Re: [Asterisk-Users] Asterisk on Dell blade servers

[Richard Scobie]

Mike Fedyk wrote:

Matt Riddell wrote:

I would instead recommend the SuperMicro 1U servers - we have had a 
really

great run with these.
 


Do you use Opteron or Intel?


I would not suggest that Supermicro are in Intel's pocket, so they must 
have had their fingers in their ears going, Laa..Laa..Laa..Laa..., 
when the AMD guys came round with benchmarks of their current hardware...


Supermicro do not do Opteron (or Athlon64) systems.

Regards,

Richard
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[Asterisk-Users] cisco/asterisk interop issues?

[James Burke]

hi,

i have an issue that when making a call from a SIP phone going as follows:

phone -- asterisk -- cisco(192.168.0.1) -- terminating voip 
platform(10.0.0.1)

i get the cisco sending up an invite to the voip platform followed 
directly with a CANCEL message, as follows:


Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bKF325E4
Remote-Party-ID: device 
sip:[EMAIL PROTECTED];party=calling;screen=no;privacy=off

From: device sip:[EMAIL PROTECTED];tag=B2A336CC-413
To: sip:[EMAIL PROTECTED]
Date: Thu, 05 Jan 2006 15:09:08 GMT
Call-ID: [EMAIL PROTECTED]
Supported: 100rel,timer,resource-priority
Min-SE:  1800
Cisco-Guid: 227404060-2100564442-3154699218-4120052929
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, 
NOTIFY, INFO, REG

ISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1136473748
Contact: sip:[EMAIL PROTECTED]:5060
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 285


Jan  5 15:09:10.642: //-1//SIP/Msg/ccsipDisplayMsg:
Sent:
CANCEL sip:[EMAIL PROTECTED]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bKF325E4
From: device sip:[EMAIL PROTECTED];tag=B2A336CC-413
to: sip:[EMAIL PROTECTED]
Date: Thu, 05 Jan 2006 15:09:08 GMT
Call-ID: [EMAIL PROTECTED]
CSeq: 101 CANCEL
Max-Forwards: 70
Timestamp: 1136473750
Reason: Q.850;cause=0
Content-Length: 0

the asterisk reports the following:

-- Executing Dial(SIP/200-c5c4, SIP/[EMAIL PROTECTED]) in new stack
-- Called [EMAIL PROTECTED]
-- SIP/192.168.0.1-a928 is making progress passing it to SIP/200-c5c4
-- Got SIP response 500 Internal Server Error back from 192.168.0.1
-- SIP/192.168.0.1-a928 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

if i send it as follows:

phone -- asterisk -- cisco(192.168.0.1) -- pstn

all is good and call is processed normally.

any help would be appreciated..
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[Asterisk-Users] bayhamsystems.com experience

[Michiel van Baak]

Hi all,

Anyone using their services ?
I'm thinking of setting up my servers with their service.
But before starting to mess with my extensions.conf I thought let's check
the community for their experience.

Thanks,

Michiel van Baak.

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Re: [Asterisk-Users] bayhamsystems.com experience

[Peter Bowyer]

On 06/01/06, Michiel van Baak [EMAIL PROTECTED] wrote:

 Anyone using their services ?
 I'm thinking of setting up my servers with their service.
 But before starting to mess with my extensions.conf I thought let's check
 the community for their experience.

I use them - the service works exactly as advertised. Recommended.

I use the perl version of their AGI (so I could hack it easily) -
actually I really only used it as a building block for a more
extensive MWI management system. The samples they provide are not
foolproof, there's more logic needed to do the job properly.

Peter

--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
VoIP: [EMAIL PROTECTED]
FWD: **275*5048707000
VoipTalk: **473*5048707000
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RE: [Asterisk-Users] bayhamsystems.com experience

[Steve Totaro]

I just signed up for an account with them yesterday.  I need to
configure my asterisk box for my needs and will test them out.  I will
post to this thread as well as the wiki after a week or two of testing.

Thanks,
Steve

 
 Hi all,
 
 Anyone using their services ?
 I'm thinking of setting up my servers with their service.
 But before starting to mess with my extensions.conf I thought let's
check
 the community for their experience.
 
 Thanks,
 
 Michiel van Baak.
 
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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

[Remco Barende]

Not really, their suggested retail price is USD 300 for the analog unit, 
probably because of the intelligent stuff in the box (which we do not 
need when using *).


At USD 300 you can find SIP capable devices, for an analog unit the SIPCE 
is 3x more expensive than the unit we were discussing.


But thanks for the tip!


On Thu, 5 Jan 2006, Cory Andrews wrote:

SICPE has a new product called the GSM Call Director that may be of interest 
to GSM enthusiasts.


http://www.sipcpe.com/fx300GSM.html

Cory Andrews
Purchasing Manager
++
VOIPSupply.com
A Division of b2 Technologies
454 Sonwil Drive
Buffalo, NY 14225

direct - 716.250.3402
mobile - 716.907.4054
email - [EMAIL PROTECTED]
AIM - b2Cory

- Original Message - From: Sam Tam [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion' 
asterisk-users@lists.digium.com

Sent: Thursday, January 05, 2006 3:30 PM
Subject: RE: [Asterisk-Users] GSM Gateway / Terminal for sale


We have ran out of stock in our office in UK. All GSM Gateway are now being
send from HK therefore the shipping will be more expensive than usual.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of bails
Sent: Friday, January 06, 2006 12:18 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale

Chris Bagnall wrote:

Single port GSM Gateway support 900 / 1800 GSM mode with
external antenna.
Brand new unit and all of them will be tested before dispatch.
Extremely easy to setup and can be used out of the box
without any configuration. So should be good alternatively of
phonecell or nokia pbx etc..
Units are located in UK and £60 GBP per unit excluding shipping.



Has anyone bought one of these and able to offer some feedback? I'm
seriously considering a GSM gateway to take advantage of the spare SIM

cards

lying around still inside their 12-month contracts.

Looking at the website in question, delivery is £17.37 for a 6-day

delivery,

or £10 for a 30+ day delivery, both of which seem a bit high for an item
apparently located in the UK.

Regards,

Chris


We were working in the area (Reading) and offered to pay cash and
collect from their site, but the response was;

that they could only be sent direct from the far east

We weren't prepared to take the risk, I mean they turned down cash!

Bails

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Re: [Asterisk-Users] bayhamsystems.com experience

[John Daragon]

Michiel van Baak wrote:

Hi all,

Anyone using their services ?
I'm thinking of setting up my servers with their service.
But before starting to mess with my extensions.conf I thought let's check
the community for their experience.


I don't use them from asterisk, but I do use their SMS service from a 
locally coded application.


Responsive, easy to do business with, absolutely no problems at all.

jd

--

John Daragon  [EMAIL PROTECTED]
argv[0] limited   (Asterisk implementation  consultancy)
Lambs Lawn Cottage,  Staple Fitzpaine,  Taunton,  TA3 5SL,  UK
v +44 (0) 1460 234068   f +44 (0) 1460 234069   m +44 (0) 7836 576127


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Re: [Asterisk-Users] TE110p and pri_cpe signalling not recognized

[[EMAIL PROTECTED]]

bchan=1-5,7-15,17-31
dchan=16
 


Why are you excluding channel 6?

jvb
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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

[Chris Mason (Lists)]

Remco Barende wrote:
Not really, their suggested retail price is USD 300 for the analog 
unit, probably because of the intelligent stuff in the box (which we 
do not need when using *).


At USD 300 you can find SIP capable devices, for an analog unit the 
SIPCE is 3x more expensive than the unit we were discussing.



Where can I find the $300 SIP capable units?

--
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 


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[Asterisk-Users] Budge Tone-100 as a Ext in the LAN

[luke devon]

HI , I installed asterisk in fedora core 3 machine perfectly. and i have 10 units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do any configurations of any files .What are the configurations has to be made with asterisk ?Thanx in advance,  Luke.Send instant messages to your online friends http://uk.messenger.yahoo.com ___
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Re: [Asterisk-Users] Budge Tone-100 as a Ext in the LAN

[Yair Hakak]

lukeuse the wiki.
(always wanted to do that)

http://www.voip-info.org/wiki/view/Asterisk+phone+grandstream+budgetone

hope this helps,
 yair

On 1/6/06, luke devon [EMAIL PROTECTED] wrote:
 HI ,

 I installed asterisk in fedora core 3 machine perfectly. and i have 10 units
 of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i use
 it as  extentions in my LAN ? Asterisk PBX alredy there. I didn't try to do
 any configurations of any files .

 What are the configurations has to be made with asterisk ?

 Thanx in advance,
 Luke.

 Send instant messages to your online friends http://uk.messenger.yahoo.com
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Re: [Asterisk-Users] Screening incoming calls.

[Philipp von Klitzing]

Hi!

 The PBX I'm getting ready to replace has a really nifty feature -- one
 that I'm not even sure Asterisk -can- do -- though I'm hoping to be proven
 wrong.  When a call goes to voicemail, the end-user can listen to the VM
 as it's being recorded, and can interrupt and answer the call if it's
 someone they want to talk to.
 
 Is there any way to implement this?

Yes, I've described the voicemail live approach here:
http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+voicemail+live

No need for ChanSpy and the manager interface, just needs MeetMe.

Cheers, Philipp


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[Asterisk-Users] RE:how many calls Asterisk gateway can handle

[Tejas Shah]

hi all,   I am newbie to asterisk. I have installed asterisk based VoIP gateway in my LAB. Now i want to how many simultaneous calls (internal and external) can this gateway can handle? hereby i m sending my system details:  1) asterisk gateway is running on P-IV 2.6GHz machine. 2) i have installed one X100P FXO card on my PC for PSTN connection. 3) i have installed 4-soft X-LITE phones on 4 different PCs. 4) I am using SIP protocol. 5) codec is G.711u.  1) Now can anyone tell me how many simultaneous calls can my asterisk gateway handle?  2) How is it to possible to simulate the performance of asterisk VoIP gateway?  3) Is any tool available so that it can generate many calls and i can check gateway performance?  4) I am thinking of SIPp tool?  5) can any body have idea that what changes i have to make in sip.c
 onf and
 extension.conf to   register those calls generated by SIPp? I mean how asterisk server make enrty of those calls?  Pls help me out..  thanks  tejas 
	
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[Asterisk-Users] Problem with integrating ISDN PBX using NT mode

[Frederik Fix]

Hi,
I'm just in the process of replacing a crappy Siemens PBX with a new and
shiny Asterisk system. To connect Legacy equipment I hooked up a small
ISDN PBX (DeTeWe OpenCom 36) to one port on a Junghanns.net quadBRI
card. That port is configured for NT Point to Multipoint
(Mehrgeraeteanschluss) mode. Now I can place calls from the ISDN PBX to
the other Asterisk extensions but the other way around does not work.
Whenever I call from the Asterisk server to one of the extensions
connected through the ISDN PBX that extension rings for a split second
and then the call is dropped. Here is what I get on the console:

-- Executing Macro(SCCP/13-002f, standard|Zap/g2/40) in new
stack
-- Executing Dial(SCCP/13-002f, Zap/g2/40|20) in new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g2/40
  == Primary D-Channel on span 4 up for TEI 64
  == Primary D-Channel on span 4 up for TEI 66
-- Zap/10-1 is proceeding passing it to SCCP/13-002f
-- Zap/10-1 is ringing
-- Channel 0/1, span 4 got hangup request
-- Hungup 'Zap/10-1'
  == No one is available to answer at this time (1:0/0/0)
-- Executing Goto(SCCP/13-002f, s-NOANSWER|1) in new stack
-- Goto (macro-standard,s-NOANSWER,1)
-- Executing VoiceMail(SCCP/13-002f, u40) in new stack
-- Executing Goto(SCCP/13-002f, default|s|1) in new stack
-- Goto (default,s,1)
  == Channel 'SCCP/13-002f' jumping out of macro 'standard'
  == Primary D-Channel on span 4 down for TEI 65
  == Primary D-Channel on span 4 down for TEI 64
  == Primary D-Channel on span 4 down for TEI 66

I think I properly configured the ISDN PBX (theres not much to configure
there). Can someone help me here? What's causing the hangup request? How
could I find out?
Below is the relevant configuration.

Thanks in advance,
Frederik Fix

zapata.conf:
[channels]
switchtype = euroisdn

pridialplan = local
prilocaldialplan = local
nationalprefix = 0
internationalprefix = 00
;usecallingpres=yes

echocancel = yes
echocancelwhenbridged = yes
echotraining = 100
debug = 2

; Festnetzanschluss
signalling = bri_cpe
context=extern
group = 1
; S/T port 1
channel = 1-2
; S/T port 2
channel = 4-5
; S/T port 3
channel = 7-8

; Interner S0-Bus
signalling = bri_net_ptmp
context = intern-isdn
group = 2
; S/T port 4
channel = 10-11

extensions.conf:
[macro-standard]
exten = s,1,Dial(${ARG1},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${MACRO_EXTEN})
exten = s-NOANSWER,2,Goto(default,s,1)
exten = s-BUSY,1,Voicemail(b${MACRO_EXTEN})
exten = s-BUSY,2,Goto(default,s,1)
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${MACRO_EXTEN})

[nebenstellen-intern]
; Konferenzzimmer
exten = 13,1,Macro(standard,SCCP/13)

; Ingrid
exten = 17,1,Macro(standard,${INGRID})
exten = 57,1,Macro(standard_ohne_ab,Zap/g2/57)

; Gavigo
exten = 60,1,Macro(standard,SCCP/60)

; Woelm
exten = 66,1,Macro(standard,SCCP/66)
exten = 68,1,Macro(standard_ohne_ab,Zap/g2/68)

; van de Beeck
exten = 40,1,Macro(standard,Zap/g2/40)
exten = 44,1,Macro(standard_ohne_ab,Zap/g2/44)

; Rohan
exten = 50,1,Macro(standard,Zap/g2/50)
exten = 59,1,Macro(standard_ohne_ab,Zap/g2/59)

; Hinterhaus
exten = 58,1,Macro(standard,Zap/g2/58)
exten = 22,1,Macro(standard,Zap/g2/22)

; fuer Testzwecke
exten = 61,1,Macro(standard,SIP/eyebeamtest)

; virtuelle Nebenstellen
exten = 30,1,Macro(virtuell)
exten = 35,1,Macro(virtuell)
exten = 48,1,Macro(virtuell)

exten = 25,1,Dial(Zap/g2/58)

[intern-isdn]
exten = 25,1,Dial(SCCP/13)
exten = s,1,DISA(no-password|intern)

[dialout]
exten = _0.,1,Dial(Zap/g1/${EXTEN:1})

[intern]
include = dialout
include = nebenstellen-intern
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[Asterisk-Users] server recommendations

[B. Keith Murphy]

OK all.  I need some help.  Looking to deploy asterisk servers and want 
to get a recommendation on what server to buy.  I love Dell's, but from 
what I see on the list they seem to have some issues.  I would like to 
stay with one brand and need systems for small offices (20 users), 
medium (50 users) and large (100 users) systems.  Thanks for the help.


Keith
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RE: [Asterisk-Users] GSM Gateway / Terminal for sale

[JCC]

I don't get it. What is the advantage of using a GSM gateway? VOIP calls are
pretty inexpensive as they are now. Is the use of a gateway intended as a
backup incase a wired network connection goes down? I have being looking
around the net for information on this. Anyone out there using it and if so
you can please share with me how you use this technology? Any information
will be appreciated.

Thanks,

Jay

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Friday, January 06, 2006 5:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale

Remco Barende wrote:
 Not really, their suggested retail price is USD 300 for the analog 
 unit, probably because of the intelligent stuff in the box (which we 
 do not need when using *).

 At USD 300 you can find SIP capable devices, for an analog unit the 
 SIPCE is 3x more expensive than the unit we were discussing.

Where can I find the $300 SIP capable units?

-- 
Chris Mason
NetConcepts
(264) 497-5670 Fax: (264) 497-8463
Int:  (305) 704-7249 Fax: (815)301-9759 
Cell: 264-235-5670
Yahoo IM: [EMAIL PROTECTED] 

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RE: [Asterisk-Users] GSM Gateway / Terminal for sale

[Pete Barnwell]

On Fri, 2006-01-06 at 07:35 -0500, JCC wrote:
 I don't get it. What is the advantage of using a GSM gateway? VOIP calls are
 pretty inexpensive as they are now. Is the use of a gateway intended as a
 backup incase a wired network connection goes down? I have being looking
 around the net for information on this. Anyone out there using it and if so
 you can please share with me how you use this technology? Any information
 will be appreciated.
 
 Thanks,
 
 Jay

Hi Jay,

I use them because:

Calls between mobiles on our package are free. Calls from * to mobile
routed via ITSP aren't. If I route them via a GSM gateway then we don't
pay any call charges for calls to any of our mobile people. Instead we
pay about £14 p.c.m. for the extra SIM card, which we save in about 2
days.

