[asterisk-users] What is needed to test issue 0013573: [Patch] Allow realtime_multi_ldap to behave like other realtime_multi functions

[Sean Brady]

Not sure if this is the right place to ask, but what do we need to do to 
get this patch merged?  How can I help?  I'm no dev, but I use LDAP with 
Asterisk and I might be of some help.

Thanks guys.

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Re: [asterisk-users] Asterisk choking on voice messages announcements

[Sean Brady]

On 04/21/2010 05:36 PM, bruce bruce wrote:

Here are result of dahdi_test:

[r...@ip-10-251-123-3 ~]# dahdi_test
Opened pseudo dahdi interface, measuring accuracy...
99.725% 96.018% 99.532% 91.934% 99.923% 99.923% 99.628% 99.434%
-434.763% 99.239% 93.770% 99.141% 99.822% 91.232% 99.727% 93.770%
99.726% -403.227% 98.069% 98.458% 95.136% 98.749% 91.229% 87.622%
98.554% 93.282% -407.620% 94.650% 96.308% 98.750% 96.993% 93.478%
94.063% 93.381% 61.745% -379.400% 99.628% 99.921% 99.142% 96.797%
98.457% 99.337% 87.909% 95.141% -396.880% 99.531% 99.923% 99.921%
91.035% 96.408% 91.916% 90.255% -402.153% 81.079% 74.534% 96.212%

What can one tell from these?

On Wed, Apr 21, 2010 at 6:59 PM, bruce bruce > wrote:


Thanks for the input.
I am going to check this once I get access to system again tonight.
But I thought the timing source dahdi_dummy is only good for
features like MeetMe or conference rooms? or am I wrong and it has
an effect on any type of calls and checking voice messages?
Thanks

On Wed, Apr 21, 2010 at 2:49 PM, Ryan Bullock mailto:rrb3...@gmail.com>> wrote:

So I be it sounds like all the recordings are underwater.

Are you using dahdi for timing? Can you run dahdi_test?

Asterisk needs a good timing source, in the case when you
don't have a physical card providing it, it relies on kernel
ticks or the RTC (or HPET). Because of the nature of virtual
machines they don't always get access to the processor when
they want and therefore their timing can get skewed and can be
bad for real-time applications.

There are some patches/work-arounds that you can do. You might
want to google 'asterisk in a virtual machine' or 'asterisk
timing virutal machine', or anything along those lines.

I think I remember in some of the recent dahdi or asterisk
release notes that they changed some settings to be more
virtual machine friendly. So maybe make sure you are running
the latest versions?

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What in the world?  Bruce, that is a measure of accuracy of your timing 
source.  I believe that is the issue.  What is this running on?
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Re: [asterisk-users] Odd Issue With Polycom Phones

[Sean Brady]

On 04/21/2010 03:08 PM, Warren Selby wrote:
On Wed, Apr 21, 2010 at 3:46 PM, Jay Vocaire > wrote:


Thanks for the tip, I did just that, and now I am more confused.

It does appear as though there is just one call ID (if my
assumption that the "tag=" determines the call.

The first time it sends like this:



Asterisk responds with a SIP/2.0 401 Unauthorized, the phone then
comes back with this:



The difference is that the CSeq is now 2 and the following line is
added:

Authorization: Digest username="3271", realm="asterisk",
nonce="393a1b1f", uri="sip:3...@y.y.y.y;user=phone",
response="c8223e261c252c12172982ee661ad307", algorithm=MD5


So maybe I do have an issue in Asterisk, okay probably.  Any clues
as to how to debug?  Let me know if need to post more information.


This is expected behavior for SIP communications.  I see this all the 
time when an end point is registering with Asterisk.  I think in those 
cases, however, it's a REGISTER request, not an INVITE.  How is your 
sip.conf configured for these end points?


Do you have any phones other than the ones experiencing this problem 
that you can test with?




Yes this is expected behavior on a REGISTER.  I didn't think that it was 
correct on an INVITE, however on reading RFC 3261, I believe that 
Asterisk is correctly responding to the request, needing credentials 
from the UA (Polycom).



