date:20090923

On Tue, Sep 22, 2009 at 07:43:51PM -0500, Martin wrote:
 I do not know if fonebridge would work here since it sends/receives
 the ~2 Mbps (for each circuit/port)
 of data over ethernet ... constantly. That could choke the USB ...

Ethernet has frames. While I'm not exactly sure how ethernet over USB
works and how TDM over Ethernet (MF) works, I would speculate that it is
far from flooding the USB bus.

-- 
   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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] New Xorcom FXS USB Bank is not loading firmware

On Wed, Sep 23, 2009 at 08:14:41AM +0200, Loic Didelot wrote:
 Thank you for the information. The tarball compiles fine. Except that it
 has no qozap module.

This is a snapshow of the upstream Zaptel tarball, and hence does not
include qozap. It should be simple to use hat tarball in the bristuff
installation or to apply the relevant patches from bristuff to this
tarball.

Note that in latest DAHDI, wcb4xxp supports all the devices supported by
qozap.

-- 
   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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] New Xorcom FXS USB Bank is not loading firmware

On Wed, Sep 23, 2009 at 08:37:19AM +0200, Loic Didelot wrote:
 Hi Tzafrir,
 I just compiled the tarball, but now there seem to be some problems with
 the script lszaptel.
 
 
 Can't call method is_twinstar on unblessed reference
 at /usr/local/share/perl/5.8.8/Zaptel/Hardware/USB.pm line 108.

Is usbfs mounted under /pruc/bus/usb ?

-- 
   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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk on a Beagleboard?

2009-09-23 Thread Jeff LaCoursiere


On Wed, 23 Sep 2009, Tzafrir Cohen wrote:

 On Tue, Sep 22, 2009 at 07:43:51PM -0500, Martin wrote:
 I do not know if fonebridge would work here since it sends/receives
 the ~2 Mbps (for each circuit/port)
 of data over ethernet ... constantly. That could choke the USB ...

 Ethernet has frames. While I'm not exactly sure how ethernet over USB
 works and how TDM over Ethernet (MF) works, I would speculate that it is
 far from flooding the USB bus.


Even USB 1.1 was 12Mbps.  Should be plenty of room for a mere 24 channels 
of ulaw :)

j


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Simple dialplan issue

On Tue, Sep 22, 2009 at 05:32:13PM -1000, Julian Yap wrote:
 I have an issue where a particular dialplan works but another doesn't.  I'm
 not sure why.  To me they look identical and it has me stumped.
 
 This works:
 [to-test]
 exten = _X., 1, SetCallerPres(allowed)
 exten = _X., 2, Monitor(wav,/tmp/test-${UNIQUEID},mb)
 exten = _X., 3, Ringing
 exten = _X., 4, Dial(SIP/9...@a-test,20,ro)
 exten = _X., 5, GotoIf($[${DIALSTATUS} = ANSWER]?9)
 exten = _X., 6, GotoIf($[${DIALSTATUS} = NOANSWER]?7)
 exten = _X., 7, Dial(SIP/9...@a-test2,20,ro)
 exten = _X., 8, GotoIf($[${DIALSTATUS} = ANSWER]?9)
 exten = _X., 9, Hangup

Does this actually work? With the extra space after the ','?

-- 
   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

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] RTPAUDIOQOS

2009-09-23 Thread Mindaugas Kezys

Thank you for answer. It was very informative, I put it in our wiki if you
don't mind.

Regards,
Mindaugas Kezys
http://www.kolmisoft.com
VoIP Billing and Routing Solutions


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: 2009 m. rugsėjo 22 d. 20:13
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] RTPAUDIOQOS

On Tuesday 22 September 2009 10:42:44 Johann Steinwendtner wrote:
 Mindaugas Kezys schrieb:
  Check this link:
  http://wiki.kolmisoft.com/index.php/RTPAUDIOQOS_Demystified

 In the given example:

*ssrc=254186206;themssrc=2024901615;lp=0;rxjitter=0.020917;rxcount=150;txji
tter=0.00;txcount=83;rlp=0;rtt=14818.715000* How do I interprete the
 jitter value ? Is the value 0.020917 good ? Bad ? Is there a unit behind
 this value ?

It's a ratio of out-of-order (jittered) to in-order packets, calculated
progressively.  Due to the progressive calculation, it's not exactly 3/147,
in
this case, but it's close enough to know that 3 packets were received
out-of-order.  The closer the value is to 0, the better.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] SIP/WiFi handsets?

2009-09-23 Thread Ken D'Ambrosio

Anyone know of any *portable* SIP/WiFi handsets?  Looking for a decent
price:quality ratio, of possible.  Keep seeing handsets for Vonage, etc.,
in Best Buy and the like, but I imagine it's locked to Vonage, and can't
be re-appropriated.

Thanks!

-Ken


-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Asterisk success

2009-09-23 Thread Thomas Mullins

For the last 10 years we have had some Tandberg video conferencing units
spread across our WAN. To make a long story short we had to work with
another entity to allow access into those units with H.323.  Calls from
public networks have never worked to those devices with private IP's.  

Over the summer, my coworker and I built an * VOIP system for one of our
high schools.  The need came up for another public school division to
call us, and we said sure, we can do it with *.  The next day, the other
school division called in just fine, no big deal.

This may not seem like a big deal, but this a huge star for Asterisk in
our school system. 

Shane


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk on a Beagleboard?

2009-09-23 Thread Martin

Even PCI has 133MB/s ... so what ? Also isn't USB only target ? It
doesn't do DMA ...
so it might be same as PCI Target chips that slow down the CPU

TDMoE has to have those frames on time all the time forever ...
these ethernet frames are sent both ways every 1ms
that might be (or not) too much load on the small CPU

loose a few frames or deliver late and your voice TDMoE won't work right

I just speculate here

Martin

On Wed, Sep 23, 2009 at 7:56 AM, Jeff LaCoursiere j...@jeff.net wrote:

 On Wed, 23 Sep 2009, Tzafrir Cohen wrote:

 On Tue, Sep 22, 2009 at 07:43:51PM -0500, Martin wrote:
 I do not know if fonebridge would work here since it sends/receives
 the ~2 Mbps (for each circuit/port)
 of data over ethernet ... constantly. That could choke the USB ...

