Re: [asterisk-users] howto set up persistent dynamic meetme
Sean wrote: Tilghman Lesher wrote: On Saturday 16 May 2009 08:21:43 sean darcy wrote: With 1.6.1, I'm trying to set up a test of meetme for creating dynamic conferences. Trimmed I don't want the conference to stay up forever, since I'd like new pin's each time. This should be a common use case. How do you do it? In Asterisk 1.6, user DEA contributed realtime capabilities to MeetMe, which allows the capability of scheduling conferences, with new pins each time. I believe this would meet the needs your question has posed. I using 1.6.1. 'core show application meetme' doesn't have anything on realtime. I found http://www.voip-info.org/wiki/view/Asterisk+RealTime+MeetMe but that's just a stub. Any references available. I should point out that I (DEA) did not contribute the basic RealTime support in app_meetme. I added the scheduling and resource limit features, and the option to store the conference flags in the database table. You will want to understand the basics of Asterisk's RealTime features to get started. Basic support for RealTime conferences can be had by using the database table defined in contrib/scripts/meetme.sql The scheduling features require a more complex database table that I was sure that I included in the contribution, but I do not see it in SVN. The correct db table is available in the Web-MeetMe package, which is a front-end to manage the database and active conferences. It is hosted at https://sf.net/projects/web-meetme I am behind schedule to release a package for 1.6.1, and I need to submit the database table to Mantis so it can be added to future release packages. Dan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TODAY May 17 Sunday Asterisk VOIP Conference server Ekiga for BerkeleyTIP
As usual, the BerkeleyTIP group programming project is to learn about work toward implementing our own Asterisk VOIP conference server, during our meeting. 10A-6P Pacific USA (-8H GMT) = 1P-9P Eastern US = 6P-2AM GMT. http://sites.google.com/site/berkeleytip/ Join the VOIP online conference help out, or chat with your buddies. :) http://sites.google.com/site/berkeleytip/remote-attendance #berekeleytip on freenode.irc.net Join the mailing list. http://groups.google.com/group/BerkTIPGlobal == This message is mainly for sending to the Ekiga Asterisk mailing lists. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Correction IRC Channel name - was TODAY May 17 Sunday Asterisk VOIP Conference server Ekiga for BerkeleyTIP
http://sites.google.com/site/berkeleytip/ Join the VOIP online conference help out, or chat with your buddies. :) http://sites.google.com/site/berkeleytip/remote-attendance er, correct spelling: Freenode channel #BerkeleyTIP, irc.freenode.net NOT - #berekeleytip on freenode.irc.net ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Capture Server header in SIP reply.
Hi,
I am trying to capture Server header in a 200 OK reply message.
My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)),
and inside of GetOtherPartyInfo macro use SIP_HEADER function.
For example:
[default]
exten = _X.,1,Dial(SIP/u...@domain,30,M(GetOtherPartyInfo))
exten = _X.,n,Hangup()
[macro-GetOtherPartyInfo]
exten = s,1,NoOp(SIP Server: ${SIP_HEADER(Server,1)})
unfortunately the above doesn't seem to work:
-- Executing [...@macro-getotherpartyinfo:1]
NoOp(SIP/dev-sip.domain.net-08dbb610, SIP src_server: ) in new
stack
Is there any way to capture SIP headers from reply messages generated
by a called party?
Regards,
Chris
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Re: [asterisk-users] Capture Server header in SIP reply.
On Sun, May 17, 2009 at 6:43 AM, Chris Maciejewski ch...@wima.co.uk wrote: I am trying to capture Server header in a 200 OK reply message. My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)), and inside of GetOtherPartyInfo macro use SIP_HEADER function. unfortunately the above doesn't seem to work: Is there any way to capture SIP headers from reply messages generated by a called party? http://www.voip-info.org/wiki/view/Asterisk+func+sip_header You might prefer the SIP_HEADER(FROM) field. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router
On Sat, May 16, 2009 at 10:22 AM, Timothy Smith timotsm...@gmail.com wrote:
I have finally managed to get voice working. I both parties can hear
each other. The problem was nating. Our network is fairly big and
these machines are atleast 2 switches from each other. I just enabled
it (nat=route or nat=yes) and it worked.
It's not yet done however. When I redirect a call to any Asterisk
application, it just hangs up! I have read some history and archives,
but none of the solutions has worked for me. e.g ip inspect udp
idle-time 900. My router (or IOS) doesn't have thet command.
Could you please assist point to what could be causing this and how to
solve it? Below are some logs and attached is the router log.
; This is the extension conf. Enter the extension you want to reach
now (something like auto attendant).
exten = _X.,1,Read(NUM,beep,4,2,3)
exten = _X.,n,Dial(SIP/${NUM})
; This is all i get when i call and the call hangs up!
Did you ever set up that reverse dial-peer? If not, do that first.
You put a three second timeout on the Read(). By any chance, is the
call hanging up 3 seconds after you call? That would be expected
behavior. Well, actually you give it two tries. So it should be
beep
three second wait
beep
three second wait
hangup
If you're actually entering numbers on your dialpad and they're not
getting read, you have a misconfiguration on your DTMF. If you enable
sip debugging on your asterisk side you can see exactly what's coming
over the wire from the Cisco side. There are a lot of choices for DTMF
on the asterisk side and the Cisco side, and they need to agree for
the button presses to be encoded and passed correctly. You can pass
them in-line as real audio, or you can convert them to a special dtmf
sip encoding. You'll notice all those choices when you go to configure
the Cisco dial-peer.
My personal preference:
on the Cisco dial-peer side
dtmf-relay rtp-nte
on the asterisk side
I left the dtmf config blank, and I don't remember which default you
end up with, but it worked in the default config for me.
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Re: [asterisk-users] Capture Server header in SIP reply.
Hi David,
Thanks for your post.
Unfortunately SIP_HEADER(FROM) is not an option for me.
