I'd also be more sold on it if it had half the features of the GXP2000
(which is only a little over half the price).
Sure, but if only half of the features in the GXP2000 actually work,
what is the point of them? I'd take a stable phone with less features
over one that has lots of features
: No
Auto Clear: 120
Again, if I do a sip show peer after pruning, I see the new values,
but it appears that * is still holding it somewhere that isn't updating.
Marc Smith wrote:
On Tue, Jul 15, 2008 at 12:05 PM, Peder @ NetworkOblivion
[EMAIL PROTECTED] wrote:
I am using realtime
I am using realtime on two boxes, one running 1.4.10.1 and one running
1.4.11. Everything works fine except for when I make a database change,
such as a phones password. I change the DB, I prune the peer, I see it
is gone and then I see it show up again in sip show peer , but
everything
Does anybody have the settings that you use on a Cisco 7970/79x1 to get
presence? I see the * side settings, but I can't find the Cisco side
settings anywhere.
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SIP.
Michiel van Baak wrote:
On 14:59, Wed 25 Jun 08, Peder @ NetworkOblivion wrote:
Does anybody have the settings that you use on a Cisco 7970/79x1 to get
presence? I see the * side settings, but I can't find the Cisco side
settings anywhere.
Sip or Skinny
They still have issues. If you use TCP and reboot the server, the phone
will never reconnect as it tries to use a closed TCP session. I opened
a ticket with them and after a week their answer is . use udp.
Rob Hillis wrote:
Doug wrote:
There is a bug in these units that won't let
you
They still make them. We use the CS70N with HL10 (headset lifter). They
are around $300 with the lifter, so they aren't cheap, but they work
well. The lifter fits on a Cisco 79xx phone pretty easily, but anything
else requires a little extra tape and some experimentation.
Peder
Steve
FYI, I have probably 10 Fortinet units with multiple SIP phones behind
each and all of the phones work flawlessly. As long as the Fortinet is
ver 3.0 or newer, it does NAT so that you don't need to have nat=yes on
*. No pinholes or static nat or anything, it just works.
As a side note, I
How does the g729 encoder/decoder count in regards to the total number
of licenses and how does it count an encoder/decoder? I looked on the
wiki and don't really see anything that explains it. In other words,
how do the calls below count (assume no reinvite)?
g729 phone calls into voicemail
the g729 side (
no license for g711 side of call ) . In short anytime u need to
convert g729 into some other codec ( transcoding ) you need 1 license
.
On Wed, Apr 2, 2008 at 1:59 AM, Peder @ NetworkOblivion
[EMAIL PROTECTED] wrote:
How does the g729 encoder/decoder count in regards
I am trying to get BLF working on Grandstream phones with 1.2.27. I
actually have it working, but I found a very strange issue and I am
wondering if anybody knows what the problem is.
Here is the scenario. If I have 3 phones, A, B and C. A monitors
presence of B and C. Right now, if I call
Enable NAT on the phone itself and leave it enabled in *.
Jerry Geis wrote:
I have a cisco 7960 phone. Worked fine in the office.
I took it home. At home I have a linksys router that the phone is
plugged into.
The linksys router has DHCP enabled. I am getting the following error on
the
Do you mean Call Manager? Unity is just their voicemail system. Yes,
you can use SIP to talk between * and CM. You can also use h.323, but
it is a big hassle.
Tony Mountifield wrote:
Has anyone here any experience in getting an Asterisk box to talk to
a Cisco Unity system? I have a
autoload=yes says to load everything, so you either need to change it
to no and then add load statements for every module you need, or leave
it as yes and then add noload for everything you don't need.
Vincent wrote:
On Wed, 20 Feb 2008 21:44:30 -0500, C F [EMAIL PROTECTED] wrote:
vi
What happens when you try it? And what do you do on the phone? We have
lots of GXP-2000 and 2020 and transfer is one feature that does work.
Gustavo Gonzalez wrote:
Hello! is there a way to get keep working the TRANSFER key of GXP-2020 with
asterisk?. Attended and blind transfer does not
We use PRI, not BRI, with Cisco gateways and it works great. Rock solid.
Razza wrote:
Is anyone using a cisco router as an ISDN gateway with Asterisk?