Cheers

Pete

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RE: [Asterisk-Users] GSM Gateway / Terminal for sale

[Remco Barende]

VOIP - GSM calls may be cheap if you call to China.

When you call a cell in The Netherlands it will cost you USD 0.25 per 
minute. I am located in NL therefore a lot of calls go to NL mobiles.


You can buy sim cards that offer minutes for USD 0.02 per minute, if you 
can recommend a carrier that offers VOIP - NL GSM calls for that amount I 
will be very happy :)




On Fri, 6 Jan 2006, JCC wrote:


I don't get it. What is the advantage of using a GSM gateway? VOIP calls are
pretty inexpensive as they are now. Is the use of a gateway intended as a
backup incase a wired network connection goes down? I have being looking
around the net for information on this. Anyone out there using it and if so
you can please share with me how you use this technology? Any information
will be appreciated.

Thanks,

Jay

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Mason
(Lists)
Sent: Friday, January 06, 2006 5:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale

Remco Barende wrote:

Not really, their suggested retail price is USD 300 for the analog
unit, probably because of the intelligent stuff in the box (which we
do not need when using *).

At USD 300 you can find SIP capable devices, for an analog unit the
SIPCE is 3x more expensive than the unit we were discussing.


Where can I find the $300 SIP capable units?



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[Asterisk-Users] Call forwarding for particular extension

[nr k]

Hi all

I need to configure call forwarding for particular
extension is busy.how to configure this my extension
configuration is like following.


exten = 2006,1,Dial(SIP/sipura2)


regards
ramakrishnan.n



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Re: [Asterisk-Users] server recommendations

[Steve Blair]

We use Dell PE 1650 upto 2850 servers for all of our Asterisk and SER
applications and they work just fine. Not sure what others are experiencing
but our systems have been rock solid.

-Steve

B. Keith Murphy wrote:

OK all.  I need some help.  Looking to deploy asterisk servers and 
want to get a recommendation on what server to buy.  I love Dell's, 
but from what I see on the list they seem to have some issues.  I 
would like to stay with one brand and need systems for small offices 
(20 users), medium (50 users) and large (100 users) systems.  Thanks 
for the help.


Keith
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ISC Network Engineering

The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104  



voice: 215-573-8396 


  215-746-8001

fax: 215-898-9348


sip:[EMAIL PROTECTED]

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Re: [Asterisk-Users] Call forwarding for particular extension

[Giovanni Miano]

exten = 2006,2,goto(s-${DIALSTATUS},1)exten = s-BUSY,1,DIAL(SIP/sipura3)exten = s-NOANSWER,1,exten = s-www.voip-info.org/wiki-Asterisk+variable+DIALSTATUS
Cheers,Giovanni Miano2006/1/6, nr k [EMAIL PROTECTED]
:Hi allI need to configure call forwarding for particularextension is 
busy.how to configure this my extension configuration is like following.exten = 2006,1,Dial(SIP/sipura2)regardsramakrishnan.n__
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RE: [Asterisk-Users] GSM Gateway / Terminal for sale

[Chris Bagnall]

 I don't get it. What is the advantage of using a GSM gateway? 
 VOIP calls are pretty inexpensive as they are now.

It largely depends on the country you're calling. Here in the UK, calls to
mobiles are maintained at an artificially high rate because the terminating
network (the mobile networks) get a cut of call revenue for calls *to* your
mobile. By contrast, in the US, the mobile customer often pays a small
charge per minute on incoming calls (as I understand the market over there).

You'll also find in the UK the mobile phone market is heavily subsidized by
the networks such that you can get phones for free if you sign up to 12
month contracts. I often find that it's cost-effective to get a new contract
every 12 months (with a free phone), even if I don't want the phone. Flog
the phone on ebay and you've got a spare SIM with lots of inclusive minutes
for almost nothing.

Regards,

Chris
-- 
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This email is made from 100% recycled electrons


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Re: [Asterisk-Users] CD (call deflection) on Bristuff/zaphfc?

[Giovanni Miano]

call deflection does not work with bristuffuse CAPI2006/1/6, Pisac [EMAIL PROTECTED]:
Do bristuff/zaphfc support CD (Call Deflection)?How to deflect call (transfer before answering) with bristuff?
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Re: [Asterisk-Users] call monitoring from 3th phone

[Matt]

I can't say for sure that it's 10.. but it's somewhere between 8 and
13 as I hit * to cycle.. when I get up in that range... it will stop
spying.. and asterisk will stop taking calls until I do a restart.

On 1/5/06, Tom Vile [EMAIL PROTECTED] wrote:
 I have not had that issue.  Are you saying 10 concurrent channels
 being spied on or after the 10th it starts to crash?

 On 1/5/06, Matt [EMAIL PROTECTED] wrote:
  I've found that chanspy crashes asterisk after about 10 channel spys..
  asterisk just stops responding, and I have to restart it.
 
  On 1/4/06, Tom Vile [EMAIL PROTECTED] wrote:
   correct it only works with bridged calls.
   On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote:
Tom Vile wrote:
   
use chanspy or zapbarge



That slipped my mind :). Had always been using the conf method since pre
1.0. Does app_chanspy work with reinvite=yes? I understand it only works
with bridged calls.
   
On 1/4/06, Leo Ann Boon [EMAIL PROTECTED] wrote:


[EMAIL PROTECTED] wrote:



is it possible only monitoring call between phone A and B from phone 
C?





I think you want to do service observation? You can do the following:
a. Use a 'stealth' meetme conference room say 1234 that doesn't need 
PIN
to log in and also doesn't play a tone on entry/exit (may not be legal
in your country).
b. Use manager API to redirect 'A' and 'B' to the conference room.
c. 'C' joins the conference room with the mute option.
d. C will now be able to hear what A and B are saying.




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--
Tom Vile
Baldwin Technology Solutions, Inc
Consulting - Web Design - VoIP Telephony
www.baldwintechsolutions.com
Phone: 518-631-2855 x205
Fax: 518-631-2856
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   --
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   Baldwin Technology Solutions, Inc
   Consulting - Web Design - VoIP Telephony
   www.baldwintechsolutions.com
   Phone: 518-631-2855 x205
   Fax: 518-631-2856
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 --
 Tom Vile
 Baldwin Technology Solutions, Inc
 Consulting - Web Design - VoIP Telephony
 www.baldwintechsolutions.com
 Phone: 518-631-2855 x205
 Fax: 518-631-2856
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Re: [Asterisk-Users] ChanSpy via external application

[Dov Bigio]

Hello,

It didn't work...

I used "Data: SIP/dov.bigio-9949" which was the 
channel being used, and the call I received just had beeps... no 
conversation.

According to the documentation on (http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy), 
ChanSpy doesn't take a channel as parameter, does it?

Thank you very much!!
Dov

  - Original Message - 
  From: 
  Giovanni 
  Miano 
  To: Dov Bigio ; Asterisk Users Mailing List - 
  Non-Commercial Discussion 
  Sent: Thursday, January 05, 2006 7:01 
  PM
  Subject: Re: [Asterisk-Users] ChanSpy via 
  external application
  Use channel of your agentChannel: SIP/dov.bigioMaxRetries: 3RetryTime: 40WaitTime: 
  25Context: 01.telecomApplication: ChanSpyData: 
  SIP/234-ssnfPriority: 1Cheers,Giovanni Miano
  2006/1/5, Dov Bigio  [EMAIL PROTECTED]:
  
Hi,

I have developped an application that monitors 
the status of my queues through the events triggered on the Manager 
Interface.

This way, I can know the status of my Agent 
real time.

Now, I have a new requirement that I must allow 
a manager to click on the Agent he wants to monitor and be able to monitor 
the call.

My idea was to, when the user clicks on the 
Agent, I would Originate a call between his extension and the 
extension I have for ChanSpy, passing as parameter the Agent 
number.

For testing this, I tried a call file on 
/var/spool/asterisk/outgoing

Channel: 
SIP/dov.bigio 
--- This is meMaxRetries: 3RetryTime: 40WaitTime: 
25Context: 01.telecomApplication: ChanSpyData: 
Agent/5450- 
This is the Agent I want to monitorPriority: 1
The problem is that ChanSpy doesn't accept 
"Agent/" as parameter, just "Agent".
Is there a way to ChanSpy a specific know 
Agent?
(Or at least to send via dtmf the Agent Number 
I want to monitor right after the ChanSpy application is 
called?

Thank you very much!Dov
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[Asterisk-Users] Xs4all VoIP service - SIP config?

[Patrick]

Hi,

Recently the Dutch ISP Xs4all started a SIP based VoIP service with free
087 numbers to their subscribers. Has anyone been able to get this
service to work with Asterisk? So far I had no luck. It seems they use
MD5 authentication with a realm of sip.xs4all.nl. And for those
interested: they use the solution from B3G.

Thanks and regards,
Patrick
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Re: [Asterisk-Users] CD (call deflection) on Bristuff/zaphfc?

[Torsten Krueger]

Hello,

Giovanni Miano schrieb:
 call deflection does not work with bristuff

this is no longer true - at least not when using a recent bristuff version and a
point-to-multipoint trunk.

exten = 37,1,Wait(0.5)
exten = 37,2,ZapCD(destination-number)
exten = 37,3,Progress()
exten = 37,4,4,Hangup

Does just what it should. Unfortunately this does not work when using a
point-to-point connection. In this case you would the facility 'reroute' and
this is not implemented in bristuff. BTW, the Sirrix channels can also do both.

Regards
Torsten

 
 use CAPI
 
 2006/1/6, Pisac [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]:
 
 Do bri
 
  
 
 stuff/zaphfc support CD (Call Deflection)?
 
 How to deflect call (transfer before answering) with bristuff?
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RE: [Asterisk-Users] Asterisk on Dell blade servers

[Dean Collins]

Lol, so Dell must be doing the same thing.

Did you ever consider that Supermicro are an enterprise setup to make
money, and that possibly their financial interests are served by
sticking with Intel?

You would have to figure that Dell is doing something right to get to
the size they currently areAMD-less as the case may be.


Cheers,

Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Richard
Scobie
Sent: Friday, 6 January 2006 3:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk on Dell blade servers



Mike Fedyk wrote:
Matt Riddell wrote:

 I would instead recommend the SuperMicro 1U servers - we have had a 
 really
 great run with these.
  

 Do you use Opteron or Intel?

I would not suggest that Supermicro are in Intel's pocket, so they must 
have had their fingers in their ears going, Laa..Laa..Laa..Laa..., 
when the AMD guys came round with benchmarks of their current
hardware...

Supermicro do not do Opteron (or Athlon64) systems.

Regards,

Richard
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Re: [Asterisk-Users] open h323 compile error

[Bob Goddard]

On Friday 06 Jan 2006 00:46, A_ Navone wrote:
 make[2]: *** [obj_linux_x86_r/simph323] Error 1
 make[2]: Leaving directory `/usr/src/openh323/samples/simple'
 make[1]: *** [opt] Error 2
 make[1]: Leaving directory `/usr/src/openh323'
 make: *** [optshared] Error 2

 
 any idea ?

None unless you tell us what the error is.
Hint, it's the first error which matters, not the last.


B

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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

[Steve Kennedy]

On Fri, Jan 06, 2006 at 01:23:26PM -, Chris Bagnall wrote:

  I don't get it. What is the advantage of using a GSM gateway? 
  VOIP calls are pretty inexpensive as they are now.
 It largely depends on the country you're calling. Here in the UK, calls to
 mobiles are maintained at an artificially high rate because the terminating
 network (the mobile networks) get a cut of call revenue for calls *to* your
 mobile. By contrast, in the US, the mobile customer often pays a small
 charge per minute on incoming calls (as I understand the market over there).
 You'll also find in the UK the mobile phone market is heavily subsidized by
 the networks such that you can get phones for free if you sign up to 12
 month contracts. I often find that it's cost-effective to get a new contract
 every 12 months (with a free phone), even if I don't want the phone. Flog
 the phone on ebay and you've got a spare SIM with lots of inclusive minutes
 for almost nothing.

In the UK the wholesale rates are set by Ofcom (like the FCC), which
works out about 7p'ish per minute.

However the operators can offer retail bundles (including phones) and
for a monthly contract they throw in various ammounts of cross network
minutes (or free to their own network or whatever). With clever
dial-plans and multiple terminals connected to multiple networks you can
generally get free calls to mobile users (basically clever least cost
routing, time of day sometimes needs to be taken into account as well).

However there are some disadvantages, the main being you cant set CLI of
the outgoing call as it will always be tied to the SIM of the mobile
terminal.

Another is that you can NOT run a GSM gateway (as they're known) for 3rd
parties. So if you want to connect your office PBX to a gateway to make
use of cheap mobile termination for your own company that's fine, but as
an ITSP (or traditional telco) you can not allow 3rd party traffic to
utilise a gateway. If networks find you are using a gateway (as a telco)
they can cut it off, no questions asked. Gateways have been determined
to be fixed infrastructure, therefore NOT mobile.

There is (or maybe was by now) an Ofcom consultation asking whether this
should be changed, the mobile operators will fight it, telcos and other
users will be asking for it to be changed.

Of course this is UK specific, other countries have more lenient
policies (I think Belgium allow gateways, France doesn't allow any kind,
and some allow them with the co-operation of the operators).


Steve

-- 
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UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM stevekennedyuk / MSN [EMAIL PROTECTED]
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Re: [Asterisk-Users] Asterisk on Dell blade servers

[Bob Goddard]

On Friday 06 Jan 2006 08:11, Richard Scobie wrote:
 Mike Fedyk wrote:
 Matt Riddell wrote:
 
  I would instead recommend the SuperMicro 1U servers - we have had a
  really
  great run with these.
 
  Do you use Opteron or Intel?

 I would not suggest that Supermicro are in Intel's pocket, so they must
 have had their fingers in their ears going, Laa..Laa..Laa..Laa...,
 when the AMD guys came round with benchmarks of their current hardware...

 Supermicro do not do Opteron (or Athlon64) systems.

Supermicro DO do Opteron.


B


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[Asterisk-Users] FATAL: Error running install command for ztdummy

[Tom]

Here is the issue:

[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# lsmod | grep zaptel
zaptel206724  0 
crc_ccitt   2113  1 zaptel
[EMAIL PROTECTED] ~]# 
[EMAIL PROTECTED] ~]# 
[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

FATAL: Error running install command for ztdummy
[EMAIL PROTECTED] ~]# 




Here is the background:


I am using the Linux 4 ES Update 2 distro.
[EMAIL PROTECTED] ~]# uname -a
Linux localhost.localdomain 2.6.9-22.EL #1 Mon Sep 19
18:20:28 EDT 2005 i686 i686 i386 GNU/Linux
[EMAIL PROTECTED] ~]# 


I am installing zaptel (just to get access to ztdummy)
and asterisk for the first time and this is the first
time that it is being installed on the server in 

question.  No PSTN hardware in the server; just a NIC
card -- I will be doing sip to sip calls in a lab
environment.

1.  Before compiling zaptel (the compile of zaptel
looks good to me -- but see below), I configured NTP
on the server so that it would sync to another server 

of ours which syncs directly to the NTP server.   I
did this via crontab:

[EMAIL PROTECTED] ~]# crontab -l
30 * * * * /usr/sbin/ntpdate -u xx  /dev/null
21
[EMAIL PROTECTED] ~]# ntpq
ntpq host xx
current host set to xx.a.net
ntpq peers
 remote   refid  st t when poll reach 
 delay   offset  jitter
==
 LOCAL(0)LOCAL(0)10 l   44   64  377  
 0.0000.000   0.000
*HOPF_S(0)   .CDMA.   0 l   10   16  377  
 0.000   -0.003   0.000
ntpq


2.  I Uncommented ztdummy in the zaptel Makefile
before doing the first compile and I left one-space
between ztd-loc and ztdummy.

MODULES:=zaptel tor2 torisa wcusb wcfxo wctdm
wctdm24xxp \
 ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp
pciradio \
 ztd-loc ztdummy

3.  In the compile of zaptel (make, make install, make
config, make clean) I noticed in the make output the
following (not sure if this is a problem or not):

[EMAIL PROTECTED] zaptel-1.2.1]# 
[EMAIL PROTECTED] zaptel-1.2.1]# make
Makefile:178: target `ztdummy.o' given more than once
in the same rule.
~SNIP~
make[1]: Entering directory
`/usr/src/kernels/2.6.9-22.EL-i686'
/usr/src/zaptel-1.2.1/Makefile:178: target `ztdummy.o'
given more than once in the same rule.
  CC [M]  /usr/src/zaptel-1.2.1/zaptel.o
/usr/src/zaptel-1.2.1/zaptel.c:187: warning: 'fcstab'
defined but not used
  CC [M]  /usr/src/zaptel-1.2.1/tor2.o
~SNIP~

4.  Making NO edits to the Makefile for asterisk, I
then compiled asterisk (make, make install, make
samples, make progdocs, make config, make clean).  

