Re: [Asterisk-Users] SPAM Notice

[Benjamin on Asterisk Mailing Lists]

On Tue, 19 Oct 2004 10:34:02 +1000, Adam Goryachev
[EMAIL PROTECTED] wrote:
 Just a heads-up that asterisk is getting a mention in spam now... oh,
 and make sure you NEVER EVER buy anything from this company.
 
 [SNIP]
 NEWS: VocalScape Inc. Announces DELETED for Asterisk IP PBX Users.

As marketeers say: Any news is good news. So, this may well be good news.

Just make sure you have your mailboxes protected or use a webmail
account to participate in the mailing lists and be prepared for the
wave of Asterisk related spam that might be coming our way. Thanks for
the warning.

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
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[Asterisk-Users] SIP video support problem

[Jacky]

Hi, List

I have used Windows Messenger for video call via Asterisk Server.
But Windows Messenger function can't match our requirement.
We are looking more SIP Video Phone can use under Asterisk.

Any suggestion for video Phone(Software or Hardware)?
Also I still have an question about video/audio codec?
Does Asterisk only bypass the codec frame when call is not softswitch?
Can * handle mpeg4 or other codec when video client use this codec?

Thanks,


-- 
Jacky
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Re: [Asterisk-Users] Re: Dial group continues to ring after answer -SNOM phones and solution

[Sven Fischer (support)]

Hi,

On Monday 11 October 2004 19:12, Dave Cotton wrote:
 On Mon, 2004-10-11 at 11:51 -0500, Mike Meyer wrote:
  Someone pointed me here 
  
  http://www.snom.com/downloads/share

  http://www.snom.com/download/share

  !
  That where the SNOM support team sent me. Seems that they may be
  suggesting a different process or URL do update from. My concern is
  whether the latest version 3.54 has been tested and is an official
  release. I hate to put something out that hasn't been through a
  sufficient QA process. I don't want to risk getting my user's mad at me
  with a bad version of software.

 I've been working though the 3.5x series and haven't noticed any real
 nasties yet.

 Out of interest has anyone worked out how to use the Action URL
 settings?

Some few specific events on the phone can trigger web get requests to the 
configured URLs. Like lifting the handset is triggering there is some action 
going on on the phone etc...  

Regards,

Sven
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Re: [Asterisk-Users] Re: Dial group continues to ring after answer -SNOM phones and solution

[Sven Fischer (support)]

Hi,

no, it is

 http://www.snom.com/download/share

!!!

Sven

On Monday 11 October 2004 17:18, Alex Barnes wrote:
 Someone pointed me here 

 http://www.snom.com/downloads/share (had to guess at URL as the Snom
 site appears to be down or uber slow but if that's not it its damn close

 :-P )

 Which lists all versions of firmware for all their phones.  Handy if you
 have a specific version in mind but don't know the correct URL.  Tho I
 haven't had problems with the auto-update so far.

 HTH

 alex

 -Original Message-
 From: Mike Meyer [mailto:[EMAIL PROTECTED]
 Sent: 11 October 2004 16:12
 To: Asterisk Users Group
 Subject: [Asterisk-Users] Re: Dial group continues to ring after answer
 -SNOM phones and solution


 Asterisk Users;

  Just wanted to let you know I fixed my problem.

 To follow up on my own testing of the situation, I find that the
 continued ringing after pickup only occured on the SNOM phones in the
 group. The Grandstream phones stop ringing when another phone picks up.

 Having turned on SIP debugging I have verified that the cancel message
 is sent to the SNOM phone (and others in the group) when one of the
 group phones is picked up, as expected.

 It appears that the SNOMs don't handle the cancel message the same as
 the Grandstream. I was using SIP 2.03o firmware on the SNOM which is the
 latest official release.

 It seems that these phones even though they are set to do automatic
 update, they do not. Or perhaps that capability was broken in the
 firmware version I had last updated to.

 THE SOLUTION:
 To remedy the problem I upgraded to version 3.52 beta version. Also
 2.04g fixes this problem as well.

 I had to create my own internal TFTP server and flash update to 3.52.
 The standard update process did not work to go beyond 2.03y or 2.04g. I
 tried 2.05e  f and these would never come out of boot.

 MORAL TO THE STORY: Keep your SIP phone firmware up to date.

 SNOM support is telling me to upgrade to 3.54. I don't see this one
 listed on the standard update URL. I am a little leery about moving to
 that one.

 Now to upgrade my GrandStream's. They seem to be stuck at an old version
 as well.

 Thanks,
 Mike Meyer

 On Tue, 2004-10-05 at 16:47, Mike Meyer wrote:
  Asterisk Users:
 
   We have our * dial plan set up to ring 5 phones in the office

 and it

  delivers the call to the first that picks up their receiver.
   The problem is that after the pickup, everyone else's SIP phone

 keeps

  ringing at least once and sometimes twice. This interferes with the
  conversation, while others pick up the phone and get nothing.
 
   Does anyone else have similar problems or have a solution to

 stop the

  ring once answered? My dial statement looks like the following and has
 
  a timeout of 15 seconds.
 
  exten = MainTeam,1,Dial(${MainTeamChannels},15,tr)
  exten = MainTeam,2,Voicemail(u${MainTeam_EXT})
  ...
  note the variables MainTeamChannels define the SIP phone channels
  defined in sip.conf and MainTeam_EXT is the voicemail box for this
  group extension.
 
   As an alternative, I am considering doing a round robin on a

 call

  group or pickup group and implementing call pickup.
 
  Any ideas welcome.
 
  Mike Meyer

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 Dear Friends of Ubiquity Software:

 As you may have noticed, Ubiquity Software began using the web domain
 ubiquity.com earlier this year in addition to the previously established
 ubiquity.net for our website and email communications to you.  However,
 since that time, a dispute has emerged with respect to actual ownership of
 the ubiquity.com domain.

 As an international software company founded over decade ago, you can
 always reach Ubiquity Software under the website www.ubiquity.net
 http://www.ubiquity.net/  and via email at @ubiquity.net.  However, we
 have also chosen to expand our domain to the more specific
 www.ubiquitysoftware.com http://www.ubiquitysoftware.com/  for web and
 @ubiquitysoftware.com for email communications.

 Please use either the historical ubiquity.net or begin to use the new
 ubiquitysoftware.com domain for all email communications to Ubiquity
 employees from now on.

 Thank you.

 Regards,

 Ubiquity Software
 www.ubiquitysoftware.com http://www.ubiquitysoftware.com/
 [EMAIL PROTECTED]
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Re: [Asterisk-Users] Intercept HOLD of Snom phones

[Sven Fischer (support)]

Hi,

do a SIP trace or PCAP trace of the scenario via the webinterface and you will 
see exactly, what is going on...

Regards,

Sven

On Thursday 14 October 2004 21:53, Magnus Jungsbluth wrote:
 Hi,

 I'm running the 1.0 release of Asterisk an it is working nicely with our
 snom 105 phones. Hold puts the caller on hold, attended / unattended
 Transfer works directly with the snom buttons ...
 I have one question though: what does the snom exactly do to tell the *
 to put the call on hold (can I intercept this somewhere)?

 I would like to decide using the callerid which music on hold is tobe
 played: That is: play free music to calls from the outside but play
 copyrighted music if I put an internal call on hold (i.e. a co-worker).
 Is this possible ?

 regards,
 Magnus Jungsbluth
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Re: [Asterisk-Users] Quick question regarding daily restart of asterisk

[David H Hickman]

This tends to be a religious issue.   I guess I am an older admin.  :)  
I come from the school of thought that it is a good idea to
reboot a server that is not meant to be used interactivly (console or 
terminal) on a schedule.  Most software does not require it.  In my 
experience, the systems that have some sort of auto reboot, typically 
run for years without any real maintenance.

My * box reboots itself on sundays at 0300.  It also rsyncs config 
files and voicemail on an hourly basis with another server ( that does 
not reboot.  All it does is act as a samba and nfs fileserver.)  Calls 
are scp as soon as they are combined after the call.

---
David Hickman
PH  314-433-0133 x31Fax 314-865-4752
AIM:fsckrmrfICQ:7059948
YahooIM:dhickman
PGP Prefered - Use current key from the keyservers.
On Oct 18, 2004, at 16:11, Matt G wrote:
Hi All,
I have a quick question regarding restarting (and/or 
stopping/restarting) asterisk daily -- Should it be done?

I've seen conflicting answers, some people have told me that the only 
reason for asterisk to be stopped/started daily was for mpg123 causing 
many childs, which has since been fixed using 'no buffer' or 'nb' 
appended to the line in musiconhold.conf.

Others have told me there is no reason whatsoever to restart/stop it, 
yet there's instructions on how to do it on the wiki, are these just 
outdated?

Is there any other reason why one would want to stop and restart 
asterisk daily? (or at any other scheduled time?)

On a related note, is asterisk -rx restart now the equivalent of 
asterisk -rx stop now  /usr/sbin/safe_asterisk (or whatever 
command is used to restart it)?. I have a job cronned on a slackware 
system to restart it daily using -rx restart now and it creates a 
new PID, and Process Time, but when I run the same thing on Redhat 9 I 
get an error saying that it exited on sig 13. I'm sure this is just a 
redhat specific thing as this isn't the only problem I'm running into, 
but it would be nice to find some answers.