As you might have seen from a couple of my threads, I have been looking
at Fritz! and Cologne cards, both of which require development
I have 20-30 GXP2000's connected to * over a T1 line. Neither end is
NAT'd and there is plenty of bandwidth available over the line. The
GXP's are 1.1.5.15, which is the latest. I have a problem where the
phones keep dropping off of * and I get a failed to register message
in the log of *.
appears to kill
registration until the Grandstream is rebooted. Has anybody else seen
this? Or maybe know how to get around it?
Peder @ NetworkOblivion wrote:
I did post most of that. Point to point T1, no firewalls and no nat,
cisco routers, bandwidth is monitored at 30 second intervals
file of the phone.
On Feb 11, 2008 4:52 PM, Peder @ NetworkOblivion
[EMAIL PROTECTED] wrote:
I have 20-30 GXP2000's connected to * over a T1 line. Neither end is
NAT'd and there is plenty of bandwidth available over the line. The
GXP's are 1.1.5.15, which is the latest. I have a problem
I've got a setup where we have 100 DID's. Our default dialplan has one
line that calls a macro:
exten = _22XX,1,Macro(STDEXT,${EXTEN})
The macro is pretty basic:
[macro-STDEXT]
exten = s,1,NoOp
exten = s,2,Dial(SIP/${ARG1},15,Tt)
exten = s,3,Goto(s-${DIALSTATUS},1)
exten =
What about the situation where there is no voicemail box for an
extension. Is there a way to tell the difference between the phone
isn't registered and there is no phone at that extension?
Doug Lytle wrote:
exten = s-CHANUNAVAIL,1,Voicemail(${ARG1}|b)
exten = s-CHANUNAVAIL,n,Hangup
Or you can prune the specific user entry and it will look it up again.
Anthony Francis wrote:
Adam Moffett wrote:
I asked this question last week and never got an answer. I also
didn't find the answer in the wiki.
I think it would be nice if asterisk would check the database again if
I am running 1.4.10.1. I have a macro that is called from default for a
certain extension (both below). I added NoCDR to s to try and stop
extra CDR records, but I am still getting them. Any idea how to stop them?
extensions.conf:
[macro-STDEXT]
exten =s,1,NoCDR()
exten
We've got an SPA-2100 connected to * and then into a paging system on
one of the FXS ports. We are having an issue where the paging system
doesn't hang up the line, so it stays offhook forever and obviously
makes in unusable. The paging company says that the SPA needs to hangup
the line once
Is there any way to see the called number when a call gets redirected to
the 'a' extension from voicemail? Say x123 calls x456 and it rolls to
voicemail. x123 hits * and gets dumped into the 'a' extension in the
original context. I need some logic in 'a' to do a database lookup
based on the
What is the issue with the Grandstream? We are getting tired of Cisco
issues, so we have started looking at Grandstream and they seem to be
pretty good. The Polycom work well, but they seem to die after about a
year or so. We bought 20 of them about 2 years ago and 7 of them have
died or
to send it. If I test it and hit * from my voicemail, I get 'a'
as the EXTEN, which doesn't help me. I need 'a' to be able to see the
called number so that I can do a db lookup and send the call to the
appropriate extension.
Peder
James FitzGibbon wrote:
On 10/26/07, *Peder @ NetworkOblivion
I know that you can set it up to where a user hits 0 from their mailbox
and goes to an operator, but can you set up other options as well?
Could I have 0 for an operator and 1 to go to another extension? I know
you can do this by building an AA, but I don't want to have to do that
for every
This is semi-related, but I have a Tmobile MDA and I couldn't play the
files either. The issue was not a codec issue, it was an email encoding
issue. If I sent the message to an email account and it was then
downloaded to my desktop via outlook and then forwarded on to my phone,
I can listen
Yes, you need to buy a license if you use it with ANY pbx, whether it is
Callmangler or Asterisk or whatever. If you buy one used, then you need
to pay to re-license it as well.
The 7940/7960 only work with Cisco PoE, not standard 802.3af, so you
will need a switch that provides Cisco PoE for
:
Peder, can you point me to the Cisco PoE swith (pre-802.3af) that can
handle the 7940G ?
The 7941G does conform to the standard but it only support SCCP (shame
on cisco).
