[asterisk-users] Re: 1.4 branch on OSX?

[Martin Joseph]

Good news!

I did an SVN update to my 1.4 branch again today, and 1.4-r46154  seems 
to have resolved the asterisk hogging the whole CPU issue.


I still can't use the regular console though (asterisk -r) as that is 
unresponsive.


Using asterisk -c to start it , works and gives me a color CLI too.

At least now it's working well enough to test a bit for real...

Awesome to see all those changes and fixes flowing in.  This project is 
really pretty incredible.


Thanks to all who contribute and make this possible!

Marty



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[asterisk-users] sip.conf - srvlookup

[Tomislav Parčina]

I would like to put srvlookup=no in my SIP conf, so that I don't get DNS issues 
(Asterisk stops responding). I use VoIP Buster and in sip.conf I use 
sip1.voipbuster.com. When I do sip show peers in CLI I get
voipbuster/tomo 194.221.62.207  5060 OK (27 ms)
And when I ping sip1.voipbuster.com
[EMAIL PROTECTED] ~]# ping sip1.voipbuster.com
PING sip1.voipbuster.com (194.221.62.206) 56(84) bytes of data.

So, Asterisk is registered at 194.221.62.207 and DNS lookup gives me 
194.221.62.206 IP address.

Question is, which IP address should I use, 206 or 207?



--
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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Re: [Asterisk-Users] rxfax problem

[Klaus Darilion]

Steve Underwood wrote:
If someone wants to take my code and make it work with Asterisk under 
GPL conditions, that's fine. The GPL gives you that right. Please make 
sure you stick to GPL conditions, though. You can't use G.729, for 
example, in an Asterisk that's using spandsp.


I do not see any problems here - Asterisk does not link against g729. 
Further, even it would be the case, it would be no problem if you do not 
distribute it. And AFAIK even about distribution there are different 
meanings (e.g. GPL applications which link against openssl).


Or do I miss an important point?

regards
klaus

--
Klaus Darilion
nic.at

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Re: [asterisk-users] Junghanns quadBRI and mISDN

[Olivier]

2006/10/24, Alberto Pastore [EMAIL PROTECTED]: (there are a couple of serious issues using bristuff
and we've been looking for alternate drivers).Hi,Which issues do you have ?Regards
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[asterisk-users] Re: rxfax problem

[Martin Joseph]

On 2006-10-24 17:25:37 -0700, Steve Underwood [EMAIL PROTECTED] said:

The development of Asterisk has now degraded to the point where I will 
no longer contribute anything to it.


I am not interested in a flame war, but would love to here a more 
explicit explanation for what is occurring within asterisk development 
community, that makes you say this?


Thanks for all your efforts.
Marty


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RE: [asterisk-users] asterisk and HMP

[Gregory Duchatelet]

 I think this is now the Eicon HMP platform. It looks like Eicon bought
 this when the fools paid good money for Dialogic.
 
 Its amazing how many companies have got on the HMP bandwagon since we
 started the Zapata work in 1999. If you do a Google search you can find
 something like 10 companies promoting HMP type products. Few look like
 coherent products, though.

I ask this question but i twas not the good question. Here a schema :
Public telephon network - PABX - computer with a soft call
center and dialogic cards. 
I want to connect this computer to an Asterisk, via a SIP trunk, so I start
with a HMP driver on the dialogic cards...

I'm not sure I'm clear...

Greg

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[asterisk-users] Re: IAX2 goes one way audio when lag gets bad

[Martin Joseph]

On 2006-10-24 13:04:02 -0700, Matt [EMAIL PROTECTED] said:


Hi,
I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.

The customer is connected via IAX2 to our softswitch.

On the customer's end I have the following config in iax.conf:
[general]
bindport = 4569   ; Port to bind to (IAX is 4569)
bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
disallow=all
allow=ulaw
notransfer=yes
trunk=no
(I have also tried trunk=yes and nothing for trunk=)
jitterbuffer=yes
forcejitterbuffer=yes
mailboxdetail=yes
dropcount=3
minexcessbuffer=80
jittershrinkrate=1

I have tried with jitterbuffer=no, and then rather then one-way-audio
I get high packet loss until the connection settles back down.Any
ideas on other things I can try?
Implement QoS that prevents the upstream bandwidth from the customers 
site from being completely hammered...


Just a thought,
Marty


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[asterisk-users] Re: Dynamic Codec Selection

[Martin Joseph]

On 2006-10-24 06:44:01 -0700, Wildheart 
[EMAIL PROTECTED] said:



Hi,

Does anyone know a what to use a different codec for calls which are
handset to handset (eg, G711) then when we have calls to the out side
world (via an asterisk server) to use a different codec(eg, G729)?

The idea is to reduce the bandwidth to the server for the majority of
calls, but get good quality on internal calls.

With thanks,


yes, this is simple,  just make it so the extensions allow both g729 
and ulaw, and set your outside world is g729.


Marty



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[asterisk-users] Re: Meetme... No channel type registered for 'zap'

[Tony Mountifield]

In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
 Kristian,
  
 I don't have any zap hardware What do I put in zaptel.conf if I don't 
 have any hardware?
 On some other systems we have, with chan_zap not loaded, and no zaptel.conf 
 (running
 1.2.9.1), meetme runs fine. This system with the problem has 1.2.12.1. I 
 wonder if something
 was changed?

Doug, it sounds to me like you don't have the /dev/zap device files.

Do you have the file /etc/udev/permissions.d/zaptel.permissions and
/etc/udev/rules.d/zaptel.rules installed?

What Linux distro are you using?

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org
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[asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI

[Alex]

Hi all!

We've released VoiceOne 0.4.0, a web-based and open source solution 
which allows to fully manage an Asterisk service hosted on a LAMP server.


We focused on an charming and overall user-friendly interface. Thanks to 
the authentication based on roles, once configured by a super user, the 
PBX may be easily maintained even by an Asterisk unskilled users.


From a technical point of view, the application is made up of two 
modules: one for the client - i.e. the user interface - and the other 
for the server. Thanks to the web services provided by the server module 
and the use of a database, VoiceOne may be easily integrated with other 
applications (e.g. CRM software).


The project has grown and has received positive response so far. 
Nowadays there's a little but enthusiastic community of developers, 
supporters and users. Translations in several languages (e.g. English, 
Spanish, Russian, etc.) are already available.


On the project website at http://www.voiceone.it you'll find the online 
demo and the links to download the source files from Sourceforge, as 
well as a support forum.


We would be pleased if you could give it a try and let us know your 
feedback, comments, ideas, or suggestions replying here or posting a 
message on our forum.


Thanks for your kind attention.

Regards,
Alex
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RE: [asterisk-users] UA - number assignment

[Paul Ianas]

Thank you.

Indeed, this is what I want to know. When somebody wants to make a call
(using a standard telephone, connected to a media gateway), he doesn't
know what user is in my Asterisk conf. He only knows that he wants to
call John, who has the number 102 for example. He dials 102 from his
phone and the call is routed to the corresponding user (this is
Asterisk's job).

When using a softphone, you can call user John, without knowing what
number he has (and that's ok). But when using a standard (physical)
phone, one doesn't need to know what user is in the database (even if he
would know, the telephone must dial a number), he only needs to know
what telephone number does user John have. John is an example. In
Asterisk I could set the username to -let's say - xxx_John_yz4230 or a
MAC address, etc.

I want to know only the way I can make an UA-number assignment.

I will try your solution and I will give you feedback.


Thank you!

--
Paul Ianas
Programming Engineer
Level 7 Software
Timisoara, 59D Bucovinei
phone: 0744137020
email: [EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marco
Mouta
Sent: Tuesday, October 24, 2006 5:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] UA - number assignment

I think I understood what you want:
1- You want when someone dials an extension, do a Lookup in a database
using FWDCIDNAME
2- Then Dial the number that corresponds to this FWDCIDNAME in database

is that?

If it is so, i would recomend you to use AstDB - Asterisk Berkeley DB
(version1) - automatically installed with your asterisk.

Example:

exten=_X.,1,Set(NumberToDial=DB(myuserlist/${FWDCIDNAME})
exten= _X.,2,Dial(SIP/${NumberToDial})
exten= _X.,3,hangup

Take a look on this function and applications on your CLI show function
DB

hope it helps.

Pls give me some feedback



On 10/24/06, Paul Ianas [EMAIL PROTECTED] wrote:




 My problem is simple and I've issued it about 3 weeks ago. I want the
UAs to
 authenticate with a number to the SIP server. Is this possible?



 For example, I configured an AT-RG613TX (Allied Telesyn Residential
 Gateway). In its configuration it is not possible for me to skip
specifying
 a number (ex. 102) along with the username. I've looked into the
source code
 (SIP implementation) of Asterisk and, as I figured out, it is not
possible
 to tell Asterisk the number the user has.



 The question is: how can I assign a number to a user in Asterisk? One
 solution would be to define two rules in extensions.conf :



 exten = 102,1,SetCallerId,${FWDCIDNAME}

 exten = 102,2,Dial(SIP/pianas)



 these would tell Asterisk that user pianas has the number 102.



 Is there any other solution for my problem? (a database for example).



 Thank you.



 --

 Paul Ianas

 Programming Engineer

 Level 7 Software

 Timisoara, 59D Bucovinei

 phone: 0744137020

 email: [EMAIL PROTECTED]


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[asterisk-users] Call is not coming through sipgate.co.uk+Asterisk

[Crazy Boy]

Hi,I have installed  Asterisk, Zaptel, Libpri, Addons, Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK phone number i.e., 0207100. I configured my Asterisk server with 0207100. When I made a call to this number from outside phone, my XLite extension is not ringing. Its directly going to Voicemail or telling that "person is unavailable". When I made a call, Asterisk console is also not showing anything. But, sipgate website is showing my calls list. I thought that When I made a call from outside to my number, call is going to sipgate.co.uk and its not routing to my server. When I execute "sip show registry", its not displaying anything. Here I am giving my configuration details:My sip.conf file contents:[general]port = 5060bindaddr =
 0.0.0.0qualify=nodisable=allallow=alawallow=alawallow=ulawallow=g729allow=gsmallow=slinearsrvlookup=yes[250]type=friendusername=250secret=dannycallerid="Danny"host=dynamiccontext=demoregister = 100:[EMAIL PROTECTED]/100[sipgate4]type=frienddisallow=allallow=alawallow=ulawfromuser=100authuser=100secret=passwordusername=100host=sipgate.co.ukcontext=demodtmfmode=infofromdomain=sipgate.co.ukinsecure=verynat=yescanreinvite=nocallerid="Danny" lt;0207100My Extensions.conf file contents:[demo]exten = 250,1,Dial(SIP/250,20)exten = 250,2,Voicemail(u250)exten = 250,3,Voicemail(b250)exten = 250,4,Hangupexten = _0207.,1,SetCallerID(""
 lt;100|a) ;Outgoingexten = _0207.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],40,tr)exten = 100,1,Dial(SIP/250,30,tr) ;IncomingAm I have to install any other libraries?Anything wrong in the above configuration?Looking forward to your response. Thanks in advance.Regards,Chandra. 
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Re: [asterisk-users] zaptel 1.2.10 make problem

[Jan Marek]

Hello,

you have to install package with kernel sources or at least with
kernel headers to compile zaptel sources...

Sincerely
Jan Marek
On Sat, Oct 21, 2006 at 08:18:32PM +0530, ram wrote:
 Hi
 
 iam installing zaptel 1.2.10 on my FC5
 
 when i make iam getting following error
 
 any one suggest me whats wrong, i have installed source also in the same
 server.
 
 
 grep: /lib/modules/2.6.15-1.2054_FC5/build/include/linux/autoconf.h: No such
 file or directory
 ZAPTELVERSION=1.2.10 build_tools/make_version_h  version.h.tmp
 if cmp -s version.h.tmp version.h ; then echo; else \
mv version.h.tmp version.h ; \
 fi
 
 rm -f version.h.tmp
 /lib/modules/2.6.15-1.2054_FC5/build
 make -C /lib/modules/2.6.15-1.2054_FC5/build SUBDIRS=/root/vici/zaptel-
 1.2.10 modules
 make[1]: Entering directory `/usr/src/kernels/2.6.15-1.2054_FC5-x86_64'
 Makefile:486: .config: No such file or directory
 
  WARNING: Symbol version dump /usr/src/kernels/2.6.15-
 1.2054_FC5-x86_64/Module.symvers
   is missing; modules will have no dependencies and modversions.
 
 grep: /lib/modules/2.6.15-1.2054_FC5/build/include/linux/autoconf.h: No such
 file or directory
  CC [M]  /root/vici/zaptel-1.2.10/zaptel.o
 cc1: error: include/linux/autoconf.h: No such file or directory
 In file included from /root/vici/zaptel-1.2.10/zconfig.h:9,
 from /root/vici/zaptel-1.2.10/zaptel.c:40:
 include/linux/config.h:6:28: error: ./linux/autoconf.h: Too many levels of
 symbolic links
 In file included from /root/vici/zaptel-1.2.10/zaptel.c:40:
 /root/vici/zaptel-1.2.10/zconfig.h:10:27: error: ./linux/version.h: Too many
 levels of symbolic links
 /root/vici/zaptel-1.2.10/zconfig.h:72:5: warning: LINUX_VERSION_CODE is
 not defined
 /root/vici/zaptel-1.2.10/zconfig.h:72:27: warning: KERNEL_VERSION is not
 defined
 /root/vici/zaptel-1.2.10/zconfig.h:72:41: error: missing binary operator
 before token (
 In file included from include/linux/kernel.h:11,
 from /root/vici/zaptel-1.2.10/zaptel.c:43:
 include/linux/linkage.h:5:25: error: asm/linkage.h: No such file or
 directory
 In file included from include/linux/types.h:13,
 from include/linux/kernel.h:13,
 from /root/vici/zaptel-1.2.10/zaptel.c:43:
 include/linux/posix_types.h:47:29: error: asm/posix_types.h: No such file or
 directory
 In file included from include/linux/kernel.h:13,
 from /root/vici/zaptel-1.2.10/zaptel.c:43:
 include/linux/types.h:14:23: error: asm/types.h: No such file or directory
 In file included from include/linux/kernel.h:13,
 from /root/vici/zaptel-1.2.10/zaptel.c:43:

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-- 
Ing. Jan Marek   | Nez mi poslete prilohu .doc, .xls 
University of South Bohemia  | nebo .ppt, prectete si, prosim,
Academic Computer Centre | WWW stranku uvedenou na poslednim
Phone: +420-38-9032080   | radku signatury...
http://www.gnu.org/philosophy/no-word-attachments.cs.html
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[asterisk-users] Choice of soundfile format

[Jon Schøpzinsky]

Hello

What soundfile format, is the one that uses least transcoding during playback?
As I can see, I can choose wav or gsm. What sucks least cpu power, during 
playback to example a Zap channel? I would guess wav, but is this correct?

