing a prepaid phone service. As far as now we're having terrific service results but there's a problem with the calls billing at our local telecom. For instance, a farmer buys a 1 dollar phone card and use it to dial a USA number, the call should lasts for 60 seconds. Asterisk is doing a great job finishing the call exactly at 60 seconds. The problem is that the telecom company billing system adds a two second delay for each call, so the bill is not for 1 but 2 minutes (they round fractions up). We're loosing money and the local telecom doesn't seem to have a solution for this matter. Have you experienced something similar? Do you have any idea of how can we solve this? Is it possible to configure Asterisk so that the system thinks that a minute has 58 seconds instead of 60? _ MSN Amor: busca tu naranja http://latam.msn.com/amor/
[Asterisk-Users] How to get started, what do I need?
Hi, I currently have a hardware PBX with its own custom phones, which I'm using pretty much as an in-house intercom system. At some point in the future I might want to convert to Asterisk. My setup is two analog POTS lines from my local phone company and currently nine stations or extensions, each of which contains the custom phone my current PBX uses. What hardware would I need for this setup, what types of Digium cards? All the different types of cards are a bit confusing to me as I've never done anything like this before. How are the cards priced? Currently I have no interest in ISDN or other digital protocols. I do have Packet8 service, but if I wanted to hook that up to Asterisk I'd probably just use a third POTS channel and hook up the convertor box. I have more questions about software implementation but I'll save those for later. Jayson. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Case studies for Asterisk Voicemail
On 6/16/05, Alistair Cunningham [EMAIL PROTECTED] wrote: Several people have responded with architecture suggestions. While these are welcome, I'm happy with the architecture options planned, having done many large voicemail implementations on products other than Asterisk. What I had hoped to get from Asterisk-Users and Asterisk-Biz was not a technical discussion (though I don't mind getting this too), but reports of people who already have systems like this so I can put the customer's mind at ease that they're not the first people to use Asterisk voicemail with 3000 users. I should have made this more clear in my first email. It was clear developers just like to put their 2 cents in I guess. :-) What you might do is contact a comapny like http://www.broadvoice.com/ or http://voicepulse.com/ and just ask if they are using Asterisk for their voicemail systems for their customers. I would think they might be using Asterisk for their systems and they have the amounts of users you're looking at. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bill seconds
The easiest way is to change another vendor asap. It is ridiculous that your carrier still uses 60+60 now(30+6 is an asset). 2 seconds doesn't matter and billing unit does. Leon Sun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: June 15, 2005 10:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bill seconds I've done a little thinking on this one If you are using ASTCC, it would be fairly straightforward to edit it and have it make a 2 second adjustment. If your using another solution it probably would be fairly easy also... Darren Wiebe [EMAIL PROTECTED] Americo Sanchez C. wrote: Hi all, We've installed Asterisk on a rural development project and we're testing a prepaid phone service. As far as now we're having terrific service results but there's a problem with the calls billing at our local telecom. For instance, a farmer buys a 1 dollar phone card and use it to dial a USA number, the call should lasts for 60 seconds. Asterisk is doing a great job finishing the call exactly at 60 seconds. The problem is that the telecom company billing system adds a two second delay for each call, so the bill is not for 1 but 2 minutes (they round fractions up). We're loosing money and the local telecom doesn't seem to have a solution for this matter. Have you experienced something similar? Do you have any idea of how can we solve this? Is it possible to configure Asterisk so that the system thinks that a minute has 58 seconds instead of 60? _ MSN Amor: busca tu naranja http://latam.msn.com/amor/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Bill seconds
If you need a SIP 30+6 a-z carrier, let me know. We may do 6+6 for you. Leon Sun -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Darren Wiebe Sent: June 15, 2005 10:06 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bill seconds I've done a little thinking on this one If you are using ASTCC, it would be fairly straightforward to edit it and have it make a 2 second adjustment. If your using another solution it probably would be fairly easy also... Darren Wiebe [EMAIL PROTECTED] Americo Sanchez C. wrote: Hi all, We've installed Asterisk on a rural development project and we're testing a prepaid phone service. As far as now we're having terrific service results but there's a problem with the calls billing at our local telecom. For instance, a farmer buys a 1 dollar phone card and use it to dial a USA number, the call should lasts for 60 seconds. Asterisk is doing a great job finishing the call exactly at 60 seconds. The problem is that the telecom company billing system adds a two second delay for each call, so the bill is not for 1 but 2 minutes (they round fractions up). We're loosing money and the local telecom doesn't seem to have a solution for this matter. Have you experienced something similar? Do you have any idea of how can we solve this? Is it possible to configure Asterisk so that the system thinks that a minute has 58 seconds instead of 60? _ MSN Amor: busca tu naranja http://latam.msn.com/amor/ ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and 2 line MGCP phone
Hi, -Original Message- Anybody here know or using Asterisk with 2 lines MGCP phone? I am trying to figure out if there are such device available and if so, how does it differenciate between the lines that is associated with extention number. Theoretically you could differentiate by the line: aaln/[EMAIL PROTECTED] aaln/[EMAIL PROTECTED] Are typical indications for this. I've never seen a phone that does this, though.. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nasty little incident ...