Thanks,
Matt
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RE: [Asterisk-Users] SMTP MTA suggestions.

[Andreas Sikkema]

Fabian Garcia wrote:

 I understand asterisk invokes sendmail in order to send email
 notifications of messages left. Is there another application less
 complicated than Sendmail, I already got mail servers else where
 and they are the ones I want to use.   

All major MTAs emulate the sendmail interface. So you can 
probably use your favourite MTA i.s.o. sendmail.

-- 
Andreas SikkemaRits tele.com
Scheepmakersstraat 11  3011 VH Rotterdam
t: +31 (0)10 2245544f: +31 (0)10 2245540
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Re: [Asterisk-Users] Sipura-3000 - silent dial out on FXO port

[Dameon D. Welch-Abernathy]

Benjamin on Asterisk Mailing Lists wrote:
When I connect to the Sipura to dial out on the PSTN line connected to
the Sipura's FXO port, it gives me the dialtone of the PSTN line and
then I can hear the DTMF for the number I dialled beforehand.
It does work but the customer perceives this delayed second DTMF
feedback as unprofessional and the sipura as a toy. I wonder if
there is anything that can be done to keep the channel to the caller
silent until after the Sipura has sent the DTMF out on the PSTN line.
Upgrade your firmware to the latest release. They solved that problem in 
the more recent releases (2.0.10 and above, IIRC).

-- PhoneBoy
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RE: [Asterisk-Users] Intercept HOLD of Snom phones

[Nick Barnes]

Magnus:
 I would like to decide using the callerid which music on hold is tobe
 played: That is: play free music to calls from the outside 
 but play copyrighted music if I put an internal call on hold 
 (i.e. a co-worker). 
 Is this possible ?

Yes, and it's easier than intercepting the hold request.

Add the following lines to your musiconhold.conf:

INTERNAL = mp3:/var/lib/asterisk/mohmp3/internal
EXTERNAL = mp3:/var/lib/asterisk/mohmp3/external

and put your music into the appropriate directories.

In the dial plan, for internal calls insert the line:

exten = whatever,whatever,SetMusicOnHold(INTERNAL)

and for external calls, insert the line:

exten = whatever,whatever,SetMusicOnHold(EXTERNAL)

in the appropriate places.

Voila.

Nick Barnes


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Re: [Asterisk-Users] OT: Opensource Sipura Profile Compiler for SPA2K, 3K

[Dameon D. Welch-Abernathy]

Kristian Kielhofner wrote:
1) There is a lot of code in the dump from /admin/advanced.
Note that if you're interested in only changing a few parameters, you 
need not post everything.

2) The password is all *'s (not good to PUT it back like that).
See previous point.
-- PhoneBoy
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Re: [Asterisk-Users] Quick question regarding daily restart of asterisk

[Peter Svensson]

On Tue, 19 Oct 2004, David H Hickman wrote:

 This tends to be a religious issue.   I guess I am an older admin.  :)  
 I come from the school of thought that it is a good idea to
 reboot a server that is not meant to be used interactivly (console or 
 terminal) on a schedule.  Most software does not require it.  In my 
 experience, the systems that have some sort of auto reboot, typically 
 run for years without any real maintenance.

Most of our servers stay up until there is some need (such as power 
reconfiguration) to power them down. The time between restarts is usually 
about a year. 

The asterisk box is close to half a year now. Asterisk itself has been 
restarted once since we needed to change a configuration that required a 
restart to reload.

Peter


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[Asterisk-Users] Voicemail and AutoAttendant for a Nortel Option 11 PBX

[Voip Business]

Hello List,,

I have a customer that has a broken voicemailof a nortel option 11 ,,
can we offer something to replace with Asterisk? anyone there that all
ready implement something as this , please contact me because I'll
need service to setup one.

right now they have 8 digital ports for the voicemail


regards


Humberto
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[Asterisk-Users] Problem with DIAL command

[Ali Riza]

Hi,
I have Digium TDM400P. I have succesfully installed and got the demo.
However I have a problem with DIAL command. I have 2 FXS port (Zap/3 and 
Zap/4).
Both of them are connected a inner telephone line.(100-Zap/3 and 101-Zap/4)
When i call 100 with phone 102, 101 redirects 102 to 103.
So i use

exten = s,1,Answer
exten = s,2,Dial(Zap/g2/103,20)
exten = s,3,Hangup
exten = s,103,Hangup
When 103 answers to 102's call, it is ok,  but when i finish the call, the 
call does not hangup. So line is busy although the call is finished.
It waits and when timeout becomes the lines are freed.(Zap/3 and Zap/4)
It there a problem with Dial command?
Another problem which is similar to that. When 102 finishes the call before 
103 answers, it continues ringing the phone 103.
What is wrong?

Thanks in advance.
_
FREE pop-up blocking with the new MSN Toolbar - get it now! 
http://toolbar.msn.com/

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[Asterisk-Users] Working Asterisk With Vonage

[usman]

Hi ! 
I have been working on making my asterisk server work with Vonage 
services. I have been able to recieve calls on my asterisk machine but i 
couldnt call through that account to other people. Means if i call a zap 
channel and then dial 1 314 652 ... then i get an error like 

Executing Dial(Zap/3-1, SIP/dialled number@sphone.vopr.vonage.net:5061) 
in new stack
-- Called dialled number@sphone.vopr.vonage.net:5061
-- Got SIP response 404 Not Found back from 216.115.25.198
-- SIP/sphone.vopr.vonage.net-ec6e is circuit-busy
  == Everyone is busy at this time
-- Executing Hangup(Zap/3-1, ) in new stack
  == Spawn extension (local, 192512100488, 2) exited non-zero on 'Zap/3-1'
-- Hungup 'Zap/3-1'


whether i dial any number ... i get the same response... and always ... 
Can anyone guess what might be the problem ? 
in sip .conf my settings are :

register = username:password@sphone.vopr.vonage.net:5061

[sphone.vopr.vonage.net]
type = peer
fromuser = username
secret = password
host = asterisk machine ip:5070
fromdomain=sphone.vopr.vonage.net
dtmfmode=rfc2833
nat = yes
canreinvite=no

In extensions.conf i have done : 

exten = _1.,1,Dial,SIP/[EMAIL PROTECTED]:5061,tr
exten = _1.,2,Hangup

Please help me in this reagard.

Regards ,

Usman.

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RE: [Asterisk-Users] FireFly w/ SIP

[Whisker, Peter]

Adam

On UK keyboards ,I have to type a £ to get a # into Firefly. The proper
# key does nothing. If you are updating the code, perhaps you might look
at this?

Many thanks
Peter

-Original Message-
From: Adam Hart [mailto:[EMAIL PROTECTED]
Sent: 16 October 2004 07:46
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] FireFly w/ SIP


The best way for me or yourself to debug it is using ethereal (google 
for it) and debugview from www.sysinternals.com. I'm happy to help, so 
send the logs, the native transfer might be the issue.

-Adam

Willem de Groot wrote:

 Is anyone succesfully using FireFly-Thirdparty in SIP modus with Asterisk?
 
 It works in IAX mode, but in SIP mode I am unable to hear anything (no 
 dialtone, no voice). I am able to setup a conversation with another SIP 
 phone though (Xlite, Grandstream) and the other side can hear the 
 FireFly user just fine (both sides using g711u).
 
 I tried different PC's with different audio hardware. They all work fine 
 using FireFly in IAX mode and using other softphones, so I guess it must 
 be related so FireFly in SIP mode.
 
 This is my SIP config:
 
 [201]
 type=friend
 host=dynamic
 dtmfmode=rfc2833
 context=sip
 canreinvite=yes
 
 FireFly is also configured for rfc2833 dtmf.
 
 Thanks for any suggestions!
 Willem
 
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[Asterisk-Users] Follow me using a loop

[Pascal C. Kocher]

Hello *-users

I'm trying to implement a simple follow me solution. The case is that I
would like to be able to pickup the incoming call on a line (whatever)
hang it up and repick it on another line (mobile)

Currently i'm using the following to accomplish this:

exten = 31xxx,1,Wait(1)
exten =
31xxx,2,Dial(IAX2/[EMAIL PROTECTED]CAPI/31xxx:079xxx,120,mgh)
exten = 31xxx,3,Goto(31xxx,1)

This successfully calls my firefly and my mobile phone using CAPI and I
can pick up the call. If I hang up on firefly for example it restarts
the calls on all phones and I'm able to pick it up on the mobile, which
works fine.

What I'm currently missing is, that I'm unable to hang up the call as
callee, as long as ther caller does not hang up, it loops indefinately,
which bothers me a little.

As you can see, I tried using the Option h to hangup the call, but of
course it will just continue and start again. Is there a possibility I
might have been missing to exit this loop as callee (maybe a
GotoIf-condition)?

Best regards,
Pascal.
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[Asterisk-Users] Snom Mass Deployment Config Problems

[Alex Barnes]

Title: Message



Hiall,

I am hoping 
that someone out there is using the Snom phones "configuration via HTTP server" 
functionality.
I have 
downloaded and read the FAQ many times but I am having trouble getting the 
settings to take effect. Probably as I haven't formatted things 
correctly. For example the "fkey" settings aren't taking 
effect.