 
Kind Regards
Jon Leren Schøpzinsky

-- 
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Re: [asterisk-users] Choice of soundfile format

[Conrad Wood]

On Wed, 2006-10-25 at 11:24 +0200, Jon Schøpzinsky wrote:
 Hello
 
 What soundfile format, is the one that uses least transcoding during playback?
 As I can see, I can choose wav or gsm. What sucks least cpu power, during 
 playback to example a Zap channel? I would guess wav, but is this correct?

The one that is encoded in the same codec as the codec of the channel.
On zap it's often alaw or ulaw so you can encode your files like that.
You can encode the same file with different codecs and save it with
different extensions (matching the codec) and asterisk will pick the
most suitable one.
If the channel is gsm, a gsm encoded file would be most efficient, as it
doesn't need transcoding at all.

Conrad

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[asterisk-users] PBAX-Group with QuadBRI card, outgoing call problem

[Giray Devlet]

Hi All ...

I'm running Asterisk 1.2.13-BRIstuffed-0.3.0-PRE-1v  which has a
Junghanns QuadBri card in it (lspci reports  Cologne Chip Designs GmbH
ISDN network Controller [HFC-4S] (rev 01))

I have a regular KPN ISDN2 Line connected which works fine with the
the zapata.conf
below.

However, I also have a new KPN ISDN Mutli (meervoudig) line (PBAX-Group),
which works fine for incoming calls, however, is not able to successfully make
an outgoing call. I get a 'busy/congested'.

When I do a bri debug I have the following, any help on what I can do to setup
this is welcome!

Regs,

gd



Enabled debugging on span 3
   -- Executing Dial(SIP/gdevlet-089534a8, ZAP/8/06x) in new stack
3 -- Making new call for cr 132
   -- Requested transfer capability: 0x00 - SPEECH
3  Protocol Discriminator: Q.931 (8)  len=51
3  Call Ref: len= 1 (reference 4/0x4) (Originator)
3  Message type: SETUP (5)
3  [04 03 80 90 a3]
3  Bearer Capability (len= 5) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
3   Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
3   Ext: 1  User information layer 1: A-Law (35)
3  [18 01 8a]
3  Channel ID (len= 3) [ Ext: 1  IntID: Implicit, Other Spare: 0,
Exclusive Dchan: 0
3 ChanSel: B2 channel
3  ]
3  [28 0c 47 69 72 61 79 20 44 65 76 6c 65 74]
3  Display (len=12) [ Giray Devlet ]
3  [6c 06 00 81 33 31 30 31]
3  Calling Number (len= 8) [ Ext: 0  TON: Unknown Number Type (0)
NPI: Unknown Number Plan (0)
3Presentation: Presentation permitted,
user number passed network screening (1) '3101' ]
3  [70 0b 80 30 36 32 34 32 36 32 38 36 37]
3  Called Number (len=13) [ Ext: 1  TON: Unknown Number Type (0)
NPI: Unknown Number Plan (0) '0624262867' ]
3  [7d 02 91 81]
3  High-layer compatibilty (len= 4) [ 3 0x91 3 0x81 3  ]
   -- Called 8/06xx
3 No response to SETUP message
3 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated,
peerstate Overlap sending
   -- Channel 0/2, span 3 got hangup, cause 18
3 NEW_HANGUP DEBUG: Destroying the call, ourstate Call Initiated,
peerstate Overlap sending
3 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Call Initiated,
peerstate Overlap sending
   -- Hungup 'Zap/8-1'
 == Everyone is busy/congested at this time (1:0/0/1)
 == Auto fallthrough, channel 'SIP/gdevlet-089534a8' status is 'CHANUNAVAIL'





/etc/asterisk/zapata.conf
switchtype = euroisdn
; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode)
signalling = bri_cpe_ptmp

pridialplan = unknown
prilocaldialplan = unknown
overlapdial=yes
nationalprefix = 0
internationalprefix = 00
;usecallingpres=yes
callerid=asreceived
;priindication = passthrough

echocancel = yes
echocancelwhenbridged = yes
echotraining = 100


context=isdn-incoming
; S/T port 1 - 4
channel = 1-2
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Re: [asterisk-users] PBAX-Group with QuadBRI card, outgoing call problem

[Steve Davies]

On 10/25/06, Giray Devlet [EMAIL PROTECTED] wrote:


/etc/asterisk/zapata.conf
switchtype = euroisdn
; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode)
signalling = bri_cpe_ptmp



Have you tried signalling = bri_cpe if you have a group of ISDN
channels, they are more often in a PTP mode than a PTMP.

Just a thought.
Steve
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Re: [asterisk-users] Adit 600 resetting

[Doug Lytle]

Don Wisdom wrote:

Hi All,
Im trying to erase the config in a addit that I got off of ebay.   I know
  


Try no password.  Just hit enter.  If that doesn't work, you'll have to 
contact Carrie Access technical support.  They'll charge you an arm and 
a leg.  Nobody has reported any successes with password cracking 
software to date.


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] All calls Hangup after receive these logs.

[Doug Lytle]

Xue Liangliang wrote:

Hi, all i receive these logs quite often, and all the calls hangup
after receiving these .



 Oct 25 11:17:44 NOTICE[5121]: chan_zap.c:8176 pri_dchannel: PRI got
event: Alarm (4) on Primary D-channel of span 1
Oct 25 11:17:44 WARNING[5121]: chan_zap.c:2289 pri_find_dchan: No
D-channels available!  Using Primary channel 16 as D-channel anyway!


Looks like your PRI dropped and then came back up.

Doug

-- Ben Franklin quote: Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Broadvoice incoming DTMF problems

[Al Bochter]

dtmf = inband

Best regards,

Al Bochter
Bochter Services

(Voip PBX) Toll Free: 866-638-1254  EXT: 250
(Voip PBX) Free World DialUp: 780217 EXT: 250

(Voip) Cellular: 712-432-5401

http://www.BochterServices.com/?t=Email

BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=goldt=email

For new and used security items
http://www.bochterservices.com/?j=storet=email_security

GOLD PLATING SERVICES
http://www.bochterservices.com/?j=platingt=email



Kevin Kiely wrote:


Is anyone having problems and Broadvoice with incoming DTMF not being
recognized from a caller originating on the PSTN connection to Broadvoice?

Broadvoice tech support confirmed this issue as a result of their carrier
connections and suggested a work around in the dial plan(SIPDtmf).  This
does work but breaks DTMF for BroadVoice callers.



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Inbound (clean). Database: 0643-2, 10/24/2006 - 10/25/2006 1:05:14 AM




 


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Re: [asterisk-users] Re: Dynamic Codec Selection

[Wildheart]

Hi Marty,

   By the outside world, I mean the PSTN connection. I am still interested
in how you would set this up. Can you paste in a sample config?

   With thanks,

Tim

 On 2006-10-24 06:44:01 -0700, Wildheart
 [EMAIL PROTECTED] said:

 Hi,

 Does anyone know a what to use a different codec for calls which are
 handset to handset (eg, G711) then when we have calls to the out side
 world (via an asterisk server) to use a different codec(eg, G729)?

 The idea is to reduce the bandwidth to the server for the majority
 of
 calls, but get good quality on internal calls.

 With thanks,

 yes, this is simple,  just make it so the extensions allow both g729
 and ulaw, and set your outside world is g729.

 Marty



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Re: [asterisk-users] UA - number assignment

[Brian Candler]

On Wed, Oct 25, 2006 at 11:11:00AM +0300, Paul Ianas wrote:
 Indeed, this is what I want to know. When somebody wants to make a call
 (using a standard telephone, connected to a media gateway), he doesn't
 know what user is in my Asterisk conf. He only knows that he wants to
 call John, who has the number 102 for example. He dials 102 from his
 phone and the call is routed to the corresponding user (this is
 Asterisk's job).

Which is controlled by the dial plan in extensions.conf

 I want to know only the way I can make an UA-number assignment.

Well, the simplest way is to put a new line in extensions.conf for each
phone number: at minimum, this might look like

exten = 101,1,Dial(SIP/john)
exten = 102,1,Dial(SIP/fred)
... etc

However it gets long-winded if you have multiple extensions which all want
the same logic (e.g. forward on no answer to voicemail)

Personally I use macros and sub-contexts to clean this up. An example is
shown below. Each new local user you add just needs a single entry under the
[extensions] section, plus an entry in their specific channel (e.g.
sip.conf). This is reasonably easy to manage.

HTH,

Brian.

- extensions.conf --

; This is the macro for placing a call to a user
[macro-ext]
exten = s,1,Dial(${ARG1},15)
exten = s,2,Playback(vm-nobodyavail)
exten = s,3,Hangup()
exten = s,102,Playback(tt-allbusy)
exten = s,103,Hangup()

; These are mappings of internal extension numbers to destinations
[extensions]
exten = 101,1,Macro(ext,Zap/1)
exten = 102,1,Macro(ext,Zap/2)
exten = 301,1,Macro(ext,SIP/john)
exten = 401,1,Macro(ext,SIP/tulip1)
exten = 402,1,Macro(ext,SIP/tulip2)

; This allows outgoing calls (prefixed with 9) via the Zaptel FXO port
[outbound]
exten = _9.,1,Dial(Zap/4/${EXTEN:1})
exten = _9.,2,Congestion()
exten = _9.,102,Congestion()

; This matches anything else, i.e. invalid numbers
[invalid]
exten = _X!,1,Answer()
exten = _X!,2,Background(pbx-invalid)

; Now you create a context for each class of user, and include whichever
; sub-contexts are permitted for those users. They are tried in sequence.

; Registered SIP clients go in this context - they can place PSTN calls
[from-sip]
include = extensions
include = outbound
include = invalid

; Directly-connected phones on FXS ports
[internal]
include = extensions
include = outbound
include = invalid

; Incoming SIP calls from arbitary hosts on the Internet
[default]
include = extensions
include = invalid

; incoming calls on the FXO port are directed to this context
; from zapata.conf
[incoming]
include = extensions
exten = s,1,Answer()
exten = s,2,Wait(1)
exten = s,3,Background(enter-ext-of-person)
exten = i,1,Background(pbx-invalid)
exten = t,1,Playback(vm-goodbye)
exten = t,2,Hangup()

; Note that people dialling into our system are NOT allowed to access
; the 'outbound' context and place toll calls back out again!

---

Because the call handling logic is in a macro, if you decide to change it -
e.g. you find that 15 seconds of ringing is too short, and you want to make
it 30 seconds - you only have to do this in one place:

exten = s,1,Dial(${ARG1},30)

and it applies to all users.

Finally, you need to put each device in the correct context.

 sip.conf 

[general]
context=default ;  NOTE

[john]
type=friend
secret=XX
context=from-sip;  NOTE
callerid=John Smith 301
nat=no
canreinvite=yes
host=dynamic

;...etc

 zapata.conf ---

; Zap/1=FXS, Zap/2=FXS, Zap/3=not installed, Zap/4=FXO
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no

context=internal;  NOTE
signalling=fxo_ls
txgain=-6.0
callerid=Red Phone 101
channel = 1

context=internal;  NOTE
signalling=fxo_ls
txgain=-6.0
callerid=Blue Phone 102
channel = 2

context=incoming;  NOTE
signalling=fxs_ls
channel = 4

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Re: [asterisk-users] One way audio half way through call

[Matt]

So no one has any solution to this, huh?  We can't be the only two
people having this problem.

On 10/24/06, Matt [EMAIL PROTECTED] wrote:

Just as a follow up.. on the OTHER server that is connected I'm seeing:
chan_iax2.c: Received VNAK: resending outstanding frames


On 10/24/06, Matt [EMAIL PROTECTED] wrote:
 I am getting the following on my server when the problem happens:

 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
 within window 209-209
 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
 within window 209-210
 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
 within window 209-211
 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
 within window 209-211
 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 207 not
 within window 209-211
 Oct 24 10:26:41 DEBUG[28361] chan_iax2.c: Received iseqno 208 not
 within window 209-212

 Any idea what this means?  To me it looks like it just is missing a
 packet, but why does it not continue?

 On 10/23/06, Matt [EMAIL PROTECTED] wrote:
  Have you tried disabling the jitterbuffer?  Maybe it is a bug in the
  jitterbuffer code, then?
 
  On 10/23/06, Pavel Jezek [EMAIL PROTECTED] wrote:
   I have same problem, but with 1.4 branch, after several minutes,
   asterisk stops sending packets resulting one way audio,
   this problem appears especialy when bigger jitter appears (300ms) on
   one connection (I have jitterbuffer enabled on IAX),
   bigger jitter resulting in bigger one way audio probability in my 
case...
   PJ
  
  
   Matt wrote:
Hi,
I have asterisk 1.2.12 running on my server.   Everything seems to be
working fine on it.  It has an IAX connection to the
terminator/orignator.   Again, everything seems to be fine.. calls
come in and go out.  However, it seems that after a call has been up
for several minutes audio will go one-way.   That is, we can hear the
other person, but they can not hear us.
   
Any thoughts?
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Re: [asterisk-users] Re: IAX2 goes one way audio when lag gets bad

[Matt]

Marty,
Thanks for the suggestion... unfortunately it is not a case of the
bandwidth being hammered.   The only things on this connection is the
voice.My thought is there is something wrong, possibly, with the
cable provider's node.  Still.. Asterisk shouldn't just barf with
one-way-audio.

On 10/25/06, Martin Joseph [EMAIL PROTECTED] wrote:

On 2006-10-24 13:04:02 -0700, Matt [EMAIL PROTECTED] said:

 Hi,
 I have a customer who experiences, once in a while, one-way audio...
 That is... they can hear the person they called, but the person can
 not hear them.

 The customer is connected via IAX2 to our softswitch.

 On the customer's end I have the following config in iax.conf:
 [general]
 bindport = 4569   ; Port to bind to (IAX is 4569)
 bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
 disallow=all
 allow=ulaw
 notransfer=yes
 trunk=no
 (I have also tried trunk=yes and nothing for trunk=)
 jitterbuffer=yes
 forcejitterbuffer=yes
 mailboxdetail=yes
 dropcount=3
 minexcessbuffer=80
 jittershrinkrate=1

 I have tried with jitterbuffer=no, and then rather then one-way-audio
 I get high packet loss until the connection settles back down.Any
 ideas on other things I can try?
Implement QoS that prevents the upstream bandwidth from the customers
site from being completely hammered...

Just a thought,
Marty


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Re: [asterisk-users] Re: IAX2 goes one way audio when lag gets bad

[Matt]

Pavel,
What version of asterisk are you connecting back to?  Is it also 1.4. branch?