What I want to do is record in CDRs User-Agent header of calling
party (this can be easily done with ${CHANNEL(useragent)}), and SIP
Server header of called party (from 200 OK response to INVITE
generated by Asterisk).
2009/5/17 David Backeberg dbackeb...@gmail.com:
On Sun, May 17, 2009 at 6:43 AM, Chris Maciejewski ch...@wima.co.uk wrote:
I am trying to capture Server header in a 200 OK reply message.
My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)),
and inside of GetOtherPartyInfo macro use SIP_HEADER function.
unfortunately the above doesn't seem to work:
Is there any way to capture SIP headers from reply messages generated
by a called party?
http://www.voip-info.org/wiki/view/Asterisk+func+sip_header
You might prefer the SIP_HEADER(FROM) field.
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Re: [asterisk-users] Capture Server header in SIP reply.
On Sun, May 17, 2009 at 9:04 AM, Chris Maciejewski ch...@wima.co.uk wrote:
Unfortunately SIP_HEADER(FROM) is not an option for me.
What I want to do is record in CDRs User-Agent header of calling
party (this can be easily done with ${CHANNEL(useragent)}), and SIP
Server header of called party (from 200 OK response to INVITE
generated by Asterisk).
Maybe you need a better name for it than server. To me server means
the hostname / address of the other side of the SIP conversation, aka:
FROM.
You can use SipAddHeader to make your own X-blah tags for your
packets, and then pick them off on the other side. I don't seem to
understand what you mean by 'server', despite my command of the
english language. Perhaps you want
${SIPUSERAGENT}
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Re: [asterisk-users] Capture Server header in SIP reply.
User-Agent header is present in SIP *request* i.e. INVITE received
by Asterisk from UAC.
RFC 3261 - 20.41 User-Agent
The User-Agent header field contains information about the UAC
originating the request. The semantics of this header field are
defined in [H14.43].
Revealing the specific software version of the user agent might allow
the user agent to become more vulnerable to attacks against software
that is known to contain security holes. Implementers SHOULD make
the User-Agent header field a configurable option.
Example:
User-Agent: Softphone Beta1.5
Server header is present in SIP *response* i.e. 200 OK generated
by UAS to INVITE generated by Asterisk.
RFC 3261 - 20.35 Server
The Server header field contains information about the software used
by the UAS to handle the request.
Revealing the specific software version of the server might allow the
server to become more vulnerable to attacks against software that is
known to contain security holes. Implementers SHOULD make the Server
header field a configurable option.
Example:
Server: HomeServer v2
My scenario:
Phone 1 - INVITE [1] - Asterisk -- INVITE [2] -- Phone 2
--- 200
OK [3] ---
What I want to do is capture Server header in 200 OK reply
generated by Phone 2.
2009/5/17 David Backeberg dbackeb...@gmail.com:
On Sun, May 17, 2009 at 9:04 AM, Chris Maciejewski ch...@wima.co.uk wrote:
Maybe you need a better name for it than server. To me server means
the hostname / address of the other side of the SIP conversation, aka:
FROM.
You can use SipAddHeader to make your own X-blah tags for your
packets, and then pick them off on the other side. I don't seem to
understand what you mean by 'server', despite my command of the
english language. Perhaps you want
${SIPUSERAGENT}
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Re: [asterisk-users] Capture Server header in SIP reply.
There is, actually, a Server header. It is the equivalent of User-
Agent for UASs.
--
Sent from mobile device
On May 17, 2009, at 9:19 AM, David Backeberg dbackeb...@gmail.com
wrote:
On Sun, May 17, 2009 at 9:04 AM, Chris Maciejewski
ch...@wima.co.uk wrote:
Unfortunately SIP_HEADER(FROM) is not an option for me.
What I want to do is record in CDRs User-Agent header of calling
party (this can be easily done with ${CHANNEL(useragent)}), and SIP
Server header of called party (from 200 OK response to INVITE
generated by Asterisk).
Maybe you need a better name for it than server. To me server means
the hostname / address of the other side of the SIP conversation, aka:
FROM.
You can use SipAddHeader to make your own X-blah tags for your
packets, and then pick them off on the other side. I don't seem to
understand what you mean by 'server', despite my command of the
english language. Perhaps you want
${SIPUSERAGENT}
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Re: [asterisk-users] Capture Server header in SIP reply.
It is fairly trivial to modify chan_sip to expose headers from final
replies to the SIP_HEADER container or some other channel variables.
Just make sure to disambiguate names of headers if you put them in a
conflicting namespace.
There is one big switch statement that dispatches handling behaviour
for various message types, including final replies. Go in the 200 OK
handler for INVITEs and add some magic.
By default, only headers from initial INVITE request can be accessed.
But, you can change that.
I would be inclined to ask if Asterisk is the right tool for the job,
though, if you need to go that low-level.
--
Sent from mobile device
On May 17, 2009, at 9:04 AM, Chris Maciejewski ch...@wima.co.uk wrote:
Hi David,
Thanks for your post.
Unfortunately SIP_HEADER(FROM) is not an option for me.
What I want to do is record in CDRs User-Agent header of calling
party (this can be easily done with ${CHANNEL(useragent)}), and SIP
Server header of called party (from 200 OK response to INVITE
generated by Asterisk).
2009/5/17 David Backeberg dbackeb...@gmail.com:
On Sun, May 17, 2009 at 6:43 AM, Chris Maciejewski
ch...@wima.co.uk wrote:
I am trying to capture Server header in a 200 OK reply message.
My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)),
and inside of GetOtherPartyInfo macro use SIP_HEADER function.
unfortunately the above doesn't seem to work:
Is there any way to capture SIP headers from reply messages
generated
by a called party?
http://www.voip-info.org/wiki/view/Asterisk+func+sip_header
You might prefer the SIP_HEADER(FROM) field.
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[asterisk-users] SHARED() variables and ZOMBIE channel
Hi,
I am using SHARED() function to push destination channel info (i.e.
audio codec) into source channel, in order to record into a customer
CDR field.