On 9/27/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
Yes, you need to buy a license if you use it with ANY pbx
Could be a mysql permission issue. Try this from the local box:
mysql -u root -p
enter asterisk as the password
use asterisk;
select * from sip_buddies;
select * from iax_buddies;
If you get that far and can see the entries in iax_buddies and
sip_buddies, you know it isn't a permissions issue.
There has to be some reasonable priced sip provider that would perform
just as well as ATT. Does it exist?
The problem is that there is no QoS control between you and the
provider, so a lot of the quality issues you have are probably not
related to the specific provider, but just the
Is there a way to decrease the volume on the native files version of MOH
in 1.4? I've had several people complain that it is too loud.
Peder
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We have users with Cisco 7900 phones running sip. When user A calls
user B, we want user B's name to appear on user A's phone. It shows the
extension they call, but not the internal name of the called user. Is
this possible? We have some people that used to be on an MGCP based
system and
You can buy a used Cisco 2600 with dual-port PRI/T1 card for VoIP for
~$1500. No worries about echo-cancelation, or IRQ issues or anything
like that. It just works. And the config for inbound/outbound calls is
maybe 20 lines total.
Alex Balashov wrote:
For a price tag that does not scale
The question I always have when someone mentions distributing the load
across multiple machines is how do you handle contexts for phones on
different machines? I want all of my phones to dial into
[companyA-phones]. I have to define it in two different places (or more
depending on the number
I am in the process of setting up several * servers using realtime and
connecting to mysql. I am trying to figure out if I should just use one
database and one set of tables for all of the user data. Or if I should
have separate databases for each * box. The boxes are independent of
each
Is there any way to get the channel of the first agent called in a
queue? Say I have a queue with 5 agents setup in roundrobin. I want
the voicemail to go to the first person that was called. Say a call
comes in and rings 1,2,3, then I want it to go to vm for 1. Say the
next call rings
Does anybody have realtime queue members working? Not the queues
themselves, just the members. I have realtime working for voicemail and
sippeers, but I can't get queue members to work. Here is what I have:
res_mysql.conf:
[general]
dbhost = 127.0.0.1
dbname = ASTERISK
dbuser = myuser
dbpass
Anthony Francis wrote:
There is no queue_members file, asterisk doesnt know hat you are
talking
about, you would have to #include queue_members from inside that queue
definition.
Huh? How is including a file going to make realtime access the
queue_members database via mysql?
realtime members without having your queue in realtime
queues. Now you can have a static queue with realtime members. Very
useful.
Peder
Julian Lyndon-Smith wrote:
I think that revision 80086 in the 1.4 subversion branch would fix this.
Julian.
Peder @ NetworkOblivion wrote:
Does anybody
A. BC are pre-packaged and are useful for some things, but if you
deviate too much, they aren't very helpful. As a matter of fact, if you
modify a text file in AsteriskNow in one of the sections that it uses,
it causes the gui to freak out and it won't parse right. Plain old
asterisk is a
First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one setup
but during that time the caller has had two rings before the local extension
has even begun to ring. Is there something I am doing wrong that
Wait(2) is what I do.
Matthew Harrell wrote:
First, it seems I have to have a 2 - 3 second wait before the AGI call in
order to get valid CID data. Usually 2 seconds suffices for this one setup
but during that time the caller has had two rings before the local extension
has even begun to
. The recording extension answers, plays a beep, records the call to
the file name that it pulls from the *DB
5. It plays the recording back and then hangs up
It works perfectly. Not quite what I planned, but it does work.
Doug Lytle wrote:
Peder @ NetworkOblivion wrote:
That's great, now say you have 5
I am trying to use Asterisk Manager via php to record auto attendant
greetings and I just can't figure out how to do it. I've got the php
page working and I can click to call between two phones. However if I
click to call just a single phone and then try to enable monitor, when
I pick up the
that one chunk rather than the whole thing. I would need
lots of extensions pre-setup for each chunk. Not very efficient.
