[asterisk-users] Use of TONE_DETECT to detect dial tone in call (Asterisk18)

[Stephen Moran]

I am trying to use TONE_DETECT in a call that is made into a call centre
and placed on hold, when the call has been answered by the agent (denoted
by a dial tone of 2 seconds of dual tone 350Hz + 440Hz followed by 4
seconds of silence (North American dial tone), to act as a trigger t to
bridge a call to our agent. I am using

exten => s,n,Set(TONE_DETECT(0,,d(5)crg(tonedetect-dialagent,s,1))=)

which should look for dial tone on the rx channel with a decibel threshold
of 5 (tried default and didn't work), and then jump to the context
tonedetect-dialagent.

The documentation on the use of the options for TONE DETECT offers no
explanation on the syntax to use, so I have guessed the above.

Does anyone know if this is correct usage of the function? Also it is still
not detecting dial tone.

I noticed in Asterisk 20 there is a new function called WaitForTone() App
on its way

https://wiki.asterisk.org/wiki/display/AST/Asterisk+20+Application_WaitForTone

Anyone know anything about this?

Thanks

Steve
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Re: [asterisk-users] Failover / modifying response time

[Stephen More]

I was able to get a packet trace of this event

Time
312.353549 -> INVITE to primary
313.222303 -> INVITE to primary ( suspected resend of frame )
314.289215 -> INVITE to backup
315.397120 -> INVITE to backup ( suspected resend of frame )

So is primary just too slow to answer ? I am not seeing anything in the
logs on primary.



On Wed, Sep 3, 2014 at 2:39 PM, Stephen More  wrote:

> I have two real time asterisk boxes configured to accept incoming or
> outgoing calls at any time.
>
> All the users are configured to send their calls to primary. If primary is
> down calls will go to backup.
> All incoming SIP calls should be sent to primary. If primary is down
> incoming calls will go to backup.
>
> 99.9% of the time it works as designed.
>
> Every once in a while my SIP provider will send a sip call to our backup.
> primary is working fine. When asked they responded with "it's possible
> primary didn't respond fast enough so backup grabbed it. You should lower
> the response time on primary and raise it on backup"
>
> What setting is he referring to ? Can I tweak a setting to make our backup
> server respond slower so that the primary answers the call ?
>
>
> -Thanks
>
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[asterisk-users] Failover / modifying response time

[Stephen More]

I have two real time asterisk boxes configured to accept incoming or
outgoing calls at any time.

All the users are configured to send their calls to primary. If primary is
down calls will go to backup.
All incoming SIP calls should be sent to primary. If primary is down
incoming calls will go to backup.

99.9% of the time it works as designed.

Every once in a while my SIP provider will send a sip call to our backup.
primary is working fine. When asked they responded with "it's possible
primary didn't respond fast enough so backup grabbed it. You should lower
the response time on primary and raise it on backup"

What setting is he referring to ? Can I tweak a setting to make our backup
server respond slower so that the primary answers the call ?


-Thanks
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Re: [asterisk-users] authoritative sql definitions for Asterisk Realtime Architecture ARA

[Stephen More]

It appears that starting with version 12, there will be a methodical
approach to maintaining schemas:
https://wiki.asterisk.org/wiki/display/AST/Managing+Realtime+Databases+with+Alembic


On Fri, May 9, 2014 at 4:32 PM, Stephen More  wrote:

> I am trying to find where the authoritative sql definitions for Asterisk
> Realtime Architecture ARA are located. I have found many locations but each
> and everyone seems to be different.
>
>
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html
>
> http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example
>
> Files included with the distribution:
>
> asterisk-11.9.0/contrib/realtime/mysql/iaxfriends.sql:CREATE TABLE
> `iaxfriends` (
> asterisk-11.9.0/contrib/realtime/mysql/meetme.sql:CREATE TABLE meetme (
> asterisk-11.9.0/contrib/realtime/mysql/musiconhold.sql:CREATE TABLE
> musiconhold (
> asterisk-11.9.0/contrib/realtime/mysql/queue_log.sql:CREATE TABLE
> queue_log (
> asterisk-11.9.0/contrib/realtime/mysql/sippeers.sql:CREATE TABLE IF NOT
> EXISTS `sippeers` (
> asterisk-11.9.0/contrib/realtime/mysql/voicemail.sql:CREATE TABLE
> voicemail (
> asterisk-11.9.0/contrib/realtime/mysql/voicemail_data.sql:CREATE TABLE
> voicemail_data (
> asterisk-11.9.0/contrib/realtime/mysql/voicemail_messages.sql:CREATE TABLE
> voicemail_messages (
>
> asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
> extensions_conf (
> asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE cdr (
> asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
> sip_conf (
> asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
> voicemail_users (
> asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
> queue_table (
> asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
> queue_member_table
> asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
> "queue_log" (
>
> Any help is appreciated.
> -Stephen More
>
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[asterisk-users] authoritative sql definitions for Asterisk Realtime Architecture ARA

[Stephen More]

I am trying to find where the authoritative sql definitions for Asterisk
Realtime Architecture ARA are located. I have found many locations but each
and everyone seems to be different.

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html/asterisk-book.html

http://www.open-voip.org/index.php?title=Asterisk_Full_RealTime_example

Files included with the distribution:

asterisk-11.9.0/contrib/realtime/mysql/iaxfriends.sql:CREATE TABLE
`iaxfriends` (
asterisk-11.9.0/contrib/realtime/mysql/meetme.sql:CREATE TABLE meetme (
asterisk-11.9.0/contrib/realtime/mysql/musiconhold.sql:CREATE TABLE
musiconhold (
asterisk-11.9.0/contrib/realtime/mysql/queue_log.sql:CREATE TABLE queue_log
(
asterisk-11.9.0/contrib/realtime/mysql/sippeers.sql:CREATE TABLE IF NOT
EXISTS `sippeers` (
asterisk-11.9.0/contrib/realtime/mysql/voicemail.sql:CREATE TABLE voicemail
(
asterisk-11.9.0/contrib/realtime/mysql/voicemail_data.sql:CREATE TABLE
voicemail_data (
asterisk-11.9.0/contrib/realtime/mysql/voicemail_messages.sql:CREATE TABLE
voicemail_messages (

asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
extensions_conf (
asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE cdr (
asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
sip_conf (
asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
voicemail_users (
asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
queue_table (
asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
queue_member_table
asterisk-11.9.0/contrib/realtime/postgresql/realtime.sql:CREATE TABLE
"queue_log" (

Any help is appreciated.
-Stephen More
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[asterisk-users] gloCOM is Banned in China

[Stephen Wingfield]

China bans gloCOM

We heard this week from a prospect in China that our former gloCOM video is 
BANNED !
See if you can work out the offending item.
https://www.youtube.com/watch?v=naL7EuwNKaw



We have removed the offending item in the latest:
GLOCOM
https://www.youtube.com/watch?v=0kq9rCEmLmg

Still it comes with a 'free' version as always :)


Has anyone else been so treated by China's censorship ? 
Have any Chinese manufacturers been so treated in Europe or the US ?

Would be grateful to hear your views.

Steve
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Re: [asterisk-users] Integration with NEC DSX - help with dial line

[Stephen More]

My asterisk box can now call our NEC DSX !

Here is what worked for:
 elastix 2.4.0
 asterisk 11.6.0

1. Configure extension 421 on the NEC DSX with a password of 123456 (
very important to use a high numbered extension 4XX )
2. vi /etc/asterisk/sip_registrations_custom.conf

register=421:123456@192.168.1.57

[mysipprovider-out]
type=peer
secret=123456
username=421
host=192.168.1.57
fromuser=421
fromdomain=192.168.1.57
nat=yes
context=from-mysipprovider
; is further defined in extensions.conf

3. /usr/sbin/asterisk -r
Elastix*CLI> sip reload
Elastix*CLI> sip show registry
Hostdnsmgr Username   Refresh
StateReg.Time
192.168.1.57:5060   N  421 25
Registered   Fri, 15 Nov 2013 15:58:46
1 SIP registrations.

4. Add Extension 402 using elastix user interface
5. Edit Extension 402 using elastix user interface - set dial
SIP/402@mysipprovider-out,20

Now if you dial extension 402 on your asterisk box, it will connect
you to extension 402 on the NEC DSX.

Thank you John Novack for all your help.
-Stephen More


On Thu, Nov 14, 2013 at 11:43 AM, John Novack
 wrote:
>
> Stephen More wrote:
>>
>> I am trying to setup an extension in asterisk which dials an extension
>> on the NEC DSX. i.e. If an asterisk user dials 402 I want it to
>> connect to the NEC DSX @ 192.168.1.57 and connect to extension 402. (
>> 404 would be the NEC DSX sip account that I have the credentials for
>> ).
>>
>> [402]
>> deny=0.0.0.0/0.0.0.0
>> secret=pass1
>> dtmfmode=rfc2833
>> canreinvite=no
>> context=from-internal
>> host=dynamic
>> type=friend
>> nat=yes
>> port=5060
>> qualify=yes
>> callgroup=
>> pickupgroup=
>> dial=SIP/404:pass2@192.168.1.57/402
>> mailbox=402@device
>> permit=0.0.0.0/0.0.0.0
>> callerid=device <402>
>> callcounter=yes
>> faxdetect=no
>>
>>
>> [Nov 14 10:35:45] VERBOSE[13117][C-0010] app_dial.c: -- Called
>> SIP/404:pass2@192.168.1.57/402
>> [Nov 14 10:35:45] VERBOSE[11623][C-0010] chan_sip.c: -- Got
>> SIP response 480 "Temporarily not available" back from
>> 192.168.1.57:5060
>> [Nov 14 10:35:45] VERBOSE[13117][C-0010] app_dial.c: --
>> SIP/192.168.1.57-0019 is circuit-busy
>> [Nov 14 10:35:45] VERBOSE[13117][C-0010] app_dial.c:   == Everyone
>> is busy/congested at this time (1:0/1/0)
>>
>>
>> I tried IAX2, but then I just get ring no answer.
>>
>>
>> What am I missing in the config - I simply want a one to one mapping.
>>
>>
>> -Thanks
>>
> The NEC DSX does NOT support IAX2.
> Sorry to say that, but since IAX2 was never submitted to the rigors of
> becoming an accepted standard, there have been few adopters.
> Digium chose back in 2005, I believe, perhaps earlier, not to put the effort
> into whatever it took to get IAX2 accepted as an international standard.
>
> I have Asterisk send a page to my NEC DSX every 15 minutes during waking
> hours that announces the time, as well as some other information.
> First, make sure your DSX has the latest firmware. This is easily done
> through the system administrator
> Of course you will need the IP daughter board installed
> Note that I use a non standard port, as I have my system linked to another
> DSX system over the Internet, and in addition to firewall settings, we chose
> to move the sip control port to discourage hackers
> Obviously, if this is ONLY within your LAN, that isn't necessary.
> You will also need to set up an extension number in the 4XX range BEYOND the
> IP boards possible extensions, why we chose 421
> I do NOT use this to place a SIP call TO Asterisk from the DSX
> Hope this helps. If you need more information, feel free to contact me off
> list
> John Novack
>
> Here is my sip.conf:
> register=421:@172.16.0.235:6065;
> ;
> [421]
> allow=ulaw
> context=internal;
> type=friend
> username=421
> secret=;
> port=6065
> host=172.16.0.235 ; DSX IP address
> fromuser=421
> fromdomain=bigjohnnovack.ckts.info ;
> dtmfmode=inband
> qualify=yes
> accountcode=NECDSX
> maxexpirey=3600
> defaultexpirey=160
> srvlookup=no
> canreinvite=yes
>
> --
>
> Dog is my Co-pilot
>

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[asterisk-users] Integration with NEC DSX - help with dial line

[Stephen More]

I am trying to setup an extension in asterisk which dials an extension
on the NEC DSX. i.e. If an asterisk user dials 402 I want it to
connect to the NEC DSX @ 192.168.1.57 and connect to extension 402. (
404 would be the NEC DSX sip account that I have the credentials for
).

[402]
deny=0.0.0.0/0.0.0.0
secret=pass1
dtmfmode=rfc2833
canreinvite=no
context=from-internal
host=dynamic
type=friend
nat=yes
port=5060
qualify=yes
callgroup=
pickupgroup=
dial=SIP/404:pass2@192.168.1.57/402
mailbox=402@device
permit=0.0.0.0/0.0.0.0
callerid=device <402>
callcounter=yes
faxdetect=no


[Nov 14 10:35:45] VERBOSE[13117][C-0010] app_dial.c: -- Called
SIP/404:pass2@192.168.1.57/402
[Nov 14 10:35:45] VERBOSE[11623][C-0010] chan_sip.c: -- Got
SIP response 480 "Temporarily not available" back from
192.168.1.57:5060
[Nov 14 10:35:45] VERBOSE[13117][C-0010] app_dial.c: --
SIP/192.168.1.57-0019 is circuit-busy
[Nov 14 10:35:45] VERBOSE[13117][C-0010] app_dial.c:   == Everyone
is busy/congested at this time (1:0/1/0)


I tried IAX2, but then I just get ring no answer.


What am I missing in the config - I simply want a one to one mapping.


-Thanks

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Re: [asterisk-users] Serviced Office operator panel

[Stephen Wingfield]

Andrew

Seems you want something like this:

GLOCOM
http://www.youtube.com/watch?v=naL7EuwNKaw

We can deliver this with PBXware - the world's first Open Source Based Turnkey 
Telephony System.

Steve

steve 'At"  bicomsystems d0T  c*m



On Mar 12, 2013, at 4:46 PM, Michael Gilleran wrote:

> -Original Message-
> From: Andrew Yager [mailto:and...@rwts.com.au] 
> Sent: Monday, March 11, 2013 5:07 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Cc: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] Serviced Office operator panel
> 
> Hi,
> 
> It's a great console, should have included it in my list. 
> 
> Sadly doesn't meet SO requirements. :(
> 
> Andrew
> 
> On 12/03/2013, at 11:00 AM, Patrick Lists  
> wrote:
> 
>> On 03/12/2013 12:07 AM, Andrew Yager wrote:
>>> Hi,
>>> 
>>> I'm trying to find (with some desperation now) a decent web based or 
>>> application based UI that integrates with an Asterisk based PBX and 
>>> is designed for a Serviced Office environment.
>>> 
>>> Key features we're looking for:
>> 
>> Don't know if it covers your requirements but here's another 
>> commercial solution: http://www.getisymphony.com/
>> 
>> Regards,
>> Patrick
> 
> The one that we use is www.fop2.com
> 
> Not sure if it's exactly what you're looking for, but worth a look.
> 
> Mike
> 
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[asterisk-users] IAX2 over OpenVPN connection.... working but

[Stephen Brown]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
 
So a friend of mine and I setup a static key based point to point
OpenVPN connection from my box to his for the express intent of carrying
IAX traffic encrypted.

His network on his lan is 172.30.1.0/24 and mine is 10.0.30.0/24. His
PBX is located at 172.30.1.48 and mine is at 10.0.30.2. We had an
existing working IAX trunk in place prior to the VPN, and after we
brought the VPN up we set the host= parameter within Asterisk
accordingly on each end to match the local IP's and discovered it did
not work. The trunk remained in an UNKNOWN status on each end, even
though we could ping each box locally, SSH, and even SIP worked.

Here's where I am baffled and I am hoping someone with intricate
knowledge of this implementation may be able to explain it to me. What
we had to do to get this working was to set the host= parameter to the
respective endpoint IP's of the VPN tunnel, 172.10.1.1 in my case, and
172.10.1.2 in his case. Calls flow normally now and we cannot understand
how or why. I would have assumed with a destination of either LAN as
defined by the routing table it would have left out on the OpenVPN
connection by default, and what's even more strange is that IAX is the
only protocol that does not appear to function as intended.

Any takers? :)


-BEGIN PGP SIGNATURE-
Version: GnuPG v2.0.17 (MingW32)
 
iEYEARECAAYFAlDE7GcACgkQ3sJXNEncx7is9QCcCciMYFJ7ZXjYxuHC2EYD0PZY
waAAniNNx8GuC5To7ajlGR5sYs3yftFK
=lcWJ
-END PGP SIGNATURE-


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Re: [asterisk-users] Realtime Hints

[Stephen Collier]

I use the following in func_odbc.conf

[FINDEXTN]
dsn=asterisk
readsql=SELECT ${ARG1} FROM extension_map as em left join sip_devices as sd on s
d.id = em.name_id WHERE em.extension ='${ARG2}' and name_id IS NOT NULL

this is for our own extension_map table which is part of our mapping to our 
Avaya users. 

A simple one would be 
[FINDEXTN]
dsn=asterisk
readsql=SELECT ${ARG1} FROM sip_devices as sd WHERE sd.name ='${ARG2}'

This allows pulling any field from sip_devices which is our realtime sip table.

You could pull some of the other data you are looking for.

