Re: [asterisk-users] Cisco + Asterisk list anyone?

[Gary Richardson]

I'm interested. I turned my last call manager off last month, but I still
use the handsets and a Cisco router for PSTN access.

On 3/16/07, Curt Shaffer <[EMAIL PROTECTED]> wrote:


I have been working with a couple companies who are interested in
integrating Cisco VoIP (mostly call manager express) but utilizing
Asterisk
for AA, VM, failover trunks etc. I have found some materials and guidance
out there but I think a list and/or wiki for general asterisk integration
with other vendors would be great and feel that it is enough off topic
that
it deserves its own space. Just throwing it out there for feedback. I'm
willing to host both. Let me know what ya'll think!

Curt

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Re: [asterisk-users] Re: Best inexpensive home office router for VoIP (QoS with maybe PoE)

[Gary Richardson]

I'd wager to say yes, it does support layer 3 routing :) That's a bit of a
redundant term (though you can route above layer 3). Depending on how many
interfaces you have on your router, you may be sending multiple vlans over a
trunk port (I'm pretty sure the 1600 series support trunk ports -- you may
want to google 'router on a stick').

Most of the layer 3 gigabit switches will still be very expensive, though
Catalyst 3500's may be getting 'cheaper' -- most of the 3500 and 3700 series
switch have multi-gigabit backplanes (usually 16 - 32 gigabits) and can
usually route packets are wire speed, or very close to it. If you are
looking for a gigabit port or two for uplink, I believe they even made a
2900G, though that won't have PoE. And now that I think about it, probably
doesn't support layer 3 routing :(

That's the Cisco world, I'm sure you can find other vendors that have
hardware for much cheaper, though this is an advantage to using the same
networking equipment most other people are using. Also, most of this is
overkill for a handful of network devices.


On 1/10/07, Ed Rubright - mail lists <[EMAIL PROTECTED]> wrote:



Do these 1600 series Cisco routers you mention that you find on eBay for
$50-$150 support Layer3 routing?  I have a managed switch setup on my
home network with several VLANs defined. (work subnet, home subnet, VOIP
subnet)   I currently have to use a Linux box to route between the
VLANs.  I'd like to move to Gigabit routing, but I'd need to replace the
Linux box(more processor power and new NICs) and that gets expensive.

I'd much rather have a router or smart switch for that matter that does
Gigabit Layer3 routing all in one unit.

Do you have any recommendationsthat wouldn't break the bank?

Thanks,
Ed
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Re: [asterisk-users] Jabber Client

[Gary Richardson]

Look for Asterisk-IM (http://www.jivesoftware.org/asterisk-im/)

It'll work with your jabber server. I don't know if it 'works' with Exodus,
but it will show if you're on the phone.

On 12/9/06, Shinji Kawamoto <[EMAIL PROTECTED]> wrote:


 Hello,



I would like to connect jabber client (Exodus) to Asterisk 1.4.

Asterisk 1.4 already supported jabber (client/component), didn't it?



Can I connect Exodus to Asterisk directly?

And if yes, please tell me how to connect?



Kawamoto



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Re: [asterisk-users] Manage Users in LDAP

[Gary Richardson]

phpldapadmin is pretty nice. I was using 2-3 different ldap clients to get
the job done until I got over my php bias and installed it. It lets me do
everything I want, without crashing.

On 11/27/06, Steven Baker <[EMAIL PROTECTED]> wrote:


Hello All,
we are using asterisk+openldap. Do is there any easy way to manage users
besides command line or the java "ldap browser?

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Re: [asterisk-users] cisco 2600

[Gary Richardson]

I'm using a 2811. Just about everything works fine, though we are experiencing a problem with redirecting calls.On 8/5/06, FaberK <
[EMAIL PROTECTED]> wrote:Hi,does anybody used cisco 2600 as * gateway with E1?
Thanks-- .:FaberK:.

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Re: [asterisk-users] LDAP athentication

[Gary Richardson]

I have some in house scripts, but it's definitely not real time.. It uses perl to generate various config files to be included. The philosophy behind it is to store the dial plan on the hard drive (and in some sort of rcs) and to generate phone objects  into separate config files. The LDAP schema is fairly abstracted -- there is no dial plan steps in it..
I could pass it along if you're interested.On 9/18/06, Andre O. <[EMAIL PROTECTED]> wrote:
Hello, Does anyone have a solution for having SIP users to authenticate against a LDAP server?Best Regards,Andre O.

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Re: [asterisk-users] Unable to make calls from CallManager to Asterisk

[Gary Richardson]

What version of call manager?On 9/5/06, Anantha Padmanabha.M.L <[EMAIL PROTECTED]> wrote:
HI,I have successfully integrated CallManager and Asterisk and was able to make call from one of Asterisk phone to CallManager 
Phone.But Could not able to make call from CallManager to asterisk.I have also tried the below link :-  
http://www.voip-info.org/wiki/index.php?page=Asterisk+Cisco+CallManager+Integration  But still not able to place calls from CallManager to Asterisk     Can anybody send me sample of Configuration that i have to make to make calls from CallManager to Asterisk.
  This is really Urgent for me!!.Thanks in Advance,Anantha 
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Re: [asterisk-users] canreinvite=yes and RTP dropping in and out

[Gary Richardson]

My next attempt at this is going to be putting a hub in between the path to the switch. I'm hoping to be able to sniff the packets to see what's going on.Also, using the network status page on the hard phones, the transmit and receive counters for the direction of the channel slows way down as if almost no data is being transmitted. 
How do I send a sip debug?Thanks.On 8/2/06, Joshua Colp <[EMAIL PROTECTED]> wrote:
- Original Message -From: Gary Richardson[mailto:[EMAIL PROTECTED]]To: Asterisk Users Mailing List -Non-Commercial Discussion [mailto:
asterisk-users@lists.digium.com]Sent:Wed, 02 Aug 2006 13:54:04 -0300Subject: [asterisk-users] canreinvite=yesand RTP dropping in and out> Hey guys,>> I'm having yet another strange problem. I've recently set canreinvite=yes,
> allowing the RTP streams to avoid our * server. Now, a few people are> experience one way audio drops on internal calls. External calls are working> fine (they re-invite directly to a Cisco router). Sometimes, if you wait 20
> seconds or more, the stream will resume. Flipping the person on and off hold> won't resume the stream.>> We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem> to happen all of the time. There are no sip messages being exchanged when
> the stream stops or restarts.>> Any suggestions?If the audio is going directly there's not too much you can do to examine it. There may be software out there to sniff the data on your network and examine the RTP stream, maybe even see when it drops out (if it really does drop out, ie: stream actually stops). I know there's some Windows software out there capable of this as I picked a copy up while at Spring VON but you might need to look around. OH - can you also send a sip debug with the reinvites? I'm just curious to see the RTP information in the SDP.
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[asterisk-users] canreinvite=yes and RTP dropping in and out

[Gary Richardson]

Hey guys,I'm having yet another strange problem. I've recently set canreinvite=yes, allowing the RTP streams to avoid our * server. Now, a few people are experience one way audio drops on internal calls. External calls are working fine (they re-invite directly to a Cisco router). Sometimes, if you wait 20 seconds or more, the stream will resume. Flipping the person on and off hold won't resume the stream.
We're using 7960 phones. Enabled_vad is set to 0 (disabled). It doesn't seem to happen all of the time. There are no sip messages being exchanged when the stream stops or restarts.Any suggestions?Thanks.

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Re: [Asterisk-Users] sip to h323 gateway ...

[Gary Richardson]

Nope, asterisk does the bridging. Asterisk can talk to SIP phones and H323 gateways/phones. It can also cross connect them.Since I have SIP users plugged into asterisk, I have a dial plan that looks something like:
exten => 100,1,Macro(local_sip_user,SIP/bill)exten => 101,1,Macro(local_sip_user,SIP/bob)exten => 102,1,Macro(local_sip_user,SIP/steve)exten => _XXX,1,Macro(call_ccm,${EXTEN})exten => _8XXX,1,Macro(call_ccm,${EXTEN:1})
So, if you dial 100-102, you get a sip call, but if you dial 103, it would try to dial my CCM. If you dial 8100, it would call CCM anyway.From the cisco side, I have some similar logic. That's pretty much it.
On 6/15/06, Cesc <[EMAIL PROTECTED]> wrote:
So, asterisk does the bridging ... I asked on another list and theanswer was that asterisk could not do the job :OThe truth is that my setup should be fairly simple ... i do not needany "cool" feature (voicemail and the like). I just need to call from
one side to the other, for a reduced amount of users (so name mappingcould even be manual ... no problem).CescOn 6/15/06, Gary Richardson <[EMAIL PROTECTED]
> wrote:> I'm bridging a Cisco Call Manager 3.2 system (h323 only) to an asterisk SIP> setup. It works. There are issues, but that has more to do with Unity> voicemail than the h323 implementations.
>>>  On 6/15/06, Cesc <[EMAIL PROTECTED]> wrote:> >>  Hi,>> I am familiar with asterisk, though never actually tinkered with one
> myself ... so i don't know the full extent of its capabilities.>> I am facing a request to bridge a sip network and an h323 network.>  I would like to operate the sip with ser as the proxy and some
> gatekeeper on the h323 side (not required though).> Actually, i have a few more points that may make it simpler> - i do not need codec negotiation: both sides are configured use> the same (g711 alaw) by default.
> - I have just a few "phones" on each side, so even "static routing"> can work, if that is of any help.> - it is not a production environment, for now. It is a demo/lab>
> The question is ... can asterisk do the job?>> Ideally, the bridge would be only signalling-wise (rtp to be direct> end-to-end). But, if someone had bad experience with this and would> recommend to use a B2BUA approach, please, tell me.
>> I don't know if it makes a difference, but most of the calls would go> from the H323 side to the SIP side ... but i don't really want to> restrict SIP->H323.>> Thanks a lot!
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Re: [Asterisk-Users] sip to h323 gateway ...

