Roy Sigurd Karlsbakk wrote:
Thanks, that's what I want to do.
Any chance of me getting my hands dirty with this code? Please?
Sorry, there's no way I can distribute it in the state it's in, it's got
bunches of other stuff in it that can't be easily separated out, and
most of it does not work.
Erick Perez wrote:
And what people are using to deploy super servers with astersik?
Itanium with linux? clusters of itanium with linux? or some RISC
processor with some *nix? cause it seems asterisk is only 100%
supported on Linux/Intel
or am i totally wrong?
The highest-performing standard
[EMAIL PROTECTED] wrote:
If we want a box that can perform 60 calls. What would be apoproximate budget
for that using AMD x86-64 ?
60 calls can easily be done on a 3.4GHz Pentium 4 box, no special
hardware is required.
___
Asterisk-Users mailing list
Matt wrote:
How exactly does Asterisk provide E911 service??
Could you ask a slightly more open-ended and ambiguous question next
time? This one might actually have some real answers...
Asterisk does not provide _any_ service, the user configuring Asterisk
makes that happen. Asterisk can be
Rich Adamson wrote:
To avoid legal issues down the road, I'd suggest handling it via a
local pstn line (one way or another), and install a Red Phone with
a normal pstn line for emergency use. (The pstn line for the Red
Phone 'could' be used for incoming faxes as well, and when combined
with
Vladyslav wrote:
Is that with channels recording ? ;)
If you have a fast disk subsystem, yes. Recording calls is not CPU
intensive, only transcoding is (at that call volume, anyway).
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Parker, Blake (MIS) wrote:
I am new to Asterisk and currently work mainly with Cisco Callmanager.
With Callmanager I can setup partitions and call search spaces to
determine where a given phone can and cant dial. Does Asterisk offer
this type of functionality, and if so how?
In Asterisk those
mattf wrote:
In our experience having more processors doesn't really matter on the x86
platform because of the limitations of the motherboard bus.
The motherboard bus is not very specific.
If you are referring to the PCI bus, then most dual/quad/etc. Opteron
boxes have multiple independent PCI
Rich Adamson wrote:
Agreed 100%. Think about how one might config a spa3k to accomplish
everything noted, plus some. :)
Well, incoming call handling on SPA-3000 kind of sucks at the moment...
but I don't see how it could be configured to ring a bunch of phones
anyway. At best it can deliver the
Paul Dugas wrote:
Maybe this would be enlightening:
http://www.samag.com/documents/s=9408/sam0411b/0411b.htm
I believe there's more that one way to connect Opterons; some better than
others.
Yes, that's an excellent article; thanks for the link.
In my previous messages I was certainly assuming
ADCOM Corp wrote:
Does anyone know a good place to find a BRI S/T and U card for north
america?
Good luck; they are very rare. Even if you can find a card, most of them
do not offer US NI-1/2 firmware, so they can't be used on the network
here anyway.
The few that do work are listed on the
Joseph wrote:
In my test it seems to work fine for a little and than soon the phone
looses its time. At first the status shows clear, and then it appears to
get confused about the ntp time source and the time goes away on it.
I don't have that problem on the 10 or so phones I've updated to 7.4.
Wolfgang S. Rupprecht wrote:
Have the spa3k use an S0 dialplan:
PSTN Line:
...
Dial Plan 8:(S0 : )
...
PSTN Caller Default DP: 8
In the little bit of testing I did with an SPA-3000, I could not get it
to automatically send the call on to the Asterisk server
Ed Greenberg wrote:
/SIP/Registry/216 : 192.168.1.80:5061:3600:216:sip:[EMAIL PROTECTED]:5061
If I want to delete one of these keys, I do 'help database del' and I get:
Usage: database del family key
Deletes an entry in the Asterisk database for a given
So what's a family? How do I do a
Ronald Wiplinger wrote:
What does that mean? Where can I get more info about it?
Mar 18 07:19:51 WARNING[9563]: chan_sip.c:7549 handle_response: Got 200
OK on REGISTER that isn't a register
It means Asterisk received a 200 OK response to a REGISTER request, but
it no longer has any evidence that
Barton Fisher wrote:
I have span in a group (ZAP/g1) - How can I make this group sequentially
select ports on the span, instead always selecting port 1?
Amazingly, a quick search on the wiki turned up this page:
http://www.voip-info.org/wiki-Asterisk+ZAP+channels
Asterisk wrote:
CVS head has an option to do this. persistentmembers is the option I think.