Cheers
Stephen


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Re: [asterisk-users] Realtime Hints

[Stephen Collier]

We use something like below

[blf]
exten =>_ZXX!,hint,SIP/${ODBC_FINDEXTN(sd.name,${EXTEN})}


This uses an odbc call to create the hint when the phone asks for it.
Using snom 760 and 821

Cheers
Stephen

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Re: [asterisk-users] iax2 trunks between asterisk servers

[Stephen Collier]

Doug,

Thanks, that answers my question I will reuse the macro I'm using with
an Avaya connection and CONNECTEDLINE(). Pity I was hoping iax2 would
transfer callee id.

Cheers
Stephen


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[asterisk-users] iax2 trunks between asterisk servers

[Stephen Collier]

Hi,

I am using iax2 trunks between asterisk servers and am having a callerid
problem. We are using realtime sip clients distributed between multiple
servers. Only in test now but have run into a calleeid problem - the
name of the called party is not displayed if the called party is on a
different server, it works if the called party is on the same server.

On each server sip clients show calleeid on calls but if the caller is
between servers calleeid doesn't work. Callerid is working fine though.

A call from a SIP client on asterisk2 to asterisk3. All phones are snom
760s.

Any ideas or suggestions appreciated.

iax.conf   (asterisk2 10.6.1)

[general]
bandwidth=high
allow=all
shrinkcallerid=no

[asterisk3]
type=friend
username=asterisk2
secret=secret
host=10.101.0.3
context=incoming
sendani=yes
trunk=yes

iax.conf   (asterisk3 11.0.0-beta1)

[general]
bandwidth=high
allow=all
shrinkcallerid=no

[asterisk2]
type=friend
username=asterisk3
secret=secret
host=10.101.0.2
context=incoming
sendani=yes
trunk=yes

Cheers
Stephen Collier

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Re: [asterisk-users] freepbx asterisk

[Stephen J Alexander]

On Fri, Jul 20, 2012 at 6:21 AM, Patrick Lists
 wrote:
> On 20-07-12 09:15, neo nortan wrote:
>>
>>
>> dear
>> i am a neewbi for asterisk, plz tell me or if any link is there where i
>> can understand how asterisk, freepbx, web-meetme, dahdi all these tools
>> works and how they are related.
>> plz help me.
>
>
> http://www.asteriskdocs.org/
>
> Regards,
> Patrick

You might get a bit of mileage out of http://www.voip-info.org/ wiki as well

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729

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Re: [asterisk-users] Maximum concurrent calls using call files

[Stephen J Alexander]

I haven't used it, so can't recommend it per se; but as I understand
it, iperf is a tool that can do that kind of simulation for you:
http://iperf.sourceforge.net/ might be worth trying before you build
your own modules.

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729


On Fri, Jul 6, 2012 at 4:01 PM, sathiish kumar  wrote:
> I am planning on building a testing module which would spawn about 500 calls
> in order to test the performance of the network by transferring audio/speech
> files to end points at that juncture.Is it possible to spawn as many
> concurrent calls (or nearly concurrent calls) using just call files.Is there
> a limit as to the maximum number that could be spawned.?
> I tried doing this for about 20 calls and found that there is
> autofallthrough after a point of time.Is this a problem with my dialplan or
> is it because of the call files (i also get a warning which states that the
> ast_queue_frame:Exceptionally long queue length)
>
> Thanks,
> Sathiish
>
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[asterisk-users] asterisk 10.4.2 realtime callgroup missing

[Stephen Collier]

I am using asterisk 10.4.2 and everything is working correctly except
callgroup is not working correctly. I have callgroup field in mysql but
if I run sip show peer  on the asterisk cli I get the callgroup
field as blank all other fields correctly populated including
pickupgroup.

I'm not sure where to look to debug the realtime sip connection

Cheers
Stephen Collier


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Re: [asterisk-users] Dual- or Quad ISDN cards for PCI-X Slots

[Stephen J Alexander]

That graphic doesn't cover PCI-X at all. PCI-X uses the long connector like
the 64-bit cards shown, but PCI-X is 32-bit.

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729


On Tue, May 29, 2012 at 8:25 AM, Eliezer Croitoru wrote:

> On 29/05/2012 15:18, Kevin P. Fleming wrote:
>
>> On 05/29/2012 01:48 AM, Michelle Konzack wrote:
>>
>>> No, it does not fit, since PCI 2.0 is 5V and has only one notch.
>>>
>>> PCI 2.1, 2.2. and 2.3 do have two notches, because they are 3.3V.
>>>
>>> In clear, you can not insert old 5V PCI 2.0 cards into a 3.3V PCI-X slot
>>>
>>
>> Ahh, your real issue is voltage then, not the PCI specification that the
>> card is compliant with. Cards can be compliant with any of the PCI
>> versions you mentioned and still be 5V only, 3.3V only, or 5V/3.3V
>> compatible.
>>
>> All modern ISDN BRI cards usable with Asterisk are both 5V and 3.3V
>> compatible, but as you say, they aren't available in your price range.
>>
>>  it seems like this picture:
> http://upload.wikimedia.org/**wikipedia/commons/6/6f/PCI_**Keying.png<http://upload.wikimedia.org/wikipedia/commons/6/6f/PCI_Keying.png>
>
> will clear every doubt about pci-x and other pci.
>
> --
> Eliezer Croitoru
> https://www1.ngtech.co.il
> IT consulting for Nonprofit organizations
> eliezer  ngtech.co.il
>
>
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Re: [asterisk-users] Vitelity Setup

[Stephen J Alexander]

If your server says it is registered, that could be part of the problem.
Vitelity doesn't use trunk registration, only IP authentication. You should
not be using a registration string in your trunk definition. I don't know
if it will hurt but it won't help.

It sounds like you might have only 1 trunk defined, but you need 2; one for
inbound and one for outbound. Their servers for incoming calls and for
outgoing calls are separate. If fixing that doesn't do the job, make sure
that incoming traffic from Vitelity is correctly routed to your PBX (and
that they have the correct IP to send SIP traffic to).

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729


On Fri, May 25, 2012 at 4:12 PM, Ralph Green  wrote:

> Howdy,
>  Since the subject is Viteiy Setup, I don't think this is off topic.
> My big problem with Vitelity is getting my server to register for
> incoming calls.  I can make outgoing calls just fine.  My server says
> it is registered with Vitelity, but no calls come in.  Every attempt
> to call the number generates an email saying there was a failed call.
> I am using IAX, not SIP, and that is probably part of the problem.
> IAX should work better in several ways, but few enough people use it.
> Vitelity support has been unhelpful so far.  My suspicion is that
> there is a setting they need to make in their server so that calls go
> to the registered IAX server, instead of looking for a SIP
> registration, which is not there.  Has anyone here worked past such a
> problem?  Was there some special thing I need to ask Vitelity?
> Thanks,
> Ralph
>
>
> On 5/24/12, Stephen J Alexander  wrote:
> > If I were troubleshooting this, the next thing I would do is verify
> > connectivity on the relevant ports – more plainly, make sure that there's
> > not a firewall rule with unintended consequences somewhere between your
> > asterisk and your ISP. Otherwise, as Alejandro suggests – check with
> > Vitelity support.
> >
> > Regards,
> >
> > Stephen J Alexander
> > MPBX, LLC
> > http://mpbx.com
> > 832-713-6729
> >
> >
> > On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass  wrote:
> >
> >> On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N
> >>  wrote:
> >> > yes I did that, even then i am not able to make outbound and inbound
> as
> >> > well.
> >> >
> >> >
> >>
> >>
> >> That's weird. Guess you're gonna have to place a detailed ticket to
> >> them. It sounds like a network problem to me but without any detailed
> >> info it's hard to say. Maybe you can try sip set debug in the console
> >> for the IP and see if you can get an idea of what is happening at the
> >> packet level.
> >>
> >> We use Vitel, Skype SIP (we recently eliminated this one), and now
> >> Gafachi and they all seem to work per there set-up instructions right
> >> away.
> >>
> >> --
> >> Alejandro
> >>
> >> --
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> >>
> >
>
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Re: [asterisk-users] Vitelity Setup

[Stephen J Alexander]

If I were troubleshooting this, the next thing I would do is verify
connectivity on the relevant ports – more plainly, make sure that there's
not a firewall rule with unintended consequences somewhere between your
asterisk and your ISP. Otherwise, as Alejandro suggests – check with
Vitelity support.

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729


On Thu, May 24, 2012 at 9:24 AM, Alejandro Imass  wrote:

> On Thu, May 24, 2012 at 4:07 AM, Gopalakrishnan N
>  wrote:
> > yes I did that, even then i am not able to make outbound and inbound as
> > well.
> >
> >
>
>
> That's weird. Guess you're gonna have to place a detailed ticket to
> them. It sounds like a network problem to me but without any detailed
> info it's hard to say. Maybe you can try sip set debug in the console
> for the IP and see if you can get an idea of what is happening at the
> packet level.
>
> We use Vitel, Skype SIP (we recently eliminated this one), and now
> Gafachi and they all seem to work per there set-up instructions right
> away.
>
> --
> Alejandro
>
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Re: [asterisk-users] Vitelity Setup

[Stephen J Alexander]

Alejandro's setup looks correct; you can also get the correct config using
Vitelity's wizard tool for setting up the trunks.

The only thing I would add is that if your account is setup with a session
border controller you will need to use the SBC's IP address instead of the
IP the wizard gives you. If you have an SBC, the fact will be noted in your
account including the IP address.

I've found Vitelity's tech support to be pretty helpful too, should you
need to contact them.

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729


On Wed, May 23, 2012 at 9:14 AM, Alejandro Imass  wrote:

> On Wed, May 23, 2012 at 7:57 AM, Gopalakrishnan N
>  wrote:
> > Hi,
> >
> > I am unable to register vitelity SIP trunk, where its keep on sending
> > registration request, and I am using Asterisk 1.4.39.2, my registration
> > procedure as follows,
> >
> > sip.conf
> >
> > register => username:sec...@sip41.vitelity.net:5060
> >
>
> We use viteity w/o registration like so:
>
> [vitel-inbound]
> type=friend
> dtmfmode=auto
> host=inbound24.vitelity.net
> context=vitelity-inbound
> allow=all
> insecure=very
>
> [vitel-outbound]
> type=friend
> dtmfmode=auto
> host=outbound.vitelity.net
> context=vitelity-outbound
> allow=all
> insecure=very
>
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Re: [asterisk-users] Configuring OpenVOX A400P issues

[Stephen J Alexander]

As a first step, I would verify the problem is not in the phone itself. Can
you duplicate the behavior with a different telephone? Does the handset act
properly when connected directly to your PSTN line?

If these tests still indicate the problem is on the trixbox, can you verify
you changed nothing in the configuration since it was working? If so, that
strongly suggests a hardware problem with your cards. I don't know the
cards specifically so I can't comment on that in detail.

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729


On Sun, May 13, 2012 at 11:05 AM, Kaya Saman  wrote:

> Hi,
>
> I'm using Trixbox 2.8 edition and having issues with my PSTN line and the
> OpenVox A400P card.
>
>
> During purchase I decided to go for the 1xFXS and 1xFXO option.
>
>
> I hooked the PSTN and my POTs DECT phone up correctly and configured the
> appropriate trunks, routes, and extensions.
>
>
> The system was working fine for a bit until all of a sudden it just went
> funny. by this I mean that an inbound call would be detected, the DECT
> POTs phone would ring but pressing the green button isn't answering the
> call.
>
>
> On the Trixbox UI I can see that somebody called and it shows as answered
> but apart from that, the callee side still rings.
>
>
> Additionally sometimes the POTs phone just won't ring either??
>
>
> Internally, using SIP to POTs extensions work fine..
>
>
> The system is a Mini-ITX Intel ATOM D510 1.66GHz driven box with 1GB RAM
> and 40GB SSD. My PSTN provider is BT in the UK.
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Re: [asterisk-users] Asterisk 1.8 Transfer CallerID

[Stephen Collier]

We are using snom 821's and it works as described with sendrpid and
trustrpid both set. We are using realtime for sip peers and users.

Version 1.8.10.0

Cheers
Stephen
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: Wednesday, 9 May 2012 12:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 1.8 Transfer CallerID

On 05/08/2012 04:32 PM, Karsten Wemheuer wrote:
> Hi,
>
> Am Dienstag, den 08.05.2012, 14:13 +0200 schrieb Jonas Kellens:
>> Hello,
>>
>> when a call comes in and is answered by colleague A, this colleague A

>> sees the CallerID of the external calling number.
>>
>> When colleague A transfers the call to colleague B, attended or 
>> unattended, then colleague B sees the number of colleague A on his 
>> screen while talking to the external calling number.
>>
>> I expect here that colleague B would see the external calling number 
>> on the screen of his IP-phone.
>>
>> How can I get this behaviour ?
> As far as I understand You, you want to update the callerID (A calls 
> B, the call is established, then A transfers the external party to B).

> In this case (and if You are using SIP endpoints), check out the 
> config params sendrpid and rpid_update in sip.conf. AFAIK this feature

> is working in versions from 1.8 and newer.

Yes this is what I want to establish. I will check out sendrpid and
rpid_update.


Thanks.

Jonas.

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Re: [asterisk-users] Asterisk 1.6.2.22 backtrace

[Stephen J Alexander]

To give a more verbose answer, the line in the backtrace that says no
debugging symbols found answers the rest of your question: It's not
possible to tell where the signal comes from, because asterisk was built
without debugging symbols. This is not a proper backtrace in that it
doesn't trace back to anything. So we can't say from this what caused the
problem.

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729


On Fri, May 4, 2012 at 3:58 PM, Doug Lytle  wrote:

> >> is this an answer or is the most of your answer missing ?
>
> That would be his answer.  If you type kill -l, you'll get a listing of
> signals:
>
> kill -l
>
>  1) SIGHUP   2) SIGINT   3) SIGQUIT  4) SIGILL   5) SIGTRAP
>  6) SIGABRT
>
> 6 would be signal abort.
>
> Doug
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
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Re: [asterisk-users] hangup problem on T1 span

[Stephen J Alexander]

Tzafrir,

Thanks for your response. I'll check into those items.

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729


On Thu, May 3, 2012 at 4:39 AM, Tzafrir Cohen wrote:

> On Wed, May 02, 2012 at 11:18:54AM -0500, Stephen J Alexander wrote:
> > Hello all,
> >
> > I'm trying to solve a problem on a T1 span setup wherein calls are
> > apparently not hanging up properly.
>
> CAS or PRI?
>
> >
> > The system in question is using a Xorcom Astribank with 1 full and 1
> > partial T1 span, and running Asterisk 1.4.36.
> >
> > The symptom is that when a call hangs up on a DAHDI channel (according to
> > Asterisk), and another outgoing call tries to open a new channel on the
> > same line as the hung-up call within approximately a minute of the
> hangup,
> > the new call gets a congestion notice ("all circuits busy") from
> > asterisk. After about a minute passes after the hangup, the line becomes
> > available again. So it seems like the channels are not hanging up when
> > Asterisk tells them to, and Asterisk doesn't know it.
> >
> > I suspected a signaling issue, and this appeared confirmed when I
> > discovered that the signalling was set in chan_dahdi.conf as "fxs_ks"
> (this
> > installation had been converted from analog lines by another company; I
> > guess that was an oversight?).
>
> The signalling and such is probably set in
> /etc/asterisk/dahdi-channels.conf so that setting does not matter.
>
> >
> > So I changed it to pri_cpe, as my reading of the docs indicated was
> proper.
> > After this change and restarting everything, though, the symptoms
> persist.
> > So I figure that either my reading of the docs is wrong (and therefore
> > pri_cpe is not the right signaling) OR something totally unrelated is
> going
> > on.
> >
> > Can someone please clue me in here? I am a bit at a loss. Let me know if
> > you need further information about the system/environment.
>
> What is the output of 'dahdi show channel N' for one such a "bad"
> channel when not in a call? Are you sure it's not in a call? See the
> output of 'core show channels'.
>
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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[asterisk-users] hangup problem on T1 span

[Stephen J Alexander]

Hello all,

I'm trying to solve a problem on a T1 span setup wherein calls are
apparently not hanging up properly.

The system in question is using a Xorcom Astribank with 1 full and 1
partial T1 span, and running Asterisk 1.4.36.

The symptom is that when a call hangs up on a DAHDI channel (according to
Asterisk), and another outgoing call tries to open a new channel on the
same line as the hung-up call within approximately a minute of the hangup,
the new call gets a congestion notice ("all circuits busy") from
asterisk. After about a minute passes after the hangup, the line becomes
available again. So it seems like the channels are not hanging up when
Asterisk tells them to, and Asterisk doesn't know it.

I suspected a signaling issue, and this appeared confirmed when I
discovered that the signalling was set in chan_dahdi.conf as "fxs_ks" (this
installation had been converted from analog lines by another company; I
guess that was an oversight?).

So I changed it to pri_cpe, as my reading of the docs indicated was proper.
After this change and restarting everything, though, the symptoms persist.
So I figure that either my reading of the docs is wrong (and therefore
pri_cpe is not the right signaling) OR something totally unrelated is going
on.

Can someone please clue me in here? I am a bit at a loss. Let me know if
you need further information about the system/environment.