Gordon Henderson wrote:
On Fri, 10 Aug 2007, Peder @ NetworkOblivion wrote:
I am trying to use Asterisk Manager via php to record auto attendant
greetings and I just can't
I've had MOH die probably 4-5 times in the last 2+ years and the only
way to get it back is to stop * and restart it. Reloading MOH or just
doing a regular reload doesn't work. I have to actually do a stop now
and then asterisk to get it to work again. * restarts and MOH works
fine. No
Does anybody know if SetCallerPres works on calls via SIP through a
Cisco gateway? We are trying to mark outbound calls as anonymous and we
set it to prohib, but calls still show outbound callerid. We are SIP
from * to the Cisco gateway and then PRI outbound. If we strip the
callerid num,
I am trying to use a macro to screen calls by calling several different
phones at the same time in different groups. Find me will not work and
queues will not work either. Trust me, I've tried them both and they
don't work like they should. Here is what I have:
A call comes into 6084 and
I am using the Find-me/Follow-me example below with screening:
http://www.voip-info.org/wiki/view/Asterisk+tips+findme
Here is my actual config:
[macro-screen]
exten = s,1,Wait(1)
exten = s,n,Background(press-1-to-be-connected-to-the-caller)
exten = s,n,Set(TIMEOUT(response=5))
exten =
I've got a very strange problem and I can't figure it out. I have a
Cisco PRI gateway connected to * via SIP. When I debug on the Cisco, I
see callerID name, but it is not getting to * via SIP. I am running *
1.4.2 and the latest Cisco IOS for my router. Here is what is happening:
A call
Is there a way to use privacy manager without requiring the user to
enter their name? Essentially I am just looking for a way to force the
called user to enter 1 to accept the call. I don't need a name
recording. I want a call to come in, a message to be played, music on
hold, call out to
I just opened 0009509 and used Explicit Call Acceptance as the name.
Ben Klang wrote:
On Monday 09 April 2007 02:48:32 pm Steve Murphy wrote:
On Mon, 2007-04-09 at 09:47 -0500, Peder @ NetworkOblivion wrote:
Is there a way to use privacy manager without requiring the user to
enter their name
Does anybody have callerid name coming in on a Cisco PRI via a Cisco
gateway via SIP to *? I've seen a few people ask and a few people that
say it should work, but I've never seen an actual working config.
I do a debug on our Cisco gateway and I can see the callerid name,
however none of the
I am trying to setup a queue in a very specific way and I can't quite
figure it out. I'm sure someone else has done this.
I want calls to come into a queue and do a ringall on a number of phones
(let's say 3). So ring them for 20 seconds or so. If there is no
answer, I want it to ring a
Does anybody know the sql type for the call-limit field under sip
peers? Everything on voip-info is missing that entry.
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Kevin P. Fleming wrote:
Peder @ NetworkOblivion wrote:
I want calls to come into a queue and do a ringall on a number of phones
(let's say 3). So ring them for 20 seconds or so. If there is no
answer, I want it to ring a second set of phones for 20 seconds. If no
answer, then go back
I did a trial as a wholesale provider and it seemed to work pretty good,
but I could never get them to activate our account. I emailed the sales
guy probably five times over a month to go ahead and fire it up and he
never responded. Also, their tech support is horrible So basically
they
Group pickup / call pickup is the feature you want.You put everybody
in a group and if you want to grab a ringing phone, you just hit the
group pickup code.
http://www.voip-info.org/wiki/view/Asterisk+callgroups+and+pickupgroups
Rob Schall wrote:
Question:
Is it possible to pickup
SNTP tcpIpApp.sntp.resyncPeriod=86400
tcpIpApp.sntp.address= tcpIpApp.sntp.gmtOffset=
tcpIpApp.sntp.daylightSavings.enable=1
tcpIpApp.sntp.daylightSavings.fixedDayEnable=0
tcpIpApp.sntp.daylightSavings.start.month=3
tcpIpApp.sntp.daylightSavings.start.date=8
Is it possible to use the include command to include other contexts if
you are using realtime for extensions? I've searched voip-info and some
people were asking about it, but I didn't find a real answer anywhere.
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is
running in realtime. Every night the include files are generated and put
in /etc/asterisk. If MySQL were to ever fail, we could just uncomment
those 3 includes and reload asterisk. It wouldn't have the same dynamic
nature to it, but it would bring functionality back online.
Rob
Peder
Check out CallRex, they list Talkswitch as a supported product (also
Asterisk):
http://www.telrex.com/callrex.htm
I've seen it being used with Cisco phones on a hosted Covad environment
and it is pretty neat.