 Ethernet has frames. While I'm not exactly sure how ethernet over USB
 works and how TDM over Ethernet (MF) works, I would speculate that it is
 far from flooding the USB bus.


 Even USB 1.1 was 12Mbps.  Should be plenty of room for a mere 24 channels
 of ulaw :)

 j


 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP/WiFi handsets?

2009-09-23 Thread Jason Baker

Ken,
I did lots of research on this for my VoIP deployment here where I
work. We have a huge manufacturing floor and all the supervisors have
wifi phones. We evetually settled on the Polycom Spectralink 8002. A
nice rugged little phone with great sound quality and some good
features. We use a managed switch to create seamless wifi coverage over
all of our AP's. Provisioning the phone is pretty easy, but no web
browser if you were planning on using the phone to travel with, some
hotels require login for internet access.

I also tried a clamshell wifi SIP phone by D-Link. This phone actually
works really well, but we had some minor issues with it so we went with
all Spectralink phones. But the D-Link phone would be good choice if
you plan to take your wifi phone on the road.

I also tested the Linksys WIP330 which I thought was a terrible phone.
Very difficult to use.

Good luck.

http://www.polycom.com/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html
http://www.dlink.com/products/?pid=485
http://www.voipsupply.com/linksys-wip330-na

Jason Baker
IT
Coordinator

Glastender, Inc.
5400 North Michigan Road
Saginaw,
Michigan 48604 USA
Phone: 989.752.4275 ext.
228
Fax: 989.752.4276
www.glastender.com

Ken D'Ambrosio wrote:

Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent
price:quality ratio, of possible. Keep seeing handsets for Vonage, etc.,
in Best Buy and the like, but I imagine it's locked to Vonage, and can't
be re-appropriated.

Thanks!

-Ken

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] MeetMe in Macro

2009-09-23 Thread Juan Cardoza

I need the same information, did you find that information Anahi???

Best regards

Juan Cardoza

 

De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Anahi Ludueña
Enviado el: Miércoles, 16 de Septiembre de 2009 09:49 a.m.
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] MeetMe in Macro

 

Thanks Miguel, It was my mistake.
So, my question is:
if I want to call the GoSub application from the Originate Action (using
AMI), what I need to put in the context parameter? The GoSub will jump to a
special context.
Thanks,




  _  

Date: Wed, 16 Sep 2009 09:34:31 -0500
From: mmol...@millenium.com.co
To: asterisk-...@lists.digium.com; asterisk-users@lists.digium.com
Subject: Re: [asterisk-dev] MeetMe in Macro

Hi,

I didn't notice on my first answer, but we are on the -dev list and this is
not related to asterisk code developing. I will answer you on the -users
list, so we can continue the discussion there.

Cheers,

-- 


Ing. Miguel Molina


Grupo de Tecnología


Millenium Phone Center





  _  

¿Quieres que tus amigos de Messenger sigan tus movimientos de Facebook?
¡Conéctalos ya! http://www.vivelive.com/feedfacebook 



Teleperformance values: Integrity - Respect - Professionalism - Innovation - 
Commitment

The information contained in this communication is privileged and confidential. 
 The content is intended only for the use of the individual or entity named 
above. If the reader of this message is not the intended recipient, you are 
hereby notified that any dissemination, distribution or copying of this 
communication is strictly prohibited.  If you have received this communication 
in error, please notify me immediately by telephone or e-mail, and delete this 
message from your systems.
Please consider the environmental impact of needlessly printing this e-mail.___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] MeetMe in Macro

You can go to  any context from AMI by using Context: context, priority: 1

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan Cardoza
Sent: Wednesday, September 23, 2009 10:10 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] MeetMe in Macro

 

I need the same information, did you find that information Anahi???

Best regards

Juan Cardoza

 

De: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Anahi Ludueña
Enviado el: Miércoles, 16 de Septiembre de 2009 09:49 a.m.
Para: asterisk-users@lists.digium.com
Asunto: [asterisk-users] MeetMe in Macro

 

Thanks Miguel, It was my mistake.
So, my question is:
if I want to call the GoSub application from the Originate Action (using
AMI), what I need to put in the context parameter? The GoSub will jump to a
special context.
Thanks,



  _  

Date: Wed, 16 Sep 2009 09:34:31 -0500
From: mmol...@millenium.com.co
To: asterisk-...@lists.digium.com; asterisk-users@lists.digium.com
Subject: Re: [asterisk-dev] MeetMe in Macro

Hi,

I didn't notice on my first answer, but we are on the -dev list and this is
not related to asterisk code developing. I will answer you on the -users
list, so we can continue the discussion there.

Cheers,

-- 






Ing. Miguel Molina






Grupo de Tecnología






Millenium Phone Center

 

  _  

¿Quieres que tus amigos de Messenger sigan tus movimientos de Facebook?
¡Conéctalos ya! http://www.vivelive.com/feedfacebook 


Teleperformance values: Integrity - Respect - Professionalism - Innovation 
Commitment 

The information contained in this communication is privileged and
confidential.  The content is intended only for the use of the individual or
entity named above. If the reader of this message is not the intended
recipient, you are hereby notified that any dissemination, distribution or
copying of this communication is strictly prohibited.  If you have received
this communication in error, please notify me immediately by telephone or
e-mail, and delete this message from your systems. 
  http://webmail.tpmex.com/users/imagenes/Pictures/earth.png Please
consider the environmental impact of needlessly printing this e-mail. 

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 1.6.0.5: I need a really simple analog SendFax dialplan

2009-09-23 Thread Martin

well maybe it doesn't work as it should ... anyways like the other
poster said that's not the way you use it ...

either call the sendfax app directly or use Originate / call file spooling...

