Re: [asterisk-users] Caller ID in Sweden not working and looking for and voices

[Mattias Andersson]

Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias 
On 15/11/06, Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias Andersson: Hi! I am getting inbound caller ID fine bout not out. I am in Sweden and suing Rixtelcom /POrt80 as provider. anyone knowing what is wrong?
Assuming that is a SIP provider, it is not your job to set the calleridbut the provider's - their interfacing to the regular landline networkis responsible. There are providers that never send callerid, some send
always (united, gmx in Germany) and some allow the user to set in hispreferences wether he wants to send callerid (sipgate.de).BRAnselm___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Mattias AnderssonStorskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41
h. +46-8-641 38 97 Email: [EMAIL PROTECTED]
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Re: [asterisk-users] Caller ID in Sweden not working and looking for and voices

[Mattias Andersson]

Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias
On 15/11/06, Mattias Andersson [EMAIL PROTECTED] wrote:
Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias 

On 15/11/06, Anselm Martin Hoffmeister [EMAIL PROTECTED]
 wrote:
Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias Andersson: Hi! I am getting inbound caller ID fine bout not out. I am in Sweden and suing Rixtelcom /POrt80 as provider. anyone knowing what is wrong?
Assuming that is a SIP provider, it is not your job to set the calleridbut the provider's - their interfacing to the regular landline networkis responsible. There are providers that never send callerid, some send
always (united, gmx in Germany) and some allow the user to set in hispreferences wether he wants to send callerid (
sipgate.de).BRAnselm___
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Mattias Andersson
Storskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41
h. +46-8-641 38 97 Email: [EMAIL PROTECTED]

-- Mattias AnderssonStorskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41h. +46-8-641 38 97 Email: 
[EMAIL PROTECTED]
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[asterisk-users] Condensing queue CDRs into single entry

[Rajkumar S]

Hi,

When a call is made to a queue and picked up by agents at least 2 CDR
entries are made, one from local to the agent's (sip) phone, and from
incoming line to Agent. There are other entries generated when other
conditions happen, like agent do not pickup phones and so on.

Going through the cdr entries, there seems to be no common filelds by
which all entries belonging to a single call can be picked up, so how
can I extract all entries belonging to a single call, say if I have
the CDR in a database.

raj
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[asterisk-users] Re: Moh stops immediately

[Martin Joseph]

[EMAIL PROTECTED] said:


I'm trying to set up the Music on Hold feature.
However, when I place a call the moh starts and stops
immediately and as a result I dont hear the audio.




On 2006-11-13 00:14:40 -0800, zen Perry [EMAIL PROTECTED] said:



Mac OS X, Asterisk 1.4 beta


Yeah,  I am also an OSX user and it does this under 1.2.13 and 1.4branch...

Don't have the answer,  but if you figure it out, please do share!

Marty



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RE: Problem found Re: [asterisk-users] Headaches with Video over SIP

[Steve Langstaff]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Peter Howard
 Sent: 14 November 2006 20:51
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: Problem found Re: [asterisk-users] Headaches 
 with Video over SIP

  Codec identifiers = 96 refer to dynamic payload types.
  
 
 Thanks, that's worth knowing.
 
  They have to be negotiated on each SDP offer/answer exchange.
  
  So for the Polycom-Asterisk traffic, Asterisk should parse the SDP 
  and say to itself Hey, the caller wants me to send it H264 marked 
  with payload type 109, and/or H263-1998 marked with payload 
 type 96. 
  and adapt it's outgoing payload type marking accordingly.
  
 
 should parse the SDP.  It's not at 1.4.0-beta3 (or, 
 seemingly, earlier versions).  Should I submit a bug report for this?

*If* Asterisk is claiming compliance with RFC 2327, *and* if you read
the RFC the same way that I do, *and* you are actually seeing what you
have reported then I guess you *could* submit a bug report, but I'm not
going to say that you *should* submit a report (is that disclaimered
enough?). As an aside, it appears that this issue might already be the
subject of bugs 6568 and 7461.

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RE: [asterisk-users] In the beginning-The first question.

[Steve Langstaff]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 James R. Stevens
 Sent: 14 November 2006 20:36
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] In the beginning-The first question.
 
 List,
 Im a Cisco certified Network guy with little telecom 
 experience (BRI/PRI at the time) so please forgive my 
 terminology. I am showing interest after the Network World 
 SHSU October 4 article. We have 3 offices (Hub-Spoke T1 Frame 
 relay to the remote offices(Data  voice on separate T)). 
 Each office currently does their own thing for telecom. Our
 Main(HUB) office currently has 14 channels of T1 into an ADIT 
 600 punched down to the DEMARC. Our Panasonic (72 port) 
 VB-43050 DBS picks up from the DEMARC and spits out 4 lines 
 for our VM server. My goal is described below, the question 
 is how to make Asterisk do it.
 
 Consolidate telecom services of the other two offices into 
 our HUB office. Try (Hard) to keep some of the current phones 
 (Panasonic-Digital_ Not a high priority). 

You could use a Citel SIP Handset Gateway (http://www.citel.com) to keep
the Panasonic DBS phones. This unit converts their proprietary
signalling to SIP.

Disclaimer: I work for Citel.


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[asterisk-users] How to use Voipjet or any Voip provider Trunk from my mobile through fxo and fxs ports?

[Crazy Boy]

Hi Friends,I have installed Asterisk and configured successfully. Now, I got a doubt. Here I am giving my configuration.1) 1 PSTN line connected to FXO port and created inbound route. (Ph. No: 233534)2) 1 Analog phone connected to FXS port and created ZAP extension with No. 1033) Configured "Voipjet" trunk to make international calls.All the above are working fine.Now, My problem is: I have to make international calls from my mobile through Voipjet trunk using my Asterisk server.When I make a call to 233534 from my mobile, call will automatically goes to 103. Its working fine. Now, I have to dial a international number (For eg: 1 718 777 3456) and call should be go through Voipjet trunk. How can I do this? Please tell me or suggest me a good link to do this.Looking forward to your response. Thank you.Regards,Chandra. 

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[asterisk-users] How to do the Call Snooping

[raviprakash sunkara]

Hello Users,I googled Call Snooping, its shows only the it means, But i didn't find How to dialplan the Call Snooping,I seen that  What is Trixbox  in Asterisk I Use only some Feature in Asterisk (20), 
Is it need Asterisk to install the TrixBox in that same System where i installed the Asterisk ServerHelp me please :P-- Thanks and RegardsRavi Prakash Sunkara		
[EMAIL PROTECTED] 	M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 		[EMAIL PROTECTED]
www.hyperion-tech.com
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Re: [asterisk-users] Add Apps to Asterisk?

[Andrea Spadaccini]

Ciao Matthew,

 What do I have to do, exactly, to install Meetme?

You have to build Zaptel before building Asterisk, because MeetMe uses
Zaptel modules for timing. Then, when you build Asterisk the MeetMe app
will automatically be built.

See http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe

HTH,

-- 
Andrea Spadaccini
Multimedia Technologies Institute s.r.l.
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Re: [asterisk-users] Add Apps to Asterisk?

[Mattias Andersson]

Hi!The problem is that I have commercial Asterisk baste switch that it works wit.My trixbox do not. I guess it has to do with the setting of system for Caller ID.//Mattias
Sorry if I have made double post! (Difficult to verify if mail was sent).On 15/11/06, Darryl Dunkin [EMAIL PROTECTED]
 wrote:First, check for app_meetme.so in /usr/lib/asterisk/modules (wherever
your modules path is). Next, in the CLI, do a 'show modules' to see ifit is there. If not, check your modules.conf and add in 'load =app_meetme.so' assuming autoload is not enabled.-Original Message-
From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED]
] On Behalf Of MatthewRubensteinSent: Tuesday, November 14, 2006 21:09To: Asterisk-UsersSubject: [asterisk-users] Add Apps to Asterisk?I've got an Asterisk (v1.2.11) installation running, but it
doesn'tseem to have the Meetme() app. At the CLI, I type Meetme , and itresponds No such command 'Meetme'; meetme doesn't show up in CLI showmodules . I'm running a SIP-only server at a datacenter where I can't
add Digium (or any other) HW, and am running under CentOS. There isan /etc/asterisk/meetme.conf file, but I don't see anything to use it.What do I have to do, exactly, to install Meetme? What about the
Conference command, or others not installed? I'd prefer to use theCentOS package system as much as possible, but I can compile source ifnecessary. Is there a HowTo on the Web somewhere that details thisprocess?
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http://lists.digium.com/mailman/listinfo/asterisk-users-- Mattias AnderssonStorskiftesvägen 6145 60 Norsborg m. +46-70-799 44 41
h. +46-8-641 38 97 Email: [EMAIL PROTECTED]
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[asterisk-users] A question on ISDN cards... (in the UK)

[Gordon Henderson]

(I'm in the UK if that makes a difference)

There seems to be a plethora of different ISDN cards available in both the
BRI and PRI range - all with varying prices too - from £25 to nearly £1000
from some popular reseller sites...

Does anyone have (or know of) a good comparison site, or have views on one
card type over another?

I'm assuming that the more expensive cards have additional features like
better echo cancellation and audio processing abilities (or less CPU
overhead?)

Would anyone like to recommend a good and reasonable quality ISDN card for
use in the UK, as after a lot of good results with TDM400P cards with
several systems installed now, I need to look at a few ISDN BRI (old
business highway about to move to ISDN2) and possibly a single-line PRI
(ISDN-30) system.

Many thanks,

Gordon
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Re: [asterisk-users] How to do the Call Snooping

[Vij]

chanspy

see: http://www.voip-info.org/wiki-Asterisk+cmd+ChanSpy

-VijOn 11/15/06, raviprakash sunkara [EMAIL PROTECTED] wrote:
Hello Users,I googled Call Snooping, its shows only the it means, But i didn't find How to dialplan the Call Snooping,
I seen that  What is Trixbox  in Asterisk I Use only some Feature in Asterisk (20), 
Is it need Asterisk to install the TrixBox in that same System where i installed the Asterisk ServerHelp me please :P-- Thanks and RegardsRavi Prakash Sunkara		

[EMAIL PROTECTED] 	M:+91 9985077535O:+91 40 23114549F:+91 40 40208727 		
[EMAIL PROTECTED]
www.hyperion-tech.com

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RE: [asterisk-users] A question on ISDN cards... (in the UK)

[Senad Jordanovic]

 Would anyone like to recommend a good and reasonable quality ISDN
 card for use in the UK, as after a lot of good results with TDM400P
 cards with several systems installed now, I need to look at a few
 ISDN BRI (old business highway about to move to ISDN2) and possibly a
 single-line PRI (ISDN-30) system.   
 
 Many thanks,
 

For low cost 1 port:
http://www.bicomsystems.com/products/C/P/319/286_2875/

For 4 ports, try:
http://www.bicomsystems.com/products/C/P/319/282/

We also have just done some tests with new Digium BRI card which is
promising since it has echo cancellation on board.

Some other people use CAPI cards so you may want to look at that too...



Senad





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[asterisk-users] Time Based Voicemail Messages

[Wildheart]

Hi,

   I want to change my voicemail message based on the time of day. I would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.

   I have come up with two ways of doing it:

 1. A cron job to replace the files (messy)

 2. Using different mailboxes at the different times (this means I have 2
mailboxes to check).

   Is there a way that the voicemail could be enhanced by adding a feature
like this?

With thanks,

  Tim

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[asterisk-users] Time Based Voicemail Messages

[Wildheart]

Hi,

   I want to change my voicemail message based on the time of day. I would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.

   I have come up with two ways of doing it:

 1. A cron job to replace the files (messy)

 2. Using different mailboxes at the different times (this means I have 2
mailboxes to check).

   Is there a way that the voicemail could be enhanced by adding a feature
like this?

With thanks,

  Tim

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Re: [asterisk-users] Time Based Voicemail Messages

[Marnus van Niekerk]

Just use two different contexts for the two times of day (open/closed)
and use Playback to play the correct message before going direct into 
voicemail without any prompt.


M

Wildheart wrote:

Hi,

   I want to change my voicemail message based on the time of day. I would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.

   I have come up with two ways of doing it:

 1. A cron job to replace the files (messy)

 2. Using different mailboxes at the different times (this means I have 2
mailboxes to check).

   Is there a way that the voicemail could be enhanced by adding a feature
like this?
  


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Re: [asterisk-users] Time Based Voicemail Messages

[Benjamin Jacob]

a lil bit of googling wud have answered you Tim.
Put in some effort next time   anyway, for now :

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

Wildheart wrote:


Hi,

  I want to change my voicemail message based on the time of day. I would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.

  I have come up with two ways of doing it:

1. A cron job to replace the files (messy)

2. Using different mailboxes at the different times (this means I have 2
mailboxes to check).

  Is there a way that the voicemail could be enhanced by adding a feature
like this?