(I have no affiliation with them whatsoever).
Cory Andrews wrote:
Apologies
I have a little call recording script that I am running and it works
fine, but I have one problem. I get CDR when a user calls into the
extension, but I do not get CDR for the call that it makes outbound on #
17. Any idea why? Here is the extensions info:
[default]
exten = 2211,1,Answer
Does anybody happen to know the input power specs for the Polycom IP 500
and IP 600? We've mixed up our power supplies and we've got a whole box
of them and can't figure out which go to the Polycoms. I would rather
not kill the phones by trying random ones
It doesn't have anything to do with hardphone versus softphone. The
issue is that it can only keep track of one registration per account.
When the hardphone gets unplugged, it will not know about the softphone
until it registers with asterisk. It's initial registration was lost
when the
Virtually any Cisco device from a 2610 up will work. 2610, 2620, 2811,
2821, 3640, 3700, 3800. I have 2610 and 3640 in production for 2+ years
with no issues.
Patrick Fortin wrote:
Hi
Can someone recommend a PRI to SIP Box that work well with asterisk
We are presently testing with a
We've had a few people complain that the beep before leaving a
voicemail is not loud enough and too short. Does anybody have a
recorded beep that they can share, that is a little louder and a little
longer? We've had this box in production for 2+ years, so I hate to
mess with the gain on the
Is the storage of actual voicemail messages in a database still limited
to ODBC? If so, why?
And is the use of mySQL and ODBC at the same time still a bad idea? If
so, why?
I want to store all of my voicemail stuff in a database so that I can
give users web access to it, but I don't want
Does anybody know how to enable CallerID name passing from a Cisco
gateway (with PRI that has name and number) to an * box via SIP?
Supposedly CID name is enabled, but we can't get it passed to * and I've
googled and I can't find what I need.
___
There is a Timeout SIP in the config. What is it set to? If it is
less than the the qualify interval, which I believe is 60 seconds, then
the PIX will close the inbound hole for qualify traffic. We've got lots
of phones at several remote sites all running behind PIX's and all being
NAT'd to
canreinvite=no will force all rtp packets through *.
Ranjeet Kumar wrote:
Hi,
Can I do RTP Proxy in asterisk? As our requirement says that voice
packet should also go though sip server, so that billing should be perfect.
Thanks,
Ranjeet
Thanks,
Ranjeet
The
How does it work?
Joshua Colp wrote:
Rushowr wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Actually, isn't there SLA work being done in the trunk right now?
It doesn't work how you think it does, you can still only have 1 SIP
device registered to a SIP peer at a time.
Our MOH died, so I finally had to kill my * process and restart it.
Interestingly, stop now didn't work. I had to kill the process. It
used to work, but it had been up so long that it must have gotten
corrupted somehow. Here is the show uptime before I killed it:
Asterisk-A*CLI show uptime
:
On 8/26/06, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote:
Who says * isn't stable enough for prime time? At least it is on
1.0.3.
What kind of abuse does that box take?
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asterisk-users
If you are running a new version of PIX sw (6.3.4 or 6.3.5), then leave
fixup on and set nat=no. The PIX is the only firewall that I have
seen that truly does nat correctly. It nat's both the source and dest
inside the packet. You can even do reinvite with multiple phones behind
a PIX and
What is the status of it anyway? I followed the bug for it and it
appears that the bug was closed and maybe it was incorporated into
Trunk. Is this true? And should it be (fully) functional now?
PA
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Can you do buddies with Cisco phones running SIP? I can't find anything
that says yes or no. I can set it up on the * server, but I don't know
what to do on the 7960's themselves.
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I read both of those links and I don't see any mention of SIP buddies on
either one.
Adrià Vidal wrote:
2006/8/10, Peder @ NetworkOblivion [EMAIL PROTECTED]:
Can you do buddies with Cisco phones running SIP? I can't find anything
that says yes or no. I can set it up on the * server
Then in the tftp config file for the phone add speeddials
for the 6000 extension (cant recal how it is done, there are
examples in the default file and on the wiki)
I found out you really have to define the speeddials in the
tftp files. speeddials configured with the phone menu or
webinterface
What is the regseconds field supposed to be used for when using ARA?