BTW there should be an Originate app executable from dialplan ...
But since there's none you can do

exten = _X.,n,System(echo -e Channel: SIP/num...@gateway\\ncontext:
send\\nExtension: s\\nPriority: 1\\n 
/var/spool/asterisk/outgoing/call-${UNIQUEID})

and at send,s,1 call sendfax

Martin

On Wed, Sep 23, 2009 at 1:44 AM, sean darcy seandar...@gmail.com wrote:
 Martin wrote:
 from RTFM

 G(context^exten^pri) - If the call is answered, transfer the calling party to
            the specified priority and the called party to the
 specified priority+1.
            Optionally, an extension, or extension and context may be 
 specified.
            Otherwise, the current extension is used. You cannot use
 any additional
            action post answer options in conjunction with this option.


 your priority+1 is Hangup ...

 is that it ?

 Martin

 On Tue, Sep 22, 2009 at 7:32 PM, sean darcy seandar...@gmail.com wrote:
 Using Digium fax I've tried a simple dialplan:

 '8447' = 1. Answer()                       [pbx_config]
           2. Set(CALLERID(num)=xxxyyy)              [pbx_config]
           3. Dial(DAHDI/g0/1bbbccc,,G(send))        [pbx_config]
 [send]    4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif) [pbx_config]
           5. HangUp()

 But I doesn't work. It executes hangup:

 DAHDI/g0/1bbbccc,,G(send)) in new stack
    -- Requested transfer capability: 0x00 - SPEECH
    -- Called g0/1bbbccc
    -- DAHDI/1-1 is proceeding passing it to SIP/173-b55f7448
    -- DAHDI/1-1 is ringing
    -- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448
    -- DAHDI/1-1 is making progress passing it to SIP/173-b55f7448
    -- DAHDI/1-1 answered SIP/173-b55f7448
    -- Executing [8...@outbound-fax:4] SendFAX(SIP/173-b55f7448,
 /var/spool/asterisk/fax/20090922_1301.tif) in new stack
    -- Channel 'SIP/173-b55f7448' sending fax
 '/var/spool/asterisk/fax/20090922_1301.tif'
    -- Channel 'SIP/173-b55f7448' fax session '16' started
    -- Executing [8...@outbound-fax:5] Hangup(DAHDI/1-1, ) in new stack
  == Spawn extension (outbound-fax, 8447, 5) exited non-zero on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'
    -- Channel 'SIP/173-b55f7448' fax session '16', [ 000.003512 ],
 STAT_EVT_STRT_TX       st: IDLE         rt: IDLENSTX



 So why does it hangup before completing the fax?

 Does anyone have a SendFax dialplan that works for an analog channel?

 Thanks for any help.

 sean



 Well, I had RTFM :) And I've tried this, without success:

  '8447' = 1. Answer()                       [pbx_config]
            2. Set(CALLERID(num)=xxxyyy)              [pbx_config]
            3. Dial(DAHDI/g0/1bbbccc,,G(send))        [pbx_config]
  [send]    4. SendFax(/var/spool/asterisk/fax/20090922_1301.tif)
 [pbx_config]
            5. Wait()                  [pbx_config]
            6. HangUp()                            [pbx_config]

 The dialplan didn't wait. Also tried without the HangUp(), but the
 dialplan just fell through. What should priority 5 (priority + 1) be?

 Does anyone use SendFax for analog faxing?

 sean


 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 1.6.0.5: I need a really simple analog SendFax dialplan

2009-09-23 Thread Jared Smith

On Wed, 2009-09-23 at 10:17 -0500, Martin wrote:
 BTW there should be an Originate app executable from dialplan ...
 But since there's none you can do

There is an Originate application, but it's only available in newer
versions of Asterisk.  (I know I have it on the 1.6.2 branch, but I
don't remember if it's available on the 1.6.1 branch.  I know it's not
available on the 1.6.0 branch.)


-- 
Jared Smith
Training Manager
Digium, Inc.


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Error When Using Postgresql Schema With Realtime Sip

2009-09-23 Thread Tilghman Lesher

On Wednesday 23 September 2009 05:49:54 stephen.hindma...@bt.com wrote:
 I am using asterisk 1.6.1.6 and have been setting up a system to use a
 Postgresql database as the realtime DB via the ODBC route. I have got
 extensions and voicemail working but am having trouble with SIP

 The problem seems to be with using a schema. If I put the table sip in
 the schema foo then I add this entry to extconfig.conf

 sippeers = odbc,psqldb,foo.sip

 Restart everything and then try to register a client. The registration
 fails and I get this set of messages in the log

snip

 So it seems to be something to do with having the schema name in the
 table name. But as I say I have already got extensions and voicemail
 working, and they both uses schemas, so it seems to be peculiar to SIP.

 Does anybody have any ideas about what it might be?

Yep, I never bothered to include support for specifying either the catalog or
the schema, since I've never had reason to use either one.  Please report this
issue on the bugtracker (https://issues.asterisk.org) and I'll get a patch up
straightaway, but I'll need your testing to ensure the patch works.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] About bug 13115

2009-09-23 Thread Miguel Molina

Hi everyone,

Does someone know why the solution for bug 13115 
(https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=13115) 
was made only for trunk? Having that this bug went solved more than a 
year ago, it means that all the 1.6.X.X branches have it applied 
already? Can this be backported to the 1.4 branch? This could be another 
good reason to upgrade to 1.6.0.16 after I do some good testing...