On 10/25/06, Pavel Jezek [EMAIL PROTECTED] wrote:

I have same problem, but only with 1.4 branch and when some bigger
jitter occur (1.2 is working fine, even in case with big jitter),
I dump packets with tcpdump and see, that asterisk stops sending packets
in one direction...
Maybe good reason to open bug report for this, because QoS settings ins
not always possible (e.g. my case with CDMA connection)
PJ



Martin Joseph wrote:
 On 2006-10-24 13:04:02 -0700, Matt [EMAIL PROTECTED] said:

 Hi,
 I have a customer who experiences, once in a while, one-way audio...
 That is... they can hear the person they called, but the person can
 not hear them.

 The customer is connected via IAX2 to our softswitch.

 On the customer's end I have the following config in iax.conf:
 [general]
 bindport = 4569   ; Port to bind to (IAX is 4569)
 bindaddr = 0.0.0.0; Address to bind to (all addresses on machine)
 disallow=all
 allow=ulaw
 notransfer=yes
 trunk=no
 (I have also tried trunk=yes and nothing for trunk=)
 jitterbuffer=yes
 forcejitterbuffer=yes
 mailboxdetail=yes
 dropcount=3
 minexcessbuffer=80
 jittershrinkrate=1

 I have tried with jitterbuffer=no, and then rather then one-way-audio
 I get high packet loss until the connection settles back down.Any
 ideas on other things I can try?
 Implement QoS that prevents the upstream bandwidth from the customers
 site from being completely hammered...

 Just a thought,
 Marty


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[asterisk-users] asterisk 1.4 problem with call queues

[Dean Bath]

Hi,



Im posting here as I have found an issue in
1.4, and hoping someone might be able to help.



I have setup a call queue in asterisk, a call comes
into the queue, asterisk calls the agents, an agent answers the call fine, but
if they try and transfer the call, asterisk drops out with a segmentation fault
(below).



Disconnected from Asterisk server

voip1:/etc/asterisk# /usr/sbin/safe_asterisk: line
157: 2985 Segmentation fault (core dumped)
nice -n $PRIORITY ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS}
1/dev/${TTY} /dev/${TTY}

Asterisk ended with exit status 139

Asterisk exited on signal 11.

Automatically restarting Asterisk.



I have got a copy of the core dump, with gdb I typed
bt  results below.



(gdb) bt

#0 0x3c202273 in ?? ()

#1 0x08078b42 in
ast_bridged_channel (chan=0x81dd400) at channel.c:3529

#2 0xb75451d6 in
process_sdp (p=0x81ef7b0, req=0xb74fcfac) at chan_sip.c:5065

#3 0xb754a41e in handle_request_invite
(p=0x81ef7b0, req=0xb74fcfac, debug=0,

 seqno=1,
sin=0xb74fe2d0, recount=0xb74fe2e0,


e=0xb74fd1cf sip:xx(masked)[EMAIL PROTECTED]) at chan_sip.c:12988

#4 0xb75515c5 in
handle_request (p=0x81ef7b0, req=0xb74fcfac, sin=0xb74fe2d0,


recount=0xb74fe2e0, nounlock=0xb74fe2e4) at chan_sip.c:14303

#5 0xb7553b1d in
sipsock_read (id=0x81a99c8, fd=16, events=1, ignore=0x0)

 at
chan_sip.c:14448

#6 0x080a1920 in
ast_io_wait (ioc=0x816a2d0, howlong=233) at io.c:279

#7 0xb753565c in
do_monitor (data="" at chan_sip.c:14641

#8 0x080ec910 in
dummy_start (data="" at utils.c:544

#9 0xb7faf0bd in
start_thread () from /lib/tls/libpthread.so.0 #10 0xb7ddf8ae in clone () from
/lib/tls/libc.so.6

(gdb)





Im running libpri and zaptel beta1 and
asterisk beta3 on Debian 2.6.17.3



I can reproduce this error if anything else is
needed.



Any help would be great.



I did post this on the
dev-list yesterday but had no response yet, has anybody else tried the call
queues? It does this for me with beta2 and beta3 releases.





Thanks,

Dean Bath








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Re: [asterisk-users] Broadvoice incoming DTMF problems

[Dovid B]

Is anyone having problems and Broadvoice with incoming DTMF not being
recognized from a caller originating on the PSTN connection to Broadvoice?

This is the reason why I left them two months after I signed up with them.



Broadvoice tech support confirmed this issue as a result of their carrier
connections and suggested a work around in the dial plan(SIPDtmf).  This
does work but breaks DTMF for BroadVoice callers.

Find a better carrier :)


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Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK

[Faris Raouf]

Henry.L.Coleman wrote:

Yep, just swop the two wires. Sometimes the Tip and Ring get reversed
and   most loop start interfaces don't really care (they work either way).
It's worth a try since if the disconnect is a reverse polarity flash then
the card may see not see this condition as it is already reversed.

I have a similar problem with Foriegn Exchange line (FX) but I haven't had
time to visit the client to check this out yet.



Thanks Henry. I'll definitely give this a go.

Faris.

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Re: [asterisk-users] PBAX-Group with QuadBRI card, outgoing call problem

[Giray Devlet]

Hi Steve,

THX!!! This works ...  couldn't really find anywhere what other
options I could use as values for signalling ... thx!

gd

From: Steve Davies [EMAIL PROTECTED]

On 10/25/06, Giray Devlet [EMAIL PROTECTED] wrote:


/etc/asterisk/zapata.conf
switchtype = euroisdn
; p2mp TE mode (for connecting ISDN lines in point-to-multipoint mode)
signalling = bri_cpe_ptmp



Have you tried signalling = bri_cpe if you have a group of ISDN
channels, they are more often in a PTP mode than a PTMP.

Just a thought.
Steve

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Re: [asterisk-users] Call is not coming through sipgate.co.uk+Asterisk

[Dovid B]

Are you behind NAT. Any firewall's ?

  - Original Message - 
  From: 
  Crazy 
  Boy 
  To: asterisk-users@lists.digium.com 
  
  Sent: Wednesday, October 25, 2006 10:54 
  AM
  Subject: [asterisk-users] Call is not 
  coming through sipgate.co.uk+Asterisk
  Hi,I have installed Asterisk, Zaptel, Libpri, Addons, 
  Sounds in my Linux system. I got registered with sipgate.co.uk and got the UK 
  phone number i.e., 0207100. I configured my Asterisk server with 
  0207100. When I made a call to this number from outside phone, my XLite 
  extension is not ringing. Its directly going to Voicemail or telling that 
  "person is unavailable". When I made a call, Asterisk console is also not 
  showing anything. But, sipgate website is showing my calls list. I thought 
  that When I made a call from outside to my number, call is going to 
  sipgate.co.uk and its not routing to my server. When I execute "sip show 
  registry", its not displaying anything. Here I am giving my 
  configuration details:My sip.conf file 
  contents:[general]port = 5060bindaddr = 
  0.0.0.0qualify=nodisable=allallow=alawallow=alawallow=ulawallow=g729allow=gsmallow=slinearsrvlookup=yes[250]type=friendusername=250secret=dannycallerid="Danny"host=dynamiccontext=demoregister 
  = 
  100:[EMAIL PROTECTED]/100[sipgate4]type=frienddisallow=allallow=alawallow=ulawfromuser=100authuser=100secret=passwordusername=100host=sipgate.co.ukcontext=demodtmfmode=infofromdomain=sipgate.co.ukinsecure=verynat=yescanreinvite=nocallerid="Danny" 
  lt;0207100My 
  Extensions.conf file contents:[demo]exten = 
  250,1,Dial(SIP/250,20)exten = 250,2,Voicemail(u250)exten = 
  250,3,Voicemail(b250)exten = 250,4,Hangupexten = 
  _0207.,1,SetCallerID("" 
  lt;100|a) 
  ;Outgoingexten = 
  _0207.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],40,tr)exten = 
  100,1,Dial(SIP/250,30,tr) 
  ;IncomingAm I have to install any other libraries?Anything wrong 
  in the above configuration?Looking forward to your response. Thanks in 
  advance.Regards,Chandra.
  
  
  All-new 
  Yahoo! Mail - Fire up a more powerful email and get things done faster.
  
  

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Re: [asterisk-users] Problem with CallerID (UK) TDM400P ( CID timed out waiting for ring )

[Faris Raouf]

[EMAIL PROTECTED] wrote:

We have a problem where callerid works 50% of the time on both lines.  What
we are seeing in the logs is:


Hi Phil,

Unfortunately your configuration looks OK to me.

Here's mine, which works 100% with CID (but not dratted hangup 
detection!). There are some duplications and things - just ignore them. 
I note that you have sendcalleridafter=2
and I have =1 but I think =2 is just fine. The only thing I can suggest 
is to play with the RX gain in case things are just too quiet:


usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=no
callwaitingcallerid=no
threewaycalling=no
transfer=no
canpark=yes
cancallforward=yes
callreturn=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
relaxdtmf=yes
group=1
immediate=no

signalling=fxo_ks
language=en
context=sip2
channel = 1

signalling=fxo_ks
language=en
context=blah2
channel = 2

usecallerid=yes
cidsignalling=v23
;cidstart=ring
cidstart=polarity
sendcalleridafter=1

busydetect=yes
busycount=3

;callsrogress=yes
progzone=uk

rxgain=2.5
txgain=2.0
ringtimeout=5000

signalling=fxs_ks

polarityonanswerdelay=1000
answeronpolarityswitch=yes
hanguponpolarityswitch=yes
resetpolarityonring=true
language=en
context=blah
channel = 3


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Re: [asterisk-users] need help using tftp for polycom 501

[Marlin Unruh]

Marlin Unruh wrote:

Hi,

I have a Polycom 501 that is currently unusable because I started a
firmware and sip upgrade that I can't complete.

The Ubuntu box address is set static at: 192.168.1.101.
The phone address is set static at 192.168.1.51.

The phone settings for the server menu are:
Server Type: Trivial FTP
Server Address: 192.168.1.101
Server User: PlcmSpIp
Server Password: PlcmSpIp (not sure what it should be)
Pro. Method: default

I am using tcpdump to watch the network messages, and I see the phone
sending messages like:


11:04:50.147597 IP 192.168.1.51.1025  192.168.1.101.69:  19 RRQ 
bootrom.ld octet
11:04:58.235875 IP 192.168.1.51.1027  192.168.1.101.69:  25 RRQ 
0004f21136a1.cfg octet
11:06:36.728815 IP 192.168.1.51.1029  192.168.1.101.69:  25 RRQ 
.cfg octet


I have the following files in the directory /srv/tftp:

0004f21136a1.cfg  bootrom.ld  phone774110.cfg  sip.cfg

I have edited 0004f21136a1.cfg to point to phone774110.cfg

I get the following message on the phone:
Could not contact boot server.
error loading 004f21136a1.cfg

If I ps -e I see tftp is active.

I am at a total lose how to setup and use tftp properly. I have searched 
the Internet and read man pages, but I can't get it into my head.


Any help will be very much appreciated.

Glad to say I got it working. Sad to say I had to go to Windows to 
accomplish it. I used tftpd32 and it worked perfect.


I would like to use tftp under Linux. May I will try again later.

--
 Marlin
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[asterisk-users] Quintum DX as gateway to PSTN for Asterisk

[doki_cti]

Hello,
I try configuring Quintum DX gateway as link to PSTN for *. Now, I can dial 
number which is connect to Quintum, and call is diverted to *. I don't know 
what I should set, if I want call from SIP_phone registred in  Asterisk to PSTN 
via Quitnum. I set in sip.conf account for Quintum 
[sip_proxy-out]
type=peer   
outboundproxy=QUINTUM_IP   

, and changed extensions.conf. When I call from SIP Phone, I see in Quintum 
log, that call is received with good caller and called numbers, but I think 
that quintum don't how route this call (he diverte this call to asterisk). So, 
can you  give me advice what I should set, when I want route all calls from IP 
to PSTN and from PSTN to * via IP?

How set password and user for quitnum and calls from SIP? Is it posible  on 
Quintum or I should use for this radius? 

Regards
Doki
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Re: [asterisk-users] need help using tftp for polycom 501

[Doug Lytle]

Marlin Unruh wrote:


Glad to say I got it working. Sad to say I had to go to Windows to 
accomplish it. I used tftpd32 and it worked perfect.


I would like to use tftp under Linux. May I will try again later.

Why not use just standard FTP?  I use ProFTP and setup a Polycom user.  
Works great.


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] need help using tftp for polycom 501

[joe, at j4computers]

Marlin Unruh[EMAIL PROTECTED] Wrote on: 10/25/2006 8:12 AM:
 Marlin Unruh wrote:
 Hi,
 
 I have a Polycom 501 that is currently unusable because I started a
 firmware and sip upgrade that I can't complete.
 
 The Ubuntu box address is set static at: 192.168.1.101.
 The phone address is set static at 192.168.1.51.
 
 The phone settings for the server menu are:
 Server Type: Trivial FTP
 Server Address: 192.168.1.101
 Server User: PlcmSpIp
 Server Password: PlcmSpIp (not sure what it should be)
 Pro. Method: default
 
 I am using tcpdump to watch the network messages, and I see the phone
 sending messages like:
 
 
 11:04:50.147597 IP 192.168.1.51.1025  192.168.1.101.69:  19 RRQ 
 bootrom.ld octet
 11:04:58.235875 IP 192.168.1.51.1027  192.168.1.101.69:  25 RRQ 
 0004f21136a1.cfg octet
 11:06:36.728815 IP 192.168.1.51.1029  192.168.1.101.69:  25 RRQ 
 .cfg octet
 
 I have the following files in the directory /srv/tftp:
 
 0004f21136a1.cfg  bootrom.ld  phone774110.cfg  sip.cfg
 
 I have edited 0004f21136a1.cfg to point to phone774110.cfg
 
 I get the following message on the phone:
 Could not contact boot server.
 error loading 004f21136a1.cfg
 
 If I ps -e I see tftp is active.
 
 I am at a total lose how to setup and use tftp properly. I have searched 
 the Internet and read man pages, but I can't get it into my head.
 
 Any help will be very much appreciated.
 
 Glad to say I got it working. Sad to say I had to go to Windows to 
 accomplish it. I used tftpd32 and it worked perfect.
 
 I would like to use tftp under Linux. May I will try again later.
 

I had a similar problem. 

Look at /etc/xinetd.d/tftp  (for SLES 10 (SUSE)).

My server_args was hosed.

I made it server_args = /tftpboot -s -c -v and everyone was happy. (except my 
ACT P160s, but that's another story)

Entire file here (/tftpboot can be whatever you set up):

# default: off
# description: tftp service is provided primarily for booting or when a \
#   router need an upgrade. Most sites run this only on machines acting as
#   boot servers.
service tftp
{
socket_type = dgram
protocol= udp
wait= yes
user= root
server  = /usr/sbin/in.tftpd
server_args = /tftpboot -s -c -v  
disable = no
}

joe a.
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[asterisk-users] SIP problem - ACT p160s error

[joe, at j4computers ([EMAIL PROTECTED])]

I have a setup with a polycom 601 and an act p160s.  All on local segment, no 
NAT.  