Looked good.

5.  I then went straight to loading zaptel (having
forgotten to first compile asterisk-sounds (make
install) and asterisk-addons (make install) [side 

question: Is it OK to go back and compile sounds and
addons after zaptel is now loaded??])

6.  I loaded zaptel sucessfully, it looks like:

[EMAIL PROTECTED] ~]# 
[EMAIL PROTECTED] ~]# modprobe zaptel
[EMAIL PROTECTED] ~]# lsmod | grep zaptel
zaptel206724  0 
crc_ccitt   2113  1 zaptel
[EMAIL PROTECTED] ~]# 


7.  ztdummy, however, does not load:

[EMAIL PROTECTED] ~]# modprobe ztdummy
Notice: Configuration file is /etc/zaptel.conf
line 0: Unable to open master device '/dev/zap/ctl'

1 error(s) detected

FATAL: Error running install command for ztdummy
[EMAIL PROTECTED] ~]# 


7.1  I see that i have no zap/ under /dev/:

[EMAIL PROTECTED] ~]# cd /dev/zap/
-bash: cd: /dev/zap/: No such file or directory
[EMAIL PROTECTED] ~]# cd /dev
[EMAIL PROTECTED] dev]# 
[EMAIL PROTECTED] dev]# 
[EMAIL PROTECTED] dev]# pwd
/dev
[EMAIL PROTECTED] dev]# ls -l
total 0
crw---  1 root  root36,   8 Dec 23 10:04 arpd
lrwxrwxrwx  1 root  root  3 Dec 23 10:04 cdrom
- hda
crw---  1 rdlab root 5,   1 Dec 23 16:21
console
lrwxrwxrwx  1 root  root 11 Dec 23 10:04 core
- /proc/kcore
crw---  1 root  root10,  63 Dec 23 10:04
device-mapper
~SNIP~
crw---  1 vcsa  tty  7, 136 Dec 23 15:05 vcsa8
drwx--  2 root  root 80 Dec 23 15:05
VolGroup00
crw---  1 root  root36,   6 Dec 23 10:04 xfrm
lrwxrwxrwx  1 root  root  4 Dec 23 10:04 XOR
- null
crw---  1 root  root   196, 254 Jan  5 17:47
zapchannel
crw---  1 root  root   196,   0 Jan  5 17:47
zapctl
crw---  1 root  root   196, 255 Jan  5 17:47
zappseudo
crw---  1 root  root   196, 253 Jan  5 17:47
zaptimer
crw-rw-rw-  1 root  root 1,   5 Dec 23 10:04 zero
[EMAIL PROTECTED] dev]# 


7.2 Per my understanding of the asterisk docs, I
belive I have to solve this ztdummy load issue BEFORE
I attempt to load asterisk for the first time.


Thanks in advance!






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[Asterisk-Users] How To - Building a VoIP-PSTN Gateway with Asterisk

[maingault]

Hi,

Im a new user of Aterisk, and I have to
configure a VoIP Gateway.

I have an Alcatel PBX with an E1 card, connected, for
the moment, to a local carrier.

I would like work with a french VoIP provider, but,
for this, I need to use a VoIP Gateway for connect my E1.

Thus, I want to build my own voip gateway.

It very simple, I want to connect my PBX to the
gateway (E1 link) for both call origination and termination with a SIP VoIP
Provider.



Is asterisk + digium card is ok for this purpose ?



What kind of configuration I have to do on Asterisk ?



Thanks

Bertrand






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[Asterisk-Users] Recording Calls at the phone

[Michael Sampson]

I work for a call center and we are looking at using asterisk to have 
our operators take calls. Our message taking software records all the 
calls on the operators computers. Right now we use these recording 
controls from radio shack that plug in between the wall jack and the 
phone and plug in via a 1/8 inch stereo connector to the mic input on 
the computer. If I buy an IP phone I can't do that. I could get an FXO 
adapter and regular phones, but I'm looking to get as little equipment 
as possible. Radio shack makes a recording control that plugs in to a 
2.5 mm headset jack, but it takes batteries so thats not going to work


Does anyone else do something similar? Does anyone have any ideas about 
what producs/setup would work for this.


--
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
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Re: [Asterisk-Users] FATAL: Error running install command for ztdummy

[Derek Whitten]

Tom wrote:
 Here is the issue:
 
 [EMAIL PROTECTED] ~]# modprobe zaptel
 [EMAIL PROTECTED] ~]# lsmod | grep zaptel
 zaptel206724  0 
 crc_ccitt   2113  1 zaptel
 [EMAIL PROTECTED] ~]# 
 [EMAIL PROTECTED] ~]# 
 [EMAIL PROTECTED] ~]# modprobe ztdummy
 Notice: Configuration file is /etc/zaptel.conf
 line 0: Unable to open master device '/dev/zap/ctl'
 
 1 error(s) detected
 
 FATAL: Error running install command for ztdummy
 [EMAIL PROTECTED] ~]# 
 
 
 
 
 Here is the background:
 
 
 I am using the Linux 4 ES Update 2 distro.
 [EMAIL PROTECTED] ~]# uname -a
 Linux localhost.localdomain 2.6.9-22.EL #1 Mon Sep 19
 18:20:28 EDT 2005 i686 i686 i386 GNU/Linux
 [EMAIL PROTECTED] ~]# 
 
 
 I am installing zaptel (just to get access to ztdummy)
 and asterisk for the first time and this is the first
 time that it is being installed on the server in 
 
 question.  No PSTN hardware in the server; just a NIC
 card -- I will be doing sip to sip calls in a lab
 environment.
 
 1.  Before compiling zaptel (the compile of zaptel
 looks good to me -- but see below), I configured NTP
 on the server so that it would sync to another server 
 
 of ours which syncs directly to the NTP server.   I
 did this via crontab:
 
 [EMAIL PROTECTED] ~]# crontab -l
 30 * * * * /usr/sbin/ntpdate -u xx  /dev/null
 21
 [EMAIL PROTECTED] ~]# ntpq
 ntpq host xx
 current host set to xx.a.net
 ntpq peers
  remote   refid  st t when poll reach 
  delay   offset  jitter
 ==
  LOCAL(0)LOCAL(0)10 l   44   64  377  
  0.0000.000   0.000
 *HOPF_S(0)   .CDMA.   0 l   10   16  377  
  0.000   -0.003   0.000
 ntpq
 
 
 2.  I Uncommented ztdummy in the zaptel Makefile
 before doing the first compile and I left one-space
 between ztd-loc and ztdummy.
 
 MODULES:=zaptel tor2 torisa wcusb wcfxo wctdm
 wctdm24xxp \
  ztdynamic ztd-eth wct1xxp wct4xxp wcte11xp
 pciradio \
  ztd-loc ztdummy
 
 3.  In the compile of zaptel (make, make install, make
 config, make clean) I noticed in the make output the
 following (not sure if this is a problem or not):
 
 [EMAIL PROTECTED] zaptel-1.2.1]# 
 [EMAIL PROTECTED] zaptel-1.2.1]# make
 Makefile:178: target `ztdummy.o' given more than once
 in the same rule.
 ~SNIP~
 make[1]: Entering directory
 `/usr/src/kernels/2.6.9-22.EL-i686'
 /usr/src/zaptel-1.2.1/Makefile:178: target `ztdummy.o'
 given more than once in the same rule.
   CC [M]  /usr/src/zaptel-1.2.1/zaptel.o
 /usr/src/zaptel-1.2.1/zaptel.c:187: warning: 'fcstab'
 defined but not used
   CC [M]  /usr/src/zaptel-1.2.1/tor2.o
 ~SNIP~
 
 4.  Making NO edits to the Makefile for asterisk, I
 then compiled asterisk (make, make install, make
 samples, make progdocs, make config, make clean).  
 
 Looked good.
 
 5.  I then went straight to loading zaptel (having
 forgotten to first compile asterisk-sounds (make
 install) and asterisk-addons (make install) [side 
 
 question: Is it OK to go back and compile sounds and
 addons after zaptel is now loaded??])
 
 6.  I loaded zaptel sucessfully, it looks like:
 
 [EMAIL PROTECTED] ~]# 
 [EMAIL PROTECTED] ~]# modprobe zaptel
 [EMAIL PROTECTED] ~]# lsmod | grep zaptel
 zaptel206724  0 
 crc_ccitt   2113  1 zaptel
 [EMAIL PROTECTED] ~]# 
 
 
 7.  ztdummy, however, does not load:
 
 [EMAIL PROTECTED] ~]# modprobe ztdummy
 Notice: Configuration file is /etc/zaptel.conf
 line 0: Unable to open master device '/dev/zap/ctl'
 
 1 error(s) detected
 
 FATAL: Error running install command for ztdummy
 [EMAIL PROTECTED] ~]# 
 
 
 7.1  I see that i have no zap/ under /dev/:
 
 [EMAIL PROTECTED] ~]# cd /dev/zap/
 -bash: cd: /dev/zap/: No such file or directory
 [EMAIL PROTECTED] ~]# cd /dev
 [EMAIL PROTECTED] dev]# 
 [EMAIL PROTECTED] dev]# 
 [EMAIL PROTECTED] dev]# pwd
 /dev
 [EMAIL PROTECTED] dev]# ls -l
 total 0
 crw---  1 root  root36,   8 Dec 23 10:04 arpd
 lrwxrwxrwx  1 root  root  3 Dec 23 10:04 cdrom
 - hda
 crw---  1 rdlab root 5,   1 Dec 23 16:21
 console
 lrwxrwxrwx  1 root  root 11 Dec 23 10:04 core
 - /proc/kcore
 crw---  1 root  root10,  63 Dec 23 10:04
 device-mapper
 ~SNIP~
 crw---  1 vcsa  tty  7, 136 Dec 23 15:05 vcsa8
 drwx--  2 root  root 80 Dec 23 15:05
 VolGroup00
 crw---  1 root  root36,   6 Dec 23 10:04 xfrm
 lrwxrwxrwx  1 root  root  4 Dec 23 10:04 XOR
 - null
 crw---  1 root  root   196, 254 Jan  5 17:47
 zapchannel
 crw---  1 root  root   196,   0 Jan  5 17:47
 zapctl
 crw---  1 root  root   196, 255 Jan  5 17:47
 zappseudo
 crw---  1 root  root   196, 253 Jan  5 17:47
 zaptimer
 crw-rw-rw-  1 root  root 1,   5 Dec 23 10:04 zero
 [EMAIL PROTECTED] dev]# 
 
 
 7.2 Per my understanding of the asterisk docs, I
 belive I have to solve this ztdummy load issue BEFORE
 I attempt to load 

Re: [Asterisk-Users] GSM Gateway / Terminal for sale

[Adrian Carter]

Is anyone aware of the details of this in Australia?

I'd love to be able to let tech's have calls route straight to their
mobiles when 'in-house'

Steve Kennedy wrote:


On Fri, Jan 06, 2006 at 01:23:26PM -, Chris Bagnall wrote:

 

I don't get it. What is the advantage of using a GSM gateway? 
VOIP calls are pretty inexpensive as they are now.
 


It largely depends on the country you're calling. Here in the UK, calls to
mobiles are maintained at an artificially high rate because the terminating
network (the mobile networks) get a cut of call revenue for calls *to* your
mobile. By contrast, in the US, the mobile customer often pays a small
charge per minute on incoming calls (as I understand the market over there).
You'll also find in the UK the mobile phone market is heavily subsidized by
the networks such that you can get phones for free if you sign up to 12
month contracts. I often find that it's cost-effective to get a new contract
every 12 months (with a free phone), even if I don't want the phone. Flog
the phone on ebay and you've got a spare SIM with lots of inclusive minutes
for almost nothing.
   



In the UK the wholesale rates are set by Ofcom (like the FCC), which
works out about 7p'ish per minute.

However the operators can offer retail bundles (including phones) and
for a monthly contract they throw in various ammounts of cross network
minutes (or free to their own network or whatever). With clever
dial-plans and multiple terminals connected to multiple networks you can
generally get free calls to mobile users (basically clever least cost
routing, time of day sometimes needs to be taken into account as well).

However there are some disadvantages, the main being you cant set CLI of
the outgoing call as it will always be tied to the SIM of the mobile
terminal.

Another is that you can NOT run a GSM gateway (as they're known) for 3rd
parties. So if you want to connect your office PBX to a gateway to make
use of cheap mobile termination for your own company that's fine, but as
an ITSP (or traditional telco) you can not allow 3rd party traffic to
utilise a gateway. If networks find you are using a gateway (as a telco)
they can cut it off, no questions asked. Gateways have been determined
to be fixed infrastructure, therefore NOT mobile.

There is (or maybe was by now) an Ofcom consultation asking whether this
should be changed, the mobile operators will fight it, telcos and other
users will be asking for it to be changed.

Of course this is UK specific, other countries have more lenient
policies (I think Belgium allow gateways, France doesn't allow any kind,
and some allow them with the co-operation of the operators).


Steve

 



--
Adrian Carter
Technical Manager
Leading Edge Internet

Web   http://www.lei.net.au http://support.lei.net.au
Direct+61 2 6163 6162  Support 1 300 662 415
E-mail[EMAIL PROTECTED]
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[Asterisk-Users] Announcing a call transfer

[Michael Sampson]

With our current pbx system, a call comes in from the PSTN to the 
receptionist. She then hits flash, which puts the caller on hold, calls 
my extension, says so and so is on the phone for you, I say ok put 
him through, she hangs up and I am connected to the caller.


With [EMAIL PROTECTED] I can it # then the extension to transfer to and it 
will ring there. But is there a simple way to announce the call before 
you transfer it. If not, does anyone have any good work arounds for this.


--
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000

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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

[Jean-Michel Hiver]

However there are some disadvantages, the main being you cant set CLI of
the outgoing call as it will always be tied to the SIM of the mobile
terminal.
 


That's true. You can however choose to mask the caller ID.



Another is that you can NOT run a GSM gateway (as they're known) for 3rd
parties. So if you want to connect your office PBX to a gateway to make
use of cheap mobile termination for your own company that's fine, but as
an ITSP (or traditional telco) you can not allow 3rd party traffic to
utilise a gateway. If networks find you are using a gateway (as a telco)
they can cut it off, no questions asked. Gateways have been determined
to be fixed infrastructure, therefore NOT mobile.
 

Yes, mobile grey routing is illegal in the UK. However it DOES happen in 
the UK, and on a large scale (you're talking dozens of E1s worth of 
capacity), I can guarantee you. I've seen it!




Of course this is UK specific, other countries have more lenient
policies (I think Belgium allow gateways, France doesn't allow any kind,
and some allow them with the co-operation of the operators).
 

France fully allows GSM gateways. In fact one of the leading IP/GSM 
manufacturer, Quescom, is French. Their latest product, the SIM server, 
is just mad: it is able so auto-swap SIM cards and IMEI remotely to 
simulate somebody roaming around and stay below mobile providers' radar.


The ARCEP (France's flavor of regulators) solution to the problem is to 
force biggest mobile phone companies to lower their off net wholesale 
rates (over a span of 3 years) until it closes the GSM gateway economic 
space.



Cheers,
Jean-Michel.

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[Asterisk-Users] Re: FATAL: Error running install command for ztdummy

[Tony Mountifield]

In article [EMAIL PROTECTED],
Tom [EMAIL PROTECTED] wrote:
 
 Here is the issue:
 
 [EMAIL PROTECTED] ~]# modprobe zaptel
 [EMAIL PROTECTED] ~]# lsmod | grep zaptel
 zaptel206724  0 
 crc_ccitt   2113  1 zaptel
 [EMAIL PROTECTED] ~]# 
 [EMAIL PROTECTED] ~]# 
 [EMAIL PROTECTED] ~]# modprobe ztdummy
 Notice: Configuration file is /etc/zaptel.conf
 line 0: Unable to open master device '/dev/zap/ctl'
 
 1 error(s) detected
 
 FATAL: Error running install command for ztdummy
 [EMAIL PROTECTED] ~]# 

Take a look at the file README.udev in the zaptel directory.
You have probably omitted to perform the steps it describes.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[Asterisk-Users] IAX2-SIP dropped calls

[Adam Moffett]

Apparently we've been having calls sporadically drop.  We're using an 
IAX outbound trunk and SIP adapters on the inside.


Below is a log excerpt detailing one of the calls which dropped, and it 
looks largely normal to me except for this:


Jan  5 13:31:07 DEBUG[3776] channel.c: Didn't get a frame from channel: 
IAX2/teliax-2
Jan  5 13:31:07 DEBUG[3776] channel.c: Bridge stops bridging channels 
IAX2/teliax-2 and SIP/davidblanco-e02c


Can missing one IAX frame result in a dropped call?  Seems pretty 
fragile if that's the case.  Would enabling the jitter buffer mitigate 
this?  Any other suggestions?