If someone is 
willing to email me (directly to save spamming the list is fine) a working 
settings file that would help me alot.

thanks a lot 
for any help

Alex

[EMAIL PROTECTED]

-

htmlpre

#Basic 
Settingsphone_name: Snom 6107dhcp: truecall_completion: 
trueauto_dial: 10admin_mode_password: 

#Line 
Settingsuser_realname[1]: Snom 200user_name[1]: snomuser_host[1]: 
172.16.0.217user_pass[1]: snomuser_transport[1]: udpuser_expiry[1]: 
3600user_mailbox[1]: 8500user_outbound[1]: 
172.16.0.217

#SIP 
Settingsnat_detection: offtcp_threshold: udppublish_presence: 
true

#Codec 
Settingsdtmf_type_inband: falseutc_offset: 0ntp_server: 
193.195.52.24

#Network Settingshttp_proxy: 
193.195.52.26http_port: 8001

#Update Settingssetting_server: http://sqa5.sqa.net/test/snom200.htm

#Misc Settingstone_scheme: GBRfkey1: dest [EMAIL PROTECTED]fkey2: 
"dest [EMAIL PROTECTED]"fkey[3]: 
"dest [EMAIL PROTECTED]"fkey[4]: 
dest [EMAIL PROTECTED]

/pre/html




Dear Friends of Ubiquity Software: 

As you may have noticed, Ubiquity Software began using the web domain ubiquity.com earlier this year in addition to the previously established ubiquity.net for our website and email communications to you. However, since that time, a dispute has emerged with respect to actual ownershipof the ubiquity.com domain.

As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net http://www.ubiquity.net/ and via email at @ubiquity.net. However,we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ for web and @ubiquitysoftware.com for email communications.

Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. 

Thank you.

Regards,

Ubiquity Software 
www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ 
[EMAIL PROTECTED] 

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Re: [Asterisk-Users] SMTP MTA suggestions.

[james]

On Mon, 2004-10-18 at 21:11, Fabian Garcia wrote:
  
 
 I understand asterisk invokes sendmail in order to send email
 notifications of messages left. Is there another application less
 complicated than Sendmail, I already got mail servers else where and
 they are the ones I want to use. 
 
  
 
 Any light in this matter will be appreciated.

There are several replacements, but sendmail isn't any harder to config.
You usually only need to change 3 lines in the sendmail config.

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Re: [Asterisk-Users] Follow me using a loop

[Brian]

How about simply doing a blind transfer to your cellphone (or other 
phone...)? You could setup a special extension, say extension *1, to 
dial your cellphone so you don't have to dial the whole number every time.

Pascal C. Kocher wrote:
Hello *-users
I'm trying to implement a simple follow me solution. The case is that I
would like to be able to pickup the incoming call on a line (whatever)
hang it up and repick it on another line (mobile)
Currently i'm using the following to accomplish this:
exten = 31xxx,1,Wait(1)
exten =
31xxx,2,Dial(IAX2/[EMAIL PROTECTED]CAPI/31xxx:079xxx,120,mgh)
exten = 31xxx,3,Goto(31xxx,1)
This successfully calls my firefly and my mobile phone using CAPI and I
can pick up the call. If I hang up on firefly for example it restarts
the calls on all phones and I'm able to pick it up on the mobile, which
works fine.
What I'm currently missing is, that I'm unable to hang up the call as
callee, as long as ther caller does not hang up, it loops indefinately,
which bothers me a little.
As you can see, I tried using the Option h to hangup the call, but of
course it will just continue and start again. Is there a possibility I
might have been missing to exit this loop as callee (maybe a
GotoIf-condition)?
Best regards,
Pascal.
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[Asterisk-Users] record

[Altus Syman]

Good day all
How do I record a call on a vpb channel?
Thanks
Altus
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RE: [Asterisk-Users] SMTP MTA suggestions.

[Low, Adam]

I usually use Qmail www.qmail.org, in my humble opinion it is more straight forward to 
configure than sendmail.

On Mon, 2004-10-18 at 21:11, Fabian Garcia wrote:
  
 
 I understand asterisk invokes sendmail in order to send email 
 notifications of messages left. Is there another application less 
 complicated than Sendmail, I already got mail servers else where and 
 they are the ones I want to use.
 
  
 
 Any light in this matter will be appreciated.

There are several replacements, but sendmail isn't any harder to config.
You usually only need to change 3 lines in the sendmail config.


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Re: [Asterisk-Users] How to make asterisk send email notification ofvoicemessages?

[Brian]

Try googling for 'linux mail server how to'
All you really need is a simple setup (sometimes called a 'smarthost' if 
I recall correctly) that forwards mail to another smtp server.

Take a look at EXIM and Postfix. IMHO they are both much easier to setup 
then sendmail.

Fabian Garcia wrote:
Hi,
Is there something you suggest to have the mail server working? Do I need to
setup sendmail?
Thanks.
Fabian

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Leif Madsen
Sent: Monday, October 18, 2004 3:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] How to make asterisk send email notification
ofvoicemessages?
On Mon, 18 Oct 2004 15:05:40 -0400, Fabian Garcia [EMAIL PROTECTED]
wrote:
Hi, 

I've been trying to have Asterisk to email user each time a voice message
is
left. I am quite lost on how to do this, where should the pop and stmp
settings be written? Or just simply how should one proceed? 

First, make sure your mail server can send emails.  Seconds, in
voicemail.conf, you are going to want to make sure that there is an
email address on your voicemail config line.  For example:
[default]
1000 = 1234,General Mailbox,[EMAIL PROTECTED]
If you want the voicemail attached, in the [general] section add:
attach=yes
HTH,
Leif Madsen.
http://www.asteriskdocs.org
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AW: [Asterisk-Users] Follow me using a loop

[Pascal C. Kocher]

Hello Brian

 How about simply doing a blind transfer to your cellphone (or other 
 phone...)? You could setup a special extension, say extension *1, to 
 dial your cellphone so you don't have to dial the whole 
 number every time.

Thank you for the reply, the log extension is really a DDI, not an
extension, so it doesn't have be dialed.

31xxx,2,Dial(IAX2/[EMAIL PROTECTED]CAPI/31xxx:079xxx,120,mgh)

Best regards,
Pascal.


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[Asterisk-Users] About Supervised Call Transfert on GS BT100

[Ronan de Kermadec]

Hi,
I have a Grandstream Budge Tone 100 and i wanted to use the supervised call 
transfert feature but i don't find any tips for that. So there is my 
question : Is this feature is implemented on GS BT100 and if it is not, it 
is possible to implement it directly on Asterisk. Juts for your infomation, 
blind transfert work fine with the transfert key.

Thanks a lot !
Ronan
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Re: [Asterisk-Users] Working Asterisk With Vonage

[Aaron Clauson]

Hi,

I haven't worked with Vonage myself but I usually get
this error back from my termination provider when the
number I have sent them is incorrect.

It might be worth checking you have used the correct
prefix (011 or 00) and area code etc.

Regards,
Aaron


Hi ! 
I have been working on making my asterisk server work
with Vonage 
services. I have been able to recieve calls on my
asterisk machine but 
i 
couldnt call through that account to other people.
Means if i call a 
zap 
channel and then dial 1 314 652 ... then i get an
error like 

Executing Dial(Zap/3-1, SIP/dialled 
number@sphone.vopr.vonage.net:5061) 
in new stack
-- Called dialled
number@sphone.vopr.vonage.net:5061
-- Got SIP response 404 Not Found back from
216.115.25.198
-- SIP/sphone.vopr.vonage.net-ec6e is circuit-busy
  == Everyone is busy at this time
-- Executing Hangup(Zap/3-1, ) in new stack
  == Spawn extension (local, 192512100488, 2) exited
non-zero on 
'Zap/3-1'
-- Hungup 'Zap/3-1'


whether i dial any number ... i get the same
response... and always ... 
Can anyone guess what might be the problem ? 
in sip .conf my settings are :

register =
username:password@sphone.vopr.vonage.net:5061

[sphone.vopr.vonage.net]
type = peer
fromuser = username
secret = password
host = asterisk machine ip:5070
fromdomain=sphone.vopr.vonage.net
dtmfmode=rfc2833
nat = yes
canreinvite=no

In extensions.conf i have done : 

exten =
_1.,1,Dial,SIP/[EMAIL PROTECTED]:5061,tr
exten = _1.,2,Hangup

Please help me in this reagard.

Regards ,

Usman.
===



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[Asterisk-Users] Setting CallerID on UK BRI line

[bj]

Hi all, 

was just wondering if there were any
special things I had to do to set the outgoing caller ID on a UK BRI (EUROISDN)
line. I've got a line in my extensions.conf which says:

exten = _9.,1,SetCallerID(3317**)

This is then followed by the dial command.


So I dial 9 followed by my mobile number
and the call comes through fine but the display says No Caller ID.


I'm at work now and don't have my access
to my asterisk box (which isn't much use as I can't post debug data or
other lines from the config files). Just wondered if anyone had done this
and where I was going wrong (I have tried different number in addition
to 3317**, like 011893317**, 93317**, 7** and entirely ficticious
numbers). The console shows the caller ID being set but nothing appears
on the remote phone. 