[Gary Richardson]

I'm bridging a Cisco Call Manager 3.2 system (h323 only) to an asterisk SIP setup. It works. There are issues, but that has more to do with Unity voicemail than the h323 implementations.
On 6/15/06, Cesc <[EMAIL PROTECTED]> wrote:
Hi,I am familiar with asterisk, though never actually tinkered with onemyself ... so i don't know the full extent of its capabilities.I am facing a request to bridge a sip network and an h323 network.
I would like to operate the sip with ser as the proxy and somegatekeeper on the h323 side (not required though).Actually, i have a few more points that may make it simpler- i do not need codec negotiation: both sides are configured use
the same (g711 alaw) by default.- I have just a few "phones" on each side, so even "static routing"can work, if that is of any help.- it is not a production environment, for now. It is a demo/lab
The question is ... can asterisk do the job?Ideally, the bridge would be only signalling-wise (rtp to be directend-to-end). But, if someone had bad experience with this and wouldrecommend to use a B2BUA approach, please, tell me.
I don't know if it makes a difference, but most of the calls would gofrom the H323 side to the SIP side ... but i don't really want torestrict SIP->H323.Thanks a lot!Cesc___
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Re: [Asterisk-Users] Audio cuts out

[Gary Richardson]

Hmm, out of curiosity, does anyone have experience with call recording and 3ware 9550SX cards? We're running a raid1 mirror.Our call recording load is more like 3-5 than 30-50, but I feel that there is some correlation between the two anyway at this point.
Thanks.On 6/12/06, Steve Totaro <[EMAIL PROTECTED]> wrote:
We battled this same issue for a couple weeks, at about 30-50simultaneous recordings the audio would get all screwy.I looked at that solution but opted for something a little morepassive.  I use orkaudio to sniff rtp streams and mux them.  I have it
to perfect quality, the same as the monitor app in asterisk.  I think itis a much better solution than ramdisk since it is so passive and putsno strain or need for additional RAM on the asterisk machine itself.
Let the phone system be a phone system and not a recording device I say.Best part about the orkaudio project is Henri.  I had audio issues withorkaudio in the beginning but Henri re-worked his software to eliminate
my problems in a matter of days.  A true credit and great contributionto opensource software.Thanks,Steve TotaroGary Richardson wrote:> That could be an issue. Would recording to a ram drive solve the problem?
>> Thanks.>> On 6/12/06, *Steve Totaro* < [EMAIL PROTECTED]> [EMAIL PROTECTED]
>> wrote:>> Recording many simultaneous calls can cause this behavior too.>> Thanks,> Steve Totaro>>> 
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Re: [Asterisk-Users] Audio cuts out

[Gary Richardson]

That could be an issue. Would recording to a ram drive solve the problem?Thanks.On 6/12/06, Steve Totaro <
[EMAIL PROTECTED]> wrote:Recording many simultaneous calls can cause this behavior too.
Thanks,Steve Totaro
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Re: [Asterisk-Users] Audio cuts out

[Gary Richardson]

We're not using any zaptel hardware though. I didn't think the echo cancellers would be doing anything? We're digital and sip from end to end. Do I need to disable echo cancellation in some way?
Thanks.On 6/12/06, Andrei (MPI) <[EMAIL PROTECTED]> wrote:
Gary,I would check echo cancelling parameters first. I've seen this to happenwith one of the zaptel echo cancellers. Try to change the default echoalgorithm in zconfig.h,  and recompile and install new zaptel. Also
zapata.conf echo parameters may need to be changed either way.AndreiGary Richardson wrote:> Hey All,>> I've been experiencing a problem for a bit. During a call to the PSTN,> audio will cut out for 2-5 seconds. It's completely random and may or
> may not happen during a call.>> Our setup is 79XX phones -> asterisk -> 2811 router -> PRI to the> PSTN. Everything is talking SIP. The asterisk box is a dual core> system. /proc/interrupts looks like:
>>  cat /proc/interrupts>CPU0   CPU1>   0:  733669449  732813122IO-APIC-edge  timer>   8:  1  0IO-APIC-edge  rtc>   9:  0  0   IO-APIC-level  acpi
>  14:65984106589174IO-APIC-edge  ide0> 169:  0  0   IO-APIC-level  uhci_hcd> 185:  0  0   IO-APIC-level  ehci_hcd, uhci_hcd> 193:  0  0   IO-APIC-level  uhci_hcd
> 201:  0  0   IO-APIC-level  uhci_hcd> 209:   11404158   10762030   IO-APIC-level  3w-9xxx> 225:  100440701136 PCI-MSI  eth0> 233: 14   10512166 PCI-MSI  eth1
> NMI:  0  0> LOC: 1466464719 1466464718> ERR:  0> MIS:  0>> Can-Reinvite is enabled, but I do have it configured to allow call> recording on outbound calls, so I think the audio streams all go
> through asterisk. There are no G.729 licenses involved and everything> should be talking G.711.>> Oh, and this is an 1.2.7.1  install. ztdummy is loaded.>> Does anyone have any insite into this problem?>> Thanks.> >> ___
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[Asterisk-Users] Audio cuts out

[Gary Richardson]

Hey All,I've been experiencing a problem for a bit. During a call to the PSTN, audio will cut out for 2-5 seconds. It's completely random and may or may not happen during a call.Our setup is 79XX phones -> asterisk -> 2811 router -> PRI to the PSTN. Everything is talking SIP. The asterisk box is a dual core system. /proc/interrupts looks like:
 cat /proc/interrupts    CPU0   CPU1     0:  733669449  732813122    IO-APIC-edge  timer  8:  1  0    IO-APIC-edge  rtc  9:  0  0   IO-APIC-level  acpi
 14:    6598410    6589174    IO-APIC-edge  ide0169:  0  0   IO-APIC-level  uhci_hcd185:  0  0   IO-APIC-level  ehci_hcd, uhci_hcd193:  0  0   IO-APIC-level  uhci_hcd
201:  0  0   IO-APIC-level  uhci_hcd209:   11404158   10762030   IO-APIC-level  3w-9xxx225:  100440701    136 PCI-MSI  eth0233: 14   10512166 PCI-MSI  eth1NMI:  0  0 
LOC: 1466464719 1466464718 ERR:  0MIS:  0Can-Reinvite is enabled, but I do have it configured to allow call recording on outbound calls, so I think the audio streams all go through asterisk. There are no 
G.729 licenses involved and everything should be talking G.711. Oh, and this is an 1.2.7.1 install. ztdummy is loaded.Does anyone have any insite into this problem?Thanks.

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Re: [Asterisk-Users] PSTN -> CCM3.2 -> Asterisk CLID

[Gary Richardson]

I've made the change, but it didn't make a difference. Unity is currently acting as our IVR. Would that make any difference?Thanks.On 5/24/06, Greg Oliver
 <[EMAIL PROTECTED]> wrote:On Wed, 2006-05-24 at 07:30 -0700, Gary Richardson wrote:
> On the route pattern configuration page, there isn't a 'redirecting> number' option. The closes thing I have is "Use Calling Party's> External Phone Number Mask". This is a 3.2 install of callmanager.
>> On the gateway configuration page, there is a 'Calling Party> Selection' box. Changing the values in that drop down does not have> any affect on the callerid.>> Thanks.>
> On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:> On Tue, 2006-05-23 at 10:46 -0700, Gary Richardson wrote:> > Hey guys,> >
> > When a call comes in via the PSTN to our Call Manager 3.2> and is> > forwarded (via unity and H323), the caller id is set to our> Unity> > Voicemail instead of the caller id from the PSTN. We're
> using the> > oh323 channel in this case.> >> > Has anyone experienced this issue before? Any solutions?> >Sorry - you're right - is "first redirect number" set on outbound calls
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Re: [Asterisk-Users] How to add H.323 channels on Asterisk 1.2.7.1

[Gary Richardson]

There are 3 different h323 providers for asterisk, chan_h323.so (which comes in the asterisk source), oh323 and ooh323.You can read more about them in 
http://www.voip-info.org/wiki-Asterisk+H323+channels. Remember that most of them require something called pwlib and openh323 to work.On 5/24/06, Daye
 <[EMAIL PROTECTED]> wrote:How do I add 
H.323 channel on Asterisk 1.2.7.1? Thanks
		New Yahoo! Messenger with Voice. 
Call regular phones from your PC and save big.
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Re: [Asterisk-Users] PSTN -> CCM3.2 -> Asterisk CLID

[Gary Richardson]

On the route pattern configuration page, there isn't a 'redirecting number' option. The closes thing I have is "Use Calling Party's External Phone Number Mask". This is a 3.2 install of callmanager.On the gateway configuration page, there is a 'Calling Party Selection' box. Changing the values in that drop down does not have any affect on the callerid.
Thanks.On 5/23/06, Greg Oliver <[EMAIL PROTECTED]> wrote:
On Tue, 2006-05-23 at 10:46 -0700, Gary Richardson wrote:> Hey guys,>> When a call comes in via the PSTN to our Call Manager 3.2 and is> forwarded (via unity and H323), the caller id is set to our Unity
> Voicemail instead of the caller id from the PSTN. We're using the> oh323 channel in this case.>> Has anyone experienced this issue before? Any solutions?>> Thanks.> ___
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http://lists.digium.com/mailman/listinfo/asterisk-usersYes - you probably have "redirecting number" set on you rroute pattern..Disabling that will send the correct CID, but vmail will not work..  As
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[Asterisk-Users] PSTN -> CCM3.2 -> Asterisk CLID

[Gary Richardson]

Hey guys,When a call comes in via the PSTN to our Call Manager 3.2 and is forwarded (via unity and H323), the caller id is set to our Unity Voicemail instead of the caller id from the PSTN. We're using the oh323 channel in this case.
Has anyone experienced this issue before? Any solutions?Thanks.
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Re: [Asterisk-Users] CCM 3.3 and Asterisk

[Gary Richardson]

I'm using Asterisk + OH323 with CCM 3.2. There are a few quirks with codec negotiation and callerid. We're moving as fast as we can to dump the CCM, so we're not too worried about them.
On 5/15/06, Gustavo Souza Queiroz <[EMAIL PROTECTED]> wrote:

Hello,
I´m have a CCM 3.3 and Asterisk in my
LAN.
I need connect my Asterisk in my CCM
3.3.
You can a help me?