My dialplan looks like:
[default]
exten = _X.,1,Set(_X-SRC_CHANNEL=${CHANNEL})
exten = _X.,n,Dial(SIP/u...@domain.net,30,M(getCalledInfo))
exten = h,1,Set(CDR(DST_CODEC)=${SHARED(X-DST-CODEC,${CHANNEL})})
[macro-getCalledInfo]
exten =
s,1,Set(SHARED(X-DST-CODEC,${X-SRC_CHANNEL})=${CHANNEL(audionativeformat)})
The above works great, however there is a problem when call is
transferred via SIP attended transfer and channel is renamed to
ChannelZOMBIE.
-- Executing [...@default:1] Set(SIP/somechannelZOMBIE,
CDR(DST_CODEC)=) in new stack
Is there any workaround for the above issue?
Regards,
Chris
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Re: [asterisk-users] Agent-Login/out in 1.6
The main problem i see with thgis is that with old-school agents, you
could easily have association with multiple queues. And that was quite
useful.
l.
2009/5/16 David Anthony O Reilly oreil...@tcd.ie
Hi Jim
Thanks for your code!! I see you use the Voicemail system to authenticate,
have you ever managed to avoid that as I don't use voicemail at all and I am
thinking if I use that solution I will need to set up a voicemail for all
the queue members just to get them to log in.
hehe What were the developers thinking by removing the old system! It
worked perfect!! and by the looks of it nobody has ever recovered from the
command removal unless they hack around with the voicemail system.
Hopefully somebody out there has managed to create an agent login/logout
without bringing voicemail into it If I find a way I will let you and
post a wiki on it as I am sure loads of people have this problem.
Thanks
Dave
; Agent login logout
exten = *20,1,Answer()
exten = *20,n,wait(.0.5)
exten = *20,n,Read(AgentNumber,agent-user)
exten = *20,n,Set(UserID=${DB(ExtenToUser/${AgentNumber})})
exten = *20,n,GotoIf($[${UserID}=]?NOUSER)
exten = *20,n,Set(AgentStatus=${DB(users/${UserID}/AgentStatus)})
exten = *20,n,GotoIf($[${AgentStatus}=1]?VERIFY)
exten = *20,n,GotoIf($[${AgentStatus}=2]?VERIFY)
exten = *20,n(NOUSER),Playback(cfmc/bad-agent)
exten = *20,n,Playback(vm-goodbye)
exten = *20,n,Hangup()
exten = *20,n(VERIFY),VMAuthenticate(${agentnumb...@ourvm)
exten = *20,n,GotoIf($[${AgentStatus}=2]?AGENTOFF)
exten = *20,n,Set(DB(users/${UserID}/AgentStatus)=2)
exten = *20,n,Set(DB(users/${UserID}/AgentDevice)=${CUT(CHANNEL,-,1)})
exten =
*20,n,AddQueueMember(support,Local/queue${agentnumb...@ansqueue
${CUT(CHA
NNEL,-,1)})
; AQMSTATUS can be ADDED | MEMBERALREADY | NOSUCHQUEUE
exten = *20,n,Playback(agent-loginok)
exten = *20,n,Verbose(2,Agent ${AgentNumber} added
${DB(users/${UserID}/AgentDevice)})
exten = *20,n,HangUp()
exten = *20,n(AGENTOFF),Set(DB(users/${UserID}/AgentStatus)=1)
exten = *20,n,Set(OldVal=${DB_DELETE(users/${UserID}/AgentDevice)})
exten = *20,n,RemoveQueueMember(support,Local/queue${agentnumb...@ansqueue
)
exten = *20,n,Playback(agent-loggedoff)
exten = *20,n,Verbose(2,Agent ${AgentNumber} removed)
exten = *20,n,Hangup()
--
_
Mr. David Anthony O'Reilly, B.Sc Comp (Hons)
M.Sc MOB Postgraduate @ University College Cork, Ireland - M.Sc (Mob) -
2009
Computer Science Graduate of The University of Dublin, Trinity College -
B.Sc (Comp) 2008
Email: oreil...@tcd.ie/d...@student.cs.ucc.ie
Tel: +353 (0) 86 030 60 32
_
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Loway - home of QueueMetrics - http://queuemetrics.com
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Re: [asterisk-users] Agent-Login/out in 1.6
Lenz Emilitri schrieb:
The main problem i see with thgis is that with old-school agents, you
could easily have association with multiple queues. And that was quite
useful.
2009/5/16 David Anthony O Reilly oreil...@tcd.ie
Hopefully somebody out there has managed to create an agent login/logout
without bringing voicemail into it If I find a way I will let you and
post a wiki on it as I am sure loads of people have this problem.
Logging into multiple queues seems perfectly doable to me, and we
do that in Gemeinschaft (http://www.amooma.de/gemeinschaft/ , german).