Thanks for your help!

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Re: [Asterisk-Users] SMTP MTA suggestions.

[Andrew Kohlsmith]

On October 18, 2004 09:11 pm, Fabian Garcia wrote:
 I understand asterisk invokes sendmail in order to send email notifications
 of messages left. Is there another application less complicated than
 Sendmail, I already got mail servers else where and they are the ones I
 want to use.

Nullmailer; why put a full blown MTA on your voice box -- nullmailer has its 
own queue and hands off everything to your favourite SMTP machine instead of 
trying to deliver to all the endpoints itself.  This keeps your mail system 
centralized and the control in one place, which is far easier to maintain 
that a full-blown MTA for ever server you have.

If you insist on a full MTA, Postfix gets my vote; it's every bit as secure as 
qmail (I have been using qmail for close to a decade now) but without the 
billion patches required to give qmail any kind of modern functionality.

-A.
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RE: [Asterisk-Users] Setting CallerID on UK BRI line

[Nick Barnes]

Benjamin: 
 
 So I dial 9 followed by my mobile number and the call comes
 through fine but the display says No Caller ID. 

Assuming that your mobile is displaying caller IDs for other numbers and
your ISDN lines are with BT.

There are two ways in which the number can be withheld:

1 - Caller ID is withheld on the line you're calling from. If this is the
case, prefix the dialed number with '1470' which should release the caller
ID.

2 - You're blocking the caller ID yourself. If this is the case, remove the
prefix '141' from the dialed number.

If you don't set a caller ID to anything or set it to an invalid number, BT
will default the caller ID to the base number of the ISDN device, so remove
the 'SetCallerID' line to do your testing. Otherwise, you need only four or
six least significant digits of the number for the SetCallerID command.

Nick Barnes


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Re: [Asterisk-Users] SMTP MTA suggestions.

[Andrew Kohlsmith]

On October 19, 2004 05:48 am, james wrote:
 There are several replacements, but sendmail isn't any harder to config.
 You usually only need to change 3 lines in the sendmail config.

I suppose the reasons people are so anti-sendmail are several:

1. Security.  Sendmail has a track record of being Unix's most insecure MTA.
2. Confusion.  Sendmail's configuration is truly unweildy and unnecessarily 
complex.  c.f. postfix for a configuration interface that doesn't suck.
3. Size.  Why use a cannon swat a fly?  Sendmail will do everything and then 
some, but this is unnecessary complexity and bloat for something as simple as 
a forwarding-only mail server, which is generally all people want if they 
already have a world-class SMTP server doing spam/virus checking and so 
forth.

Sendmail is dead.  Long live the alternatives.

-A.
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Re: [Asterisk-Users] Setting CallerID on UK BRI line

[Peter Svensson]

On Tue, 19 Oct 2004 [EMAIL PROTECTED] wrote:

 I'm at work now and don't have my access to my asterisk box (which isn't 
 much use as I can't post debug data or other lines from the config files). 
 Just wondered if anyone had done this and where I was going wrong (I have 
 tried different number in addition to 3317**, like  011893317**, 93317**, 
 7** and entirely ficticious numbers). The console shows the caller ID 
 being set but nothing appears on the remote phone. 

You need to ask your BT contact how many callerid digits they expect. 
A common configuration is the same number of digits as are given to you 
for DDI. Also you will often need prilocaldialplan=unknown for EuroISDN. 
This means the number is just digits in the form agreed to by you and 
your telco which is what is normally used for outgoing number presentation 
for EuroISDN.

Again, check with your BT contact:
 * verify that the line is set to accept calling number identification
 * the expected Type Of Number and Numbering Plan for the id
 * the expected number of digits to send.

Peter


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[Asterisk-Users] I can't solve my problems with the IVR

[ismaelg]

Hello all,
I'm still having problems with the IVR options.
When I press on my mobile phone one of the digits related in the IVR
options, press 1 for .,press 2 for.., press 3 for..
After I press the one, the second or the tirth key on my mobile phone, I
can't hear nothing more, I can't hear the following menu.
I just search info about dtmf but i can't find information witch help my
to solve my problem.
Any clue will be appreciated.
Here is my channel definition in
Zapata.conf
signalling=fxs_ks
   callwaiting=yes
   language=en
   context=incoming
   callerid=asreceibed
   relaxdtmf=yes
   channel =1
And here a user defined in SIP.conf (All users I have have the same config)
[pepe]
type=friend
;secret=lele
host=dynamic
;dtmfmode=inband; Choices are inband, rfc2833, or info
dtmfmode=info
defaultip=xxx.xxx.xxx.xxx
mailbox=122  ; Mailbox for message waiting indicator
;restrictcid=yes; To have the callerid restriced - sent
as ANI
pickupgroup=1
callgroup=1
username=pepel ; usually matches the section title
fromuser=22 ; overrides the callerid, e.g. required by FWD
callerid=pepe

Thank you in advice
Ismael Gil.


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RE: [Asterisk-Users] VoIP over 1xRTT

[Joe Dennick]

I also have a Samsung i700 phone, and their newer i600 (Mobile Windows
2003), both through Verizon.  With both phones you can purchase
unlimited Internet access for $79 per month.  Both phones are able to
browse the Internet, send and receive email, and use the roll-up
keyboard.  As such, I'm having a tough time understanding why you would
want to use wireless (802.11?) to connect these things to your local
network when the freedom comes from being able to use the build-in
Internet Access.  Verizon plans to have their EVD Network throughout the
US by the end of 2005 (it's currently only in Washington DC and San
Diego.

I cannot understand why you would want to try to get your laptop to go
online through the phone or vice-versa, the phones can do just about
everything you would want to do online, so why bother trying to connect
a laptop?  Many of our company's traveling executives travel with the
i700 in order to keep up with email.  This keeps them from having to lug
their laptops around and search out an Internet connection for them.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deon
Rodden
Sent: Monday, October 18, 2004 5:25 PM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] VoIP over 1xRTT


My concern is Sprint would consider this a threat to their Bread and
Butter. Why buy one of their $100/mo plans for 2000 Anytime Minutes
when you can get a $50/mo data plan and use an unlimited VOIP carrier.
Why pay their overpriced International rates, when you can just go
through a VOIP carrier.

With Verizon, I'm getting the Samsung i700 PDA phone. It runs Windows
Mobile 2003, which is based off CE. I believe I found a SIP client for
it. Using a wireless SD card in the SDIO slot it comes with, anytime I'm
in range of a hotspot, I'll be able to go online and use the SIP client.
Effectively turning my Cell phone into a wireless SIP phone when in
range of a hot spot. Would be really nice when I'm in Australia. Also
found an ssh client for it and a small roll out keyboard.

I know I can get my laptop to go online through this phone, but I wonder
if the phone can go online through my laptop. If so, anytime my laptop
has internet access, such as from Ethernet, the sip client on the
pda-phone would work. Thus turning it into a usb phone. Hehe. 

I love technology... Most of the time.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
McSpadden
Sent: Monday, October 18, 2004 5:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoIP over 1xRTT

It is not always the bandwidth, you are correct. There are however times
on the Sprint network that the bandwidth is reduced, or the bandwidth is
zero, because it can't connect to anything because the network is so
busy. WIth Sprint's CDMA 1xRTT network, voice and data share the same
network, but voice is the bread and butter of the business, so it will
get priority over data.

This is why I'm saying, EV-DO (and later EV-DV) will do great things for
VoIP over cellular networks. EV-DO (DO stands for Data Only), dedicates
a high speed data network, more available bandwidth for everybody, and
less latency, hence less jitter. I'm excited to see these developments,
as I believe it will make VoIP more reliable over these types of
networks. At the moment, there are simply too many variables to trust
it.

Brian


On Mon, 18 Oct 2004 17:36:14 -0400, Deon Rodden [EMAIL PROTECTED]
wrote:
 It's not the bandwidth. I have Sprint and am switching to Verizon with

 a week. When I go online through my Sprint phone, I get 250+ms 
 response
times.
 That can not be VOIP friendly. I have clocked downloads at up to 130 
 kbits per second, so the speed is ok, but the ping response times are 
 bad.
 
 I've heard reports from Verizon users who get an average of 60-80 
 kbits
per
 second, so I 'feel' Sprint's network may be a little faster as their
average
 is higher, at least in my area. But Verizon is already doing the 2nd 
 stage rollout, which is nice and fast.
 
 But the latency issue will probably still be there, for Sprint or 
 Verizon.
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RE: [Asterisk-Users] SIP video support problem

[Doug Reid -Stormcorp]

Hi Jacky

Try using Eye Beam from X-Ten for vidio with Asterisk.

www.Xten.com

Doug Reid

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Jacky
Sent: Tuesday, October 19, 2004 8:20 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP video support problem


Hi, List

I have used Windows Messenger for video call via Asterisk Server.
But Windows Messenger function can't match our requirement.
We are looking more SIP Video Phone can use under Asterisk.

Any suggestion for video Phone(Software or Hardware)?
Also I still have an question about video/audio codec?
Does Asterisk only bypass the codec frame when call is not softswitch?
Can * handle mpeg4 or other codec when video client use this codec?