I'm running 1.2.10 and when a phone registers, it is a HUGE number, like
1155074046. I assumed it would be the same as expire under a sip
show peer, but it's not. That field shows 30 seconds (well it varies,
but it's under
Are they both being NAT'd to the same external IP?
Dovid Bender wrote:
Hi List,
I am not sure what this issue is. I am having a problem where I have 2
phones that are behind NAT on the same internet connection. The
asterisk server has a public IP. Using asterisk real time1.2.10 on
CentOS
When I looked several months ago, the only Sipura that supported T.38
was the SPA-2100. I haven't searched in a while, but I think it is
still true. We go directly from a Cisco gateway to the SPA-2100 and it
works great. It is the only ATA that we've seen that works right.
Joshua Colp
Does anybody know if shared appearance / BLA is on the * roadmap? And
if so, when it might appear? I've seen people asking for it for quite a
while, but I've never seen anybody say that it is in process or on
the roadmap.
___
--Bandwidth and
Did anyone ever get an answer to this problem? I just brought a new
gateway on line and it is running 12.4 and I have to use g729br8 and
there are lots of quality issues with noise and errors on * about extra
frames. If I drop the br8 codec, the phones can call out and the
quality is great,
I seem to remember reading somewhere about a setting on Cisco gateway's
(with PRI) where you can have it send inbound (from PSTN) callerID name
via SIP to *. Does anybody know what that setting is? I searched the
archives and can't quite find the right set of keywords to locate that
info.
Two follow on questions:
1. Wouldn't that be for calls from * to the gateway out to the PSTN? I
want incoming calls from the PSTN to the gateway to deliver CNAM via SIP
to my * box.
2. How would I know if I want/need codeset 6?
[EMAIL PROTECTED] wrote:
In the interface Serial section
I've got a question about voicemail and callerid and I can't quite
figure it out. I've got extensions 100, 101 and 102. For outbound
callerID (calls from the phones to the PSTN), I want the callerid to say
100 on all phones, so under sip.conf, I added:
callerid=Bill 100
The problem is that
I've got a strange problem. I have two Cisco gateways each with one PRI
and they each go to a different provider. One has been working for 2+
years with no problems. We recently added the second one and I have a
problem where calls come in, but I can't answer them. The call comes
into *,
What is the current recommended version of firmware for SIP on
7960/7940's. I was reading through some of the stuff on voip-info and
it looks like the 8.x's have pretty serious bugs in regards ti *. Thanks.
PA
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Right now we have several companies all within on context in voicemail.
The users want to have dial by name, so we are going to split into
multiple contexts so that they don't accidentally dial each other (and
complain). I've been reading over the voicemail context info and I'm
somewhat
Is there a buddies feature on the Cisco phones, like there is on the
Polycom? If not, how are people getting around the issue where a
receptionist wants to see who is on the phone? Or are they just living
with the limitation? Thanks.
Peder
___
Our asterisk server has been up and running for over a year and it works
great. I have emails going to my account as an attachment and I can
listen to them on the desktop and it works fine. I just got a T-Mobile
MDA that runs Windows Pocket (or whatever they call it) and it can check
email.
Not to be a smarta**, but you have to ask them to do it. We do the same
thing and it works for us. Depending on the CLEC, they may do it or
they may say no. If they say no, there isn't anything you can do about it.
Hugh L. Johnson wrote:
Central business location has a PRI with a CLEC.
Is anybody using 1.2 (or 1.2.1) in a production network using Realtime
(voicemail, sip or extensions) with 100+ SIP phones? If so, what are
your experiences? We've been running 1.0.3 for about a year and it's
been rock-solid. We'd like to upgrade to Realtime and 1.2, but I'm
afraid of
It just re-directs the RTP stream. The SIP stream still goes through *.
Mike Hammett wrote:
According to this page: http://www.asterisk.org/doxygen/Config_sip.html
canreinvite=yes redirects just the RTP. I was under the impression that
the entire SIP connection got redirected, therefore
Does anybody have a DaemonTools Supervise script for Asterisk? I
searched google and the archives (and voip-info.org) and I see people
mention using Supervise, but I don't see any actual sample scripts. Thanks.
Peder
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Is anybody running * on a SunFire X4100? If so, any issues?
Peder
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