Regards,

Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729
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[asterisk-users] CallerId back to incoming

[Stephen Collier]

I'm currently doing some testing with Asterisk ( 1.8.11.0) on RHEL6
using realtime for sippeers, sipusers and musiconhold

I have  Avaya definity <-> PRI E1 <-> Asterisk 1 <-> IAX2  <-> Asterisk
2 

I have peers (sip) snom 821s on both Asterisk 1 and 2 all calls working
between all systems.

CallerID from Asterisk to Avaya is working correctly.

The problem is a caller from Avaya to Asterisk displays correctly the
CID of the Asterisk Extension to the calling party on the Avaya but only
if the peer is on Asterisk 1. If the peer is on Asterisk 2 only the CID
of the PRI on the avaya side is displayed. I hope this makes sense.

I'm not sure where to start looking or whether its even possible.

I can of course supply any of the configs that may help.

Cheers

Stephen Collier

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Re: [asterisk-users] Streaming musiconhold via mpg123

[Stephen Brown]

DAHDI it is are there any known workarounds? I use the FreePBX
distro and they are a bit behind, so no telling when they will update.

On 2/21/2012 6:45 PM, Israel Gottlieb wrote:
> that bug is running since the start of 1.8 and has been fixed in 1.8.9
>
> https://issues.asterisk.org/jira/browse/ASTERISK-17474
>
> i know it says that after the first time asterisks starts it works but
> thats true only if the moh was loaded before the timing
>
> its a long story but the fix is finally in
>
> when typing timing test in the cli what timer to get if its dahdi then
> thats the probably problem
>
>
>
> On Wed, Feb 22, 2012 at 1:34 AM, Stephen Brown
> mailto:stephen.brow...@gmail.com>> wrote:
>
> On 2/21/2012 3:38 PM, isr...@gmail.com <mailto:isr...@gmail.com>
> wrote:
> > There is a bug in up to version 1.8.9 with external moh sources
> and dahdi timers
>
> Do you have a link to the bug report? I was unable to find
> anything but
> it's possible I'm not looking hard enough ;)
>
> > Share with us your musiconhold.conf configuration please.
>
> Here it is... please excuse the mess, it's been a wild ride so my
> formatting/commenting has been left in-tact:
>
> ;
> ; Music on hold class definitions
> ; This is using the new 1.2 config file format, and will not work
> with 1.0
> ; based Asterisk systems
> ;
> ; #include musiconhold_custom.conf
> ; #include musiconhold_additional.conf
> ;[default]
> ;mode=custom
> ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r
> 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074
> ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0
> http://scfire-ntc-aa03.stream.aol.com:80/stream/1074
>
> [test]
> mode=custom
> ;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r
> 8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074
> ;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0
> http://scfire-ntc-aa03.stream.aol.com:80/stream/1074
> application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000
> /var/lib/asterisk/sounds/music/Rolling In The Deep.mp3
>
> I setup a simple 2 digit extension to call the test context and my MP3
> file nor my stream will play, and here's something else
> interesting: If
> use the MP3Player application to play an MP3, mpg123 spawns and plays
> it. I came to this conclusion by running ps aux | grep mpg while the
> song was playing.
>
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Re: [asterisk-users] Streaming musiconhold via mpg123

[Stephen Brown]

On 2/21/2012 3:38 PM, isr...@gmail.com wrote:
> There is a bug in up to version 1.8.9 with external moh sources and dahdi 
> timers

Do you have a link to the bug report? I was unable to find anything but
it's possible I'm not looking hard enough ;)

> Share with us your musiconhold.conf configuration please.

Here it is... please excuse the mess, it's been a wild ride so my
formatting/commenting has been left in-tact:

;
; Music on hold class definitions
; This is using the new 1.2 config file format, and will not work with 1.0
; based Asterisk systems
;
; #include musiconhold_custom.conf
; #include musiconhold_additional.conf
;[default]
;mode=custom
;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r
8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074
;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0
http://scfire-ntc-aa03.stream.aol.com:80/stream/1074

[test]
mode=custom
;application=/usr/src/mpg123/mpg123-1.13.4/src/mpg123 -q -s --mono -r
8000 -f 8192 -b 0 http://scfire-ntc-aa03.stream.aol.com:80/stream/1074
;application=/usr/bin/mpg123 -q -s --mono -r 8000 -f 8192 -b 0
http://scfire-ntc-aa03.stream.aol.com:80/stream/1074
application=/usr/bin/mpg123 -q -s -f 8192 --mono -r 8000
/var/lib/asterisk/sounds/music/Rolling In The Deep.mp3

I setup a simple 2 digit extension to call the test context and my MP3
file nor my stream will play, and here's something else interesting: If
use the MP3Player application to play an MP3, mpg123 spawns and plays
it. I came to this conclusion by running ps aux | grep mpg while the
song was playing.

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[asterisk-users] Streaming musiconhold via mpg123

[Stephen Brown]

At my wits end with this, and can't proceed any further so I'm hoping 
someone has seen this and can assist. I can not get streaming 
musiconhold to work with Asterisk.


My Asterisk version is 1.8.8.0 and the mpg123 version is 1.9.1, OS is 
CentOS 5.7. When I call the musiconhold class (default for example) I 
get nothing but silence. I've exhausted my troubleshooting capabilities 
at this point, I've tried everything I can think of to include:


- a newer version of mpg123, I went with the latest version
- verified I could play an MP3 file by itself in Asterisk by using the 
MP3Player application


What does not work, is if I use the mpg123 application for musiconhold 
to play a standalone file or a streaming source. I seem to be missing 
something and I just can't quite put a finger on it.






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Re: [asterisk-users] PRI Issues After Upgrade

[Stephen H. Gerstacker]

I'm digging this back up since the problem persists.  I've been attempting to 
figure out what's been going on and I'm at a stopping point again.

Even though I swore I checked it, it turns out the two cards and the ethernet 
controller were all on the same IRQ.  I moved the cards around so they are all 
isolated on their own IRQs, but the same problems persist.  I'm getting all of 
the following:

PRI Span: 1 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state 
7(Multi-frame established)

PRI Span: 1 !! Unknown IE 128 (cs0)
-- Span 1: Channel 0/23 got hangup, cause 87

[Sep 29 11:55:51] WARNING[1331]: sig_pri.c:1054 pri_find_dchan: Span 1: No 
D-channels available!  Using Primary channel as D-channel anyway!
  == Primary D-Channel on span 1 up

PRI Span: 1 !! Unknown IE 128 (cs0)
-- Span 1: Channel 0/23 got hangup, cause 14
[Sep 29 11:52:57] WARNING[15586]: app_dial.c:1452 wait_for_answer: Unable to 
forward frametype: 2


I've also replaced the cable from the PRI to the card, just in case…

Any ideas?

Stephen H. Gerstacker
Sr. Database Developer
Electronic Data Payment Systems
Phone: 866.578.9740 ext. 114
Fax: 866.528.3854
www.edpaymentsystems.com<http://www.edpaymentsystems.com>

On Sep 14, 2011, at 11:04 AM, Doug Lytle wrote:


Stephen H. Gerstacker wrote:
Came in this morning to more of the same:


Then, if you have the ability, I'd drop 1.2 back into place and see if it's 
happy.  But, my feeling is that you'll need to contact the provider.

The other thing that comes to mind is that your PRI card is having issues.

Doug

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Re: [asterisk-users] PRI Issues After Upgrade

[Stephen H. Gerstacker]

Came in this morning to more of the same:

PRI Span: 1 !! Unknown IE 128 (cs0)
-- Span 1: Channel 0/23 got hangup, cause 81



Also, I got a lot of this as well:

[Sep 14 05:54:05] WARNING[16624]: sig_pri.c:1054 pri_find_dchan: Span 1: No 
D-channels available!  Using Primary channel as D-channel anyway!
  == Primary D-Channel on span 1 up
  == Primary D-Channel on span 1 down
[Sep 14 06:02:35] WARNING[16624]: sig_pri.c:1054 pri_find_dchan: Span 1: No 
D-channels available!  Using Primary channel as D-channel anyway!
  == Primary D-Channel on span 1 up
  == Primary D-Channel on span 1 down
[Sep 14 06:15:26] WARNING[16624]: sig_pri.c:1054 pri_find_dchan: Span 1: No 
D-channels available!  Using Primary channel as D-channel anyway!
  == Primary D-Channel on span 1 up
PRI Span: 1 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state 
7(Multi-frame established)
PRI Span: 1 TEI=0 MDL-ERROR (A): Got supervisory frame with F=1 in state 
7(Multi-frame established)

I believe those popped up under the dms100 config as well.

Stephen H. Gerstacker
Sr. Database Developer
Electronic Data Payment Systems
Phone: 866.578.9740 ext. 114
Fax: 866.528.3854
www.edpaymentsystems.com<http://www.edpaymentsystems.com>

On Sep 14, 2011, at 5:31 AM, Doug Lytle wrote:


Stephen H. Gerstacker wrote:
Is there a big difference between the two?

>From what I've read, a DMS100 can redirect a call off of your system, meaning 
>that if you have an inbound call and you want to redirect it to a different 
>number, the DMS100 will redirect the call and take your system out of the 
>call's path.

In the current setup, you'd end up using 2 lines to do the forward and keep 
them busy until 1 of the parties hang up.

Doug

--
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] PRI Issues After Upgrade

[Stephen H. Gerstacker]

I made the switch and everything seems to be working. It's hard to tell, since 
it never seems to fail for me, but fails once people get in. 

A question, though. When we moved the original box to our new office, they 
asked if we could support the dms100 setting, which worked. National seems to 
be working.  Is there a big difference between the two? 

I'm just a simple programmer who happens to be the only IT guy in the office. 

- Stephen H. Gerstacker

On Sep 13, 2011, at 10:23, "Doug Lytle"  wrote:

> 
> Stephen H. Gerstacker wrote:
>> Anything else I can try?
> 
> Try switchtype=national just for testing.
> 
> Doug
> 
> 
> -- 
> 
> Ben Franklin quote:
> 
> "Those who would give up Essential Liberty to purchase a little Temporary 
> Safety, deserve neither Liberty nor Safety."
> 
> 
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Re: [asterisk-users] PRI Issues After Upgrade

[Stephen H. Gerstacker]

I disabled the echo cancelled on the PRI and the same issues are still popping 
up:

PRI Span: 1 !! Unknown IE 128 (cs0)
-- Span 1: Channel 0/22 got hangup, cause 16

Anything else I can try?

Stephen H. Gerstacker
Sr. Database Developer
Electronic Data Payment Systems
Phone: 866.578.9740 ext. 114
Fax: 866.528.3854
www.edpaymentsystems.com<http://www.edpaymentsystems.com>

On Sep 11, 2011, at 9:52 AM, Stephen H. Gerstacker wrote:

When we moved buildings, the PRI provider specifically asked to switch to 
dms100. That's how the old server was as well.

I'll try the echo canceller first.

- Stephen H. Gerstacker

On Sep 11, 2011, at 9:07, "Doug Lytle" 
mailto:supp...@drdos.info>> wrote:


Stephen H. Gerstacker wrote:

switchtype=dms100


Are you sure that your switchtype is correct and your provider has a dms100?  
If not, change this to national.

If so, the two things I'd try to narrow it down are:

1.)  Temporarily remove the AEX410 card and test again
2.)  Temporarily disable OSLEC for your echo canceller and test again.

Doug

--
Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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Re: [asterisk-users] PRI Issues After Upgrade

[Stephen H. Gerstacker]

When we moved buildings, the PRI provider specifically asked to switch to 
dms100. That's how the old server was as well.

I'll try the echo canceller first.

- Stephen H. Gerstacker

On Sep 11, 2011, at 9:07, "Doug Lytle"  wrote:

> 
> Stephen H. Gerstacker wrote:
>> 
>> switchtype=dms100
>> 
> 
> Are you sure that your switchtype is correct and your provider has a dms100?  
> If not, change this to national.
> 
> If so, the two things I'd try to narrow it down are:
> 
> 1.)  Temporarily remove the AEX410 card and test again
> 2.)  Temporarily disable OSLEC for your echo canceller and test again.
> 
> Doug
> 
> -- 
> Ben Franklin quote:
> 
> "Those who would give up Essential Liberty to purchase a little Temporary 
> Safety, deserve neither Liberty nor Safety."
> 
> 
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Re: [asterisk-users] PRI Issues After Upgrade

[Stephen H. Gerstacker]

/etc/dahdi/system.conf:

# Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug 31 18:10:31 2011
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: WCT1/0 "Digium Wildcard TE110P T1/E1 Card 0" (MASTER) B8ZS/ESF
span=1,1,0,esf,b8zs
# termtype: te
bchan=1-23
dchan=24
echocanceller=oslec,1-23

# Span 2: WCTDM/0 "Wildcard AEX410 Board 1"
fxsks=25
echocanceller=oslec,25
fxsks=26
echocanceller=oslec,26
# channel 27, WCTDM/0/2, no module.
# channel 28, WCTDM/0/3, no module.

# Global data

loadzone = us
defaultzone = us

~

/etc/asterisk/chan_dahdi.conf

[trunkgroups]

[channels]

usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=256
echocancelwhenbridged=no

callerid = asreceived
faxdetect=both

signalling=fxs_ks
context=incoming-local
group=1,2
channel => 25-26

context=incoming-ld
switchtype=dms100
signalling=pri_cpe
group=1,3
channel => 1-23

~

Stephen H. Gerstacker
Sr. Database Developer
Electronic Data Payment Systems
Phone: 866.578.9740 ext. 114
Fax: 866.528.3854
www.edpaymentsystems.com<http://www.edpaymentsystems.com>

On Sep 9, 2011, at 1:05 PM, Doug Lytle wrote:


Stephen H. Gerstacker wrote:
so any help would be appreciated

You could start by supplying snippets of your config for the PRI.  Telling us 
what PRI card you're using.

For example, my dahdi config for my PRI:

cd /etc/dahdi

cat system.conf

span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
defaultzone=us
loadzone=us

span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
defaultzone=us
loadzone=us


cd /etc/asterisk


Show snippets of your chan_dahdi.conf


switchtype=national
context=definity
signalling=pri_cpe
echocancel=no
echotraining=no
echocancelwhenbridged=no
pridialplan=unknown
group=1
rxgain=0.0
txgain=0.0
usecallerid=yes
callerid=asreceived
channel=1-23

switchtype=national
context=cts
signalling=pri_cpe
echocancel=no
echotraining=no
echocancelwhenbridged=no
pridialplan=unknown
rxgain=0.0
txgain=0.0
group=2
usecallerid=yes
callerid=asreceived


Doug

--

Ben Franklin quote:

"Those who would give up Essential Liberty to purchase a little Temporary 
Safety, deserve neither Liberty nor Safety."


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[asterisk-users] PRI Issues After Upgrade

[Stephen H. Gerstacker]

I've finally moved from my 5 year old 1.2 installation to a 1.8 installation.

I'm using the packages supplied by asterisk.org<http://asterisk.org> for Ubuntu 
10.04, so I am at Asterisk 1.8.6.

On the console, I am seeing:

PRI Span: 1 !! Unknown IE 128 (cs0)
-- Span 1: Channel 0/22 got hangup, cause 15

And then the call hangs up.  The cause code changes all the time.  It never 
seems to be consistent.  31, 34, 54, 102, 122, etc.  Before it was a different 
type of error.  I've tried multiple versions of libpri and am now on 1.4.12.  
I'm also on DAHDI 2.4.1.  The card in question is a Digium Wildcard TE110P 
T1/E1 Card.

I've tried IRC and the forum and gotten no help, so any help would be 
appreciated.

Stephen H. Gerstacker
Sr. Database Developer
Electronic Data Payment Systems
Phone: 866.578.9740 ext. 114
Fax: 866.528.3854
www.edpaymentsystems.com<http://www.edpaymentsystems.com>

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[asterisk-users] Auto Reply: asterisk-users Digest, Vol 84, Issue 15

[craig . stephen]

I am out of the office on vacation through July 20th, 2011.

I am checking email, and will get back to you as soon as I can.

For urgent matters, contact:

Angie Besse for Oracle Labs, M&A, and Corporate Security Architecture issues.
Tami Sisneros regarding Corporate Architecture Approvals.

Craig


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Re: [asterisk-users] Simplifying dial-plan

[Stephen Reese]

> To answer your first question - ${MACRO_EXTEN} is a macro-specific
> variable.  It's the ${EXTEN} that called the macro, since using ${EXTEN}
> inside a Macro would just give you a value of "s".
>
> As for your second question, that's pretty easy to do.  If every outbound
> call needs to be formatted in the format 1NXXNXX, you would do this
> (again, untested, but should be good along with the macro I gave you
> earlier):
>
> [globals]
> DEFAULT_AREA_CODE=555 ; swap with your default area code
>
> [outbound-context]
>
> exten => _1NXXNXX,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
> exten => _NXXNXX,1,Goto(outbound-context,1${EXTEN},1)
> exten => _NXX,1,Goto(outbound-context,1${DEFAULT_AREA_CODE}${EXTEN},1)
> exten => _011.,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
> exten => 911,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
>
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.selbytech.com

Thanks again Warren, that works quite well!