My Ekiga softphone is doing the exact same thing, however it's not 
creating the same "2 call" issue that your Polycoms are having.  The 
Ekiga call setup is not including credentials on the first INVITE, 
receives a 401 not authorized, and sends another INVITE with 
credentials, and receives a "100 TRYING" from Asterisk.


This is most likely an issue with the firmware on the Polycom.  Bottom 
line is that another UA is doing the same thing, the call is setup 
properly, and it appears to work.


I respectfully request that someone smarter than me take a look at this 
and verify my conclusions, or correct me accordingly.


Thanks.

According to RFC 3261 (note that the RFC uses the word "request" instead 
of "register" or "registration request"):


"... If a 401 (Unauthorized) or 407 (Proxy Authentication Required)
response is received, the UAC SHOULD follow the authorization
procedures of Section 22.2 and Section 22.3 to retry the request with
credentials. ..."


Read more: http://www.faqs.org/rfcs/rfc3261.html#ixzz0llyASXyI

" ...

22.2 User-to-User Authentication

   When a UAS receives a request from a UAC, the UAS MAY authenticate
   the originator before the request is processed.  If no credentials
   (in the Authorization header field) are provided in the request, the
   UAS can challenge the originator to provide credentials by rejecting
   the request with a 401 (Unauthorized) status code.

   The WWW-Authenticate response-header field MUST be included in 401
   (Unauthorized) response messages.  The field value consists of at
   least one challenge that indicates the authentication scheme(s) and
   parameters applicable to the realm.

   An example of the WWW-Authenticate header field in a 401 challenge
   is:

  WWW-Authenticate: Digest
  realm="biloxi.com",
  qop="auth,auth-int",
  nonce="dcd98b7102dd2f0e8b11d0f600bfb0c093",
  opaque="5ccc069c403ebaf9f0171e9517f40e41"

   When the originating UAC receives the 401 (Unauthorized), it SHOULD,
   if it is able, re-originate the request with the proper credentials.
   The UAC may require input from the originating user before
   proceeding.  Once authentication credentials have been supplied
   (either directly by the user, or discovered in an internal keyring),
   UAs SHOULD cache the credentials for a given value of the To header
   field and "realm" and attempt to re-use these values on the next
   request for that destination.  UAs MAY cache credentials in any way
   they would like.

   If no credentials for a realm can be located, UACs MAY attempt to
   retry the request with a username of "anonymous" and no password (a
   password of "").

   Once credentials have been located, any UA that wishes to
   authenticate itself with a UAS or registrar -- usually, but not
   necessarily, after receiving a 401 (Unauthorized) response -- MAY do
   so by including an Authorization header field with the request.  The
   Authorization field value consists of credentials containing the
   authentication information of the UA for the realm of the resource
   being requested as well as parameters required in support of
   authentication and replay protection.

..."

Read more: http://www.faqs.org/rfcs/rfc3261.html#ixzz0llyY2M2W

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[asterisk-users] Dozens of SIP NOTIFY messages with unique call ID's, and the same mailbox repeated multiple times on 1.6.2.6

[Sean Brady]

(sorry this is so long)

I could really use a helping hand.  I have a 1.6.2.6 installation using 
LDAP as the realtime engine for voicemail users, SIP users, queues, and 
some custom hotdesking families.  I'm also using ODBC voicemail storage.

The issue that I am having is that the UA's (Polycom 501's, 301's and 
430's) are receiving multiple SIP NOTIFY messages when the user has a 
new voicemail, to the point where it will crash the UA.  I have also 
noticed that the same mailbox is repeated in the CLI output several 
times (see below).

CLI output on "sip show peer Desk004" as of right now:
Mailbox  : 
2...@gtf,2...@gtf,2...@gtf,2...@gtf,2...@gtf,2...@gtf,2...@gtf,2...@gtf,2...@gtf,2...@gtf

The actual LDAP attribute on the peer is: 2...@gtf

It appears- and I am trying to confirm- that the number of SIP NOTIFY 
messages is related to the number of extra mailbox entries.  I am using 
realtime cache on SIP peers, and I have qualify enabled.