   With thanks,

 Tim

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[asterisk-users] Asterisk as a SIP client, Need to auto-answer

[Ehsan Khosrowshahi]

Hi all,I want to initiate a call from the asterisk to an extension, where I will forward   the asterisk side to another extension later (to the conference extension). I can   initiate a call uning originate call from an extension to the desired extension,   but it would need someone from the originator extension to answer the phone. How   can i register an extension to asterisk where it automatically answers the phone   and creates a channel where I may be able to redirect that channel later to the   conference room.This is what I have done and didnt work:SIP.confregister = 7:[EMAIL PROTECTED]  [7]type=friendauth=md5username=7secret=7callerid=7host=191.21.21.21reinvite=nocanreinvite=noqualify=1500nat=yes   
 and in Extension.conf I got:  exten = 7,1,Answer  and when I originate a call using Manager API with these parameters:  Channel: SIP/[EMAIL PROTECTED]CallerID: 7Exten: Any number  I got the following error in asterisk CLI:   == Manager 'manager' logged on from 191.21.21.21 -- Got SIP response 482 "Loop Detected" back from 191.21.21.21  Channel SIP/0041435215309-3c5a was never answered. == Manager 'manager' logged off from 191.21.21.21  I want to create a dump connection between a dump extension to any extension then   redirect the channel from the dump extension side to the conferece. but How can i make the dump extension to auto-answer and create a channel when I Originate Call using manager API?   
 BestEhsan 

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RE: [asterisk-users] A question on ISDN cards... (in the UK)

[Conrad Wood]

On Wed, 2006-11-15 at 11:23 +, Senad Jordanovic wrote:
  Would anyone like to recommend a good and reasonable quality ISDN
  card for use in the UK, as after a lot of good results with TDM400P
  cards with several systems installed now, I need to look at a few
  ISDN BRI (old business highway about to move to ISDN2) and possibly a
  single-line PRI (ISDN-30) system.   
  
I'd say the most important distinction is the choice between HFC based
ISDN cards (starting around £9) and active cards, like Diva, Digium etc
(~£300).

Whilst the HFC cards work (with bristuff) you need to be prepared to
reload modules regularly and go through other hoops.
I used to work in a IT company, and there it's perfectly allright to use
cheap cards, because the skills to reset modules etc are available at
all times.
For a clients' system I wouldn't go down that route and spend the money.
I have no complains on call quality or dropped calls on cheap cards nor
on expensive cards, it's the administration and 'shinyness of the
product'.

Conrad


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Re: [asterisk-users] Time Based Voicemail Messages

[Wildheart]

Isn't that covered in point 2? Admittedly, I did not consider using
Playback rather than voicemail to play the message. But you didn't point
that out anyway.



 a lil bit of googling wud have answered you Tim.
 Put in some effort next time   anyway, for now :

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

 Wildheart wrote:

Hi,

   I want to change my voicemail message based on the time of day. I
 would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.

   I have come up with two ways of doing it:

 1. A cron job to replace the files (messy)

 2. Using different mailboxes at the different times (this means I have 2
mailboxes to check).

   Is there a way that the voicemail could be enhanced by adding a
 feature
like this?

With thanks,

  Tim

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[asterisk-users] Re: Is asterisk able to integrate with MS SQL

[Tony Mountifield]

In article [EMAIL PROTECTED],
Sharon Lim [EMAIL PROTECTED] wrote:
 -=-=-=-=-=-
 -=-=-=-=-=-
 
 Thanks, will do more research on that part. By the way, Im trying to do IVR
 where caller enter the pin the retrieve some information out of the MS SQL.
 I am wondering, what is the constraints or how to go about it. As per said
 MS SQL is about CDR. Now like i want to match and retrieve data out of the
 DB through IVR. Any guidance?

I don't think there is any direct access to MS SQL via FreeTDS from the
dialplan, but there are ODBC functions you could use. See this page:

http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc

Alternatively, implement your IVR using AGI or the ExternalIVR application
and then you can do what you like with the database.

See http://www.voip-info.org/wiki-Asterisk+AGI
and http://www.voip-info.org/wiki-Asterisk+cmd+ExternalIVR

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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Re: [asterisk-users] A question on ISDN cards... (in the UK)

[Marco Mouta]

Beronet cards have 2 or 4 ports are very good.Those guys produced the misdn driver, that is now Digium uses for their new BRI card.www.beronet.comtheir tech support has been very very good.
On 11/15/06, Conrad Wood [EMAIL PROTECTED] wrote:
On Wed, 2006-11-15 at 11:23 +, Senad Jordanovic wrote:  Would anyone like to recommend a good and reasonable quality ISDN  card for use in the UK, as after a lot of good results with TDM400P
  cards with several systems installed now, I need to look at a few  ISDN BRI (old business highway about to move to ISDN2) and possibly a  single-line PRI (ISDN-30) system. 
I'd say the most important distinction is the choice between HFC basedISDN cards (starting around £9) and active cards, like Diva, Digium etc(~£300).Whilst the HFC cards work (with bristuff) you need to be prepared to
reload modules regularly and go through other hoops.I used to work in a IT company, and there it's perfectly allright to usecheap cards, because the skills to reset modules etc are available atall times.
For a clients' system I wouldn't go down that route and spend the money.I have no complains on call quality or dropped calls on cheap cards noron expensive cards, it's the administration and 'shinyness of theproduct'.
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 http://lists.digium.com/mailman/listinfo/asterisk-users-- Com os melhores cumprimentos,
Marco Mouta
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Re: [asterisk-users] A question on ISDN cards... (in the UK)

[Steve Kennedy]

On Wed, Nov 15, 2006 at 11:15:16AM +, Gordon Henderson wrote:

 (I'm in the UK if that makes a difference)
 There seems to be a plethora of different ISDN cards available in both the
 BRI and PRI range - all with varying prices too - from ?25 to nearly ?1000
 from some popular reseller sites...
 Does anyone have (or know of) a good comparison site, or have views on one
 card type over another?
 I'm assuming that the more expensive cards have additional features like
 better echo cancellation and audio processing abilities (or less CPU
 overhead?)
 Would anyone like to recommend a good and reasonable quality ISDN card for
 use in the UK, as after a lot of good results with TDM400P cards with
 several systems installed now, I need to look at a few ISDN BRI (old
 business highway about to move to ISDN2) and possibly a single-line PRI
 (ISDN-30) system.

Remember that ISDN may not be ISDN (well it is), but you specifically
need ISDN2e for BRI and make sure a PRI is configured as EuroISDN (ISDN
v110, the UK default is v85).

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/Gizmo/Mac stevekennedyuk / MSN [EMAIL PROTECTED]
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[asterisk-users] Zaptel configurations in India

[K Y Iyer]

Hi

I am testing Asterisk connecting to our Alcatel 4400 PBX.  I have a
wcte11xp card - all is well - but we cannot communicate with the Alcatel
- when we try to call an Alcatel extension, we see Error 34 no channels
available on the CLI.

I suspect that this is because of invalid span and signalling parameters
in my config files.  Have seen a lot of information on this - but none
of seems to be helping.

We get no errors when we start Asterisk 

What does the YELLOW indicator mean when I do cat /proc/zaptel/1?

What does Status: Provisioned, In Alarm, Down, Active when I see pri
show span 1

Thanks very much for any assistance

Best wishes

Iyer
 



Zaptel.conf
---
span=1,1,0,cas,hdb3
bchan=1-15
dchan=16
bchan=17-31
loadzone=uk
defaultzone=uk



Zapata.conf

[trunkgroups]
[channels]
language=uk
context=default
switchtype=euroisdn
signalling=pri_cpe
rxwink=300
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=0.0
txgain=0.0
group=1
immediate=no
channel = 1-15,17-31



*CLI zap show channels
   Chan Extension  Context Language   MusicOnHold
 pseudodefault uk
  1default uk
  2default uk
  3default uk
  4default uk
  5default uk
  6default uk
  7default uk
  8default uk
  9default uk
 10default uk
 11default uk
 12default uk
 13default uk
 14default uk
 15default uk
 17default uk
 18default uk
 19default uk
 20default uk
 21default uk
 22default uk
 23default uk
 24default uk
 25default uk
 26default uk
 27default uk
 28default uk
 29default uk
 30default uk
 31default uk



*CLI pri show span 1
Primary D-channel: 16
Status: Provisioned, In Alarm, Down, Active
Switchtype: EuroISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
T200 Timer: 1000
T203 Timer: 1
T305 Timer: 3
T308 Timer: 4000
T313 Timer: 4000
N200 Counter: 3



Cat /proc/interrupts
-
   CPU0   CPU1
  0:43005644296314IO-APIC-edge  timer
  1:363429IO-APIC-edge  i8042
  2:  0  0  XT-PIC  cascade
  8:  0  1IO-APIC-edge  rtc
 15:  38352  38153IO-APIC-edge  ide1
137:  14622  14304   IO-APIC-level  aic7xxx
145: 113222 59   IO-APIC-level  eth0
161:   5192   5981   IO-APIC-level  uhci_hcd
169:42215654213646   IO-APIC-level  wcte11xp
NMI:  0  0
LOC:85980068597974
ERR:  0
MIS:  0



Ztcvg -v

[EMAIL PROTECTED] ~]# ztcfg -

Zaptel Configuration
==

SPAN 1: CAS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

Channel map:

Channel 01: Clear channel (Default) (Slaves: 01)
Channel 02: Clear channel (Default) (Slaves: 02)
Channel 03: Clear channel (Default) (Slaves: 03)
Channel 04: Clear channel (Default) (Slaves: 04)
Channel 05: Clear channel (Default) (Slaves: 05)
Channel 06: Clear channel (Default) (Slaves: 06)
Channel 07: Clear channel (Default) (Slaves: 07)
Channel 08: Clear channel (Default) (Slaves: 08)
Channel 09: Clear channel (Default) (Slaves: 09)
Channel 10: Clear channel (Default) (Slaves: 10)
Channel 11: Clear channel (Default) (Slaves: 11)
Channel 12: Clear channel (Default) (Slaves: 12)
Channel 13: Clear channel (Default) (Slaves: 13)
Channel 14: Clear channel (Default) 

[asterisk-users] T38 problem

[Tomislav Parčina]

I have problem with fax machine Panasonic DX600. It's connected to Grandstream 
Handy Tone 386 which is connected to Asterisk. Asterisk is connected to my SIP 
provider.

To some numbers I can't send FAX, and I get following error on CLI.
WARNING[2237] chan_sip.c: Unknown SDP media type in offer: image 31358 udptl t38

I believe that Panasonic DX600 machine supports T38. And when I have another 
T38 fax machine on other end they try to send FAX using T38 protocol. And than 
I believe I get above error and sending FAX fails.

Is there any way to solve this? I hear that there is T38 support in Asterisk 
1.4, but I can't wait for version 1.4. In manual for Panasonic DX600 I didn't 
find any instructions how to turn T38 off.

Please suggest something.



--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] Re: unable to get channel lock BAD BAD BAD

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 I am seeing the following in my log file (standard trixbox install).
 One seems to be complaining about an error in the dialplan but it
 won't tell me what file or what line. The other (maybe related) is
 complaining about a channel lock.
 
 How to do go about trying to figure out what the problem is and how to solve 
 it?
 
 Nov 14 07:20:44 ERROR[24091] chan_sip.c: BAD! BAD! BAD!

Yes, I get same error message in my log. Anybody has any info on this one?


--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[asterisk-users] some questions about atxfer usage

[Antonio Almodóvar]

Hi all.

I have enabled the attended transfer feature in features.conf. I'm
using it and I want to resolve some questions, I hope someone can help
me :)

When I transfer a call to an extension:
- The extension rings during 15 seconds and the call returns to the
transferer. Is there any possibility to recover the call before the
timeout of 15 seconds expires?

I mean, I would like to personalize the way of making transfers using
the feature of atxfer. How can I do that?


Thanks in advance.
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Re: [asterisk-users] Re: Is asterisk able to integrate with MS SQL

[Bruce Reeves]

I have an IVR for employees to enter certain information, like employee number and such and then I pass that to a simple agi/php script that build the query string and uses freetds. It took me a while to get it working and reproduce it on several systems, but I am rather new to Linux in general.
On 11/15/06, Tony Mountifield [EMAIL PROTECTED] wrote:
In article [EMAIL PROTECTED],Sharon Lim 
[EMAIL PROTECTED] wrote: -=-=-=-=-=- -=-=-=-=-=- Thanks, will do more research on that part. By the way, Im trying to do IVR where caller enter the pin the retrieve some information out of the MS SQL.
 I am wondering, what is the constraints or how to go about it. As per said MS SQL is about CDR. Now like i want to match and retrieve data out of the DB through IVR. Any guidance?I don't think there is any direct access to MS SQL via FreeTDS from the
dialplan, but there are ODBC functions you could use. See this page:http://www.voip-info.org/wiki/view/Asterisk+app_dbodbcAlternatively, implement your IVR using AGI or the ExternalIVR application
and then you can do what you like with the database.See http://www.voip-info.org/wiki-Asterisk+AGIand 
http://www.voip-info.org/wiki-Asterisk+cmd+ExternalIVRCheersTony--Tony MountifieldWork: [EMAIL PROTECTED] - http://www.softins.co.uk
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-- BruceNortex Networks
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[asterisk-users] Re: ATA with reliable FAX?

[Tomislav Parčina]

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
 (a) If you are not running a version of Asterisk that has working SIP 
 jitter buffering (is there such a thing?), then abandon all hope now.
 
 (b) We have no experience with the Cisco ATAs, but the Linksys (nee 
 Sipura) SPA-210x is markedly better than the SPA-100x and SPA-200x, 
 probably because they have better jitter buffering (it goes without 
 saying we do not pass our fax traffic through Asterisk).
 