Thanks,


-- 
Jacky
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RE: [Asterisk-Users] Cheap, Highquality IP Phones

[Doug Reid -Stormcorp]

Hi

We use the Grandstream range, the work very well with Asterisk
although the run at 10BASET so best to keep them on a separate
network. They have all the functionality and work very well, not
the best looking phone but you get what you pay for!

Doug Reid

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kanuri,
Seshu (Company IT)
Sent: Monday, October 18, 2004 3:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Cheap, Highquality IP Phones


If you are for bulk deployment of the phones in large numbers, without
losing your skin along with your shirt, I would recommend buying ATCOM
Phones. You can get them at $55.00 a pop in Bulk and $65 to $70 in
retail. These phones have all the basic features.

Try the link below for an OEM version available in USA:
http://ipphone.eezeephone.com

You can find them on ebay on sale.



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Boehm
Sent: Friday, October 15, 2004 4:50 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Cheap, Highquality IP Phones

I know that there is a list of phones on the wiki, but most of them are
now out of date by months if not a year. Our whole office is using Cisco
7960s.
Nice phones. Works great with asterisk. However, $300 each.

If people could send the phone they use with asterisk, a quick pros/cons
and its price, it would be appreciated.

Basically, I am looking for a high quality $100 2-line SIP phone that
supports g729 and works well with asterisk.

Much appreciated,
Matthew

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NOTICE: If received in error, please destroy and notify sender.  Sender does
not waive confidentiality or privilege, and use is prohibited.

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Re: [Asterisk-Users] About Supervised Call Transfert on GS BT100

[Craig Guy]

There is currently no such feature on the BT100 although someone did post
two weeks or so ago that firmware 1.0.5.12 would have it.  As yet, there is
no hint of this new firmware.  Alternately I think there is a patch around
somewhere to do it within Asterisk, play detective and see if you can find
it.

Craig

- Original Message - 
From: Ronan de Kermadec [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, October 19, 2004 6:18 PM
Subject: [Asterisk-Users] About Supervised Call Transfert on GS BT100


 Hi,

 I have a Grandstream Budge Tone 100 and i wanted to use the supervised
call
 transfert feature but i don't find any tips for that. So there is my
 question : Is this feature is implemented on GS BT100 and if it is not, it
 is possible to implement it directly on Asterisk. Juts for your
infomation,
 blind transfert work fine with the transfert key.

 Thanks a lot !

 Ronan

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RE: [Asterisk-Users] AGI RECORD FILE BUG!

[Simon Smith]

 
I am experiencing a problem with the  RECORD FILE functionality in AGI when
I am doing a Record_file. 
After approx 20 mins + the Record_file ceases to accept escape digits and
therefore records for ever or until my timeout I set. It acts like a dead
application, just recording without the ability to stop.
 
It basically does not allow you to use the escape with the DTMF string you
give and for some reason it works perfectly fine at the beginning of the
call and on small recordings. Please help
 
It is consuming me, we have tried everything and read all the forums. Any
ideas?
 
Simon



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronan de
Kermadec
Sent: Tuesday, 19 October 2004 8:18 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] About Supervised Call Transfert on GS BT100

Hi,

I have a Grandstream Budge Tone 100 and i wanted to use the supervised call
transfert feature but i don't find any tips for that. So there is my
question : Is this feature is implemented on GS BT100 and if it is not, it
is possible to implement it directly on Asterisk. Juts for your infomation,
blind transfert work fine with the transfert key.

Thanks a lot !

Ronan

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#== gPopper Menu ===#
Delete from Gmail inbox:   mailto:del|[EMAIL PROTECTED]
Mark message as unread:mailto:unr|[EMAIL PROTECTED]
Mark message as read:  mailto:rea|[EMAIL PROTECTED]

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RE: [Asterisk-Users] Setting CallerID on UK BRI line

[bj]

Hi Nick, 

yep, my mobile displays caller id for
other numbers - and it even works perfectly displaying caller id information
set by a cheap ISDN pbx on the *same* ISDN line as the Asterisk box. Curious.
Even without setting a callerid on the outgoing calls I get No Caller
ID on my mobile (or other phones - including other BT lines).
BT are not withholding a number and I can change the callerID presented
on the other phone system and it works perfectly. 

Strange, I will investigate more later.


Thanks, 

Benjamin







Nick Barnes
[EMAIL PROTECTED] 
Sent by: [EMAIL PROTECTED]
19/10/2004 12:18



Please respond to
Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED]





To
'Asterisk Users Mailing List -
Non-Commercial Discussion' [EMAIL PROTECTED]


cc



Subject
RE: [Asterisk-Users] Setting CallerID
on UK BRI line









Benjamin: 
 
 So I dial 9 followed by my mobile number and the call comes
 through fine but the display says No Caller ID. 

Assuming that your mobile is displaying caller IDs for other numbers and
your ISDN lines are with BT.

There are two ways in which the number can be withheld:

1 - Caller ID is withheld on the line you're calling from. If this is the
case, prefix the dialed number with '1470' which should release the caller
ID.

2 - You're blocking the caller ID yourself. If this is the case, remove
the
prefix '141' from the dialed number.

If you don't set a caller ID to anything or set it to an invalid number,
BT
will default the caller ID to the base number of the ISDN device, so remove
the 'SetCallerID' line to do your testing. Otherwise, you need only four
or
six least significant digits of the number for the SetCallerID command.

Nick Barnes


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This email has been scanned for all viruses by the MessageLabs SkyScan
service - please check that your Hichrom Virus Scanner is running and up
to date!. 

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Re: [Asterisk-Users] Intercept HOLD of Snom phones

[Magnus Jungsbluth]

Yeah, thats what I figured, BUT, if you transfer an incoming call to 
another internal user, music on hold switches to INTERNAL, and if the 
2nd agent does a another transfer, the incoming call gets INTERNAL 
music. I search for a way to define somewhere in extensions.conf a 
extension that is used when the call is put on hold, so I can decide by 
callerid.

I tryied the snom Music on Hold Server Option and it seems to work:
Define an extension like
1000,1,MusicOnHold(Something)
and set [EMAIL PROTECTED] as Music on Hold Server in the snom phone.
But I still see in the Asterisk CLI when pressing hold(verbose)
-playing Music On Hold (default)
-playing Music On Hold (Something)
So it triggers twice somehow, but anyway, doesn't seem to cause trouble
Nick Barnes wrote:
Magnus:
 

I would like to decide using the callerid which music on hold is tobe
played: That is: play free music to calls from the outside 
but play copyrighted music if I put an internal call on hold 
(i.e. a co-worker). 
Is this possible ?
   

Yes, and it's easier than intercepting the hold request.
Add the following lines to your musiconhold.conf:
INTERNAL = mp3:/var/lib/asterisk/mohmp3/internal
EXTERNAL = mp3:/var/lib/asterisk/mohmp3/external
and put your music into the appropriate directories.
In the dial plan, for internal calls insert the line:
exten = whatever,whatever,SetMusicOnHold(INTERNAL)
and for external calls, insert the line:
exten = whatever,whatever,SetMusicOnHold(EXTERNAL)
in the appropriate places.
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RE: [Asterisk-Users] SMTP MTA suggestions.

[Deon Rodden]

Laugh. I use a bare-bones install of QMail on my main asterisk server. It of
course emulates sendmail and the like.

But on my remote Asterisk server, I use ssmtp, it came as a prerequisite to
Asterisk. When I emerged asterisk, ssmtp came with it. Works great.
Configured it to use my main Asterisk server as the relay, my main asterisk
server only relays from the remote asterisk servers, and all is well. I also
have the remotes running ntpdate every 5 minutes to synchronize with ntpd
running on the main asterisk server. This way the times are in sync. 

By the time I got into Mail clients and building mail servers, sendmail was
already dying, and there was so much negativity about security, so I jumped
straight to QMail. Almost gave up on it (in favor of Exim) but then
discovered all the patches and enhancements people were continuing to make
on QMail. Gentoo has a nice QMail install.

Anyways, for maximum simplicity, I would recommend ssmtp. It works great,
gets the job done.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Andrew
Kohlsmith
Sent: Tuesday, October 19, 2004 7:17 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SMTP MTA suggestions.

On October 19, 2004 05:48 am, james wrote:
 There are several replacements, but sendmail isn't any harder to config.
 You usually only need to change 3 lines in the sendmail config.

I suppose the reasons people are so anti-sendmail are several:

1. Security.  Sendmail has a track record of being Unix's most insecure MTA.
2. Confusion.  Sendmail's configuration is truly unweildy and unnecessarily 
complex.  c.f. postfix for a configuration interface that doesn't suck.
3. Size.  Why use a cannon swat a fly?  Sendmail will do everything and then

some, but this is unnecessary complexity and bloat for something as simple
as 
a forwarding-only mail server, which is generally all people want if they 
already have a world-class SMTP server doing spam/virus checking and so 
forth.

Sendmail is dead.  Long live the alternatives.

-A.
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RE: [Asterisk-Users] VoIP over 1xRTT

[Deon Rodden]

For the SIP client. I just can't imagine using a SIP client over a
connection that has 250+ ms response times. If I make it go online to the
802.11 networks, I can use the SIP Client with ULaw and get high quality SIP
calls at any hotspot. I wouldn't do this for every hot spot, but it'd be a
nice feature to have for when I'm in other countries, or I'm in an area
where Verizon has no coverage or their reception is terrible. Maybe even a
supplement for when I'm running low on minutes. 