Thank´s

Gustavo Souza Queiroz.
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Re: [Asterisk-Users] How does asterisk behave when multiple phones are logged in on a single SIP/account?

[Gary Richardson]

The last sip device to register gets the call. The way around this is to have your sip devices register under different accounts and create a ring group (dial(SIP/dev1&SIP/dev2&SIP/devN))AFAIK, there isn't a reliable method of determining if a sip device is busy other than calling it. 
On 5/1/06, Arne Morten Johansen <[EMAIL PROTECTED]> wrote:













Hi.

How does this work?

What if this SIP/account was a member (agent) of
a queue? 

Ex: dial(SIP/account,20,tT). Would the dialstatus
be set as busy when one of the phones is actively talking, or will the other
phones continue to ring?

You may have seen my other submissions to this
list. I'm looking for a way to make the other phones in a group
unavailable when one of them is busy. Because one person will have multiple
phones. 

Thanks

Arne Morten Johansen. 







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Re: [Asterisk-Users] Asterisk redundancy

[Gary Richardson]

On 4/17/06, Joseph Rothstein <[EMAIL PROTECTED]> wrote:
Is anyone using a PRI to Ethernet bridge, or any other kind of E1 GW thatwould allow failover to an alternate Asterisk box without manually switchingthe cable? This one is a litteexpensive(
http://www.mapleleaf-technologies.com/webstore/ethernetbridges.php), but seems like it would do the trick. But I would have to run TDMoEbetween the Asterisk boxes and the bridge. Not a big deal probably, but I
have no experience with TDMoEI'm using the PRI to Ethernet bridge from Cisco. It's call a 2800 router with a T1 card. Of course, if my router dies, I'm SOL. I could have two routers and balance calls across them. Of course, any active calls to the dead router would be lost, but it's pretty cheap, all considering it uses SIP.

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Re: [Asterisk-Users] Monitor or mixmonitor

[Gary Richardson]

I'm using MixMonitor. Be aware that some people encounter a bug where MixMonitor stops recording at random (see http://bugs.digium.com/view.php?id=6457). There are a couple of working patches for it.
Thanks.On 4/3/06, Wai Wu <[EMAIL PROTECTED]> wrote:
Hi all,I am setting up a script to record all the call. There are two app for recording. "Monitor" and "Mixmonitor", one mixing the audio on the fly and one mixing it at the end but also allow a option not to mixing the audio at all. If mixing the audio on the fly is not that taxing on the CPU, I would like to use 'mixmonitor' app. My question is, what is penalty on the CPU when mixing the audio on the fly? I know this is the better option, but I don't really need the 'in' and 'out' audio mixed until it's played back, and which happens less than 5% of the time. What are your thoughts?
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Re: [Asterisk-Users] Testing asterisk faxing functionality

[Gary Richardson]

I was playing with the fax stuff over IP on Friday. Unless you're
receiving faxes from a PSTN circuit, it doesn't work so well.

Also, I don't think you can chain txfax and rxfax like that. When you
hit the s,2 part, it's going to play the fax out to the handset you
dialed from. You'll need something like hylafax to send the fax.

And you probably want to call Dial(Local/[EMAIL PROTECTED]) to call a
local extension..

On 3/27/06, patryk <[EMAIL PROTECTED]> wrote:
> I have asterisk with rxfax txfax modules.I want
> to test fax sendig and reciving in one asterisk
> instance, in extensions.conf I have :
>
> exten => 1234567,1,rxfax(/home/patryk/fax-new.tif|debug)
>
> exten => s,1,Dial(1234567)
> exten => s,2,txfax(/home/patryk/fax.tif|caller|debug)
>
> but I doesn't seem to work.But when I'm calling on this number I can
> hear fax tones.
> I can't find sip client with fax fuctionality for linux I think it would
> help with testing.
>
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Re: [Asterisk-Users] Config File Management

[Gary Richardson]

I'm using CVS. I only have one server right now. I use it on other
clusters to sync files and it works for me..

On 3/27/06, Douglas Garstang <[EMAIL PROTECTED]> wrote:
> I'm curious (ok, well I admit it - it's for perosnal gain) what methods 
> people are using to manage asterisk config files when they have multiple 
> asterisk systems?
>
> Some sort of revision control such as cvs,rcs or subversion?
>
> A central 'config server' where you edit the files and then rsync them out?
>
> I have 5 systems to manage, and it seems that about the only common file is 
> extensions.conf. All the other files, even sip.conf have subtle differences 
> which preclude them from being the same file (binaddr for example).
>
> Thanks,
> Doug.
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Re: FW: [Asterisk-Users] Asterisk Users

[Gary Richardson]

You wouldn't have to post anonymously -- only if it makes you feel better.

I could have really used such a resource in January -- Digium's list
of success stories is a little thin.

On 3/24/06, Aaron Daniel <[EMAIL PROTECTED]> wrote:
> > Perhaps a page on the wiki would work?  We could set the ground rules
> > similar to other industries:  no names, nothing more defining than a
> > region, the number of units, etc.  Would that be useful?
> >
> > For example, I can describe this organization as a security company in
> > Southwest Missouri using asterisk with 60 sets and 16 lines.
> >
> > When you strip off my name and email, it gets a little less certain who
> > I am talking about...
> >
> >
> > Bob McDowell
>
> I like the idea of having the information on the wiki, makes it simpler
> for everyone to see just how well the project is doing.  I'm not sure
> about the removing identifying information part is such a good idea, since
> the best way for people to trust a system is to talk to people that have
> used it before.  Or do we just want the information to filter through the
> asterisk-users list?
>
>
> --
> Aaron Daniel
> Computer Systems Technician
> Sam Houston State University
> [EMAIL PROTECTED]
> (936) 294-4198
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Re: [Asterisk-Users] Call Monitoring?

[Gary Richardson]

You could use contexts for this. By default put everyone into the
'internal' context. Managers would go into the 'managers' context,
which would include the 'internal' context.

The manager context specifically would have the exten's to monitor or
barge into calls. By including the internal context, they'd have the
same dialplan otherwise.

You determine which context a user gets by default in sip.conf (if
you're using sip phones..).

On 3/23/06, Charles Marcus <[EMAIL PROTECTED]> wrote:
> 1. Is Asterisk capable of allowing for setting up Groups so that only
> one extension in a Group can selectively monitor one of the other
> extensions in the Group (but none of the others can initiate it)?
>
> This would be for Managers to listen to Sales Calls of other members of
> their Team, to provide feedback to the Rep for training purposes.
>
> 2. Alternatively, can a Group be defined that will allow multiple
> extensions to listen in on another call in progress?
>
> Again, we want to use this kind of functionality to do some Sales
> Technique Training calls.
>
> --
>
> Best regards,
>
> Charles
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Re: [Asterisk-Users] Asterisk Users

[Gary Richardson]

We've been running on an ICS7750 for almost 4 years. It's ridiculously
expensive. We've looked at the cost of setting up call center like
features, call recording etc and it boils down to a forklift upgrade
that will end up costing anywhere from $100-$250K. That's partially
our problem, since we bought the ICS unit in the first place.

Plus, the ICS7750 in particular has been very unreliable for us. The
switching backplain died 3 times in it. The memory for the SPE blades
was underspeced for unity (especially after SQL server memory leaks).

We never upgraded off of callmanager 3.2. We got our system barely
stable, but never working properly. It seems like every time I walk
past it in the lan room, somethings goes wrong. TAC couldn't really
solve our problems. The software itself is far too complex and
unstable. It takes 15-25 screens to configure a new user with
voicemail. We've never be able to properly use conference bridges --
they die after 3 minutes.

Possibly CCM4/5 is better. We'll never know at this point.

I suppose I'm just whining now. Cisco has tried really hard to help us out.

-- A jaded Cisco user.

On 3/23/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote:
> Just out of curiosity what was the reasons for migrating off of Cisco?  This
> is interesting since I've run across people who swear by Cisco, but the
> costs involved are just too unreasonable for me.
>
>
>
> On 3/23/06, Gary Richardson <[EMAIL PROTECTED]> wrote:
> > I'm currently 1 PRI with 25 phones. We'll be 2 PRI's and over 100
> > phones by mid summer. We're currently 5 DID's, but I'm pretty sure
> > we'll be around 50 when were done.
> >
> > We're currently migrating off of a cisco call manager. I recommend
> > going slow with your transition -- do a chunk of phones, wait a week
> > or two before doing more. Asterisk excels at interoping with other
> > systems; take advantage of this.
> >
> > Thanks.
> >
> > On 3/22/06, QUICK, RANDY <[EMAIL PROTECTED]> wrote:
> > >
> > >
> > > Can you guys and girls give me some examples of companies using Asterisk
> and
> > > how many DIDs you have.  I have built a small system and tested it with
> > > AASTRA 480i's and all is working perfectly.  I go in front of my
> Management
> > > Board tomorrow to demo the app and show them it is a viable solution.
> We
> > > are a medical facility with 12 facilities and a total of 1700 phones.
> Any
> > > info you have would be a huge help when they ask who else is using it.
> > >
> > > Thanks in advance!
> > >
> > >
> > > Randy Quick
> > > Communications Technician II
> > > Texoma Healthcare Systems
> > > 903.416.4398
> > > [EMAIL PROTECTED]
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> > >
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> > >
> > >
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Re: [Asterisk-Users] Asterisk Users

[Gary Richardson]

I'm currently 1 PRI with 25 phones. We'll be 2 PRI's and over 100
phones by mid summer. We're currently 5 DID's, but I'm pretty sure
we'll be around 50 when were done.