I'll try to quote the relevant parts from extensions.ael
(https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/etc/asterisk/e.ael):
//-
// Queue Login/Logout
//-
macro queue-login( queue ) {
Set(CDR(amaflags)=OMIT);
Answer();
Verbose(1,### User ${user_name} wants to log in to queue ${queue});
AGI(/opt/gemeinschaft/dialplan-scripts/queue-login-logout.agi,${user_name},${queue},login);
if (${agent_login_status} = loggedin) {
// fake login for the statistics:
QueueLog(NONE,${UNIQUEID},${CHANNEL(channeltype)}/${user_name},AGENTLOGIN,fake);
}
Verbose(1,### agent_login_status: ${agent_login_status});
Wait(0.5);
if (${agent_login_status} = loggedin || ${agent_login_status} =
alreadyon) {
Playback(agent-loginok);
} else {
Playback(beeperr);
}
Hangup();
}
macro queue-logout( queue ) {
Set(CDR(amaflags)=OMIT);
Answer();
Verbose(1,### User ${user_name} wants to log out from queue ${queue});
AGI(/opt/gemeinschaft/dialplan-scripts/queue-login-logout.agi,${user_name},${queue},logout);
Wait(0.5);
if (${agent_login_status} = loggedout) {
Playback(agent-loggedoff);
} else {
Playback(beeperr);
}
Hangup();
}
macro queue-logout-all() {
Set(CDR(amaflags)=OMIT);
Answer();
Verbose(1,### User ${user_name} wants to log out from all queues);
AGI(/opt/gemeinschaft/dialplan-scripts/queue-login-logout.agi,${user_name},0,logoutall);
Wait(0.5);
if (${agent_login_status} = loggedout) {
Playback(agent-loggedoff);
} else {
Playback(beeperr);
}
Hangup();
}
macro queue-logout-all-silent() {
// like queue-logout-all() but silent
Set(CDR(amaflags)=OMIT);
Verbose(1,### User ${user_name} wants to log out from all queues);
AGI(/opt/gemeinschaft/dialplan-scripts/queue-login-logout.agi,${user_name},0,logoutall);
}
context queues-login-logout {
*5* = queue-logout-all();
_*5. = queue-login(${EXTEN:2});
//_*5.* = queue-logout(${EXTEN:2}); // does not work
_*5X* = queue-logout(${EXTEN:2:-1});
_*5XX* = queue-logout(${EXTEN:2:-1});
_*5XXX* = queue-logout(${EXTEN:2:-1});
_*5* = queue-logout(${EXTEN:2:-1});
_*5X* = queue-logout(${EXTEN:2:-1});
_*5XX* = queue-logout(${EXTEN:2:-1});
}
Instead of ${user_name} you might want to use something like
${CALLERID(num)} if you don't use Gemeinschaft.
The queue-login-logout.agi AGI script
(https://svn.amooma.com/gemeinschaft/trunk/opt/gemeinschaft/dialplan-scripts/queue-login-logout.agi)
inserts/removes the interface (SIP/...) into/from the queue_members
Realtime family which is stored in a MySQL database.
Log in to a queue by dialing *5queuenumber e.g. *5300. You're
not limited to only one queue at a time.
Log out of a specific queue by dialing *5queuenumber*.
Log out of all queues by dialing *5*.
Philipp Kempgen
--
AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] SHARED() variables and ZOMBIE channel
On Sunday 17 May 2009 09:10:47 Chris Maciejewski wrote:
Hi,
I am using SHARED() function to push destination channel info (i.e.
audio codec) into source channel, in order to record into a customer
CDR field.
My dialplan looks like:
[default]
exten = _X.,1,Set(_X-SRC_CHANNEL=${CHANNEL})
exten = _X.,n,Dial(SIP/u...@domain.net,30,M(getCalledInfo))
exten = h,1,Set(CDR(DST_CODEC)=${SHARED(X-DST-CODEC,${CHANNEL})})
[macro-getCalledInfo]
exten =
s,1,Set(SHARED(X-DST-CODEC,${X-SRC_CHANNEL})=${CHANNEL(audionativeformat)})
The above works great, however there is a problem when call is
transferred via SIP attended transfer and channel is renamed to
ChannelZOMBIE.
-- Executing [...@default:1] Set(SIP/somechannelZOMBIE,
CDR(DST_CODEC)=) in new stack
Is there any workaround for the above issue?
I suppose you could use CUT to guarantee that the ZOMBIE portion won't show
up in the channel name, i.e.
exten = h,1,Set(CDR(DST_CODEC)=${SHARED(X-DST-CODEC,${CUT(CHANNEL,,1)})})
--
Tilghman
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[asterisk-users] Fwd: Asterisk With Cisco Voice Router
Hi, We have AS5400's set up with asterisk boxes. Initially we had similar issues, but as described, you need to have dial peers to handle both incoming and outgoing peers. Please post your dial peer configs as well as the serial interface configs. I also found that until I add [isdn incoming-voice modem ] I could not get incoming calls on that serial interface to route to my * box. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of asterisk-users-requ...@lists.digium.com Sent: Saturday, 16 May 2009 10:00 AM To: asterisk-users@lists.digium.com Subject: asterisk-users Digest, Vol 58, Issue 40 Send asterisk-users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-requ...@lists.digium.com You can reach the person managing the list at asterisk-users-ow...@lists.digium.com When replying, please edit your Subject line so it is more specific than Re: Contents of asterisk-users digest... Today's Topics: 1. Fwd: Asterisk With Cisco Voice Router (Timothy Smith) 2. Re: Fwd: Asterisk With Cisco Voice Router (Steve Howes) 3. Re: Fwd: Asterisk With Cisco Voice Router (Timothy Smith) 4. Re: Fwd: Asterisk With Cisco Voice Router (David Backeberg) 5. Re: meetme dies looking for conf-getconfno (sean darcy) 6. howto set up persistent dynamic meetme (sean darcy) 7. Agent-Login/out in 1.6 (David Anthony O Reilly) 8. Agent-Login/out in 1.6 (David Anthony O Reilly) 9. Re: Agent-Login/out in 1.6 (Stefan Reuter) 10. Re: Fwd: Asterisk With Cisco Voice Router (Timothy Smith) 11. Re: Agent-Login/out in 1.6 (Jim Dickenson) 12. Re: howto set up persistent dynamic meetme (Tilghman Lesher) 13. Re: Fwd: Asterisk With Cisco Voice Router (Philipp Kempgen) -- Message: 1 Date: Sat, 16 May 2009 14:46:27 +0300 From: Timothy Smith timotsm...@gmail.com Subject: [asterisk-users] Fwd: Asterisk With Cisco Voice Router To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 416fc8170905160446r5815fd87m67e62506ad9ac...