On Thu, 16 Jun 2005, Rich Adamson wrote: The E1 card does not receive clocking from any span. It sync's the on-board clock to whatever span you choose. If you watch what others have posted on the list over many months, you'll notice many have never specified a clock sync source. The problem they have is typically associated with clicking and other audio distortion; not a total failure. Thanks for the correction - by getting clocking I didn't mean anything more than syncing clock. Nevertheless, I have had customers, though, with dead TE410P setups - exactly this person's symptoms. The cause was having a span selected as the sync span but having nothing connected to that port. Adjust zaptel.conf so that span is not a sync source (0 in position 2) and the board starts to work. Steve ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] #(transfer) no longer working
Anyone who can help me with this ?
I tried everything :(
On 14:26, Tue 14 Jun 05, Michiel van Baak wrote:
Hi list,
For months everything worked super here in our setup.
This week I implemented some new idea in our webbased
calendar system. I thought it would be nice to have an
option that tells asterisk you are not available for calls
during an appointment.
For this to work I could no longer use the ringgroup setup:
Dial(SIP/10SIP/11SIP/12,40,tr)
So I thought, why not use the Local channel and a smaal
macro for each device so we can check a dbfield and decide
if we can call the device or not.
But now I cannot transfer calls with the # key anymore. We
use this a lot to put ppl in a parkedcall slot.
Here is my setup (incoming number obfuscated):
(phones 11 and 12 match 10, only different
defaultip,username,secret
Any idea what I miss ?
sip.conf:
[general]
context=default
port=5060
bindaddr=0.0.0.0
srvlookup=yes
musicclass=default
[10]
host=dynamic
defaultip=192.168.1.91
type=friend
username=10
secret=secret
nat=yes
qualify=yes
context=terrazur
callgroup=2
pickupgroup=2
extensions.conf
[general]
static=yes
writeprotect=no
[macro-stdexten]
include = parkedcalls
exten = s,1,DBget(temp=CFIM/${ARG1})
exten = s,102,Dial(${ARG1},,Ttr)
[remote]
include = parkedcalls
;Incoming lines.
exten = 31X,1,SetCallerID(${CALLERID})
exten = 31X,2,Agi(covide.agi)
exten = 31X,3,Goto,ringgroup-terrazur|s|1
[ringgroup-terrazur]
include = parkedcalls
exten = s,1,Wait,1
exten = s,2,DigitTimeout,5
exten = s,3,ResponseTimeout,10
exten = s,4,Dial(Local/[EMAIL PROTECTED],5,tTr)
exten = s,5,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED],10,tTr)
exten = s,6,Dial(Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]Local/[EMAIL PROTECTED],40,Ttr)
[default]
[terrazur]
include = parkedcalls
include = remote
include = speakup-out
exten = 10,1,Macro(stdexten,SIP/10)
exten = 11,1,Macro(stdexten,SIP/11)
exten = 12,1,Macro(stdexten,SIP/12)
exten = 701,1,ParkedCall(701)
exten = 702,1,ParkedCall(702)
exten = 703,1,ParkedCall(703)
features.conf:
[general]
parkext = 700
parkpos = 701-720
context = parkedcalls
parkingtime = 999
pickupexten = *8
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--
Michiel van Baak
http://michiel.vanbaak.info
[EMAIL PROTECTED]
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x7E0B9A2D
Two of the most famous products of Berkeley are LSD and BSD. I don't think
that this is a coincidence.