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Re: [asterisk-users] Simplifying dial-plan

[Stephen Reese]

On Wed, Dec 22, 2010 at 12:59 PM, Warren Selby  wrote:
> On Tue, Dec 21, 2010 at 6:59 PM, Stephen Reese  wrote:
>>
>> On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese  wrote:
>> > Is there a way to include:
>> >
>> > _NXXNXX
>> > _NXX
>> > _011.
>> > _911
>> >
>> > into my current plan:
>> >
>>
>> Sorry, here's the rest.
>>
>> exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
>> exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
>> exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "201"]?20:10)
>> exten => _1NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
>> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
>> exten => _1NXXNXX,n,Goto(h,1)
>> exten => _1NXXNXX,20,Set(CALLERID(all)=${EXTERNAL_CALLERID})
>> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
>> exten => _1NXXNXX,n,Goto(h,1)
>>
>
> Why not make a Macro (or GoSub) to handle this block of code, and then your
> outbound dial lines are just one line calling the Macro?  Saves a lot of
> repeating blocks of code. Something like this (not tested):
>
> [macro-OutboundDial]
> ; ${ARG1} = CHANNEL
> ; ${ARG2} = EXTERNAL_CALLERID
> exten => s,1,Set(Outgoing=${CUT(${ARG1},/,2)})
> exten => s,n,Set(Outgoing=${CUT(Outgoing,-,1)})
> exten => s,n,GotoIf($["${Outgoing}" = "201"]?outbound2:outbound1)
> exten => s,n(outbound1),Set(CALLERID(all)=${ARG2})
> exten => s,n,Dial(SIP/${macro_ext...@vitel-outbound)
> exten => s,n,Goto(h,1)
> exten => s,n(outbound2),Set(CALLERID(all)=${ARG2})
> exten => s,n,Dial(SIP/${macro_ext...@vitel-outbound2)
> exten => s,n,Goto(h,1)
>
> [outbound-context]
>
> exten => _NXXNXX,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
> exten => _NXX,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
> exten => _011.,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
> exten => _911,1,Macro(OutboundDial,${CHANNEL},${EXTERNAL_CALLERID})
>
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.selbytech.com

Thanks Warren, that's what I'm looking to do.

First question is where did ${MACRO_EXTEN} come from, I assumed
${EXTEN} is a built in variable?

Secondly, where would the 1 and/or area-code need to be placed? Could
an additional argument be used to specify the prefix, i.e. a third
variable be specified in the outbond-context to implement the
OutboundDial macro, or is the MACRO_EXTEN suppose to be an
implementation of this?

exten => s,n,Dial(SIP/{$arg3}${macro_ext...@vitel-outbound2)

As Jeroen mentioned previously a goto may be used, would this help,
seems similar to what I am trying to accomplish.

exten => _NXXNXX,1,Goto(1${EXTEN},1)
exten => _NXX,1,Goto(1555${EXTEN},1)

Thanks,
Stephen

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Re: [asterisk-users] Simplifying dial-plan

[Stephen Reese]

On Wed, Dec 22, 2010 at 2:01 AM, Jeroen Eeuwes  wrote:
> Hi Stephen,
>
>> _NXXNXX
>> _NXX
>> _011.
>> _911
>
> Of course it can, but it depends on what you want to do when those
> numbers are called...
>
> I didn't know about the setvar in the sip.conf and actually I think it
> is a much "cleaner" solution. Since you are already using it I would
> suggest to not only use it for CallerID but also for the
> @vitel-outbound like this:
>
> exten => _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
> exten => _1NXXNXX,n,Dial(SIP/${ext...@${outbound})
> exten => _1NXXNXX,n,Goto(h,1)
>
> Of course you'll need to set setvar=Outbound=vitel-outbound or
> setvar=Outbound=vitel-outbound2 in sip.conf.
>
> What do you want to do with the other numbers? If you want to do the
> same as with _1NXXNXX you can just add things like this in your
> extensions.conf:
>
> exten => _NXXNXX,1,Goto(_1NXXNXX,1)
> exten => 911,1,Goto(_1NXXNXX,1)
>
> Or you can do different things if you want that like this:
>
> exten => _NXX,1,Set(CALLERID(all)="No one cares" <0>)
> exten => _NXX,n,Dial(SIP/${ext...@abcdefgh)
> exten => _NXX,n,Goto(h,1)
>
> Best regards,
> Jeroen Eeuwes

Jeroen, I'm trying to avoid rewriting the outgoing block for the
patterns mentioned above. I've placed a pseudo dial-plan below. The
plan needs to dial the 1 and/or also the area code depending on the
pattern they enter. Any tips, thanks.

exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "201"]?20:10)
exten => _1NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten => _NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten => _NXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _NXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)
exten => _NXXNXX,n,Goto(h,1)
exten => _NXX,n,Goto(h,1)
exten => _1NXXNXX,20,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten => _1NXXNXX,n,Goto(h,1)

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[asterisk-users] Simplifying dial-plan

[Stephen Reese]

Is there a way to include:

_NXXNXX
_NXX
_011.
_911

into my current plan:

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Re: [asterisk-users] Simplifying dial-plan

[Stephen Reese]

On Tue, Dec 21, 2010 at 7:58 PM, Stephen Reese  wrote:
> Is there a way to include:
>
> _NXXNXX
> _NXX
> _011.
> _911
>
> into my current plan:
>

Sorry, here's the rest.

exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "201"]?20:10)
exten => _1NXXNXX,10,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)
exten => _1NXXNXX,20,Set(CALLERID(all)=${EXTERNAL_CALLERID})
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten => _1NXXNXX,n,Goto(h,1)

Thanks

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Re: [asterisk-users] Specifying DID for outbound calls

[Stephen Reese]

> I believe I have made a little headway. I have two outgoing DID
> contexts and have changed the GotoIf statement to the extension name.
> User One acts as expected and User two now displays unknown when
> calling so I believe it is trying to to goto 20 but it's not quite
> making it. Any tips? Thanks
>
> [outgoing]
> exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
> exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
> exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "201"]?20:10)
> exten => _1NXXNXX,10,Set(CALLERID(all)="User One" <3012323434>)
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
> exten => _1NXXNXX,20,Set(CALLERID(num)="User Two" <3013232322>)
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
> exten => _1NXXNXX,n,Goto(h,1)
>

Disregard, I had num instead of all for the CALLERID statement.

Thanks for all of the help!

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Re: [asterisk-users] Specifying DID for outbound calls

[Stephen Reese]

On Sun, Dec 19, 2010 at 4:36 AM, Jeroen Eeuwes  wrote:
> Hi Stephen,
>
>> Thanks for the heads up, I have been setting the caller-ID but the
>> trouble I'm running into is specifying the which number to call out
>> as. How can an extension specify a different number? See below for my
>> current extension.conf, thanks.
>
> You can check the channel-name to see which extension is making the
> call and set the CallerID accordingly. The channel-name will be
> something like "SIP/201-abc23ef34" or "SIP/User1-def34abc51". The 201
> or User1 part depends on how you put the username in sip.conf  You can
> use the CUT function to get the calling extension and then jump to the
> correct CallerID. I've used something like this:
>
> [outgoing]
> exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
> exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
> exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "User2"]?20:10)
> exten => _1NXXNXX,10,Set(CALLERID(num)=3012323434)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User1")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
> exten => _1NXXNXX,20,Set(CALLERID(num)=3013232322)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User2")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
>
> But in my case I had two different domains. E.g.
> Dial(SIP/${ext...@provider-id1) and Dial(SIP/${ext...@provider-id2)
> instead of setting the CallerID.
>
> Not that the Cut doesn't work correctly if you use a minus-sign in the 
> username.
>
> Best regards,
> Jeroen Eeuwes

I believe I have made a little headway. I have two outgoing DID
contexts and have changed the GotoIf statement to the extension name.
User One acts as expected and User two now displays unknown when
calling so I believe it is trying to to goto 20 but it's not quite
making it. Any tips? Thanks

[outgoing]
exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "201"]?20:10)
exten => _1NXXNXX,10,Set(CALLERID(all)="User One" <3012323434>)
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)
exten => _1NXXNXX,20,Set(CALLERID(num)="User Two" <3013232322>)
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound2)
exten => _1NXXNXX,n,Goto(h,1)

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Re: [asterisk-users] Specifying DID for outbound calls

[Stephen Reese]

>> So I should have multiple incoming and outgoing contexts? Vitelity
>> will allow me to use IP routing or user/pass auth, the latter would
>> allow me to specify the outgoing context, this would also guarantee
>> the correct account is billed and not alone rely on caller-ID.
>
> Let me clarify further. For calls FROM vitelity you are pretty much
> limited to a single context in sip.conf doing IP based matching. Most
> equipment will not authenticate to you, and chan_sip currently has no
> additional method for separating the accounts into separate contexts.
>
> For calls TO vitelity you should probably have separate contexts.
>
>> Thanks for being responsive, I do not work with Asterisk much,
>> actually I do not touch it unless I need to add more functionality
>> outside of regular patching so my fu is not strong in this area ;-)
>
>
> --
> Joshua Colp
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org
>

Great, I'll get it changed and see if it helps, thanks.

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Re: [asterisk-users] Specifying DID for outbound calls

[Stephen Reese]

>> First, when using multiple accounts from the same DID provider, is it
>> ideal to use IP based routing using one context as I currently am or
>> have a separate contexts for each account in the sip.conf?
>
> That's really the only way to do it presently.

So I should have multiple incoming and outgoing contexts? Vitelity
will allow me to use IP routing or user/pass auth, the latter would
allow me to specify the outgoing context, this would also guarantee
the correct account is billed and not alone rely on caller-ID.

Thanks for being responsive, I do not work with Asterisk much,
actually I do not touch it unless I need to add more functionality
outside of regular patching so my fu is not strong in this area ;-)

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Re: [asterisk-users] Specifying DID for outbound calls

[Stephen Reese]

On Sun, Dec 19, 2010 at 2:40 PM, Joshua Colp  wrote:
> I'm surprised nobody has suggested using the setvar functionality. It's 
> extremely
> useful for stuff like this and would allow you to keep all CallerID 
> information
> with the actual configuration of the device.
>
> Using a configuration entry for sip.conf in another response as an example:
>
> [101]
> type=friend
> username=101
> secret=
> mailbox=101
> callerid="User One" <101>
> host=dynamic
> nat=yes
> dtmfmode=rfc2833
> canreinvite=no
> reinvite=no
> qualify=yes
> setvar=EXTERNAL_CALLERID="User One" <3012323434>
>
> And then in extensions.conf:
>
> exten => _1NXXNXX,1,Set(CALLERID(all)=${EXTERNAL_CALLERID})
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
>
> Of course you could add some sanity checking there to make sure that
> ${EXTERNAL_CALLERID} contains a value and if not default to your
> main DID.
>
> --
> Joshua Colp
> Digium, Inc. | Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at:  www.digium.com  & www.asterisk.org

Joshua, that seems reasonable. I have two "best practice" questions
before moving further if anyone would like to chime in.

First, when using multiple accounts from the same DID provider, is it
ideal to use IP based routing using one context as I currently am or
have a separate contexts for each account in the sip.conf?

Secondly, it never crossed my mind that the caller-ID was being set in
the sip.conf and extensions.conf. I guess the extension.conf takes
precedence. At this point is was not my intention to use the sip.conf
and I can easily remove it and set the variable in the extension.conf.
I am just not familiar with what is ideal.

Most of my configuration has come from snippets I've found per example.

Thanks

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Re: [asterisk-users] Specifying DID for outbound calls

[Stephen Reese]

>On Sun, Dec 19, 2010 at 1:52 PM, William Stillwell  
>wrote:
> You can also just use an agi script to look up their current caller-id in a
> database, and set it to the correct caller-id needed.
>
> exten => _NXXNXX,1,AGI(getcid.pl,${CALLERID(NUM)},1)
> exten => _NXXNXX,n,Dial(SIP/+1${ext...@providerx,60)
> exten => _NXXNXX,n,congestion()
>
> my getcid.pl expects two values, extension callerid, and a type.
>
> 911 gets 0, inhouse gets 1, outside 2 etc. (as I ust the getcid for
> different Dial() options.
>
> The script then looks up there "station" callerid, and set it to an
> apporiate value, 911 always gets local in house direct number, regular stuff
> gets a toll number, inhouse gets there extension number, and if there
> callerid is not found in the database it returns a 'default' value.
>
> This way every user can have multiple caller id's .

William, I'm not familiar with AGI scripts, it might be a little
overkill for what I am trying to accomplish.

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Re: [asterisk-users] Specifying DID for outbound calls

[Stephen Reese]

> You can check the channel-name to see which extension is making the
> call and set the CallerID accordingly. The channel-name will be
> something like "SIP/201-abc23ef34" or "SIP/User1-def34abc51". The 201
> or User1 part depends on how you put the username in sip.conf  You can
> use the CUT function to get the calling extension and then jump to the
> correct CallerID. I've used something like this:
>
> [outgoing]
> exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
> exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
> exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "User2"]?20:10)
> exten => _1NXXNXX,10,Set(CALLERID(num)=3012323434)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User1")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
> exten => _1NXXNXX,20,Set(CALLERID(num)=3013232322)
> exten => _1NXXNXX,n,Set(CALLERID(name)="User2")
> exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
> exten => _1NXXNXX,n,Goto(h,1)
>
> But in my case I had two different domains. E.g.
> Dial(SIP/${ext...@provider-id1) and Dial(SIP/${ext...@provider-id2)
> instead of setting the CallerID.
>
> Not that the Cut doesn't work correctly if you use a minus-sign in the 
> username.
>
> Best regards,
> Jeroen Eeuwes

Thanks Jeroen, though it is still not firing correct, I have provided
a little more information.

Here are the channel-names:

SIP/201-000a

SIP/101-0012

Here is the extension information from the sip.conf:

[101]
type=friend
username=101
secret=
mailbox=101
callerid="User One" <101>
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes

[201]
type=friend
username=201
secret=
mailbox=201
callerid="User Two" <201>
host=dynamic
nat=yes
dtmfmode=rfc2833
canreinvite=no
reinvite=no
qualify=yes

Here is the updated outgoing context that you provided with a few updates.

[outgoing]
exten => _1NXXNXX,1,Set(Outgoing=${CUT(CHANNEL,/,2)})
exten => _1NXXNXX,n,Set(Outgoing=${CUT(Outgoing,-,1)})
exten => _1NXXNXX,n,GotoIf($["${Outgoing}" = "User Two"]?20:10)
exten => _1NXXNXX,10,Set(CALLERID(num)=3012323434)
exten => _1NXXNXX,n,Set(CALLERID(name)="User One")
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)
exten => _1NXXNXX,20,Set(CALLERID(num)=3013232322)
exten => _1NXXNXX,n,Set(CALLERID(name)="User Two")
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)
exten => _1NXXNXX,n,Goto(h,1)

Based on the information above, what should be altered to correctly
associated the number with the relevant extension?

Thanks

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Re: [asterisk-users] Specifying DID for outbound calls

[Stephen Reese]

> The outgoing caller-id is probably just the extension number, so the
> provider is setting it to a default (usually the main billing number).  You
> can set what Asterisk sends as the outbound Caller-ID in the outbound
> context before the Dial statement.  Make sure your provider will honor what
> you set, as many filter what you can send to only the DIDs they provide for
> you.
>
> Take a look here for more information on setting the caller-id in the
> dialplan:
>
> http://www.voip-info.org/wiki/view/Asterisk+func+callerid
>
> -Jonathan

Thanks for the heads up, I have been setting the caller-ID but the
trouble I'm running into is specifying the which number to call out
as. How can an extension specify a different number? See below for my
current extension.conf, thanks.

[default]
exten => 201,1,Dial(SIP/201@,30)
exten => 201,n,Voicemail(2...@default)
exten => 201,n,Hangup

exten => 202,1,Dial(SIP/202,30)
exten => 202,n,Voicemail(2...@default)
exten => 202,n,Hangup


include => inbound
include => outgoing

[inbound]
exten => 3012323434,1,Goto(default,201,1)
exten => 3013232322,1,Goto(default,202,1)

[outgoing]

exten => _1NXXNXX,1,Set(CALLERID(num)=3012323434)
exten => _1NXXNXX,n,Set(CALLERID(name)="User1")
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)

[outgoing2]

exten => _1NXXNXX,1,Set(CALLERID(num)=3013232322)
exten => _1NXXNXX,n,Set(CALLERID(name)="User2")
exten => _1NXXNXX,n,Dial(SIP/${ext...@vitel-outbound)

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[asterisk-users] Specifying DID for outbound calls

[Stephen Reese]

The host I am working with has two accounts from the same DID
provider. Incoming calls work correctly and dial the appropriate
extensions. This also allows incoming calls to be billed appropriately to the
individual DID accounts.

Outgoing calls from either extension default to the first DID, i.e.
calls from either extension have the same callerID. How can an
extension specify separate outgoing contexts so the correct number is
associated with it, also allowing the SIP provider to recognize the
difference for billing purposes, or is there a better way?

In short I'm looking to associate an outgoing call from an extension
with a specific number.