@mail.gmail.com Content-Type: text/plain; charset=iso-8859-1 Hi, In our office, we're slowly migrating from a cisco call manager set up to asterisk. Problem is management doesn't want to buy any other hardware ?as they had already invested a lot in cisco. The main cause of this is asterisk's added features like unique FAX number for everyone in the company (which will be the same as phone DID), Voice mail, Auto Answer etc yet we need thousands of dollars to add those to our cisco call manager 4.1 set up. I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9), and also a dialpeer to forward on the router to forward calls to my asterisk. It works properly but the problem is there is NO AUDIO! I have tried to change codec but no sucess! Has anyone had the above set up working successfully? Attached are some confs. Thanks a lot for your assistance. Kind Regards, Wilson -- next part -- An embedded and charset-unspecified text was scrubbed... Name: show-run.txt Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/67 2c1104/attachment-0002.txt -- next part -- cs-intranet*CLI sip show peers Name/username HostDyn Nat ACL Port Status 103172.17.3.2495060 OK (3 ms) 102172.17.3.2485060 OK (3 ms) 101172.17.10.150 5060 OK (1 ms) 100/100172.19.4.102 D N 32544 Unmonitored 4 sip peers [Monitored: 3 online, 0 offline Unmonitored: 1 online, 0 offline] ; 102 and 103 are cisco routers, 101 is the call manager, 100 is a SIP phone -- next part -- An embedded and charset-unspecified text was scrubbed... Name: show-dialpeer.txt Url: http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/67 2c1104/attachment-0003.txt -- next part -- A non-text attachment was scrubbed... Name: sip.conf Type: application/octet-stream Size: 3327 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090516/67 2c1104/attachment-0001.obj -- Message: 2 Date: Sat, 16 May 2009 13:25:40 +0100 From: Steve Howes st...@geekinter.net Subject: Re: [asterisk-users] Fwd: Asterisk With Cisco Voice Router To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 659dc612-4035-4d7e-a73c-77b5a16d6...@geekinter.net Content-Type: text/plain;
Re: [asterisk-users] howto set up persistent dynamic meetme
Dan Austin wrote:
Sean wrote:
Tilghman Lesher wrote:
On Saturday 16 May 2009 08:21:43 sean darcy wrote:
With 1.6.1, I'm trying to set up a test of meetme for creating dynamic
conferences.
Trimmed
I don't want the conference to stay up forever, since I'd like new pin's
each time.
This should be a common use case. How do you do it?
In Asterisk 1.6, user DEA contributed realtime capabilities to MeetMe, which
allows the capability of scheduling conferences, with new pins each time. I
believe this would meet the needs your question has posed.
I using 1.6.1. 'core show application meetme' doesn't have anything on
realtime. I found
http://www.voip-info.org/wiki/view/Asterisk+RealTime+MeetMe but that's
just a stub.
Any references available.
I should point out that I (DEA) did not contribute the basic RealTime
support in app_meetme. I added the scheduling and resource limit features,
and the option to store the conference flags in the database table.
You will want to understand the basics of Asterisk's RealTime features to
get started. Basic support for RealTime conferences can be had by using
the database table defined in contrib/scripts/meetme.sql
The scheduling features require a more complex database table that I was
sure that I included in the contribution, but I do not see it in SVN.
The correct db table is available in the Web-MeetMe package, which is
a front-end to manage the database and active conferences. It is hosted
at https://sf.net/projects/web-meetme
I am behind schedule to release a package for 1.6.1, and I need to submit
the database table to Mantis so it can be added to future release packages.
Dan
Dan,
Thanks. I'll follow Web-MeetMe. It's pretty neat to be able to set up
a conference from wherever. Right now, though, I'd settle for setting it
up from an office extension, I've written this hacky snippet to set them up:
[setup-conf-room]
exten = _6000XXXNXXX,1,Set(meetme=/etc/asterisk/meetme.conf)
exten = _6000XXXNXXX,n,Set(Time-in-secs=${STRFTIME(${EPOCH},,%s} )
exten = _6000XXXNXXX,n,System('echo conf = ${EXTEN:4:4},${EXTEN:7} ;
_${Time-in-secs} ${meetme}')
exten = _6000XXXNXXX,n,Playback(our-conf-confno)
exten = _6000XXXNXXX,n,SayDigits(${EXTEN:4:4} ) ; 6000 XXX
exten = _6000XXXNXXX,n,Playback(our-conf-pin-for-conf)
exten = _6000XXXNXXX,n,SayDigits(${EXTEN:7} ) ; 6000 xxx
exten = _6000XXXNXXX,n,Playback(our-conf-confup-24-hours)
exten = _6000XXXNXXX,n,Hangup()
exten = _6000.,1,Playback(our-conf-invalid-plus-3-plus-4)
exten = _6000.,2,Hangup()
where a user dials 6000{3digit-conf-rm}{4digit-pin)
then I'll use a cron.hourly job to delete the conf after 24 hours.
I'd like to find some way to check for conflicting existing conference
numbers.
sean
sean
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Re: [asterisk-users] Is anyone keeping up with the versions?
On Tue, May 12, 2009 at 3:05 PM, James A. Shigley j...@answeringserv.com wrote: Unless there is a new feature or your making a new system. Don’t fix it if it aint broke. BUT do stay current on reading about new feature and things in the releases. James Shigley From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle Dupuis Sent: Tuesday, May 12, 2009 1:45 PM To: 'Asterisk Users List' Subject: Re: [asterisk-users] Is anyone keeping up with the versions? Pick a release and stick with it as long as you can. Only when you have to jump, pick a new release, test the hell out of it, and then leave it alone. Too many people try to keep on the latest release... From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thermal Wetland Sent: Tuesday, May 12, 2009 2:32 PM To: Asterisk Users List Subject: [asterisk-users] Is anyone keeping up with the versions? We are still using 1.4 and were going to start testing with 1.6.0, but then 1.6.1 was released and now 1.6.2 is already in beta 2. That seems like a lot of independent releases to maintain. I read about all the regressions ans hurried dot releases, makes us nervous. How is everyone doing their testing? -Matt ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users But I'm concerned about getting stranded. I'm now on 1.4.24.1, but I'm considering going to 16.0.9 _before_ I have to. That means I can always go back to 1.4.x before it's EOL if I have any problems. It seems to me that more and more security/bug issues will be dealt with faster in the 1.6.x releases. sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SHARED() variables and ZOMBIE channel
2009/5/17 Tilghman Lesher tilgh...@mail.jeffandtilghman.com:
On Sunday 17 May 2009 09:10:47 Chris Maciejewski wrote:
Hi,
I am using SHARED() function to push destination channel info (i.e.
audio codec) into source channel, in order to record into a customer
CDR field.