 (c) T.38 is the way to go, G.711 a poor and distant second choice 
 (again, Asterisk's T.38 pass-through is far from ready for prime time).

Hi George!

You said that T.38 is the way to go. I have problems with T.38 and I don't know 
how to solve them. Maybe you can help me. I often get this message on CLI:

Nov 15 14:56:03 WARNING[2237]: chan_sip.c:3602 process_sdp: Unknown SDP media ty
pe in offer: image 31512 udptl t38

What could be the reason and how to solve it? I have

Fax machine - Grandstream Handy Tone 386 - Asterisk - my SIP provider

Thank you for your time!



--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)270248
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail: tparcina#lama.hr
http://www.lama.hr
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[Asterisk-Users] The best available CAPI BRI card for Asterisk ?

[Olivier]

Hi,It seems AVM C2 and C4 ISDN-BRI active boards are not distributed anymore (is true everywhere).Eicon-Dialogic boards seem to have good Asterisk support, thanks to chan-capi.What are the best other CAPI-compliant boards (with embeded fax DSP) one could use with Asterisk ?
Regards
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Re: [asterisk-users] Re: Re: Voicemail Press '0'

[Brian Roy]

On 10/10/06, LJ [EMAIL PROTECTED] wrote:
In my Asterisk 1.2.9.1 installation I use the following:in voicemail.conf
 include the following:exitcontext=vmloginoperator=yes

Sorry to revive a month old thread but here was the easy button solution for me. 

With debugging on I did a reload app_voicemail.so from the CLI I saw the following

Nov 15 06:28:16 DEBUG[24613]: app_voicemail.c:6012 load_config: VM Operator break disabled globally

That happened even with the operator=yes in my voicemail context. So I moved the operator=yes up to the general contextandthat message went away, and my 0 option worked fine.

Hope this helps,

-Brian
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[asterisk-users] Asterisk - big installation

[doki_cti]

Hello
I want build big asterisk server. Server will be work as gateway between PSTN 
and VoIP network. I think about 1000 SIP users accounts and 4 E1 ports. I know 
that preformance in this case depend on codeck which will be use. I want use 
card with CAPI interface. Can you describe me your experience with this?
If you have some big installaion, please wriete some info about server 
(procesor, ram etc), numbers of user and simultaneous calls beetwene VoIP and 
PSTN. How often server crash?
Regards
Doki
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[asterisk-users] How to disable the 482 Loop Detected messages sent by Asterisk

[Ricardo Carvalho]

Is there a way to make Asterisk don't send 482 Loop Detected error 
messages and continue with the transaction that is taking place?



Thanks,
Ricardo.
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Re: [asterisk-users] How to disable the 482 Loop Detected messages sent by Asterisk

[Eric \ManxPower\ Wieling]

Ricardo Carvalho wrote:
Is there a way to make Asterisk don't send 482 Loop Detected error 
messages and continue with the transaction that is taking place?


Not that I know of since a loop is an error.
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RE: [asterisk-users] Re: Is asterisk able to integrate with MS SQL

[Wes Baehr]

Func_odbc (which is new in 1.4) was
backported to 1.2. See http://www.asterisk.org/func_odbc



While it only will return one row (there
are patches to make it return multiple rows), its very useful for our
purposes. You set up the function in func_odbc.conf, call it with
${ODBC_FunctionName(arg1,arg2,)} and it executes and returns the
specified data.



--

Wes
 Baehr















From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bruce Reeves
Sent: Wednesday, November 15, 2006
7:56 AM
To: Asterisk
 Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Re:
Is asterisk able to integrate with MS SQL





I have an IVR for
employees to enter certain information, like employee number and such and then
I pass that to a simple agi/php script that build the query string and uses
freetds. It took me a while to get it working and reproduce it on several
systems, but I am rather new to Linux in general. 



On 11/15/06, Tony
Mountifield [EMAIL PROTECTED]
wrote:

In article [EMAIL PROTECTED],
Sharon Lim 
[EMAIL PROTECTED] wrote:
 -=-=-=-=-=-
 -=-=-=-=-=-

 Thanks, will do more research on that part. By the way, Im trying to do
IVR
 where caller enter the pin the retrieve some information out of the MS
SQL. 
 I am wondering, what is the constraints or how to go about it. As per said
 MS SQL is about CDR. Now like i want to match and retrieve data out of the
 DB through IVR. Any guidance?

I don't think there is any direct access to MS SQL via FreeTDS from the 
dialplan, but there are ODBC functions you could use. See this page:

http://www.voip-info.org/wiki/view/Asterisk+app_dbodbc

Alternatively, implement your IVR using AGI or the ExternalIVR application 
and then you can do what you like with the database.

See http://www.voip-info.org/wiki-Asterisk+AGI
and http://www.voip-info.org/wiki-Asterisk+cmd+ExternalIVR

Cheers
Tony
--
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk 
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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-- 
Bruce
Nortex Networks 








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Re: [asterisk-users] trixbox + agi

[blackwater dev]

For the Asterisk side of things, are you using Asterisk directly or Trixbox? I'm just trying to get a prototype working so don't want to spend a lot of time on the initial asterisk setup. If Trixbox will allow me to do the php+agi integration, I'll do that, if not, will try to just try to install Asterisk from source.
Thanks!On 11/15/06, Tim Uckun [EMAIL PROTECTED] wrote:
If I were you I would go the AGI way. Use ruby, python, php, perl,java, c# or even erlang. Aything but the asterisk dialplan commands.There is no sense in putting yourself through that pain.___
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[asterisk-users] SIP NOTIFY routing problem

[Steve Langstaff]

Title: SIP NOTIFY routing problem






In version 1.2.7.1 I have an endpoint (number 5302) registered.


'sip show peer 5302' shows that the Reg. Contact address is:

 sip:[EMAIL PROTECTED]:5066


When I call 5302 I see INVITE messages correctly routed to the contact address with request lines like:


 INVITE sip:[EMAIL PROTECTED]:5066 SIP/2.0


But when NOTIFY messages are sent, the request lines are incorrect, like this:


NOTIFY sip:[EMAIL PROTECTED] SIP/2.0


Can anyone tell me whether Asterisk 1.4 (or a later version of 1.2) has the NOTIFY routing correct?


__

Steve Langstaff



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Re: [asterisk-users] Time Based Voicemail Messages

[C F]

On 11/15/06, Wildheart [EMAIL PROTECTED] wrote:

Hi,

   I want to change my voicemail message based on the time of day. I would
like a message that says Sorry the office is now closed. after a
certain time, and says Sorry I am unavailable / Busy / etc before.

   I have come up with two ways of doing it:

 1. A cron job to replace the files (messy)

 2. Using different mailboxes at the different times (this means I have 2
mailboxes to check).


No, you can have 2 mailboxes for different times like this:
in extensions.conf:
exten = s,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED],u); for day
exten = s,1,Voicemail([EMAIL PROTECTED],u);for night
in voicemail.conf:
 = ,User User,,,delete=yes
 = ,User User,,,delete=no
now you only have to check the voicemail for mailbox 




   Is there a way that the voicemail could be enhanced by adding a feature
like this?

With thanks,

  Tim

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[asterisk-users] State of a public number

[Giordano Grandis]

Hi 
guys,
i would check the 
state of a number on a Zap channel, i suppose that i cannot use ExtensionState 
that works only for SIP and IAX. 
Anyone has any ides 
? Could i check the state of a pubblic number before transfer it a internal 
call?

Thanks in 
advance

Giordano


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Re: [asterisk-users] Time Based Voicemail Messages

[tracinet]

What you *could* do is record one greeting as the unavailable message and another as the busy message and during the day, just play the unavailable one and at night play the busy one...
On 11/15/06, C F [EMAIL PROTECTED] wrote:
On 11/15/06, Wildheart [EMAIL PROTECTED] wrote: Hi,I want to change my voicemail message based on the time of day. I would
 like a message that says Sorry the office is now closed. after a certain time, and says Sorry I am unavailable / Busy / etc before.I have come up with two ways of doing it:
1. A cron job to replace the files (messy)2. Using different mailboxes at the different times (this means I have 2 mailboxes to check).No, you can have 2 mailboxes for different times like this:
in extensions.conf:exten = s,1,Voicemail([EMAIL PROTECTED][EMAIL PROTECTED],u); for dayexten = s,1,Voicemail([EMAIL PROTECTED],u);for nightin voicemail.conf: = ,User User,,,delete=yes = ,User User,,,delete=no
now you only have to check the voicemail for mailbox Is there a way that the voicemail could be enhanced by adding a feature like this? With thanks,
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Re: [asterisk-users] Soundfiles adding during phone calls

[Ronald Wiplinger]

Tom Lynn wrote:

Ron,
The guy is trying to help you.  Go to the link and read it.  There is 
a feature that you can use to play a recording into the voice 
channel.  Mine is set so when you press #9, the caller hears the lots 
of monkeys recording.


The best part of it is that you can hang up and the recording will 
continue to play to the caller.  When it expires, so does the call


I tried this:
features.conf
[featuremap]
blindxfer = ##; Blind transfer  was #1 - now press # twice
disconnect = *0; Disconnect
automon = *1; One Touch Record
atxfer = *2; Attended transfer
tortore= *9,callee,Playback,tt-monkeys


extensions.conf
exten = 601,1,Set(DYNAMIC_FEATURES=hangup#play#tortore#automon) ; 
enable One-touch

exten = 601,2,Dial(${PHONE_601},30,tTwWr)


I make a call from 615 to 601
601 hits *9   but nothing happens!

when 601 hits *1  it records the conversion.

What do I miss?


bye

Ronald Wiplinger


On 11/11/06, * Ronald Wiplinger* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Andrew Joakimsen wrote:
 http://www.voip-info.org/wiki-Asterisk+config+features.conf

... and where exactly did you see this feature


bye

Ronald Wiplinger

 On 11/11/06, *Ronald Wiplinger * [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
 mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]  wrote:

 I want to add some sound filed on demand during a phone call
only
 possible on some extension numbers.


 I get many phone calls from local companies, but don't
understand
 Chinese! I would like to record the call, but also ask the
caller some
 questions, which should be added into the call with some
keys on the
 phone, ... e.g.  *66554 should add into the call: How are
you? or What
 is your phone number?


 But I do have another application for that too.
 I get many fake phone calls, where Chinese people tell you
that your
 phone bill is not paid, your court fee is not paid,  and
ask the
 caller to go to the ATM machine and key in a series of key
 strokes, 
 most likely it will clear out your account.
 For such fake callers I would like to add a terrible noise
to the
 call
 and make scare them as much as possible.

 Such fake calls I get now for each of my phone lines at least 10
 each!!!
 Either the caller-id is not set, is 0 or is a tollfree number.


 bye

 Ronald Wiplinger
 



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[asterisk-users] ODBC Voicemail Storage

[Edwin Horton]

I current have a working Asterisk 1.2.12 server with ODBC voicemail storage,
realtime static maps for voicemail, sip and iax configuration files.
Realtime extensions, etc.  All works great.  I have verified that this
configuration works on my test server as well.  Now I am trying to test the
1.4B3 version on the same test server, and all works well except for ODBC
voicemail.  I am using the same table structure as before (extended ODBC),
and the ODBC system works well in that I can use it for the static maps
(extconfig.conf), or mysql native from the addons package.  With Asterisk
compiled without ODBC voicemail, it works flawless.  Anyway, Asterisk with
ODBC voicemail compile option will not start with the following console
message:

  == Parsing '/etc/asterisk/voicemail.conf': Found
[Nov 15 09:18:47] DEBUG[23599]: app_voicemail.c:7056 load_config: VM
Temperary Greeting Reminder Option disabled globally
[Nov 15 09:18:47] DEBUG[23599]: app_voicemail.c:7082 load_config: ENVELOPE
before msg enabled globally
[Nov 15 09:18:47] DEBUG[23599]: app_voicemail.c:7110 load_config: found
dialout context: fromvm
[Nov 15 09:18:47] DEBUG[23599]: app_voicemail.c:7117 load_config: found
callback context: fromvm
  == Parsing '/etc/asterisk/users.conf': Found
app_voicemail.so = (Comedian Mail (Voicemail System) with ODBC Storage)
  == Registered channel type 'Local' (Local Proxy Channel Driver)
chan_local.so = (Local Proxy Channel)
  == Parsing '/etc/asterisk/codecs.conf': Found
-- codec_gsm: using generic PLC
  == Registered translator 'gsmtolin' from format gsm to slin, cost 5
asterisk: symbol lookup error: /usr/lib/asterisk/modules/app_voicemail.so:
undefined symbol: odbc_request_obj

I get no other information in the debug or message files.  An attempt to
backtrace, does not yield a crash dump regardless of the compile options.
Does anyone have any ideas?
Ed Horton
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RE: [asterisk-users] Add Apps to Asterisk?

[Matthew Rubenstein]

Thanks for the reply. There's
no /usr/lib/asterisk/modules/app_meetme.so , though that dir has all the
libraries for all the other modules I see in CLI 'show modules' (no
meetme there, either, as I noted). /etc/asterisk/modules.conf starts
with

[modules]
autoload=yes

and there's no 'noload =' directive specifying meetme. So it's clearly
not installed.