When I'm at home, could use it like a Cordless phone. Work off my house
line, which is tied in through my Asterisk server. To be honest, it's just a
kewl feature and a neat thing to do with the phone, what can I say, I'm a
geek like that, hehe.  The Verizon plan is coming from my work, so I don't
want to put TOO many personal calls on it. 

Even with the EVD Network (2nd stage rollout), the ping responses are going
to be 250+ ms. Also, I don't think these phones support 2nd stage, only
Stage 1, 1xRTT. I'd have to get a new phone or something.

As far as the laptop, there are times when I can do things easier from my
laptop. Full sized keyboard, access to all my documents, etc. Right now I'm
used to plugging my laptop into my Sprint phone. So whenever I want to do
SSH or email, I have to use the laptop. Not used to having a PDA, nor one
that can go online on the click of a button. You're probably right, with the
PDA features, me needing to make my laptop go online through the phone will
probably be 5-10% of what it used to be. 

Which codec should be used for 250ms response times? G729A? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
Sent: Tuesday, October 19, 2004 7:14 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] VoIP over 1xRTT

I also have a Samsung i700 phone, and their newer i600 (Mobile Windows
2003), both through Verizon.  With both phones you can purchase
unlimited Internet access for $79 per month.  Both phones are able to
browse the Internet, send and receive email, and use the roll-up
keyboard.  As such, I'm having a tough time understanding why you would
want to use wireless (802.11?) to connect these things to your local
network when the freedom comes from being able to use the build-in
Internet Access.  Verizon plans to have their EVD Network throughout the
US by the end of 2005 (it's currently only in Washington DC and San
Diego.

I cannot understand why you would want to try to get your laptop to go
online through the phone or vice-versa, the phones can do just about
everything you would want to do online, so why bother trying to connect
a laptop?  Many of our company's traveling executives travel with the
i700 in order to keep up with email.  This keeps them from having to lug
their laptops around and search out an Internet connection for them.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deon
Rodden
Sent: Monday, October 18, 2004 5:25 PM
To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
Discussion'
Subject: RE: [Asterisk-Users] VoIP over 1xRTT


My concern is Sprint would consider this a threat to their Bread and
Butter. Why buy one of their $100/mo plans for 2000 Anytime Minutes
when you can get a $50/mo data plan and use an unlimited VOIP carrier.
Why pay their overpriced International rates, when you can just go
through a VOIP carrier.

With Verizon, I'm getting the Samsung i700 PDA phone. It runs Windows
Mobile 2003, which is based off CE. I believe I found a SIP client for
it. Using a wireless SD card in the SDIO slot it comes with, anytime I'm
in range of a hotspot, I'll be able to go online and use the SIP client.
Effectively turning my Cell phone into a wireless SIP phone when in
range of a hot spot. Would be really nice when I'm in Australia. Also
found an ssh client for it and a small roll out keyboard.

I know I can get my laptop to go online through this phone, but I wonder
if the phone can go online through my laptop. If so, anytime my laptop
has internet access, such as from Ethernet, the sip client on the
pda-phone would work. Thus turning it into a usb phone. Hehe. 

I love technology... Most of the time.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
McSpadden
Sent: Monday, October 18, 2004 5:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoIP over 1xRTT

It is not always the bandwidth, you are correct. There are however times
on the Sprint network that the bandwidth is reduced, or the bandwidth is
zero, because it can't connect to anything because the network is so
busy. WIth Sprint's CDMA 1xRTT network, voice and data share the same
network, but voice is the bread and butter of the business, so it will
get priority over data.

This is why I'm saying, EV-DO (and later EV-DV) will do great things for
VoIP over cellular 

[Asterisk-Users] Called number Callerid with Sip

[Joseph]

Does anyone know if the sip firmware on the 79xx phones would support *
pushing the called name back to the calling phone?

Maybe using the SIP Info method?

-- 
respectfully, Joseph ===
-= **  =

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Re: [Asterisk-Users] VoIP over 1xRTT

[Matthew Marlowe]

I didn't read this whole discussion but I've used the G729A codec
using X-Ten Pro on my laptop while connected to the 1xRTT network to
my Verizon phone via Bluetooth and it worked rather well.  (as long as
I had a full signal on my cell phone).  It's not something you can
call stable, or probably ever will be called stable but I was well
found it very cool that it did work.


On Tue, 19 Oct 2004 08:20:04 -0400, Deon Rodden [EMAIL PROTECTED] wrote:
 For the SIP client. I just can't imagine using a SIP client over a
 connection that has 250+ ms response times. If I make it go online to the
 802.11 networks, I can use the SIP Client with ULaw and get high quality SIP
 calls at any hotspot. I wouldn't do this for every hot spot, but it'd be a
 nice feature to have for when I'm in other countries, or I'm in an area
 where Verizon has no coverage or their reception is terrible. Maybe even a
 supplement for when I'm running low on minutes.
 
 When I'm at home, could use it like a Cordless phone. Work off my house
 line, which is tied in through my Asterisk server. To be honest, it's just a
 kewl feature and a neat thing to do with the phone, what can I say, I'm a
 geek like that, hehe.  The Verizon plan is coming from my work, so I don't
 want to put TOO many personal calls on it.
 
 Even with the EVD Network (2nd stage rollout), the ping responses are going
 to be 250+ ms. Also, I don't think these phones support 2nd stage, only
 Stage 1, 1xRTT. I'd have to get a new phone or something.
 
 As far as the laptop, there are times when I can do things easier from my
 laptop. Full sized keyboard, access to all my documents, etc. Right now I'm
 used to plugging my laptop into my Sprint phone. So whenever I want to do
 SSH or email, I have to use the laptop. Not used to having a PDA, nor one
 that can go online on the click of a button. You're probably right, with the
 PDA features, me needing to make my laptop go online through the phone will
 probably be 5-10% of what it used to be.
 
 Which codec should be used for 250ms response times? G729A?
 
 
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick
 Sent: Tuesday, October 19, 2004 7:14 AM
 To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
 Subject: RE: [Asterisk-Users] VoIP over 1xRTT
 
 I also have a Samsung i700 phone, and their newer i600 (Mobile Windows
 2003), both through Verizon.  With both phones you can purchase
 unlimited Internet access for $79 per month.  Both phones are able to
 browse the Internet, send and receive email, and use the roll-up
 keyboard.  As such, I'm having a tough time understanding why you would
 want to use wireless (802.11?) to connect these things to your local
 network when the freedom comes from being able to use the build-in
 Internet Access.  Verizon plans to have their EVD Network throughout the
 US by the end of 2005 (it's currently only in Washington DC and San
 Diego.
 
 I cannot understand why you would want to try to get your laptop to go
 online through the phone or vice-versa, the phones can do just about
 everything you would want to do online, so why bother trying to connect
 a laptop?  Many of our company's traveling executives travel with the
 i700 in order to keep up with email.  This keeps them from having to lug
 their laptops around and search out an Internet connection for them.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Deon
 Rodden
 Sent: Monday, October 18, 2004 5:25 PM
 To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial
 Discussion'
 Subject: RE: [Asterisk-Users] VoIP over 1xRTT
 
 My concern is Sprint would consider this a threat to their Bread and
 Butter. Why buy one of their $100/mo plans for 2000 Anytime Minutes
 when you can get a $50/mo data plan and use an unlimited VOIP carrier.
 Why pay their overpriced International rates, when you can just go
 through a VOIP carrier.
 
 With Verizon, I'm getting the Samsung i700 PDA phone. It runs Windows
 Mobile 2003, which is based off CE. I believe I found a SIP client for
 it. Using a wireless SD card in the SDIO slot it comes with, anytime I'm
 in range of a hotspot, I'll be able to go online and use the SIP client.
 Effectively turning my Cell phone into a wireless SIP phone when in
 range of a hot spot. Would be really nice when I'm in Australia. Also
 found an ssh client for it and a small roll out keyboard.
 
 I know I can get my laptop to go online through this phone, but I wonder
 if the phone can go online through my laptop. If so, anytime my laptop
 has internet access, such as from Ethernet, the sip client on the
 pda-phone would work. Thus turning it into a usb phone. Hehe.
 
 I love technology... Most of the time.
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Brian
 McSpadden
 Sent: Monday, October 18, 2004 5:45 PM
 To: Asterisk Users Mailing List - 

[Asterisk-Users] Speex wideband mode

[Michael Graves]

Hi All,

Does anyone here use the Speex codec on their * server? I see that
Voicepulse Connect supports Speex and I'd like to try using it in
wideband mode. I'm wondering if it might be a suitable alternative to
GSM at comparable data rates. Any idea how I setup the codec for
wideband mode?

Thanks,

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

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

I know nothing, but I keep listening. - INXS
 
** Tag(s) inserted by Bandit Tagger98 - http://www.gbar.dtu.dk/~c918704




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[Asterisk-Users] mISDN, CAPI, ISDN ???

[Erwan DESVERGNES]

Did someone have experience with:



-
Chan_modem

-
Chan_capi

-
Chan_misdn



What is the best???