I'm having this issue on asterisk 1.4.22, this is a quick grep for ERROR 
of the last lines of /var/log/asterisk/messages file:

[Sep 23 10:51:45] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad66bcf8!
[Sep 23 10:51:45] ERROR[26172] chan_sip.c: SIP transaction failed: 
57d7bbb010e5b40105f0fe5b43644...@192.168.130.25
[Sep 23 10:51:45] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad66bcf8!
[Sep 23 10:51:45] ERROR[26172] chan_sip.c: SIP transaction failed: 
57d7bbb010e5b40105f0fe5b43644...@192.168.130.25
[Sep 23 10:51:47] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/512-ab3bec38!
[Sep 23 10:51:47] ERROR[26172] chan_sip.c: SIP transaction failed: 
ODc5N2NiNWQxMmJmNjhhMjE2ZjUwYmNhMmZkOGQ0ZjY.
[Sep 23 10:51:50] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-09c5a400!
[Sep 23 10:51:50] ERROR[26172] chan_sip.c: SIP transaction failed: 
36959dc8707e78a3660a1bdc22a01...@192.168.130.26
[Sep 23 10:52:48] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-09d5cae8!
[Sep 23 10:52:48] ERROR[26172] chan_sip.c: SIP transaction failed: 
595fdb4e33e821df52940e716f1d4...@192.168.130.26
[Sep 23 10:52:48] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-09d5cae8!
[Sep 23 10:52:48] ERROR[26172] chan_sip.c: SIP transaction failed: 
595fdb4e33e821df52940e716f1d4...@192.168.130.26
[Sep 23 10:52:50] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-09d5cae8!
[Sep 23 10:52:50] ERROR[26172] chan_sip.c: SIP transaction failed: 
595fdb4e33e821df52940e716f1d4...@192.168.130.26
[Sep 23 10:53:01] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad51b0c0!
[Sep 23 10:53:01] ERROR[26172] chan_sip.c: SIP transaction failed: 
0f90abe5604baaed4e52aff222143...@192.168.130.25
[Sep 23 10:53:02] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad51b0c0!
[Sep 23 10:53:02] ERROR[26172] chan_sip.c: SIP transaction failed: 
0f90abe5604baaed4e52aff222143...@192.168.130.25
[Sep 23 10:53:03] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad51b0c0!
[Sep 23 10:53:03] ERROR[26172] chan_sip.c: SIP transaction failed: 
0f90abe5604baaed4e52aff222143...@192.168.130.25
[Sep 23 10:53:04] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad51b0c0!
[Sep 23 10:53:04] ERROR[26172] chan_sip.c: SIP transaction failed: 
0f90abe5604baaed4e52aff222143...@192.168.130.25
[Sep 23 10:53:08] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad51b0c0!
[Sep 23 10:53:09] ERROR[26172] chan_sip.c: SIP transaction failed: 
0f90abe5604baaed4e52aff222143...@192.168.130.25
[Sep 23 10:53:11] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ab3c2bb0!
[Sep 23 10:53:11] ERROR[26172] chan_sip.c: SIP transaction failed: 
5a950b092204a55a686b4e1b0c0a4...@192.168.130.25
[Sep 23 10:55:48] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad56a9a0!
[Sep 23 10:55:48] ERROR[26172] chan_sip.c: SIP transaction failed: 
3779df8e48834898534a09c576389...@192.168.130.25
[Sep 23 10:55:50] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad606448!
[Sep 23 10:55:50] ERROR[26172] chan_sip.c: SIP transaction failed: 
0fd7662648a103195b9d01a047e7e...@192.168.130.25
[Sep 23 10:55:53] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad56a9a0!
[Sep 23 10:55:53] ERROR[26172] chan_sip.c: SIP transaction failed: 
3779df8e48834898534a09c576389...@192.168.130.25
[Sep 23 10:55:54] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad56a9a0!
[Sep 23 10:55:54] ERROR[26172] chan_sip.c: SIP transaction failed: 
3779df8e48834898534a09c576389...@192.168.130.25
[Sep 23 10:56:01] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/TRUNK_SWITCH4PRI-ad56a9a0!
[Sep 23 10:56:01] ERROR[26172] chan_sip.c: SIP transaction failed: 
3779df8e48834898534a09c576389...@192.168.130.25
[Sep 23 10:58:32] ERROR[26172] chan_sip.c: We could NOT get the channel 
lock for SIP/200-ad136a78!
[Sep 23 10:58:32] ERROR[26172] chan_sip.c: SIP transaction failed: 
NWExZjI5NDM2ZjQ4N2U5OGMzM2VmZGNiYjk5MzQyMGI.

On the time where the error appeared, the CPU consumption of the 
asterisk was sky high and were in trouble because of choppy calls.

Re: [asterisk-users] Error When Using Postgresql Schema With Realtime Sip

2009-09-23 Thread Roderick A. Anderson

Tilghman Lesher wrote:
 On Wednesday 23 September 2009 05:49:54 stephen.hindma...@bt.com wrote:
 I am using asterisk 1.6.1.6 and have been setting up a system to use a
 Postgresql database as the realtime DB via the ODBC route. I have got
 extensions and voicemail working but am having trouble with SIP

 The problem seems to be with using a schema. If I put the table sip in
 the schema foo then I add this entry to extconfig.conf

 sippeers = odbc,psqldb,foo.sip

 Restart everything and then try to register a client. The registration
 fails and I get this set of messages in the log
 
 snip
 
 So it seems to be something to do with having the schema name in the
 table name. But as I say I have already got extensions and voicemail
 working, and they both uses schemas, so it seems to be peculiar to SIP.

 Does anybody have any ideas about what it might be?
 
 Yep, I never bothered to include support for specifying either the catalog or
 the schema, since I've never had reason to use either one.  Please report this
 issue on the bugtracker (https://issues.asterisk.org) and I'll get a patch up
 straightaway, but I'll need your testing to ensure the patch works.

++  But I won't be able to test for awhile.

Stephen.  As a test/work-around/option you could try setting the 
search_path for the user connecting to the database.

This has worked for me with RT and LedgerSMB.


\\||/
Rod
-- 


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP/WiFi handsets?

2009-09-23 Thread mgraves

I had a good experience with that Polycom/Spectralink phone. Very rugged
as you say. The experience did highlight the weaknesses in consumer
Wifi AP, which reinforced my commitment to continue using DECT around my
office.

Michael

Original Message
Subject: Re: [asterisk-users] SIP/WiFi handsets?
From: Jason Baker jba...@glastender.com
Date: Wed, September 23, 2009 10:02 am
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com

Ken,
I did lots of research on this for my VoIP deployment here where I work. We
have a huge manufacturing floor and all the supervisors have wifi phones. We
evetually settled on the Polycom Spectralink 8002. A nice rugged little phone
with great sound quality and some good features. We use a managed switch to
create seamless wifi coverage over all of our AP's. Provisioning the phone is
pretty easy, but no web browser if you were planning on using the phone to
travel with, some hotels require login for internet access.