Maybe there's a package that's not installed? Or maybe I have to
download/compile/install a source tarfile. How can I find out if a
package that includes Meetme is available (for CentOS) but not
installed, or just get an installable tarfile (with instructions for
installing and running my upgraded Asterisk)? Thanks for your insights.


On Tue, 2006-11-14 at 22:53 -0800, Darryl Dunkin wrote:
 First, check for app_meetme.so in /usr/lib/asterisk/modules (wherever
 your modules path is). Next, in the CLI, do a 'show modules' to see if
 it is there. If not, check your modules.conf and add in 'load =
 app_meetme.so' assuming autoload is not enabled.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Matthew
 Rubenstein
 Sent: Tuesday, November 14, 2006 21:09
 To: Asterisk-Users
 Subject: [asterisk-users] Add Apps to Asterisk?
 
   I've got an Asterisk (v1.2.11) installation running, but it
 doesn't
 seem to have the Meetme() app. At the CLI, I type Meetme , and it
 responds No such command 'Meetme'; meetme doesn't show up in CLI show
 modules . I'm running a SIP-only server at a datacenter where I can't
 add Digium (or any other) HW, and am running under CentOS. There is
 an /etc/asterisk/meetme.conf file, but I don't see anything to use it.
 
   What do I have to do, exactly, to install Meetme? What about the
 Conference command, or others not installed? I'd prefer to use the
 CentOS package system as much as possible, but I can compile source if
 necessary. Is there a HowTo on the Web somewhere that details this
 process?
-- 

(C) Matthew Rubenstein

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Re: [asterisk-users] trixbox + agi

[Tom Vile]

yes, you can use Trixbox.On 11/15/06, blackwater dev [EMAIL PROTECTED] wrote:
For the Asterisk side of things, are you using Asterisk directly or Trixbox? I'm just trying to get a prototype working so don't want to spend a lot of time on the initial asterisk setup. If Trixbox will allow me to do the php+agi integration, I'll do that, if not, will try to just try to install Asterisk from source.
Thanks!On 11/15/06, Tim Uckun 
[EMAIL PROTECTED] wrote:
If I were you I would go the AGI way. Use ruby, python, php, perl,java, c# or even erlang. Aything but the asterisk dialplan commands.There is no sense in putting yourself through that pain.___
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 --asterisk-users mailing listTo UNSUBSCRIBE or update options visit:  
http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony
www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856
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[asterisk-users] Re: Broken Call Screening

[Gary T. Giesen]

There's no other way? Surely I can't be the first person that hasn't
wanted to do this before.

On 11/14/06, Justin Newman [EMAIL PROTECTED] wrote:


You need to modify app_queue.c to hold off on bridging until the receiving
party has accepted the call. If the receiving party rejects (hangup, digit
other than '1', timeout, etc), leave or put the calling party back in at
close to the same level.

Justin

--

Date: Tue, 14 Nov 2006 10:14:04 -0500
From: Gary T. Giesen [EMAIL PROTECTED]
Subject: [asterisk-users] Broken Call Screening

...

I have a cell phone added to a queue as a local extension (member =
Local/299). I want the cell phone to be able to reject calls to the
queue without the person sitting in the queue being hung up on, etc.
The way my dialplan is set up, the person hits 1 to answer the call
and any other key to reject it. It works flawlessly in that regard.
If it goes to the cell phone voicemail, it works great too, it times
out and rejects the call, all without the caller knowing. Where it
breaks is when the person answers the cell phone and then hangs up
without any input or letting it time out. The music on hold is stopped
and the caller is left there with dead air. Does anyone have any ideas
on how to fix this or a better way to implement this?

Output when the call is dropped:

   -- Channel 0/3, span 1 got hangup request
   -- User disconnected
   -- Stopped music on hold on Local/[EMAIL PROTECTED],2
Nov 13 16:21:26 WARNING[12709]: res_features.c:1374 ast_bridge_call:
Bridge failed on channels Local/[EMAIL PROTECTED],2 and Zap/3-1
   -- Hungup 'Zap/3-1'
   -- Local/[EMAIL PROTECTED],1 answered SIP/7960A-Gary1-63f2
   -- Stopped music on hold on SIP/7960A-Gary1-63f2


...

Regards,


Gary


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[asterisk-users] Setting the CallerID

[Tobias Wolf]

Hi,

I have some trouble with setting my CallerID if i make an international
Call. No Problems with National Calls, i can set whatever I want. We pay
for this service but our telephone provider was not able to state clear,
wether the number we set on an international call should be shown on the
other side.

Actually only our base number shows up.

If I understand it correctly, in every call the base number is embedded
as ANI, so that we can be billed.

Is it possible, that, if a calls goes international, they only refer to
the ANI and forget the set number ?

The only route I am trying is from Germany to England, maybe it is a
problem between the providers and not of my setup.

Maybe someone could explain to me how CallerID transmission is done on
the technical level I could guess where i have to look for an solution.

Thx.

Tobias Wolf
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[asterisk-users] Disabling Features Temporarily

[Mailing List]

There is a company that I call that requires a * be dialed to break out of 
their IVR.
The problem is Asterisk is grabbing that * for itself. Is there a way to get 
this sent?


asterisk1*CLI show features 
Builtin Feature   Default Current

---   --- ---
Pickup*8  *8 
Blind Transfer#   *2 
Attended Transfer
One Touch Monitor *1 
Disconnect Call   *   *3 



asterisk1*CLI show version 
Asterisk 1.2.12.1 built by root @ asterisk1.local on a i686 running Linux on 2006-09-27 19:41:58 UTC


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[asterisk-users] Problems with language support

[Diego Andres Asenjo G.]

Hi!

I have configured the language support in asterisk to reproduce spanish 
prompts. I have lines for it in sip.conf, iax.conf, zapata.conf and 
voicemail.conf as shown:


[general]
...
language=es
...

In zaptel.conf

loadzone = es
defaultzone = es

When I check my voicemail I get in the CLI:

-- Playing 'digits/4' (language 'en')
-- Playing 'vm-Old' (language 'en')
-- Playing 'vm-messages' (language 'en')

Almost all the messages are played in english.

If I make a

*CLI add extension 7548,1,Playback,digits/day-3 into phones

and call 7548 I hear miércole as should be.

Can you help me debug the problem?

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RE: [asterisk-users] ODBC Voicemail Storage

[Dan Austin]

Edwin wrote:
 I current have a working Asterisk 1.2.12 server with ODBC
 voicemail storage, realtime static maps for voicemail, sip
 and iax configuration files.  Realtime extensions, etc.  
 All works great.  I have verified that this configuration 
 works on my test server as well.  Now I am trying to test 
 the 1.4B3 version on the same test server, and all works 
 well except for ODBC voicemail.  I am using the same table 
 structure as before (extended ODBC), and the ODBC system 
 works well in that I can use it for the static maps 
 (extconfig.conf), or mysql native from the addons package.  
 With Asterisk compiled without ODBC voicemail, it works 
 flawless.  Anyway, Asterisk with ODBC voicemail compile 
 option will not start with the following console message:

I believe this issue was fixed in SVN branches/1.4 in the last
couple of days.

Dan

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Re: [asterisk-users] some questions about atxfer usage

[Conrad Wood]

On Wed, 2006-11-15 at 13:54 +0100, Antonio Almodóvar wrote:
 Hi all.
 
 I have enabled the attended transfer feature in features.conf. I'm
 using it and I want to resolve some questions, I hope someone can help
 me :)
 
 When I transfer a call to an extension:
  - The extension rings during 15 seconds and the call returns to the
 transferer. Is there any possibility to recover the call before the
 timeout of 15 seconds expires?
 

I just press * to retrieve the caller again - Have you tried that?

 I mean, I would like to personalize the way of making transfers using
 the feature of atxfer. How can I do that?

anything in particular?

Conrad

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Re: [asterisk-users] Recording outbound analog calls with X100P

[Matthew J. Roth]

Matthew J. Roth wrote:

Is it possible to record outbound analog calls using an X100P?

I was asked if I knew how to record all calls for a shop with 4 analog 
phones transparently to the end users.  I thought Asterisk was a good 
fit for this and I envisioned using either Digium TDM400Ps or Sangoma 
A200s with 4 FXO and 4 FXS modules.  The FXO modules would be 
connected to the existing PBX and the FXS modules to the existing 
analog phones.  Then with a simple dialplan, all inbound and outbound 
calls could be recorded by Monitor.


I wanted to mock this up using some X100Ps that I had laying around, 
but found that I could only record inbound calls.  I believe that I 
need an FXS interface to record outbound analog calls but my past 
experience is with T1 interfaces, so I could be mistaken.


If anyone could suggest any improvements to my recording scheme, they 
would also be appreciated.
As per ManxPower at #asterisk, it is not possible to record a call 
dialed from an analog phone connected to the Phone In port of an X100P 
because the two ports on the card are hard-wired together.


I'm still looking for feedback on the recording scheme.  Right now I'm 
leaning towards using Sangoma A200s.  Any input from the community would 
be appreciated.


Thank you,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

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[asterisk-users] Page() Function Timeout

[Ken Williams]

I'm trying to use a 
simple page function. It starts a MeetMe conference with the devices I've 
listed, but the devices hang up after 3-5 seconds. After doing some 
research I found this was a problem, and I needed to remove a (5) from 
app_page.c

Well, my app_page.c 
didn't have the (5). I did make clean; make install again just in case I 
had some weird compiled version installed that had the (5) in it. After 
compiling I restarted the asterisk service and tried paging again and still had 
the same problem.

In the CLI I get the 
following, which you can see the (5) is still in there somehow. 


 
-- Playing 'beep' (language 'en') -- Launching 
MeetMe(1010553064d|mqxdw(5)) on SIP/710-09a50038 -- 
Created MeetMe conference 1023 for conference 
'1010553064d' -- Launching MeetMe(1010553064d|mqxdw(5)) on 
SIP/717-09a48758
I've grep'd the 
entire src folder for \(5\) as well as qxd trying to find all instances of this, 
and the only ones are listed in the app_page.c file. Any suggestions on 
where to get this rogue (5) out of here?

 snprintf(meetmeopts, 
sizeof(meetmeopts), "%ud|%sqxdw", confid, ast_test_flag(flags, PAGE_DUPLEX) 
? "" : "m");

and 


 if (!res) 
{ 
snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%sqxd", confid, 
$ 
pbx_exec(chan, app, meetmeopts, 
1); }
are the only 
sections of the app_page.c that have the meetme call in it.

My page functions, 
fwiw, both have the same problem:

;Paging

exten = 
760,1,SIPAddHeader(Call-Info: answer-after=0)exten = 
760,2,Page(SIP/717SIP/710SIP/702|d)exten = 
760,3,Hangup

exten = 
761,1,SIPAddHeader(Call-Info: answer-after=0)exten = 
761,2,Page(SIP/717SIP/710SIP/702)exten = 
761,3,Hangup
Any suggestions 
would be very helpful.
Ken
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Re: [asterisk-users] Setting the CallerID

[Conrad Wood]

On Wed, 2006-11-15 at 17:50 +0100, Tobias Wolf wrote:
 Hi,
 
 I have some trouble with setting my CallerID if i make an international
 Call. No Problems with National Calls, i can set whatever I want. We pay
 for this service but our telephone provider was not able to state clear,
 wether the number we set on an international call should be shown on the
 other side.
 
 Actually only our base number shows up.

With that, do you mean the number without international prefix? That's
odd.
What's the provider you are calling here in uk?
I regular get calls from Germany with the correct international callerid
showing.

 
 If I understand it correctly, in every call the base number is embedded
 as ANI, so that we can be billed.
 
 Is it possible, that, if a calls goes international, they only refer to
 the ANI and forget the set number ?

 The only route I am trying is from Germany to England, maybe it is a
 problem between the providers and not of my setup.

I don't receive any callerid from some (cheaper) german telco providers
(but most work correctly)
I vaguely remember that there are some pretty dodgy contractual
agreements lingering around. You might want to google for 'europe
callerid', specificially [1].
Are you sure your calls goes straight from Germany to UK? I found that
many German telecom providers terminate through the US, particularly to
landlines.

Maybe you could route your call to via sip to a uk voip provider and
persuade them to set the callerid to whatever you have in Germany? or
simply get a UK number routed via sip ? ;)


[1]
http://www.ainslie.org.uk/callerid/cli_faq.htm
In the UK, Oftel will allow European Caller ID if the other country has
implemented the Telecoms Privacy Directive



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Re: [asterisk-users] Recording outbound analog calls with X100P

[Conrad Wood]

 As per ManxPower at #asterisk, it is not possible to record a call 
 dialed from an analog phone connected to the Phone In port of an X100P 
 because the two ports on the card are hard-wired together.

A bit off-topic maybe, but does that then mean you can't 
make 2 simultaneous calls through the card? E.g. 
1. Call:  pstn-phone - asterisk - sip...
2. Call:  sip-phone - asterisk - pstn...

Because if you could, you could try some trickery with Meetme or Local
channels but it sounds like a pretty big limitation of the card, right?