We're currently migrating off of a cisco call manager. I recommend
going slow with your transition -- do a chunk of phones, wait a week
or two before doing more. Asterisk excels at interoping with other
systems; take advantage of this.

Thanks.

On 3/22/06, QUICK, RANDY <[EMAIL PROTECTED]> wrote:
>
>
> Can you guys and girls give me some examples of companies using Asterisk and
> how many DIDs you have.  I have built a small system and tested it with
> AASTRA 480i's and all is working perfectly.  I go in front of my Management
> Board tomorrow to demo the app and show them it is a viable solution.  We
> are a medical facility with 12 facilities and a total of 1700 phones.  Any
> info you have would be a huge help when they ask who else is using it.
>
> Thanks in advance!
>
>
> Randy Quick
> Communications Technician II
> Texoma Healthcare Systems
> 903.416.4398
> [EMAIL PROTECTED]
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Re: [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud

[Gary Richardson]

It looks like there is lots of discussion already going on about it at
http://bugs.digium.com/view.php?id=6457

On 3/7/06, Matt Riddell [NZ] <[EMAIL PROTECTED]> wrote:
> The only catalyst to getting it fixed will be if someone posts a bug
> entry with full details on bugs.digium.com
>
> If you do, post again here with the ID and discussion and testing can
> continue there.
>
> --
> Cheers,
>
> Matt Riddell
> ___
>
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Re: [Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud

[Gary Richardson]

I'm running 1.2.4 and just about every call is cut short. I'm using Cisco IP phones as end points. All the outbound calls are routed via SIP through a PRI line attached to a Cisco 2811..
On 3/2/06, Johnathan Corgan <[EMAIL PROTECTED]> wrote:
Gary Richardson wrote:> Now it seems that if I'm really loud on a call, MixMonitor stops> recording. The wav file stops growing. The log says nothing. When you> hang up the call, MixMonitor reports that it is exiting, even though
> it hasn't been recording since that loud noise.>> Has anyone experienced such a problem with MixMonitor? Is MixMonitor> well tested?I've seen exactly this with MixMonitor in 1.2.1, but I hadn't isolated
it to volume issues, just random occurrences.I haven't seen it yet in a week on 1.2.4, but I don't know if the bug isgone or it just hasn't triggered yet.-Johnathan___
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Re: [Asterisk-Users] Program Buttons on Cisco 79xx Phones

[Gary Richardson]

Cisco has a book about it: http://www.amazon.com/gp/product/1587050609/sr=8-1/qid=1141401215/ref=pd_bbs_1/103-7257053-6939020?%5Fencoding=UTF8
While this isn't specifically about the SIP image, the XML browser is the same. I also use Cisco::IPPhone (http://search.cpan.org/~mrpalmer/Cisco-IPPhone-0.05/IPPhone.pm
) for the backend.On 3/3/06, Kevin Steil <[EMAIL PROTECTED]> wrote:













Does anyone have a good resource to learn how to program the
soft and hard buttons on a Cisco 7940 or 7960 phone?  Using SIP Firmware…thanks.







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Re: [Asterisk-Users] Autofill phonebook??

[Gary Richardson]

A lot of phones support LDAP for this purpose. I don't think X-Lite does, but I've never checked.On 3/3/06, Amaury Rodriguez <
[EMAIL PROTECTED]> wrote:Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server?
Amaury Rodríguez 
http://liberadospucmm.blogspot.com http://groups.msn.com/telematicaPUCMM2002 
  Go OpenSource And Be FREE!!   

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Re: [Asterisk-Users] Autofill phonebook??

[Gary Richardson]

A lot of phones support LDAP for this purpose. I don't think X-Lite does, but I've never checked.On 3/3/06, Amaury Rodriguez <
[EMAIL PROTECTED]> wrote:Most softphnes (like Idefisk and X-Lite) have a phonebook. Is there a way I can fill those phonebooks with info from the Asterisk server?
Amaury Rodríguez 
http://liberadospucmm.blogspot.com http://groups.msn.com/telematicaPUCMM2002 
  Go OpenSource And Be FREE!!   

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[Asterisk-Users] MixMonitor Problems -- sssshh, don't be too loud

[Gary Richardson]

Hey,I've come across two interesting problems today.First, when recording long calls using Monitor(), it appears the in and out channels become out of sync. It seems like one channel happens faster or has data missing when sox mixes them together. 
Digging around, I found MixMonitor, which skips the whole soxmix process. I figured that removing that step could only help.Now it seems that if I'm really loud on a call, MixMonitor stops recording. The wav file stops growing. The log says nothing. When you hang up the call, MixMonitor reports that it is exiting, even though it hasn't been recording since that loud noise.
Has anyone experienced such a problem with MixMonitor? Is MixMonitor well tested?Thanks.
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Re: [Asterisk-Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)

[Gary Richardson]

You don't need switchport mode trunk when using switchport voice vlan.. On 3/1/06, Nicholas Kathmann <
[EMAIL PROTECTED]> wrote:Joao Pereira wrote:> Hello to all
> I would like to know If some of you have already configured an Cisco> IP Phone (7940 or 7960) to work in a different VLAN than the PC that> is connected through the phone switch?> I know that this can be done with the Skinny firmware, but I dont if
> it works with the SIP firmware.>> The Cisco technical staff told me that these phones dont support> 802.1x but can work as pass-through. This way I can still use the PCs> with 802.1x and the phones in the same Ethernet plug.
>> Did someone made it with the Cisco IP phones? What configuration do I> need in the phones and in the switch?> Thanks> Joao Pereira>If configuring with Cisco switches, I'm pretty sure they pull the
information for which VLAN to operate in from the switch.  You have toconfigure the switchports on the Cisco switch like so:interface fastethernet 0/1   switchport trunk native vlan 
   switchport mode trunk   switchport voice vlan    spanning-tree portfast trunketc.Thanks,Nicholas Kathmann, CISSPKathmann Consulting, LLC___
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Re: [Asterisk-Users] Lowering Server Load

[Gary Richardson]

A good hardware SATA raid card should use less CPU time as well. I like to use 3ware cards myself.On 3/2/06, Ron McCarthy <
[EMAIL PROTECTED]> wrote:Also, SATA on a onboard SATA card will eat more CPU then a SCSI system.
Are you running software RAID by chance with your SATA? SCSI or SCSI
Raid will not each CPU near as much since the HBA does all the work and
does tie up the CPU with all its I/O's. We have successfulyl recorded
5+ calls at a time via dual xeon 3.0 with 10K SCSI drives in RAID-5
with no issuses running about 30 PRI channels and anywhere from 50-75
SIP channels, all with g729 encoding.

Hope this helps!
RonOn 3/2/06, Anton Krall <
[EMAIL PROTECTED]> wrote:
Yep, I tried it and indeed, it lowers cpu usage, so I switched from wav togsm format and Im thinking about doing the ramdisk solution for recording...Sounds like a good move?|-Original Message-

|From: [EMAIL PROTECTED]|[mailto:
[EMAIL PROTECTED]] On Behalf Of
|Matt Riddell [NZ]|Sent: Thursday, March 02, 2006 2:04 AM|To: Asterisk Users Mailing List - Non-Commercial Discussion|Subject: Re: [Asterisk-Users] Lowering Server Load||Can you try not recording for a bit and see if that helps?
||--|Cheers,||Matt Riddell|___||http://www.sineapps.com/news.php (Daily Asterisk News - html)|http://freevoip.gedameurope.com (Free Asterisk Voip Community)
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Re: [Asterisk-Users] Re: Call queue design issues and suggestions

[Gary Richardson]

I haven't done anything with call queues specifically.I do own a copy of "Developing Cisco IP Phone Services" (
http://www.amazon.ca/exec/obidos/ASIN/1587050609/qid%3D1140624389/701-7618472-4656313), though I only have ever used about 45 pages of it. I also use 
http://search.cpan.org/~mrpalmer/Cisco-IPPhone-0.05/IPPhone.pm to make my life a little easier.On 2/22/06, Tomislav Parčina <
[EMAIL PROTECTED]> wrote:In article <
[EMAIL PROTECTED]>, [EMAIL PROTECTED] says...> I don't know if this works for you, but I use the following mechanism. I
> don't use the agent call back stuff, just the (Add|Remove)QueueMember stuff.>> For each queue, dialing the extension (), puts the caller into the queue> (ie, a customer calling for reservations). I use ** to sign a phone into
> the queue and * to sign out of a queue.Good idea, maybe sometimes I'll need it.> You can use the manager to see who is currently logged into a port. It> doesn't take much to write a cgi script that outputs the Cisco XML for the
> phones. I've built a few apps that do interesting things. It would be quite> easy to write an app that:It could be easy for someone with experience, but if you have never done it before (like me) it isn't like that. Can you send us what you have done?
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Re: [Asterisk-Users] Call queue design issues and suggestions

[Gary Richardson]