Here's the sip.conf for both accounts as they are using IP routing,
I'm assuming I do not have to perform auth based to separate the two
accounts for outgoing calls:

[vitel-inbound]
type=friend
dtmfmode=auto
host=inbound18.vitelity.net
context=inbound
allow=all
insecure=very

[vitel-outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
context=outbound
insecure=very
allow=all

Thanks

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[asterisk-users] Asterisk 1.8 debian packages?

[Stephen Brown]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

Anyone know of any Asterisk 1.8 deb's available or when they might be
included in backports or (hopefully) Squeeze?

I can compile from source... but would much rather have a pre-packaged
binary if one exists...

Thanks,
Stephen
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Re: [asterisk-users] Installing Asterisk to it's own directory

[Stephen Brown]

Thanks... I actually did a ./configure --prefix=/root/asterisk18 and 
ended up with this:

r...@debian-squeeze:~/asterisk18# pwd
/root/asterisk18

r...@debian-squeeze:~/asterisk18# ls -al
total 32
drwxr-xr-x 8 root root 4096 Nov 19 18:09 .
drwx-- 5 root root 4096 Nov 19 18:37 ..
drwxr-xr-x 3 root root 4096 Nov 19 18:09 etc
drwxr-xr-x 3 root root 4096 Nov 19 18:09 include
drwxr-xr-x 3 root root 4096 Nov 19 18:09 lib
drwxr-xr-x 2 root root 4096 Nov 19 18:09 sbin
drwxr-xr-x 3 root root 4096 Nov 19 18:09 share
drwxr-xr-x 6 root root 4096 Nov 19 18:09 var

Have I essentially accomplished the same thing by doing it this way? 
This is in a virtual machine alongside an Asterisk 1.6 install (for 
testing), I'm still a little gunshy to touch my production box as of 
yet. but the 1.8 install did work, I was able to make a call to the 
demo context :)

Thanks,
Stephen

On 11/19/10 10:13 PM, Jose P. Espinal wrote:
> Hi Stephen,
>
> That's what people do when building precompiled packages for certain
> distros (along with a few more things).
>
> I use to do the following when building packages (with a few more options):
>
> ./configure --prefix=/usr --sysconfdir=/etc
> make
> make install DESTDIR=/my/destination/directory
>
> That would create the complete installation structure under
> '/my/destination/directory'
>
>
> Regards,
>
>
>
> Stephen Brown wrote:
>> I'd like to start playing with 1.8, however I don't want to potentially
>> damage anything on my existing 1.6.2 install on my production server.
>>
>> I'd like to test 1.8 against my existing configs leaving my 1.6.2
>> install untouched. Looking at the output of ./configure --help suggests
>> that it's possible to install Asterisk into another prefix of my
>> choosing, but as this is unfamiliar territory to me I'm not exactly sure
>> how to accomplish this?
>>
>> Ideally, I'd like to just dump the newly compiled 1.8 and all it's
>> dependencies into a standalone directory (say /testing/asterisk or
>> something) and update my init script to point to the new binaries. I
>> also run a Sangoma USB FXO card and DAHDI for a POTS line that I would
>> like to test as well, should it work with the pre-compiled binaries that
>> are already there? (DAHDI, etc)
>>
>> I've never tried this before, and before I potentially break something
>> I'd like to know if it's possible and how to implement it?
>>
>> Thanks,
>> Stephen
>>
>>
>>


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[asterisk-users] Installing Asterisk to it's own directory

[Stephen Brown]

I'd like to start playing with 1.8, however I don't want to potentially 
damage anything on my existing 1.6.2 install on my production server.

I'd like to test 1.8 against my existing configs leaving my 1.6.2 
install untouched. Looking at the output of ./configure --help suggests 
that it's possible to install Asterisk into another prefix of my 
choosing, but as this is unfamiliar territory to me I'm not exactly sure 
how to accomplish this?

Ideally, I'd like to just dump the newly compiled 1.8 and all it's 
dependencies into a standalone directory (say /testing/asterisk or 
something) and update my init script to point to the new binaries. I 
also run a Sangoma USB FXO card and DAHDI for a POTS line that I would 
like to test as well, should it work with the pre-compiled binaries that 
are already there? (DAHDI, etc)

I've never tried this before, and before I potentially break something 
I'd like to know if it's possible and how to implement it?

Thanks,
Stephen


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Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice

[Stephen Reese]

> Had the same issue, but have not had a chance to find a good solution.
>  You could change your status to DND.  I tried invisible put seems not
> to be supported.
>
> --
> Paul Belanger | dCAP
> Polybeacon | Consultant
> Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) |
> Blog: http://blog.polybeacon.com | Twitter: http://twitter.com/pabelanger
>

Not a bad idea. The number I am using I have already given out to a
number of people or else I would of registered a second account just
for this purpose.

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Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice

[Stephen Reese]

> I keep the AGI in Git as a version control system.  But, you can view the AGI
> source here:
>
> http://messinet.com/trac/browser/gv/gv.agi
>
> And at the very bottom of that page is a link to download it as an individual
> file here:
>
> http://messinet.com/trac/export/b3229dbba3e01c887b3bdf6b0e0d93e897bd8a59/gv/gv.agi
>
> This is not the same thing as what is in the Changelog.  I am using Asterisk
> 1.6 with this AGI.
>
> -A
> --
> Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
> 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
>

Thanks Anthony,

Interestingly enough outbound dialing started working. Had no clue
until someone called and told me my Google Chat status was updated.

Is there a way to prevent Google Chat from staying logged in but still
be able to dial outbound? People think I'm logged in persistently and
send me messages that I miss. Even if I set a status message in
asterisk most users are not going to understand...

-Stephen

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Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice

[Stephen Reese]

> Since Google Voice (GV) doesn't let us connect diretly via SIP, IAX2, etc.,
> for outbound calls, it acts basically like a fancy click-to-call application.
>
> So...
>
> You need Asterisk to "login" into GV, and "initiate" the call.  GV will dial
> the number you tell it to, then connect it to one of your GV numbers.
>
> In my case, the AGI is what connects to GV and initiates the call.  GV, then
> dials the number I told it to dial, then connects it with my ipKall number
> (which I have as one of my GV numbers).
>
> In Asterisk, the outbound call runs the AGI and places the channel in the DB,
> then waits for an incoming call via my inbound ipKall trunk.
>
> Once the ipKall comes into Asterisk, the Bridge command is used to bridge the
> original (with the matching DB entry) call-- the call that is coming in from
> GV through ipKall.
>
> I suppose you don't need that AGI and could probably do this using Curl in the
> dialplan.
>
> -A
>
> --
> Anthony - http://messinet.com - http://messinet.com/~amessina/gallery
> 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E
>

That makes sense but I do not see where the new feature is in Asterisk
1.8 which include Google Voice support per
http://svn.asterisk.org/svn/asterisk/tags/1.8.0/asterisk-1.8.0-summary.txt:

290973  |dvossel|Make outbound Google Voice calls.   |  |

It seems that the GV has been a feature for sometime with previous
versions? I'm just trying to keep the process as simple as possible
and seeing three different methods is a little confusing:

http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/
(no script referenced)
http://www.davidvossel.com/?p=28 (python script and listed in the
change log above)
http://messinet.com/trac/wiki/AsteriskGVGateway (AGI script)

Is your .agi and .git the same script? I do not have a git client on
this host to see for myself.

Thanks,
Stephen

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Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice

[Stephen Reese]

On Mon, Oct 25, 2010 at 12:50 AM, Anthony Messina  wrote:
> On Sunday, October 24, 2010 05:23:13 pm Stephen Reese wrote:
>> Evening,
>>
>> Has anyone seen a how-to on getting Asterisk to work with Google Talk
>> and Google Voice?
>>
>> Thanks
>
> For Google Voice, I use an ipKall number for the inbound trunk.  Here are the
> relevant sections of my extensions.conf:
>
> ; inbound ipKall trunk (to which Google Voice makes the connection)
> [ipkall]
> exten => ,1,GotoIf($["${DB_EXISTS(gv/channel)}" = "1"]?gv)
> same => n,Goto(default,s,1)
> same => n(gv),Bridge(${DB_DELETE(gv/channel)})
> same => n,AGI(gv/gv.agi,hangup)
> same => n,Hangup()
>
> ; outbound Google Voice initiation
> [gv-out]
> exten => _X.,1,AGI(gv/gv.agi,call)
> same => n,While($["${DB_EXISTS(gv/channel)}" = "1"])
> same => n,Wait(0.3)
> same => n,EndWhile()
> same => n,Hangup()
>
> And the AGI (written in Bash) is here:
> http://messinet.com/trac/wiki/AsteriskGVGateway
> http://messinet.com/trac/browser/gv/gv.agi
>

Does the AGI have to be used? In this example
http://www.davidvossel.com/?p=28 I see mention of a script, but not in
this one: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/

I believe I missing the connection in how the whole process actually
works therefore making troubleshooting a little difficult. I was
hoping with the release of 1.6.0 there wouldn't be a lot of bandage
work to get it to play nicely with Google Voice.

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Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice

[Stephen Reese]

On Sun, Oct 24, 2010 at 9:24 PM, Stephen Reese  wrote:
> On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger
>  wrote:
>> On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese  wrote:
>>> Has anyone seen a how-to on getting Asterisk to work with Google Talk
>>> and Google Voice?
>>>
>> I wrote one last week:
>> http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/
>>
>> Also: http://www.davidvossel.com/?p=28
>>
>
> Paul,
>
> It seems you were using a beta/SVN release for your example. Do the
> following two packages need to be installed if using the stable 1.6.0
> release before building from source? I ask as I am unable to dial out.
>
> $ apt-get install libikesemel-dev
> $ apt-get install libssl-dev
>
> Secondly, do you know if the username/password are sent in clear text
> to the Google?
>

I installed the two packages previously mentioned but still lack
outbound dialing. I enabled debugging and am getting the following
messages. I double checked the password and even changed it to one
without special characters but still the same results.

JABBER: gmail INCOMING: 
[Oct 24 23:07:55] ERROR[28785]: res_jabber.c:1693 aji_act_hook:
JABBER: encryption failure. possible bad password.

JABBER: gmail INCOMING: 
[Oct 24 23:07:55] ERROR[28785]: res_jabber.c:1576 aji_act_hook:
aji_act_hook was called with out a packet
[Oct 24 23:07:55] WARNING[28785]: res_jabber.c:1391 aji_recv: Parsing
failure: Hook returned an error.
[Oct 24 23:07:55] WARNING[28785]: res_jabber.c:2742 aji_recv_loop:
JABBER: Got hook event.
[Oct 24 23:07:55] WARNING[28785]: res_jabber.c:2753 aji_recv_loop:
JABBER: socket read error

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Re: [asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice

[Stephen Reese]

On Sun, Oct 24, 2010 at 7:06 PM, Paul Belanger
 wrote:
> On Sun, Oct 24, 2010 at 6:23 PM, Stephen Reese  wrote:
>> Has anyone seen a how-to on getting Asterisk to work with Google Talk
>> and Google Voice?
>>
> I wrote one last week:
> http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/
>
> Also: http://www.davidvossel.com/?p=28
>

Paul,

It seems you were using a beta/SVN release for your example. Do the
following two packages need to be installed if using the stable 1.6.0
release before building from source? I ask as I am unable to dial out.

$ apt-get install libikesemel-dev
$ apt-get install libssl-dev

Secondly, do you know if the username/password are sent in clear text
to the Google?

Thanks,
Stephen

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[asterisk-users] Integrating Asterisk 1.8 with Google Talk and Google Voice

[Stephen Reese]

Evening,

Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?

Thanks

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[asterisk-users] Running SIP on non-standard ports

[Stephen Brown]

-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

I want to have the ability to have anonymous SIP calls hit my server but
I want to run it on different ports and create an SRV record for my
target domain.

My understanding of SIP is limited, but in a nutshell I want to
accomplish the following:

- - run SIP signaling on port 6200
- - create RTP ports on 6201-62XX

Do I really need 10k ports open for RTP!!?? I don't plan on doing more
than a 5-10 calls simultaneously, maybe less than that. Does each RTP
port represent one channel, or does it take two, one for each end
perhaps? Just trying to come up with a number...

And my assumption is also that I will only need to create an SRV record
for the SIP signaling, will I need to create SRV record(s) for the RTP
ports as well? I'm assuming the SIP signaling handles that instead?

Thanks in advance...

Stephen
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Re: [asterisk-users] Dual Atom mobo - call capacity

[Stephen Brown Jr]

Ditto I'm running a Supermicro Atom based dual core server and it's rock
solid!!!

These make excellent servers for Asterisk installation IMHO.


On Fri, Jun 11, 2010 at 05:42, --[ UxBoD ]--  wrote:

>
> - Original Message -
> > On Thu, 10 Jun 2010, Michelle Dupuis wrote:
> >
> > > I'm looking for a small formfactor mobo for an install that needs to
> > > handle 25 phone sets (no transcoding). I found a new dual atom
> > > 1.66GHz
> > > mobo - anyone know what kinds of call volume that will handle?
> >
> > On Thu, 10 Jun 2010, mgra...@mstvp.com wrote:
> >
> > > Based on comments from Ward Mundy during a recent VUC call I'd
> > > expect
> > > even a single CPU Atom system to handle that many phones in an
> > > office
> > > application. Perhaps there may be merit in dual CPU in more of a
> > > call
> > > center application.
> >
> > Assuming you're talking about something like the Atom 330...
> >
> > My guess is you will have plenty of horsepower for 25 phone sets --
> > probably even 25 simultaneous calls.
> >
> > The 330 is dual-core and hyper-threaded so it shows up as 4 CPUs in
> > "top."
> >
> > Asterisk is multi-threaded and should distribute the workload. Another
> > advantage is that if you have something CPU heavy like bzip2'ing your
> > database dump or compiling Asterisk from source, there are still
> > several
> > "CPUs" available for Asterisk.
> >
>
> I have a single rack server with a Atom 330 and 2GB RAM, six phones
> connected and probably a couple of simultaneous calls at one time.  This is
> how it looks at the moment:
>
> total   used   free sharedbuffers cached
> Mem:   20498561346480 703376  0 181920 990376
> -/+ buffers/cache: 1741841875672
> Swap:  4095992  04095992
>
> top - 10:41:59 up 12 days, 16:03,  1 user,  load average: 0.01, 0.00, 0.00
> Tasks: 122 total,   1 running, 121 sleeping,   0 stopped,   0 zombie
> Cpu0  :  0.0%us,  0.0%sy,  0.0%ni, 99.9%id,  0.0%wa,  0.0%hi,  0.0%si,
>  0.0%st
> Cpu1  :  0.0%us,  0.0%sy,  0.0%ni, 98.4%id,  0.0%wa,  1.5%hi,  0.0%si,
>  0.0%st
> Cpu2  :  0.1%us,  0.0%sy,  0.1%ni, 99.8%id,  0.0%wa,  0.0%hi,  0.0%si,
>  0.0%st
> Cpu3  :  0.1%us,  0.0%sy,  0.0%ni, 99.8%id,  0.0%wa,  0.0%hi,  0.0%si,
>  0.0%st
> Mem:   2049856k total,  1346232k used,   703624k free,   181920k buffers
> Swap:  4095992k total,0k used,  4095992k free,   990376k cached
>
> Have a TDM card in the server and also use G729 codec and Skype for
> Asterisk.
> --
> Thanks, Phil
>
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Re: [asterisk-users] Evaluating Asterisk

[Stephen Wingfield]

Ted

Funnily enough a Call Center we did for Legal Aid just asked to  
increase from 40 to 60 seats. This replaced the Nortel - the first big  
improvement was ease of configuration.


We have also set up a Directory Inquiry (yes - like 411) that does  
40,000 calls per day. Their reason to go ahead was they wanted more  
stability vis-a-vis (again) Nortel.


If you wish to move beyond 70 - 100 concurrent calls it can become  
more complicated on organising queues, particularly if there is call  
recording that eats resource and so spreads you to more than one server.


You also need to consider how to provide integrated statistics and the  
desktop applications for supervisors to maintain a proper management  
of agents.


Steve

 you welcome to call : 619 760 7772


On Apr 19, 2010, at 3:06 PM, Ted Foote wrote:

I am thinking of moving from a traditional PBX to an asterisk box.  
Many of my leadership group are skeptical of asterisk. So I was  
hoping to find a call center that is currently using this technology  
that would not mind spending some time on a conference call to  
address some concerns that my team has.


Thanks
Ted Foote
Allied Business Services, Inc.
616-741-0437


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Re: [asterisk-users] Maximum number of PRI calls on 1 asterisk box (no HW echo)

[Stephen Davies]

On 25 March 2010 02:42, James Lamanna  wrote:

> Hi,
> I have a TE410P T1 (1st gen) card and I'm seeing interesting errors of
> D-Channels going down and then coming back up (See below).
>
>
Read all the discussion about many spans - and I've run 16 E1 spans in one
box, and run 8 spans under 200+ concurrent calls.

Your 1st gen TE410 card is very old and I'd suggest to contact Digium about
a firmware upgrade or a hardware upgrade.