Yes, CVS HEAD has both persistent dynamic members and persistent agent
logins.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Wei Su wrote:
How to let asterisk just forward this request to the other endpoint and
instead processing it as a UAS?
Asterisk _IS_ a UAS, it is not a proxy.
(Also, in the future, please send plain-text (not HTML) messages to
public mailing lists)
___
Ken D'Ambrosio wrote:
I bought a couple Polycom Soundpoint 300's, and have them working nicely
with SIP... but I'd like to be able to do automatic config via FTP, but it
requires some XML config files. The docs discuss them in detail, but I
can't seem to d/l them from Polycom. [No, it doesn't
Matt Darnell wrote:
There was some talk last June about some folks trying GR303 with *.
Asterisk supports GR-303 access concentrators now; I do not know if the
support is in stable, or only in CVS HEAD.
Asterisk does not know how to act _as_ an access concentrator, however.
Matt Darnell wrote:
Do you have an recomendations for the GR-303 concentrator?
There are a number of them; Digium has an Adtran unit (IIRC) that they
demonstrate in their trade show booth.
I was read that the GR-303 protocol is very similar to ISDN-PRI NFAS.
Yes it is.
I don't understand what
Rob Scott wrote:
It seems to me silly to have a T1/E1 card to connect to a channel bank
when you could just have a 24/30 way FXS card in the slot in the first
place.
Don't be surprised if you see something like that soon.
___
Asterisk-Users mailing list
Jerry wrote:
I would suggest contacting a dealer until you find one who will sell you
a maintenance contract for the phone. Last I checked, over a year ago,
they were somewhere around $10-$20. Once you have a contract you may
register online and download all the software you need. However to
Ken D'Ambrosio wrote:
I can't figure out how one upgrades the boot ROMs for the Polycoms.
I've read the wiki, I've checked the docs, I've looked here, there,
everywhere; I hoped it might be through the web interface, through
DHCP... but I can't see how to do it.
Put the new bootrom.ld and
Yves wrote:
I receive G729 G723 calls that I send to a provider who can handle
both too, is it impossible to tell Asterisk to keep using the same codec
for in out ? It seems that he only follows the codec list in order.
You are correct. Since Asterisk is a UAS/UAC and not a proxy, it
Kristian Kielhofner wrote:
After checking out CVS HEAD from yesterday (for those new
PLC/Jitterbuffer patches), I was affected by bug 3795 with my Polycom
IP600's. After seing it resolved as of this morning (thanks Mark), I
decided to try again...
There have been more chan_sip fixes
Adam Rothschild wrote:
1) Caller ID name data comes in on the PRI, but doesn't appear to get
handed off to the Asterisk server via SIP, at least not in any
format that Asterisk understands. Caller ID _number_ works fine.
(I'm guessing this has something to do with the 'remote-party-id'
Tom Samplonius wrote:
What does progressinband do exactly? Does it disable 180 responses?
I can't find any references to what effect no, yes, and never
have on the SIP exhange. In fact, why is it called inband if it
involves the SIP messages? Wouldn't inband refer to messaging in
the media
Kevin P. Fleming wrote:
If set to 'yes', and '183 Session Progress' has not already been sent,
then '180 Ringing' is sent _and_ audio ringback is also generated
(although I can't seem to figure out how that could work, since if '183
Session Progress' has not been sent, there is no early media
Rich Adamson wrote:
Likewise for a pc card supporting 24 fxs lines. The probability of three
or more lines ringing at exactly the same time are very small. With at
least a little engineering forethought, its not that difficult to
create ring cycles where ports 1 through 6 ring during some period,
Tom wrote:
Configuring VLAN 100 seconds
TFTP SIP loads a few seconds
back to Configuring VLAN the rest of the time.
That's about normal; I wish Cisco would let us turn off CDP in these
phones, it would help tremendously.
___
Asterisk-Users mailing list
Ken D'Ambrosio wrote:
tried option 66/next-server, and option 150, to no avail. And the docs
just don't -- leastwise, in the way I'm reading them -- make sense.
I've checked the admin guide, I've checked the wiki... no dice. Do I
need to *specify* my boot server from the LCD panel?
I've
Matt Gibson wrote:
This is what I'm sending from my dhcpd server.
option ntp-servers 10.x.x.x;
option tftp-server-name ftp.x.x.x;
option time-offset -18000;
Keep in mind that using TFTP for a Polycom boot server is sub-optimal,
because you have to rename files to get new
Eric Wieling wrote:
This was fixed in 1.4.1. TFTP and FTP now work the same for deciding to
download the firmware or not.