Jan  5 13:29:51 VERBOSE[29852] logger.c: -- Accepting 
UNAUTHENTICATED call from 208.139.204.245:

   requested format = ulaw,
   requested prefs = (g729|ulaw|g726|gsm),
   actual format = gsm,
   host prefs = (gsm|ulaw),
   priority = mine
Jan  5 13:29:51 VERBOSE[3776] logger.c: -- Executing 
Dial(IAX2/teliax-2, SIP/davidblanco|30|tr) in new stack

Jan  5 13:29:51 DEBUG[3776] chan_sip.c: Setting NAT on RTP to 524288
Jan  5 13:29:51 DEBUG[3776] chan_sip.c: Outgoing Call for davidblanco
Jan  5 13:29:51 VERBOSE[3776] logger.c: -- Called davidblanco
Jan  5 13:29:51 DEBUG[29854] chan_sip.c: (Provisional) Stopping 
retransmission (but retaining packet) on 
'[EMAIL PROTECTED]' Request 102: Found
Jan  5 13:29:51 DEBUG[29854] chan_sip.c: (Provisional) Stopping 
retransmission (but retaining packet) on 
'[EMAIL PROTECTED]' Request 102: Found
Jan  5 13:29:51 VERBOSE[3776] logger.c: -- SIP/davidblanco-e02c is 
ringing

Jan  5 13:29:57 DEBUG[29854] chan_sip.c: Acked pending invite 102
Jan  5 13:29:57 DEBUG[29854] chan_sip.c: build_route: Contact hop: 
sip:[EMAIL PROTECTED]:5060
Jan  5 13:29:57 VERBOSE[3776] logger.c: -- SIP/davidblanco-e02c 
answered IAX2/teliax-2

Jan  5 13:29:57 DEBUG[29852] chan_iax2.c: Ooh, voice format changed to 2
Jan  5 13:29:59 DEBUG[29852] chan_iax2.c: Peer lastms 70, historicms 70, 
maxms 2000
Jan  5 13:30:15 DEBUG[29852] chan_iax2.c: Peer lastms 28, historicms 28, 
maxms 2000
Jan  5 13:30:59 DEBUG[29852] chan_iax2.c: Peer lastms 71, historicms 71, 
maxms 2000
Jan  5 13:31:07 DEBUG[29852] chan_iax2.c: Immediately destroying 2, 
having received hangup
Jan  5 13:31:07 DEBUG[3776] channel.c: Didn't get a frame from channel: 
IAX2/teliax-2
Jan  5 13:31:07 DEBUG[3776] channel.c: Bridge stops bridging channels 
IAX2/teliax-2 and SIP/davidblanco-e02c
Jan  5 13:31:07 DEBUG[3776] chan_sip.c: update_call_counter(davidblanco) 
- decrement call limit counter

Jan  5 13:31:07 DEBUG[3776] app_dial.c: Exiting with DIALSTATUS=ANSWER.
Jan  5 13:31:07 VERBOSE[3776] logger.c:   == Spawn extension (default, 
6078210976, 1) exited non-zero on 'IAX2/teliax-2'
Jan  5 13:31:07 DEBUG[3776] cdr_addon_mysql.c: cdr_mysql: inserting a 
CDR record.

***CDR STUFF OMITTED***
Jan  5 13:31:07 DEBUG[3776] chan_iax2.c: We're hanging up IAX2/teliax-2 
now...
Jan  5 13:31:07 DEBUG[3776] chan_iax2.c: Really destroying IAX2/teliax-2 
now...

Jan  5 13:31:07 VERBOSE[3776] logger.c: -- Hungup 'IAX2/teliax-2'

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Re: [Asterisk-Users] Recording Calls at the phone

[Francesco Peeters (Asterisk)]

On Fri, January 6, 2006 15:37, Michael Sampson said:
 I work for a call center and we are looking at using asterisk to have
 our operators take calls. Our message taking software records all the
 calls on the operators computers. Right now we use these recording
 controls from radio shack that plug in between the wall jack and the
 phone and plug in via a 1/8 inch stereo connector to the mic input on
 the computer. If I buy an IP phone I can't do that. I could get an FXO
 adapter and regular phones, but I'm looking to get as little equipment
 as possible. Radio shack makes a recording control that plugs in to a
 2.5 mm headset jack, but it takes batteries so thats not going to work

 Does anyone else do something similar? Does anyone have any ideas about
 what producs/setup would work for this.


Asterisk has a built in monitoring system. You can chose to do Always,
Never or On Demand monitoring, depending on your setup and dialplan

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor

Good luck!

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
  AMD Duron 1GHz - 1GB - * 1.2.1
  2 Sweex HFC-PCI cards
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Re: [Asterisk-Users] Announcing a call transfer

[C F]

Look for the option of attended transfer.

On 1/6/06, Michael Sampson [EMAIL PROTECTED] wrote:
 With our current pbx system, a call comes in from the PSTN to the
 receptionist. She then hits flash, which puts the caller on hold, calls
 my extension, says so and so is on the phone for you, I say ok put
 him through, she hangs up and I am connected to the caller.

 With [EMAIL PROTECTED] I can it # then the extension to transfer to and it
 will ring there. But is there a simple way to announce the call before
 you transfer it. If not, does anyone have any good work arounds for this.

 --
 Michael Sampson
 Information Systems Manager
 Customer Contact Services
 [EMAIL PROTECTED]
 952-936-4000

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Re: [Asterisk-Users] Announcing a call transfer

[Francesco Peeters (Asterisk)]

On Fri, January 6, 2006 15:46, Michael Sampson said:
 With our current pbx system, a call comes in from the PSTN to the
 receptionist. She then hits flash, which puts the caller on hold, calls
 my extension, says so and so is on the phone for you, I say ok put
 him through, she hangs up and I am connected to the caller.

 With [EMAIL PROTECTED] I can it # then the extension to transfer to and it
 will ring there. But is there a simple way to announce the call before
 you transfer it. If not, does anyone have any good work arounds for this.

 --

It is called attended transfer.

See http://www.voip-info.org/wiki/view/Asterisk+PBX+functions
And
http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf

HTH!

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
  AMD Duron 1GHz - 1GB - * 1.2.1
  2 Sweex HFC-PCI cards
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Re: [Asterisk-Users] Screening incoming calls.

[C F]

This can be accomplished in the DP with ChanSpy, and this bug:
http://bugs.digium.com/view.php?id=5841



On 1/6/06, Philipp von Klitzing [EMAIL PROTECTED] wrote:
 Hi!

  The PBX I'm getting ready to replace has a really nifty feature -- one
  that I'm not even sure Asterisk -can- do -- though I'm hoping to be proven
  wrong.  When a call goes to voicemail, the end-user can listen to the VM
  as it's being recorded, and can interrupt and answer the call if it's
  someone they want to talk to.
 
  Is there any way to implement this?

 Yes, I've described the voicemail live approach here:
 http://www.voip-info.org/wiki/index.php?page=Asterisk+tips+voicemail+live

 No need for ChanSpy and the manager interface, just needs MeetMe.

Thanks for that post thats a good one, just one thing, what happens if
the user doesn't want to connect to the caller? does it get saved as
VM?
Looking thru the code I couldn't see where that happens.


 Cheers, Philipp


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Re: [Asterisk-Users] Announcing a call transfer

[Adam Moffett]

With our current pbx system, a call comes in from the PSTN to the 
receptionist. She then hits flash, which puts the caller on hold, 
calls my extension, says so and so is on the phone for you, I say 
ok put him through, she hangs up and I am connected to the caller.


With [EMAIL PROTECTED] I can it # then the extension to transfer to and it 
will ring there. But is there a simple way to announce the call before 
you transfer it. If not, does anyone have any good work arounds for this.


There is a feature called attended transfer which does what you want.  
Receptionist dials the attended transfer code, followed by your 
extension.  The caller hears hold music while the receptionist announces 
the call to you.  When she hangs up you get the call.  If you hang up 
before she does, the call goes back to her.


It can be enabled in the features.conf file.  Under the [featuremap] 
section add

atxfer = code
on my system it's
atxfer =*2
so I dial *2 followed by the extension to do attended transfer.

However, I don't know anything specific to [EMAIL PROTECTED], so if it's 
different than a stock asterisk setup then I don't know.

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Re: [Asterisk-Users] Budge Tone-100 as a Ext in the LAN

[stotaro]

Grandstream has been very well detailed on the 
wiki.
www.voip-info.org

  - Original Message - 
  From: 
  luke 
  devon 
  To: Astericks 
  Sent: Friday, January 06, 2006 6:10 
  AM
  Subject: [Asterisk-Users] Budge Tone-100 
  as a Ext in the LAN
  
  HI , 
  
  I installed asterisk in fedora core 3 machine perfectly. and i have 10 
  units of GrandStream IP phone ( Budge Tone-100 ) . I wanted to know how can i 
  use it as extentions in my LAN ? Asterisk PBX alredy there. I didn't try 
  to do any configurations of any files .
  
  What are the configurations has to be made with asterisk ?
  
  Thanx in advance,
  Luke.
  Send instant messages to your online friends http://uk.messenger.yahoo.com 
  
  

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RE: [Asterisk-Users] Recording Calls at the phone

[Dean Collins]

Asterisk has call recording capabilities built in. it will offer you far
more functionality than what you currently are using (better control,
archiving and ability to export to third party analysis).

I suggest you do some research on this area of asterisk capability and
then suggest to the call centre manager you migrate this functionality
to asterisk.


Cheers,

Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Sampson
Sent: Friday, 6 January 2006 9:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Recording Calls at the phone

I work for a call center and we are looking at using asterisk to have 
our operators take calls. Our message taking software records all the 
calls on the operators computers. Right now we use these recording 
controls from radio shack that plug in between the wall jack and the 
phone and plug in via a 1/8 inch stereo connector to the mic input on 
the computer. If I buy an IP phone I can't do that. I could get an FXO 
adapter and regular phones, but I'm looking to get as little equipment 
as possible. Radio shack makes a recording control that plugs in to a 
2.5 mm headset jack, but it takes batteries so thats not going to work

Does anyone else do something similar? Does anyone have any ideas about 
what producs/setup would work for this.

-- 
Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000

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Re: [Asterisk-Users] GSM Gateway / Terminal for sale

[Steve Kennedy]

On Fri, Jan 06, 2006 at 06:48:27PM +0400, Jean-Michel Hiver wrote:

 However there are some disadvantages, the main being you cant set CLI of
 the outgoing call as it will always be tied to the SIM of the mobile
 terminal.
 That's true. You can however choose to mask the caller ID.

Yup, for telcos (in the broadest sense) offering a service, generally
people want to be able to call back the number that dialed them.

 Another is that you can NOT run a GSM gateway (as they're known) for 3rd
 parties. So if you want to connect your office PBX to a gateway to make
 use of cheap mobile termination for your own company that's fine, but as
 an ITSP (or traditional telco) you can not allow 3rd party traffic to
 utilise a gateway. If networks find you are using a gateway (as a telco)
 they can cut it off, no questions asked. Gateways have been determined
 to be fixed infrastructure, therefore NOT mobile.
 Yes, mobile grey routing is illegal in the UK. However it DOES happen in 
 the UK, and on a large scale (you're talking dozens of E1s worth of 
 capacity), I can guarantee you. I've seen it!

Of course it does, but generally the networks can find them quite
quickly (as local cells get congested) and they cut off the SIMs.

 Of course this is UK specific, other countries have more lenient
 policies (I think Belgium allow gateways, France doesn't allow any kind,
 and some allow them with the co-operation of the operators).
 France fully allows GSM gateways. In fact one of the leading IP/GSM 
 manufacturer, Quescom, is French. Their latest product, the SIM server, 
 is just mad: it is able so auto-swap SIM cards and IMEI remotely to 
 simulate somebody roaming around and stay below mobile providers' radar.

OK wrong way round there ...


Steve

-- 
NetTek Ltd  UK mob +44-(0)7775 755503
UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455
Skype/GoogleTalk/AIM stevekennedyuk / MSN [EMAIL PROTECTED]
Euro Tech News Blog http://eurotechnews.blogspot.com
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RE: [Asterisk-Users] GSM Gateway / Terminal for sale

[Robert Augustyn]

Are GSM gateways allowed in Canada?
And can we resell it?
Robert


 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Steve Kennedy
 Sent: Friday, January 06, 2006 9:17 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] GSM Gateway / Terminal for sale
 
 On Fri, Jan 06, 2006 at 01:23:26PM -, Chris Bagnall wrote:
 
   I don't get it. What is the advantage of using a GSM gateway? 
   VOIP calls are pretty inexpensive as they are now.
  It largely depends on the country you're calling. Here in the UK, 
  calls to mobiles are maintained at an artificially high 
 rate because 
  the terminating network (the mobile networks) get a cut of call 
  revenue for calls *to* your mobile. By contrast, in the US, 
 the mobile 
  customer often pays a small charge per minute on incoming 
 calls (as I understand the market over there).
  You'll also find in the UK the mobile phone market is heavily 
  subsidized by the networks such that you can get phones for free if 
  you sign up to 12 month contracts. I often find that it's 
  cost-effective to get a new contract every 12 months (with a free 
  phone), even if I don't want the phone. Flog the phone on ebay and 
  you've got a spare SIM with lots of inclusive minutes for 
 almost nothing.
 
 In the UK the wholesale rates are set by Ofcom (like the 
 FCC), which works out about 7p'ish per minute.
 
 However the operators can offer retail bundles (including 
 phones) and for a monthly contract they throw in various 
 ammounts of cross network minutes (or free to their own 
 network or whatever). With clever dial-plans and multiple 
 terminals connected to multiple networks you can generally 
 get free calls to mobile users (basically clever least cost 
 routing, time of day sometimes needs to be taken into account 
 as well).
 
 However there are some disadvantages, the main being you cant 
 set CLI of the outgoing call as it will always be tied to the 
 SIM of the mobile terminal.
 
 Another is that you can NOT run a GSM gateway (as they're 
 known) for 3rd parties. So if you want to connect your office 
 PBX to a gateway to make use of cheap mobile termination for 
 your own company that's fine, but as an ITSP (or traditional 
 telco) you can not allow 3rd party traffic to utilise a 
 gateway. If networks find you are using a gateway (as a 
 telco) they can cut it off, no questions asked. Gateways have 
 been determined to be fixed infrastructure, therefore NOT mobile.
 
 There is (or maybe was by now) an Ofcom consultation asking 
 whether this should be changed, the mobile operators will 
 fight it, telcos and other users will be asking for it to be changed.
 
 Of course this is UK specific, other countries have more 
 lenient policies (I think Belgium allow gateways, France 
 doesn't allow any kind, and some allow them with the 
 co-operation of the operators).
 
 
 Steve
 
 --
 NetTek Ltd  UK mob +44-(0)7775 755503
 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 
 7483 2455 Skype/GoogleTalk/AIM stevekennedyuk / MSN 
 [EMAIL PROTECTED] Euro Tech News Blog 
 http://eurotechnews.blogspot.com 
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RE: [Asterisk-Users] Incoming PSTN Calls - Stumped

[Aisling O'Driscoll]

Hi,

Yes InternalExtension is the context and 2093 the extension.

Just to explain something odd that’s happening (and I’m very stumped
with this)….I think my contexts are definately the reason that I
can’t interrupt the menu for incoming pstn calls to choose a submenu:

My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a ‘dummy’ extension.

sip.conf

register = username:[EMAIL PROTECTED]/2093

[provider-in]
type=peer
host=sip.provider.ie
context=onecontext

[2092]
type=peer
other stuff
context=onecontext

So the dummy extension here is ‘2093’ and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
‘onecontext’ context.

Now in my extensions.conf ‘onecontext’ includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of ‘onecontext’ is to allow outgoing access to the
PSTN.

[onecontext]
include = createmenu   //creating an IVR menu
include = createconf   //creating a conf call
etc
include = default  //used for voicemail

[createmenu]
;does something

[createconf]
;does something

;outgoing calls – main purpose of onecontext
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup

[default]

;mailbox for 2092 and other users


Now this is where the problems start! For incoming calls I tried to
do “include = incomingpstn” in ‘onecontext’ which I thought would
call a new context called ‘incomingpstn’ which would have an entry
for the dummy user. i.e.

[incomingpstn]

exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension

I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn’t work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get the incoming
calls working was to add the contents of the ‘incomingpstn’ context
to the default context i.e.

[default]

exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension

With this I can hear the MainMenu when I dial my DDI but I can’t seem
to interrupt to divert to another submenu. In the testing that I have
done the user that is making the call is 2092 registered with SER. If
I change the context of 2092 directly in sip.conf to incomingpstn,
then I can hear the menu and interrupt to go to the submenu. But
obviously then I don’t have access to the other features in Asterisk.
The point is that I’m stumped as to why it only works in the default
context and if this is the case how do I get it to call the submenu.