When the peer first registers, there is only one mailbox entry, which 
matches the LDAP attribute.  I have noticed that if I check the mailbox 
using VoicemailMain, passing in the username and mailbox as arguments, 
the number of mailboxes in the Mailbox field on the "sip show peer 
Desk004" output increases from 1 to 3.  I just did a test where I 
restarted Asterisk, rebooted the phone, and it registered showing 1 
mailbox.  I checked the voicemail on the phone, then did a "sip show 
peer Desk004".  There are now 3 entries.  Every time that I check the 
voicemail, the number of entries increments by 2.

I also counted 28 retransmits of 102 NOTIFY messages with a MWI payload 
sent to the same peer.  It appears to send 28 NOTIFY messages, and 
retransmits each of them 4 times (I need to get a pcap of this, I've 
just been looking at the SIP debug on the peer).

I have tested this with several extensions, all exhibit the same 
behavior.  It appears that the UA is acting properly.  The same UA's 
with the same firmware and virtually the same configuration files work 
perfectly on 1.4.23.1 (the reg username and display name are different 
on 1.6.2.6).  I do not see this behavior on the same system with a peer 
specified in sip.conf.

I believe that this may be a bug with res_ldap in 1.6.2.6, however I 
don't think that I can rule out configuration issues until I pass it by 
the list first.  Is there something that I am doing wrong here, or is 
this a bug?


Thanks in advance for your help, it's greatly appreciated.  Feel free to 
contact off-list as well.

System:
RHEL 5.5 kernel 2.6.18-194.el5xen
Asterisk 1.6.2.6 built from source
DAHDI 2.3 built from source

LDAP packages managed by YUM:

mozldap.x86_64   6.0.5-1.el5
nss_ldap.i386   253-25.el5
nss_ldap.x86_64   253-25.el5
openldap.i386  2.3.43-12.el5
openldap.x86_64  2.3.43-12.el5
python-ldap.x86_64  2.2.0-2.1

LDAP Server:
389 Directory Server   1.2.5-1.el5



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Re: [asterisk-users] Odd Issue With Polycom Phones

[Sean Brady]

On 04/19/2010 02:22 PM, Jay Vocaire wrote:
> I have searched everywhere, but cannot seem to find anyone else talking about 
> this issue.  Maybe I am just using the wrong search terms.
>
> I am running Asterisk 1.6.2 and multiple Polycom phones all with 3.2.3 (the 
> latest) firmware on them.
>
> I am having an issue with my 550's and my 6000's (but oddly enough, not my 
> 320's).  Whenever a number is dialed on hook, and then the speakerphone 
> button is pressed, the number is dialed twice.  If the handset is picked up, 
> or the "Dial" softkey is pressed, the call is only sent once.  This leads me 
> to believe it is a phone issue, not a * config issue, but I have no way of 
> telling.
>
> I can verify that there are two call started via the snippet below:
>
>== Using SIP RTP CoS mark 5
>  -- Executing [3...@dlpn_ipausers:1] Macro("SIP/3271-0528", 
> "stdexten,3261,SIP/3261") in new stack
>  -- Executing [...@macro-stdexten:1] Set("SIP/3271-0528", 
> "__DYNAMIC_FEATURES=") in new stack
>  -- Executing [...@macro-stdexten:2] Set("SIP/3271-0528", 
> "ORIG_ARG1=3261") in new stack
>  -- Executing [...@macro-stdexten:3] GotoIf("SIP/3271-0528", "0?6:4") 
> in new stack
>  -- Goto (macro-stdexten,s,4)
>  -- Executing [...@macro-stdexten:4] Dial("SIP/3271-0528", 
> "SIP/3261,30,") in new stack
>== Using SIP RTP CoS mark 5
>  -- Called 3261
>== Spawn extension (macro-stdexten, s, 4) exited non-zero on 
> 'SIP/3271-0528' in macro 'stdexten'
>== Spawn extension (DLPN_IPAUsers, 3261, 1) exited non-zero on 
> 'SIP/3271-0528'
>== Using SIP RTP CoS mark 5
>  -- Executing [3...@dlpn_ipausers:1] Macro("SIP/3271-052a", 
> "stdexten,3261,SIP/3261") in new stack
>  -- Executing [...@macro-stdexten:1] Set("SIP/3271-052a", 
> "__DYNAMIC_FEATURES=") in new stack
>  -- Executing [...@macro-stdexten:2] Set("SIP/3271-052a", 
> "ORIG_ARG1=3261") in new stack
>  -- Executing [...@macro-stdexten:3] GotoIf("SIP/3271-052a", "0?6:4") 
> in new stack
>  -- Goto (macro-stdexten,s,4)
>  -- Executing [...@macro-stdexten:4] Dial("SIP/3271-052a", 
> "SIP/3261,30,") in new stack
>== Using SIP RTP CoS mark 5
>  -- Called 3261
>  -- SIP/3261-052b is ringing
>== Spawn extension (macro-stdexten, s, 4) exited non-zero on 
> 'SIP/3271-052a' in macro 'stdexten'
>== Spawn extension (DLPN_IPAUsers, 3261, 1) exited non-zero on 
> 'SIP/3271-052a'
>
> The first hangup was triggered right away (without me doing anything), the 
> second hangup was me actually hanging up the calling phone.
>
> It does the same thing if I dial an outside line.
>
> Any idea where to start trying to solve this?  Has anyone else seen it, and 
> can point me to the fix that I could not find with Google?
>
> Thanks.
>
>