Interesting... I'll stick with FTP anyway, since I can partially secure
it, and it works across NAT :-)
___
Asterisk-Users
Ken D'Ambrosio wrote:
Kevin: I don't think Matt made himself quite clear enough -- going with
his dhcpd.conf setup works for FTP, regardless of what the option name
is; it's working great! And I don't even have tftpd installed, much
less populated. (Honestly, I'm not even sure the phone does
Nabeel Jafferali wrote:
Xetricom Networks, who only have Toronto DIDs, do provide incoming
CallerID name using IAX.
We also provide Calling Name delivery for our DIDs. It's not an issue of
the VOIP protocol in use (it can be done over SIP or IAX), it's an issue
of what sort of PSTN connectivity
Michael Devenijn wrote:
Is it possible to modify the caller id on the phone during a call (session) ?
If not does anybody know with which SIP request this could be handled ?
Do you mean what is displayed on the phone's display? If so, yes, with
some phones this is possible, by performing a
Sys Admin wrote:
After reading this and other google results for IAX vs SIP is there
any reason why i should use SIP anywhere !!
Well, let's see.. 99.99% of the available VOIP hardware only support
SIP, MGCP and H.323, but not IAX2. Is that a good reason?
IAX2 calls between servers carry the
Remco Barende wrote:
Now compiling chan_sccp.c 744 lines
chan_sccp.c:67: conflicting types for `sccp_request'
chan_sccp.h:1723: previous declaration of `sccp_request'
sccp_pbx.h:5: storage size of `sccp_tech' isn't known
make: *** [.tmp/chan_sccp.o] Error 1
That module is
Remco Barende wrote:
Are you sure? This is in the makefile:
# Asterisk version, currently only v1_0 and HEAD are supported
ASTERISK_VERSION=v1_0
Well, then the code is buggy, because the channel technology structure
stuff is only in HEAD, not 1.0.
___
Christopher Jacob wrote:
Would it be possible to put a T1 card in an asterisk box and use it to
simulate a PRI from the CO? As I build Asterisk boxes for customers I would
like to test the installation before getting on site.
Sure, Asterisk can act as the network end of a PRI as well, so you
Tim Chandler wrote:
Let me further clarify this. I am looking to buy the TE110P. The website says that The TE110P with Asterisk will route voice and data traffic, and eliminate the need for an external router. How does this work? How is the data transferred - as a pass-through like a NAT to
Parker, Blake (MIS) wrote:
My users will be stationary businesses
There is not currently a good solution for doing what you want to do,
but there are possibilities in the works. The most likely candidate at
this time seems to be Intrado's V9-1-1 service.
Wei Su wrote:
I understand Asterisk is more like a B2BUA. But when this INFO request is
sent to asterisk, asterisk is supposed to bridge the request to the other
endpoint, right? In what situation, it decides to send a reply; in what
situation, it decides to bridge the request?
It is not required
Robert Goodyear wrote:
Does anyone know if there's a service out there to -- for a fee --
inject our DID into the LEC's CLI database so a called party gets our
associated name?
No, only if the LEC servicing the number offers it to you. It is the
responsibility of the operator running the switch
Rob Gillan wrote:
Mar 23 11:42:44 WARNING[15003]: config.c:579 cfg_process: parse error:
No category context for line 14 of cdr_mysql.conf
This doesn't mean it can't find it, this means there is a syntax error
in the file and it cannot be parsed and loaded.
Tom Samplonius wrote:
I had be using a group of two PRIs for more than a year on a Nortel
PBX. After I started testing with Asterisk through a AS5300 gateway,
I quickly noticed that I could present any calling number.
Yes, we all know we can do that (and do it every day). The poster's
question
Rich Adamson wrote:
I'm a little confused on whether the GR303 support in * will accept
calls from a Siemens central office that has GR303.
I don't know for sure (sorry for responding anyway), but I believe that
Asterisk's GR-303 support is the 'network' end only, so that it can
control access
M.N.A.Smadi wrote:
say i have two users A and B registered with asterisk. A sends an
INVITE to B thru *. My question is how can i re-write some of the
parameters in the SIP or SDP message sent from A to B?
Asterisk is not a SIP proxy. A cannot send an INVITE to B _through_
Asterisk.
Stewart Nelson wrote:
I'm running [EMAIL PROTECTED] (Asterisk 1.0). Is it possible that this bug
has already been fixed in a later version (I can't find anything that
seems relevant at bugs.digium.com)?