Can call the act p160s, from the polycom, rings, connects, and a conversation 
can take place.  The reverse is not true, Dialing from the act to the polycom 
does not work.  SIP debug shows, at the end, Incoming call: got sip response 
416 unsupported URI Scheme back from 192.168.0.xxx.  Which is the act phone, 
the orginator.

One presumes this is a configuration issue with the Act phone.  Any clues?   
Such as what a proper config for this phone should look like?  Act support has 
made an initial response, but there is a big time lag them being on the other 
side of the earth.

joe a.

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Re: [asterisk-users] AstFax Sending a Fax

[Barry Fawthrop]

Thanks Andrew
I have no plans to VoIP my Faxes to a VoIP provider

I just would like to send them from my  desktop (which is windows) to my 
PBX (which is AstLinux inside  a net 4801)

The PBX connects to PSTN lines via a FXO Gateway (CG-410 in my case)

So really it's trying to get Windows to detect the modem or phone line 
one the 410.


Can this still be done ?
Thanks again
Barry


Andrew Joakimsen wrote:
You can use the fax server Hylafax ( 
http://www.voip-info.org/wiki/index.php?page=Asterisk+IAXmodem ) with 
IAXmodem ( http://iaxmodem.sourceforge.net/howto.php )


You really don't want to be sending faxes over the internet via VoIP 
providers, not yet because there is no t.38 support for that. As long 
as the connection to the PSTN is on a card on the same machine or 
possibly over a network connection perhaps over a private line maybe 
using TDMoE then it should work fine



On 10/24/06, *Barry Fawthrop* [EMAIL PROTECTED] 
mailto:[EMAIL PROTECTED] wrote:


Hi All

I'm trying to understand how I would send my fax ?

If I use  Word  or what ever word processor  or even an email
client to
create what I want faxed.

I have *asterisk setup with and FXO Gateway that will make the
call to
the fax number I dial
SIP extension 320  is the FXO gateway.

How do I now get my email or word document to TIFF to then fax to the
FXO gateway or SIP/320 ?

I don't understand that part.  They all talk about an email with a
TIF
attachment
and the TIF attachment is sent to the number in the subject line.

Thanks all
Barry
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http://lists.digium.com/mailman/listinfo/asterisk-users




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[asterisk-users] Maximum talktime in a queue?

[Rajkumar S]

Hi,

Is it possible to define maximum talk time in a queue? ie any one who
joins a queue should not be able to talk more than say 5 minutes to
the agent.

raj
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[asterisk-users] Nerdvittle's Reminders and Zaptel

[John McCollough]

I am attempting to implement Nerdvittle's Call Reminders on my *
1.2.12.1 PBX.  It has 8 Zaptel trunks provided by 2 Digium TDM400P
cards.

If I use the call reminders internally, it works flawlessly.  The
problem happens when I set the call-back number to an external number so
that the call goes out the Zaptel trunk.  The reminder message starts
playing before the person being called picks up the phone.  As a result,
the person picks up the phone and either hears only part of the message
or just gets the menu prompt.

What appears to be happening is that the reminder script simply waits
for a connected call, then starts playing it's message, but * reports a
connected call when it connects to the trunk, not when the other party
picks up.  The result is the message starts playing while the remote
phone is still ringing.

I was wondering if anyone had any suggestions on how to work around this
problem.  The only thing I can think of that is within my ability is to
repeatedly play a prompt to press a number to hear the message.
However, this is not a good solution since this system is going to be
used to automate patient appointment reminders, and I don't want it
sounding like a sales call that will cause people to just hang up
without listening first.  My other thought is to perhaps wait until it
hears a word spoken, but I don't know how to do that and have the system
distinguish between a spoken word and the sound of the ring indicator.
Of course, I'm open to any other suggestions as well.

John
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Re: [asterisk-users] ASterisk Start problem

[J. Oquendo]

ram wrote:

Hi all
 
I have installed 1.2.12.1 http://1.2.12.1 in FC5 with libpri.1.2.4
 
when i start
 
iam getting the following error and it quits
 
  == Registered channel type 'Local' (Local Proxy Channel Driver)
 [chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325 
__load_resource: libpri.so.1.0: cannot open shared object file: No 
such file or directory
Oct 23 16:16:07 WARNING[11084]: loader.c:554 load_modules: Loading 
module chan_zap.so failed!

[EMAIL PROTECTED] agc]# Ouch ... error while writing audio data: : Broken pipe
 
what is the problem, any suggestions ?
 
Ram



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Did you build Zapata and libpri

--

J. Oquendo
http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743
sil . infiltrated @ net http://www.infiltrated.net 


The happiness of society is the end of government.
John Adams



smime.p7s
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Re: [asterisk-users] need help using tftp for polycom 501

[Anthony Rodgers]

IMHO, FTP really is the way to go - you get the ability to have the  
phones detect config file changes and automatically reboot, and you  
get the ability to upload logs, custom configs and directories from  
the phones.


We use vsftpd, with the default user and password for the phone.

CP

On 25-Oct-06, at 7:29 AM, Doug Lytle wrote:


Marlin Unruh wrote:

 Glad to say I got it working. Sad to say I had to go to Windows to
 accomplish it. I used tftpd32 and it worked perfect.

 I would like to use tftp under Linux. May I will try again later.

Why not use just standard FTP?  I use ProFTP and setup a Polycom user.
Works great.

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] Nerdvittle's Reminders and Zaptel

[Doug Lytle]

John McCollough wrote:

I was wondering if anyone had any suggestions on how to work around this
problem.  The only thing I can think of that is within my ability is to
  


The common work around for analog lines it to loop a message asking the 
caller to press 1 to accept the call.  Loop it long enough that the 
caller has time to respond.  I used to loop 5 times before hanging.  
After I moved to a PRI, it was no longer necessary.


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] Broadvoice incoming DTMF problems

[Al Bochter]

That too.

I never used Broadvoice but from what users have told me high priced 
poor service.


There are better with no connect fees

Best regards,

Al Bochter
Bochter Services

(Voip PBX) Toll Free: 866-638-1254  EXT: 250
(Voip PBX) Free World DialUp: 780217 EXT: 250

(Voip) Cellular: 712-432-5401

http://www.BochterServices.com/?t=Email

BUY and sell Coins, Silver and Gold
http://www.bochterservices.com/?j=goldt=email

For new and used security items
http://www.bochterservices.com/?j=storet=email_security

GOLD PLATING SERVICES
http://www.bochterservices.com/?j=platingt=email



Dovid B wrote:


Is anyone having problems and Broadvoice with incoming DTMF not being
recognized from a caller originating on the PSTN connection to 
Broadvoice?


This is the reason why I left them two months after I signed up with 
them.



Broadvoice tech support confirmed this issue as a result of their 
carrier

connections and suggested a work around in the dial plan(SIPDtmf).  This
does work but breaks DTMF for BroadVoice callers.


Find a better carrier :)


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Inbound (clean). Database: 0643-3, 10/25/2006 - 10/25/2006 9:34:56 AM





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RE: [asterisk-users] Nerdvittle's Reminders and Zaptel

[John McCollough]

 
So a PRI line resoves this issue as well?  That's good.  I believe there
are plans for upgrading to one.

Thank you

John

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Wednesday, October 25, 2006 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Nerdvittle's Reminders and Zaptel

John McCollough wrote:
 I was wondering if anyone had any suggestions on how to work around 
 this problem.  The only thing I can think of that is within my ability

 is to
   

The common work around for analog lines it to loop a message asking the
caller to press 1 to accept the call.  Loop it long enough that the
caller has time to respond.  I used to loop 5 times before hanging.  
After I moved to a PRI, it was no longer necessary.

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] Nerdvittle's Reminders and Zaptel

[Andrew Kohlsmith]

On Wednesday 25 October 2006 09:20, John McCollough wrote:
 What appears to be happening is that the reminder script simply waits
 for a connected call, then starts playing it's message, but * reports a
 connected call when it connects to the trunk, not when the other party
 picks up.  The result is the message starts playing while the remote
 phone is still ringing.

If you're playing out of a POTS interface, that's about as good as you can 
get... You can Wait() but POTS has no real answer supervision.  As soon as 
the line is dialed it is considered answered.

 I was wondering if anyone had any suggestions on how to work around this
 problem.  The only thing I can think of that is within my ability is to
 repeatedly play a prompt to press a number to hear the message.
 However, this is not a good solution since this system is going to be
 used to automate patient appointment reminders, and I don't want it
 sounding like a sales call that will cause people to just hang up
 without listening first.  My other thought is to perhaps wait until it
 hears a word spoken, but I don't know how to do that and have the system
 distinguish between a spoken word and the sound of the ring indicator.
 Of course, I'm open to any other suggestions as well.

There is an app called amd - answering machine detection.  You could modify 
this app to listen for voice energy and only continue after the person on the 
other end stops talking (person or machine).

-A.
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Re: [asterisk-users] Call is not coming through sipgate.co.uk+Asterisk

[Brian Candler]

On Wed, Oct 25, 2006 at 01:54:43AM -0700, Crazy Boy wrote:
My sip.conf file contents:
...
[250]
type=friend
username=250
secret=danny
callerid=Danny
host=dynamic
context=demo
register = 100:[EMAIL PROTECTED]/100
...
My Extensions.conf file contents:
[demo]
exten = 250,1,Dial(SIP/250,20)
exten = 250,2,Voicemail(u250)
exten = 250,3,Voicemail(b250)
exten = 250,4,Hangup
exten = _0207.,1,SetCallerID( lt;100|a)
;Outgoing
exten = _0207.,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED],40,tr)
exten = 100,1,Dial(SIP/250,30,tr)
;Incoming
Am I have to install any other libraries?

No.

In the first case, getting incoming calls to work is easy. Start with a
configuration which has nothing to do with sipgate in it. At the top of
sip.conf you should have a [general] section, and you can put the
registration statement there, i.e.

[general]
register = 100:[EMAIL PROTECTED]/101
context=default

In this case, incoming calls to your sipgate.co.uk PSTN number will ring as
101 in context 'default'. I've just tested this with a sipgate.co.uk and it
works fine. (I actually have two accounts, with two register statements,
pointing at two different extensions)

Now, getting outbound to work is a little harder. You need a new entry in
sip.conf to place outbound calls. My first attempt was:

[sipgate-out]
type=peer
host=sipgate.co.uk
username=100
secret=
fromuser=100
fromdomain=sipgate.co.uk

With the correct extensions.conf config (see below), outbound calls worked.
Unfortunately, doing this stopped incoming calls from working; they are
rejected with 401 unauthorised because Asterisk now explictly matches this
SIP entry for incoming calls from sipgate.co.uk, in preference to [general]

So what I eventually ended up with was:

[sipgate-out]
type=friend
host=sipgate.co.uk
username=100
secret=
fromuser=100
fromdomain=sipgate.co.uk
insecure=invite
;context=default  ; not required because I have this in [general] still

I'm not sure if this is the best way to go, but it does seem to work. I
tried moving the register lines under [sipgate-out] and Asterisk no longer
registered. Perhaps register doesn't work for friend entries?

Finally, you need a rule in extensions.conf to route outbound calls via this
link, in whichever context(s) your local phone(s) sit where you want to
allow outbound calls. For example:

[internal]
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],15,r)
exten = _9.,2,Congestion()
exten = _9.,102,Congestion()

This will match all numbers which begin with 9, and route them via sipgate,
stripping off the leading 9.

Regards,

Brian.

P.S. All my testing was with SVN trunk, which is close to 1.4. Behaviour may
be different in earlier versions.
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Re: [asterisk-users] Linksys SPA3102 - PSTN hangup detection in the UK

[Henry.L.Coleman]

You are welcome. Please let me know if this makes any difference.




Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


 Henry.L.Coleman wrote:
 Yep, just swop the two wires. Sometimes the Tip and Ring get reversed
 and   most loop start interfaces don't really care (they work either
 way).
 It's worth a try since if the disconnect is a reverse polarity flash
 then
 the card may see not see this condition as it is already reversed.

 I have a similar problem with Foriegn Exchange line (FX) but I haven't
 had
 time to visit the client to check this out yet.


 Thanks Henry. I'll definitely give this a go.

 Faris.



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Re: [asterisk-users] ASterisk Start problem

[Brian Candler]

 I have installed 1.2.12.1 http://1.2.12.1 in FC5 with libpri.1.2.4
  
 when i start
  
 iam getting the following error and it quits
  
   == Registered channel type 'Local' (Local Proxy Channel Driver)
  [chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325 
 __load_resource: libpri.so.1.0: cannot open shared object file: No 
 such file or directory
 Oct 23 16:16:07 WARNING[11084]: loader.c:554 load_modules: Loading 
 module chan_zap.so failed!
 [EMAIL PROTECTED] agc]# Ouch ... error while writing audio data: : Broken 
 pipe
  
 what is the problem, any suggestions ?

Was libpri installed in /usr/local/lib ?

If so, try

# echo /usr/local/lib /etc/ld.so.conf.d/local.conf
# ldconfig
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Re: [asterisk-users] Polycom SP4000 ftp problem

[Noah Miller]

Hi Edwin -


 rename bootrom.ld to something else like bootrom.ld-disabled.

did that. it hung on sip.ld, rename sip.ld, it hung on
phone1.cfg. seems like if the file is bigger than say 1k.
it'll hang.


I like ProFTPd - it's my ftp daemon of choice for configuring Polycom
phones (including several 4000's), but you might try something else.
Maybe just a different version of ProFTPd would do the trick, or
another daemon altogether.

- Noah
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[asterisk-users] Without ZapTel inferface or Card install , is Conference working or Not

[sunkara]

Hello Users, 

Is Without Zaptel interface Installed, conference Bridge is worked
or not. 

Why it need, For SIP conferences through OpenSER 

Please Help me
For me its Giving Some Errors and warnings. 

== Parsing '/etc/asterisk/meetme.conf': Found
Oct 25 18:16:13 WARNING[12281]: chan_zap.c:913 zt_open: Unable to open
'/dev/zap/pseudo': No such file or directory
Oct 25 18:16:13 ERROR[12281]: chan_zap.c:7396 chandup: Unable to dup
channel: No such file or directory
Oct 25 18:16:13 WARNING[12281]: app_meetme.c:460 build_conf: Unable to
open pseudo channel - trying device
Oct 25 18:16:13 WARNING[12281]: app_meetme.c:463 build_conf: Unable to
open pseudo device
 -- Playing 'conf-invalid' (language 'en')




-- 

  

  Thanks  Regards, 
  Ravi
Prakash Sunkara


  
  
  M:+91
9985077535
O:+91 40 23114549
F:+91 40 40208727 


  
  [EMAIL PROTECTED]
www.hyperion-tech.com
   
  

  




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Re: [asterisk-users] Voicemail help

[Noah Miller]

Hi Bill -


I would like to setup Asterisk for voicemail with CallManager 3.3(5).  I
would like to know what would be the best Distro of Linux to use and
version, what version of Asterisk works best to interact with CallManager,
and what H323 ChannelType works.  As you probably read in another thread I
tried FC5 with Asterisk 1.4 and OOH323 (included with the addons package).
This doesn't seem to work to well, as somewhere along the line either CCM or
OOH323 is disconnecting the call as soon as the playback application is run.