This is what comes up on my asterisk console:
-- Executing Dial (“SIP/2092-2829”, “SIP/[EMAIL PROTECTED]) in
new stack
-- Called [EMAIL PROTECTED]
-- Playing ‘MainMenu’ (language ‘en’)
-- other messages (not relevant I think)
== Spawn extension (outgoing, 021123456, 1) exited non-zero on
‘SIP/2092-5837’
== Spawn extension (default, 2093, 2) exited non zero etc etc

I’m very stuck on this and can’t figure it out.
Any help appreciated.

Many thanks,
Aisling.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Giovanni Miano
Sent: 05 January 2006 21:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls

Is Exist InternalExtension context ? and 2093 exten ?
2006/1/5, Aisling  [EMAIL PROTECTED]:
Hi all,
 
I am having difficulty getting incoming PSTN calls working. I have
set up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc
 
My provider told me to change my sip.conf as follows
 
register = username:[EMAIL PROTECTED]/2093  

; To receive incoming calls specify this block and replace
yourcontext for your dial plan. 
[blueface-in] 
type=peer 
host=sip.blueface.ie 
context=incomingpstn
 
And then in my extensions.conf to have something similar to the
following (or however I wanted to handle my incoming calls)
 
[incomingpstn]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)   
//press 1 for internal extensions.
 
 
This didn't work and I kept getting a 404 not found error saying the
user didn't exist. I tried creating the user in sip.conf and pointing
it to the appropriate context but that didn't work either. The only
way I can get it to work is to copy the code I had in the
'incomingpstn' context of my extension.conf to the 'default' context.
i.e.
 
[default]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)
 
Then the file would play. First of all I don't get why this is…It
doesn't even 

Re: [Asterisk-Users] Asterisk on Dell blade servers

[Walt Reed]

On Fri, Jan 06, 2006 at 02:17:47PM +, Bob Goddard said:
 On Friday 06 Jan 2006 08:11, Richard Scobie wrote:
  Supermicro do not do Opteron (or Athlon64) systems.
 
 Supermicro DO do Opteron.

Model numbers please? Searching through SuperMicro's web site shows ZERO
AMD based models. ONLY Intel.

They do have a few chassis that claim to support AMD based motherboards,
but NO superservers or motherboards.
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[Asterisk-Users] 3RD REQUEST - Any Help Is Appreciated

[casasterisk]

Is there a protocol I'm supposed to use here?  It seems that people are asking 
100 questions a day and SOMEONE is helping them, and I've posted this three 
times and not even an I Don't Know.

My third repost:

Ok, I've been trying to figure out why my [EMAIL PROTECTED] won't answer the 
lines when I can call out and the panel shows the call coming in - well 
something bizarre has happened. 
I set up inbound routing to ring my extension if a call comes in - and my 
extension rings but when I pick it up I get a dial tone. The whole time after I 
answer I hear the phone I originated the call on just ring and ring and ring, 
even though I answer the IP phone 
Ok, so then I set it to go to VM, and it does - but it's just a dial tone. 
So, why would the originating phone ring and ring if the PBX is picking up and 
routing? And why would I get dial tone on the answering phone when the incoming 
call rings to it? 
Bizarre! 
Here is the real time status from CLI: 
asterisk1*CLI 
-- Starting simple switch on 'Zap/2-1' 
-- Executing SetVar(Zap/2-1, FROM_DID=s) in new stack 
-- Executing Answer(Zap/2-1, ) in new stack 
-- Executing Wait(Zap/2-1, 0) in new stack 
-- Executing Goto(Zap/2-1, ext-local|*101|1) in new stack 
-- Goto (ext-local,*101,1) 
-- Executing Macro(Zap/2-1, vm|101) in new stack 
-- Executing Macro(Zap/2-1, user-callerid) in new stack 
-- Executing DBget(Zap/2-1, AMPUSER=DEVICE//user) in new stack 
-- DBget: varname=AMPUSER, family=DEVICE, key=/user 
-- DBget: Value not found in database. 
-- Executing DBget(Zap/2-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack 
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname 
-- DBget: Value not found in database. 
-- Executing GotoIf(Zap/2-1, 1?5) in new stack 
-- Goto (macro-user-callerid,s,5) 
-- Executing NoOp(Zap/2-1, Using CallerID ) in new stack 
-- Executing Goto(Zap/2-1, s-|1) in new stack 
-- Goto (macro-vm,s-,1) 
-- Executing VoiceMail(Zap/2-1, u101) in new stack 
-- Playing '/var/spool/asterisk/voicemail/default/101/unavail' (language 'en') 
-- Playing 'vm-intro' (language 'en') 
-- Playing 'beep' (language 'en') 
-- Recording the message 
-- x=0, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg 
format: wav49, 0x9f56790 
-- x=1, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg 
format: wav, 0x9f73680 
Any clues?


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Re: [Asterisk-Users] 3RD REQUEST - Any Help Is Appreciated

[Pete Barnwell]

On Fri, 2006-01-06 at 08:47 -0700, [EMAIL PROTECTED] wrote:
 Is there a protocol I'm supposed to use here?  It seems that people are 
 asking 100 questions a day and SOMEONE is helping them, and I've posted this 
 three times and not even an I Don't Know.
 
 My third repost:
 
 Ok, I've been trying to figure out why my [EMAIL PROTECTED] won't answer the 
 lines when I can call out and the panel shows the call coming in - well 
 something bizarre has happened. 
 I set up inbound routing to ring my extension if a call comes in - and my 
 extension rings but when I pick it up I get a dial tone. The whole time after 
 I answer I hear the phone I originated the call on just ring and ring and 
 ring, even though I answer the IP phone 
 Ok, so then I set it to go to VM, and it does - but it's just a dial tone. 
 So, why would the originating phone ring and ring if the PBX is picking up 
 and routing? And why would I get dial tone on the answering phone when the 
 incoming call rings to it? 
 Bizarre! 
 Here is the real time status from CLI: 
 asterisk1*CLI 
 -- Starting simple switch on 'Zap/2-1' 
 -- Executing SetVar(Zap/2-1, FROM_DID=s) in new stack 
 -- Executing Answer(Zap/2-1, ) in new stack 
 -- Executing Wait(Zap/2-1, 0) in new stack 
 -- Executing Goto(Zap/2-1, ext-local|*101|1) in new stack 
 -- Goto (ext-local,*101,1) 
 -- Executing Macro(Zap/2-1, vm|101) in new stack 
 -- Executing Macro(Zap/2-1, user-callerid) in new stack 
 -- Executing DBget(Zap/2-1, AMPUSER=DEVICE//user) in new stack 
 -- DBget: varname=AMPUSER, family=DEVICE, key=/user 
 -- DBget: Value not found in database. 
 -- Executing DBget(Zap/2-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack 
 -- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname 
 -- DBget: Value not found in database. 
 -- Executing GotoIf(Zap/2-1, 1?5) in new stack 
 -- Goto (macro-user-callerid,s,5) 
 -- Executing NoOp(Zap/2-1, Using CallerID ) in new stack 
 -- Executing Goto(Zap/2-1, s-|1) in new stack 
 -- Goto (macro-vm,s-,1) 
 -- Executing VoiceMail(Zap/2-1, u101) in new stack 
 -- Playing '/var/spool/asterisk/voicemail/default/101/unavail' (language 
 'en') 
 -- Playing 'vm-intro' (language 'en') 
 -- Playing 'beep' (language 'en') 
 -- Recording the message 
 -- x=0, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg 
 format: wav49, 0x9f56790 
 -- x=1, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg 
 format: wav, 0x9f73680 
 Any clues?

You'd probably do better to ask on the [EMAIL PROTECTED] list.

Pete

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Re: [Asterisk-Users] 3RD REQUEST - Any Help Is Appreciated

[Jean-Michel Hiver]

[EMAIL PROTECTED] a écrit :


Is there a protocol I'm supposed to use here?  It seems that people are asking 100 
questions a day and SOMEONE is helping them, and I've posted this three times and not 
even an I Don't Know.
 

You know, if thoushands of people had to answer I don't know, it would 
blow a bit. Your other options are to check #asterisk on freenode, or 
hire a consultant.


BTW: I don't know. Sorry :(

Cheers,
Jean-Michel.

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Re: [Asterisk-Users] Asterisk on Dell blade servers

[Mailing List]

- Original Message - 
Sent: Friday, January 06, 2006 10:44 AM

Subject: Re: [Asterisk-Users] Asterisk on Dell blade servers



On Fri, Jan 06, 2006 at 02:17:47PM +, Bob Goddard said:

On Friday 06 Jan 2006 08:11, Richard Scobie wrote:
 Supermicro do not do Opteron (or Athlon64) systems.

Supermicro DO do Opteron.


Model numbers please? Searching through SuperMicro's web site shows ZERO
AMD based models. ONLY Intel.

They do have a few chassis that claim to support AMD based motherboards,
but NO superservers or motherboards.


http://www.supermicro.com/Aplus/motherboard/


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Re: [Asterisk-Users] 3RD REQUEST - Any Help Is Appreciated

[Steve Blair]

People don't usually respond with I don't know. They just don't respond
unless they can help. This helps reduce the clutter on the list. And for the
record I do not have an answer to this issue.

[EMAIL PROTECTED] wrote:


Is there a protocol I'm supposed to use here?  It seems that people are asking 100 
questions a day and SOMEONE is helping them, and I've posted this three times and not 
even an I Don't Know.

My third repost:

Ok, I've been trying to figure out why my [EMAIL PROTECTED] won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. 
I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring and ring, even though I answer the IP phone 
Ok, so then I set it to go to VM, and it does - but it's just a dial tone. 
So, why would the originating phone ring and ring if the PBX is picking up and routing? And why would I get dial tone on the answering phone when the incoming call rings to it? 
Bizarre! 
Here is the real time status from CLI: 
asterisk1*CLI 
-- Starting simple switch on 'Zap/2-1' 
-- Executing SetVar(Zap/2-1, FROM_DID=s) in new stack 
-- Executing Answer(Zap/2-1, ) in new stack 
-- Executing Wait(Zap/2-1, 0) in new stack 
-- Executing Goto(Zap/2-1, ext-local|*101|1) in new stack 
-- Goto (ext-local,*101,1) 
-- Executing Macro(Zap/2-1, vm|101) in new stack 
-- Executing Macro(Zap/2-1, user-callerid) in new stack 
-- Executing DBget(Zap/2-1, AMPUSER=DEVICE//user) in new stack 
-- DBget: varname=AMPUSER, family=DEVICE, key=/user 
-- DBget: Value not found in database. 
-- Executing DBget(Zap/2-1, AMPUSERCIDNAME=AMPUSER//cidname) in new stack 
-- DBget: varname=AMPUSERCIDNAME, family=AMPUSER, key=/cidname 
-- DBget: Value not found in database. 
-- Executing GotoIf(Zap/2-1, 1?5) in new stack 
-- Goto (macro-user-callerid,s,5) 
-- Executing NoOp(Zap/2-1, Using CallerID ) in new stack 
-- Executing Goto(Zap/2-1, s-|1) in new stack 
-- Goto (macro-vm,s-,1) 
-- Executing VoiceMail(Zap/2-1, u101) in new stack 
-- Playing '/var/spool/asterisk/voicemail/default/101/unavail' (language 'en') 
-- Playing 'vm-intro' (language 'en') 
-- Playing 'beep' (language 'en') 
-- Recording the message 
-- x=0, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg format: wav49, 0x9f56790 
-- x=1, open writing: /var/spool/asterisk/voicemail/default/101/INBOX/msg format: wav, 0x9f73680 
Any clues?



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--
 
ISC Network Engineering

The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104  



voice: 215-573-8396 


  215-746-8001

fax: 215-898-9348


sip:[EMAIL PROTECTED]

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Re: [Asterisk-Users] Incoming PSTN Calls - Stumped

[Iqbal]

I had a similar problem , and then used GoTo instead of include

Iqbal

Aisling O'Driscoll wrote:


Hi,

Yes InternalExtension is the context and 2093 the extension.

Just to explain something odd that’s happening (and I’m very stumped
with this)….I think my contexts are definately the reason that I
can’t interrupt the menu for incoming pstn calls to choose a submenu:

My users register with my sip proxy (SER). Therefore when I create an
entry for them in sip.conf I set only one context. Also to allow for
incoming calls from my provider it seems I must direct the calls
firstly to a ‘dummy’ extension.

sip.conf

register = username:[EMAIL PROTECTED]/2093

[provider-in]
type=peer
host=sip.provider.ie
context=onecontext

[2092]
type=peer
other stuff
context=onecontext

So the dummy extension here is ‘2093’ and 2092 is a phone who
registers with SER and when SER redirects to Asterisk uses the
‘onecontext’ context.

Now in my extensions.conf ‘onecontext’ includes other contexts. This
is how I get access to conference calls, creating IVR menus etc. Also
the main purpose of ‘onecontext’ is to allow outgoing access to the
PSTN.

[onecontext]
include = createmenu//creating an IVR menu
include = createconf//creating a conf call
etc
include = default   //used for voicemail

[createmenu]
;does something

[createconf]
;does something

;outgoing calls – main purpose of onecontext
exten = _X.,1,Dial(SIP/[EMAIL PROTECTED])
exten = _X.,2,Hangup

[default]

;mailbox for 2092 and other users


Now this is where the problems start! For incoming calls I tried to
do “include = incomingpstn” in ‘onecontext’ which I thought would
call a new context called ‘incomingpstn’ which would have an entry
for the dummy user. i.e.

[incomingpstn]

exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension

I also changed the [provider-in] for context=incomingpstn in my
sip.conf. However this didn’t work and I kept getting directed to the
voicemail of my pstn provider. The ONLY way I could get the incoming
calls working was to add the contents of the ‘incomingpstn’ context
to the default context i.e.

[default]

exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)//directs to another
context called Internal Extension

With this I can hear the MainMenu when I dial my DDI but I can’t seem
to interrupt to divert to another submenu. In the testing that I have
done the user that is making the call is 2092 registered with SER. If
I change the context of 2092 directly in sip.conf to incomingpstn,
then I can hear the menu and interrupt to go to the submenu. But
obviously then I don’t have access to the other features in Asterisk.
The point is that I’m stumped as to why it only works in the default
context and if this is the case how do I get it to call the submenu.

This is what comes up on my asterisk console:
-- Executing Dial (“SIP/2092-2829”, “SIP/[EMAIL PROTECTED]) in
new stack
-- Called [EMAIL PROTECTED]
-- Playing ‘MainMenu’ (language ‘en’)
-- other messages (not relevant I think)
== Spawn extension (outgoing, 021123456, 1) exited non-zero on
‘SIP/2092-5837’
== Spawn extension (default, 2093, 2) exited non zero etc etc

I’m very stuck on this and can’t figure it out.
Any help appreciated.

Many thanks,
Aisling.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Giovanni Miano
Sent: 05 January 2006 21:09
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Incoming PSTN Calls

Is Exist InternalExtension context ? and 2093 exten ?
2006/1/5, Aisling  [EMAIL PROTECTED]:
Hi all,

I am having difficulty getting incoming PSTN calls working. I have
set up an account with a third party provider. In my system, the user
register with SER and use Asterisk for PSTN access, voicemail etc

My provider told me to change my sip.conf as follows

register = username:[EMAIL PROTECTED]/2093  


; To receive incoming calls specify this block and replace
yourcontext for your dial plan. 
[blueface-in] 
type=peer 
host=sip.blueface.ie 
context=incomingpstn


And then in my extensions.conf to have something similar to the
following (or however I wanted to handle my incoming calls)

[incomingpstn]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 1,1,Goto(InternalExtension,2093,1)   
//press 1 for internal extensions.



This didn't work and I kept getting a 404 not found error saying the
user didn't exist. I tried creating the user in sip.conf and pointing
it to the appropriate context but that didn't work either. The only
way I can get it to work is to copy the code I had in the
'incomingpstn' context of my extension.conf to the 'default' context.
i.e.

[default]
exten = 2093,1,Wait(1)
exten = 2093,n,Background(MainMenu)
exten = 

Re: [Asterisk-Users] 3RD REQUEST - Any Help Is Appreciated

[Jason Becker]

[EMAIL PROTECTED] wrote:

Is there a protocol I'm supposed to use here?  It seems that people are asking 100 
questions a day and SOMEONE is helping them, and I've posted this three times and not 
even an I Don't Know.

My third repost:

Ok, I've been trying to figure out why my [EMAIL PROTECTED] won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. 


You should be posting to the [EMAIL PROTECTED] Help forum:

http://sourceforge.net/forum/forum.php?forum_id=420324

or the AMP Help forum:

http://sourceforge.net/forum/forum.php?forum_id=414452

or amportal-users mailing list:

http://lists.sourceforge.net/lists/listinfo/amportal-users

Regards,

--
Jason Becker
Director  CEO
Coalescent Systems Inc.
Enabling Open Source Telephony
403.244.8089
www.coalescentsystems.ca
www.gabcast.com
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Re: [Asterisk-Users] Asterisk on Dell blade servers

[Bob Goddard]

On Friday 06 Jan 2006 15:44, Walt Reed wrote:
 On Fri, Jan 06, 2006 at 02:17:47PM +, Bob Goddard said:
  On Friday 06 Jan 2006 08:11, Richard Scobie wrote:
   Supermicro do not do Opteron (or Athlon64) systems.
 