This issue (multiple c= lines) has already been fixed in CVS HEAD (if
'pedantic' SIP parsing
Steve Blair wrote:
I'm unclear on the requirements between Comedian mail and the
spool directory where it looks for managing messages. Has anyone
implemented voicemail in this way? If so could you share some
thoughts on how the integration is accomplished?
There are discussions currently going
[EMAIL PROTECTED] wrote:
asterisk*CLI pri show span 1
Primary D-channel: 16
Status: Provisioned, Down, Active
Any ideas?
Uhhh... yeah. What do you think 'Down' means? It means the PRI span is
not up, which means it cannot be used for calls.
Call your telco, tell them you have your end
Stewart Nelson wrote:
How should I proceed? IMO, this provider offers an excellent combination
of price, reliability, quality, and support, and I believe that many in
Asterisk community would want to use them. AFAICT, their SIP/SDP does
not actually violate any RFCs.
The next step would to be
Eric Wieling aka ManxPower wrote:
the IP500's hold button works just fine. I think the previous poster
assumed you were using analog phones.
... or one of the low-end SIP phones (BT101) that don't use SIP hold,
but generate MOH locally.
___
Stewart Nelson wrote:
Well, provider is now sending a different tag, so Asterisk does not
find a match, assumes that this response is for a call it does not know
about, and discards it.
Yes, that is what is happening here.
That makes sense, but since Asterisk always generates a unique Call-ID for
Brian G wrote:
I'm looking to use Asterisk with Verizon ISDN centex service in the US.
I'd be connecting to an NT1 so I'd need an S/T interface. Users would
have SIP phones registered with Asterisk and sharing the ISDN lines.
ISDN BRI interfacing into a PC is hard to do in the US... there are
Nitesh Divecha wrote:
But I found one strange thing that, on my Snom 220 display it shows
Asterisk Asterisk when someone tries to make a call...?
Any idea where I could find this Asterisk Asterisk parameter.
Those are the default when no CLID or CNAM have been supplied when the
call was
Steven Critchfield wrote:
Does the Adtran way differ significantly enough to make this become
easy?
Yeah, the Adtran actually does ISDN PRI to ISDN BRI conversion (it's a
very simple switch), not just encapsulation. It's not cheap, though, so
it's not something you want to use unless PRI is not
Robert Goodyear wrote:
Anyone know if WAIT is not advisable to workaround the problem Noah's
asking about?
I always Wait(1) before answering an incoming PSTN or SIP call; there
are just too many cases where the media path is not quite ready to start
sending audio. There is no need to using
Mike Miller wrote:
I'm not sure where to start even -- It seems that the problem is with
the response to the digest authentication, but I'm not sure how to fix
that. The log below is from linphone, but I see the exact same thing
with kphone and xten from a indows box as well.
You are right
Stewart Nelson wrote:
I never get such good support for commercial software, even on high-end
packages that charge an arm and a leg for maintenance.
Many thanks to Mark, Kevin, and the Asterisk team.
Thanks for the kind words, we appreciate it!
___
Alex wrote:
Could anyone explain to me what is the difference between Call-ID and
UniqueID of SIP calls, please?
Which one could be used as an ID to trace, for example, the status of a call
with Manager API and PHP?
The Call-ID is internal to the SIP protocol, and not exposed inside
Asterisk (or
Adam Robins wrote:
Has anyone come up with a way to get power to a TDM400P card installed
in a Dell PowerEdge 1750?
I've not tried, but based on what I see in my 1750s, I would say 'good
luck'. There are no drive power connectors anywhere, and you can't steal
power from a fan connector because
Mike Miller wrote:
Based on what you wrote -- I'm using type=friend, not type=peer. This
should be ok, though, correct? (As friend == peer + user, right?)
Yes, type=friend is fine.
sip.conf:
[general]
context=default; Default context for incoming calls
realm=192.168.1.100; Realm for digest
Mike Miller wrote:
1.0.6 from an ubuntu package. I'd also tried a version compiled from
source, but with the same results.
I tried taking out username, but it didn't help.