I don't know if this works for you, but I use the following mechanism. I don't use the agent call back stuff, just the (Add|Remove)QueueMember stuff.For each queue, dialing the extension (), puts the caller into the queue (ie, a customer calling for reservations). I use ** to sign a phone into the queue and * to sign out of a queue. 
You can use the manager to see who is currently logged into a port. It doesn't take much to write a cgi script that outputs the Cisco XML for the phones. I've built a few apps that do interesting things. It would be quite easy to write an app that:
   - displays what queues a phone is signed into   - globally sign out of all queues   - sign into or out of individual queuesThanks.On 2/21/06, 
Joe <[EMAIL PROTECTED]> wrote:
Greetings to all.I am currently implementing call queues for a customer and have come acrossseveral "problems".The customer is an airline representative, and will be using call queues for
different airline reservations. The customer requires that any agent be ableto login to any number of queues. This means that queue members have to bedynamic, not using "member => agent/101" for example.
I am not sure of the best way to accomplish this.I initially just setup agentcallback, and hard coded the agents in eachqueue, but this means that when an agent logs in he/she will be in allqueues where member => agent/xxx.
My next thought was to use a combination of agentcallback and addQueueMemberto add SIP extensions to particular queues. I currently have a mechanism bywhich the user can dial a number, enter the two letter airline code, mysql
translates this airline code into a real queue name, and the user is thenadded to this queue. Of course the two letter airline code could be used forthe queue name to avoid the mysql lookup, something like queue-xx. Along
these lines, does anyone know if it is possible to use AddQueueMember withAgent/xxx, or just with real extensions? The main problem with this is thatthere would be no way to globally logoff agents (if real extensions had to
be used) from all the queues they are logged in to.My current idea is to use agentcallback in combination with a php/myslinterface. This of course would require realtime queue configuration. Theuser would use agentcallback to login, and the web interface to choose the
queues he/she wanted to join.The customer also wants a way of seeing which queues the agents are loggedinto. This could also be run from mysql backend. I would also like to somehow integrate this into the Cisco 7940 xml capabilities.
Would love to hear form anyone regarding these issues.Regards,Joe___--Bandwidth and Colocation provided by 
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[Asterisk-Users] 79xx's and call queues

[Gary Richardson]

Hey,I'm testing out some call queues. I have 7940's and 7960's with the SIP 7.4 image.I have a queue that looks something like:[testqueue]strategy = rrmemorytimeout = 15retry = 5weight = 0
announce-frequency = 0joinemtpy = yesreportholdtime = yesI dynamically add a phone or two to the queue (AddQueueMember, not agents). When a caller calls in, connections are made and everything is fine. When a second person calls in, each queue member that is currently in a call gets a call waiting beep in a round robin fashion.
Is this how it happens on non-Cisco phones, or is there something with how Cisco does line appearances causing this? This happens when 1 or more line appearances are configured on the phone.Thanks.
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Re: [Asterisk-Users] Cisco 2620 as PRI gateway

[Gary Richardson]

Hey Tim,My 2800 does H.323 to a CCM and SIP to my asterisk box. I actually don't forward calls directly to my asterisk box from the 2800. As Juan pointed out, you need to set up your dial peers so that your 2600 knows what to do with the calls. I'm not a guru when it comes to configuring Cisco routers for T1 voice connectivity. I'm not going to pretend to know how to do it or recommend how you do it. 
Here are a couple of snippets from my config that may get you on the right path:dial-peer voice 430 pots destination-pattern 9[2-9]11 port 0/0/0:23 forward-digits 3!dial-peer voice 400 pots
 destination-pattern 91.. port 0/0/0:23 forward-digits 11!dial-peer voice 410 pots destination-pattern 9604... port 0/0/0:23 forward-digits 10!My area code is 604.. 
I have several more of them for international dialing and a few other patterns. I hope that gets you on the right track.On 2/8/06, Tim Reimers
 <[EMAIL PROTECTED]> wrote:





Gary-
 
You have a Cisco 2600 acting as both a SIP gateway to the 
Asterisk box -and- as an H.323 gateway for your CCM?
 
Can you provide me with config details?
 
 12.3(8)T3,c2600-ipvoice-mz.123-8.T3
with T1 (2 Port) Multi-Flex Trunk
 
I tried to set up h.323 to the Asterisk box-- didn't know 
there was a possibility of running SIP and H.323 at the same 
time...
 
thanks, Tim


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED]] On Behalf Of Gary 
RichardsonSent: Tuesday, February 07, 2006 9:09 PMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: 
[Asterisk-Users] Cisco 2620 as PRI gateway
I have a 2811 working as a SIP gateway. My IOS version is 12.3(11)T5. 
Looking through my config I notice:sip-ua  sip-server 
ipv4:Everything else in the config 
file is for our h323 call manager gear. I can't remember if I needed to add the 
above line to make a sip server run on the router. In order to place a call to 
the PSTN, I Dial(SIP/9XX@) and everything 
works. As for how much of this applies to a 2600.. you'll have to 
see.
On 2/6/06, Schochet, 
Wes <[EMAIL PROTECTED] 
> wrote:
I 
  just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it.  Can I 
  make this thing into MGCP gateway or even a SIP gateway for 
  asterisk?  Seems likeit should bee useful for 
  something!I'm perfectly happy to do my homework, but also don't feel 
  thee need toreinvent the wheel!  So, links with relevant info 
  would be appreciated.  If there is a config for a 2621 being 
  used as a gateway out there somewhere, Iwouldn't be too proud to take a 
  look at that either!  Asterisk configs wouldbe great 
  too!Thanks,Wes___ 
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Re: [Asterisk-Users] Cisco 2620 as PRI gateway

[Gary Richardson]

I have a 2811 working as a SIP gateway. My IOS version is 12.3(11)T5. Looking through my config I notice:sip-ua  sip-server ipv4:Everything else in the config file is for our h323 call manager gear. I can't remember if I needed to add the above line to make a sip server run on the router. In order to place a call to the PSTN, I Dial(SIP/9XX@) and everything works.
As for how much of this applies to a 2600.. you'll have to see.On 2/6/06, Schochet, Wes <[EMAIL PROTECTED]
> wrote:I just inherited a Cisco 2621 with a VWIC-1MFT-T1 card in it.  Can I make
this thing into MGCP gateway or even a SIP gateway for asterisk?  Seems likeit should bee useful for something!I'm perfectly happy to do my homework, but also don't feel thee need toreinvent the wheel!  So, links with relevant info would be appreciated.  If
there is a config for a 2621 being used as a gateway out there somewhere, Iwouldn't be too proud to take a look at that either!  Asterisk configs wouldbe great too!Thanks,Wes___
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[Asterisk-Users] moh about twice as fast

[Gary Richardson]

Hey guys,I'm trying to get music on hold working. I have a wav file. It plays fine on my windows laptop in all sorts of audio applications. If I put it on our asterisk 1.2.4 box and do something like:sox -V nov_2005.wav /var/lib/asterisk/mohmp3/nov_2005.raw
sox: Detected file format type: wavsox: Chunk fmtsox: Chunk factsox: Chunk datasox: Reading Wave file: Microsoft U-law format, 1 channel, 8000 samp/secsox: 8000 byte/sec, 1 block align, 8 bits/samp, 3414263 data bytes
sox: Input file nov_2005.wav: using sample rate 8000    size bytes, encoding u-law, 1 channelsox: Output file nov_2005.raw: using sample rate 8000    size bytes, encoding u-law, 1 channeland then hook it up in 
musiconhold.conf like:[default]mode=filesdirectory=/var/lib/asterisk/mohmp3/And make a call and stick it on hold, the music is playing roughly twice too fast. If I use the stock mp3's that come with 
[EMAIL PROTECTED], all is good. If I dosox -V -r 4000 nov_2005.wav -r 8000 nov_2005.wavThe file is played back at the right speed, but is highly distorted. I'm sure this is some rookie mistake I'm making.. Can anyone help me out?
Thanks.
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Re: [Asterisk-Users] Re: CallerID Problem

[Gary Richardson]

Hmm, that's annoying.

If I Set(CALLERID(num)=) (ie, I unset it), the callerid is set to the
default on the router and everything works as expected..

Thanks guys :)

On 2/1/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> This is what i found on Cisco's site:
>
> "Symptoms: Media negotiation fails for SIP calls and the terminating gateway 
> replies with a "488" message to an Invite message.
>
> Conditions: This symptom is observed on a Cisco platform when the terminating 
> gateway is configured with the G279B (annex B) codec and when the Session 
> Description Protocol (SDP) for the incoming Invite message does not have any 
> FMTP attribute line, which means that the default value, that is, the G279B 
> (annex B) codec, is used.
>
> Workaround: There is no workaround."
>
> Regards,
> Jan
>
> -Ursprungligt meddelande-
> Från: [EMAIL PROTECTED] genom Gary Richardson
> Skickat: on 2006-02-01 21:45
> Till: Asterisk Users Mailing List - Non-Commercial Discussion
> Ämne: Re: [Asterisk-Users] Re: CallerID Problem
>
> No, I'm not including the <> -- I was trying to show that it was
> something that I removed from my example..
>
> Thanks.
>
> On 2/1/06, Bromont Quebec <[EMAIL PROTECTED]> wrote:
> > Are you actually putting the < > in there?
> >
> > try:
> >
> > exten => _9.,1,Set(CALLERID(number)=MAINNUMBER)
> >
> > Hey,
> >
> > I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
> > connects to it using SIP. The asterisk version is 1.2.0.
> >
> > In my sip.conf, I set callerid="First Last" 
> >
> > When I make a an outbound call with the following macro:
> >
> > exten => _9.,1,Dial(SIP/${EXTEN}@,,w)
> > exten => _9.,2,Congestion()
> >
> > The caller id is set to the extension that's defined in sip.conf.
> >
> > If I try something like:
> >
> > exten => _9.,1,Set(CALLERID(number)=)
> > exten => _9.,2,Dial(SIP/${EXTEN}@,,w)
> > exten => _9.,3,Congestion()
> >
> > I get the following error:
> >
> > -- Got SIP response 488 "Not Acceptable Media" back from 
> >
> > It all works fine if I don't set the caller id.. Any ideas on why this
> > may be happening?
> >
> > Thanks.
> >
> >
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Re: [Asterisk-Users] Re: CallerID Problem

[Gary Richardson]

No, I'm not including the <> -- I was trying to show that it was
something that I removed from my example..