  Supermicro DO do Opteron.

 Model numbers please? Searching through SuperMicro's web site shows ZERO
 AMD based models. ONLY Intel.

 They do have a few chassis that claim to support AMD based motherboards,
 but NO superservers or motherboards.

And those chassis are for Supermicro motherboards.
One (only?) mb is H8DAR-T but as you have found,
they are not listed on their website. I think they
have been available for a few months now.


B

-- 
http://www.mailtrap.org.uk/
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[Asterisk-Users] Macro DialPlan

[scott]

Hi All

I am trying to simplify a dialplan for a few thousand users.
Would what I have below work?

If someone dials exten 710001 would it go through answer and then to the macro 
to try dialing the SIP phone thats registered on 710001 and then onto voicemail 
if no answer or not signed on?


exten = 71,1,Answer()
exten = 71,2,Macro(71macro,${EXTEN})
exten = 71,3,Hangup()


[macro-71macro]

exten = s,1,Dial(SIP/${ARG1},30,tr)
exten = s,2,VoiceMail(${ARG1})
exten = s,3,PlayBack(vm-goodbye)


Many Thanks in Advance
Scott Pinhorne
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[Asterisk-Users] Annoying Notice Message: Don't know what to do with control frame 15

[Joan Bautista]

Hi, I haven't found anything about the message below on the mailing list, Does anyones knows why this notice is being appearing?
 -- Executing Dial(Local/[EMAIL PROTECTED],2, IAX2/CallOut/12365533643|30|otT) in new stack -- Called CallOut/12365533643
 -- Call accepted by 12.11.11.11 (format ulaw) -- Format for call is ulaw -- IAX2/10.11.240.110:4569-3 is proceeding passing it to 
Local/[EMAIL PROTECTED],2Jan 6 13:20:41 NOTICE[26911]: channel.c:2416 __ast_request_and_dial: Don't know what to do with control frame 15 -- IAX2/10.11.240.110:4569-3 is circuit-busy -- Hungup 'IAX2/12.11.11.11:4569-3'
 == Everyone is busy/congested at this time (1:0/1/0) -- Executing Goto(Local/[EMAIL PROTECTED],2, s-CONGESTION|1) in new stack
 -- Goto (default,s-CONGESTION,1) -- Executing NoOp(Local/[EMAIL PROTECTED],2, CONG) in new stack -- Executing Congestion(
Local/[EMAIL PROTECTED],2, ) in new stack  Channel Local/[EMAIL PROTECTED],1
 was never answered. == Spawn extension (default, s-CONGESTION, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' 
My calling scenario is like this:server01 server02 pstn server --IAX trunking-- agents/sip server 
server01: Asterisk 1.2.1server02: Asterisk 1.2.1
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Re: [Asterisk-Users] Macro DialPlan

[C F]

If you have to ask this question, please get professional help to
install this, otherwise you might end up with a few thousand users
picketing at your door.


On 1/6/06, scott [EMAIL PROTECTED] wrote:
 Hi All

 I am trying to simplify a dialplan for a few thousand users.
 Would what I have below work?

 If someone dials exten 710001 would it go through answer and then to the 
 macro to try dialing the SIP phone thats registered on 710001 and then onto 
 voicemail if no answer or not signed on?


 exten = 71,1,Answer()
 exten = 71,2,Macro(71macro,${EXTEN})
 exten = 71,3,Hangup()


 [macro-71macro]

 exten = s,1,Dial(SIP/${ARG1},30,tr)
 exten = s,2,VoiceMail(${ARG1})
 exten = s,3,PlayBack(vm-goodbye)


 Many Thanks in Advance
 Scott Pinhorne
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Re: [Asterisk-Users] SIP/IAX softphones for use in call centre environments

[Michael Van Donselaar]

Chris,

I've done several customized versions of iaxComm (including two for call
centers)

Contact me off-list if you're interested.

On Thu, 5 Jan 2006 05:37:59 -, Chris Bagnall [EMAIL PROTECTED] wrote:

I've been working my way through the softphones listed on voip-info over the
last few weeks and I've not really found anything to fit the bill. Has
anyone had more luck?

The environment is a small call centre of 5 users. Operators often need to
be able to transfer calls to other operators with different specialties, so
the softphone needs to be easy to use and quick to transfer calls. Operators
also have a full-screen web application open most of the time to assist them
with callers, so if possible, the softphone needs to either run always on
top, or (possibly) have keyboard hotkeys for common functions.

Most importantly it needs to work with 96dpi fonts (rather than Windows'
default of 72dpi). The TFTs they have are 1280x1024 and operators prefer the
larger font size. Many of the softphones I've tried end up with data
elements appearing in weird places (or not visibile at all) with the larger
font size.

Any thoughts / suggestions / pointers?

Thanks in advance.

Regards,

Chris
-- 
C.M. Bagnall, Director, Minotaur I.T. Limited
This email is made from 100% recycled electrons


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Re: [Asterisk-Users] Macro DialPlan

[Nathan Alberti]

Don't forget, patterns (for matching) must begin with an underscore (_)

I find it nicer to just use ${MACRO_EXTEN} rather than declaring $ 
{ARG1} for the sake of it.


-
exten = _71,1,Answer()
exten = _71,2,Macro(71macro)
exten = _71,3,Hangup()

[macro-71macro]
exten = s,1,Dial(${MACRO_EXTEN},30,t) 	; Ring the  
interface, 20 seconds maximum
exten = s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status  
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = s-BUSY,1,Voicemail(bs${MACRO_EXTEN})   ; If busy, send  
to voicemail w/ busy announce
; you may want to use a variable such as ${MACRO_OFFSET} to exit back  
to  where you called the macro in the dialplan with a Goto().
exten = s-NOANSWER,1,Voicemail(us${MACRO_EXTEN})   ; If unavailable,  
send to voicemail w/ unavail announce
; you may want to use a variable such as ${MACRO_OFFSET} to exit back  
to  where you called the macro in the dialplan with a Goto().
exten = _s-.,1,Goto(s-NOANSWER,1)  ; Treat anything else  
as no answer

-

btw.. its late and I have not double checked it so expect it not to  
work :)


Regards,

Nathan



exten = 71,1,Answer()
exten = 71,2,Macro(71macro,${EXTEN})
exten = 71,3,Hangup()


[macro-71macro]

exten = s,1,Dial(SIP/${ARG1},30,tr)
exten = s,2,VoiceMail(${ARG1})
exten = s,3,PlayBack(vm-goodbye)







On 06/01/2006, at 6:27 PM, scott wrote:


Hi All

I am trying to simplify a dialplan for a few thousand users.
Would what I have below work?

If someone dials exten 710001 would it go through answer and then  
to the macro to try dialing the SIP phone thats registered on  
710001 and then onto voicemail if no answer or not signed on?



exten = 71,1,Answer()
exten = 71,2,Macro(71macro,${EXTEN})
exten = 71,3,Hangup()


[macro-71macro]

exten = s,1,Dial(SIP/${ARG1},30,tr)
exten = s,2,VoiceMail(${ARG1})
exten = s,3,PlayBack(vm-goodbye)


Many Thanks in Advance
Scott Pinhorne
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Re: [Asterisk-Users] Fax with Asterisk and Sipura 2100

[Jorge Cisneros]

Change the RTP Packet Size: 0.010

to 

RTP Packet Size: 0.020

Asterisk only work with 2 frames. 

I can't send any fax with other values.






On 1/5/06, Joash Herbrink [EMAIL PROTECTED] wrote:
You could use a cisco ata 186.There aren't very cheap, but I have made them work on several of mycustomer sites with faxes.The ata just registers to the * server as a SIP endpoint.Also, echo cancelling and other intelligent things are bad when
dealing with faxes and modems.Just use the cisco ATA (or any simple vegastream ATA device) to send /receive faxes.Codec should always be G.711 and no CNG or VAD or echo canceling shouldbe used, fax machines take care of that themselves.
(Contact me of list if you any of the mentioned devices)Joash[EMAIL PROTECTED]-Original Message-From: 
[EMAIL PROTECTED][mailto:[EMAIL PROTECTED]] On Behalf Of RemcoBarendeSent: Thursday, January 05, 2006 4:42 PM
To: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Fax with Asterisk and Sipura 2100I tried to get it working for a very long time (over a year) with everypossible set of config parameters I could find both for * as well as for
the Sipura's. Echo cancelling etc. etc. all changed but still problems.I tried to get it working on an * box with a BRI line.Finally I have given up and attached a traditional ISDN - Analog (A/B)
converter to the ISDN line for the faxing bit next to Asterisk.I have yet to find a similar solution for faxing with a PRI, I'm afraiditwill be impossible because as far as I know it's not possible to hook up
some sort of A/B adapter next to the * box on one pri line.I think it can work if your fax machines are capable of capping faxtx/rxspeeds to 9600 baud maximum without error correction. However it seems
that not a single producer of FAX equipment (be it modems, all-in-onedevices or even dedicated fax machines) offer such an option. HP doesn'tseem very interested in capping the fax speeds for their all-in-one
thingies.All fax products keep trying to transmit/receive at higher speedsafter which the fax will fail completely or after the second page.Maybe there is a solution coming for PRI faxing. Junghanns informed me
some time ago that they were working on a PRI card with a possibility tosync the clock to other cards.If this works in theory you could use a Junghanns PRI card and aJunghannsBRI card, sync the clocks and keep the path fully digital without lost
frames. On their website however they only mention the possibility tointerconnect the PRI cards, not (yet?) PRI - BRI.On Thu, 5 Jan 2006, Darrell Long wrote: I know the subject of faxing has been covered in some detail, but I
was wondering if anyone has a hardware configuration similar to ours thathas faxes working successfully and would be willing to share any settings/insight. We are unable to fax reliably with a Sipura 2100 connected to
Asterisk. We do not route calls over the Internet and our network has very lowlatency. The Asterisk servers connect to Cisco Routers that have PRIs from various carriers. We have all the recommended settings in the Sipura ATA, with
Echo Cancellation and Silence Suppression off, uLaw only for the codec,etc. While I realize that no faxes going through passthrough like this willwork 100% of the time, we currently have a less than 40% success rate with
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[Asterisk-Users] Problem with Call Monitoring

[Waldo Rubinstein]

I'm running * 1.2.1 on Slackware.

I have several queues configured to record incoming calls once  
answered (without joining the in and out files). Yesterday, I showed  
my agents how to transfer a call received from a queue to another agent.


What I realized today is that when listening to some of the  
recordings, I can hear the agent answer the call, speak to the  
customer and then say to hold a moment while she transfered the call  
to the assigned agent for that customer.


This is fine, except that the recording stops there. What I think  
it's happening is that when a call is transfered, it stops recording  
the call.


I don't know if this is a bug or the way it's supposed to work.

Additionally, my agents are using eyeBeam softphone and they are  
using the XFER button of the softphone, instead of the '#' key or any  
other * related soft key.


Please advise.

Thanks,
Waldo
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Re: [Asterisk-Users] Recording Calls at the phone

[Michael Sampson]

Since not all of our operators are going through asterisk I can't switch 
over to using asterisk. I agree that it is a much better system to 
record the calls at the server, but thats just not an option. The call 
recording software we use now is too integrated into our message taking 
system not to use. Also the operators just make one 8 hour phone call 
into our message taking system to get their remote audio so asterisk 
would just record that as one long call, which won't work either. Anyone 
have any other ideas.


Michael Sampson
Information Systems Manager
Customer Contact Services
[EMAIL PROTECTED]
952-936-4000



Dean Collins wrote:


Asterisk has call recording capabilities built in. it will offer you far
more functionality than what you currently are using (better control,
archiving and ability to export to third party analysis).

I suggest you do some research on this area of asterisk capability and
then suggest to the call centre manager you migrate this functionality
to asterisk.


Cheers,

Dean


-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Sampson
Sent: Friday, 6 January 2006 9:38 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Recording Calls at the phone

I work for a call center and we are looking at using asterisk to have 
our operators take calls. Our message taking software records all the 
calls on the operators computers. Right now we use these recording 
controls from radio shack that plug in between the wall jack and the 
phone and plug in via a 1/8 inch stereo connector to the mic input on 
the computer. If I buy an IP phone I can't do that. I could get an FXO 
adapter and regular phones, but I'm looking to get as little equipment 
as possible. Radio shack makes a recording control that plugs in to a 
2.5 mm headset jack, but it takes batteries so thats not going to work


Does anyone else do something similar? Does anyone have any ideas about 
what producs/setup would work for this.


 


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Re: [Asterisk-Users] Problem with show channels

[Matt Florell]

show channels concise

it spits out colon : delimited fields with lots of information

That was one of the more frustrating changes from 1.0 to 1.2 but in
the end it provides much more data. Just beware if you use SIP or IAX
trunks that have colons in them, it will throw off the order of the
fields after it. I just submitted a patch for that to the bug tracker.
http://bugs.digium.com/view.php?id=6086

MATT---


On 1/6/06, Jerry Geis [EMAIL PROTECTED] wrote:
  All,

  when I do show channels:

  Channel  Location State   Application(Data)
  SIP/201-e478 (None)   Up  Bridged
 Call(IAX2/muncie_to_ge
  IAX2/muncie_to_georg [EMAIL PROTECTED]:7  Up  Dial(SIP/201|20)

  I am getting TRUNCATED call information
  the IAX2/muncie_to_ge is truncated. How do I
  get the need call information to transfer the call.

  Jerry


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RE: [Asterisk-Users] OT: SIP aware firewalls?

[Michael Graves]

On Thu, 5 Jan 2006 17:57:47 +0100, Erwin de Raad wrote:


 You should be able to run SIP through m0n0wall quite happily - we have a
 number of client sites with SIP phones offsite which connect to the *
server
 behind a m0n0wall box. You'll need to allow 5060 (UDP) for SIP, then an
 appropriate port range (as definted in rtp.conf) for the RTP streams.

 You'll obviously also need to apply any QoS rules to both the SIP and RTP
 streams.


Totally agree. I moved from Kerio WinRoute (claims to be SIP aware  not) to
Monowall and all SIP/NAT issues went away.
It doesn't do QoS but you can do bandwith/traffic shaping which also should
work fine.


Surely there's something more to the truly SIP-aware device, such as
the Ingate IX66, that merits their use in some specific circumstances?

I truly love my m0n0wall. It's been 100% solid and totally managable,
even for a relative novice such as myself. I don't generally have
problems with getting the mechanics of SIP setup through m0n0. But I
thought that there must be some advantage to the proxy services
provided in SIP aware devices or they simply wouldn't exist.

I know that I can stay with m0n0. The question still stands; are there
circumstances when something more is required? Would something be
gained by such a migration.

Michael

--
Michael Graves   [EMAIL PROTECTED]
Sr. Product Specialist  www.pixelpower.com
Pixel Power Inc. [EMAIL PROTECTED]

o713-861-4005
o800-905-6412
c713-201-1262
fwd 54245



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Re: [Asterisk-Users] Problem with show channels

[Kevin P. Fleming]

Jerry Geis wrote:


I am getting TRUNCATED call information
the IAX2/muncie_to_ge is truncated. How do I
get the need call information to transfer the call.


'show channels' is used for human-readable output on a console screen. 
If you need the information in a complete form for some automated 
purpose, use the manager interface.

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[Asterisk-Users] Problem with show channels

[Jerry Geis]

Jerry Geis wrote:


/ I am getting TRUNCATED call information

// the IAX2/muncie_to_ge is truncated. How do I
// get the need call information to transfer the call.
/
'show channels' is used for human-readable output on a console screen. 
If you need the information in a complete form for some automated 
purpose, use the manager interface.


Kevin can you be more precise? I am using the manager interface 
and the command show channels. It is truncating the data.


Is there another wayin the manager API I'm not aware of to get this
information?

jerry


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RE: [Asterisk-Users] Dialer

[Wiley Siler]

If this or any other example is available, I would be most thankful to
have it.

I got the go ahead on this project to day so now I have to start seeing
how to do this.

Thanks,
Wiley
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Darren
Wiebe
Sent: Tuesday, January 03, 2006 5:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Dialer

I'm supposed to have a mostly canned script that will do this done
already.  It will pull the list of people to call out of a db and play
them the file specified in the db table.  Contact me offlist if you're
interested.  It will be done real soon but I'm not done testing yet.