Conrad


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Re: [asterisk-users] Asterisk - big installation

[Conrad Wood]

On Wed, 2006-11-15 at 14:41 +0100, doki_cti wrote:
 Hello
 I want build big asterisk server. Server will be work as gateway between PSTN 
 and VoIP network. I think about 1000 SIP users accounts and 4 E1 ports. I 
 know that preformance in this case depend on codeck which will be use. I want 
 use card with CAPI interface. Can you describe me your experience with this?
 If you have some big installaion, please wriete some info about server 
 (procesor, ram etc), numbers of user and simultaneous calls beetwene VoIP and 
 PSTN. How often server crash?
 Regards
 Doki
 _

why capi? why not a sangoma or digium card?
If you have a 1000 Sip users you'll need to do more than just a 'big'
server - google and browse this list, there are plenty of people who
published their server hardware specifications and call lists.
Have you looked at sip proxying? Maybe use multiple smallish servers?
The server should not crash more than you want it to, whatever that is.
If you plan to run it
non-stop for say 4 years, plan it out accordingly and it can be done. If
you run multiple small ones, maybe you can accept 1 crashed server every
3 months.
Unless you have lots of time and patience, your best next step is to ask
on the -biz list for someone to help with this installation.

Conrad

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[asterisk-users] dtmf tones not always recognized

[Don Pobanz]

Title: dtmf tones not always recognized 






We have analog phones (Aastra 390) connected to channels banks (Adtran TA750) connected to a 4 port digium card( TE410P).

Because of echo problems we purchased external T1 echo cancellers from Orion Telecom. (The TE412P did not eliminate enough of the echo).

The Orioan echo cancellers eliminated our echo, however 5-15% of the time the first digit pressed is not recognized by * and the dialtone continues. By removing the echo cancellers, the dtmf works 100% of the time.

By changing the losses and gains in the echo canceller, in the channel bank or in *, we can see changes in the % of recognized dtmf tones. (I believe, that if the 1st digit is recognized, then all of the following are recognized.)

Orion tells me that it is a volume issue since they have thousands of the same product in use elseware. (They have offered significant assistance in trouble shooting the probem). We also have a different location with no problems with the same setup. In fact we moved an echo canceller to that location and when set up the same it worked well.

My thought is that the dialtone is interfering with the echo cancellers ability to pass the dtmf tones through cleanly.

Is there a way to decrease the dial tone volume?

Any other suggestions is appreciated!

Don Pobanz




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RE: [asterisk-users] Page() Function Timeout

[Ken Williams]

BAH!

My Makefile in the apps folder was missing 
app_page.c. I added it, recompiled, page is working 
properly.


From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Ken 
WilliamsSent: Wednesday, November 15, 2006 10:33 AMTo: 
Asterisk Users Mailing List - Non-Commercial DiscussionSubject: 
[asterisk-users] Page() Function Timeout

I'm trying to use a 
simple page function. It starts a MeetMe conference with the devices I've 
listed, but the devices hang up after 3-5 seconds. After doing some 
research I found this was a problem, and I needed to remove a (5) from 
app_page.c

Well, my app_page.c 
didn't have the (5). I did make clean; make install again just in case I 
had some weird compiled version installed that had the (5) in it. After 
compiling I restarted the asterisk service and tried paging again and still had 
the same problem.

In the CLI I get the 
following, which you can see the (5) is still in there somehow. 


 
-- Playing 'beep' (language 'en') -- Launching 
MeetMe(1010553064d|mqxdw(5)) on SIP/710-09a50038 -- 
Created MeetMe conference 1023 for conference 
'1010553064d' -- Launching MeetMe(1010553064d|mqxdw(5)) on 
SIP/717-09a48758
I've grep'd the 
entire src folder for \(5\) as well as qxd trying to find all instances of this, 
and the only ones are listed in the app_page.c file. Any suggestions on 
where to get this rogue (5) out of here?

 snprintf(meetmeopts, 
sizeof(meetmeopts), "%ud|%sqxdw", confid, ast_test_flag(flags, PAGE_DUPLEX) 
? "" : "m");

and 


 if (!res) 
{ 
snprintf(meetmeopts, sizeof(meetmeopts), "%ud|A%sqxd", confid, 
$ 
pbx_exec(chan, app, meetmeopts, 
1); }
are the only 
sections of the app_page.c that have the meetme call in it.

My page functions, 
fwiw, both have the same problem:

;Paging

exten = 
760,1,SIPAddHeader(Call-Info: answer-after=0)exten = 
760,2,Page(SIP/717SIP/710SIP/702|d)exten = 
760,3,Hangup

exten = 
761,1,SIPAddHeader(Call-Info: answer-after=0)exten = 
761,2,Page(SIP/717SIP/710SIP/702)exten = 
761,3,Hangup
Any suggestions 
would be very helpful.
Ken
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[asterisk-users] Got 200 OK on REGISTER that isn't a register

[Stas Khromoy]

chan_sip.c: Got 200 OK on REGISTER that isn't a register.

i'm getting the above warning
while trying to register a phone from outside of asterisk network.
( so no registration what so ever, no dial tone and what not)


it registered once for about 20 minutes
exepted calls and i could call out
but with no audio on either end.

any ideas ?

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[asterisk-users] quadbri + kernel 2.6.18.1

[Paco Brufal]

Hello,

I have an Asterisk system with kernel 2.6.18.1 and one quadbri. I
have installed the latest bristuff patches (0.3.0-PRE-1s). The system works
fine, but when I do a reboot, the system hangs unloading module qozap.

Is there any known problem with latest 2.6 kernels and qozap module?

Thanks.

-- 

Paco Brufal[EMAIL PROTECTED]
ServiTux Servicios Informáticos S.L.
Tel. 966 160 600 / Fax. 966 160 601
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Re: [asterisk-users] Page() Function Timeout

[Steven Ringwald]

Ken Williams wrote:
I'm trying to use a simple page function.  It starts a MeetMe 
conference with the devices I've listed, but the devices hang up after 
3-5 seconds.  After doing some research I found this was a problem, 
and I needed to remove a (5) from app_page.c
 
Well, my app_page.c didn't have the (5).  I did make clean; make 
install again just in case I had some weird compiled version installed 
that had the (5) in it.  After compiling I restarted the asterisk 
service and tried paging again and still had the same problem.
 
In the CLI I get the following, which you can see the (5) is still in 
there somehow. 
 
-- Playing 'beep' (language 'en')

-- Launching MeetMe(1010553064d|mqxdw(5)) on SIP/710-09a50038
-- Created MeetMe conference 1023 for conference '1010553064d'
-- Launching MeetMe(1010553064d|mqxdw(5)) on SIP/717-09a48758
I've grep'd the entire src folder for \(5\) as well as qxd trying to 
find all instances of this, and the only ones are listed in the 
app_page.c file.  Any suggestions on where to get this rogue (5) out 
of here?
 
snprintf(meetmeopts, sizeof(meetmeopts), %ud|%sqxdw, confid, 
ast_test_flag(flags, PAGE_DUPLEX) ?  : m);
 
and
 
if (!res) {
snprintf(meetmeopts, sizeof(meetmeopts), %ud|A%sqxd, 
confid, $

pbx_exec(chan, app, meetmeopts, 1);
}
are the only sections of the app_page.c that have the meetme call in it.
 
My page functions, fwiw, both have the same problem:
 
;Paging
 
exten = 760,1,SIPAddHeader(Call-Info: answer-after=0)

exten = 760,2,Page(SIP/717SIP/710SIP/702|d)
exten = 760,3,Hangup
 
exten = 761,1,SIPAddHeader(Call-Info: answer-after=0)

exten = 761,2,Page(SIP/717SIP/710SIP/702)
exten = 761,3,Hangup
Any suggestions would be very helpful.


I had the same problem and ended up changing the 5 to a 300. If you 
don't specify a (N) after the 'w', I believe it defaults to 5.


Steve


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[asterisk-users] Handy tip for intercom with FreePBX Grandstream phones

[Ken Williams]

We use intercom 100% 
inter-office. To get FreePBX to do this with Grandstreams by default 
without having to create intercom or paging groups, just change the following 
line (line #58) in your extensions.conf from:

exten = s,10,Dial(${ds}) 
; dialparties will set the priority to 10 if $ds is not null
to:

exten = s,10,SIPAddHeader(Call-Info: 
answer-after=0) ;dialparties will set 
thepriorityto10if$dsisnotnullexten = 
s,11,Dial(${ds}) 


Hope this helps 
someone in the future.
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Re: [asterisk-users] Got 200 OK on REGISTER that isn't a register

[Stas Khromoy]

as far as i know sip-based

i came across something that said
this could be due to too much traffic
but the mesage was not clear on what side


 Original Message  
Subject: Re:[asterisk-users] Got 200 OK on REGISTER that isn't a register
From: Ron McLeod [EMAIL PROTECTED]
To: [EMAIL PROTECTED], 'Asterisk Users Mailing List - Non-Commercial 
Discussion' asterisk-users@lists.digium.com

Date: 11/15/2006 2:03 PM

I think this message is saying that it received a 200 OK for a REGISTER
message that Asterisk does not know about (anymore).

Is you system trying to register with an ITSP or other SIP-based system?

  

-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Stas Khromoy
Sent: Wednesday, November 15, 2006 10:38 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Got 200 OK on REGISTER that isn't a register

chan_sip.c: Got 200 OK on REGISTER that isn't a register.

i'm getting the above warning
while trying to register a phone from outside of asterisk network.
( so no registration what so ever, no dial tone and what not)


it registered once for about 20 minutes
exepted calls and i could call out
but with no audio on either end.

any ideas ?

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RE: [asterisk-users] Got 200 OK on REGISTER that isn't a register

[Ron McLeod]

I think this message is saying that it received a 200 OK for a REGISTER
message that Asterisk does not know about (anymore).

Is you system trying to register with an ITSP or other SIP-based system?

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Stas Khromoy
 Sent: Wednesday, November 15, 2006 10:38 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Got 200 OK on REGISTER that isn't a register
 
 chan_sip.c: Got 200 OK on REGISTER that isn't a register.
 
 i'm getting the above warning
 while trying to register a phone from outside of asterisk network.
 ( so no registration what so ever, no dial tone and what not)
 
 
 it registered once for about 20 minutes
 exepted calls and i could call out
 but with no audio on either end.
 
 any ideas ?
 
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Re: [Asterisk-Users] Got 200 OK on REGISTER that isn't a register

[Stas Khromoy]

i figured that out
what i can't find is a solution to the problem

 Original Message  
Subject: Re:[Asterisk-Users] Got 200 OK on REGISTER that isn't a register
From: BJ Weschke [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Date: 1/1/2006 12:26 PM

On 1/1/06, Robert La Ferla [EMAIL PROTECTED] wrote:
  

What does this warning mean?

WARNING[11065]: chan_sip.c:9596 handle_response_register: Got 200 OK on
REGISTER that isn't a register




 Your SIP device is returning a 200 OK message about a registration
attempt, but Asterisk doesn't believe there is a registration attempt
in progress with this phone. This is what's generating the message.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[asterisk-users] safe_asterisks pawning multiple asterisk process???

[Andre Courchesne - Consultant]

We have 1 server that after a few hours operating has multiple process 
of asterisk running. Here is the pstree output:


# pstree
init-+-atftpd
|-auditd---{auditd}
|-bash---safe_opserver---op_server.pl
|-crond
|-cwASTcall.pl
|-dbus-daemon
|-events/0
|-hald-+-hald-addon-acpi
|  `-2*[hald-addon-stor]
|-httpd---3*[httpd]
|-khelper
|-klogd
|-ksoftirqd/0
|-kswapd0
|-kthread-+-aio/0
| |-ata/0
| |-hda_codec/0
| |-kacpid
| |-kauditd
| |-kblockd/0
| |-khubd
| |-kseriod
| |-2*[pdflush]
| |-reiserfs/0
| |-rpciod/0
| |-scsi_eh_0
| |-scsi_eh_1
| `-scsi_eh_2
|-2*[mingetty]
|-mysqld_safe---mysqld---16*[{mysqld}]
|-ntpd
|-safe_asterisk---asterisk-+-45*[asterisk]
|  `-22*[{asterisk}]
|-sshd---sshd---bash---pstree
|-syslogd
|-udevd
|-usb-storage
`-wan_ecd---wan_ecd

And ps aux | grep asterisk:

# ps aux | grep asterisk
asterisk  2047  0.0  0.1   9200  1516 ?SNov14   0:00 
/usr/sbin/httpd
asterisk  2084  0.0  0.2  11544  2388 ?SNov14   0:00 
/usr/sbin/httpd
asterisk  2085  0.0  0.2  11544  2384 ?SNov14   0:00 
/usr/sbin/httpd
root  2196  0.0  0.0   2172   456 ?SNov14   0:00 /bin/sh 
/usr/sbin/safe_asterisk -U asterisk -G asterisk
asterisk  2215  1.6 10.0 122496 90984 ?Sl   Nov14  38:07 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk  2250  0.0  0.0   2176   376 ?SNov14   0:00 -bash 
-c cd /var/www/AMP/panel  /var/www/AMP/panel/safe_opserver 
asterisk  2251  0.0  0.0   2128   868 ?SNov14   0:00 
/bin/bash /var/www/AMP/panel/safe_opserver
asterisk  2253  3.2  0.8   8988  7336 ?RNov14  73:54 
/usr/bin/perl -w ./op_server.pl
asterisk 12105  0.0  0.8  31440  7804 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 13080  0.0  0.8  32096  7616 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 18352  0.0  0.9  36080  8684 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 19352  0.0  0.9  36528  8764 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 25402  0.0  0.9  39196  8972 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 26497  0.0  1.0  40448  9372 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 30901  0.0  1.0  42064  9308 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk  3160  0.0  0.6  43968  5624 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 12444  0.0  0.5  49636  5148 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 22538  0.0  0.5  54532  5148 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 26032  0.0  0.5  56948  5148 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 27517  0.0  0.5  57056  5148 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 31806  0.0  1.0  58956  9800 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk  3655  0.0  1.0  60088  9932 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk  4636  0.0  1.1  60956 10316 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk  7494  0.0  1.2  62200 10952 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk  9276  0.0  1.3  64856 12040 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 11592  0.0  1.4  65404 12720 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 12757  0.0  1.4  66808 13504 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 13167  0.0  1.4  66576 13296 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 13363  0.0  1.4  65936 13156 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 16251  0.0  1.6  68812 14664 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 16942  0.0  1.6  68600 14676 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 24818  0.0  1.6  72740 15308 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 29714  0.0  1.7  75332 15824 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk  5114  0.0  1.6  78932 15144 ?SNov14   0:00 
/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk  

[asterisk-users] PHPAGI example usage of input.php

[Tom Vile]

I am trying to get the example input.php working from PHPAGI but it will not playback the letters that I put in because of this error:Nov 15 14:25:22 WARNING[18678] file.c: File /tmp/swift_f87b365372c500c76e497087ac7e470a does not exist in any format
But the file does exist and I see the entries for the key presses that I put in but it will not stream the file back to me using Cepstral.Asterisk 1.2.9CentOS 4.2Thanks,Tom Vile

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Re: [asterisk-users] Recording outbound analog calls with X100P

[Hadley Rich]

On Thursday 16 November 2006 06:44, Conrad Wood wrote:
 On Thursday 16 November 2006 06:42, Matthew J. Roth wrote:
  As per ManxPower at #asterisk, it is not possible to record a call
  dialed from an analog phone connected to the Phone In port of an X100P
  because the two ports on the card are hard-wired together.