Thanks.

On 2/1/06, Bromont Quebec <[EMAIL PROTECTED]> wrote:
> Are you actually putting the < > in there?
>
> try:
>
> exten => _9.,1,Set(CALLERID(number)=MAINNUMBER)
>
> Hey,
>
> I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
> connects to it using SIP. The asterisk version is 1.2.0.
>
> In my sip.conf, I set callerid="First Last" 
>
> When I make a an outbound call with the following macro:
>
> exten => _9.,1,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,2,Congestion()
>
> The caller id is set to the extension that's defined in sip.conf.
>
> If I try something like:
>
> exten => _9.,1,Set(CALLERID(number)=)
> exten => _9.,2,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,3,Congestion()
>
> I get the following error:
>
> -- Got SIP response 488 "Not Acceptable Media" back from 
>
> It all works fine if I don't set the caller id.. Any ideas on why this
> may be happening?
>
> Thanks.
>
>
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Re: [Asterisk-Users] CallerID Problem

[Gary Richardson]

Num and Number are aliases, I believe.

I just tried it and I got the same error..

Thanks.

On 2/1/06, Jason Adams <[EMAIL PROTECTED]> wrote:
> Have you tried this:
>
> exten => _9.,1,Set(CALLERID(num)=)
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gary
> Richardson
> Sent: Wednesday, February 01, 2006 12:47 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] CallerID Problem
>
> Hey,
>
> I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
> connects to it using SIP. The asterisk version is 1.2.0.
>
> In my sip.conf, I set callerid="First Last" 
>
> When I make a an outbound call with the following macro:
>
> exten => _9.,1,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,2,Congestion()
>
> The caller id is set to the extension that's defined in sip.conf.
>
> If I try something like:
>
> exten => _9.,1,Set(CALLERID(number)=)
> exten => _9.,2,Dial(SIP/${EXTEN}@,,w)
> exten => _9.,3,Congestion()
>
> I get the following error:
>
> -- Got SIP response 488 "Not Acceptable Media" back from 
>
> It all works fine if I don't set the caller id.. Any ideas on why this
> may be happening?
>
> Thanks.
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[Asterisk-Users] CallerID Problem

[Gary Richardson]

Hey,

I'm using a Cisco 2811 to make calls out to a PRI. My asterisk box
connects to it using SIP. The asterisk version is 1.2.0.

In my sip.conf, I set callerid="First Last" 

When I make a an outbound call with the following macro:

exten => _9.,1,Dial(SIP/${EXTEN}@,,w)
exten => _9.,2,Congestion()

The caller id is set to the extension that's defined in sip.conf.

If I try something like:

exten => _9.,1,Set(CALLERID(number)=)
exten => _9.,2,Dial(SIP/${EXTEN}@,,w)
exten => _9.,3,Congestion()

I get the following error:

-- Got SIP response 488 "Not Acceptable Media" back from 

It all works fine if I don't set the caller id.. Any ideas on why this
may be happening?

Thanks.
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Re: [Asterisk-Users] extension to extension dialing

[Gary Richardson]

In your sip.conf, make sure these phones have a Type=Friend entry and
a qualify=yes. I don't think the qualify=yes is required, but it helps
in debuging.

About the port, I'm not too sure about sipura and snom phones (I only
have Cisco phones :(). That could have something to do with it..

On 1/26/06, Nora Lavelle <[EMAIL PROTECTED]> wrote:
>
> Hi there gary. thanks so much for your help. we're using sipura-841 and snom 
> 320s.
>
> Here's the sip show peers.. that's weird that extension 130 has port 2057.. 
> could that be the problem ?
>
> -nora
>
> Name/usernameHostDyn Nat ACL Mask Port Status
>
> 201/201  10.200.0.56  D  255.255.255.255  5060 
> Unmonitor
> ed
> 130/130  10.200.0.10  D  255.255.255.255  2057 
> Unmonitor
> ed
> 129/129  10.200.0.5   D  255.255.255.255  5060 
> Unmonitor
> ed
> 127/127  10.201.0.30  D  255.255.255.255  5060 
> Unmonitor
> ed
> 126/126  10.201.0.29  D  255.255.255.255  5060 
> Unmonitor
> ed
> 125/125  10.201.0.35  D  255.255.255.255  5060 
> Unmonitor
> ed
> 124/124  10.201.0.31  D  255.255.255.255  5060 
> Unmonitor
> ed
> 102/102  10.200.0.48  D  255.255.255.255  5060 
> Unmonitor
> ed
>
> -Original Message-
> From: [EMAIL PROTECTED] on behalf of Gary Richardson
> Sent: Thu 1/26/2006 5:18 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] extension to extension dialing
>
> Check your error messages in you asterisk console. Perhaps your sip
> secret or caller id is broken?
>
> What type of phones are you using?
>
> On 1/26/06, Nora Lavelle <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> > Sorry for all the newbie questions. I really appreciate everyone's help
> > today.
> >
> >
> >
> > Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm
> > having an issue with SIP extension to extension calling. Any time I dial
> > another extension it goes right into voice mail.  My extensions.conf is
> > pretty small and rough but, here's what I have right now. Most of it was
> > taken from the voip-info website. Any help as always VERY appreciated.
> >
> >
> >
> > Thanks again!
> >
> > Nora Lavelle
> >
> >
> >
> > # cat extensions.conf
> >
> > [incoming]
> >
> > exten => s,1,Answer();
> >
> > exten => s,2,Background(ssn-greeting);
> >
> > exten => *,1,Directory(default)
> >
> > exten => 205,1,Wait(2)
> >
> > exten => 205,2,Record(/tmp/asterisk-recording:gsm)
> >
> > exten => 205,3,Wait(2)
> >
> > exten => 205,4,Playback(/tmp/asterisk-recording)
> >
> > exten => 205,5,Wait(2)
> >
> > exten => 205,6,Hangup
> >
> >
> >
> > [internal]
> >
> > exten => 101,1,Macro(stdexten,SIP/101)
> >
> > exten => 102,1,Macro(stdexten,SIP/102)
> >
> > exten => 103,1,Macro(stdexten,SIP/103)
> >
> > exten => 123,1,Macro(stdexten,SIP/123)
> >
> > exten => 124,1,Macro(stdexten,SIP/124)
> >
> > exten => 125,1,Macro(stdexten,SIP/125)
> >
> > exten => 126,1,Macro(stdexten,SIP/126)
> >
> > exten => 127,1,Macro(stdexten,SIP/127)
> >
> > exten => 128,1,Macro(stdexten,SIP/128)
> >
> > exten => 129,1,Macro(stdexten,SIP/129)
> >
> > exten => 130,1,Macro(stdexten,SIP/130)
> >
> > exten => 135,1,Macro(stdexten,SIP/135)
> >
> > exten => 117,1,Macro(stdexten,SIP/117)
> >
> > exten => 201,1,Macro(stdexten,SIP/201)
> >
> >
> >
> > [voicemail]
> >
> > exten => 300,1,Answer
> >
> > exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
> >
> > exten => 300,3,Hangup
> >
> >
> >
> > [local]
> >
> > exten => _9NXX,1,Dial(Zap/g1/${EXTEN:1})
> >
> > exten => _9NXX,2,Congestion
> >
> >
> >
> > [longdistance]
> >
> > exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1})
> >
> > exten => _91NXXNXX,2,Congestion
> >
> >
> >
> > [macro-stdexten]
> >
> > exten => s,1,Dial(${ARG1},20)
> >
> > exten => s,2,Goto(s-${DIALSTATUS},1)
> >
> > 

Re: [Asterisk-Users] extension to extension dialing

[Gary Richardson]

What sort of phones are you using?

What does "sip show peers" show? Are you phones properly registered?