I also tried a clamshell wifi SIP phone by D-Link. This phone actually works
really well, but we had some minor issues with it so we went with all
Spectralink phones. But the D-Link phone would be good choice if you plan to
take your wifi phone on the road.

I also tested the Linksys WIP330 which I thought was a terrible phone. Very
difficult to use.

Good luck.

http://www.polycom.com/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html
http://www.dlink.com/products/?pid=485
http://www.voipsupply.com/linksys-wip330-na
Jason Baker
IT Coordinator
Glastender, Inc.
5400 North Michigan Road
Saginaw, Michigan 48604 USA
Phone: 989.752.4275 ext. 228
Fax: 989.752.4276
www.glastender.com

Ken D'Ambrosio wrote: Anyone know of any *portable* SIP/WiFi handsets?
Looking for a decent
price:quality ratio, of possible. Keep seeing handsets for Vonage, etc.,
in Best Buy and the like, but I imagine it's locked to Vonage, and can't
be re-appropriated.

Thanks!

-Kenhr___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP/WiFi handsets?

2009-09-23 Thread Vinícius Fontes

- Jason Baker jba...@glastender.com escreveu:

Ken,
I did lots of research on this for my VoIP deployment here where I
work. We have a huge manufacturing floor and all the supervisors have
wifi phones. We evetually settled on the Polycom Spectralink 8002. A
nice rugged little phone with great sound quality and some good
features. We use a managed switch to create seamless wifi coverage
over all of our AP's. Provisioning the phone is pretty easy, but no
web browser if you were planning on using the phone to travel with,
some hotels require login for internet access.

I also tried a clamshell wifi SIP phone by D-Link. This phone actually
works really well, but we had some minor issues with it so we went
with all Spectralink phones. But the D-Link phone would be good choice
if you plan to take your wifi phone on the road.

I also tested the Linksys WIP330 which I thought was a terrible phone.
Very difficult to use.

Good luck.

http://www.polycom.com/products/voice/wireless_solutions/wifi_communications/handsets/spectralink_8002_wireless.html
http://www.dlink.com/products/?pid=485
http://www.voipsupply.com/linksys-wip330-na

Jason Baker
IT Coordinator Glastender, Inc.
5400 North Michigan Road
Saginaw, Michigan 48604 USA
Phone: 989.752.4275 ext. 228
Fax: 989.752.4276
www.glastender.com

Ken D'Ambrosio wrote:

Anyone know of any *portable* SIP/WiFi handsets? Looking for a decent
price:quality ratio, of possible. Keep seeing handsets for Vonage,
etc.,
in Best Buy and the like, but I imagine it's locked to Vonage, and
can't
be re-appropriated.

Thanks!

-Ken
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

I second that. Spectralink 8002 phones are very good, specially when using a
managed wifi solution like the 3Com WX1200.

The only thing you must pay attention is no matter what kind of access point
you have, they *must* support WMM or else the phones won't work at all.

Vinícius Fontes
www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] 1.6.0.5: I need a really simple analog SendFax dialplan

2009-09-23 Thread Martin

Well 1.6.2 is not yet released - it's rc2  now of course the app
is somewhere ... since it's very easy to code ...
actually it should have been added at the time when originate was
added to CLI ... it's a pity someone who added cli originate
did not think about writing a few more lines for originate app

Martin

On Wed, Sep 23, 2009 at 11:00 AM, Jared Smith jsm...@digium.com wrote:
 On Wed, 2009-09-23 at 10:17 -0500, Martin wrote:
 BTW there should be an Originate app executable from dialplan ...
 But since there's none you can do

 There is an Originate application, but it's only available in newer
 versions of Asterisk.  (I know I have it on the 1.6.2 branch, but I
 don't remember if it's available on the 1.6.1 branch.  I know it's not
 available on the 1.6.0 branch.)


 --
 Jared Smith
 Training Manager
 Digium, Inc.


 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Testers Wanted for IMAP Voicemail patch

2009-09-23 Thread Noah Miller

Hi All -

At Leif's suggestion, I'm soliciting testers for a patch to IMAP voicemail.

Currently, when asterisk checks for voicemails in an IMAP folder, it
only looks for messages in the same context and with the same
voicemail box number as the person dialing in to VoicemailMain().  I
believe this artificially limits what can be done with IMAP voicemail.
 For example, I'd like to have an administrator who can drag and drop
messages using an IMAP client from his/her voicemail account to other
users' voicemail accounts.  This is not possible with the current
implementation of IMAP voicemail.

The patch under this bug:

https://issues.asterisk.org/view.php?id=15670

changes the VoicemailMain() app to look for any voicemail messages
regardless of what context or user the message was originally created
for.

I'd love to see this make it into some version of asterisk sooner
rather than later.  Comments and suggestions are welcome.

FYI: The patch is incredibly simple and small so stability issues
should not be a concern.


Thanks!
Noah

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk on a Beagleboard?

2009-09-23 Thread astgroups

- Original Message - 
From: Martin asteriskl...@callthem.info 
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com 
Sent: Wednesday, September 23, 2009 11:01:04 AM GMT -05:00 US/Canada Eastern 
Subject: Re: [asterisk-users] Asterisk on a Beagleboard? 

Even PCI has 133MB/s ... so what ? Also isn't USB only target ? It 
doesn't do DMA ... 
so it might be same as PCI Target chips that slow down the CPU 

TDMoE has to have those frames on time all the time forever ... 
these ethernet frames are sent both ways every 1ms 
that might be (or not) too much load on the small CPU 

loose a few frames or deliver late and your voice TDMoE won't work right 

I just speculate here 

Martin 

On Wed, Sep 23, 2009 at 7:56 AM, Jeff LaCoursiere j...@jeff.net wrote: 

 On Wed, 23 Sep 2009, Tzafrir Cohen wrote: 

 On Tue, Sep 22, 2009 at 07:43:51PM -0500, Martin wrote: 
 I do not know if fonebridge would work here since it sends/receives 
 the ~2 Mbps (for each circuit/port) 
 of data over ethernet ... constantly. That could choke the USB ... 