_

Erwan
 Desvergnes - ANDIUM -

82/86 rue Château Gaillard

69100 Villeurbanne



Tel. 04 3743 44 45
/ Fax 04 37 43 44 44

E-mail: [EMAIL PROTECTED]








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[Asterisk-Users] SPA-3k *

[jeffpowen]

I have my brother-n-law in Australia who just purchased a SPA-3k.

He is wanting to connect to my * server.

For the * entry I have the following:

sip.conf:
[2203] ; Dustintype=friendhost=dynamiccontext=defaultsecret=supersecretpasscodemaxexpirey=1800defaultexpirey=1600callerid="Dustin-Debbie" 2203mailbox=2203dtmfmode=rfc2833canreinvite=nonat=always
I also have the ports opened on my firwall for 5060 TCP/UDP and 1-2 UDP pointing to my NAT'd * server.

He is also NAT on his side and has the SPA-3k with Firmware 2.0.9 with settings all default except for the following:

Line 1:

Proxy: DynDnsAddrofMy*Host
Use OutBound Proxy: No
Register: Yes
UserID: 2203
AuthID: supersecretpasscode

And all that * is giving is errors saying the following:

ct 19 07:05:55 NOTICE[6150]: Registration from 'Dustin  Debbie sip:[EMAIL PROTECTED]' failed for 'HisIPAddressinAU'Oct 19 07:06:00 NOTICE[6150]: Peer '2203' is now REACHABLE!Oct 19 07:07:04 NOTICE[6150]: Peer '2203' is now UNREACHABLE!
Any suggestions on where to research to get working?

Thanks,

Jeff
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RE: [Asterisk-Users] SPA-3k *

[Alex Barnes]

Title: Message



Can you enable "sip debug ip 'HisIPAddressinAU'"

And copy out the REGISTER message and responses. 
Might help narrow down what the problem is.

  
  -Original Message-From: 
  [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: 19 
  October 2004 13:51To: 
  [EMAIL PROTECTED]Subject: [Asterisk-Users] SPA-3k 
   *
  I have my brother-n-law in Australia who just purchased a SPA-3k.
  
  He is wanting to connect to my * server.
  
  For the * entry I have the following:
  
  sip.conf:
  [2203] ; 
  Dustintype=friendhost=dynamiccontext=defaultsecret=supersecretpasscodemaxexpirey=1800defaultexpirey=1600callerid="Dustin-Debbie" 
  2203mailbox=2203dtmfmode=rfc2833canreinvite=nonat=always
  I also have the ports opened on my firwall for 5060 TCP/UDP and 
  1-2 UDP pointing to my NAT'd * server.
  
  He is also NAT on his side and has the SPA-3k with Firmware 2.0.9 with 
  settings all default except for the following:
  
  Line 1:
  
  Proxy: DynDnsAddrofMy*Host
  Use OutBound Proxy: No
  Register: Yes
  UserID: 2203
  AuthID: supersecretpasscode
  
  And all that * is giving is errors saying the following:
  
  ct 19 07:05:55 NOTICE[6150]: Registration from 'Dustin  Debbie 
  sip:[EMAIL PROTECTED]' failed for 'HisIPAddressinAU'Oct 19 
  07:06:00 NOTICE[6150]: Peer '2203' is now REACHABLE!Oct 19 07:07:04 
  NOTICE[6150]: Peer '2203' is now UNREACHABLE!
  Any suggestions on where to research to get working?
  
  Thanks,
  
  Jeff


Dear Friends of Ubiquity Software: 

As you may have noticed, Ubiquity Software began using the web domain ubiquity.com earlier this year in addition to the previously established ubiquity.net for our website and email communications to you. However, since that time, a dispute has emerged with respect to actual ownershipof the ubiquity.com domain.

As an international software company founded over decade ago, you can always reach Ubiquity Software under the website www.ubiquity.net http://www.ubiquity.net/ and via email at @ubiquity.net. However,we have also chosen to expand our domain to the more specific www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ for web and @ubiquitysoftware.com for email communications.

Please use either the historical ubiquity.net or begin to use the new ubiquitysoftware.com domain for all email communications to Ubiquity employees from now on. 

Thank you.

Regards,

Ubiquity Software 
www.ubiquitysoftware.com http://www.ubiquitysoftware.com/ 
[EMAIL PROTECTED] 

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Re: [Asterisk-Users] mISDN, CAPI, ISDN ???

[daschi]

I've just used chan_capi it's very easy to use with Fritz!Cards and
therefore I like it ;-)

 Did someone have experience with:
 
  
 
 -  Chan_modem
 
 -  Chan_capi
 
 -  Chan_misdn
 
  
 
 What is the best???
 
  
 
  
 
 _
 
 Erwan Desvergnes - ANDIUM -
 
 82/86 rue Château Gaillard
 
 69100 Villeurbanne
 
  
 
 Tel. 04 37 43 44 45 / Fax 04 37 43 44 44
 
 E-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
 
  
 
 

-- 
+++ GMX DSL Premiumtarife 3 Monate gratis* + WLAN-Router 0,- EUR* +++
Clevere DSL-Nutzer wechseln jetzt zu GMX: http://www.gmx.net/de/go/dsl

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Re: [Asterisk-Users] False Hangup detected on Digium TDM400P

[Scott Wolf]

Ruben Fagundo wrote:
I have an asterisk server running on Redhat 8.0 with a Digium TDM400P 
w/4 FXO modules (TDM04P)

There are 2 lines going into the Digium card. One line is a Vonage 
digital line, and the other line is a Comcast voice line. I have a SIP 
Grandstream 100 phone connected to the Asterisk server.

The problem is that on occasionally, after talking for about 20 
minutes or so, the call gets hung up and I get a fast paced busy 
signal. The caller gets dead air.  I have called Digium wondering if 
their is a hardware problem, but they don't seem to think so. Is  
there a way to deactivate the other 2 channels, if they aren't being 
use. Perhaps having all 4 channels active is causing the false 
detects. The problem occurs on both lines for incoming calls and it 
just happened again today on an outgoing call after 15 minutes of talk.

I have tried busydetect=no and yes and neither one make a difference. 
suggestions?

/etc/asterisk/zapata.conf
 busydetect=yes
 busycount=10

We have a 4 port FXO doing the exact same thing. Normally only happens 
after a few minues, but today it happene 4 times on a call from England 
after anywhere from 30 seconds to 10 minutes.

Scott Wolf
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Re: [Asterisk-Users] record

[Steven Critchfield]

On Tue, 2004-10-19 at 11:54 +0200, Altus Syman wrote:
 Good day all
 How do I record a call on a vpb channel?

Part of the point behind the way asterisk is built is that at the
application point of view, the channel is mostly irrelavent. Therefore
you record by using record.
-- 
Steven Critchfield [EMAIL PROTECTED]

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RE: [Asterisk-Users] AGI RECORD FILE BUG!

[Steven Critchfield]

You also are having a problem realizing that we have now seen your
message SEVERAL times and shoved into other threads that are irrelavent
to recording or AGI. You are not helping yourself by doing this. 

-- 
Steven Critchfield [EMAIL PROTECTED]

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Re: [Asterisk-Users] SPA-3k *

[Benjamin on Asterisk Mailing Lists]

On Tue, 19 Oct 2004 12:50:41 +, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
  
 UDP pointing to my NAT'd * server. 
   
 He is also NAT on his side

Are you saying this is NAT on both ends? aka double NAT?

If so, use tunneling. Double NAT is a bitch.

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.
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Re: [Asterisk-Users] Specify location of ADSI Softkeys ?

[Lance Arbuckle]

I've come up with a temporary solution to locate the softkeys where I
want them...

set up the following key:
KEY blank IS  OR Blank
GOTO offhook
ENDKEY

then you can do
SHOWKEYS park
SHOWKEYS blank
SHOWKEYS xfer 
SHOWKEYS hold 
SHOWKEYS blank 
SHOWKEYS flash

and end up with softkeys laid out like this:
-PARK   HOLD-

-X-FER FLASH-

If anyone has a better solution, please let me know.
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Re: [Asterisk-Users] Snom Mass Deployment Config Problems

[Sven Fischer (support)]

Hi,

I'm sure a lot people can help you here, maybe I'm the first. See below 
inline:

On Tuesday 19 October 2004 11:20, Alex Barnes wrote:
 Hi all,

 I am hoping that someone out there is using the Snom phones
 configuration via HTTP server functionality.
 I have downloaded and read the FAQ many times but I am having trouble
 getting the settings to take effect.  Probably as I haven't formatted
 things correctly.  For example the fkey settings aren't taking effect.