On 1/26/06, Nora Lavelle <[EMAIL PROTECTED]> wrote:
> Here's what I get in the the log this is when extension 130 dials
> extension 129. Thanks again !
>
> nora
>
> -- Executing Macro("SIP/130-a644", "stdexten|SIP/129") in new stack
> -- Executing Dial("SIP/130-a644", "SIP/129|20") in new stack
> -- Called 129
> Jan 26 17:20:48 WARNING[28243]: chan_sip.c:694 retrans_pkt: Maximum
> retries exceeded on call [EMAIL PROTECTED]
> for seqno 102 (Critical Request)
>   == No one is available to answer at this time
> -- Executing Goto("SIP/130-a644", "s-NOANSWER|1") in new stack
> -- Goto (macro-stdexten,s-NOANSWER,1)
> -- Executing VoiceMail("SIP/130-a644", "u129") in new stack
> -- Playing 'voicemail/default/129/unavail' (language 'en')
>   == Spawn extension (macro-stdexten, s-NOANSWER, 1) exited non-zero on
> 'SIP/130-a644' in macro 'stdexten'
>   == Spawn extension (default, 129, 1) exited non-zero on 'SIP/130-a644'
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Gary
> Richardson
> Sent: Thursday, January 26, 2006 5:19 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] extension to extension dialing
>
> Check your error messages in you asterisk console. Perhaps your sip
> secret or caller id is broken?
>
> What type of phones are you using?
>
> On 1/26/06, Nora Lavelle <[EMAIL PROTECTED]> wrote:
> >
> >
> >
> > Sorry for all the newbie questions. I really appreciate everyone's
> help
> > today.
> >
> >
> >
> > Okay I've got outgoing and incoming calls working with no echo. yay!
> Now I'm
> > having an issue with SIP extension to extension calling. Any time I
> dial
> > another extension it goes right into voice mail.  My extensions.conf
> is
> > pretty small and rough but, here's what I have right now. Most of it
> was
> > taken from the voip-info website. Any help as always VERY appreciated.
> >
> >
> >
> > Thanks again!
> >
> > Nora Lavelle
> >
> >
> >
> > # cat extensions.conf
> >
> > [incoming]
> >
> > exten => s,1,Answer();
> >
> > exten => s,2,Background(ssn-greeting);
> >
> > exten => *,1,Directory(default)
> >
> > exten => 205,1,Wait(2)
> >
> > exten => 205,2,Record(/tmp/asterisk-recording:gsm)
> >
> > exten => 205,3,Wait(2)
> >
> > exten => 205,4,Playback(/tmp/asterisk-recording)
> >
> > exten => 205,5,Wait(2)
> >
> > exten => 205,6,Hangup
> >
> >
> >
> > [internal]
> >
> > exten => 101,1,Macro(stdexten,SIP/101)
> >
> > exten => 102,1,Macro(stdexten,SIP/102)
> >
> > exten => 103,1,Macro(stdexten,SIP/103)
> >
> > exten => 123,1,Macro(stdexten,SIP/123)
> >
> > exten => 124,1,Macro(stdexten,SIP/124)
> >
> > exten => 125,1,Macro(stdexten,SIP/125)
> >
> > exten => 126,1,Macro(stdexten,SIP/126)
> >
> > exten => 127,1,Macro(stdexten,SIP/127)
> >
> > exten => 128,1,Macro(stdexten,SIP/128)
> >
> > exten => 129,1,Macro(stdexten,SIP/129)
> >
> > exten => 130,1,Macro(stdexten,SIP/130)
> >
> > exten => 135,1,Macro(stdexten,SIP/135)
> >
> > exten => 117,1,Macro(stdexten,SIP/117)
> >
> > exten => 201,1,Macro(stdexten,SIP/201)
> >
> >
> >
> > [voicemail]
> >
> > exten => 300,1,Answer
> >
> > exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
> >
> > exten => 300,3,Hangup
> >
> >
> >
> > [local]
> >
> > exten => _9NXX,1,Dial(Zap/g1/${EXTEN:1})
> >
> > exten => _9NXX,2,Congestion
> >
> >
> >
> > [longdistance]
> >
> > exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1})
> >
> > exten => _91NXXNXX,2,Congestion
> >
> >
> >
> > [macro-stdexten]
> >
> > exten => s,1,Dial(${ARG1},20)
> >
> > exten => s,2,Goto(s-${DIALSTATUS},1)
> >
> > exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
> >
> > exten => s-NOANSWER,2,Goto(default,s,1)
> >
> > exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
> >
> > exten => 

Re: [Asterisk-Users] extension to extension dialing

[Gary Richardson]

Check your error messages in you asterisk console. Perhaps your sip
secret or caller id is broken?

What type of phones are you using?

On 1/26/06, Nora Lavelle <[EMAIL PROTECTED]> wrote:
>
>
>
> Sorry for all the newbie questions. I really appreciate everyone's help
> today.
>
>
>
> Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm
> having an issue with SIP extension to extension calling. Any time I dial
> another extension it goes right into voice mail.  My extensions.conf is
> pretty small and rough but, here's what I have right now. Most of it was
> taken from the voip-info website. Any help as always VERY appreciated.
>
>
>
> Thanks again!
>
> Nora Lavelle
>
>
>
> # cat extensions.conf
>
> [incoming]
>
> exten => s,1,Answer();
>
> exten => s,2,Background(ssn-greeting);
>
> exten => *,1,Directory(default)
>
> exten => 205,1,Wait(2)
>
> exten => 205,2,Record(/tmp/asterisk-recording:gsm)
>
> exten => 205,3,Wait(2)
>
> exten => 205,4,Playback(/tmp/asterisk-recording)
>
> exten => 205,5,Wait(2)
>
> exten => 205,6,Hangup
>
>
>
> [internal]
>
> exten => 101,1,Macro(stdexten,SIP/101)
>
> exten => 102,1,Macro(stdexten,SIP/102)
>
> exten => 103,1,Macro(stdexten,SIP/103)
>
> exten => 123,1,Macro(stdexten,SIP/123)
>
> exten => 124,1,Macro(stdexten,SIP/124)
>
> exten => 125,1,Macro(stdexten,SIP/125)
>
> exten => 126,1,Macro(stdexten,SIP/126)
>
> exten => 127,1,Macro(stdexten,SIP/127)
>
> exten => 128,1,Macro(stdexten,SIP/128)
>
> exten => 129,1,Macro(stdexten,SIP/129)
>
> exten => 130,1,Macro(stdexten,SIP/130)
>
> exten => 135,1,Macro(stdexten,SIP/135)
>
> exten => 117,1,Macro(stdexten,SIP/117)
>
> exten => 201,1,Macro(stdexten,SIP/201)
>
>
>
> [voicemail]
>
> exten => 300,1,Answer
>
> exten => 300,2,VoicemailMain(ssn-voicemail-greeting)
>
> exten => 300,3,Hangup
>
>
>
> [local]
>
> exten => _9NXX,1,Dial(Zap/g1/${EXTEN:1})
>
> exten => _9NXX,2,Congestion
>
>
>
> [longdistance]
>
> exten => _91NXXNXX,1,Dial(Zap/g1/${EXTEN:1})
>
> exten => _91NXXNXX,2,Congestion
>
>
>
> [macro-stdexten]
>
> exten => s,1,Dial(${ARG1},20)
>
> exten => s,2,Goto(s-${DIALSTATUS},1)
>
> exten => s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
>
> exten => s-NOANSWER,2,Goto(default,s,1)
>
> exten => s-BUSY,1,Voicemail(b${MACRO_EXTEN})
>
> exten => s-BUSY,2,Goto(default,s,1)
>
> exten => s-.,1,Goto(s-NOANSWER,1)
>
> exten => a,1,VoicemailMain(${MACRO_EXTEN})
>
>
>
> [default]
>
> include => incoming
>
> include => internal
>
> include => voicemail
>
> include => local
>
> include => longdistance
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Re: [Asterisk-Users] Asterisk Setup Question -- Please Help

[Gary Richardson]

I've had success using oh323 to create a trunk between a CCM3.2 and
asterisk.. I wouldn't put any analog devices in your asterisk box if
you already have CM.

You could also move completely to Asterisk and flash your 79xx's into
SIP mode and turn your Cisco boxes off.

BTW, you could also have asterisk just do something like:

exten => _9XX,1,Dial(OH323/[EMAIL PROTECTED])

So you don't have to do fancy 78 stuff to get an outside line. I'm
also playing around with something like:

exten => 1234,1,Dial(SIP/SomeUser)
exten => 1235,1,Dial(SIP/SomeOtherUser)
; we didn't find any users here .. let's try Cisco
exten => _,1,Dial(OH323/[EMAIL PROTECTED])

On this Cisco side, when I move people to Asterisk, I'll just hard
code a dial plan to the h323 gate on my asterisk box. Then, I can
flash hardphone users one at a time and get them using asterisk as
their call server at my leisure (or not move people at all).

It's mostly working now, I'm just having problems with including
contexts and precedence -- I haven't quite figured it all out yet. I
hope that's a source of inspiration :)

On 1/25/06, Goran Donev <[EMAIL PROTECTED]> wrote:
> I have a question on Asterisk and whether it will work with the following
> design.
>
>
> Install ASTERISK on the external side of the Network. Purchase an AudioCodes
> 4/8 port Analog Fx0 gateway. So far everything seems straight forward. Here
> is the twist.
>
> The company currently has Cisco Call Manager 3.3 which does not support SIP
> Trunking. But it does have a VG248. I would like to place 4 lines through
> the Cisco Call Manager.
>
> I want to set up a dial plan where 7 would grab the fx0 line for internal
> and the users would be able to place internal calls through the Cisco Call
> Manager. I envision people dialing 7 (4 digit extension.) This would
> call internally.
>
> I then envision setting up a calling plan where 7 would grab the trunk and 8
> would grab an outside line from the Cisco Call Manager and then dial the 10
> digit telephone number.
>
> 78xx. This would allow them to place external calls through the call
> manager. Is this something that would be feasible?
> Since the company is not looking to invest a lot in upgrading the Cisco yet
> they want to allow external sales reps to work from home.
>
> Would there be a way through Asterisk where I can then program the FX0
> extension coming in from the Cisco Call Manager to ring into the Audiocodes
> and be dialed directly to an extension in the Asterisk server?
>
> Example - 1300---200 on the Asterisk.
> This would allow people calling the company to directly dial their sales
> people and be forwarded to the extension attached to the audiocodes.
> If this is feasible please let me know as I would like to propose this
> solution to the company.
>
> Thanks.
>
>
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Re: [Asterisk-Users] Hardware recommendations

[Gary Richardson]

>From my understanding this is more like a 'Key' system than a 'PBX'.

You can make all you phones ring when a certain number is dialed. The
first one to pick up gets the call. I can't think of exactly what this
functionality is called, but I believe there are menus for it in
[EMAIL PROTECTED] Perhaps it's call groups?