 Ethernet has frames. While I'm not exactly sure how ethernet over USB 
 works and how TDM over Ethernet (MF) works, I would speculate that it is 
 far from flooding the USB bus. 

 Even USB 1.1 was 12Mbps. Should be plenty of room for a mere 24 channels 
 of ulaw :) 

 j 

The test we did was actually with 2x T1s worth of calls (48 uLaw calls) on the 
Beagleboard using the Dual port fonebridge. 
I'm not suggesting this would be a good production quality system. I think a 
native Ethernet connection and not via a USB adapter would be more efficient 
but the CPU was able to handle the call volume no problem. 

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] SIP/WiFi handsets?

2009-09-23 Thread Alex Samad

On Wed, Sep 23, 2009 at 09:39:09AM -0700, mgra...@mstvp.com wrote:
I had a good experience with that Polycom/Spectralink phone. Very rugged
as you say. The experience did highlight the weaknesses in consumer
Wifi AP, which reinforced my commitment to continue using DECT around my
office.

I concur, I settle on the snom M3, but i did not have any requirements
to leave the office with the device. DECT seems to drain the battery a
lot less then Wifi

Michael

Ken,
I did lots of research on this for my VoIP deployment here where I work.
We have a huge manufacturing floor and all the supervisors have wifi
phones. We evetually settled on the Polycom Spectralink 8002. A nice rugged
little phone with great sound quality and some good features. We use a
managed switch to create seamless wifi coverage over all of our AP's.
Provisioning the phone is pretty easy, but no web browser if you were
planning on using the phone to travel with, some hotels require login for
internet access.

I also tried a clamshell wifi SIP phone by D-Link. This phone actually
works really well, but we had some minor issues with it so we went with all
Spectralink phones. But the D-Link phone would be good choice if you plan
to take your wifi phone on the road.

I also tested the Linksys WIP330 which I thought was a terrible phone.
Very difficult to use.

Good luck.

Thanks!

-Kenhr___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

--
And, most importantly, Alma Powell, secretary of Colin Powell, is with us.

- George W. Bush
01/30/2003
Washington, DC

signature.asc
Description: Digital signature
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] Parking - How to transfer the other party to a given slot

Hi,

I'm having trouble to figure out how I could implement this feature :
When on call with a contact, local operator would dial a sequence which
would park the remote party to a specific parking slot, among the hundred of
existing slots.
(to each extension, a single specific parking slot is attached and there are
too many extensions to dedicate BLF or short DTMF sequence to each) .

Example:
Operator receives a call from 0123456789.
He talks to remote party and then decides the call is for extension 1234.
As extension 1234 is busy at the moment, Operator forwards the incoming call
to slot 11234, typing *911234, for instance.
The person using extension 1234 would see that slot 11234 is busy and would
try to shorten ongoing call.


Should I use features.conf's dynamic features for that (to allow a specific
DTMF sequence while on call) ?
Then how can I let Operator type digits after *91 prefix ? Should I use
Incomplete() application ?

Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Parking - How to transfer the other party to agiven slot

This stands to be corrected, but for your purpose, a dynamic conference is
preferable to a parking lot.  The Park application is designed to
sequentially use/reuse a series of lots.  By transferring the caller to
conference 11234, you would be able to have the agent pick up the call by
going to conference 11234.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, September 23, 2009 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Parking - How to transfer the other party to
agiven slot

 

Hi,

I'm having trouble to figure out how I could implement this feature :
When on call with a contact, local operator would dial a sequence which
would park the remote party to a specific parking slot, among the hundred of
existing slots.
(to each extension, a single specific parking slot is attached and there are
too many extensions to dedicate BLF or short DTMF sequence to each) .

Example:
Operator receives a call from 0123456789.
He talks to remote party and then decides the call is for extension 1234.
As extension 1234 is busy at the moment, Operator forwards the incoming call
to slot 11234, typing *911234, for instance.
The person using extension 1234 would see that slot 11234 is busy and would
try to shorten ongoing call.


Should I use features.conf's dynamic features for that (to allow a specific
DTMF sequence while on call) ?
Then how can I let Operator type digits after *91 prefix ? Should I use
Incomplete() application ?

Regards

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Parking - How to transfer the other party to agiven slot

2009/9/23 Danny Nicholas da...@debsinc.com

  This stands to be corrected, but for your purpose, a dynamic conference
 is preferable to a parking lot.  The Park application is designed to
 sequentially use/reuse a series of “lots”.  By transferring the caller to
 conference 11234, you would be able to have the agent pick up the call by
 going to conference 11234.

Yes, I think I like this idea ...

How do you transfer the remote party to conference 11234 ?
(Please, apologize if this question seems stupid but I'm really a newbie on
this topic).

Is it easy to mimic parking lot timeout feature (to be certain a caller is
not left alone in a dynamic conference) ?


  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
 *Sent:* Wednesday, September 23, 2009 2:32 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Parking - How to transfer the other party to
 agiven slot



 Hi,

 I'm having trouble to figure out how I could implement this feature :
 When on call with a contact, local operator would dial a sequence which
 would park the remote party to a specific parking slot, among the hundred of
 existing slots.
 (to each extension, a single specific parking slot is attached and there
 are too many extensions to dedicate BLF or short DTMF sequence to each) .

 Example:
 Operator receives a call from 0123456789. Call mydialer:0123456789
 He talks to remote party and then decides the call is for extension 1234.
 As extension 1234 is busy at the moment, Operator forwards the incoming
 call to slot 11234, typing *911234, for instance.
 The person using extension 1234 would see that slot 11234 is busy and would
 try to shorten ongoing call.


 Should I use features.conf's dynamic features for that (to allow a specific
 DTMF sequence while on call) ?
 Then how can I let Operator type digits after *91 prefix ? Should I use
 Incomplete() application ?