I would recommend that you enable debugging on the peer only and check 
to see if you see two invites come from the phone.  Two invites with 
different call ID's would indicate it is indeed the phone making two 
calls.  One would indicate that it MAY be an Asterisk issue.

Are you using the latest Polycom firmware, btw?

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Re: [asterisk-users] On CLI SIP don't appear

[Sean Brady]

On 04/16/2010 10:10 AM, Renato bianchini wrote:

Hi,

Anyone know why sometimes on CLI disappear parameters as sip, stop,...?

Thank you very much by reply.

Renato






AFAIK when a CLI option is not available it means that module isn't 
loaded.  Check the logs to make sure that module was loaded properly.
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Re: [asterisk-users] Testing a sip call through Asterisk?

[Sean Brady]

On 04/16/2010 03:39 PM, Nathan Clemons wrote:
I'm looking to find a test tool that will register with our Asterisk 
(Trixbox) server here at work and place an outgoing call via our main 
SIP trunk (BroadVoice) to confirm that things are working. I've looked 
around but I can't seem to find any tools that will do what I'm 
looking for.


I can't just monitor the status of the trunk inside Asterisk, as this 
is the normal status:


asterisk*CLI> sip show peers
Name/username  HostDyn Nat ACL Port Status
BroadVoice/425256  147.135.32.221   N  5060 
Unmonitored

...
37 sip peers [Monitored: 5 online, 31 offline Unmonitored: 1 online, 0 
offline]

asterisk*CLI>

Alternatively, any suggestions as to how I can change the trunk 
configuration so that it is monitored would be appreciated. The peer 
config is set as:


allow=ulaw
disallow=all
canreinvite=no
context=from-trunk
dtmf=inband
dtmfmode=inband
fromdomain=sip.broadvoice.com 
fromuser=425256
host=sip.broadvoice.com 
insecure=very
nat=yes
secret=XX
type=peer
username=425256


Any assistance would be appreciated. I'd rather know when things fail 
via an automated system rather than learning it's down from the users.


-- Nathan Clemons


I believe that adding qualify= to your 
trunk configuration is what you are looking for for the monitoring 
state.  This will send SIP OPTIONS packets to the trunk periodically.  
See "qualify" in the sip.conf samples or documentation.


From there you can use a monitoring solution to monitor the state of 
the trunk.  Alternatively you can use a OSS tool called SIPp to test SIP 
devices.  See *http://sipp*.sourceforge.net for more information.  This 
is an indispensable tool for SIP and Asterisk troubleshooting.


I hope this helps.
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Re: [asterisk-users] Polycom DHCP Option 66 FTP provisioning

[Sean Brady]

Is anyone successfully using DHCP option 66 to specify an FTP [sic] 
provisioning of Polycom Sounpoint phones instead of TFTP?  I know option 66 
is typically used TFTP booting, but the Polycom doc doesn't appear to 
specify that option 66 implies TFTP instead of FTP (since you explicitly 
call out the protocol).  TFTP option 66 booting was working fine.