Darren Wiebe
[EMAIL PROTECTED]

Kerry Garrison wrote:

 You actually aren't far from it. If the system only needs to play the 
 same file to each person, a simple script can be used to pull from a 
 database and create call files. Asterisk will use the call files to 
 place the calls and play a sound. A few minutes of searching on that 
 should get you started. I haven't seen anyone else have a canned 
 script ready to go, but would like to know if anyone does.
 -Kerry
  



 *From:* [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] *On Behalf Of
 *Wiley Siler
 *Sent:* Tuesday, January 03, 2006 3:32 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [Asterisk-Users] Dialer

 Hello All,

 I am having trouble finding a specific * piece of software so I
 thought I would see If you guys can help me get my terminology
clear.

 First off let me premise this with no, this is absolutely not for
 doing call marketing.
 I need to make my Asterisk box call a group of people and play
 them a message.
 My company deals with education so we need to do follow ups if
 students are not logging on.
 We do this manually now but it would be easier and cheaper to just
 play them a message.

 What is the term I should be looking for?  I keep thinking auto
 dialer or something like that but I am not quite getting there.

 Any help would be appreciated.  I have been learning a bit of Perl
 so I was thinking I could auto generate and AGI file and then just
 do a Play() of the mp3 when they pick up at the other end?  Seems
 a little kludge though.


 Thanks,
 Wiley


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--
Darren Wiebe
[EMAIL PROTECTED]
Aleph Communications
ASTPP - Open Source Voip Billing  Calling Cards www.aleph-com.net/astpp

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Re: [Asterisk-Users] PHP Manager

[Alex Montoanelli]

try this
?php
$socket = fsockopen(localhost,5038, $errno, $errstr, $timeout);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: 1212\r\n);
fputs($socket, Secret: 1212\r\n\r\n);
fputs($socket, Action: Command\r\n);
fputs($socket, Command: reload\r\n\r\n);
* fputs($socket, Action: Command\r\n);*
fputs($socket, Command: show channels\r\n\r\n);
$wrets=fgets($socket,128);

?

Code Lover wrote:

Hi all,

I have a small problem to execute Asterisk Commands in Asterisk
Manager using PHP.
I am able to run all Asterisk Manager command but the problem is
comming with asterisk command.

here is the code i am trying to run.

?php
 $socket = fsockopen(localhost,5038, $errno, $errstr, $timeout);
 fputs($socket, Action: Login\r\n);
 fputs($socket, UserName: 1212\r\n);
 fputs($socket, Secret: 1212\r\n\r\n);
 fputs($socket, Action: Command\r\n);
 fputs($socket, Command: reload\r\n\r\n); #Working well
 fputs($socket, Command: show channels\r\n\r\n); #Not working Working well
 fputs($socket, Command: 'show channels'\r\n\r\n); #Not working Working well
 $wrets=fgets($socket,128);

?



If you see in my code when i am calling only reload command working
but when i am trying to call piar command it is just prompting :
== Manager '1212' logged off from localhost

without showing channels

Please advice me to solve this problem.
--
Thank You,
Code Lover
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[Asterisk-Users] Alphanumeric pattern match in extensions.conf

[Dan Austin]

I need to match an incoming call based on a prefixed string, and this
solution was suggested to me some time back.

exten = _conf.,1,Answer
exten = _conf.,2,MeetMe(${EXTEN:4}|d)
exten = _conf.,3,Hangup

However incoming calls never match this pattern, and I cannot
find any evidence in the wiki or on google that such a pattern
is valid.  I'm currently running a SVN trunk, but have tested
with 1.0.X and 1.2.X.

Is anyone using alphanumeric patterns in their dialplan?

Dan
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Re: [Asterisk-Users] Alphanumeric pattern match in extensions.conf

[Sergey Okhapkin]

The match doesn't work because n in conf will never match to the
letter n (it's a pattern for a digit).

try _co[n]f. instead.

On Fri, 2006-01-06 at 10:33 -0800, Dan Austin wrote:
 I need to match an incoming call based on a prefixed string, and this
 solution was suggested to me some time back.
 
 exten = _conf.,1,Answer
 exten = _conf.,2,MeetMe(${EXTEN:4}|d)
 exten = _conf.,3,Hangup
 
 However incoming calls never match this pattern, and I cannot
 find any evidence in the wiki or on google that such a pattern
 is valid.  I'm currently running a SVN trunk, but have tested
 with 1.0.X and 1.2.X.
 
 Is anyone using alphanumeric patterns in their dialplan?
 
 Dan
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RE: [Asterisk-Users] Dialer

[Ben Higley]

A really neat thing about this, you could make it interactive, and also
post the response back from each user on if they accepted it or not. and
then call them back in 5 min again :) LOL

But someone could be seeing what the system is doing realtime...
./Ben



 Hello All,

 I am having trouble finding a specific * piece of software so I
 thought I would see If you guys can help me get my terminology
 clear.

 First off let me premise this with no, this is absolutely not for
 doing call marketing.
 I need to make my Asterisk box call a group of people and play
 them a message.
 My company deals with education so we need to do follow ups if
 students are not logging on.
 We do this manually now but it would be easier and cheaper to just
 play them a message.

 What is the term I should be looking for?  I keep thinking auto
 dialer or something like that but I am not quite getting there.

 Any help would be appreciated.  I have been learning a bit of Perl
 so I was thinking I could auto generate and AGI file and then just
 do a Play() of the mp3 when they pick up at the other end?  Seems
 a little kludge though.


 Thanks,
 Wiley


---
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 --
 Darren Wiebe
 [EMAIL PROTECTED]
 Aleph Communications
 ASTPP - Open Source Voip Billing  Calling Cards www.aleph-com.net/astpp

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RE: [Asterisk-Users] Dialer

[Wiley Siler]

Just to make it easy, I will be reading the caller list from a another
server via a web page, parsing it and dialing.
After each pass, I just post back to the server web page and it updates
the other system.
Our tech just needs to review the log once daily.

W
 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Higley
Sent: Friday, January 06, 2006 11:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Dialer


A really neat thing about this, you could make it interactive, and also
post the response back from each user on if they accepted it or not. and
then call them back in 5 min again :) LOL

But someone could be seeing what the system is doing realtime...
./Ben



 Hello All,

 I am having trouble finding a specific * piece of software so I
 thought I would see If you guys can help me get my terminology
 clear.

 First off let me premise this with no, this is absolutely not
for
 doing call marketing.
 I need to make my Asterisk box call a group of people and play
 them a message.
 My company deals with education so we need to do follow ups if
 students are not logging on.
 We do this manually now but it would be easier and cheaper to
just
 play them a message.

 What is the term I should be looking for?  I keep thinking auto
 dialer or something like that but I am not quite getting there.

 Any help would be appreciated.  I have been learning a bit of
Perl
 so I was thinking I could auto generate and AGI file and then
just
 do a Play() of the mp3 when they pick up at the other end?  Seems
 a little kludge though.


 Thanks,
 Wiley


--
-
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 --
 Darren Wiebe
 [EMAIL PROTECTED]
 Aleph Communications
 ASTPP - Open Source Voip Billing  Calling Cards 
 www.aleph-com.net/astpp

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Re: [Asterisk-Users] Problem with show channels

[Kevin P. Fleming]

Jerry Geis wrote:


Is there another wayin the manager API I'm not aware of to get this
information?


No, I was mistaken. Matt Florell's response about using 'show channels 
concise' is probably the best way to go, since it produces output 
designed for automated interpretation.

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[Asterisk-Users] SPA-3000 is translating vocal sounds into DTMF

[Brian Capouch]

I'm sure there must be a setting I'm missing somewhere, so I thought I 
might was well ask here.


Conversations are punctuated by sudden replacement of a given syllable 
or so of conversation with a DTMF tone.


I would hope perhaps there's some kind of setting that has to do with 
the way it detects inband DTMF?  I'm pretty sure it's an artifact of 
this particular ATA; my other SIP devices are just fine.


Thx.

B.
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Re: [Asterisk-Users] Asterisk on Dell blade servers

[Richard Scobie]

Dean Collins wrote:

Lol, so Dell must be doing the same thing.

Did you ever consider that Supermicro are an enterprise setup to make
money, and that possibly their financial interests are served by
sticking with Intel?


Absolutely. However, it looks as though their lack of AMD product is 
finally hurting enough for them to do something.


http://www.forbes.com/technology/feeds/afx/2005/11/20/afx2347168.html

To Bob,

My apologies. I had spent a bit of time recently looking for Opteron 
systems on their site without success. The fact that they do not seem to 
feel them worthy of mention on their home page I regard as an indication 
of their commitment. The page listing their AMD boards :


http://www.supermicro.com/Aplus/motherboard/

is headed For OEM Customers, so I take it from that I cannot order one 
from my local supplier.


Regards,

Richard
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[Asterisk-Users] Not Able to Connect Two Asterisk Servers Using IAX2

[Chandan Mishra]

Hi I have two asterisk servers. I just want to connect two asterisk server using IAX2.But the Asterisk Servers are not able to register each other. If some body have done thisthen Please send me the configuration they have done in 
iax.conf and extensions.conf.I simply want to connect and call from one sever to another.ThanksChandan Kumar MishraSoftware Engg.
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RE: [Asterisk-Users] Problem with show channels

[Douglas Garstang]

I have a question on this. It isn't readily obvious to me, upon issueing a 'sip 
show channels' command which call legs are related to which call.

For example:

*CLI sip show channels
Peer User/ANRCall ID  Seq (Tx/Rx)  Form  Hold Last 
Message   
192.168.10.121   a00090601   29a98b1708f  00102/0  g729  No   Tx: ACK   
 
192.168.10.4 a00090301   c08e095b-c1  00101/1  g729  No   Rx: ACK   
  

Apart from the fact it's obvious here because there's one call, how can you 
determine that these are the same call? The 'show channels' command is a little 
easier, but still cryptic. It appears that the format isn't standard and 
interpreting this from a script would be difficult. It would be nice if some 
identifier was printed. Maybe 'From number' and 'To number' or the call-id for 
the call.

*CLI show channels
Channel  Location State   Application(Data) 
SIP/a00090601-14cf   (None)   Up  Bridged Call(SIP/a00090301-403
SIP/a00090301-4033   [EMAIL PROTECTED]:1   Up  Dial(SIP/a00090601|20|tr)
 
2 active channels
1 active call


-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Friday, January 06, 2006 12:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Problem with show channels


Jerry Geis wrote:

 Is there another wayin the manager API I'm not aware of to get this
 information?

No, I was mistaken. Matt Florell's response about using 'show channels 
concise' is probably the best way to go, since it produces output 
designed for automated interpretation.
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Re: [Asterisk-Users] Dialer

[Jonathan Attwood]

Here, this may be of use:

http://mundy.org/blog/index.php?p=95
On 1/6/06, Wiley Siler [EMAIL PROTECTED] wrote:
If this or any other example is available, I would be most thankful tohave it.I got the go ahead on this project to day so now I have to start seeing
how to do this.Thanks,Wiley-Original Message-From: [EMAIL PROTECTED][mailto:
[EMAIL PROTECTED]] On Behalf Of DarrenWiebeSent: Tuesday, January 03, 2006 5:00 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Dialer
I'm supposed to have a mostly canned script that will do this donealready.It will pull the list of people to call out of a db and playthem the file specified in the db table.Contact me offlist if you're
interested.It will be done real soon but I'm not done testing yet.Darren Wiebe[EMAIL PROTECTED]Kerry Garrison wrote: You actually aren't far from it. If the system only needs to play the
 same file to each person, a simple script can be used to pull from a database and create call files. Asterisk will use the call files to place the calls and play a sound. A few minutes of searching on that
 should get you started. I haven't seen anyone else have a canned script ready to go, but would like to know if anyone does. -Kerry
 *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
] *On Behalf Of *Wiley Siler *Sent:* Tuesday, January 03, 2006 3:32 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [Asterisk-Users] Dialer
 Hello All, I am having trouble finding a specific * piece of software so I thought I would see If you guys can help me get my terminologyclear. First off let me premise this with no, this is absolutely not for
 doing call marketing. I need to make my Asterisk box call a group of people and play them a message. My company deals with education so we need to do follow ups if
 students are not logging on. We do this manually now but it would be easier and cheaper to just play them a message. What is the term I should be looking for?I keep thinking auto
 dialer or something like that but I am not quite getting there. Any help would be appreciated.I have been learning a bit of Perl so I was thinking I could auto generate and AGI file and then just
 do a Play() of the mp3 when they pick up at the other end?Seems a little kludge though. Thanks, Wiley---
-___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:
 http://lists.digium.com/mailman/listinfo/asterisk-users--Darren Wiebe
[EMAIL PROTECTED]Aleph CommunicationsASTPP - Open Source Voip Billing  Calling Cards www.aleph-com.net/astpp___
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Re: [Asterisk-Users] Recording Calls at the phone

[brett]

On 1/6/2006, Michael Sampson [EMAIL PROTECTED] wrote:
 Since not all of our operators are going through asterisk I can't switch
 over to using asterisk. I agree that it is a much better system to
 record the calls at the server, but thats just not an option. The call
 recording software we use now is too integrated into our message taking
 system not to use. Also the operators just make one 8 hour phone call
 into our message taking system to get their remote audio so asterisk
 would just record that as one long call, which won't work either. Anyone
 have any other ideas.

Michael - the clues are in what you originally wrote:

 Right now we use these recording controls from radio shack that plug
 in between the wall jack and the phone and plug in via a 1/8 inch
 stereo connector to the mic input on the computer.

These phones have to be straight analog phones.

Just put in a channel bank/TDM24XX/Sangoma whatever for the call center.
Do not go IP phones there.  Just wire them up they way they are now.
If your investment in the call recording configuration is so great that
you can't/won't change it - there is no reason not to give the rest of
the company the benefits of VoIP.  You just have to kick in some more
money 8-)

Since you say 'not all' are going thru asterisk - just put in a TDM4XX
to test 4 agents or go with the ATA (re use them later for 'at home'
users)
once you have sold the idea to the 'powers that be'.

Brett
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[Asterisk-Users] PRI problem

[Joseph Rothstein]

We have an Asterisk server with a single Digium E1. Everzthign works as it
should except for one minor issue.

When we place a call to a number that is busy, Asterisk does not seem to
properly send the busy signal back to the SIP phones. There is no indication
on the phone of anything at all, just silence, like the call did not go
through. As you might imagine, this can be quite frustrating. The only
indication is that we see a 403 Forbidden SIP message on softphones.

I would appreciate any ideas of how to solve this issue. I have yet to do
extensive PRI debugging to see what the Telecom provider sends back, so I am
assuming that it correct signaling.

Regards,
Joe

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RE: [Asterisk-Users] Recording Calls at the phone

[Douglas Garstang]

On Demand-monitoring? If your referring to monitoring specific agents calls, 
I'm still trying to work out how to do that. You can either monitor all calls 
for a queue, or all calls for all agents, but not all calls for a specific 
agent. I tried to use the Monitor() command on it's own to start recording when 
an agent receives a call, but that does not appear to work.

-Original Message-
From: Francesco Peeters (Asterisk) [mailto:[EMAIL PROTECTED]
Sent: Friday, January 06, 2006 7:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Recording Calls at the phone


On Fri, January 6, 2006 15:37, Michael Sampson said:
 I work for a call center and we are looking at using asterisk to have
 our operators take calls. Our message taking software records all the
 calls on the operators computers. Right now we use these recording
 controls from radio shack that plug in between the wall jack and the
 phone and plug in via a 1/8 inch stereo connector to the mic input on
 the computer. If I buy an IP phone I can't do that. I could get an FXO
 adapter and regular phones, but I'm looking to get as little equipment
 as possible. Radio shack makes a recording control that plugs in to a
 2.5 mm headset jack, but it takes batteries so thats not going to work

 Does anyone else do something similar? Does anyone have any ideas about
 what producs/setup would work for this.


Asterisk has a built in monitoring system. You can chose to do Always,
Never or On Demand monitoring, depending on your setup and dialplan

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Monitor

Good luck!

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
  AMD Duron 1GHz - 1GB - * 1.2.1
  2 Sweex HFC-PCI cards
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[Asterisk-Users] pwlib compile error

[A_ Navone]

pwlib ver 1.5.2

/usr/bin/ld: 
./obj_linux_x86_d/asn_grammar.o(.gnu.linkonce.r._ZTV5PListI7PStringE[vtable 
for PListPString]+0x1c): unresolvable relocation against symbol 
`PAbstractList::Compare(PObject const) const'

/usr/bin/ld: final link failed: Nonrepresentable section on output
collect2: ld returned 1 exit status
make[3]: *** [obj_linux_x86_d/asnparser] Error 1
make[3]: Leaving directory `/usr/src/pwlib/tools/asnparser'
make[2]: *** [debug] Error 2
make[2]: Leaving directory `/usr/src/pwlib'
make[1]: *** [libs] Error 2
make[1]: Leaving directory `/usr/src/pwlib'
make: *** [debuglibs] Error 2

any ideas ?
thx in advance

_
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Security. http://clinic.mcafee.com/clinic/ibuy/campaign.asp?cid=3963


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Re: [Asterisk-Users] PRI problem

[C F]

Looks like you got a configuration issue, you should test for the
${DIALSTATUS} variable and set the signalling to the phones based on
that.