 Regards

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Parking - How to transfer the other party toagiven slot

Here's a snippet from a reply from Jared Smith (Digium, Huntsville AL) -
untested

exten = 11234,1,Set(TIMEOUT(absolute)=60) 

exten = 11234,n,MeetMe(11234,d1M)

 

This should create a dynamic room 11234 and send the caller to it for 60
seconds.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, September 23, 2009 2:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Parking - How to transfer the other party
toagiven slot

 

 

2009/9/23 Danny Nicholas da...@debsinc.com

This stands to be corrected, but for your purpose, a dynamic conference is
preferable to a parking lot.  The Park application is designed to
sequentially use/reuse a series of lots.  By transferring the caller to
conference 11234, you would be able to have the agent pick up the call by
going to conference 11234.

Yes, I think I like this idea ...

How do you transfer the remote party to conference 11234 ?
(Please, apologize if this question seems stupid but I'm really a newbie on
this topic).

Is it easy to mimic parking lot timeout feature (to be certain a caller is
not left alone in a dynamic conference) ?

 


  _  


From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Wednesday, September 23, 2009 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Parking - How to transfer the other party to
agiven slot

 

Hi,

I'm having trouble to figure out how I could implement this feature :
When on call with a contact, local operator would dial a sequence which
would park the remote party to a specific parking slot, among the hundred of
existing slots.
(to each extension, a single specific parking slot is attached and there are
too many extensions to dedicate BLF or short DTMF sequence to each) .

Example:
Operator receives a call from 0123456789. Call mydialer:0123456789 
He talks to remote party and then decides the call is for extension 1234.
As extension 1234 is busy at the moment, Operator forwards the incoming call
to slot 11234, typing *911234, for instance.
The person using extension 1234 would see that slot 11234 is busy and would
try to shorten ongoing call.


Should I use features.conf's dynamic features for that (to allow a specific
DTMF sequence while on call) ?
Then how can I let Operator type digits after *91 prefix ? Should I use
Incomplete() application ?

Regards


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] MeetMe in Macro

2009-09-23 Thread Anahi Ludueña


Hi Juan, I didn't use the GoSub application, I put the name of the context in 
the Originate and the variables and their values in the Variable field.
See http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate.
Good luck!






Anahi Ludueña
 From: jcard...@tpmex.com
To: asterisk-users@lists.digium.com
Date: Wed, 23 Sep 2009 10:09:52 -0500
Subject: Re: [asterisk-users] MeetMe in Macro




















I need the same information, did you find that information
Anahi???

Best regards

Juan Cardoza

 





De:
asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] En nombre de Anahi
Ludueña

Enviado el: Miércoles, 16 de Septiembre de 2009 09:49 a.m.

Para: asterisk-users@lists.digium.com

Asunto: [asterisk-users] MeetMe in Macro





 

Thanks Miguel, It was my mistake.

So, my question is:

if I want to call the GoSub application from the Originate
Action (using AMI), what I need to put in the context parameter? The GoSub will
jump to a special context.

Thanks,













Date:
Wed, 16 Sep 2009 09:34:31 -0500

From: mmol...@millenium.com.co

To: asterisk-...@lists.digium.com; asterisk-users@lists.digium.com

Subject: Re: [asterisk-dev] MeetMe in Macro



Hi,



I didn't notice on my first answer, but we are on the -dev list and this is not
related to asterisk code developing. I will answer you on the -users list, so
we can continue the discussion there.



Cheers,

-- 

Ing. Miguel Molina

Grupo de Tecnología

Millenium Phone Center













¿Quieres
que tus amigos de Messenger sigan tus movimientos de Facebook? ¡Conéctalos ya!


































Teleperformance values:





 





Integrity - Respect - Professionalism - Innovation – Commitment







The information

contained in this communication is privileged and confidential.  The

content is intended only for the use of the individual or entity named above.

If the reader of this message is not the intended recipient, you are hereby

notified that any dissemination, distribution or copying of this communication

is strictly prohibited.  If you have received this communication in error,

please notify me immediately by telephone or e-mail, and delete this message

from your systems. 










 

 Please consider the

environmental impact of needlessly printing this e-mail. 

  
_
Descubre todas las formas en que puedes estar en contacto con amigos y 
familiares.
http://www.microsoft.com/windows/windowslive/default.aspx___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Parking - How to transfer the other party toagiven slot

2009-09-23 Thread John A. Sullivan III

Won't that hangup the call after 60 seconds? - John

On Wed, 2009-09-23 at 15:22 -0500, Danny Nicholas wrote:
 Here’s a snippet from a reply from Jared Smith (Digium, Huntsville AL)
 - untested
 
 exten = 11234,1,Set(TIMEOUT(absolute)=60) 
 
 exten = 11234,n,MeetMe(11234,d1M)
 
  
 
 This should create a dynamic room 11234 and send the caller to it for
 60 seconds.
 
  
 

 __
 From:asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
 Sent: Wednesday, September 23, 2009 2:59 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Parking - How to transfer the other
 party toagiven slot
 
 
  
 
  
 
 2009/9/23 Danny Nicholas da...@debsinc.com
 
 This stands to be corrected, but for your purpose, a dynamic
 conference is preferable to a parking lot.  The Park application is
 designed to sequentially use/reuse a series of “lots”.  By
 transferring the caller to conference 11234, you would be able to have
 the agent pick up the call by going to conference 11234.
 
 
 Yes, I think I like this idea ...
 
 How do you transfer the remote party to conference 11234 ?
 (Please, apologize if this question seems stupid but I'm really a
 newbie on this topic).
 
 Is it easy to mimic parking lot timeout feature (to be certain a
 caller is not left alone in a dynamic conference) ?
 
 
  
 

 __
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
 Olivier
 Sent: Wednesday, September 23, 2009 2:32 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Parking - How to transfer the other
 party to agiven slot
 
 
  
 
 Hi,
 
 I'm having trouble to figure out how I could implement this
 feature :
 When on call with a contact, local operator would dial a
 sequence which would park the remote party to a specific
 parking slot, among the hundred of existing slots.
 (to each extension, a single specific parking slot is attached
 and there are too many extensions to dedicate BLF or short
 DTMF sequence to each) .
 