You need to think of asterisk as a multiplexor -- you have x number of
lines coming in from the PSTN and y number of phones. Not all phones
are active at one time and it is completely indescriminate when it
comes to the next available line. It doesn't matter which line gets
picked up when you dial 9, just that you get an outside line.

You should be able to get your telco to assign the same phone number
on mutliple lines and it will ring through to the next available line
(similar to how a T1 works).

On 1/24/06, Dane Reugger <[EMAIL PROTECTED]> wrote:
> Maybe I am getting this wrong - every phone I look at says it handles a
> given number of lines.
>
> I don't want to spend the extra for 4 appearances when all I need is 2.
> Where I must be missing something  is:
>
> Imagine w/ have 2 appearances phones - no operator - the phones just ring.
>
> Lets say a call comes in and its for Joe, Joe picks up
> another call comes in, this time for Fred - he picks up
> now a call comes for me - wouldn't their above calls occupy all of our
> appearances?
>
> If not I would think we would need some type of operator forwarding the
> call to the phones instead of just having them ring.
>
> Sorry, I'm not getting it - maybe I'm just too old fashioned. I'm trying
> to do this as simply and economically as possibly w/o sacrificing quality.
>
> Your help is GREATLY appreciated.
>
> -Dane
>
>
>
>
> Kerry Garrison wrote:
> > You need to separate lines from call appearances. Asterisk has lines (actual
> > phone lines) and phones have call appearances (number of simultaneous calls
> > the phone can handle). You could have 1000 lines going into your Asterisk
> > box but the typical user doesn't need more than 2 - 4 simultaneous calls.
> > On the flip side, you could have 4 "lines" coming into your asterisk server
> > and have 100 phones with 4 call appearances each. By using Asterisk to
> > manage the lines, you don't need 400 phone lines to support 100 phones w/4
> > call appearances each.
> >
> > Kerry Garrison
> > Publisher - http://GeekGazette.com - http://VOIPSpeak.net
> > (949) 502-7819 x200 - [EMAIL PROTECTED]
> > http://www.techdatapros.com
> >
> >
> >
> >> -Original Message-
> >> From: [EMAIL PROTECTED]
> >> [mailto:[EMAIL PROTECTED] On Behalf Of
> >> Dane Reugger
> >> Sent: Tuesday, January 24, 2006 9:09 AM
> >> To: Asterisk Users Mailing List - Non-Commercial Discussion
> >> Subject: Re: [Asterisk-Users] Hardware recommendations
> >>
> >> If you have 16 call appearances or lines - how do you get to
> >> line 16 - type in some code?
> >>
> >>
> >> Adam Goryachev wrote:
> >>
> >>> On Mon, 2006-01-23 at 23:00 -0700, Douglas Garstang wrote:
> >>>
> >>>
>  Polycom SoundPoint 601 has 4 'lines'. :)
> 
> 
> >>> Actually, it has 6 'lines' :)
> >>>
> >>>
> >>>
> Needing a 4 line phone is going to decrease your
> 
> >> choices of phones.
> >>
> 
> Why do you need 4 lines?
> 
> 
> >>> He probably hasn't worked out the difference between 'call
> >>>
> >> appearances'
> >>
> >>> and lines yet Even a polycom 301 (with 2 'lines' can
> >>>
> >> handle loads
> >>
> >>> of calls, I think the limit is something like 16 per line,
> >>> configurable in the xml file).
> >>>
> >>> Regards,
> >>> Adam
> >>>
> >>> ___
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> >>>
> >>>
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> >>
> >>
> >
> >
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Re: [Asterisk-Users] Asterisk SIP phones to Cisco Unity via CCM 4.0 SIP Trunk

[Gary Richardson]

You can run a SIP image on a 7940. [EMAIL PROTECTED] has pretty good
support for it. Check the voip-info.org wiki for instructions on
switching the firmware.

Hopefully that will take a step out of the plan -- you could
completely ditch your Cisco system :)

On 1/23/06, sys read <[EMAIL PROTECTED]> wrote:
>
> Hi,
>
> I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and
> about 45 SCCP phones on the ccm, and 200 users on unity.   we want to
> migrate all users to IP Phones to ditch our ancient phone system.   I would
> love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet
> and run sip to an asterisk server, but have their voicemail on Unity.
>
> these phones are $150 each, the alternative is cisco 7940s ( around $250 )
> running SCCP through the CCM.  at the quantities I'm talking about, $100 is
> significant.
>
> Does anyone have any idea how to get this done?
>
> I've tried this:
>
> exten => 123,1,Dial(SIP/sipphone,20)
> exten => 123,2,Dial(SIP/ccm/3040)
>
> where 3040 is our VM pilot for ccm.  but all it does is take us to the main
> greeting.
>
> we have smartnet, but they haven't been helpful at all
>
> I called digium to see if they could help if we paid, but they said they've
> never heard of cisco unity
>
> help?
>
> thanks.
>
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Re: [Asterisk-Users] Asterisk and Linux-HA

[Gary Richardson]

Could you possibly use the redhat init scripts instead? Or at least
duplicate the functionality under Debian.

(I'm not too familiar with Debian, so I don't know how it does such things).

On 1/19/06, Tron <[EMAIL PROTECTED]> wrote:
>
> Yes, I know, but I have I was think that heartbeat use status function in
> init asterisk script to check if asterisk is alive, but status function is
> for redhat.Are there any similar function in Debian?. And in respect of
> slave, when slave get all resources and master wakeup, maste request for all
> resources, slave give it all resources, but asterisk continues alive in
> slave. My questions are:
>
> What I need to say to heartbeat that asterisk is alive or dead and why when
> slave give all resources to master doesn't goes down itself asterisk
> service. My init script is on /etc/hearbeat/resource.d and in /etc/init.d/.
>
>
> Any idea?
>
>
> regards,
>
> tron
>
> -Mensaje original-
> De: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] En nombre de Gary
> Richardson
> Enviado el: jueves, 19 de enero de 2006 16:28
> Para: Asterisk Users Mailing List - Non-Commercial Discussion
> Asunto: Re: [Asterisk-Users] Asterisk and Linux-HA
>
> Doesn't heartbeat take care of this? It's been awhile since I've configured
> it. If two servers join back together as master, one of them shuts down its
> services. Maybe I'm just wishfully thinking..
>
> There's also a directive to determine if a secondary should fail back over
> to the master if it comes back up.
>
> All your 'shared' services should be controlled by heartbeat -- you
> shouldn't have to do anything except use the supplied init scripts.
>
> On 1/19/06, Tron <[EMAIL PROTECTED]> wrote:
> >
> > Hi Srs.,
> >
> > we have installing two machines with Asterisk and Linux-HA. I just
> > copy conf files and voicemail files and more with rsync, and now I
> > want to test with Linux-HA if asterisk is up. I'm installing Asterisk
> > over Debian, but I haven't a status function in asterisk script.
> >
> > Any one help me to know how can I check if asterisk is up? If I switch
> > off master machine or I cut network cable, second machine goes up OK,
> > but if I switch on or replug cable in Main machine, all works fine but
> > I realize that slave machine doesn't down asterisk.
> >
> > Any one has installed this system?
> >
> >
> > regards,
> >
> > tron
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> >
> >
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Re: [Asterisk-Users] Asterisk and Linux-HA

[Gary Richardson]

Doesn't heartbeat take care of this? It's been awhile since I've
configured it. If two servers join back together as master, one of
them shuts down its services. Maybe I'm just wishfully thinking..

There's also a directive to determine if a secondary should fail back
over to the master if it comes back up.

All your 'shared' services should be controlled by heartbeat -- you
shouldn't have to do anything except use the supplied init scripts.

On 1/19/06, Tron <[EMAIL PROTECTED]> wrote:
>
> Hi Srs.,
>
> we have installing two machines with Asterisk and Linux-HA. I just copy
> conf files and voicemail files and more with rsync, and now I want to test
> with Linux-HA if asterisk is up. I'm installing Asterisk over Debian, but I
> haven't a status function in asterisk script.
>
> Any one help me to know how can I check if asterisk is up? If I switch off
> master machine or I cut network cable, second machine goes up OK, but if I
> switch on or replug cable in Main machine, all works fine but I realize that
> slave machine doesn't down asterisk.
>
> Any one has installed this system?
>
>
> regards,
>
> tron
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Re: [Asterisk-Users] Failover Device?

[Gary Richardson]

Is there any documentation around for running Asterisk in a Cluster (I
assume you mean a n+1 cluster as you list a failover cluster as a
different option). I was under the impression that it can't be done..

Thanks.

On 1/11/06, Carlos Alperin <[EMAIL PROTECTED]> wrote:
> Do you need failover on wich side? PRI or Asterisk? Both?
>
> Straight to the last option:
>
> PRI: the best if you have more than one PRI is to do hunt on the provider
> side, so when one is full or down, all calls are going to be directed to the
> second one.
>
> Asterisk: Do redundancy, so you need to have a second Asterisk box ready for
> failover, taken all the traffic of the first one in such case. You can do
> Hearthbeat, or DNS handling for this. I never try to run asterisk in a
> Cluster, that can be a third option.
>
> Any experience on that direction???
>
> Regards,
>
> Carlos
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of Matt
> Sent: Wednesday, January 11, 2006 8:43 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] Failover Device?
>
> First,
> Something seems to be wrong with the list.  I'm not the only person
> who has expressed seeing their messages either arrive late, or not at
> all.
>
> With that out of the way..
>
> Is anyone aware of any type of failover device for PRI on asterisk?
> I've found the ISDNGuard, however it is currently not made in the
> U.S., nor does it run on U.S. power.
>
> Is anyone aware of a device that will detect (heartbeat?) if Asterisk
> is running, and if not, failover to a backup server?
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