 A bit off-topic maybe, but does that then mean you can't
 make 2 simultaneous calls through the card? E.g.
 1. Call:  pstn-phone - asterisk - sip...
 2. Call:  sip-phone - asterisk - pstn...

As he said above, the ports are wired together. There is no FXS device on that 
card.

-- 
http://nicegear.co.nz
New Zealand's VoIP supplier
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Re: [asterisk-users] PHPAGI example usage of input.php

[Jay Moore]

Are you including the file extension?

Jay

Tom Vile wrote:
I am trying to get the example input.php working from PHPAGI but it will 
not

playback the letters that I put in because of this error:

Nov 15 14:25:22 WARNING[18678] file.c: File
/tmp/swift_f87b365372c500c76e497087ac7e470a does not exist in any format

But the file does exist and I see the entries for the key presses that I 
put

in but it will not stream the file back to me using Cepstral.

Asterisk 1.2.9
CentOS 4.2

Thanks,

Tom Vile




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Re: [asterisk-users] PHPAGI example usage of input.php

[Tom Vile]

There are no file extensions. It is just-rw-r--r-- 1 asterisk asterisk 32 Nov 15 12:52 swift_082da06a422be49e3a475925d9fc50e6-rw-r--r-- 1 asterisk asterisk 7 Nov 15 12:52 swift_6fc422233a40a75a1f028e11c3cd1140
-rw-r--r-- 1 asterisk asterisk 13 Nov 15 12:52 swift_80339585692b0188288da14748213dcc-rw-r--r-- 1 asterisk asterisk 11 Nov 15 12:54 swift_f87b365372c500c76e497087ac7e470a
On 11/15/06, Jay Moore [EMAIL PROTECTED] wrote:
Are you including the file extension?JayTom Vile wrote: I am trying to get the example input.php working from PHPAGI but it will not playback the letters that I put in because of this error:
 Nov 15 14:25:22 WARNING[18678] file.c: File /tmp/swift_f87b365372c500c76e497087ac7e470a does not exist in any format But the file does exist and I see the entries for the key presses that I
 put in but it will not stream the file back to me using Cepstral. Asterisk 1.2.9 CentOS 4.2 Thanks, Tom Vile 
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-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephonywww.baldwintechsolutions.com
Phone: 518-631-2855 x205Fax: 518-631-2856
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Re: [asterisk-users] Problem with FXS ports of TDM400P

[Noah Miller]

Hi Gustavo -


 I just received two TDM400P cards, but I'm having problems with them.
 On a x86 stable Gentoo box.
 Kernel: 2.6.17
 gcc-4.1.1, glibc-2.4-r4

 Is that an hardware problem? Should I try the other card?

I tried the other card and the problem is still there. REALY NEED HELP


What is your hardware?  The TDM400 cards require PCI 2.2 compliant
hardware.  If you run them with PCI 2.1 or earlier hardware yhou are
liable to run into issues.

- Noah
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Re: [asterisk-users] Dialing from Placed Calls on Polycom IP501doesn't always work

[Noah Miller]

  Has anyone noticed that attempting to place a call from the Placed
  Calls list on a Polycom IP501 by pressing the 'Dial' softkey
 sometimes
  simply returns the phone to the idle screen?

 Yes, I've seen it. We're running 1.6.6, what firmware version do you
 have?

We're running SIP 1.6.6.0036 on the 3.1.3.0131 BootROM.

Did you come up with any reason/fix for this?


I never ran 1.6.6 for any length of time.  1.6.7 and 2.0.1 don't seem
to suffer this issue.  2.0.1 has some buddy watch problems, so you may
not want to use it, but 1.6.7 should be OK.

- Noah
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Re: [asterisk-users] safe_asterisks pawning multiple asterisk process???

[Vicky]

its normal .if there are many calls going . You should worry if your load or memory usage is very high .On 16/11/06, Andre Courchesne - Consultant 
[EMAIL PROTECTED] wrote:
We have 1 server that after a few hours operating has multiple processof asterisk running. Here is the pstree output:# pstreeinit-+-atftpd |-auditd---{auditd} |-bash---safe_opserver---op_server.pl
 |-crond |-cwASTcall.pl |-dbus-daemon |-events/0 |-hald-+-hald-addon-acpi |`-2*[hald-addon-stor] |-httpd---3*[httpd] |-khelper |-klogd |-ksoftirqd/0
 |-kswapd0 |-kthread-+-aio/0 | |-ata/0 | |-hda_codec/0 | |-kacpid | |-kauditd | |-kblockd/0 | |-khubd | |-kseriod
 | |-2*[pdflush] | |-reiserfs/0 | |-rpciod/0 | |-scsi_eh_0 | |-scsi_eh_1 | `-scsi_eh_2 |-2*[mingetty] |-mysqld_safe---mysqld---16*[{mysqld}]
 |-ntpd |-safe_asterisk---asterisk-+-45*[asterisk] |`-22*[{asterisk}] |-sshd---sshd---bash---pstree |-syslogd |-udevd |-usb-storage `-wan_ecd---wan_ecd
And ps aux | grep asterisk:# ps aux | grep asteriskasterisk20470.00.1 92001516 ?SNov14 0:00/usr/sbin/httpdasterisk20840.00.2115442388 ?SNov14 0:00
/usr/sbin/httpdasterisk20850.00.2115442384 ?SNov14 0:00/usr/sbin/httpdroot21960.00.0 2172 456 ?SNov14 0:00 /bin/sh/usr/sbin/safe_asterisk -U asterisk -G asterisk
asterisk22151.6 10.0 122496 90984 ?Sl Nov1438:07/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk22500.00.0 2176 376 ?SNov14 0:00 -bash-c cd /var/www/AMP/panel  /var/www/AMP/panel/safe_opserver 
asterisk22510.00.0 2128 868 ?SNov14 0:00/bin/bash /var/www/AMP/panel/safe_opserverasterisk22533.20.8 89887336 ?RNov1473:54/usr/bin/perl -w ./op_server.pl
asterisk 121050.00.8314407804 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 130800.00.8320967616 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 183520.00.9360808684 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 193520.00.9365288764 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 254020.00.9391968972 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 264970.01.0404489372 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 309010.01.0420649308 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk31600.00.6439685624 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 124440.00.5496365148 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 225380.00.5545325148 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 260320.00.5569485148 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 275170.00.5570565148 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 318060.01.0589569800 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk36550.01.0600889932 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk46360.01.160956 10316 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk74940.01.262200 10952 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk92760.01.364856 12040 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 115920.01.465404 12720 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 127570.01.466808 13504 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 131670.01.466576 13296 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 133630.01.465936 13156 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 162510.01.668812 14664 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 169420.01.668600 14676 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 248180.01.672740 15308 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 297140.01.775332 15824 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk51140.01.678932 15144 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk57160.01.778560 15408 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk96050.01.781680 16228 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 102350.01.881020 16864 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 174290.01.984996 17896 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 180900.02.085480 18176 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 205420.02.086980 18732 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 243700.04.088652 36340 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 302470.06.092268 54432 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 309040.06.192492 55920 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -casterisk 313630.06.292500 56396 ?SNov14 0:00/usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
asterisk 

[asterisk-users] Question about TFTPD server

[Christian]

Hi all,
I have installed this package onto my Debian and placed the files i want the 
Cisco 7960 phone to get from the tftpdboot directory. But it doesn't seem to 
work. Are ther any special settings I should do to this server?
Many thanks for all your help,
Christian


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Re: [asterisk-users] PHPAGI example usage of input.php

[Jay Moore]

1) Try giving it an extension (say .gsm) and seeing if that works.  Make 
sure you change both the file and your script.


2) Does the rest of the script work?  If you run './test.php', do you 
get any errors?


Jay

Tom Vile wrote:

There are no file extensions.  It is just

-rw-r--r--   1 asterisk asterisk   32 Nov 15 12:52
swift_082da06a422be49e3a475925d9fc50e6
-rw-r--r--   1 asterisk asterisk7 Nov 15 12:52
swift_6fc422233a40a75a1f028e11c3cd1140
-rw-r--r--   1 asterisk asterisk   13 Nov 15 12:52
swift_80339585692b0188288da14748213dcc
-rw-r--r--   1 asterisk asterisk   11 Nov 15 12:54
swift_f87b365372c500c76e497087ac7e470a


On 11/15/06, Jay Moore [EMAIL PROTECTED] wrote:


Are you including the file extension?

Jay

Tom Vile wrote:
 I am trying to get the example input.php working from PHPAGI but it 
will

 not
 playback the letters that I put in because of this error:

 Nov 15 14:25:22 WARNING[18678] file.c: File
 /tmp/swift_f87b365372c500c76e497087ac7e470a does not exist in any 
format


 But the file does exist and I see the entries for the key presses 
that I

 put
 in but it will not stream the file back to me using Cepstral.

 Asterisk 1.2.9
 CentOS 4.2

 Thanks,

 Tom Vile


 



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Re: [asterisk-users] Recording outbound analog calls with X100P

[Time Bandit]

Is it possible to record outbound analog calls using an X100P?

I was asked if I knew how to record all calls for a shop with 4 analog
phones transparently to the end users.  I thought Asterisk was a good
fit for this and I envisioned using either Digium TDM400Ps or Sangoma
A200s with 4 FXO and 4 FXS modules.  The FXO modules would be connected
to the existing PBX and the FXS modules to the existing analog phones.
Then with a simple dialplan, all inbound and outbound calls could be
recorded by Monitor.

I wanted to mock this up using some X100Ps that I had laying around, but
found that I could only record inbound calls.  I believe that I need an
FXS interface to record outbound analog calls but my past experience is
with T1 interfaces, so I could be mistaken.


Of course you can, if you have 4 FXO and 4 FXS, you could make a
really simple dialplan and record the calls that pass through it,
incoming or outgoing, and the users wouldn't even know that there is a
pbx between them and the PSTN.

You will need a lot of space to keep them all, but you could make a
simple cron job that would erase any recording older then, say, 2
months.

Also, you would have the benefit of having CDR

hth
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RE: [asterisk-users] Question about TFTPD server

[Ron McLeod]

You may need to create (or modify) the tftp file in /etc/xinetd.d.  For
example:

service tftp
{
disabled= no
socket_type = dgram
protocol= udp
wait= yes
user= root
server  = /usr/sbin/in.tftpd
server_args = -s /var/lib/tftpboot
per_source  = 11
cps = 100 2
flags   = IPv4
}


 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Christian
 Sent: Wednesday, November 15, 2006 12:12 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Question about TFTPD server
 
 Hi all,
 I have installed this package onto my Debian and placed the files i want
 the Cisco 7960 phone to get from the tftpdboot directory. But it doesn't
 seem to work. Are ther any special settings I should do to this server?
 Many thanks for all your help,
 Christian
 
 
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[asterisk-users] Huawei Videophone

[Nabeel Jafferali]

Does anyone have any experience using the Huawei Videophones in a
point-to-multipoint configuration using Asterisk?

Thanks,

Nabeel

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Re: [asterisk-users] PHPAGI example usage of input.php

[Tom Vile]

It does error when I run it from the CLI with error:

sh: -p: command not found

I am assuming that it is referring to this line in phpagi.php

shell_exec({$this-config['cepstral']['swift']} -p
audio/channels=1,audio/sampling-rate=$frequency $voice -o $fname.wav -f
$fname.txt);

On 11/15/06, Jay Moore [EMAIL PROTECTED] wrote:


1) Try giving it an extension (say .gsm) and seeing if that works.  Make
sure you change both the file and your script.