Does anyone know whether FTP provisioning of Polycom definitely requires a 
custom DHCP option like 160?

Usually in a situation like this I'd just creatively try different things in 
a divide-and-conquer approach to find something that works.  However in THIS 
case the phone tries to contact the boot server for SO LONG that the 
aforementioned 'brute-force' option would take me a decade.

Therefore I'm trolling for tips, which would be very mcuh appreciated!

Thanks!
-Karl
-

Use option 66.  This is from my dhcpd.conf:

option boot-server code 66 = string;

The later on in the file:

option boot-server "ftp://someone:stuffyoudontneedtok...@my.ip.ad.dy

Hope this helps.  My poly's are working fine.  

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Re: [asterisk-users] CDR: Add Dialed Number Identifierfield (DNID) field into MySQL

[Sean Brady]

> I have read 2 solutions

> (a) Changing the Dial plan and capturing DNID and inserting it into
> one of the existing column in CDR table.

> (b) Copy new CDR related .c & .h files which have added the
> functionality of recording DNID into MySQL.
> For this, CDR table structure needs to be changed and a new field has
> be created in CDR table.

> But I am still not very sure on how to go about doing this.
> Since I only have a production server, I do not have the options of
> experimenting.
> Can someone help with a step-by-step?

> Thx
> Sanjay




>> On Mon, Mar 15, 2010 at 3:08 PM, Lee Archer  
>> wrote:
>> Isn't the use of DNID separate to the userfield?  I'd like to have this
>> working also.
>>
>> Lee
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex
>> Balashov
>> Sent: 15 March 2010 08:34
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] CDR: Add Dialed Number Identifierfield
>> (DNID) field into MySQL
>>
>> Use the userfield.
>>
>> On 03/15/2010 04:25 AM, RSCL Mumbai wrote:
>>
>>> Hi,
>>>
>>> I would like to see the DNID in my MySQL CDR logs.
>>>
>>> I have read one big thread in the Asterisk Developer List, but I could
>>> not figure out how to implement it ?
>>> Is there a simple step-by-step.


If this is Asterisk 1.6.*, then you can use the adaptive ODBC, which is 
configured using /etc/asterisk/cdr_adaptive_odbc.conf.  If you compiled 
Asterisk with samples, you will find a sample file that has pretty much 
everything that you need.  From there, simply set the fieldname that you wish 
to write to the CDR, like this:

; Using Adaptive ODBC CDR's, sets the caller ID DNID to the CDR's custom field 
named "DNID"
Set(CDR(DNID)=${CALLERID(DNID)})

Personally, I like to set the DNID to a variable, just in case, when the 
inbound call first hits Asterisk from the trunk.  This probably isn't 
necessary, but I am always afraid that the CALLERID(DNID) value will change 
with a transfer or a channel redirect, which we use.  From there I write the 
variable to the CDR.

For more information on the adaptive concept, please see 
http://www.asterisk.org/node/48492.  There is also more detail from Tilghman 
Lesher here: 
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg210573.html

It's very elegant in it's design and it works like a champ- we use it in 
production.

If you are using Asterisk 1.4.*, you can use the the CDR's userfield. This is 
an optional, user defined field that can store just about whatever data you 
wish depending on the data type defined in the database.  You will have to 
google around to find out more information on how to enable it, although I 
believe that it's an option in the /etc/asterisk/cdr.conf configuration file 
that you are using.  

Again, if you are using Asterisk 1.6.* I would strongly recommend that you take 
advantage of the Adaptive CDR system.  
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[asterisk-users] Help with playing a recorded message in a conference.

[Sean Brady]

Hello all,

My folks would like to play a message to answering machines automatically after 
hanging up the phone.  So, when the caller dials the number of the callee, 
hears an answering machine, they would like to enter a code on the phone and 
hang up.  After the hangup the message plays to the callee and disconnects.  
The message that is played uses text to speech that is tailored to the callee, 
and there will be multiple callers playing a message to multiple callees 
simultaneously.

I was thinking about doing an features.conf application map, but I am not sure 
how to play the message.  One possibility is to have the dialplan create 
another channel, bridge the two channels together, then play the message, but 
I'm not sure of the best method of accomplishing this.