As for the spans going down:-

1) Make sure you are syncing your clock to your telco (span=1,1,0,...)
2) Make sure you are using the right IDE driver module for your chipset and
not the generic one
3) Avoid long runs (>25m?) of unscreened cable on the T1/E1 span

Regards,
Steve
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Re: [asterisk-users] Sending a hook flash to a DAHDI channel

[Stephen Brown]

> You only get past this point when this call is over.
That makes sense now that I think about it.
> Can the CO send Flash to the CPE?
I am the CO in this case :)

I started looking at the "D" option of the dial command, is there any 
way to send a DTMF digit followed by a hookflash at call origination? 
For example maybe something like this:
exten => 1234,1,Dial(DAHDI/2/D(2,someoptopiontohookflashhere)

Or is there an alternative method that I could use for what I am trying 
to accomplish?




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[asterisk-users] Sending a hook flash to a DAHDI channel

[Stephen Brown]

I've got a piece of CPE equipment that has an FXS port that I have tied 
to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a 
standard telephone connected to it, I get a dialtone. If I dial a digit, 
and send a hookflash, the device will provide a dialtone back for the 
next available channel on the device.

I'm trying to recreate this same behavior with Asterisk, however, it's 
not working. I'm not sure if it's me, the CPE device, Asterisk, my 
dialplan code or a combination of all. I am running Asterisk 1.6.0.21, 
FreePBX 2.6, and the latest version of DAHDI.

Here's the dialplan logic I am trying to execute, this is in 
/etc/asterisk/extensions_custom.conf:

[from-internal-custom]
exten => 1234,1,Dial(DAHDI/2); DAHDI channel 2, FXO
exten => 1234,n,SendDTMF(2)  ; I expect to send DTMF digit 2 after 
the channel answers?
exten => 1234,n,Wait(1)  ; I added a wait statement
exten => 1234,n,Flash()   ; Send the hookflash

After the hookflash is when I expect to receive dialtone from the CPE 
device, but it does not appear to ever execute the SendDTMF or Flash events:

[Feb 20 15:45:52] -- Starting simple switch on 'DAHDI/4-1'
[Feb 20 15:46:00] -- Executing [1...@from-internal:1] 
ResetCDR("DAHDI/4-1", "") in new stack
[Feb 20 15:46:00] -- Executing [1...@from-internal:2] 
Dial("DAHDI/4-1", "DAHDI/2") in new stack
[Feb 20 15:46:00] -- Called 2
[Feb 20 15:46:01] WARNING[27502]: chan_dahdi.c:2008 dahdi_enable_ec: 
Unable to enable echo cancellation on channel 2 (No such device)
[Feb 20 15:46:01] -- DAHDI/2-1 answered DAHDI/4-1
[Feb 20 15:46:01] -- Native bridging DAHDI/4-1 and DAHDI/2-1
[Feb 20 15:46:06] WARNING[27502]: chan_dahdi.c:2008 dahdi_enable_ec: 
Unable to enable echo cancellation on channel 2 (No such device)
[Feb 20 15:46:06] -- Executing [...@from-internal:1] 
Macro("DAHDI/4-1", "hangupcall") in new stack
[Feb 20 15:46:06] -- Executing [...@macro-hangupcall:1] 
GotoIf("DAHDI/4-1", "1?skiprg") in new stack
[Feb 20 15:46:06] -- Goto (macro-hangupcall,s,4)
[Feb 20 15:46:06] -- Executing [...@macro-hangupcall:4] 
GotoIf("DAHDI/4-1", "1?skipblkvm") in new stack
[Feb 20 15:46:06] -- Goto (macro-hangupcall,s,7)
[Feb 20 15:46:06] -- Executing [...@macro-hangupcall:7] 
GotoIf("DAHDI/4-1", "1?theend") in new stack
[Feb 20 15:46:06] -- Goto (macro-hangupcall,s,9)
[Feb 20 15:46:06] -- Executing [...@macro-hangupcall:9] 
Hangup("DAHDI/4-1", "") in new stack
[Feb 20 15:46:06]   == Spawn extension (macro-hangupcall, s, 9) exited 
non-zero on 'DAHDI/4-1' in macro 'hangupcall'
[Feb 20 15:46:06] -- Hungup 'DAHDI/2-1'
[Feb 20 15:46:06]   == Spawn extension (from-internal, 1234, 2) exited 
non-zero on 'DAHDI/4-1'
[Feb 20 15:46:06] -- Hungup 'DAHDI/4-1'

I've also tried it with this variation:
exten => 1234,1,Dial(DAHDI/2/D(2)); DAHDI channel 2, FXO send DTMF 
digit 2 before call gets bridged
exten => 1234,n,Wait(1)  ; Wait 1 second
exten => 1234,n,Flash()   ; Send the hookflash

This yields the results of dialing "2" to the CPE device, again I never 
see the Flash event get executed (as evidenced from the console, it's 
also possible my verbosity is set too low?)

Any help appreciated

Thanks,
Stephen




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Re: [asterisk-users] SIP tunnel

[Stephen Davies]

Problem is that the port 80 you are talking about is a TCP port.  Voip
(iax and rtp) use UDP

On 2/11/10, mosbah.abdelkader  wrote:
> Thank you Jamie for your good reply.
>
>
> It is a very good idea to hava the media and control transported over the
> same port with IAX protocol.
>
>
> The difficulty is in that the port is not well known by the network admins.
> It is usually blocked.
>
>
> My idea is to use a well know port like port 80 (that is not blocked). Skype
> for example uses this port.
>
>
> I need recommendations and help.
>
> Thanks.
>
> *--
> Please discover scientific miracles of CORAN
>
> http://www.55a.net/*
>

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Re: [asterisk-users] VERY HIGH LOAD AVERAGE: top - 10:27:57 up 199 days, 5:18, 2 users, load average: 67.75, 62.55, 55.75

[Stephen Davies]

On 9 February 2010 06:42, Muro, Sam  wrote:

> Hi Team
>
> Can someone advice me on how i can lower the load average on my asterisk
> server?
>
> dahdi-linux-2.1.0.4
> dahdi-tools-2.1.0.2
> libpri-1.4.10.1
> asterisk-1.4.25.1
>
> 2 X TE412P Digium cards on ISDN PRI
>
> Im using the system as an IVR without any transcoding or bridging
>
> **
> top - 10:27:57 up 199 days,  5:18,  2 users,  load average: 67.75, 62.55,
> 55.75
>

Hi Sam!

Are there any side-effects from the high load average?  The system doesn't
seem to be CPU or disk bound from the look of the CPU stats.  System %age is
high by way - software echo cancellaton?, and Asterisk is using a lot of cpu
which isn't suprising.

I'm guessing you are running 8 spans and 200+ calls into your IVR?

If the system is actually performing fine then I'd just say that there is
something about the Asterisk threads that makes them look runnable and that
accounts for the high load average.  Is the IVR an agi or fastagi or what? -
the code path may have a "spinlock" logic to it that means that many threads
are runnable but when scheduled just go back to sleep.  That would account
for high load average with lots of spare CPU.  If that's what is happening
then I wouldn't worry much more about it.

Regards,
Steve

PS: Alex - why the dig about ALL CAPS?  The post wasn't in caps?
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Re: [asterisk-users] Beginners Guide to setting up a Call Centre

[Stephen Davies]

2010/1/15 Doug Lytle 

> > Decide if you are going to be a zealot for your preferred approach
>
> That's a little harsh, wouldn't you say?  Do whatever your most
> comfortable with.  But, to call me and those like me a zealot, for
> offering advice that was asked for is a little off, in my opinion.
>
>
Hi Doug,

Maybe I read too much into the original poster's question, and I didn't mean
to be harsh.  But I used to get called in often here in South Africa to
sites where the "usual way" wasn't good enough for someone so they'd put the
whole system together the way they thought it should be done and in the
process bumped into all the subtle gotchas that are mostly worked out in the
standard builds.  Then discovered that its harder than they thought it would
be and PBX users are ungrateful b*ggers sometimes and they've walked away.
 Our efforts to recover these installs are always twice the work because
they are tainted by what went before.  But we hate to see failed Asterisk
projects so we try to get them right.

If your objective is to run a simple inbound call centre and get good
metrics into the bargain then a FreePBX-based ISO-install (Elastic,
AsteriskNow, Trixbox-CE, whathaveyou) plus Queuemetrics will have you up an
running in short order.

Build from the bare metal using your-own-install-of-your-preferred-distro
plus raw Asterisk plus dialplan from scratch plus DIY reportage and you'll
be working away after a month and cursing Asterisk.  Once you're an expert
then you may indeed be able to do a better job for your application than the
all-in-one distros.  But not first time.

So apologies to the poster if I read too much into the question, but this is
the sort of situation I thought of.

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

[Stephen Davies]

2010/1/15 Peter Childs 

> Elastix, Trixbox, or AsterixNow, or DIY (ie Ubuntu or whatever
> installed with OpenPBX, Asterix etc by hand)
>
> I've got a new server to run Asterix on and want to get it working
> quickly and yet be configurable in the future with out having to
> reisntall and start again regally.
>
> Currently no VoIP hardware but that will come once I prove the concept. I
> guess
>
> Oh the machine does not have a CD Rom Drive so a network/USB install
> would be nice.. But I guess I can open the case and plug one in
> for installation if I must!
> (Says he who has just installed Ubuntu over the network to check the
> computer works!)
>
>
Decide if you are going to be a zealot for your preferred approach - Ubuntu
and all that - or if you want a solution that works without tons of extra
work.  If you wisely decide that you want the latter, then get Elastix and
install it.  Buy QueueMetrics and install on your Elastix build.  Start
running your inbound call centre.

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

[Stephen Davies]

2010/1/12 Jeff LaCoursiere 

> That is so not true.  FreePBX has hooks in a million places to do custom
> dialplan stuff - I do it all the time.  I also link in custom AGI/AMI
> applications, custom provisioning, custom LCR, and am even working with
> one customer that has mastered making FreePBX multi-tenant.
>
> If you want to get your hands dirty there is plenty of dirt underneath
> FreePBX.  On the other hand, if you want a simple setup that is easily
> managed, the GUI is fantastic and saves a LOT of time.  And if you are a
> PHP programmer you can easily modify the operation of any part of it.
>
>
Preach it brother.  We take the same approach and have never had any
difficulty integrating our customisations into the FreePBX dialplan.  The
common structure makes it EASIER for my techies to work on systems that we
built and support.

On asterisk-users its traditional to be hard core and raw-dialplan and look
down on those who have projects to deliver and are happy to have the help.
 I'm not the insecure - each of you writing your raw dialplans runs some of
my code every time you run Asterisk.

Regards,
Steve
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Re: [asterisk-users] hints through a Local channel

[Stephen Davies]

What you are missing is the new state-interface parameter to AddQueueMember.

You can't use functions in a hint exten.

Steve

On 12/14/09, Lenz Emilitri  wrote:
> Hello all,
> I am trying to set up a dynamic channel to be used as an Agent dialer for a
> queue - you know, trying to replace AgentCallBackLogin for an Asterisk 1.6.
>
> I would like to do something like:
>
> [myagents]
> exten => XXX,1,Set(realchan=${DB(myagent/${EXTEN})})
> exten => XXX,n,Dial(${realchan},tT,60)
>
> This basically fetches the actual channel to be used for dialling and dials
> it. What I would like now is to make app_queue aware in advance of the state
> of each channel, something like:
>
> exten => 100,hint,SIP/705   (and this works)
>
> But more dynamical, so I would try and look up the actual channel in the
> AstDB, like:
>
> exten => XXX,hint,${DB(myagent/${EXTEN})}
>
> This does not seem to be working - is there a way to work around this issue?
>
> (I admit this is the fist time I'm trying to use devoce state and the
> related functions, so maybe there is a very simple slution right in front of
> my big nose and I'm not seeing it).
>
> Thanks a lot for your help,
> l.
>
>
> --
> Loway - home of QueueMetrics - http://queuemetrics.com
>

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Re: [asterisk-users] Trouble registering Cisco 7942

[Stephen Reese]

On Wed, Nov 11, 2009 at 9:34 PM, Warren Selby  wrote:
> The 7960 and 79x2 use different sip firmwares and as far a I have seen
> the 7960 does not have the same port issue the 7941/2 seems to have
> (which technically is not a problem, just an implementation of the sip
> protocol that you don't typically see).
>
> As to your issue, are you still seeing the same error messages in the
> ssh logs?  I haven't ever had to use the register with proxy settings
> in my configs, but I've only worked with the 79x1 series phones, not
> the x2.
>
> I've actually got a post up on my blog addressing setting up a 7941 in
> a situation similar to yours:
>
> http://www.selbytech.com/2009/10/setup-cisco-7941-or-7961-with-asterisk/
>
> In that post is a sanitized version of my conf file that I use on my
> own deskphone, if you'd like to download it and try it out with your
> setup.
>

My config is very similar though my only question is you have
registerWithProxy set to true though nothing defined. Was this a
sanitation mistake?








true


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Re: [asterisk-users] Can't configure Cisco 7942 avec factory reset

[Stephen Reese]

This is possible as I was just able to get the latest SIP firmware
loaded on my 7942. Make sure to follow the guide using the 7941 as the
SIP firmware differs from the 79x0 versions. Here's two links to help:

http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP

http://www.greenwireit.com/blog/2009/09/cisco-7961-and-7941-sip-configuration-sepmac-cnf-xml/

On Thu, Oct 22, 2009 at 10:48 AM, Olivier  wrote:
> Hi,
>
> (I think) I followed instructions here
> (http://www.voip-info.org/wiki/view/Firmware+issues+on+7940+-+7960   section
> "Notes added Nov 2005, revised May 2006:"
> at the bottom of the page) to factory reset a Cisco 7942 I wanted to
> configure to SIP firmware.
>
> When booting, I can see this requesting and obtaining file
> term42.default.loads from TFTP server.
> Then it would send a request (recognized as CDP request by Wireshark) a
> couple of times, then loop again asking for term42.default.loads file.
>
> My question is :
> Is it possible to upload a SIP firmware with a factory reset 7942, without
> any Call Manager ?
>
> Cheers
>
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Re: [asterisk-users] Trouble registering Cisco 7942

[Stephen Reese]

On Tue, Nov 10, 2009 at 10:13 AM, Warren Selby  wrote:
> In your sip.conf file, be sure to specify nat=no for the phone, even
> though the phone is behind a nat device. The cisco phones handle sip
> packets differently than the way asterisk expects, so you have to do
> this in order to make asterisk send the way the phone will accept.
>
>
>
> Thanks,
> --Warren Selby

Thanks, as a test I changed both a 7960 and 7942 both to "nat=no" the
latter being the one I'm having trouble registering. The 7960 then was
unable to register so I changed it back to "nat=yes". When I changed
the 7942 to "nat=no" and disabled registerWithProxy I can get a dial
tone but can't dial out due to the following:

SIP/2.0 407 Proxy Authentication Required

I tried re-enabling the proxy but then I get nothing as before.

>From the sip.conf I would assume that  would be the
same as the realm and the  statement for the phone be the same
as the domain?

realm=ns1.domain.net
domain=domain.net

Thanks again for any help.

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Re: [asterisk-users] Trouble registering Cisco 7942

[Stephen Reese]

On Sat, Nov 7, 2009 at 11:36 AM, Warren Selby  wrote:
> I think your featureLabel definition is wrong.
>
> On the login issue, ssh to the ip of the phone and login first with
> the user/pass you defined in the file (admin/123), then at the second
> login prompt use log/log. That should get you the log files which will
> show you your error.

Thanks for the insight. After you mentioned that the syntax of the XML
file may be wrong I looked around and found a more complete
configuration I could find since mine was a copy and paste special.
Using the new configuration the phone comes up but is unable register
I *think* it may be an issue with NAT. When the phone fires up for the
first time it tries to register for a while and the log didn't help
much so I took a peak at the asterisk logging. It seems like packets
are not getting back to the phone. I've enabled NAT in the
configuration similar to how the other phones are configured but no
dice. Note that the Asterisk device is not NATed but the phones are
behind a NAT device.