At the present time, I would not use Asterisk 1.4, especially if you
plan on using this in production.  It's definitely still a beta
release.  1.2.12.1 should work for what you're trying to do,
especially with any version of FC (but FC4 or FC5 are probably safest
for right now).  Have you taken a look at the WIKI article:

http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration


- Noah
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[asterisk-users] chan_misdn

[Mark Hannessen]

Hi list,

I ran into some trouble trying to get asterisk (1.4beta2) to compile with 
misdn support. (I need to run a hfc card in NT mode)

when I run ./configure --with-misdn=/usr
it results into the following error:

checking for mISDN_open in -lmISDN... yes
checking /usr/include/mISDNuser/mISDNlib.h usability... no
checking /usr/include/mISDNuser/mISDNlib.h presence... no
checking for /usr/include/mISDNuser/mISDNlib.h... no
configure: ***
configure: *** It appears that you do not have the mISDN development package 
installed.
configure: *** without explicitly specifying --with-misdn

I installed the latest snapshots of mISDN and mISDNuser fron:
http://ftp.uni-bayreuth.de/linux/drivers/isdn4linux/CVS-Snapshots/

mISDN-CVS-2006-10-21 and mISDNuser-CVS-2006-10-20 (both compiled without 
errors for as far as i could see)

ls /usr/include/mISDNuser/
asn1_diversion.h bchannel.h g711.h ibuffer.h isdn_msg.h isound.h mISDNlib.h  
net_l3.h tone.h asn1.h fsm.h helper.h isdn_debug.h isdn_net.h  l3dss1.h 
net_l2.h suppserv.h

anyone any idea what I should try next?

thanks!

Mark Hannessen
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Re: [asterisk-users] IAX2 goes one way audio when lag gets bad

[Noah Miller]

Hi Matt -


I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.

On the customer's end I have the following config in iax.conf:
trunk=no
(I have also tried trunk=yes and nothing for trunk=)
jitterbuffer=yes
forcejitterbuffer=yes
dropcount=3
minexcessbuffer=80
jittershrinkrate=1


If you're using Asterisk 1.2.x, dropcount, jittershrinkrate and
minexcesbuffer don't do anything.  They are ignored by 1.2.x unless
you specify that you want to use the old 1.0.x jitterbuffer.  Instead
you might try the parameters maxjitterbuffer, resyncthreshold, and
maxjitterinterps.  For more, you can check out the sample iax.conf.

I believe, also, that you are correct in setting trunk=no.  I know in
the 1.0.x jitterbuffer, trunk was not fully supported.  I think this
is still the case with the 1.2.x jitterbuffer.

- Noah
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Re: [asterisk-users] Choice of soundfile format

[Matthew Rubenstein]

What's the native soundfile format for SIP? Any idea which soundfile
takes the least CPU for mixing together in conferences?

How about whether the CPU load for conferencing native data is
greater/less than the CPU load for transcoding non-native data that is
CPU lighter in the conference mixing phase?


On Wed, 2006-10-25 at 04:19 -0700,
[EMAIL PROTECTED] wrote:
 Date: Wed, 25 Oct 2006 10:29:32 +0100
 From: Conrad Wood [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] Choice of soundfile format
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Message-ID: [EMAIL PROTECTED]
 Content-Type: text/plain; charset=UTF-8
 
 On Wed, 2006-10-25 at 11:24 +0200, Jon Schpzinsky wrote:
  Hello
  
  What soundfile format, is the one that uses least transcoding during
 playback?
  As I can see, I can choose wav or gsm. What sucks least cpu power,
 during playback to example a Zap channel? I would guess wav, but is
 this correct?
 
 The one that is encoded in the same codec as the codec of the channel.
 On zap it's often alaw or ulaw so you can encode your files like that.
 You can encode the same file with different codecs and save it with
 different extensions (matching the codec) and asterisk will pick the
 most suitable one.
 If the channel is gsm, a gsm encoded file would be most efficient, as
 it
 doesn't need transcoding at all.
 
 Conrad 
-- 

(C) Matthew Rubenstein

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Re: [asterisk-users] Without ZapTel inferface or Card install , is Conference working or Not

[Carlos Chavez]

On Wed, 2006-10-25 at 19:52 +0530, sunkara wrote:
 Hello Users, 
 
 Is Without Zaptel interface Installed,  conference  Bridge is worked
 or not.   
 
 Why it need, For SIP  conferences through  OpenSER 
 
 Please Help me
 For me  its Giving Some Errors and warnings.  
 
You need to install Ztdummy so you can use meetme and Music on Hold.
If you are using kernel 2.4 you need to have USB ports on your machine
for timing, for kernel 2.6 there are no external requirements to compile
ztdummy.

-- 
Carlos Chavez Prats
Director de Tecnología
Telecomunicaciones Abiertas de México S.A. de C.V.
Tel: +52-55-91169161 Ext 2001


signature.asc
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RE: [asterisk-users] Re: Meetme... No channel type registered for 'zap'

[Douglas Garstang]

 -Original Message-
 From: Tony Mountifield [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, October 25, 2006 1:26 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Re: Meetme... No channel type registered for
 'zap'
 
 
 In article 
 [EMAIL PROTECTED],
 Douglas Garstang [EMAIL PROTECTED] wrote:
  Kristian,
   
  I don't have any zap hardware What do I put in 
 zaptel.conf if I don't have any hardware?
  On some other systems we have, with chan_zap not loaded, 
 and no zaptel.conf (running
  1.2.9.1), meetme runs fine. This system with the problem 
 has 1.2.12.1. I wonder if something
  was changed?
 
 Doug, it sounds to me like you don't have the /dev/zap device files.
 
 Do you have the file /etc/udev/permissions.d/zaptel.permissions and
 /etc/udev/rules.d/zaptel.rules installed?

Tony, I don't have /etc/udev/permissions.d/, but I do have the other file.

demeter:(acd1)ipt # ls -l /etc/udev/rules.d/zaptel.rules
-r--r--r--  1 root root 498 Oct 24 15:50 /etc/udev/rules.d/zaptel.rules

 
 What Linux distro are you using?

I'm using Gentoo Linux, and have been for a number of months. This is the first 
time this problem has cropped up. If I have ztdummy installed, why do I need 
the device files? Isn't that what ztdummy is supposed to do?

Doug.

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Re: [asterisk-users] Without ZapTel inferface or Card install , is Conference working or Not

[Noah Miller]

Is Without Zaptel interface Installed, conference Bridge is worked
or not. 

Why it need, For SIP conferences through OpenSER Zaptel interfaces provide timing that is necessary for meetme conferences. When you start a conference, on the cli you can see that asterisk opens a ZAP/pseudo channel. So, if you don't have a zaptel card, as Carlos said, you need to emulate a zaptel interface, and you do that with ztdummy.
- Noah
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[asterisk-users] Re: IAX2 goes one way audio when lag gets bad

[Martin Joseph]

On 2006-10-25 08:14:43 -0700, Noah Miller [EMAIL PROTECTED] said:


Hi Matt -


I have a customer who experiences, once in a while, one-way audio...
That is... they can hear the person they called, but the person can
not hear them.

On the customer's end I have the following config in iax.conf:
trunk=no
(I have also tried trunk=yes and nothing for trunk=)
jitterbuffer=yes
forcejitterbuffer=yes
dropcount=3
minexcessbuffer=80
jittershrinkrate=1


If you're using Asterisk 1.2.x, dropcount, jittershrinkrate and
minexcesbuffer don't do anything.  They are ignored by 1.2.x unless
you specify that you want to use the old 1.0.x jitterbuffer.  Instead
you might try the parameters maxjitterbuffer, resyncthreshold, and
maxjitterinterps.  For more, you can check out the sample iax.conf.

I believe, also, that you are correct in setting trunk=no.  I know in
the 1.0.x jitterbuffer, trunk was not fully supported.  I think this
is still the case with the 1.2.x jitterbuffer.


If the audio is dropping out completely, then I suspect the whole 
jitter buffer thing is a red herring (waste of time).


Perhaps it's a nat issue?  What kind of router if any is involved?  I 
am reaching here... Also, please do tell us which version of asterisk 
you are running...



Marty


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[asterisk-users] Simple example for call transfer.

[Jonson Player]

Hello,
i hev a subscription to a international voip provider and I want
all calls for numbers _001xx to go through my voip provider. I
tried many settings in sip.conf, extensions.conf and iax.conf. Please
give me some simple example for how can i transfer the specified calls
to my external voip provider. What may I put and where in witch file.
Thank you for your support.



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[asterisk-users] Conference is Not Working.... with OpenSER And Asterisk

[sunkara]

Hello Users, 
Good Morning, 
I'm doing on Conference Bridge with Asterisk + OpenSER with CBMySql
modules.
And I'm not Using the Zapptel Cards. 

9001 -- dial 19001(conference Users)---openSER
- Asterisk
In Extension.conf 
 [from-sip]
exten = 19001,1,Playback(conf-hasentered)
;exten = 19001,2,Answer
exten = 19001,2,Wait(2)
exten = 19001,3,CBMysql()

exten = 19002,1,Playback(conf-hasentered)
;exten = 19002,,Answer
exten = 19002,2,Wait(2)
exten = 19002,3,MeetMe(1234|pm)
exten = 19003,1,Playback(conf-hasentered)
;exten = 19003,2,Answer
exten = 19003,2,Wait(2)
exten = 19003,3,MeetMe(1234|pm)
 In cbmysql.conf
[global]
hostname=localhost
dbname=conference
password=
user=root
port=3306
sock=/var/lib/mysql/mysql.sock
DBOpts=yes
;OptsAdm=asdp
;OptsUsr=sdp
ConfApp=MeetMe
ConfAppCount=MeetMeCount
; Choose one of the following to modify early join behaviour
earlyalert=300 ; Tell the participant if they are too early (seconds)
;fuzzystart= ; Allow participants to join early (seconds)

 In sip.conf
[19001]
type=friend
username=9001
secret=august
context=from-sip
host=192.168.2.75
fromdomain=192.168.2.76
realm=192.168.2.75
;[EMAIL PROTECTED]
insecure=very
callerid="Ravi" 9001009
disallow=all
allow=ulaw
allow=gsm
nat=yes

9001,9002,and 9003 is register from openSER, When They dial 19001 to
enter the conference. 

Following are Showing the Errors and Warning 

Executing Playback("SIP/9001-08f8d7e0", "conf-hasentered") in new
stack
 -- Playing 'conf-hasentered' (language 'en')
 -- Executing Wait("SIP/9001-08f8d7e0", "2") in new stack
 -- Executing CBMySQL("SIP/9001-08f8d7e0", "") in new stack
 -- Playing 'conf-getconfno' (language 'en')
Oct 25 18:15:47 NOTICE[12281]: app_cbmysql.c:373 cb_exec: getConf: 1
 -- Playing 'agent-pass' (language 'en')
Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags:
Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags:
Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:146 passQuery: CBMySQL:
Invalid room or pass
Oct 25 18:15:55 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY:
 -- Playing 'auth-incorrect' (language 'en')
Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags:
Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags:
Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:146 passQuery: CBMySQL:
Invalid room or pass
Oct 25 18:16:05 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY:
 -- Playing 'auth-incorrect' (language 'en')
Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:126 passQuery: Admin flags:
Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:127 passQuery: user flags:
Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:149 passQuery: PASSQUERY:
Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:376 cb_exec: getPass: 1
 == Parsing '/etc/asterisk/meetme.conf': Found
Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:97 checkMax: Currentusers:
0
Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:390 cb_exec: checkMax: 1
Oct 25 18:16:13 NOTICE[12281]: app_cbmysql.c:276 enterConf: Roomtype:
1234||
 == Parsing '/etc/asterisk/meetme.conf': Found
Oct 25 18:16:13 WARNING[12281]: chan_zap.c:913 zt_open: Unable to open
'/dev/zap/pseudo': No such file or directory
Oct 25 18:16:13 ERROR[12281]: chan_zap.c:7396 chandup: Unable to dup
channel: No such file or directory
Oct 25 18:16:13 WARNING[12281]: app_meetme.c:460 build_conf: Unable to
open pseudo channel - trying device
Oct 25 18:16:13 WARNING[12281]: app_meetme.c:463 build_conf: Unable to
open pseudo device
 -- Playing 'conf-invalid' (language 'en')
Oct 25 18:16:17 NOTICE[12281]: app_cbmysql.c:393 cb_exec: enterConf: -1
 == Spawn extension (from-sip, 19001, 3) exited non-zero on
'SIP/9001-08f8d7e0'



Please Help me.

-- 

  

  Thanks  Regards, 
  Ravi
Prakash Sunkara


  
  
  M:+91
9985077535
O:+91 40 23114549
F:+91 40 40208727 


  
  [EMAIL PROTECTED]
www.hyperion-tech.com
   
  

  




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Re: [asterisk-users] Choice of soundfile format

[Noah Miller]

What soundfile format, is the one that uses least transcoding

during playback?

As I can see, I can choose wav or gsm. What sucks least cpu power, during
playback to example a Zap channel? I would guess wav, but is this correct?


When you get down to it, the asterisk native format is slinear.
Fortunately, you're in luck as Kristian Kielhofner did the asterisk
community a big favor and had Alison re-record all the asterisk
sounds, and he put them in slinear format.  You can find them on the
astlinux website (there's other formats, too):

http://www.astlinux.org/index.php?option=com_contenttask=viewid=38Itemid=43

- Noah



On 10/25/06, Matthew Rubenstein [EMAIL PROTECTED] wrote:

What's the native soundfile format for SIP? Any idea which soundfile
takes the least CPU for mixing together in conferences?

How about whether the CPU load for conferencing native data is
greater/less than the CPU load for transcoding non-native data that is
CPU lighter in the conference mixing phase?


On Wed, 2006-10-25 at 04:19 -0700,
[EMAIL PROTECTED] wrote:
 Date: Wed, 25 Oct 2006 10:29:32 +0100
 From: Conrad Wood [EMAIL PROTECTED]
 Subject: Re: [asterisk-users] Choice of soundfile format
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 asterisk-users@lists.digium.com
 Message-ID: [EMAIL PROTECTED]
 Content-Type: text/plain; charset=UTF-8

 On Wed, 2006-10-25 at 11:24 +0200, Jon Schpzinsky wrote:
  Hello
 
  What soundfile format, is the one that uses least transcoding during
 playback?
  As I can see, I can choose wav or gsm. What sucks least cpu power,
 during playback to example a Zap channel? I would guess wav, but is
 this correct?