 Can someone help me with some ideas on this?  Does this sound like the correct 
approach, or is there a better or easier way to do this that anyone can think 
of?

Asterisk version is 1.6.2.6.  Thank you very much for your help in advance.

Thanks in advance.
Sean



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Re: [asterisk-users] Followme broken

[Sean Brady]

 
> > I am seeing the exact same behavior on 1.6.2.5.   Could this have
> > anything to do with issue #16816?  I'm no developer here, the reason
> > that I think it might be related is that both apps depend on the
> > second leg of the call to be answered, and it appears that for some
> > reason the apps don't think that the answer has occurred.  
> > 
> > Please let me know what you would like to see from debugging,
> > messages, etc. if you need more information.
> > 

> Please try the patch here as it worked for me :- 
> https://issues.asterisk.org/view.php?id=16929
>
> -- 
> Thanks, Phil

Worked for me on 1.6.2.5 as well.  Thanks Phil.  Will update the issue on 
Mantis as well.

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Re: [asterisk-users] Followme broken

[Sean Brady]

> - "--[ UxBoD ]--"  wrote:

> > - "--[ UxBoD ]--"  wrote:
> >  
> > > Hi,
> > > 
> > > we are running Asterisk 1.6.1.14 and have a issue that when we use
> > > followme the call is correctly placed to the mobile phone, the
> > mobile
> > > rings, but when answered we do not hear the normal followme
> > > introduction message.  If we press 1 to accept there is just
> > silence. 
> > > Has anybody else seen this issue before ?
> > > 
> > Ran a tcpdump against the session this morning and all SIP packets
> > look fine.  If I phone the mobile without using followme, so go
> > direct, all is okay.  Any ideas please ?
> >

> Ran a full trace against the issue and I see that SIP is established between 
> the PBX and the mobile phone, but at that  >point no RTP packets are sent.  
> We are using NAT but the firewall takes care of that and the session state.  
> I have >checked all the SIP packets and both the header and payload contains 
> our public IP.
>-- 
>Thanks, Phil



I am seeing the exact same behavior on 1.6.2.5.   Could this have anything to 
do with issue #16816?  I'm no developer here, the reason that I think it might 
be related is that both apps depend on the second leg of the call to be 
answered, and it appears that for some reason the apps don't think that the 
answer has occurred.  

Please let me know what you would like to see from debugging, messages, etc. if 
you need more information.

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Re: [asterisk-users] audio glitches in conference

[Sean Brady]

> I'm having a problem with conferences both meetme and app_conference, 
> though I've done most of the testing with meetme.


What version of DAHDI are you running?

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Re: [asterisk-users] Virtual machine timing (KVM)

[Sean Brady]

>> Sean Brady wrote:

To get MeetMe working properly, I know some sort of timing device

provided by the zaptel package is required (even if it means the

zt_dummy).  But, on a virtual machine I know that the Linux timing won't

work as expected.  Is it possible to then dedicate a physical device

like a USB port or something to the virtual machine to use for the

timing interrupts?





>>The 2.2.1 version of DAHDI using DAHDI dummy seems to be working adequately 
>>in a Xen environment on CentOS for me, although I haven't been using MeetMe.  
>>Have you run into issues with it specifically?  Which version of DAHDI are 
>>you using?  If there are some issues that you have found I would like to 
>>know...



>> Thanks,



>> Sean




>To be honest I haven't tried it with Asterisk version 1.4 or higher.  I only 
>tried it with 1.2 and when the DAHDI was called "Zaptel".  I have been a 
>little >afraid to upgrade to 1.6 from 1.2 just in case there are
>some incompatibilities in my config that'll bring down the phone system here 
>at the office for a while.

> The issue that I had was that the even the calls were choppy.  Not even 
> specifically just the MeetMe ones.  But that was on VirtualBox.  I am using 
> >KVM now.  I'm not sure if that matters.