My dialplan looks like:
[default]
exten = _X.,1,Set(_X-SRC_CHANNEL=${CHANNEL})
exten = _X.,n,Dial(SIP/u...@domain.net,30,M(getCalledInfo))
exten = h,1,Set(CDR(DST_CODEC)=${SHARED(X-DST-CODEC,${CHANNEL})})
[macro-getCalledInfo]
exten =
s,1,Set(SHARED(X-DST-CODEC,${X-SRC_CHANNEL})=${CHANNEL(audionativeformat)})
The above works great, however there is a problem when call is
transferred via SIP attended transfer and channel is renamed to
ChannelZOMBIE.
-- Executing [...@default:1] Set(SIP/somechannelZOMBIE,
CDR(DST_CODEC)=) in new stack
Is there any workaround for the above issue?
I suppose you could use CUT to guarantee that the ZOMBIE portion won't show
up in the channel name, i.e.
exten = h,1,Set(CDR(DST_CODEC)=${SHARED(X-DST-CODEC,${CUT(CHANNEL,,1)})})
I tried that already, but Asterisk throws the following error:
-- Executing [...@default:1] Set(SIP/OpenSER-0831a618ZOMBIE,
X-CHAN-NAME=SIP/OpenSER-0831a618) in new stack
[May 17 18:24:32] ERROR[6101]: func_global.c:106 shared_read: Channel
'SIP/OpenSER-0831a618' not found! Variable 'X-DST-CODEC' will be
blank.
as OpenSER-0831a618 doesn't exist any more. Looks like maybe SHARED()
variables are not inherited by ZOMBIE channel?
--
Tilghman
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Re: [asterisk-users] Not receiving voicemail message in mailbox
I have it up and running o my system with this line in voicemail.conf and a symlink named sendmail to the actual msmtp program. mailcmd=/usr/sbin/sendmail -v -t -f lmy e-ail name@gmail.com This is the install log based on info provided by others and the install process itself. Hope it is useful /Erik Step 1 Installing needed packages/libs on your system install this packages (I'm not sure if all the packages are needed but with this packages it works) apt-get install libwww-perl apt-get install openssl apt-get install libcrypt-ssleay apt-get install libnet-ssleay-perl apt-get install libcrypt-ssleay-perl Step 2 download msmtp download msmtp van sourceforge (http://sourceforge.net/projects/ msmtp/) to /usr/src/ Step 3 bunzip2 msmtp.tar.bz2 tar -xvf msmtp.tar cd /usr/src/msmtp Step 4 built msmtp ./configure make make install Step 5 check if msmtp is on the system and if the output looks like below. # msmtp --version msmtp version 1.4.9 TLS/SSL library: GnuTLS Authentication library: GNU SASL Supported authentication methods: plain cram-md5 digest-md5 gssapi external login IDN support: enabled NLS: enabled, LOCALEDIR is /usr/share/locale System configuration file name: /etc/msmtprc User configuration file name: /root/.msmtprc Copyright (C) 2006 Martin Lambers and others. This is free software. You may redistribute copies of it under the terms of the GNU General Public License http://www.gnu.org/licenses/gpl.html. There is NO WARRANTY, to the extent permitted by law. Step 6 Make a symlink from /usr/local/bin/msmtp to /usr/sbin/sendmail (the name of the symlink is sendmail) # ln -s /usr/local/bin/msmtp /usr/sbin/sendmail Step 7 Add /root/.msmtprc (be aware of the dot) to the system with only owner read and write permissions and with this lines (adjust to your x...@gmail.com account). This way it works for a gmail account defaults logfile /var/log/msmtp.log account default from xx@gmail.com protocol smtp host smtp.gmail.com port 587 user xxx@gmail.com password password auth on tls on tls_certcheck on tls_trust_file /root/cert.pem Step 8 certificate file copy the certificate file to the root directory /root/cert.pem copied on system (see attachement) Step 9 configuration of /etc/asterisk/voicemail.conf Add this to /etc/asterisk/voicemail.conf as a replacement of the mailcmd = line mailcmd=/usr/sbin/sendmail -v -t -f your_gemail_name@gmail.com and uncomment attach = yes Add a vociemailbox to the system in [default] of voicemail.conf [default] ; Define maximum number of messages per folder for a particular context. ;maxmsg=50 500 = 1234,name,e-mail adress step 10 adding a test extension to the system Add an extension to /etc/asterisk/extension.conf to test de setup something like exten = 888,1,Answer() exten = 888,n,Voicemail(500) If you call 888 with in internal phone you enter the voicemail routine and a recording will be made. After finishing you will receive an e- mail with the recording as an attachement. And you are done Message: 2 Date: Sat, 16 May 2009 21:47:58 +0200 From: jonas kellens jonas.kell...@telenet.be Subject: Re: [asterisk-users] Not receiving voicemail message in mailbox To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: 1242503278.3667.4.ca...@localhost.localdomain Content-Type: text/plain; charset=us-ascii I have put the following in my voicemail.conf-file : mailcmd=/usr/local/bin/msmtp -d --syslog=on -d and syslog=on are to debug some information, because I am still not receiving my voicemail-messages via mail as an attachment ! I don't know which mailcommand I need to put here to make Asterisk use msmtp as 'mailing server'. It is currently not working... The logfile /root/.msmtp.log is not mentioning anything. I think this is because Asterisk is really not using msmtp to send the message. Can someone help me figure this out... ? Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Load, Asterisk Disconnected
I have Asterisk 1.2.29, Zaptel 1.2.24 , TE 121P Digium Card, and Freepbx Setup
for a queue up to 15 agents through a PRI line, it was working fine for more
than 1 year, suddenly, when there is a load on the queue, the asterisk service
disconnects and the calls are dropped. And the service starts again after few
seconds, and so on.