 The one that is encoded in the same codec as the codec of the channel.
 On zap it's often alaw or ulaw so you can encode your files like that.
 You can encode the same file with different codecs and save it with
 different extensions (matching the codec) and asterisk will pick the
 most suitable one.
 If the channel is gsm, a gsm encoded file would be most efficient, as
 it
 doesn't need transcoding at all.

 Conrad
--

(C) Matthew Rubenstein

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Re: [asterisk-users] Re: Meetme... No channel type registered for 'zap'

[Tzafrir Cohen]

On Wed, Oct 25, 2006 at 10:06:02AM -0600, Douglas Garstang wrote:
  -Original Message-
  From: Tony Mountifield [mailto:[EMAIL PROTECTED]
  Sent: Wednesday, October 25, 2006 1:26 AM
  To: asterisk-users@lists.digium.com
  Subject: [asterisk-users] Re: Meetme... No channel type registered for
  'zap'
  
  
  In article 
  [EMAIL PROTECTED],
  Douglas Garstang [EMAIL PROTECTED] wrote:
   Kristian,

   I don't have any zap hardware What do I put in 
  zaptel.conf if I don't have any hardware?
   On some other systems we have, with chan_zap not loaded, 
  and no zaptel.conf (running
   1.2.9.1), meetme runs fine. This system with the problem 
  has 1.2.12.1. I wonder if something
   was changed?
  
  Doug, it sounds to me like you don't have the /dev/zap device files.
  
  Do you have the file /etc/udev/permissions.d/zaptel.permissions and
  /etc/udev/rules.d/zaptel.rules installed?
 
 Tony, I don't have /etc/udev/permissions.d/, but I do have the other file.
 
 demeter:(acd1)ipt # ls -l /etc/udev/rules.d/zaptel.rules
 -r--r--r--  1 root root 498 Oct 24 15:50 /etc/udev/rules.d/zaptel.rules

And its contents is?

But do you actually have the channels? Anything in /dev/zap ? Anything
in /sys/class/zaptel ? Specifically pseudo/zapseudo .

-- 
Tzafrir Cohen   iax:[EMAIL PROTECTED]/tzafrir
icq#16849755   mailto:[EMAIL PROTECTED] 
+972-50-7952406  jabber:[EMAIL PROTECTED]
 http://www.xorcom.com 
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[asterisk-users] Asterisk Manager

[Maps]

Dear Friends and Supporters!

I try to write a php application to monitor the 
asterisk, but when I open the .php to access to asterisk to retrieve the 
information about the queues status, sip show peers, realtime mysql status 
etc... However, It just return to me "Unable to connect to remote 
asterisk (does /var/run/asterisk.ctl exist?)"

Asterisk is current running with the a file in 
/var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the 
owner of the folder /var/run, and start asterisk with user is asterisk. 
HTTPD is run under asterisk user. My manager.conf has an 
entry.
[admin]secret = 
passworddeny=0.0.0.0/0.0.0.0permit=127.0.0.1/255.255.255.0read = 
system,call,log,verbose,command,agent,userwrite = 
system,call,log,verbose,command,agent,user
However, my php still unable to retrieve the 
information for asterisk. 
Did I miss somethings?

Your help would be very appreciated!

Regards,

Lan
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Re: [asterisk-users] Maximum talktime in a queue?

[Lenz]

Hi Raj,
if you use Local channels for agents (or callback agents), you can easily  
do this in the Dial() command after the Local channel is called. Of course  
your clients may get a bit angry at being disconnected, it is usually  
better to jave each agent stay aware od the call length and occasionally  
tolerate longer calls :)

Just my $0.02
l.


On Wed, 25 Oct 2006 15:06:35 +0200, Rajkumar S  
[EMAIL PROTECTED] wrote:



Hi,

Is it possible to define maximum talk time in a queue? ie any one who
joins a queue should not be able to talk more than say 5 minutes to
the agent.

raj




--
Loway Research - Home of QueueMetrics
http://queuemetrics.loway.it
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[asterisk-users] Re: Dynamic Codec Selection

[Martin Joseph]

On 2006-10-24 06:44:01 -0700, Wildheart
[EMAIL PROTECTED] said:


Hi,

Does anyone know a what to use a different codec for calls which a

re

handset to handset (eg, G711) then when we have calls to the out side
world (via an asterisk server) to use a different codec(eg, G729)?
snip



I responded:
yes, this is simple,  just make it so the extensions allow both g729
and ulaw, and set your outside world is g729.
On 2006-10-25 03:31:39 -0700, Wildheart 
[EMAIL PROTECTED] said:



Hi Marty,

   By the outside world, I mean the PSTN connection. I am still intereste d
in how you would set this up. Can you paste in a sample config?


One internal phone from SIP.conf:

;
; SIP entry for users test rig
[2004]
type=friend
secret=footest
dtmfmode=inband  ; my stupid PSTN gateway doesn't like rfc2833
auth=md5
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=g729
context=autocontext
callerid=Alton Wireless Phone 2004

Another internal extension

; IAX entry for user karma
[3000]
type=friend
secret=testfoo
auth=md5
host=dynamic
disallow=all
allow=ulaw
allow=g729
context=karma
callerid=Karma206500

Ok, now these two extensions when one calls the other should use uLaw.

Now here is my extension for my PSTN gateway:


;
; SIP entry for user (FXO)
[2003]
type=friend
secret=testPSTN
dtmfmode=inband
auth=md5
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=g729
context=autocontext
callerid=Alton Qwest Line2065551183

Depending how your PSTN is setup the last bit could be quite different, 
but the premise is the same.  Since the PSTN only allows g729, this 
will force other connections to that also.  Of course you need to be 
sure your devices support this, or else you will need to buy licenses 
for G729 to transcode, which is also a significant hit for CPU.


Further more, this only makes sense to do if your PSTN calls are 
being terminated by someone OFF your local network.  If your PSTN calls 
(like mine) are being routed to a local gateway, then using ulaw should 
be ok also (it's your network, make it work!).



Marty



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RE: [asterisk-users] Voicemail help

[Ward, Bill]

That Wiki covers CCM4 and my company doesnt have the cash to upgrade to that 
yet.  I have to stick with H323.  I actually started from scratch and went to 
the 1.2 version of Asterisk. 

-Original Message-
From: [EMAIL PROTECTED] on behalf of Noah Miller
Sent: Wed 10/25/2006 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Voicemail help
 
Hi Bill -

 I would like to setup Asterisk for voicemail with CallManager 3.3(5).  I
 would like to know what would be the best Distro of Linux to use and
 version, what version of Asterisk works best to interact with CallManager,
 and what H323 ChannelType works.  As you probably read in another thread I
 tried FC5 with Asterisk 1.4 and OOH323 (included with the addons package).
 This doesn't seem to work to well, as somewhere along the line either CCM or
 OOH323 is disconnecting the call as soon as the playback application is run.

At the present time, I would not use Asterisk 1.4, especially if you
plan on using this in production.  It's definitely still a beta
release.  1.2.12.1 should work for what you're trying to do,
especially with any version of FC (but FC4 or FC5 are probably safest
for right now).  Have you taken a look at the WIKI article:

http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration


- Noah
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RE: [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323

[Ward, Bill]

I don't have any calling search spaces defined.  

-Original Message-
From: [EMAIL PROTECTED] on behalf of Pavel Jezek
Sent: Wed 10/25/2006 3:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323
 
Did you apply correct calling search space on callmanager gateway 
configuration page for incomming calls from asterisk to callmanager?
imho, oh323 is obsolete/unmaintained, I'm using original chan_h323 with 
callmanager 4.1 and it working fine (including dtmf),
ooh323 is probably perspektive and working also, but currently is not 
working dtmf between callmanager and asterisk (ooh323)
see: http://bugs.digium.com/view.php?id=8191
PJ


Ward, Bill wrote:
 I started from scratch again. This time i went with oh323 instead of ooh323.  
 I still get the same issue but this time it says call refused by remote host. 
  It doesn't explain why though.  I can't understand the CCM trace logs to see 
 why it might be refusing the connection.

 -Original Message-
 From: [EMAIL PROTECTED] on behalf of Ward, Bill
 Sent: Tue 10/24/2006 10:57 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323
  

 I have tried GSL and ulaw.  tried one at a time.  tried both and even tried 
 g729 and g7231.  still same issue.  Unless i am thinking of something 
 different


 -Original Message-
 From: [EMAIL PROTECTED] on behalf of Michael Araba
 Sent: Tue 10/24/2006 10:36 PM
 To: asterisk-users@lists.digium.com
 Subject: RE: [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323
  
 I have experience problems like this in a different scenario. It is
 usually due to codec translation problem.

 What is the default codec set on CCM for the IP Phone and the default
 set in Asterisk? Make sure the defaults are the same. Try G.711

 Michael

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RE: [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323

[Dan Austin]

PJ Wrote:
 Did you apply correct calling search space on callmanager 
 gateway configuration page for incomming calls from asterisk
 to callmanager?  imho, oh323 is obsolete/unmaintained, I'm using
 original chan_h323 with callmanager 4.1 and it working fine 
 (including dtmf), ooh323 is probably perspektive and working also,
 but currently is not working dtmf between callmanager and asterisk
 (ooh323) see: http://bugs.digium.com/view.php?id=8191
 PJ
PJ is correct the the most likely cause of call failures from
Asterisk to CCM is an improper Calling Search Space on CCM.  It
could also be a case that the number format that Asterisk is
sending to CCM is not correct, if CCM expects four digit extensions
Asterisk should not send six digit extensions.

 Ward, Bill wrote:
 I started from scratch again. This time i went with oh323 instead 
 of ooh323.  I still get the same issue but this time it says call 
 refused by remote host.  It doesn't explain why though.  I can't 
 understand the CCM trace logs to see why it might be refusing the 
 connection.

On the DTMF issues with chan_ooh323, I've used the channel with
CCM 4.0, 4.1 and 5.0 with no issues.  At one point I tested all
of the DTMF methods ooH323 provides.  CCM definately works with
h245signal, and in the past has worked with q931keypad and 
h245alphanumeric.  Cisco's support of RFC2833 in CCM has not
been impressive.  It works OK as of CCM 5.0 with SIP, but still
not so solid with H323.

Dan
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Re: [asterisk-users] Choice of soundfile format

[Conrad Wood]

On Wed, 2006-10-25 at 12:15 -0400, Noah Miller wrote:
  What soundfile format, is the one that uses least transcoding
 during playback?
  As I can see, I can choose wav or gsm. What sucks least cpu power, during
  playback to example a Zap channel? I would guess wav, but is this correct?
 
 When you get down to it, the asterisk native format is slinear.
 Fortunately, you're in luck as Kristian Kielhofner did the asterisk
 community a big favor and had Alison re-record all the asterisk
 sounds, and he put them in slinear format.  You can find them on the
 astlinux website (there's other formats, too):
 
 http://www.astlinux.org/index.php?option=com_contenttask=viewid=38Itemid=43
 
 - Noah
 

As I understand it, if you have a channel that has a given codec the
least amount of cpu power is required if the voiceprompt is recorded in
that same codec because then asterisk doesn't transcode.
Slinear is good, because you can re-encode them without loss of quality.

Conrad

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[asterisk-users] Re: Choice of soundfile format

[Martin Joseph]

On 2006-10-25 08:14:56 -0700, Matthew Rubenstein [EMAIL PROTECTED] said:


What's the native soundfile format for SIP?
??? I think you might need to do some research (the above is a nonsense 
question I think).

 Any idea which soundfile
takes the least CPU for mixing together in conferences?
Probably slinear for prerecorded stuff, as that will only have to 
transcoded one way (the direction of your output devices). Unless all 
your devices are using ulaw or alaw, then the previous posters advice 
is correct.


How about whether the CPU load for conferencing native data is
greater/less than the CPU load for transcoding non-native data that is
CPU lighter in the conference mixing phase?

Transcoding is a bigger hit then mixing as i understand it.

If all the conference members are using ulaw for example, then having 
the playback material encoded in ulaw is the big winner.  If there are 
different codecs connecting, then there is a lot of 
decoding/mixing/recoding that will need to occur.


Marty


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RE: [asterisk-users] Re: Meetme... No channel type registered for'zap'

[Douglas Garstang]

 -Original Message-
 From: Tzafrir Cohen [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, October 25, 2006 10:18 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] Re: Meetme... No channel type registered
 for'zap'
 
 
 On Wed, Oct 25, 2006 at 10:06:02AM -0600, Douglas Garstang wrote:
   -Original Message-
   From: Tony Mountifield [mailto:[EMAIL PROTECTED]
   Sent: Wednesday, October 25, 2006 1:26 AM
   To: asterisk-users@lists.digium.com
   Subject: [asterisk-users] Re: Meetme... No channel type 
 registered for
   'zap'
   
   
   In article 
   [EMAIL PROTECTED],
   Douglas Garstang [EMAIL PROTECTED] wrote:
Kristian,
 
I don't have any zap hardware What do I put in 
   zaptel.conf if I don't have any hardware?
On some other systems we have, with chan_zap not loaded, 
   and no zaptel.conf (running
1.2.9.1), meetme runs fine. This system with the problem 
   has 1.2.12.1. I wonder if something
was changed?
   
   Doug, it sounds to me like you don't have the /dev/zap 
 device files.
   
   Do you have the file 
 /etc/udev/permissions.d/zaptel.permissions and
   /etc/udev/rules.d/zaptel.rules installed?
  
  Tony, I don't have /etc/udev/permissions.d/, but I do have 
 the other file.
  
  demeter:(acd1)ipt # ls -l /etc/udev/rules.d/zaptel.rules
  -r--r--r--  1 root root 498 Oct 24 15:50 
 /etc/udev/rules.d/zaptel.rules
 
 And its contents is?

Contents are:

demeter:(acd1)ipt # cat /etc/udev/rules.d/zaptel.rules
# zaptel devices with ownership/permissions for running as non-root
KERNEL==zapctl, NAME=zap/ctl, OWNER=asterisk, GROUP=asterisk, 
MODE=0660
KERNEL==zaptimer, NAME=zap/timer, OWNER=asterisk, GROUP=asterisk, 
MODE=0660
KERNEL==zapchannel, NAME=zap/channel, OWNER=asterisk, GROUP=asterisk, 
MODE=0660
KERNEL==zappseudo, NAME=zap/pseudo, OWNER=asterisk, GROUP=asterisk, 
MODE=0660
KERNEL==zap[0-9]*, NAME=zap/%n, OWNER=asterisk, GROUP=asterisk, 
MODE=0660

 But do you actually have the channels? Anything in /dev/zap ? Anything
 in /sys/class/zaptel ? Specifically pseudo/zapseudo .