2) Does the rest of the script work?  If you run './test.php', do you
get any errors?

Jay

Tom Vile wrote:
 There are no file extensions.  It is just

 -rw-r--r--   1 asterisk asterisk   32 Nov 15 12:52
 swift_082da06a422be49e3a475925d9fc50e6
 -rw-r--r--   1 asterisk asterisk7 Nov 15 12:52
 swift_6fc422233a40a75a1f028e11c3cd1140
 -rw-r--r--   1 asterisk asterisk   13 Nov 15 12:52
 swift_80339585692b0188288da14748213dcc
 -rw-r--r--   1 asterisk asterisk   11 Nov 15 12:54
 swift_f87b365372c500c76e497087ac7e470a


 On 11/15/06, Jay Moore [EMAIL PROTECTED] wrote:

 Are you including the file extension?

 Jay

 Tom Vile wrote:
  I am trying to get the example input.php working from PHPAGI but it
 will
  not
  playback the letters that I put in because of this error:
 
  Nov 15 14:25:22 WARNING[18678] file.c: File
  /tmp/swift_f87b365372c500c76e497087ac7e470a does not exist in any
 format
 
  But the file does exist and I see the entries for the key presses
 that I
  put
  in but it will not stream the file back to me using Cepstral.
 
  Asterisk 1.2.9
  CentOS 4.2
 
  Thanks,
 
  Tom Vile
 
 
 


 
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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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[asterisk-users] PortSip and Astericks new install

[Charlie Grosvenor]

I have just installed Asterisk and installed the  sample configuration
files. Asterisks appears to be working and I have added a SIP client:

 

[John]

type=friend

secret=test

host=dynamic

allow=all

 

I have been trying to dial the demo number 500 when using PortSip,
Asterisks answers the phone but PortSip gives me the error:

 

Call failed: codec not accepted 488.

 

I have tried changing the enabled codecs in PortSip but this makes no
difference. I have also tried various other SoftPhone but none of them
seem to work.

 

Anybody know what I have missed / doing wrong?

 

Thanks 

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Re: [asterisk-users] trixbox + agi

[Tim Uckun]

For the Asterisk side of things, are you using Asterisk directly or Trixbox?
 I'm just trying to get a prototype working so don't want to spend a lot of
time on the initial asterisk setup.  If Trixbox will allow me to do the
php+agi integration, I'll do that, if not, will try to just try to install
Asterisk from source.



You can use trixbox but be aware of the following.

Trixbox scatters it's config files. Some stuff is kept in the
database, some in the conf files.
You have to keep your configuration in specific files that won't be overrritten.
Trixbox has it's own contexts for everything so when people give you
instructions that work on a plain jane asterisk box it won't work.

Trixbox does not have a mailing list. The forums suck. There is no
real support from anybody. Everybody is asking questions and maybe
somebody will answer your question maybe they wont.

People who use trixbox are not writing AGI scripts by and large so I
don't think you will get any help in that regard at all.

I am using trixbox and I am starting to feel like I would have been
better off with just a plain asterisk box for my agi work.
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Re: [asterisk-users] Re: unable to get channel lock BAD BAD BAD

[Tim Uckun]

Yes, I get same error message in my log. Anybody has any info on this one?



Are you using trixbox? It would be nice to try and isolate this
problem by ruling out a bad config in trixbox.
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[asterisk-users] Monitor Zap Status

[Ken Williams]

I've installed Grandstream GPX-2000 phones and have successfully enabled
one of my buttons to use Asterisk BLF for an extension.  I can tell when
this extension is available, is being rung, or is on the line.
 
I'd like to do the same for my Zaptel channels, to be able to see when a
line is onhook, ringing or offhook.
 
I tried the following but alas, it doesn't seem to be working:
 
exten = 102,hint,ZAP/2

I based that on:
 
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Re: [asterisk-users] PortSip and Astericks new install

[Vicky]

try :
[John]
type=friend
secret=test
host=dynamic
disallow=all
allow =gsmilbculawalaw
Also try other sip phone slike sjphone just to make sure there is no prob .
On 16/11/06, Charlie Grosvenor [EMAIL PROTECTED] wrote:


I have just installed Asterisk and installed the  sample configuration
files. Asterisks appears to be working and I have added a SIP client:



[John]

type=friend

secret=test

host=dynamic

allow=all



I have been trying to dial the demo number 500 when using PortSip,
Asterisks answers the phone but PortSip gives me the error:



Call failed: codec not accepted 488.



I have tried changing the enabled codecs in PortSip but this makes no
difference. I have also tried various other SoftPhone but none of them seem
to work.



Anybody know what I have missed / doing wrong?



Thanks

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[asterisk-users] Monitor Zap Status - Full E-mail...

[Ken Williams]

I've installed Grandstream GPX-2000 phones and have successfully enabled
one of my buttons to use Asterisk BLF for an extension.  I can tell when
this extension is available, is being rung, or is on the line.
 
I'd like to do the same for my Zaptel channels, to be able to see when a
line is onhook, ringing or offhook.
 
I tried the following but alas, it doesn't seem to be working:
 
exten = 102,hint,ZAP/2

 
I based that on:
 
exten = 732,hint,SIP/732

which does work for the SIP phones.
 
If I do show hints in the CLI, I get
 
   102 : ZAP/2 State:Idle
Watchers  0
   732 : ZAP/1 State:Idle
Watchers  0
 
When a call is made on the ZAP/2 line the State changes to InUse, so I
know it's working on that side.
 
Any thoughts or suggestions as to how I can monitor a ZAP line on my
GPX-2000?  The problem is we have 6 lines, so my plan was to use the 7
buttons down the side, the top 6 for lines and the 7th for paging.
 
Thanks for the help,
Ken
 
 
 
*Sorry about duplicate e-mail, accidentally hit ENTER when holding CTRL
instead of V to paste
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Re: [asterisk-users] PortSip and Astericks new install

[Doug Lytle]

Charlie Grosvenor wrote:


 


[John]

type=friend

secret=test

host=dynamic

allow=all


Try:

disallow=all
allow=alaw
allow=ulaw
allow=gsm

Doug

--

Ben Franklin quote:

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

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RE: [asterisk-users] trixbox + agi

[Michael Collins]

 Trixbox scatters it's config files. Some stuff is kept in the
 database, some in the conf files.
 You have to keep your configuration in specific files that won't be
 overrritten.

True - TB does a lot of very specific stuff.  If you want to have a
plain Jane dial plan for your stuff then use the file
extensions_custom.conf or create a whole new
extensions_whatever.conf file, then #include it in
extensions_custom.conf.


 Trixbox has it's own contexts for everything so when people give you
 instructions that work on a plain jane asterisk box it won't work.
 

While it can be more complicated to integrate dialplan suggestions from
others, this statement is not entirely true.  Your best bet when using
TB is to study their dialplan.  If you can understand *why* they have
what they have in there then you will be able to add your own stuff
without any hitches.  Just be sure that you know what is there already.
The TB stock dialplan is not small, so you should plan on a few hours of
studying it before you feel totally comfortable with it.  (Be sure to
have a reference handy so that you can look up commands.  I bought the
TFOT book whose reference I found invaluable when studying dialplans
from TB and others.)


 Trixbox does not have a mailing list. The forums suck. There is no
 real support from anybody. Everybody is asking questions and maybe
 somebody will answer your question maybe they wont.
 

Again, not entirely true.  The forums aren't the best I've ever seen,
but I've never had a post go unanswered.  In some cases, I've had
answers within an hour.  It depends entirely upon the depth of the
question.  Like all forums, the easier the question, the more likely to
have a greater number of answers and to have them more quickly.  Deeper
questions tend to require more effort and the pool of available brains
to think about them is smaller, so it usually takes more time.


 People who use trixbox are not writing AGI scripts by and large so I
 don't think you will get any help in that regard at all.

There might be some truth to this.  Those who choose TB do so for
certain reasons.  That being said, I've done a number of AGI operations
with TB and the Asterisk Perl module.  Again, it all depends...


 
 I am using trixbox and I am starting to feel like I would have been
 better off with just a plain asterisk box for my agi work.

Quite possibly the case.  If you have the resources to do a plain *
install next to your TB install then that would be ideal because you
could get your feet wet in doing a plain * install and you could also
use the TB install for reference.  You may find that a plain install
suits your tastes and situation just fine.


-MC, plain Jane and TB user
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[asterisk-users] Set port to which Asterisk should send its answer

[Andre Kirchner]

Hi,

I'm sending the following message from port X to port 5060 of another box 
running Asterisk, and it is answering back to port X from port 5060. Shouldn't 
Asterisk use the Via header to find out where to answer, and in this case send 
its answer to port 4000?

OPTIONS sip:192.168.0.103 SIP/2.0\r\n
Via: SIP/2.0/UDP 192.168.1.130:4000;branch=0.0\r\n
CSeq: 4711 OPTIONS\r\n\r\n

Thanks,

Andre

 
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[asterisk-users] web interface to control zap interface

[Jordan Novak]

I am looking for a web interface to control my zap agents. Allowing them
to do conferences and transfers. I am familiar with flash operator panel
but am unsure of how I would set it up to allow the agent, caller, to
dial another number and have a three way conference. I have setup
features.conf to do a attended transfer but can't figure out how to make
it three-way.
 
Jordan Novak
Senior Telecommunications Engineer
Logistics Health Inc.
 
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RE: [asterisk-users] Monitor Zap Status - Full E-mail...

[Ken Williams]

Upon further investigation I must be doing something wrong.
 
It was my understanding that a hint extension could be anything, it
wasn't the same as a real extension, though you could make it the same
to make it easier.
 
That being said exten = 702,hint,SIP/702 works, while exten =
102,hint,SIP/702  doesn't.
 
I've got a GXP-2000 with the first button set to AsteriskBLF username
102 and the second button set to AsteriskBLF username 702, only the
second button actively monitors 702.
 
I've read, reread, rereread and so on a ot of examples of hint files and
I can't figure out why the GXP-2000 doesn't like them.  When I do show
hints in CLI it is registering both 102  702 properly.
 



From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
Williams
Sent: Wednesday, November 15, 2006 1:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Monitor Zap Status - Full E-mail...


I've installed Grandstream GPX-2000 phones and have successfully enabled
one of my buttons to use Asterisk BLF for an extension.  I can tell when
this extension is available, is being rung, or is on the line.
 
I'd like to do the same for my Zaptel channels, to be able to see when a
line is onhook, ringing or offhook.
 
I tried the following but alas, it doesn't seem to be working:
 
exten = 102,hint,ZAP/2

 
I based that on:
 
exten = 732,hint,SIP/732

which does work for the SIP phones.
 
If I do show hints in the CLI, I get
 
   102 : ZAP/2 State:Idle
Watchers  0
   732 : ZAP/1 State:Idle
Watchers  0
 
When a call is made on the ZAP/2 line the State changes to InUse, so I
know it's working on that side.
 
Any thoughts or suggestions as to how I can monitor a ZAP line on my
GPX-2000?  The problem is we have 6 lines, so my plan was to use the 7
buttons down the side, the top 6 for lines and the 7th for paging.
 
Thanks for the help,
Ken
 
 
 
*Sorry about duplicate e-mail, accidentally hit ENTER when holding CTRL
instead of V to paste
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Re: [asterisk-users] Sending '#' with Dial

[Mark Hulber]

Have you tried setting the CALLERID variables?  If the provider is 
ignoring those then I guess they are asking you to set per call 
blocking?  I don't know how to do that.


exten = s,1,Set(CALLERID(number)=3025551212|a)
exten = s,n,Set(CALLERID(name)=Joe Smith|a)

MARK.

Emil Thelin wrote:

Hi!

I have a working asterisk-setup with four sip-clients. Everything 
works great but when the users call someone the phonenumber shows up 
on the receiving ends callerid-display.


To correct this my provider told me to send #31# before the 
phonenumber, tried this with: Dial(SIP/[EMAIL PROTECTED]) but my 
asterisk tells me that it isn't a valid extension.


The INVITE looks fine, '#31#phonenumber@provider' but my provider 
then sends SIP/2.0 404 Not Found back to me.


Any thoughts?

/e

--
http://hostname.nu/~emil
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Re: [asterisk-users] Set port to which Asterisk should send its answer

[Eric \ManxPower\ Wieling]

nat=yes might cause this, since with NAT we cannot trust the IP or the 
port that is in the data part of the packet.


Andre Kirchner wrote:

I'm sending the following message from port X to port 5060 of another box 
running Asterisk, and it is answering back to port X from port 5060. Shouldn't 
Asterisk use the Via header to find out where to answer, and in this case send 
its answer to port 4000?

OPTIONS sip:192.168.0.103 SIP/2.0\r\n
Via: SIP/2.0/UDP 192.168.1.130:4000;branch=0.0\r\n
CSeq: 4711 OPTIONS\r\n\r\n

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[asterisk-users] Re: safe_asterisks pawning multiple asterisk process???