OK, then we need a _full_ log, with:
- sip debug
- set verbose 255
- set debug 255
There should be (at least) a message on the
Mike Miller wrote:
Mar 30 04:39:30 VERBOSE[32543]: == Parsing '/etc/asterisk/sip.conf':
Mar 30 04:39:30 VERBOSE[32543]: == Parsing '/etc/asterisk/sip.conf': Found
Mar 30 04:39:30 DEBUG[32543]: Unable to find key '203' in family 'SIP/Registry'
Mar 30 04:39:30 VERBOSE[32543]: == SIP Listening
MDS wrote:
I have Asterisk CVS-HEAD-03/19/05. been running Asterisk for over 6
months, no problems with my grandstreams. I'm fairly familiar with the
ins and outs of asterisk...
If you are going to use CVS HEAD, you _must_ stay up to date. There have
been a large number of SIP-related fixes in
Mike Miller wrote:
They're both running on 192.168.1.100
Sorry -- I probably should've clarified that.
Yeah. that would have helped! For some reason, they were not only
running on the same machine, but sharing the same port number, which
shouldn't really be possible...
But in any case, if you
Jan Johansson wrote:
Uhm, doesn't the 1750 have the same feature as the 2600-series? As in
a female molex hidden away for the possible addition of a tape-streamer?
No, it does not. There is no place in a 1750 where a tape drive could be
installed. There are five device bays, one for the floppy,
Carlos M wrote:
Unable to allocate channel structure
Unable to create/find channel
Your machine has run out of memory.
When this happens im unable to make and receive calls.
The only way to fix this is restarting asterisk. The asterisk version
im running on all servers is Asterisk
Pepe Aracil wrote:
How can i match in sip.conf by the (TO: ) header in sip negotiation?
When you register with your provider, add /extension suffixes to the two
register = lines, which will direct the incoming calls to different
extensions in your incoming context.
Tyler wrote:
I've searched the archives but have been unable to find the answer to
this. I have a 2-port Patton Datafire 2977 T1 card that I had
originally picked up for a HylaFax project.
No, this card will not work with Asterisk.
___
Asterisk-Users
The recent discussions about mailing lists vs. forums have resulted in
Digium management deciding to offer a forum site on a provisional basis,
to determine if it will benefit the community.
You will find a brand-new set of phpBB forums at forums.digium.com.
Membership and posting are open to
Andrew Kohlsmith wrote:
Now they MAY have incorporated the TJ320 chip logic in the Xilinx Spartan II
FPGA but I would be **VERY** surprised if they did that. Just my opinion,
but I think that level 2 digium tech is full of shit.
Andrew is correct; there are no TigerJet parts on the quad-span
Remco Barende wrote:
like it says, the equivalent of 20 E1's or 28 T1's
and I guess you know how many channels a E1 or T1 PRI is
That is correct; the DS3000P will support full access to every channel
on the DS-3 (or E-3), however it is provisioned. In a T1-RBS signaling
mode, that means 672
Andres wrote:
Can you confirm if there will be some sort of DSP daughther card add on
of some sort for the DS3000 so that we can run G729 transcoding? I
don't see how the DS3 interface would be usefull unless we could offload
transcoding stuff to onboard DSPs. Or is Digium only going to
Andrew Kohlsmith wrote:
secondary card for DSP functions is very inefficient of the PCI bus. I'd be
curious to know if the Digium cards can even do PCI-PCI DMA.
The Digium TDM cards can DMA into any RAM accessible over the PCI bus,
regardless of whether it is located on the motherboard or on a
Bicom Systems wrote:
What is target release date for DS3000P?
That has not been announced; sometime after today would be a safe
assumption :-)
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
[EMAIL PROTECTED] wrote:
In other words, a PCI-based co-processor would double the PCI bus
bandwidth necessary. And with a latency-sensitive product like voice, bus
contention is not something you want to add to! :)
It only 'doubles the bandwidth required' when compared to a single-board
Matthew Boehm wrote:
So, no hardware encoding on this beast?
The announcement on the website makes no mention of transcoding, echo
cancellation or toast-and-jam making, so at this time, no, there is no
hardware transcoding apparently included. (Besides, would you really
want a board that could
Andrew Kohlsmith wrote:
Yes, but then what are you doing with it? You're shuttling the new data
to/from a network card in a lot of cases. Combined with other traffic over
the PCI bus for normal system operation I could see you coming close to the
limitations of regular ole PCI.
Absolutely.
Steve Underwood wrote:
Since encoding typically requires 5 times as much compute as decoding,
for CELP based codecs, an encode onyl board would not be as dumb as it
seems at first sight :-)
Hah! I knew someone would say that!
___
Asterisk-Users mailing
Matthew Boehm wrote:
I'd like to see an S200I, a S600I, a S120I and a S240I. But the S240I needs
to be cheaper than an Adtran channel bank off ebay.