Do I have the channels? No, I don't think so. I don't have any zap hardware 
installed. That's why I am using ztdummy.

demeter:(acd1)ipt # ls -l /dev/zap
total 0
crw-rw  1 root root 196, 254 Oct 24 16:01 channel
crw-rw  1 root root 196,   0 Oct 24 16:01 ctl
crw-rw  1 root root 196, 255 Oct 24 16:01 pseudo
crw-rw  1 root root 196, 253 Oct 24 16:01 timer

demeter:(acd1)ipt # ls -l /sys/class/zaptel
total 0
drwxr-xr-x  2 root root 0 Oct 24 16:01 zapchannel
drwxr-xr-x  2 root root 0 Oct 24 16:01 zapctl
drwxr-xr-x  2 root root 0 Oct 24 16:01 zappseudo
drwxr-xr-x  2 root root 0 Oct 24 16:01 zaptimer

Doug
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Re: [asterisk-users] Simple example for call transfer.

[Brian Candler]

On Wed, Oct 25, 2006 at 07:14:23PM +0300, Jonson Player wrote:
i hev a subscription to a international voip provider and I want all
calls for numbers _001xx to go through my voip provider. I
tried many settings in sip.conf , extensions.conf and iax.conf. Please
give me some simple example for how can i transfer the specified calls
to my external voip provider. What may I put and where in witch file.

How about showing us:
(a) what you tried, i.e. the contents of those files;
(b) what happened (in terms of what the call did, and what Asterisk
displayed on the console); and
(c) what you think should have happened.

Then we might be able to help you - and you might also solve the problem
yourself in the process. See
http://www.catb.org/~esr/faqs/smart-questions.html#intro for more
suggestions on how to ask questions in a useful way.

In any case, a good read is Asterisk: The Future of Telephony,
downloadable for free from www.asteriskdocs.org. The chapters on dialplans
(ch 5 and 6) should tell you what you need.

Regards,

Brian.
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Re: [asterisk-users] make menuselect question- Module Embedding

[Tim Panton]

On 24 Oct 2006, at 01:05, Carla Schroder wrote:

What does option '11. Module Embedding' do in Asterisk 1.4? The  
default is

none of them are selected:

 [ ] 1.  apps
 [ ] 2.  cdr
 [ ] 3.  channels
 [ ] 4.  codecs
 [ ] 5.  formats
 [ ] 6.  funcs
 [ ] 7.  pbx
 [ ] 8.  res



It allows you to select modules you want to have statically compiled
into the asterisk binary, as opposed to having them as dynamically  
loaded

modules.

According to the presentation at astricon there are a couple of  
reasons you might want

to do this:
1) because your platform does not support dynamic loading :-)
2) because you are debugging multiple versions and you want
certainty about which version you are running.

I'm sure there are other reasons folks can come up with.

Tim.


Tim Panton

www.mexuar.com



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[asterisk-users] Re: Meetme... No channel type registered for 'zap'

[Tony Mountifield]

In article [EMAIL PROTECTED],
Douglas Garstang [EMAIL PROTECTED] wrote:
 Tony Mountifield [mailto:[EMAIL PROTECTED] said:
  
  Doug, it sounds to me like you don't have the /dev/zap device files.
  
  Do you have the file /etc/udev/permissions.d/zaptel.permissions and
  /etc/udev/rules.d/zaptel.rules installed?
 
 Tony, I don't have /etc/udev/permissions.d/, but I do have the other file.
 
 demeter:(acd1)ipt # ls -l /etc/udev/rules.d/zaptel.rules
 -r--r--r--  1 root root 498 Oct 24 15:50 /etc/udev/rules.d/zaptel.rules
 
  
  What Linux distro are you using?
 
 I'm using Gentoo Linux, and have been for a number of months. This is the 
 first time this
 problem has cropped up. If I have ztdummy installed, why do I need the device 
 files? Isn't
 that what ztdummy is supposed to do?

I'm not familiar with Gentoo, so I'm afraid I can only help in general
terms.

In fact I've gone back and re-read your original message and found that
I had misinterpreted it, so I'll start from the beginning again. It's
nothing to do with udev or device files after all.

The messages you mentioned were:

-- Executing Answer(IAX2/xxx.yyy.142.204:4569-2, ) in new stack
-- Executing MeetMe(IAX2/xxx.yyy.142.204:4569-2, |||d) in new stack
-- Playing 'conf-getconfno' (language 'en')
Warning, flexible rate not heavily tested!
Oct 24 16:16:59 WARNING[1732]: channel.c:2597 ast_request: No channel type 
registered for 'zap'
Oct 24 16:16:59 WARNING[1732]: app_meetme.c:465 build_conf: Unable to open 
pseudo channel - trying device
-- Created MeetMe conference 1023 for conference '5000'
-- Playing 'conf-onlyperson' (language 'en')
-- Hungup 'IAX2/xxx.yyy.142.204:4569-2'

What you didn't say was whether the conference worked despite those
messages.

When you create a conference, MeetMe tries to create a full Asterisk
channel for the zaptel pseudo device. The two warnings above indicate
that it was unable to do so, meaning that chan_zap.so is not loaded.
If Meetme fails to create a full asterisk channel, it falls back to
opening a file descriptor on /dev/zap/pseudo directly. That's what the
trying device part in the second message means. It evidently
succeeded, or there would have been a third error message.

If conferences are working ok for you, you can ignore the warnings.
However, certain options such as 'i' will not work, as they rely on the
full Asterisk channel.

The best solution is to make sure that chan_zap was built when you
compiled Asterisk on this box, AND that you don't have an entry in
modules.conf preventing it being loaded (noload=chan_zap.so).

To make sure chan_zap is built, you must have built AND installed zaptel
BEFORE you start to build Asterisk.

Hope this all helps!

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] problem with setting outbound caller id when calling another asterisk

[Chris Mazuc]

Asterisk seems to have a bug which is not letting me set the caller id 
to another peer's caller id.


http://www.mail-archive.com/asterisk-dev@lists.digium.com/msg23230.html

I've sent this to the asterisk-users mailing list, hopefully I get a 
response soon if there is a workaround.


I'm going to see if there is a way to blindly accept calls from a known 
IP address, but I don't think there is a way that would retain CDR 
information.


Chris Mazuc wrote:
I have an asterisk box at a remote location (which I will call remote), 
which registers to my local asterisk box (I'll call that one local), and 
uses that to route calls to the outside world. The problem I am having 
is that the remote location needs to lie about it's callerid sometimes, 
however if I set a callerid that matches the extension of another peer 
that exists, local returns a 403 to remote. I can set the callerid 
to the did and it will work fine, or I can set the callerid to something 
random and it will work fine.


What does * do with the proxy-authorization header, because it seems to 
be ignoring the username part.


Any help is greatly appreciated.

Thanks,
Chris Mazuc

-- SIP read from REMOTE:1025:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP REMOTE:5060;branch=z9hG4bK1757eacd;rport
From: My Name sip:[EMAIL PROTECTED];tag=as4f42dab4
To: sip:[EMAIL PROTECTED]
Contact: sip:[EMAIL PROTECTED]
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username=1XX1205, realm=asterisk, 
algorithm=MD5, uri=sip:[EMAIL PROTECTED], nonce=45a347bc, 
response=934b409f19a0ebf28d1cf266db29f497, opaque=

Date: Tue, 24 Oct 2006 20:26:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 2238 2239 IN IP4 REMOTE
s=session
c=IN IP4 REMOTE
t=0 0
m=audio 15384 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

--- (14 headers 11 lines)---
Using INVITE request as basis request - 
[EMAIL PROTECTED]

Sending to REMOTE : 5060 (NAT)
Found user '1XX1200'
Reliably Transmitting (NAT) to REMOTE:1025:
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 
REMOTE:5060;branch=z9hG4bK1757eacd;received=REMOTE;rport=1025

From: My Name sip:[EMAIL PROTECTED];tag=as4f42dab4
To: sip:[EMAIL PROTECTED];tag=as1f40e0ec
Call-ID: [EMAIL PROTECTED]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:[EMAIL PROTECTED]
Content-Length: 0
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Re: [asterisk-users] Asterisk Manager

[Lacy Moore - Aspendora]

Asterisk is current running with the a file in /var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the owner of the folder /var/run, and start asterisk with user is asterisk. HTTPD is run under asterisk user. My 
manager.conf has an entry.

Check to make sure the file is actually /var/run/asterisk.ctl and not /var/run/asterisk/asterisk.ctl.
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Re: [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI

[Melcon Moraes]

Same here with Brazilian Portuguese. :)

Nicolas S. wrote:
Hi, 


I can help in French translation if needed.
Drop me the procedure to do it. 

Regards 



Le mercredi 25 octobre 2006 à 09:51 +0200, Alex a écrit :

Hi all!

We've released VoiceOne 0.4.0, a web-based and open source solution 
which allows to fully manage an Asterisk service hosted on a LAMP server.


We focused on an charming and overall user-friendly interface. Thanks to 
the authentication based on roles, once configured by a super user, the 
PBX may be easily maintained even by an Asterisk unskilled users.


 From a technical point of view, the application is made up of two 
modules: one for the client - i.e. the user interface - and the other 
for the server. Thanks to the web services provided by the server module 
and the use of a database, VoiceOne may be easily integrated with other 
applications (e.g. CRM software).


The project has grown and has received positive response so far. 
Nowadays there's a little but enthusiastic community of developers, 
supporters and users. Translations in several languages (e.g. English, 
Spanish, Russian, etc.) are already available.


On the project website at http://www.voiceone.it you'll find the online 
demo and the links to download the source files from Sourceforge, as 
well as a support forum.


We would be pleased if you could give it a try and let us know your 
feedback, comments, ideas, or suggestions replying here or posting a 
message on our forum.


Thanks for your kind attention.

Regards,
Alex
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RE: [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323

[Ward, Bill]

Well I seem to have removed my call pattern too many times and now CCM isn't 
routing it anymore.  


-Original Message-
From: [EMAIL PROTECTED] on behalf of Dan Austin
Sent: Wed 10/25/2006 11:46 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323
 
PJ Wrote:
 Did you apply correct calling search space on callmanager 
 gateway configuration page for incomming calls from asterisk
 to callmanager?  imho, oh323 is obsolete/unmaintained, I'm using
 original chan_h323 with callmanager 4.1 and it working fine 
 (including dtmf), ooh323 is probably perspektive and working also,
 but currently is not working dtmf between callmanager and asterisk
 (ooh323) see: http://bugs.digium.com/view.php?id=8191
 PJ
PJ is correct the the most likely cause of call failures from
Asterisk to CCM is an improper Calling Search Space on CCM.  It
could also be a case that the number format that Asterisk is
sending to CCM is not correct, if CCM expects four digit extensions
Asterisk should not send six digit extensions.

 Ward, Bill wrote:
 I started from scratch again. This time i went with oh323 instead 
 of ooh323.  I still get the same issue but this time it says call 
 refused by remote host.  It doesn't explain why though.  I can't 
 understand the CCM trace logs to see why it might be refusing the 
 connection.

On the DTMF issues with chan_ooh323, I've used the channel with
CCM 4.0, 4.1 and 5.0 with no issues.  At one point I tested all
of the DTMF methods ooH323 provides.  CCM definately works with
h245signal, and in the past has worked with q931keypad and 
h245alphanumeric.  Cisco's support of RFC2833 in CCM has not
been impressive.  It works OK as of CCM 5.0 with SIP, but still
not so solid with H323.

Dan
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Re: [asterisk-users] Re: IAX2 goes one way audio when lag gets bad

[Matt]

If the audio is dropping out completely, then I suspect the whole
jitter buffer thing is a red herring (waste of time).

Perhaps it's a nat issue?  What kind of router if any is involved?  I
am reaching here... Also, please do tell us which version of asterisk
you are running...


I apologize.. I thought I told already.  I am running 1.2.6 and have
tried 1.2.12.  At any rate, I believe it is actually the cable modem
connection dropping, and someone from Comcast is coming to look at it
tomorrow.  My question is.. why is the jitterbuffer just dieing?   I
understand there may not be audio if the connection dropped for like 4
or 5 seconds, but shouldn't it pick back up?

Using pingplotter I've determined when they are losing audio, I also
get a red 100% packet loss from their node lasts about 5 seconds
usually, and then the jitterbuffer is dead.  Is this just too much for
the jitterbuffer to handle?  Can't it get back on track at least?
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[asterisk-users] Re: Asterisk Manager

[Maps]

Dear Friends and Supporters!

I try to write a php application to monitor the 
asterisk, but when I open the .php to access to asterisk to retrieve the 
information about the queues status, sip show peers, realtime mysql status 
etc... However, It just return to me "Unable to connect to remote 
asterisk (does /var/run/asterisk.ctl exist?)"

Asterisk is current running with the a file in 
/var/run/asterisk.ctl for the user asterisk. I have set asterisk to be the 
owner of the folder /var/run, and start asterisk with user is asterisk. 
HTTPD is run under asterisk user. My manager.conf has an 
entry.
[admin]secret = 
passworddeny=0.0.0.0/0.0.0.0permit=127.0.0.1/255.255.255.0read = 
system,call,log,verbose,command,agent,userwrite = 
system,call,log,verbose,command,agent,user
However, my php still unable to retrieve the 
information for asterisk. 
Did I miss somethings?

Your help would be very appreciated!

Regards,

Lan
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[asterisk-users] Trixbox installation - ZAP channels becoming upresponsive

[Cory Andrews]

I have a colleague who had an IP PBX solution put in by a reseller and
they are having an issue with their ZAP channels becoming unresponsive.

They are using a Digium TDM2400 Series, all inbound and outbound through
the FXO ports, VOIP is internal only.

Anyone aware of any known issues with Digium/Trixbox ZAP channels going
awol?

Thanks

Cory
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[asterisk-users] Asterisk on Embedded platforms

[Prasad Kandikonda]

We are looking at porting asterisk onto a embedded platform based on IXP or ARM. I would like to know the experiences of anybody who has already ported to these platforms. I am also particularly interested in issues related to performance and scaling on these platforms.Also, is anybody aware of any embedded asterisk products. I know recently Digium announced a platform based on Blackfin.Thanks,  Prasad Kandikonda. 
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[asterisk-users] Asterisk on Embedded platforms

[Prasad Kandikonda]

We are looking at porting asterisk onto a embedded platform based on IXP or ARM. I would like to know the experiences of anybody who has already ported to these platforms. I am also particularly interested in issues related to performance and scaling on these platforms.Also, is anybody aware of any embedded asterisk products. I know recently Digium announced a platform based on Blackfin.Thanks,  Prasad Kandikonda. 
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RE: [asterisk-users] Re: Meetme... No channel type registered for 'zap'

[Douglas Garstang]

 -Original Message-
 From: Tony Mountifield [mailto:[EMAIL PROTECTED]
 Sent: Wednesday, October 25, 2006 11:10 AM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] Re: Meetme... No channel type registered for
 'zap'
 
 
 In article 
 [EMAIL PROTECTED],
 Douglas Garstang [EMAIL PROTECTED] wrote:
  Tony Mountifield [mailto:[EMAIL PROTECTED] said:
   
   Doug, it sounds to me like you don't have the /dev/zap 
 device files.
   