I get multiple of the following message in the phone:

ERR 16:40:16.273722 JVM: %REG send failure: REGISTER

On the asterisk server I keep getting NAT retries:

Retransmitting #4 (NAT) to 71.226.175.137:1026:
OPTIONS sip:1...@ip of NAT device:1027;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP ASTERISK IP:5060;branch=z9hG4bK53121c03;rport
From: "asterisk" ;tag=as5b0b32f5
To: 
Contact: 
Call-ID: 090e1e583f29f9f000dd30ff5719f...@209.251.157.91
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 10 Nov 2009 02:26:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

Below is the full XML config for the phone:


  SIP
  admin
  123
  

  M/D/Ya
  Eastern Standard/Daylight Time
  

  192.43.244.18
  directedbroadcast

  


  

  

  2000
  5060
  5061

Asterisk IP
  

 

  

  




Asterisk IP
5060
true
  
  
true
x--serviceuri-cfwdall
x-cisco-serviceuri-pickup
x-cisco-serviceuri-opickup
x-cisco-serviceuri-gpickup
x-cisco-serviceuri-meetme
x-cisco-serviceuri-abbrdial
false
2
true
true
2
2
0
true
  
  
6
10
180
3600
5
120
120
5
500
4000
70
false
None
  
  1
  false
  true
  false
  false
  g711ulaw
  101
  3
  avt
  false
  false
  3
  true
  IP outside of NAT Device
  ATLAS
  1
  true
  
10
  false
  16384
  32766
  

  9
  Line 102
  Asterisk IP
  5060
  102
  ATLAS
  
2
  
  3
  102
  pass
  false
  1
  *97
  4
  5
  102
  
true
false
false
true
  


5060
184
0
dialplan.xml
softkeys.xml
  
  

true
2
  
  
false
false
1
1
0
1
0
0
1
0
1

  
  
  
English_United_States
1
en_US
1.0.0.0-1
iso-8859-1
  
  United_States
  
United_States
64
1.0.0.0-1
  
  1
  
  
  
  
  
  
  
  96
  0
  96
  4
  0
  

  3804

  
  
  false


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Re: [asterisk-users] Trouble registering Cisco 7942

[Stephen Reese]

On Sat, Nov 7, 2009 at 12:56 AM, Warren Selby  wrote:
> That typically means you've got an error in your phone specific config file,
> the SEP[MAC].cnf.xml.
>
> You need to login to the phone via ssh and use the log/log login.  Once
> you've done that, look at the logs and see what line of the config is giving
> it grief.  Once you know that, you'll know what's causing the Unprovisioned
> message.

I set the username and password but am unable to log into the phone. I
provided an updated config below. I am prompted for the username and
password though.

Secondly should I be using IP or hostnames for the  and
 or does it not matter? Thanks


SIP
admin
123


   
  
 

   2000
   5060
   5061

SIPSERVER
 
  
   



   true
   x--serviceuri-cfwdall
   x-cisco-serviceuri-pickup
   x-cisco-serviceuri-opickup
   x-cisco-serviceuri-gpickup
   x-cisco-serviceuri-meetme
   x-cisco-serviceuri-abbrdial
   false
   2
   true
   true
   2
   2
   0
   true

 true
 172.16.2.1
 102

  
  9
  102
  102
  SIPSERVER
  5060
  102
  Atlas
  102
  PASS
false




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[asterisk-users] Suggestions for low level RTP stream generator?

[Stephen Davies]

Hi,
I need to build a simple, command-line method to generate a legal and
"perfect" RTP stream across a network link, and analyse it on the other side
and measure network performance.  Want to do this for a number of links and
over long periods.  I'm trying to characterise performance of various
available Internet links locally as input into a design project.

Asterisk isn't ideal for me because I just want a one-way stream and in the
case of things like Echo, Milliwatt, MusicOnHold etc the generated RTP
stream is "synchronised" to the incoming RTP and so issues in the one
direction affect the other.  And I'm just looking for a lean-and-mean
command line program!

rtptools from Columbia seems to be almost what I want, except that for
rtpsend I have to specify the desired RTP stream in more detail than I want.
 (I just want to generate a valid stream containing say "silence")

I also looked at rat (from mbone days) - it can generate a "tone test"

Can anyone suggest other old-time utilities that might help me?

Thanks,
Steve
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Re: [asterisk-users] DAHDI disconnect supervision timing

[Stephen Brown]

ting [...@macro-vm:5] NoOp("DAHDI/1-1", 
"Checking if ext 100 is enabled: ") in new stack
[Sep 25 13:34:56] -- Executing [...@macro-vm:6] GotoIf("DAHDI/1-1", 
"1?s-NOANSWER,1") in new stack
[Sep 25 13:34:56] -- Goto (macro-vm,s-NOANSWER,1)
[Sep 25 13:34:56] -- Executing [s-noans...@macro-vm:1] 
Macro("DAHDI/1-1", "get-vmcontext,100") in new stack
[Sep 25 13:34:56] -- Executing [...@macro-get-vmcontext:1] 
Set("DAHDI/1-1", "VMCONTEXT=default") in new stack
[Sep 25 13:34:56] -- Executing [...@macro-get-vmcontext:2] 
GotoIf("DAHDI/1-1", "0?200:300") in new stack
[Sep 25 13:34:56] -- Goto (macro-get-vmcontext,s,300)
[Sep 25 13:34:56] -- Executing [...@macro-get-vmcontext:300] 
NoOp("DAHDI/1-1", "") in new stack
[Sep 25 13:34:56] -- Executing [s-noans...@macro-vm:2] 
VoiceMail("DAHDI/1-1", "1...@default,u""") in new stack
[Sep 25 13:34:56] --  Playing 'vm-theperson.gsm' 
(language 'en')
[Sep 25 13:34:58] --  Playing 'digits/1.gsm' (language 'en')
[Sep 25 13:34:58] --  Playing 'digits/0.gsm' (language 'en')
[Sep 25 13:34:59] --  Playing 'digits/0.gsm' (language 'en')
[Sep 25 13:35:00] --  Playing 'vm-isunavail.gsm' 
(language 'en')
[Sep 25 13:35:01] --  Playing 'vm-intro.gsm' (language 'en')
[Sep 25 13:35:07] --  Playing 'beep.gsm' (language 'en')
[Sep 25 13:35:07] -- Recording the message
[Sep 25 13:35:07] -- x=0, open writing:  
/var/spool/asterisk/voicemail/default/100/tmp/Tjm8w6 format: wav49, 
0x86c7f18
[Sep 25 13:35:07] -- x=1, open writing:  
/var/spool/asterisk/voicemail/default/100/tmp/Tjm8w6 format: gsm, 0x865d768
[Sep 25 13:35:07] -- x=2, open writing:  
/var/spool/asterisk/voicemail/default/100/tmp/Tjm8w6 format: wav, 0x8683e70
[Sep 25 13:38:05] -- Recording automatically stopped after a silence 
of 10 seconds
[Sep 25 13:38:05] --  Playing 'auth-thankyou.gsm' 
(language 'en')
[Sep 25 13:38:06] -- Executing [s-noans...@macro-vm:3] 
Goto("DAHDI/1-1", "exit-SUCCESS,1") in new stack
[Sep 25 13:38:06] -- Goto (macro-vm,exit-SUCCESS,1)
[Sep 25 13:38:06] -- Executing [exit-succ...@macro-vm:1] 
GotoIf("DAHDI/1-1", "0?exit-RETURN,1") in new stack
[Sep 25 13:38:06] -- Executing [exit-succ...@macro-vm:2] 
Playback("DAHDI/1-1", "goodbye") in new stack
[Sep 25 13:38:06] --  Playing 'goodbye.ulaw' (language 'en')
[Sep 25 13:38:07] -- Executing [exit-succ...@macro-vm:3] 
Hangup("DAHDI/1-1", "") in new stack
[Sep 25 13:38:07]   == Spawn extension (macro-vm, exit-SUCCESS, 3) 
exited non-zero on 'DAHDI/1-1' in macro 'vm'
[Sep 25 13:38:07]   == Spawn extension (macro-exten-vm, s, 18) exited 
non-zero on 'DAHDI/1-1' in macro 'exten-vm'
[Sep 25 13:38:07]   == Spawn extension (from-did-direct, 100, 1) exited 
non-zero on 'DAHDI/1-1'
[Sep 25 13:38:07] -- Hungup 'DAHDI/1-1'

Then I am left with a nice 3 minute long message full of dialtone 
followed by the ever famous "howler" tone to indicate an off-hook phone :/

Any clues!!!??? I should also mention I am running FreePBX 2.5.2

Thanks,
Stephen



On 9/25/09 1:28 PM, Tzafrir Cohen wrote:
> On Fri, Sep 25, 2009 at 11:19:39AM -0400, Stephen Brown Jr wrote:
>
>> Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
>> install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
>> FXO port and 1 FXS port. I have a POTS line from my phone company attached
>> to the POTS line.
>>
>> I have asked for "disconnect supervision" to be provisioned on my line and
>> they claim to have added it. However, my scenario is as follows:
>>
>> I receive a call, if the caller hangs up before hitting voice mail, the
>> DAHDI channel is released as to be expected (evidenced from console
>> messaging)
>> If the call gets to voicemail and the caller hangs up during the greeting,
>> no hangup condition is ever detected and I am greeted with a useless
>> voicemail moments later.
>>  
> I don't understand this.
>
> Can you enable debug logging and provide log of a call that disconnect
> successfully and a log of a call that fails to disconnect?
>
>

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Re: [asterisk-users] DAHDI disconnect supervision timing

[Stephen Brown]

Have you been able to find a satisfiable config? My biggest headache is 
the useless voicemails being left if the caller hangs up during the 
greeting, otherwise it appears to work as intended.


Agree on the Zoloft, this is driving me nuts!

On 9/25/09 11:45 AM, Danny Nicholas wrote:


If you're really going to pursue this, I'd buy stock in Zoloft -- I've 
got a TDM400 and TDP410 and they both drive me nuts on POTS issues...




*From:* asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Stephen Brown

*Sent:* Friday, September 25, 2009 10:41 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] DAHDI disconnect supervision timing

I have, but I wanted to see if I could fix this problem before I 
started experimenting with that.


On 9/25/09 11:24 AM, Danny Nicholas wrote:

Have you looked into minimum message length and/or silence parameters?



*From:* asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com> 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Stephen Brown Jr

*Sent:* Friday, September 25, 2009 10:20 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] DAHDI disconnect supervision timing

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card 
with 1 FXO port and 1 FXS port. I have a POTS line from my phone 
company attached to the POTS line.


I have asked for "disconnect supervision" to be provisioned on my line 
and they claim to have added it. However, my scenario is as follows:


I receive a call, if the caller hangs up before hitting voice mail, 
the DAHDI channel is released as to be expected (evidenced from 
console messaging)
If the call gets to voicemail and the caller hangs up during the 
greeting, no hangup condition is ever detected and I am greeted with a 
useless voicemail moments later.


I am using kewlstart signaling etc. I came across this page from Digium:

http://kb.digium.com/entry/6/

This suggests adjusting a variable in zaptel.h, as I don't use zaptel, 
can this same logic be applied to DAHDI somewhere? My theory is that 
the "disconnect supervision" signal coming from the phone company may 
be less than 1000ms.


Desperately trying to fix this.

Thanks,
Stephen

  

  
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Re: [asterisk-users] DAHDI disconnect supervision timing

[Stephen Brown]

I've tried the hanguponpolarityswitch parameter as well with no success :(

Any clues where in the DAHDI code I might find reference to disconnect 
supervision timing?

On 9/25/09 11:39 AM, Martin wrote:
> find the code in dahdi and put printk so you can see in dmesg or
> /var/log/messages
> if that gets ever detected
>
> also you may try hanguponpolarityswitch=yes in chan_dahdi.conf
>
> Martin
>
>
> On Fri, Sep 25, 2009 at 10:19 AM, Stephen Brown Jr
>   wrote:
>
>> Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
>> install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
>> FXO port and 1 FXS port. I have a POTS line from my phone company attached
>> to the POTS line.
>>
>> I have asked for "disconnect supervision" to be provisioned on my line and
>> they claim to have added it. However, my scenario is as follows:
>>
>> I receive a call, if the caller hangs up before hitting voice mail, the
>> DAHDI channel is released as to be expected (evidenced from console
>> messaging)
>> If the call gets to voicemail and the caller hangs up during the greeting,
>> no hangup condition is ever detected and I am greeted with a useless
>> voicemail moments later.
>>
>> I am using kewlstart signaling etc. I came across this page from Digium:
>>
>> http://kb.digium.com/entry/6/
>>
>> This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can
>> this same logic be applied to DAHDI somewhere? My theory is that the
>> "disconnect supervision" signal coming from the phone company may be less
>> than 1000ms.
>>
>> Desperately trying to fix this.
>>
>> Thanks,
>> Stephen
>>
>>
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>>  
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Re: [asterisk-users] DAHDI disconnect supervision timing

[Stephen Brown]

I have, but I wanted to see if I could fix this problem before I started 
experimenting with that.


On 9/25/09 11:24 AM, Danny Nicholas wrote:


Have you looked into minimum message length and/or silence parameters?



*From:* asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Stephen Brown Jr

*Sent:* Friday, September 25, 2009 10:20 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] DAHDI disconnect supervision timing

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card 
with 1 FXO port and 1 FXS port. I have a POTS line from my phone 
company attached to the POTS line.


I have asked for "disconnect supervision" to be provisioned on my line 
and they claim to have added it. However, my scenario is as follows:


I receive a call, if the caller hangs up before hitting voice mail, 
the DAHDI channel is released as to be expected (evidenced from 
console messaging)
If the call gets to voicemail and the caller hangs up during the 
greeting, no hangup condition is ever detected and I am greeted with a 
useless voicemail moments later.


I am using kewlstart signaling etc. I came across this page from Digium:

http://kb.digium.com/entry/6/

This suggests adjusting a variable in zaptel.h, as I don't use zaptel, 
can this same logic be applied to DAHDI somewhere? My theory is that 
the "disconnect supervision" signal coming from the phone company may 
be less than 1000ms.


Desperately trying to fix this.

Thanks,
Stephen


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[asterisk-users] DAHDI disconnect supervision timing

[Stephen Brown Jr]

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6
install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1
FXO port and 1 FXS port. I have a POTS line from my phone company attached
to the POTS line.

I have asked for "disconnect supervision" to be provisioned on my line and
they claim to have added it. However, my scenario is as follows:

I receive a call, if the caller hangs up before hitting voice mail, the
DAHDI channel is released as to be expected (evidenced from console
messaging)
If the call gets to voicemail and the caller hangs up during the greeting,
no hangup condition is ever detected and I am greeted with a useless
voicemail moments later.

I am using kewlstart signaling etc. I came across this page from Digium:

http://kb.digium.com/entry/6/

This suggests adjusting a variable in zaptel.h, as I don't use zaptel, can
this same logic be applied to DAHDI somewhere? My theory is that the
"disconnect supervision" signal coming from the phone company may be less
than 1000ms.

Desperately trying to fix this.

Thanks,
Stephen
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Re: [asterisk-users] DAHDI hangup detection

[Stephen Brown]

Following up on this with some testing I have done this afternoon. I 
have lengthened the default ringtime to 25 seconds from 15, and have 
tried both kewlstart and loopstart. Kewlstart appears to be the most 
functional, if I hang up before it hits app_voicemail, it appears to 
work as intended, and I get this message in the console:

== Spawn extension (macro-dial, s, 7) exited non-zero on 'DAHDI/1-1' in 
macro 'dial'
== Spawn extension (macro-exten-vm, s, 9) exited non-zero on 'DAHDI/1-1' 
in macro 'exten-vm'
== Spawn extension (from-did-direct, 2001, 1) exited non-zero on 'DAHDI/1-1'
 -- Hungup 'DAHDI/1-1'

However, if I hangup while the voicemail greeting is playing, then it 
leaves a 0:34 second message with busy tone and I am notified with the 
new message.

Am I to assume the following?

- after 25 seconds, app_voicemail takes over and has no way of knowing 
the calling party has hung up, therefore recording dial/busy/offhook 
tone (whatever is present)

Is there any conceivable way to fix this? My fear is being bombarded 
with bogus voice mails if people call the number and decide to hang up 
while the voice mail greeting is playing.









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Re: [asterisk-users] DAHDI hangup detection

[Stephen Brown]

Does linetest.agi exist as part of the Asterisk install? I can't find it 
on my box. Also, does this detect dialtone?


Dialtone is being recorded (left as voicemail) without my original 
workarounds in place.


On 9/15/09 1:08 PM, Danny Nicholas wrote:


This is how I'd do it with the "old" dialplan (extensions.conf) --

- exten => s,1,dial.

- exten => s,n,Macro(voicemail-test,${EXTEN})

- exten => s,n,hangup

- [macro-voicemail-test]

- Exten => s,1,AGI(linetest.agi,${ARG1})

- Exten => s,n,gotoif($["${VAR}" = "AVAIL"]?hangup)

- Exten => s,n,voicemail...

- Exten => s,n(hangup),hangup

Linetest.agi would run an AMI session to see if DAHDI-X was in use and 
return a variable as AVAIL or undef.  If the line is in use, record 
voicemail, else hangup.




*From:* asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Stephen Brown Jr

*Sent:* Tuesday, September 15, 2009 11:52 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] DAHDI hangup detection

Ok on the workaround, how would I implement it? I'd like to give that 
a shot.


On Tue, Sep 15, 2009 at 12:23, Danny Nicholas <mailto:da...@debsinc.com>> wrote:


The issue is that POTS as a technology does not have Answer/Hangup
Supervision control (This is per the good folks at Digium).  Your local
Telco may or may not be of assistance, but a simpler workaround might 
be to

hangup on silence or query the line through AMI before transferring to
voicemail.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com>
[mailto:asterisk-users-boun...@lists.digium.com 
<mailto:asterisk-users-boun...@lists.digium.com>] On Behalf Of Stephen 
Brown

Sent: Tuesday, September 15, 2009 11:18 AM
To: asterisk-users@lists.digium.com 
<mailto:asterisk-users@lists.digium.com>

Subject: [asterisk-users] DAHDI hangup detection

I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled
from source on a Debian Lenny box, also running FreePBX 2.5.2. I also
have an OpenVOX TDM400 card installed with an FXO port on port 1, and an
FXS port on port 2. I have a POTS line installed and working on the FXO
port.