 Example:
 Operator receives a call from 0123456789. Call
 He talks to remote party and then decides the call is for
 extension 1234.
 As extension 1234 is busy at the moment, Operator forwards the
 incoming call to slot 11234, typing *911234, for instance.
 The person using extension 1234 would see that slot 11234 is
 busy and would try to shorten ongoing call.
 
 
 Should I use features.conf's dynamic features for that (to
 allow a specific DTMF sequence while on call) ?
 Then how can I let Operator type digits after *91 prefix ?
 Should I use Incomplete() application ?
 
 Regards
 
 
 
 ___
 -- Bandwidth and Colocation Provided by
 http://www.api-digital.com --
 
 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users
 
  
 
 
 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 
 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society


___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Parking - How to transfer the otherparty toagiven slot

I SAID it was untested...  I tried to look up this thread in my emails, but
that repository has about 8K messages.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John A.
Sullivan III
Sent: Wednesday, September 23, 2009 4:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Parking - How to transfer the otherparty
toagiven slot

Won't that hangup the call after 60 seconds? - John

On Wed, 2009-09-23 at 15:22 -0500, Danny Nicholas wrote:
 Here's a snippet from a reply from Jared Smith (Digium, Huntsville AL)
 - untested

 exten = 11234,1,Set(TIMEOUT(absolute)=60) 

 exten = 11234,n,MeetMe(11234,d1M)

 This should create a dynamic room 11234 and send the caller to it for
 60 seconds.

 __
 From:asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
 Sent: Wednesday, September 23, 2009 2:59 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] Parking - How to transfer the other
 party toagiven slot

 2009/9/23 Danny Nicholas da...@debsinc.com

 This stands to be corrected, but for your purpose, a dynamic
 conference is preferable to a parking lot.  The Park application is
 designed to sequentially use/reuse a series of lots.  By
 transferring the caller to conference 11234, you would be able to have
 the agent pick up the call by going to conference 11234.

 Yes, I think I like this idea ...

 How do you transfer the remote party to conference 11234 ?
 (Please, apologize if this question seems stupid but I'm really a
 newbie on this topic).

 Is it easy to mimic parking lot timeout feature (to be certain a
 caller is not left alone in a dynamic conference) ?

 __
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
 Olivier
 Sent: Wednesday, September 23, 2009 2:32 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] Parking - How to transfer the other
 party to agiven slot

 Hi,

 I'm having trouble to figure out how I could implement this
 feature :
 When on call with a contact, local operator would dial a
 sequence which would park the remote party to a specific
 parking slot, among the hundred of existing slots.
 (to each extension, a single specific parking slot is attached
 and there are too many extensions to dedicate BLF or short
 DTMF sequence to each) .

 Example:
 Operator receives a call from 0123456789. Call
 He talks to remote party and then decides the call is for
 extension 1234.
 As extension 1234 is busy at the moment, Operator forwards the
 incoming call to slot 11234, typing *911234, for instance.
 The person using extension 1234 would see that slot 11234 is
 busy and would try to shorten ongoing call.

 Should I use features.conf's dynamic features for that (to
 allow a specific DTMF sequence while on call) ?
 Then how can I let Operator type digits after *91 prefix ?
 Should I use Incomplete() application ?

 Regards

 ___
 -- Bandwidth and Colocation Provided by
 http://www.api-digital.com --

 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

 ___
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --

 AstriCon 2009 - October 13 - 15 Phoenix, Arizona
 Register Now: http://www.astricon.net

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
John A. Sullivan III
Open Source Development Corporation
+1 207-985-7880
jsulli...@opensourcedevel.com

http://www.spiritualoutreach.com
Making Christianity intelligible to secular society

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] SIP/WiFi handsets?

2009-09-23 Thread Hans Witvliet

On Wed, 2009-09-23 at 09:39 -0700, mgra...@mstvp.com wrote:
I had a good experience with that Polycom/Spectralink phone. Very rugged
as you say. The experience did highlight the weaknesses in consumer
Wifi AP, which reinforced my commitment to continue using DECT around my
office.

Michael

Ken,
I did lots of research on this for my VoIP deployment here where I work.
We have a huge manufacturing floor and all the supervisors have wifi
phones. We evetually settled on the Polycom Spectralink 8002. A nice rugged
little phone with great sound quality and some good features. We use a
managed switch to create seamless wifi coverage over all of our AP's.
Provisioning the phone is pretty easy, but no web browser if you were
planning on using the phone to travel with, some hotels require login for
internet access.

I also tried a clamshell wifi SIP phone by D-Link. This phone actually
works really well, but we had some minor issues with it so we went with all
Spectralink phones. But the D-Link phone would be good choice if you plan
to take your wifi phone on the road.

I also tested the Linksys WIP330 which I thought was a terrible phone.
Very difficult to use.

Good luck.

Thanks!

For a field trial we bought about 50 flutstars F1000.
Audio quality reasonable, reach also good.
Minus-points were: wep-only and the capacity/quality of the batteries.

Instead of DECT, we bought our own base-stations and made our own
GSN-network. Much cheaper as all of the people allready have an
GSM-phone.

___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users

[asterisk-users] DYNAMIC FEATURES, AEL2 - how to use Goto, Gosub or Macro ?

Hello,

I'm using AEL2 (in Asterisk 1.6.1.6) and I can't find a way to successfully
come back into my dialplan.

I've tried things like this (in features.conf) :
toto = #9,peer,Goto,mylocal2,s,1

But typing #9 (from channel SIP/7275, in example bellow) I've got:
--  Feature Found: toto exten: toto
-- Started music on hold, class 'default', on SIP/7275-08b7fbe0
-- Goto (dial-with-user-events,s,64)
-- Stopped music on hold on SIP/7275-08b7fbe0

In extensions.ael, I've got:
macro dial-with-user-events (caller,callee,dst,fwdcount) {
   ...
   Dial (...)
   ...
   return;

mylocal2:
NoOp(Before starting anything);
DumpChan();
return;
};


From my point of view, it seems Asterisk is looking for something in context
in which Dial originally occurred, but for an unknown reason, it can't find
the appropriate hook to keep on.

Do you have any working sample ?

Regards
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Parking - How to transfer the otherparty toagiven slot