> What is your timer frequency set to in the kernel btw?
With DAHDI dummy in 2.2.1 you don't have to even do that, AFAIK.  At least I 
didn't on my test box.
I do get choppy audio when playing recordings occasionally.  I haven't had time 
to figure that one out, but I haven't put it into production yet.
I have been told repeatedly that Asterisk shouldn't be virtualized, and that 
timing was an issue, however I have never been given a reason that I consider 
acceptable to preclude me from doing so.  I have also seen presentations 
talking about using Asterisk in Xen environments as well as Amazon's EC2 (also 
Xen).  So there is some real contradictions and FUD surrounding Asterisk 
virtualized.  Perhaps I am just stubborn, but I am determined to run Asterisk 
virtualized in production with conferencing (be it meetme or confbridge) until 
it's been proven without doubt that it just doesn't work.
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Re: [asterisk-users] Virtual machine timing (KVM)

[Sean Brady]

> To get MeetMe working properly, I know some sort of timing device
> provided by the zaptel package is required (even if it means the
> zt_dummy).  But, on a virtual machine I know that the Linux timing won't
> work as expected.  Is it possible to then dedicate a physical device
> like a USB port or something to the virtual machine to use for the
> timing interrupts?

The 2.2.1 version of DAHDI using DAHDI dummy seems to be working adequately in 
a Xen environment on CentOS for me, although I haven't been using MeetMe.  Have 
you run into issues with it specifically?  Which version of DAHDI are you 
using?  If there are some issues that you have found I would like to know...

Thanks,

Sean

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Re: [asterisk-users] 2 Asterisk Boxes, Single Voicemail

[Sean Brady]

>On 5 Feb 2010, at 16:55, Greg Blakely wrote:
>> If so, how?

>NFS or rsync?

>S

Use ODBC voice message storage and realtime voicemail configuration.

- Sean

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Re: [asterisk-users] CDR / billsec / originate / local chan

[Sean Brady]

>On 1.6.2 I have also tried using a local channel for the outbound leg 
>with the originate looking like the following:
>
>   action:.Originate..
>   actionid:.1306903_89#AJ_ORIGINATE_25
>   timeout:.4
>   exten:.s
>   async:.true
>   callerid:."".<612>
>   context:.campaignType_5
>   priority:.1
>   channel:.Local/61212142...@outboundsip/n
>
>And the Local context as follows;
>
>[outboundsip]
>exten = _XX.,1,Dial(SIP/trunk1/${EXTEN})
>exten = _XX.,n,Hangup
>
>exten = h,1,NoOp(Billsec is: ${CDR(billsec)})
>
>In this configuration, whilst the outbound call goes out and billsec 
>gets reported correctly in the h exten, the call does not get bridged 
>back into the campaignType_5 context so none of the call processing 
>occurs. I cannot see any options that can be passed to the dial command 
>that may affect the bridging of the call back into the campaignType_5 
>context???

Have you tried removing the /n option from the local channel?  Just a thought, 
but it's probably worth a try.  You could also try calculating the billsec in 
the dialplan and write it to the CDR with the adaptive CDR feature in 1.6.2.

Not sure if this is helpful but it was a thought.

-Sean



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Re: [asterisk-users] Error and call drops

[Sean Brady]

>Hi, does anyone have an info into what could cause

>[Nov 28 14:24:48] ERROR[11964] utils.c: write() returned error: Broken pipe

I have had the same issue with a PHP script that logs into the manager 
interface.  If you don't wait for the AMI response, then log off before closing 
the connection you will get those errors.  I also haven't seen any call drops.  
I would urge you to check your scripts, and put some 2 second waits before a 
logoff and closing the socket and see if that helps.
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Re: [asterisk-users] Asterisk & LDAP authentification

[Sean Brady]

>Hi everybody,
>
>I would like to use realtime authentification with my LDAP.



It depends on what you are doing with LDAP.  There is an LDAP realtime engine 
for SIP/IAX peers, voicemail users, asterisk configurations and extensions with 
a sample ldif included with the distro, although I recommend using the one from 
the trunk at this point.  The included ldif is for OpenLDAP.

These links can help you get started, but DO NOT use the schema listed within 
as is, as they are not up to date.
http://www.flyn.org/astldap/
http://magazine.redhat.com/2008/07/24/open-source-telephony-a-fedora-based-voip-server-with-asterisk/

I would pull the schema from here:

http://svnview.digium.com/svn/asterisk/branches/1.6.1/contrib/scripts/


I hope that this helps get you started.