Really? You think that an Ethernet-connected 24-port FXS channel bank
with built-in codecs and all that should be cheaper than raw channels
via a
Andrew Latham wrote:
What would be a real example of use. EG: A Tyan Transport TX46 and
this card could handle what ever you could put on it. What is the
realistic low end of system that could support this card.
Unfortunately that is a question without an answer, because support
this card means
Matthew Boehm wrote:
If it costs less for me to buy an adtran and another asterisk box to handle
the encoding/decoding of that adtran, then there is no point to buying an
S240I.
That may be true in some very specific cases, but I can think of lots of
reasons why an Ethernet-connected channel bank
Matthew Boehm wrote:
(for us at least, subtract the price of the TE110P cause all our T1's
come to us on DS3s, and we already have DS3 routers in place and paid for.)
I'm confused: why would you terminate a T1 from a provider using a
channel bank, rather than directly into an Asterisk server?
[EMAIL PROTECTED] wrote:
We are moving to a new building late in January and I have been tasked with
the job of setting up the phone system. I have ordered a full T-1 for voice and
am considering using Asterisk as the PBX. I have a Dell PowerEdge 2650 that I
can use for Asterisk. I have also
Matthew Boehm wrote:
Well, according to Digium, they don't support T1 data mode on their cards.
The reason for this stems from the fact that you have to do a kernel
recompile to add in HDLC support; it is not present in standard RH/FC
install.
The OP didn't ask about data mode at all. He said he
Paul Rodan wrote:
The thing I dislike the most about the 79xx phones is that in DHCP mode,
they expect the DHCP server to tell them their TFTP server address. They
won't let you set it manually. So if I don't have DHCP server that gives
TFTP server info, which is most of the DHCP servers at out
Scott Gruby wrote:
sip show peers
and it is blank with the system being hung.
snip
Any ideas on what is causing this? Is there any additional information I
can provide for assistance?
You can start with actually telling us what version of Asterisk you are
using, and how you installed it (from a
Adi Linden wrote:
http://voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf%20sorting
This page on voip-info.org describes how it is possible to affect the sort
order of patterns in extensions.conf. What is doesn't explain is how
asterisk really does sort patterns. How does this
Roger Schreiter wrote:
did the digium registration server for g.729 change
or is it currently just down?
It was down over the weekend, should have been brought back up by now.
Since you have purchased licenses, you can call Digium support and ask
them to take care of it.
Matthew Boehm wrote:
Does this work for crashes and complete shutdowns/restarts of asterisk?
Yes, it has worked for us through both without a problem.
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Joao Pereira wrote:
Hello to all
I would like to know if someone tried the VPT1000 WiFi phone from:
http://pcphoneline.com/
Where do you see that it is a WiFi phone? That page says it uses USB to
connect to your computer. In fact, there is no mention of WiFi or any
802.11 on that page at all.
John Bittner wrote:
Anyone know how to get app_queue to send logs to MySQL or
any other sql server.
I found info for cdr's and even configs but nothing on
queue_log.
If sql is not supported in the current app_queue I will be
willing to pay someone to add it.
It does not currently do anything
Lars Fredriksson wrote:
I have benn playing a little with quesues tonight and I found out if there
are at least one member-extension free the announcement with p'the place
in the queue wont be played to the person who called in.
This is a change that went into CVS (and changed the default
Noah Miller wrote:
I guess the phone just doesn't register as busy when there is only one
call on a line. It has to have two calls on a line appearance to
register as busy. Has anyone figured out how to disable this hold
feature and just have the second call go to the second line, the third
[EMAIL PROTECTED] wrote:
I've got a SIP channel that appears to be hung up. It's an extension
that records a .gsm file and fortunately the recording has stopped. I
tried zap destroy channel but I guess that doesn't apply to SIP channels.
Uhh, no, why would zap destroy channel affect a SIP
Joe Dennick wrote:
Yeah, set the queue timeout to be about 1 second less than the voicemail
timeout (ya know, where you say Dial(SIP/, 15)). That way the queue
times out the agent before the dialplan goes to voicemail.
The more reasonable solution is to just put the agent's direct path
Matthew Boehm wrote:
If I add a line like this: member = SIP/3044, can I still get
login/logoff functionality? We need agent login/logff functionality AND for
calls to not goto voicemail.
No, I was suggesting using SIP/3044 in agents.conf, not in queues.conf.
If you put it into queues.conf,
1 - 100 of 2272 matches
Mail list logo