However I've encountered a weird problem that I can't seem to figure
out. I have incoming POTS line calls set to ring a SIP extension (a
cisco IP phone). It works, but here is the issue:

- If an inbound call comes in on the POTS line and hangs up before the
call hits voicemail or during the voicemail greeting, Asterisk does not
appear to detect this condition.
- As a result of this, a voicemail is being left with a dialtone and
notifying me

As a workaround, I have set busy detection in chan_dahdi.conf and set 12
seconds for a minimum voicemail message in voicemail.conf, but I must be
missing something or something is potentially broke. While this
workaround works, I don't suspect this behavior I am experiencing is
normal and I am having a hard time putting a finger on it.

Thanks for any replies.

Stephen

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Re: [asterisk-users] DAHDI hangup detection

[Stephen Brown Jr]

Ok on the workaround, how would I implement it? I'd like to give that a
shot.

On Tue, Sep 15, 2009 at 12:23, Danny Nicholas  wrote:

> The issue is that POTS as a technology does not have Answer/Hangup
> Supervision control (This is per the good folks at Digium).  Your local
> Telco may or may not be of assistance, but a simpler workaround might be to
> hangup on silence or query the line through AMI before transferring to
> voicemail.
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen
> Brown
> Sent: Tuesday, September 15, 2009 11:18 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] DAHDI hangup detection
>
> I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled
> from source on a Debian Lenny box, also running FreePBX 2.5.2. I also
> have an OpenVOX TDM400 card installed with an FXO port on port 1, and an
> FXS port on port 2. I have a POTS line installed and working on the FXO
> port.
>
> However I've encountered a weird problem that I can't seem to figure
> out. I have incoming POTS line calls set to ring a SIP extension (a
> cisco IP phone). It works, but here is the issue:
>
> - If an inbound call comes in on the POTS line and hangs up before the
> call hits voicemail or during the voicemail greeting, Asterisk does not
> appear to detect this condition.
> - As a result of this, a voicemail is being left with a dialtone and
> notifying me
>
> As a workaround, I have set busy detection in chan_dahdi.conf and set 12
> seconds for a minimum voicemail message in voicemail.conf, but I must be
> missing something or something is potentially broke. While this
> workaround works, I don't suspect this behavior I am experiencing is
> normal and I am having a hard time putting a finger on it.
>
> Thanks for any replies.
>
> Stephen
>
>
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[asterisk-users] DAHDI hangup detection

[Stephen Brown]

I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled 
from source on a Debian Lenny box, also running FreePBX 2.5.2. I also 
have an OpenVOX TDM400 card installed with an FXO port on port 1, and an 
FXS port on port 2. I have a POTS line installed and working on the FXO 
port.

However I've encountered a weird problem that I can't seem to figure 
out. I have incoming POTS line calls set to ring a SIP extension (a 
cisco IP phone). It works, but here is the issue:

- If an inbound call comes in on the POTS line and hangs up before the 
call hits voicemail or during the voicemail greeting, Asterisk does not 
appear to detect this condition.
- As a result of this, a voicemail is being left with a dialtone and 
notifying me

As a workaround, I have set busy detection in chan_dahdi.conf and set 12 
seconds for a minimum voicemail message in voicemail.conf, but I must be 
missing something or something is potentially broke. While this 
workaround works, I don't suspect this behavior I am experiencing is 
normal and I am having a hard time putting a finger on it.

Thanks for any replies.

Stephen


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Re: [asterisk-users] G.722 problems with IAX

[Stephen Davies]

2009/9/9 Armin Schindler 

> >> No, I didn't miss that. See my text.
> >> I mentioned this because I think this might be the reason of the problem
> >> and
> >> the incorrect handling in jitterbuffer, if it is the jitterbuffer. It is
> >> just a guess, since everything else seems to work good.
> >> The question is why does G.722 via IAX has problems.
> >> Is anyone using it and can say it works in his setup?
>


Hi,

I'm not sure if Steve Kann is still around the project, but if not, I'm
familiar with chan_iax2.c and mostly familiar with the iax2 jitter buffer so
I might have a go at fixing the problem.  Will you open a bug on the
bugs.digium.com bug tracker.

I did do a test from a SNOM820 (yum) via an IAX trunk with jitter buffer and
got the same nasty jerky audio.  This is a recent checkout of branch-1.4.

Regards,
Steve Davies
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Re: [asterisk-users] Should digium build a 2FXO / 2FXS 4-port daughterboard?

[Stephen Davies]

2009/9/9 Karl Fife 

> ...of course you need one of these to dial SIP URI's or navigate IVR's from
> the rotary mechanism.
> http://www.oldphoneworks.com/rotatone-pulse-to-tone-converter.html
>
>
On Asterisk I don't think that's true.  At least for IVRs on the local
Asterisk box, Asterisk will process a rotary dialled "4" just as if you'd
pressed 4 on a DTMF phoneset.

I guess if you want to operate a remote IVR via the dial phone you'd need
the converter.

Are you in the US?  Hope you use loadzone=us-old.

Steve
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Re: [asterisk-users] Asterisk remote calls with low bandwith and high latency

[Stephen Davies]

2009/9/8 James Mutuku 

> I have 2 sites. One(Site 1) has an asterisk PBx and the Other(site 2) has 2
> remote soft phones. The latency btw both sites is btw 500ms-700ms.  I know
> this is a shot in the dark...but are there ways of improving the voice
> quality for the remote calls(btw site 1 and site 2), Other than increasing
> bandwidth?
>
>
Hi,

The latency will result in long delay on the call and you can't do anything
about that.  I suppose its a VSAT link?

With respect to the bandwidth all you can do is carefully select a low
bandwidth codec.  This will depend on the codecs that your soft phone
supports.  Speex can be tuned to use little bandwidth.  G729 can be used
with licensed codecs from Digium.  Rumour is that G.723 is technically
possible with some googling and provided that you have some sort of licence
to use the G723 patent.

Regards,
Steve Davies
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Re: [asterisk-users] Inquiry:Problem with Call Parking

[Stephen Davies]

In any event, the real problem is probably that you forgot to 'include
=> parkedcalls' in your dialplan.

Steve

On 9/2/09, Lyle Giese  wrote:
> And now that the whole world of Asterisk has your sip user ids and
> passwords, you should change all of the passwords that are in that file
> and yes, change the passwords in all your phones.
>
> Lyle Giese
> LCR Computer Services, Inc.
>
> hadi motamedi wrote:
>> Thank you for your reply . Please find attached my Asterisk sip.conf .
>> Can you please let me know what modifications are needed ?
>> Regards
>> H.Motamedi
>>
>>
>>
>> On Tue, Sep 1, 2009 at 5:55 AM, Lee, John (Sydney)
>> mailto:john@compuware.com>> wrote:
>>
>> Just a quick guess - is it because you did not program your
>> Polycom digit plan properly in sip.cfg?
>>
>> 
>> From: asterisk-users-boun...@lists.digium.com
>> 
>> [mailto:asterisk-users-boun...@lists.digium.com
>> ] On Behalf Of
>> hadi motamedi
>> Sent: Tuesday, 1 September 2009 2:39 PM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] Inquiry:Problem with Call Parking
>>
>> Dear All
>> Can you please do me favor and let me know what is the problem
>> with my Asterisk call parking as it is not functioning correctly
>> on my Asterisk ? Please find attached my "features.conf" .
>> According to my configuration , the subscriber needs to press hash
>> (pound) key and dial 700 to initiate the transfer . We tried but
>> it didn't get through on our Asterisk . Can you please let me know
>> what extra config needs to be done for putting it into operation ?
>> Regards
>> H.Motamedi
>>
>>
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>

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Re: [asterisk-users] I find this incomprehensible ?!

[Stephen Davies]

2009/8/30 jonas kellens 

>  I am totally not understanding this :
>
> My IAX.conf :
> register => BOX-YOCAN:pas...@remote_asterisk_ip 
>
> On remote Asterisk :
> *CLI> [Aug 30 20:37:07] -- Registered IAX2 'BOX-YOCAN' (AUTHENTICATED)
> at ip:4569
>
> So this is normal... Now the following :
>
> [remoteasterisk]
> type=peer
> host=ip remote asterisk
> auth=md5
> secret=passwd
>
>

I think you are missing a username=BOX-YOCAN in the [remoteasterik] peer
entry.

Steve
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[asterisk-users] Analog FXO or IAX DIDS for new facility?

[Stephen Fierbaugh (PBT)]

I am a Linux sysadmin who has been tasked with developing the phone 
system for our nonprofit's new US headquarters building.  We cannot 
bring our legacy phone system with us, so I am building this completely 
from scratch.  I have already read "Asterisk: The Future of Telephony" 
and done a fair amount of googling.  I am completely sold on Asterisk, 
and the new building's phones will be a mix of SIP handsets and softphones.

I am confused about one thing:  Should we be getting a block of analog 
circuits from the local telco (probably AT&T), connected to the server's 
FXO cards for in-bound and out-bound POTS calls; or should we get a 
block of DIDS numbers from one of the plethora of providers available 
over the Internet, and then have our server connect POTS calls by IAX to 
the DIDS provider?

We are unsure whether we are going to have separate numbers for everyone 
in the organization, or just 1 US phone number, with everyone in the org 
having their own extension number.  That probably largely depends upon cost.

We will have 75 people in the building.  We have no data on call 
patterns or usage (because our legacy system belongs to our current 
facilities host), but we currently have 4 lines for 35 people and on 
unusual occasions they all get busy.

An additional consideration is that we also have 300 other people 
scattered literally world-wide, and the next logical future step is to 
start providing VOIP links for them, as well.

Thanks in advance for your advice.  Any other suggestions, such as # of 
lines sizing info or reputable DIDS vendors (if that's the answer) are 
also appreciated.

-- 
Sincerely Yours,
Stephen P. Fierbaugh
Pioneer Bible Translators
Pronounced: "Fire" as in hot, "Bah" as in humbug!

"" John 3:16 in over 3,000 
languages.


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Re: [asterisk-users] Dialplan step that I do not have

[Stephen Davies]

2009/7/21 Jim Dickenson 


>
> How can the first step of the extension be a playback when I do a verbose?
>
>

Because you have a "exten =>" line that matches *9901 in the empl context or
some context that is included into the empl context "before" (above)
dorecord.

Steve
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Re: [asterisk-users] Suggest Multi-tenant Predictive Dialer ?

[Stephen Wingfield]

Kashif

This changes things. We can do, but this is hardly a simple off-the- 
shelf.

I will call you midday Tuesday if I might

Steve
On Jun 15, 2009, at 3:46 PM, Kashif Naeem wrote:


Hello All,

We have a requirement of multi-tenant Predictive Dialer which we can  
sell to multiple call centers. Each call center will have saperate  
interface for setting up campaigns and Reporting. Please suggest  
some solution or let us know if have it to sell ?


Regards,
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com

Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766

Email: kas...@haditelecom.com
MSN: kashif__na...@hotmail.com
Gmail: meet.kas...@gmail.com
Skype: kashif.naeem

302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.
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Re: [asterisk-users] LoadAvg , Codec and Bandwidth Utilisation

[Stephen Davies]

Hii,
Looking at this, your problem appears to be that you are diskbound.  Note
the 60% "wait" time.

Use "hdparm -t" to find out the throughput of each of your disks.  If its
not 20MB/sec or more then you need to look into the drivers you are using
for your disk.  If you do have good disk throughput and still stuck with
high wait time then you need to look into why you have so much disk IO going
on.

By the way - there's no big problem with an Asterisk box with a load average
of 4.  I've seen 10 or so without audible issues.  Obviously depends on
available CPUs and what exactly is happening on the box.

Steve


2009/5/1 David @ULC 

>
> [root]# top
> top - 20:19:40 up 53 min,  1 user,  load average: 9.54, 7.85, 6.44
> Tasks: 224 total,   1 running, 223 sleeping,   0 stopped,   0 zombie
> Cpu(s):  7.5%us,  3.8%sy,  0.0%ni, 28.6%id, 59.1%wa,  0.2%hi,  0.8%si,
>  0.0%st
>
>   PID USER  PR  NI  VIRT  RES  SHR S %CPU %MEMTIME+  COMMAND
>  2835 root  15   0 41252  14m 4932 S5  0.4   1:08.82 asterisk
>  2456 mysql 15   0  154m  28m 4480 S2  0.8   0:42.05 mysqld
>  2995 root  16   0 12380 7220 2740 S1  0.2   0:21.56 AST_update.pl
> 10700 root  15   0  2304 1088  788 R1  0.0   0:00.03 top
>   413 root  10  -5 000 D0  0.0   0:02.54 kjournald
>   428 apache15   0 58168 6460 2952 S0  0.2   0:00.10 httpd
>  3004 root  15   0 11096 5804 2640 S0  0.2   0:03.12
> AST_VDauto_dial
>  6725 root  15   0 12644 7400 2600 S0  0.2   0:00.90
> AST_CRON_audio_
> 10428 root  18   0 21536  15m 2600 S0  0.4   0:00.28
> AST_CRON_audio_
> 13471 apache15   0 58168 6780 3260 S0  0.2   0:00.50 httpd
> 17253 root  15   0 13248 6696 1456 S0  0.2   0:00.30 FastAGI_log.pl
> 29074 apache15   0 58168 6692 3172 S0  0.2   0:00.38 httpd
> 30454 root  16   0 21536  15m 2596 S0  0.4   0:00.65
> AST_CRON_audio_
> 1 root  15   0  2044  664  572 S0  0.0   0:00.58 init
> 2 root  RT   0 000 S0  0.0   0:00.02 migration/0
> 3 root  34  19 000 S0  0.0   0:00.00 ksoftirqd/0
> 4 root  RT   0 000 S0  0.0   0:00.00 watchdog/0
> 5 root  RT   0 000 S0  0.0   0:00.01 migration/1
> 6 root  34  19 000 S0  0.0   0:00.00 ksoftirqd/1
>
>
>
>
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Re: [asterisk-users] SIP trunk with > 250 lines

[Stephen Davies]

Hi,

We have a customer who used a strong quad-core Xeon box to convert up
to 800 simultneous calls from SIP to IAX and trunk them to another
box.

So your requirement doesn't look like a big problem.

Steve

On 3/24/09, Christian Victor  wrote:
> Hi!
>
> A customer of mine wants to connect an asterisk system with 240 to 480 lines
> to a PSTN switch. To save the costs for E1 cards and the corresponding E1
> mainlines he wants to connect the system to the switch by a SIP trunk.
>
> Phones will be connected to the server through the same SIP trunk as this
> will be some kind of a "hosted pbx".
>
> Given he finds a provider wich has this much SIP capacity and IP bandwith
> and no codec conversion is needed - do you think this is possible with pure
> asterisk on a decent system? Is there anything I shoudl watch out for?
>
> Your help is much appreciated!
>
> Chris
>

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Re: [asterisk-users] Special Information Tones

[Stephen Davies]

Hi,

Are you sure that Verizon amswers the call?  They should play that
message as 'early media' without answering, after which they cam clear
the call with an appropriate cause code.

That would work for you and still give callers the audible ,essage they want.

Steve

On 3/20/09, drew einhorn  wrote:
> I'm having a problem with Verizon Wireless.
>
> I would be extremely surprised if I was the only one having this problem.
>
> It seems to me that Verizon Wireless might be able to use one of the
> Special Information Tones to allow us to solve the problem.
>
> But I really do not whether my suggestion is compliant with the ITU-T
> standards.
>
> Perhaps someone can give me an expert opinion on whether I should try
> to get Verizon to implement my suggestion.
>
> First I'll describe the problem.
>
> I'm trying to implement Single Number Reach.  For example, when a call
> comes in to one of my DIDs, it simultaneously rings on a couple
> extensions in my home office and a couple of Verizon Wireless cell
> phone numbers.  Everything works just the way it is supposed to if the
> cell phones are powered up, and within the range of a cell tower.
>
> The problem is if a cellphone is turned off, or out of range and
> unable to talk to a cell tower, Verizon is unable to find the
> cellphone on their network, Verizon answers the call and plays a
> recorded message, instead of allowing the number to continue ringing,
> and allowing one of the voip extensions, or another cellphone to
> answer the call.
>
> Verizon really wants to get rid of the call as quickly as possible to
> free up their equipment to handle other calls.
>
> Unfortunately we spend a lot of time in rural areas where there is no
> cell tower to talk to.  In that case we really someone else to pick up
> the call.
>
> I'm hoping that if Verizon would precede the voice message with one of
> the Special Information Tones, we could recognize the fact that the
> call has not really been answer, and continue to ring on the other
> lines.
>
> Two questions.  1) would the approach be compliant with ITU-T
> standards?  2) Assuming that it is, and we can convince Verizon to
> implement this. How difficult would it be to configure asterisk to
> handle this as I suggest?
>
> --
> Drew Einhorn
>
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