Sean



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Re: [asterisk-users] Beginners Guide to setting up a Call Centre

[Sean Brady]

>Looking at all the docs I can find Asterisks looks like it should be
>able to do the job and a whole lot more.

>This is for a small call centre so ideally we want all the features of
>an average call centre, ACD, Call Recording, Queue's etc etc.

>Any pointers on how to get started would be most helpful.

>Peter.
---
(sorry this is so long)

Peter, 

I figured that I would chime in, as I run IT and am a managing partner of a 
small call center based on Asterisk and I think that my experience will be 
helpful (hate to beat a dead horse)...

Asterisk can definitely do what you need, so I am not going to talk about that 
any further.  I wouldn't waste my time with anything else.

I would strongly recommend either of the two following methods to get started, 
with the deciding factors being time and money.  There are lots of factors that 
will sway this argument, such as the complexity of your workflow, CTI needs, 
etc., but those time and money are the biggies.  You also have to carefully 
weigh your support requirements, uptime, and your desire to manage a phone 
system.  Asterisk doesn't have to take that much work once it's installed and 
tuned, but it will require some maintenance.  You will need to evaluate whether 
or not you want to take on that maintenance role or whether you want to pay to 
have it done for you.


Method 1: A professional installation by a Digium Certified Asterisk 
Professional. 

It will cost you some money, how much depends on your needs and how clearly you 
articulate them.  There are lots of great people out there that can help you 
get EXACTLY what you want and design a system that will grow with your 
business.  Call Digium for recommendations, or reply to this with your contact 
info and we can talk off list (I'm not trying to sell anything, but I have some 
people that I can recommend).  This can be a great option for a solid Asterisk 
system with good support and reliable operation with little maintenance.

There's a couple different approaches to this method- managed and developed 
with support.  Managed is where the team that developed the dialplan and 
asterisk environment for you manages the system for you as well for a recurring 
support fee.

Drawbacks to this method:

A. You will have to find a good vendor that will charge fairly and deliver on 
their SLA (always get an SLA with enforceable penalties).  This isn't that 
tough, but it's important.

B. The recurring support costs can eat into your budget quickly

C. This will take some time to develop properly, and for simple environments it 
may be overkill.

D. Adds/changes/ and deletes can be costly as well.  This can be mitigating by 
communicating the need to accommodate staff turnover with a user maintainable 
system.  


Method 2: Get a distro, install it, be dialing in about 8 hours or less (the 
route that I took when we started).  

This method is by far and away the easiest, cheapest, 
get-it-up-and-running-consequences-be-damned method.  You will take less time, 
effort  and money to get going like this than any other way I know of.  If your 
call flow is simple to moderately complex, this is the way to go in my opinion. 
 The FreePBX distros (Trixbox, AsteriskNOW [I think], Elastix, etc) all are 
very well put together, and will do everything that you listed in your original 
message and then some.  Of the distro's, I would probably either go with 
AsteriskNOW or, if you are up for a little more setup work, FreePBX on it's 
own.   


Drawbacks to this method:

A. I can't speak for others, but I found that the configuration engines have 
their limitations when it comes to call centers.  They simply weren't designed 
to do some of the specific things that we needed to do as we grew.  This 
doesn't mean that they wont do everything you need though, each case is unique. 
 They were fine for us in the beginning, but as our business grew so did our 
specific needs, and we outgrew these solutions.  There is nothing wrong with 
that if you understand from the outset that you may have needs that aren't met 
in the future.  These distros have to factor in the needs of their respective 
communities, and what may be good for one organization might not be good for 
others.

B. Troubleshooting issues can be more complex as you start to understand 
Asterisk and increase your level of sophistication.  I had a hard time 
troubleshooting FreePBX until I understood it's dialplan more, and it made 
troubleshooting complicated as I didn't fully understand the call flow through 
it's dialplan. The more you work with it, the easier it gets, but there can be 
a learning curve. 

C. Integration with other vendor's products can sometimes be a challenge if 
they don't already support your configuration GUI.

D. You cannot, no matter what anyone tells you (I know this to be cold, hard 
fact), modify the built in dialplan of FreePBX.  When you upgrade, or ev