[Steve Murphy]

On Wed, 2006-11-15 at 13:34 -0700, Andre
Courchesne[EMAIL PROTECTED] wrote:

I remember seeing a ton of asterisk lines in ps if you had just the
exact right (wrong?)
declare in safe_asterisk. I had it myself and erased the line a while
back. I can't see it in the svn repository at all. It was KERNEL= or
something like that. Sorry I can't remember.
If you have such a line in your safe_asterisk, remove it. It causes a
line in ps for
each thread running in asterisk. It's not harmful-- just irritating.

murf

 We have 1 server that after a few hours operating has multiple
 process 
 of asterisk running. Here is the pstree output:
 

 
 And ps aux | grep asterisk:
 
 # ps aux | grep asterisk
 asterisk  2047  0.0  0.1   9200  1516 ?SNov14
 0:00 
 /usr/sbin/httpd
 asterisk  2084  0.0  0.2  11544  2388 ?SNov14
 0:00 
 /usr/sbin/httpd
 asterisk  2085  0.0  0.2  11544  2384 ?SNov14
 0:00 
 /usr/sbin/httpd
 root  2196  0.0  0.0   2172   456 ?SNov14
 0:00 /bin/sh 
 /usr/sbin/safe_asterisk -U asterisk -G asterisk
 asterisk  2215  1.6 10.0 122496 90984 ?Sl   Nov14
 38:07 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk  2250  0.0  0.0   2176   376 ?SNov14
 0:00 -bash 
 -c cd /var/www/AMP/panel  /var/www/AMP/panel/safe_opserver 
 asterisk  2251  0.0  0.0   2128   868 ?SNov14
 0:00 
 /bin/bash /var/www/AMP/panel/safe_opserver
 asterisk  2253  3.2  0.8   8988  7336 ?RNov14
 73:54 
 /usr/bin/perl -w ./op_server.pl
 asterisk 12105  0.0  0.8  31440  7804 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 13080  0.0  0.8  32096  7616 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 18352  0.0  0.9  36080  8684 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 19352  0.0  0.9  36528  8764 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 25402  0.0  0.9  39196  8972 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 26497  0.0  1.0  40448  9372 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 30901  0.0  1.0  42064  9308 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk  3160  0.0  0.6  43968  5624 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 12444  0.0  0.5  49636  5148 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 22538  0.0  0.5  54532  5148 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 26032  0.0  0.5  56948  5148 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 27517  0.0  0.5  57056  5148 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 31806  0.0  1.0  58956  9800 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk  3655  0.0  1.0  60088  9932 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk  4636  0.0  1.1  60956 10316 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk  7494  0.0  1.2  62200 10952 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk  9276  0.0  1.3  64856 12040 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 11592  0.0  1.4  65404 12720 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 12757  0.0  1.4  66808 13504 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 13167  0.0  1.4  66576 13296 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 13363  0.0  1.4  65936 13156 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 16251  0.0  1.6  68812 14664 ?SNov14
 0:00 
 /usr/sbin/asterisk -U asterisk -G asterisk -vvvg -c
 asterisk 16942  0.0  1.6  68600 14676 ?S

Re: [asterisk-users] Caller ID in Sweden not working and looking for and voices

[Mattias Andersson]

Hi!
The problem is that I have commercial Asterisk baste switch that it works wit.
My trixbox do not. I guess it has to do with the 
setting of system for Caller ID.

//Mattias

I do not now way, but my posting are not coming trow. Or are the?
//Mattias
On 15/11/06, Mattias Andersson mailto:[EMAIL PROTECTED][EMAIL PROTECTED] 
wrote:
Hi!
The problem is that I have commercial Asterisk baste switch that it works wit.
My trixbox do not. I guess it has to do with the 
setting of system for Caller ID.

//Mattias


On 15/11/06, Anselm Martin Hoffmeister 
mailto:[EMAIL PROTECTED][EMAIL PROTECTED]  wrote:

Am Mittwoch, den 15.11.2006, 01:06 +0100 schrieb Mattias Andersson:

 Hi!
 I am getting inbound caller ID fine bout not out.
 I am in Sweden and suing Rixtelcom /POrt80 as provider.
 anyone knowing what is wrong?

Assuming that is a SIP provider, it is not your job to set the callerid
but the provider's - their interfacing to the regular landline network
is responsible. There are providers that never send callerid, some send
always (united, gmx in Germany) and some allow the user to set in his
preferences wether he wants to send callerid (http://sipgate.de sipgate.de).

BR
Anselm

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--
Mattias Andersson

Storskiftesvägen 6
145 60 Norsborg

m. +46-70-799 44 41
h. +46-8-641 38 97

Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED]




--
Mattias Andersson

Storskiftesvägen 6
145 60 Norsborg

m. +46-70-799 44 41
h. +46-8-641 38 97

Email: mailto:[EMAIL PROTECTED][EMAIL PROTECTED]


Adress:
Mattias Andersson
Storskiftesvägen 6
S-145 60 Norsborg

Mobil: +46-70-799 44 41
Email: [EMAIL PROTECTED]
Skype: eskes1 



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RE: [asterisk-users] Page() Function Timeout

[David Gagnon]

Which version are you using? There was a problem in 1.2.12.1 with the page
application. Update to 1.2.13.

David

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Steven
Ringwald
Envoyé : 15 novembre 2006 13:45
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [asterisk-users] Page() Function Timeout

Ken Williams wrote:
 I'm trying to use a simple page function.  It starts a MeetMe 
 conference with the devices I've listed, but the devices hang up after 
 3-5 seconds.  After doing some research I found this was a problem, 
 and I needed to remove a (5) from app_page.c
  
 Well, my app_page.c didn't have the (5).  I did make clean; make 
 install again just in case I had some weird compiled version installed 
 that had the (5) in it.  After compiling I restarted the asterisk 
 service and tried paging again and still had the same problem.
  
 In the CLI I get the following, which you can see the (5) is still in 
 there somehow. 
  
 -- Playing 'beep' (language 'en')
 -- Launching MeetMe(1010553064d|mqxdw(5)) on SIP/710-09a50038
 -- Created MeetMe conference 1023 for conference '1010553064d'
 -- Launching MeetMe(1010553064d|mqxdw(5)) on SIP/717-09a48758
 I've grep'd the entire src folder for \(5\) as well as qxd trying to 
 find all instances of this, and the only ones are listed in the 
 app_page.c file.  Any suggestions on where to get this rogue (5) out 
 of here?
  
 snprintf(meetmeopts, sizeof(meetmeopts), %ud|%sqxdw, confid, 
 ast_test_flag(flags, PAGE_DUPLEX) ?  : m);
  
 and
  
 if (!res) {
 snprintf(meetmeopts, sizeof(meetmeopts), %ud|A%sqxd, 
 confid, $
 pbx_exec(chan, app, meetmeopts, 1);
 }
 are the only sections of the app_page.c that have the meetme call in it.
  
 My page functions, fwiw, both have the same problem:
  
 ;Paging
  
 exten = 760,1,SIPAddHeader(Call-Info: answer-after=0)
 exten = 760,2,Page(SIP/717SIP/710SIP/702|d)
 exten = 760,3,Hangup
  
 exten = 761,1,SIPAddHeader(Call-Info: answer-after=0)
 exten = 761,2,Page(SIP/717SIP/710SIP/702)
 exten = 761,3,Hangup
 Any suggestions would be very helpful.

I had the same problem and ended up changing the 5 to a 300. If you 
don't specify a (N) after the 'w', I believe it defaults to 5.

Steve


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RE: [asterisk-users] Set port to which Asterisk should send its answer

[Ron McLeod]

I am new to the Asterisk code, but it looks it the response to OPTIONS is
always sent to the IP address and UDP port that the request was received
from.  Also, it looks like Asterisk doesn't deal well the VIA header anyway.
In chan_sip.c it looks like to gives-up if the VIA contains SIP/2.0/UDP:

 

if (strcasecmp(via, SIP/2.0/UDP)) {

ast_log(LOG_WARNING, Don't know how to respond via
'%s'\n, via);

return -1;

}

 

Check you log and see if the warning message is there.

 

Ron

 

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre Kirchner
Sent: Wednesday, November 15, 2006 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set port to which Asterisk should send its answer

 

Hi,

I'm sending the following message from port X to port 5060 of another box
running Asterisk, and it is answering back to port X from port 5060.
Shouldn't Asterisk use the Via header to find out where to answer, and in
this case send its answer to port 4000?

OPTIONS sip:192.168.0.103 SIP/2.0\r\n
Via: SIP/2.0/UDP 192.168.1.130:4000;branch=0.0\r\n
CSeq: 4711 OPTIONS\r\n\r\n

Thanks,

Andre

  

  _  

Everyone is raving about the
http://us.rd.yahoo.com/evt=42297/*http:/advision.webevents.yahoo.com/mailbe
ta  all-new Yahoo! Mail beta.

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Re: [asterisk-users] Dialing from Placed Calls on PolycomIP501doesn't always work

[Anthony Rodgers]

Thanks, Noah - we'll try 1.6.7 and see if the problem goes away.

CP

On 15-Nov-06, at 11:55 AM, Noah Miller wrote:

   Has anyone noticed that attempting to place a call from the  
Placed

   Calls list on a Polycom IP501 by pressing the 'Dial' softkey
  sometimes
   simply returns the phone to the idle screen?
 
  Yes, I've seen it. We're running 1.6.6, what firmware version  
do you

  have?
 
 We're running SIP 1.6.6.0036 on the 3.1.3.0131 BootROM.

 Did you come up with any reason/fix for this?

I never ran 1.6.6 for any length of time.  1.6.7 and 2.0.1 don't seem
to suffer this issue.  2.0.1 has some buddy watch problems, so you may
not want to use it, but 1.6.7 should be OK.

- Noah
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Re: [asterisk-users] Re: Linksys doesn't resync properly and doesn't get provisioning from TFTP and HTTP

[Tzafrir Cohen]

On Mon, Nov 13, 2006 at 09:50:08PM -0500, Zeeshan Zakaria wrote:
 On checking tail -f /var/log/atftpd, I can see that on reboots, other phones
 get served by the TFTP, but not the linksys ones. Now I don't understand how
 was it was updating itself and was being provisioned resyncing for so many
 hours initially, and why TFTP has stopped serving it now?

Do you see any request at all? Any chance it is asking the wrong host?
Try tcpdump or any other network sniffer.

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] Set port to which Asterisk should send its answer

[Ron McLeod]

Whoops, sorry - it only handles SIP/2.0/UDP; which is what is expected,
but it seems like it only checks for the VIA header for REGISTER, INVITE,
CANCEL, BYE, and SUBSCRIBE requests.

 

 

Ron

 

 

  _  

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andre Kirchner
Sent: Wednesday, November 15, 2006 1:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Set port to which Asterisk should send its answer

 

Hi,

I'm sending the following message from port X to port 5060 of another box
running Asterisk, and it is answering back to port X from port 5060.
Shouldn't Asterisk use the Via header to find out where to answer, and in
this case send its answer to port 4000?

OPTIONS sip:192.168.0.103 SIP/2.0\r\n
Via: SIP/2.0/UDP 192.168.1.130:4000;branch=0.0\r\n
CSeq: 4711 OPTIONS\r\n\r\n

Thanks,

Andre

  

  _  

Everyone is raving about the
http://us.rd.yahoo.com/evt=42297/*http:/advision.webevents.yahoo.com/mailbe
ta  all-new Yahoo! Mail beta.

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[asterisk-users] Auth Issue using Asterisk as Voicemail AND as Normal SIP Extension.

[Jean-Marc Salsa]

Hi,

My case is a little bit complicated.
I would like to use my Asterisk Box for 2 different services/providers :
- Voicemail server for one
- SIP Registrar and Proxy for some other extensions

The problem is that Voicemail service is for another provider which has
defined Extension like ABC ...
We are connected to them through a SIP Trunk.
Everything works fine 
Except IF ABC is also defined in the sip.conf as one of the other Extensions
of the second virtual provider ...
Thus, Asterisk doesn't ask anymore password for the SIP Trunk, but for the
SIP Extension . Which is WHAT I DO NOT WANT !
because as you might guess, the call to Voicemail will be rejected !

I have been thinking of implementing SIP Multi-Domain,
Would it be the way ?
Very hard to find good doc on the subject ... could someone point it out
please ?

How should I implement it ? what domain should I put in Asterisk General SIP
config ?
Should I use for each of the defined extension a setting specifying from
which domain they belong ?


Thanks a lot for your help !

JM
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[asterisk-users] Grandstream Programmable Buttons Retrieving On Hold Lines

[Ken Williams]

A day of banging my head against a wall and spamming this list is about
done
 
I've got everything working beautifully, and I'm ready to go full out
and implement all across the board...save for one stupid little thing.
 
We have 6 phone lines and I'd like the GXP-2000 to show the status of
these 6 phone lines down, as well as allow a user to pick a specific
line, the right hand side.  I can get it show the status, but when you
push that button it just rings the system as if you're an outside
caller.  I've gone in loops on the internet  in the dialplan, tried a
ton of different things, but this simple trick I cannot get to perform.
Do I need to create a new extension that ties directly to line 1 and
monitor it?  If so, how?
 
I'm using TDM400's in the server.  I'd like to have button 1 on the
right hand side tied directly to ZAP/1, button 2 to ZAP/2 and so on.
Please tell me this is possible and give some tips or recommendations to
me.  Happy to paste parts of configuration files, but don't want to spam
this poor list any more than I have, so I'll wait til they're requested.
 
Thanks,
Ken
 
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