   Do you have the file 
 /etc/udev/permissions.d/zaptel.permissions and
   /etc/udev/rules.d/zaptel.rules installed?
  
  Tony, I don't have /etc/udev/permissions.d/, but I do have 
 the other file.
  
  demeter:(acd1)ipt # ls -l /etc/udev/rules.d/zaptel.rules
  -r--r--r--  1 root root 498 Oct 24 15:50 
 /etc/udev/rules.d/zaptel.rules
  
   
   What Linux distro are you using?
  
  I'm using Gentoo Linux, and have been for a number of 
 months. This is the first time this
  problem has cropped up. If I have ztdummy installed, why do 
 I need the device files? Isn't
  that what ztdummy is supposed to do?
 
 I'm not familiar with Gentoo, so I'm afraid I can only help in general
 terms.
 
 In fact I've gone back and re-read your original message and 
 found that
 I had misinterpreted it, so I'll start from the beginning again. It's
 nothing to do with udev or device files after all.
 
 The messages you mentioned were:
 
 -- Executing Answer(IAX2/xxx.yyy.142.204:4569-2, ) in 
 new stack
 -- Executing MeetMe(IAX2/xxx.yyy.142.204:4569-2, 
 |||d) in new stack
 -- Playing 'conf-getconfno' (language 'en')
 Warning, flexible rate not heavily tested!
 Oct 24 16:16:59 WARNING[1732]: channel.c:2597 ast_request: No 
 channel type registered for 'zap'
 Oct 24 16:16:59 WARNING[1732]: app_meetme.c:465 build_conf: 
 Unable to open pseudo channel - trying device
 -- Created MeetMe conference 1023 for conference '5000'
 -- Playing 'conf-onlyperson' (language 'en')
 -- Hungup 'IAX2/xxx.yyy.142.204:4569-2'
 
 What you didn't say was whether the conference worked despite those
 messages.
 
 When you create a conference, MeetMe tries to create a full Asterisk
 channel for the zaptel pseudo device. The two warnings above indicate
 that it was unable to do so, meaning that chan_zap.so is not loaded.
 If Meetme fails to create a full asterisk channel, it falls back to
 opening a file descriptor on /dev/zap/pseudo directly. That's what the
 trying device part in the second message means. It evidently
 succeeded, or there would have been a third error message.
 
 If conferences are working ok for you, you can ignore the warnings.
 However, certain options such as 'i' will not work, as they 
 rely on the
 full Asterisk channel.
 
 The best solution is to make sure that chan_zap was built when you
 compiled Asterisk on this box, AND that you don't have an entry in
 modules.conf preventing it being loaded (noload=chan_zap.so).
 
 To make sure chan_zap is built, you must have built AND 
 installed zaptel
 BEFORE you start to build Asterisk.
 
 Hope this all helps!

Tony,

Thanks for the reply. chan_zap was built, but I am not loading it. The meetme 
conference works, but user entry/exit is not being announced (that's option 
'i', right?). I tried loading chan_zap, but it complains that I have no 
zaptel.conf file. So, if I have no zap hardware, what should I put in 
zaptel.conf?

Doug.
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[asterisk-users] Multiple queue_log files based on queue - is it possible??

[Christopher Aloi]

Hello List,

Question: Has anyone been able to create multiple queue_log files in
/var/log/asterisk for multiple queues?

We are designing a multi-tenant system and separating the log files
would be useful, instead of dropping all queue actions into one file.

Is it possible this is a user configurable option I am missing?

Cheers,
--
--
Christopher T Aloi
--
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[asterisk-users] Looking for Wireless Heaset for Polycom 501

[Jim Freeze]

Hi

I am looking for a good wirless headset to use with the Polycom Soundpoint 501
phone. I would greatly appreciate hearing from anyone with good experiences
with a specific device.

Thanks

--
Jim Freeze
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Re: [asterisk-users] VoiceOne 0.4.0 released: a new web-based and open source GUI

[Michiel van Baak]

On 09:51, Wed 25 Oct 06, Alex wrote:
snip
/snip

Any plans to support multiple virtual pbx-en on one asterisk
instance ?
That's something almost no webbased tool implements. It's
all one asterisk, one pbx while asterisk is very capable
of virtualhosting PBX-en on one instance.

Would be a great feature :)
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

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[asterisk-users] Add second account to Xlite 3.0

[Tielin Xu]

Hi List:

I have been testing Xlite 2.0 and 3.0. The Xlite 2.0 is slow on
initiate time, but I can add second sip proxy account, which is very
critical to my testing. I installed Xlite 3.0, which I could not add
second account on  SIP account settings. After I add the first one, the
Add button is grayed out, I can not do anything to add extra account.
Does anyone know how to get second account added in?
 
Many thanks,

Tielin

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Re: [asterisk-users] Multiple queue_log files based on queue - is it possible??

[BJ Weschke]

On 10/25/06, Christopher Aloi [EMAIL PROTECTED] wrote:

Hello List,

Question: Has anyone been able to create multiple queue_log files in
/var/log/asterisk for multiple queues?

We are designing a multi-tenant system and separating the log files
would be useful, instead of dropping all queue actions into one file.

Is it possible this is a user configurable option I am missing?


No It isn't user configurable. If you want to split them externally
you can, but there nothing native to do that at this point.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/
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[asterisk-users] No Authority Found

[Andrew Joakimsen]

In over three years of using Asterisk in the lab and also in real-world deployments and supporting other Asterisk users, the single most common problem I have encountered and seen others encounter is the message No Authority Found and the inability to call between machines when using IAX. This is always a configuration problem which is solved after some tinkering with the 
iax.conf, however I still do not understand the behaviour fully, every time I am able to get it to work it is by pure luck, not by a set formula Even using the example iax.conf files I have seen the no authority found messages, and what puzzles me even more is that I can't find a detailed explanation of this error. Is there any resource with a detailed explanation of No Authority Found messages and how to troubleshoot them? Maybe it relates to the second part of my inquiry?
Another thing is my understanding of the peer, user and friend. I thought that a peer can only receive calls from either a user or a friend, a user sends calls to a peer or friend and a friend is both a peer and a user, however in my production machine I have the following configured:
register = user:[EMAIL PROTECTED][provider-ingress]type=peerusername=userhost=providerip[provider-egress]type=userusername=userhost=provideripThat's the basics, user, password, providerip are all the same. Now when the provier sends us a call, it always comes in through (according to the CLI and CDR) provider-egress. How can this be if a user is supposed to send calls, not receive them??

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Re: [asterisk-users] Re: Asterisk Manager

[Michiel van Baak]

On 13:12, Wed 25 Oct 06, Maps wrote:
 Dear Friends and Supporters!
 
 I try to write a php application to monitor the asterisk, but when I open the 
 .php to access to asterisk to retrieve the information about the queues 
 status, sip show peers, realtime mysql status etc...  However, It just return 
 to me Unable to connect to remote asterisk (does /var/run/asterisk.ctl 
 exist?)
 
 Asterisk is current running with the a file in /var/run/asterisk.ctl for the 
 user asterisk.  I have set asterisk to be the owner of the folder /var/run, 
 and start asterisk with user is asterisk.  HTTPD is run under asterisk user.  
 My manager.conf has an entry.
 [admin]
 secret = password
 deny=0.0.0.0/0.0.0.0
 permit=127.0.0.1/255.255.255.0
 read = system,call,log,verbose,command,agent,user
 write = system,call,log,verbose,command,agent,user
 
 However, my php still unable to retrieve the information for asterisk.  
 Did I miss somethings?

How are you connecting to asterisk?
Maybe you can paste some code so we can actually see why it
is not working.

-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

Why is it drug addicts and computer afficionados are both called users?

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Re: [asterisk-users] Looking for Wireless Heaset for Polycom 501

[Andrew Joakimsen]

I've used the Plantronics ones, similar to these: http://www.plantronics.com/north_america/en_US/products/cat29880043/cat29880058/prod5510016
 and they work very well with the headset lifter, The range is pretty good too.However there are more elegant and complete solutions, with those headsets you need to be by the phone to see who is calling and to use the keypad.
On 10/25/06, Jim Freeze [EMAIL PROTECTED] wrote:
HiI am looking for a good wirless headset to use with the Polycom Soundpoint 501phone. I would greatly appreciate hearing from anyone with good experienceswith a specific device.Thanks--
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Re: [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323

[Pavel Jezek]

Dan, can you supply your ooh323.conf for me? I would like resolve my 
issue with not recognizing dtmf by ooh323 from callmanager
my ooh323 is quite simple, also on callmanager config page for gateway 
to asterisk is nothing special, no faststart, no mtp; ccm v4.1.3sr3a



[general]
disallow=all
allow=alaw
;faststart=no
;h245tunneling=no

[ccm]
type=peer
ip=192.168.40.7
port=1720
;dtmfmode=rfc2833
;dtmfmode=h245signal
;h245tunneling=no
;faststart=no



Dan Austin wrote:


On the DTMF issues with chan_ooh323, I've used the channel with
CCM 4.0, 4.1 and 5.0 with no issues.  At one point I tested all
of the DTMF methods ooH323 provides.  CCM definately works with
h245signal, and in the past has worked with q931keypad and 
h245alphanumeric.  Cisco's support of RFC2833 in CCM has not

been impressive.  It works OK as of CCM 5.0 with SIP, but still
not so solid with H323.


  

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Re: [asterisk-users] Add second account to Xlite 3.0

[Brian Candler]

On Wed, Oct 25, 2006 at 11:37:35AM -0700, Tielin Xu wrote:
 I have been testing Xlite 2.0 and 3.0. The Xlite 2.0 is slow on
 initiate time, but I can add second sip proxy account, which is very
 critical to my testing. I installed Xlite 3.0, which I could not add
 second account on  SIP account settings. After I add the first one, the
 Add button is grayed out, I can not do anything to add extra account.
 Does anyone know how to get second account added in?

http://support.counterpath.com/viewtopic.php?t=7919

And see also
http://www.xten.com/index.php?menu=Productssmenu=compare
(scroll down to where it says Multiple Accounts)
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Re: [asterisk-users] Multiple queue_log files based on queue - is it possible??

[Michiel van Baak]

On 14:29, Wed 25 Oct 06, Christopher Aloi wrote:
 Hello List,
 
 Question: Has anyone been able to create multiple queue_log files in
 /var/log/asterisk for multiple queues?
 
 We are designing a multi-tenant system and separating the log files
 would be useful, instead of dropping all queue actions into one file.
 
 Is it possible this is a user configurable option I am missing?

Asterisk is not able to do this itself.
It should be easy to write a shellscript or something to do
the splitting for you. The queuename is the 3rd field in the
logfile so this should be giving you the queuename of a
line:
cut -d | -f 3

good luck
-- 

Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD

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Re: [asterisk-users] Looking for Wireless Heaset for Polycom 501

[Jim Rice]

On Wed, 2006-10-25 at 13:31 -0500, Jim Freeze wrote:
 Hi
 
 I am looking for a good wirless headset to use with the Polycom Soundpoint 501
 phone. I would greatly appreciate hearing from anyone with good experiences
 with a specific device.
 
 Thanks

We've used the Plantronics CS50 wireless Headset with the HL10 Handset
Lifter.  About $240.

The handset lifter leaves a lot to be desired with the 501.
It lifts the handset off the cradle, but doesn't completely hang it up
properly.  We've had to place items under the phone to tilt it back.

Other than that, the headset is great.

Jim

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RE: [SPAM] - [asterisk-users] Looking for Wireless Heaset for Polycom 501 - Email found in subject

[Cory Andrews]

I like the Plantronics CS55/HL10, it's a DECT Wireless boom headset with
a lifter kit for the phone, works like a charm, great range.

-Cory 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Freeze
Sent: Wednesday, October 25, 2006 2:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [SPAM] - [asterisk-users] Looking for Wireless Heaset for
Polycom 501 - Email found in subject

Hi

I am looking for a good wirless headset to use with the Polycom
Soundpoint 501 phone. I would greatly appreciate hearing from anyone
with good experiences with a specific device.

Thanks

--
Jim Freeze
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Re: [asterisk-users] Dynamic Codec Selection

[Andrew Joakimsen]

In your configuration files, for the providers, put:disallow=allallow=g729For the phones leave them as it is, they might use G711 between the phones and the server, but if its a local lan it really wont matter unless its not well designed and managed.
On 10/24/06, Wildheart [EMAIL PROTECTED] wrote:
Hi,Does anyone know a what to use a different codec for calls which arehandset to handset (eg, G711) then when we have calls to the out sideworld (via an asterisk server) to use a different codec(eg, G729)?
The idea is to reduce the bandwidth to the server for the majority ofcalls, but get good quality on internal calls.With thanks, Tim___
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Re: [asterisk-users] Looking for Wireless Heaset for Polycom 501

[Jim Freeze]

On 10/25/06, Andrew Joakimsen [EMAIL PROTECTED] wrote:

I've used the Plantronics ones, similar to these:
http://www.plantronics.com/north_america/en_US/products/cat29880043/cat29880058/prod5510016
and they work very well with the headset lifter, The range is pretty good
too.

However there are more elegant and complete solutions, with those headsets
you need to be by the phone to see who is calling and to use the keypad.


What are these other, more elegant complete solutions you are talking about?

--
Jim Freeze
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RE: [asterisk-users] Callmanager 3.3(5) and Asterisk with ooh323

[Dan Austin]

PJ Wrote:
 Dan, can you supply your ooh323.conf for me? I would like resolve my 
 issue with not recognizing dtmf by ooh323 from callmanager
 my ooh323 is quite simple, also on callmanager config page for gateway

 to asterisk is nothing special, no faststart, no mtp; ccm v4.1.3sr3a

There's nothing secret about it, but I think I can skip that if the
settings below are what you are using-

 [general]
 disallow=all
 allow=alaw
 ;faststart=no
 ;h245tunneling=no

 [ccm]
 type=peer
 ip=192.168.40.7
 port=1720
 ;dtmfmode=rfc2833
 ;dtmfmode=h245signal
 ;h245tunneling=no
 ;faststart=no

I tend to use type=friend since I want calls in both directions.
From the list above, you have not set the dtmfmode in [general],
which is where I set it, or in the peer.  The sample ooh323.conf
shows that the default is RFC2833, which does not work with CCM.
Either uncomment ;dtmfmode=h245signal in the peer, or uncomment it
and move it to [general].



 Dan Austin wrote:

 On the DTMF issues with chan_ooh323, I've used the channel with
 CCM 4.0, 4.1 and 5.0 with no issues.  At one point I tested all
 of the DTMF methods ooH323 provides.  CCM definately works with
 h245signal, and in the past has worked with q931keypad and 
 h245alphanumeric.  Cisco's support of RFC2833 in CCM has not
 been impressive.  It works OK as of CCM 5.0 with SIP, but still
 not so solid with H323.


Dan
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