I am not using fax.
I checked PRI by zttool and there are no alarms.
The cdr logs
2195,2,,Busy,,2009-05-13 10:06:53,,2009-05-13 10:06:55,2,0,NO
ANSWER,DOCUMENTATION
,0225167604,237,from-internal,0225167604
0225167604,Local/2...@from-internal-a07f,2,,Busy,,2009-05-13
10:06:55,,2009-05-13 10:06:57,2,0,NO ANSWER,DOCUMENTATION
,0225167604,229,from-internal,0225167604
0225167604,Local/2...@from-internal-c662,2,,Busy,,2009-05-13
10:06:57,,2009-05-13 10:06:59,2,0,NO ANSWER,DOCUMENTATION
,0225167604,224,from-internal,0225167604
0225167604,Local/2...@from-internal-3a3a,2,,Busy,,2009-05-13
10:06:59,,2009-05-13 10:07:00,1,0,NO ANSWER,DOCUMENTATION
/usr/sbin/safe_asterisk: line 57: 14229 Segmentation fault (core dumped)
${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
cat: /var/run/asterisk.pid: No such file or directory
Automatically restarting Asterisk.
Verbosity logs:
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090513-092729|1242196049.184: Inbound recording enabled.
recordingcheck|20090513-092729|1242196049.184: CALLFILENAME=1242196049.184
-- AGI Script recordingcheck completed, returning 0
-- Executing Monitor(Local/2...@from-internal-b759,2, wav49|1242196049.184|
mb) in new stack
-- Executing Macro(Local/2...@from-internal-b759,2, dial|30|Ttr|211) in new
stack
-- Executing AGI(Local/2...@from-internal-b759,2, dialparties.agi) in new
stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
dialparties.agi: Starting New Dialparties.agi
-- dialparties.agi: priority is 1
dialparties.agi: Caller ID name is '0227559600' number is '0227559600'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 211 to extension map
-- dialparties.agi: Extension 211 cf is disabled
dialparties.agi: Extension 211 has do not disturb enabled
-- AGI Script dialparties.agi completed, returning 0
-- Executing NoOp(Local/2...@from-internal-b759,2, Returned from dialparties
with no extensions to call) in new stack
-- Executing Set(Local/2...@from-internal-b759,2, DIALSTATUS=BUSY) in new
stack
-- Executing GotoIf(Local/2...@from-internal-b759,2, 1?s-BUSY|1) in new
stack
-- Goto (macro-exten-vm,s-BUSY,1)
-- Executing NoOp(Local/2...@from-internal-b759,2, Extension is reporting
BUSY and has no Voicemail) in new stack
-- Executing Busy(Local/2...@from-internal-b759,2, ) in new stack
-- Local/2...@from-internal-b759,1 is busy
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on
'Local/2...@from-internal-b759,2' in macro 'exten-vm'
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on
'Local/2...@from-internal-b759,2'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on
'Local/2...@from-internal-6cb4,2'
== Spawn extension (ext-queues, 100, 7) exited non-zero on 'Zap/25-1'
-- Hungup 'Zap/25-1'
-- Stopped music on hold on Zap/27-1
-- Playing periodic announcement
-- Playing 'custom/Busy' (language 'en')
-- Called Local/2...@from-internal/n
-- Executing Macro(Local/2...@from-internal-e5d7,2, exten-vm|novm|221) in
new stack
-- Executing Macro(Local/2...@from-internal-e5d7,2, user-callerid) in new
stack
-- Executing Set(Local/2...@from-internal-e5d7,2, AMPUSER=) in new stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 0?109) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2, EMERGENCYCID=) in new
stack
-- Executing Set(Local/2...@from-internal-e5d7,2, AMPUSERCIDNAME=) in new
stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 1?7) in new stack
-- Goto (macro-user-callerid,s,7)
-- Executing NoOp(Local/2...@from-internal-e5d7,2, Using CallerID
0227559600 0227559600) in new stack
-- Executing Set(Local/2...@from-internal-e5d7,2, FROMCONTEXT=exten-vm) in
new stack
-- Executing Macro(Local/2...@from-internal-e5d7,2, record-enable|221|IN)
in new stack
-- Executing GotoIf(Local/2...@from-internal-e5d7,2, 0 0?2:4) in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI(Local/2...@from-internal-e5d7,2,
recordingcheck|20090513-092731|1242196051.186) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20090513-092731|1242196051.186: Inbound recording enabled.
recordingcheck|20090513-092731|1242196051.186: CALLFILENAME=1242196051.186
-- AGI Script recordingcheck completed, returning 0
-- Executing
[asterisk-users] Can YOU find a trailing parenthesis?
On 1.6.1, I must be losing my eyesight:
[internal]
include = outbound-pstn
.
include = meetme; 2663
include = setup-meetme-conf-room ; 6000xxx
[setup-meetme-conf-room]
exten = _6000XXXNXXX,n,Set(Time-in-secs=${STRFTIME(${EPOCH},,%s} )
CLI:
-- Starting simple switch on 'DAHDI/1-1'
[2009-05-17 14:54:49] WARNING[13433]: pbx.c:2846 func_args: Can't find
trailing parenthesis?
-- Executing [60001234...@internal:1] Set(DAHDI/1-1,
Time-in-secs= 1242586489 ) in new stack
.
I've tried it with and without quotes around STRFTIME.
Now it works, so I can't really complain,
But I'm worried this come bite me later, somehow.
sean
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Re: [asterisk-users] Can YOU find a trailing parenthesis?