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RE: [Asterisk-Users] Case studies for Asterisk Voicemail
Vonage uses Asterisk, and they have a lot more than 3000 customers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stearne Sent: Thursday, June 16, 2005 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Case studies for Asterisk Voicemail On 6/16/05, Alistair Cunningham [EMAIL PROTECTED] wrote: Several people have responded with architecture suggestions. While these are welcome, I'm happy with the architecture options planned, having done many large voicemail implementations on products other than Asterisk. What I had hoped to get from Asterisk-Users and Asterisk-Biz was not a technical discussion (though I don't mind getting this too), but reports of people who already have systems like this so I can put the customer's mind at ease that they're not the first people to use Asterisk voicemail with 3000 users. I should have made this more clear in my first email. It was clear developers just like to put their 2 cents in I guess. :-) What you might do is contact a comapny like http://www.broadvoice.com/ or http://voicepulse.com/ and just ask if they are using Asterisk for their voicemail systems for their customers. I would think they might be using Asterisk for their systems and they have the amounts of users you're looking at. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] iax2 registry - auto reconnect ?
I use the Teliax service with the IAX2 protocol. I noticed 2 days ago that I was not registered with the Teliax server. I used the iax2 show registry command and found I was not registered with Teliax. I issued a reload command in asterisk in order to connect again. I went to the Teliax website and noticed a message which stated clients may need to reboot due to changes made with their dns servers. My question is ... If an iax2 entry is not registered, will I fail to get inbound calls? Is there any way to have * automatically detect and re-register periodically? Thanks, Jim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and 2 line MGCP phone
Are you using, putting those lines in the mgcp.conf file, should handle two lines? Did anybody tried it? Thanks Quoting Florian Overkamp [EMAIL PROTECTED]: Hi, -Original Message- Anybody here know or using Asterisk with 2 lines MGCP phone? I am trying to figure out if there are such device available and if so, how does it differenciate between the lines that is associated with extention number. Theoretically you could differentiate by the line: aaln/[EMAIL PROTECTED] aaln/[EMAIL PROTECTED] Are typical indications for this. I've never seen a phone that does this, though.. Florian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users CM Rahman Jr. CTO CCS Internet www.ccsi.com ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Case studies for Asterisk Voicemail
On 6/16/05, Bill McLaughlin [EMAIL PROTECTED] wrote: Vonage uses Asterisk, and they have a lot more than 3000 customers. That should help your argument! Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Case studies for Asterisk Voicemail
Bill McLaughlin wrote: Vonage uses Asterisk, and they have a lot more than 3000 customers. ?? You have documentation of that assertion? Not saying you're wrong, but I've never seen such a thing before. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Case studies for Asterisk Voicemail
Are you sure about that? I know Freshtel.net uses a highly customized version of asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Bill McLaughlin Sent: Thursday, 16 June 2005 2:12 PM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Case studies for Asterisk Voicemail Vonage uses Asterisk, and they have a lot more than 3000 customers. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Stearne Sent: Thursday, June 16, 2005 11:53 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Case studies for Asterisk Voicemail On 6/16/05, Alistair Cunningham [EMAIL PROTECTED] wrote: Several people have responded with architecture suggestions. While these are welcome, I'm happy with the architecture options planned, having done many large voicemail implementations on products other than Asterisk. What I had hoped to get from Asterisk-Users and Asterisk-Biz was not a technical discussion (though I don't mind getting this too), but reports of people who already have systems like this so I can put the customer's mind at ease that they're not the first people to use Asterisk voicemail with 3000 users. I should have made this more clear in my first email. It was clear developers just like to put their 2 cents in I guess. :-) What you might do is contact a comapny like http://www.broadvoice.com/ or http://voicepulse.com/ and just ask if they are using Asterisk for their voicemail systems for their customers. I would think they might be using Asterisk for their systems and they have the amounts of users you're looking at. Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