You can do:
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Goto,s-${DIALSTATUS},1)
exten = s-CANCEL,1,Playtones(congestion)
exten = s-CANCEL,2,Congestion
exten = s-NOANSWER,1,Goto(s-CANCEL,1)
exten = s-BUSY,1,Playtones(busy)
exten = s-BUSY,2,Busy
exten = s-CONGESTION,1,Goto(s-CANCEL,1)
exten = s-CHANUNAVAIL,1,Goto(s-CANCEL,1)

Check this:
http://www.voip-info.org/wiki-asterisk+cmd+dial
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+DIALSTATUS

On 1/6/06, Joseph Rothstein [EMAIL PROTECTED] wrote:
 We have an Asterisk server with a single Digium E1. Everzthign works as it
 should except for one minor issue.

 When we place a call to a number that is busy, Asterisk does not seem to
 properly send the busy signal back to the SIP phones. There is no indication
 on the phone of anything at all, just silence, like the call did not go
 through. As you might imagine, this can be quite frustrating. The only
 indication is that we see a 403 Forbidden SIP message on softphones.

 I would appreciate any ideas of how to solve this issue. I have yet to do
 extensive PRI debugging to see what the Telecom provider sends back, so I am
 assuming that it correct signaling.

 Regards,
 Joe

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[Asterisk-Users] controlling SIP subscriptions from SNOM phones

[Joseph Rothstein]

We recently deployed 10 SNOMs as part of a PBX hosted solution. We have one
phone setup as the receptionist phone, using hints to show busy office
lines. This all works as expected.

This is a new installation, and people are just starting to setup their
phones. For those of you not familiar with SNOM phones, there is a row of
keys on the right side of the phone which SNOM calls function keys. In order
to get the receptionist phone to see the hinted extensions, you have to
set DESTINATION and the extension in the SNOM setup web page. This causes
the receptionist phone to issue SIP SUBSCRIBE messages for the state of
these extensions. 

Now, one user, not the receptionist, has gone in and set his personal
numbers to these function keys thinking that DESTINATION meant setting a
number to dial out. So now I have a ton of SIP SUBSCRIBE messages for his
numbers.

I thought that in order for SIP SUBSCRIBE to work, I had to use 's
subscribecontext=x in the user's config in sip.conf so that SIP peers
could only subscribe to extensions in that particular context. See the
output below:

asterisk_test*CLI
 -- Registered SIP '959' at 82.135.81.211 port 5062 expires 3600
 -- Saved useragent snom320/4.4 for peer 959 Jan  6 20:11:29
ERROR[8677]: chan_sip.c:10790
handle_request_subscribe: Got SUBSCRIBE for extensions without hint.  
Please add hint to 008971940494 in context sanset Jan  6 20:11:29
ERROR[8677]: chan_sip.c:10790
handle_request_subscribe: Got SUBSCRIBE for extensions without hint.  
Please add hint to 0089956010 in context sanset Jan  6 20:11:29 ERROR[8677]:
chan_sip.c:10790
handle_request_subscribe: Got SUBSCRIBE for extensions without hint.  
Please add hint to 904 in context sanset Jan  6 20:11:29 ERROR[8677]:
chan_sip.c:10790
handle_request_subscribe: Got SUBSCRIBE for extensions without hint.  
Please add hint to 001708130105 in context sanset Jan  6 20:11:29
ERROR[8677]: chan_sip.c:10790
handle_request_subscribe: Got SUBSCRIBE for extensions without hint.  
Please add hint to 001709114321 in context sanset asterisk_test*CLI sip
show peer 959 asterisk_test*CLI

   * Name   : 959
   Secret   : Not set
   MD5Secret: Not set
   Context  : sanset
   Subscr.Cont. : Not set
   Language : de
   Accountcode  : sanset
   AMA flags: Unknown
   CallingPres  : Presentation Allowed, Not Screened
   Callgroup: 
   Pickupgroup  : 
   Mailbox  : [EMAIL PROTECTED]
   ...
   SIP Options  : (none)
   Codecs   : 0x100 (g729)
   Codec Order  : (g729)
   Status   : Unmonitored
   Useragent: snom320/4.4
   Reg. Contact : sip:[EMAIL PROTECTED]:5062;line=uq15mxj7

I would think that if there is no subscribecontext set, that Asterisk should
ignore these subscribe messages instead of creating an error. I know that
subscribe is not complete in Asterisk, so I am thinking that this is a bug.

Has anyone else had this same problem? Or know of a way around this? Would
setting the LINE option on the web page work for callout? 

Regards to all,
Joe

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[Asterisk-Users] SJPhone with external ringer

[Chuck Bunn]

Hi,

Does anyone know if it is possible to setup an SJPhone with an external 
ringer of some sort. One of the operators may not always be at her desk 
and when she is not wearing a headset she cannot hear the phone ring - 
is there some way to fix this?


Thanks
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Re: [Asterisk-Users] Not Able to Connect Two Asterisk Servers Using IAX2

[Erik Anderson]

On 1/6/06, Chandan Mishra [EMAIL PROTECTED] wrote:
 Hi
 I have two asterisk servers. I just want to connect two asterisk server
 using IAX2.
 But the Asterisk  Servers are not able to register each other. If some body
 have done this
 then Please send me the configuration they have done in iax.conf and
 extensions.conf.
 I simply want to connect and call from one sever to another.

If you want a helpful response, you're going to need to offer more
information.  Error messages, appropriate snippets from your iax.conf
files, etc.

http://www.catb.org/~esr/faqs/smart-questions.html
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RE: [Asterisk-Users] controlling SIP subscriptions from SNOM phones

[Alexander Lopez]

It may be better to change it to NOTICE instead of ERROR. You would want
to know it you are trying to subscribe to an unknown 'hint', misconfigs
or lack of should cause NOTICE instead of ERROR.
 

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Joseph Rothstein
 Sent: Friday, January 06, 2006 2:57 PM
 To: asterisk-users@lists.digium.com
 Subject: [Asterisk-Users] controlling SIP subscriptions from 
 SNOM phones
 
 We recently deployed 10 SNOMs as part of a PBX hosted 
 solution. We have one phone setup as the receptionist phone, 
 using hints to show busy office lines. This all works as expected.
 
 This is a new installation, and people are just starting to 
 setup their phones. For those of you not familiar with SNOM 
 phones, there is a row of keys on the right side of the phone 
 which SNOM calls function keys. In order to get the 
 receptionist phone to see the hinted extensions, you have 
 to set DESTINATION and the extension in the SNOM setup web 
 page. This causes the receptionist phone to issue SIP 
 SUBSCRIBE messages for the state of these extensions. 
 
 Now, one user, not the receptionist, has gone in and set his 
 personal numbers to these function keys thinking that 
 DESTINATION meant setting a number to dial out. So now I have 
 a ton of SIP SUBSCRIBE messages for his numbers.
 
 I thought that in order for SIP SUBSCRIBE to work, I had to 
 use 's subscribecontext=x in the user's config in 
 sip.conf so that SIP peers could only subscribe to extensions 
 in that particular context. See the output below:
 
 asterisk_test*CLI
  -- Registered SIP '959' at 82.135.81.211 port 5062 expires 3600
  -- Saved useragent snom320/4.4 for peer 959 Jan  6 20:11:29
 ERROR[8677]: chan_sip.c:10790
 handle_request_subscribe: Got SUBSCRIBE for extensions without hint.  
 Please add hint to 008971940494 in context sanset Jan  6 20:11:29
 ERROR[8677]: chan_sip.c:10790
 handle_request_subscribe: Got SUBSCRIBE for extensions without hint.  
 Please add hint to 0089956010 in context sanset Jan  6 
 20:11:29 ERROR[8677]:
 chan_sip.c:10790
 handle_request_subscribe: Got SUBSCRIBE for extensions without hint.  
 Please add hint to 904 in context sanset Jan  6 20:11:29 ERROR[8677]:
 chan_sip.c:10790
 handle_request_subscribe: Got SUBSCRIBE for extensions without hint.  
 Please add hint to 001708130105 in context sanset Jan  6 20:11:29
 ERROR[8677]: chan_sip.c:10790
 handle_request_subscribe: Got SUBSCRIBE for extensions without hint.  
 Please add hint to 001709114321 in context sanset 
 asterisk_test*CLI sip show peer 959 asterisk_test*CLI
 
* Name   : 959
Secret   : Not set
MD5Secret: Not set
Context  : sanset
Subscr.Cont. : Not set
Language : de
Accountcode  : sanset
AMA flags: Unknown
CallingPres  : Presentation Allowed, Not Screened
Callgroup: 
Pickupgroup  : 
Mailbox  : [EMAIL PROTECTED]
...
SIP Options  : (none)
Codecs   : 0x100 (g729)
Codec Order  : (g729)
Status   : Unmonitored
Useragent: snom320/4.4
Reg. Contact : sip:[EMAIL PROTECTED]:5062;line=uq15mxj7
 
 I would think that if there is no subscribecontext set, that 
 Asterisk should ignore these subscribe messages instead of 
 creating an error. I know that subscribe is not complete in 
 Asterisk, so I am thinking that this is a bug.
 
 Has anyone else had this same problem? Or know of a way 
 around this? Would setting the LINE option on the web page 
 work for callout? 
 
 Regards to all,
 Joe
 
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RE: [Asterisk-Users] Sharing SIP Info with Realtime

[Doug G]

We have done some work on this since my last post.We added some code
to update new fields in the realtime SIP database.  Status, Qualify, and
Host Server. We then place the call directly to the phone the SIP
full contact (i.e. dial(sip/[EMAIL PROTECTED]:5060) Via a AGI
script. Our AGI looks at the new fields to determine the status of the
UA.  We are still testing, but so far so good.   

Doug





-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Friday, January 06, 2006 2:02 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Sharing SIP Info with Realtime

I've asked this question several times before and was always told it
wasn't possible. However, after reading a thread posted to the list
today, I'm not so sure my question was understood.
 
So, here I go again...
 
Is it possible to have multiple Asterisk systems share a common realtime
database for the purposes of accepting registrations from phones, and
maintaining SIP location (ie ip address, port, expiry) information? The
phones are type=friend as they both make and receive calls.
 
I've tried to implement this on three ocassions and on each occasion it
has failed. Phones registered to the database from one asterisk system
can't find the location of other phones registered to the database from
a different asterisk system.
 
Kevin Fleming said it wasn't possible. I even called Digium Sales and
they also seemed aware of the issue and said it wasn't possible. Have I
misunderstood something? I'd really like to know if it's supposed to
work or not before I try for a 4th time.
 
Doug.
 
 
 
 
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Re: [Asterisk-Users] Not Able to Connect Two Asterisk Servers UsingIAX2

[Ioan Indreias]

 On 1/6/06, Chandan Mishra [EMAIL PROTECTED] wrote:
 Hi
 I have two asterisk servers. I just want to connect two asterisk server
 using IAX2.
 But the Asterisk  Servers are not able to register each other. If some
 body
 have done this
 then Please send me the configuration they have done in iax.conf and
 extensions.conf.
 I simply want to connect and call from one sever to another.

 If you want a helpful response, you're going to need to offer more
 information.  Error messages, appropriate snippets from your iax.conf
 files, etc.

 http://www.catb.org/~esr/faqs/smart-questions.html
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You can start connecting Asterisk Server using information provided here:
http://sourceforge.net/docman/display_doc.php?docid=26418group_id=121515
Even if you not use AMP, it give you some guides on how to configure
iax.conf and extensions.conf files

--//--
Ioan Indreias
IT Consultant
Modulo Consulting - http://www.modulo.ro

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Re: [Asterisk-Users] Not Able to Connect Two Asterisk Servers Using IAX2

[Francesco Peeters (Asterisk)]

On Fri, January 6, 2006 20:20, Chandan Mishra said:
 Hi
 I have two asterisk servers. I just want to connect two asterisk server
 using IAX2.
 But the Asterisk  Servers are not able to register each other. If some
 body
 have done this
 then Please send me the configuration they have done in iax.conf and
 extensions.conf.
 I simply want to connect and call from one sever to another.

 Thanks

 Chandan Kumar Mishra
 Software Engg.
 

As always, the Wiki is your friend...

http://www.voip-info.org/wiki-Asterisk+-+dual+servers

I am using a modified version of method 3...

You have to make sure that you have a user entry in IAX.conf for the other
server as mentioned above...

So if your serverA logs in using passwd SECRET, make sure that you have an
entry
[serverA]
secret=SECRET
type=user
context=IncomingContext
auth=md5(this one is optional of course...)

Good luck!

-- 
F Peeters
  PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch
  2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0
Cologne HFC-S pins #52, #54, #55 connected in parallel for synching.
  AMD Duron 1GHz - 1GB - * 1.2.1
  2 Sweex HFC-PCI cards
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[Asterisk-Users] Using local\number

[Matt]

Hi,
What do I have to do to get local\number to work in a context?

It works from my [from-internal]... however from subcontexts it does not work:

Jan  6 15:55:32 VERBOSE[20237] logger.c: -- AGI Script Executing
Application: (Dial) Options: (Local/570323)
Jan  6 15:55:32 NOTICE[20237] chan_local.c: No such extension/context
[EMAIL PROTECTED] creating local channel
Jan  6 15:55:32 NOTICE[20237] app_dial.c: Unable to create channel of
type 'Local' (cause 0 - Unknown)
Jan  6 15:55:32 VERBOSE[20237] logger.c:   == Everyone is
busy/congested at this time (1:0/0/1)
Jan  6 15:55:32 DEBUG[20237] app_dial.c: Exiting with DIALSTATUS=CHANUNAVAIL.

I'm dialing this as:  Local/570323
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RE: [Asterisk-Users] Not Able to Connect Two Asterisk Servers Usi ng IAX2

[Mark Welch]

Here is what I used to do it:

http://www.voip-info.org/wiki/index.php?page=Asterisk+-+dual+servers



Worked for me J











From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 
Sent: Friday, January 06, 2006
2:20 PM
To:
asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Not Able
to Connect Two Asterisk Servers Using IAX2





Hi 
I have two asterisk servers. I just want to connect two asterisk server using
IAX2.
But the Asterisk Servers are not able to register each other. If some
body have done this
then Please send me the configuration they have done in iax.conf and
extensions.conf.
I simply want to connect and call from one sever to another.

Thanks

Chandan Kumar Mishra
Software Engg.






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Re: [Asterisk-Users] SPA-3000 is translating vocal sounds into DTMF

[Matt]

Is it a female talking on the other end from which hears the DTMF?  If
so, that's life.   It happens on the PSTN network too from time to
time.

On 1/6/06, Brian Capouch [EMAIL PROTECTED] wrote:
 I'm sure there must be a setting I'm missing somewhere, so I thought I
 might was well ask here.

 Conversations are punctuated by sudden replacement of a given syllable
 or so of conversation with a DTMF tone.

 I would hope perhaps there's some kind of setting that has to do with
 the way it detects inband DTMF?  I'm pretty sure it's an artifact of
 this particular ATA; my other SIP devices are just fine.

 Thx.

 B.
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Re: [Asterisk-Users] Macro DialPlan

[trixter aka Bret McDanel]

On Fri, 2006-01-06 at 10:27 +, scott wrote:
 [macro-71macro]
 
 exten = s,1,Dial(SIP/${ARG1},30,tr)
 exten = s,2,VoiceMail(${ARG1})
 exten = s,3,PlayBack(vm-goodbye)
You appear to be missing some potential error codes.  Here is what I use
which is just an extended version of what you have.


[macro-dialvmb]
exten = s,1,Dial(${ARG1},16,t)
exten = s,2,Voicemail(u${ARG2})
exten = s,3,Hangup
exten = s,102,Voicemail(b${ARG2})
exten = s,103,Hangup 

I have set this up so I pass 2 args, one is the extension to dial the
other is the voicemail box to goto if dial fails.  I added the hangups
cause I am paranoid about that :)


-- 
Trixter http://www.0xdecafbad.com Bret McDanel
UK +44 870 340 4605   Germany +49 801 777 555 3402
US +1 360 207 0479 or +1 516 687 5200
FreeWorldDialup: 635378
http://www.sacaug.org/ Sacramento Asterisk Users Group


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[Asterisk-Users] How to properly use GROUP

[Brent Torrenga]

Can someone explain how to use groups? I can't seem to wrap myself around
this, though I know it is something simple.

I have 3 zap lines, and when placing an outgoing call, would like to 1) use
a zap line if and only if 1 or fewer zap lines are being used at the time,
and 2) if more than 1 zap lines are in use then to go ahead and use VOIP to
place the call.

Also, is there a difference between how this is implemented in 1.0.9 and
1.2.1?


Sincerely,

Brent A. Torrenga
[EMAIL PROTECTED]

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

219.836.8918x325 Voice
219.836.1138 Facsimile
www.torrenga.com

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