On Sun, 17 May 2009, sean darcy wrote:
[setup-meetme-conf-room] exten =
_6000XXXNXXX,n,Set(Time-in-secs=${STRFTIME(${EPOCH},,%s} )
CLI:
-- Starting simple switch on 'DAHDI/1-1' [2009-05-17 14:54:49]
WARNING[13433]: pbx.c:2846 func_args: Can't find trailing parenthesis?
This seems too obvious...
exten = _6000XXXNXXX,n,Set(Time-in-secs=${STRFTIME(${EPOCH},,%s} )
exten = _6000XXXNXXX,n,Set(Time-in-secs=${STRFTIME(${EPOCH},,%s)})
I didn't test it, but this looks right to me. I don't think you want a
trailing space assigned to Time-in-secs either.
Emacs has an excellent match-paren facility that works on (){}[] in any
nesting combination.
Thanks in advance,
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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[asterisk-users] Calls Declined
All my calls are getting DECLINED when I am trying from xlite : CLI shows : May 18 00:00:32 WARNING[4617]: channel.c:2781 ast_channel_make_compatible: No pa th to translate from SIP/cc101-b790c1d8(4) to SIP/sip19-090e87d8(256) May 18 00:00:32 WARNING[4617]: app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make SIP/cc101-b790c1d8 compatible with SIP/sip19-090e87d8 == Spawn extension (default, 71954509, 2) exited non-zero on 'SIP/cc101-b7 90c1d8' -- Executing DeadAGI(SIP/cc101-b790c1d8, agi:// 127.0.0.1:4577/call_log--H Vcauses--PRI-NODEBUG-16---) in new stack -- AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-NODEBUG-1 6--- completed, returning 0 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can YOU find a trailing parenthesis?
On 17 May 2009, at 20:07, sean darcy wrote:
exten = _6000XXXNXXX,n,Set(Time-in-secs=${STRFTIME(${EPOCH},,%s} )
STRFTIME is never closed with a ). You have two ( and only one )
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Re: [asterisk-users] Can YOU find a trailing parenthesis?
Steve Edwards schrieb:
Emacs has an excellent match-paren facility that works on (){}[] in any
nesting combination.
So does any other decent text editor. :-P
Philipp Kempgen
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Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] Can YOU find a trailing parenthesis?
sean darcy schrieb:
On 1.6.1, I must be losing my eyesight:
exten = _6000XXXNXXX,n,Set(Time-in-secs=${STRFTIME(${EPOCH},,%s} )
exten = _6000XXXNXXX,n,Set(Time_in_secs=${STRFTIME(${EPOCH},,%s)})
^
CLI:
-- Starting simple switch on 'DAHDI/1-1'
[2009-05-17 14:54:49] WARNING[13433]: pbx.c:2846 func_args: Can't find
trailing parenthesis?
-- Executing [60001234...@internal:1] Set(DAHDI/1-1,
Time-in-secs= 1242586489 ) in new stack
Philipp Kempgen
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AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de
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Re: [asterisk-users] Calls Declined
On Sun, May 17, 2009 at 2:35 PM, David @ULC ucoms2...@gmail.com wrote: All my calls are getting DECLINED when I am trying from xlite : Codecs. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls Declined
All my calls get OK when i try . From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kai-Uwe Jensen Sent: May-17-09 5:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Calls Declined On Sun, May 17, 2009 at 2:35 PM, David @ULC ucoms2...@gmail.com wrote: All my calls are getting DECLINED when I am trying from xlite : Codecs. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can YOU find a trailing parenthesis?
Philipp Kempgen wrote:
sean darcy schrieb:
On 1.6.1, I must be losing my eyesight:
exten = _6000XXXNXXX,n,Set(Time-in-secs=${STRFTIME(${EPOCH},,%s} )
exten = _6000XXXNXXX,n,Set(Time_in_secs=${STRFTIME(${EPOCH},,%s)})
^
CLI:
-- Starting simple switch on 'DAHDI/1-1'
[2009-05-17 14:54:49] WARNING[13433]: pbx.c:2846 func_args: Can't find
trailing parenthesis?
-- Executing [60001234...@internal:1] Set(DAHDI/1-1,
Time-in-secs= 1242586489 ) in new stack
Philipp Kempgen
Doh.
Any time you get to New York, I'll buy you a beer.
Thanks.
sean
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Re: [asterisk-users] Switchvox
I just inherited a client that is using a Switchvox system. I normally install a CentOS based system with freePBX and some custom endpoint management stuff for Polycom phones. This Switchvox is making me feel a bit stifled. I am having nightmares of another recent encounter with Trixbox Pro. Can I really not ssh into this box? If I could is there anything useful that I might change without breaking things and/or endangering their warranty or support? Google seems to be very quiet about customer experiences with Switchvox. I had a client who purchased a Switchvox AA350 (for about $7k). The box was plug-n-play and the endpoint manager for the Polycom phones worked flawlessly on the first try. The server hummed along for a year. One day the ISP changed out some equipment and suddenly there were double/missing DTMF tones, static and dropped calls. I spent quite a bit of time in email and on the phone with Digium trying to resolve the problem before we finally gave up. They blamed the ISP and the ISP blamed Digium. I didn't have the heart to tell the client that he wasted $7000 so I ended up swapping out the AA350 for a Dell GX745 with FreePBX. The basic installation of FreePBX exhibited the same problems on the Dell that we experienced on the AA350 which means the server itself probably was not the problem. After spending some time working with the sip trunk provider we were able to make some changes to sip.conf that resolved the issues with jitter and DTMF. The AA350 now sits idle in the closet. I will probably EBay the AA350 or use it for another client. You will never catch me speaking ill of Digium. They sell an appliance that works great under ideal conditions. I am also thrilled that Digium continues to support Asterisk as an open source product. Unfortunately we live in an imperfect world and the lack of SSH control over my client's servers prevents me from recommending their server equipment. Does this help?David Walker(602)410-3210 -0700GMT ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