 If someone is willing to email me (directly to save spamming the list is
 fine) a working settings file that would help me alot.

 thanks a lot for any help

 Alex

 [EMAIL PROTECTED]

 -

 html
 pre

 #Basic Settings
 phone_name: Snom 6107
 dhcp: true
 call_completion: true
 auto_dial: 10
 admin_mode_password: 

 #Line Settings

the brackets are wrong:

 user_realname[1]: Snom 200
 user_name[1]: snom
 user_host[1]: 172.16.0.217
 user_pass[1]: snom
 user_transport[1]: udp
 user_expiry[1]: 3600
 user_mailbox[1]: 8500
 user_outbound[1]: 172.16.0.217


it should be like:

user_realname1: Snom 200
user_name1: snom
user_host1: 172.16.0.217
user_pass1: snom
user_transport1: udp
user_expiry1: 3600
user_mailbox1: 8500
user_outbound1: 172.16.0.217

 #SIP Settings
 nat_detection: off
 tcp_threshold: udp
 publish_presence: true

 #Codec Settings
 dtmf_type_inband: false
 utc_offset: 0
 ntp_server: 193.195.52.24

 #Network Settings
 http_proxy: 193.195.52.26
 http_port: 8001

 #Update Settings
 setting_server: http://sqa5.sqa.net/test/snom200.htm

what is this ? remove it.

 http://sqa5.sqa.net/test/snom200.htm


 #Misc Settings
 tone_scheme: GBR

here again the brackets are wrong:

 fkey1: dest [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 fkey2: dest [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
 fkey[3]: dest [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
 fkey[4]: dest [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]


fkey1: dest sip:[EMAIL PROTECTED];user=phone
fkey2: dest sip:[EMAIL PROTECTED];user=phone
fkey3: dest sip:[EMAIL PROTECTED];user=phone
fkey4: dest sip:[EMAIL PROTECTED];user=phone


 /pre
 /html


BTW did you saw our FAQ regarding massdeployment ?

kind regards,

Sven Fischer





 Dear Friends of Ubiquity Software:

 As you may have noticed, Ubiquity Software began using the web domain
 ubiquity.com earlier this year in addition to the previously established
 ubiquity.net for our website and email communications to you.  However,
 since that time, a dispute has emerged with respect to actual ownership of
 the ubiquity.com domain.

 As an international software company founded over decade ago, you can
 always reach Ubiquity Software under the website www.ubiquity.net
 http://www.ubiquity.net/  and via email at @ubiquity.net.  However, we
 have also chosen to expand our domain to the more specific
 www.ubiquitysoftware.com http://www.ubiquitysoftware.com/  for web and
 @ubiquitysoftware.com for email communications.

 Please use either the historical ubiquity.net or begin to use the new
 ubiquitysoftware.com domain for all email communications to Ubiquity
 employees from now on.

 Thank you.

 Regards,

 Ubiquity Software
 www.ubiquitysoftware.com http://www.ubiquitysoftware.com/
 [EMAIL PROTECTED]

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RE: [Asterisk-Users] mISDN, CAPI, ISDN ???

[Erwan DESVERGNES]

Have you got any problem with sound on the 2nde chanel ???

-Message d'origine-
De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
Envoyé : mardi 19 octobre 2004 14:59
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] mISDN, CAPI, ISDN ???

I've just used chan_capi it's very easy to use with Fritz!Cards and
therefore I like it ;-)

 Did someone have experience with:
 
  
 
 -  Chan_modem
 
 -  Chan_capi
 
 -  Chan_misdn
 
  
 
 What is the best???
 
  
 
  
 
 _
 
 Erwan Desvergnes - ANDIUM -
 
 82/86 rue Château Gaillard
 
 69100 Villeurbanne
 
  
 
 Tel. 04 37 43 44 45 / Fax 04 37 43 44 44
 
 E-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] 
 
  
 
 

-- 
+++ GMX DSL Premiumtarife 3 Monate gratis* + WLAN-Router 0,- EUR* +++
Clevere DSL-Nutzer wechseln jetzt zu GMX: http://www.gmx.net/de/go/dsl

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Re: [Asterisk-Users] mISDN, CAPI, ISDN ???

[Massimo De Nadal]

Definitely choose chan_capi.
Chan_modem is almost deprecated, bad quality and very few features.
Chan_misdn seems to be a very good project but it is still young.
Zaphfc in theory it's wonderful (zap echo cancellation, timing etc.) but 
you have to use older * versions, (till new kapejod's release) and here 
in Italy (with italian nt1) I have many stability issues.

Chan_capi works really great, you have to choose isdn boards with good 
capi drivers (avm, eicon) but the results is really stable and full 
featured.

regards
maxx
Erwan DESVERGNES wrote:
Did someone have experience with:
 

-  Chan_modem
-  Chan_capi
-  Chan_misdn
 

What is the best???
 

 

**_**
**Erwan Desvergnes **- **ANDIUM **-
//82/86 rue Château Gaillard//
//69100 Villeurbanne//
 

//Tel. 04 37 43 44 45 / Fax 04 37 43 44 44//
E-mail: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
 


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[Asterisk-Users] txgain usage with T100P

[dawson]

Has anyone tweaked the txgain values on an T100P 
card(hooked up to PRI)with success?
People complain about loudness.

Thanks,

Don Dawson.

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[Asterisk-Users] Re: IAX2 Nat issue, Any help greatly appreciated

[Gene Willingham]

I am using a Sonicwall 3060.  The SonicWall has 6 hardware interfaces. My
asterisk box is on one interface configured as a DMZ.  It still goes through
NAT, but is exposed as a public ip of x.x.x.56, and private IP 192.168.3.2.
The public ip of the firewall is x.x.x.50.

I am using the connect service from Voicepulse.  They are initiating the
call.  IT appears when I register with them I 

What I think is happening is:  If I receive an inbound call on IAX during an
IAX registration, the call does not get setup.  I appear to be unavailable
to the other server. When a call fails I noticed using tcpdump that the
inbound packets are destined for port 13081.  When the call succeeds the
inbound packets are destined for port 4569.  Port 13081 seems to make sense
when looking at iax2 show registry.  But it does not match the output from
tcpdump when compared to calls that succeed.



gw1*CLI iax2 show registry
Host  UsernamePerceived Refresh  State
66.234.228.170:4569   QSa55JPy58  x.x.x.50:13081   60  Registered

 

[IAX2 debug enabled]
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
REGREQ 

   Timestamp: 00017ms  SCall: 2  DCall: 0 [66.234.228.170:4569]
   USERNAME: QSa55JPy58
   REFRESH : 60

gw1*CLI 
Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REGACK 
   Timestamp: 00015ms  SCall: 00186  DCall: 2 [66.234.228.170:4569]
   USERNAME: QSa55JPy58
   DATE TIME   : 156437288
   REFRESH : 60
   APPARENT ADDRES : IPV4 x.x.x.50:13081

gw1*CLI 
Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK

   Timestamp: 00015ms  SCall: 2  DCall: 00186 [66.234.228.170:4569]
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
HANGUP 
   Timestamp: 09779ms  SCall: 00518  DCall: 0 [66.234.228.170:4569]

 

Output from tcpdump:
22:02:48.246092 x.x.com.4569  170-228-234-66.cosmoweb.net.4569: udp 12 (DF)
[tos 0x10]
22:03:18.597719 170-228-234-66.cosmoweb.net.4569  x.x.com.13081: udp 84
(DF)
22:03:20.601668 170-228-234-66.cosmoweb.net.4569  x.x.com.13081: udp 84
(DF)
22:03:28.406522 170-228-234-66.cosmoweb.net.4569  x.X.com.13081: udp 12
(DF)
22:03:30.406566 170-228-234-66.cosmoweb.net.4569  x.x.com.13081: udp 12
(DF)
22:03:30.601889 170-228-234-66.cosmoweb.net.4569  X.X.com.13081: udp 84
(DF)
22:03:38.236056 X.x.com.4569  170-228-234-66.cosmoweb.net.4569: udp 28 (DF)
[tos 0x10]
22:03:38.246584 170-228-234-66.cosmoweb.net.4569  x.x.com.4569: udp 52 (DF)

 
Configuration:
  Asterisk 1.0.1.
  Sonicwall 3060 Firewall.


Message: 3
Date: Tue, 19 Oct 2004 14:27:29 +0900
From: Benjamin on Asterisk Mailing Lists
[EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX2 Nat issue, Any help greatly
appreciated
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=US-ASCII

On Mon, 18 Oct 2004 18:20:17 -0400, Gene Willingham
[EMAIL PROTECTED] wrote:
 
   My asterisk box is behind a firewall, but in a DMZ.

Is this a hardware or software DMZ?


  The Asterisk Box is
 published with a public IP address.  My provider appears to be ignoring
the
 Public IP address and using the received from ip.

Can you be a bit more specific. What's the setup of your NAT/DMZ?
Which address is published? The NAT router's? The DMZ's? Who is
initiating the calls? etc

rgds
benjk

-- 
Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya,
Tokyo, Japan.

NB: Spam filters in place. Messages unrelated to the * mailing lists
may get trashed.




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RE: [Asterisk-Users] SIP video support problem

[Tomica Crnek]

Hi Doug,

Are you using eyeBeam with Asterisk? I posted in another message to this
group this text:

---
Hi everyone,
 
Is anyone using Xten eyeBeam Video softphone with Asterisk? It supports
few types of H.263 codecs for video. I have tried to use it with
Asterisk with enabled video support in sip.conf and allowed h263, but in
the moment I click to start sending video I get this error in Asterisk:
 
NOTICE[445923361]: Unknown RTP codec 127 received
 
Can anyone help?
---

Can you help?

Thanks,

Tomica

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Doug Reid -Stormcorp
 Sent: Tuesday, October 19, 2004 1:39 PM
 To: Jacky; Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] SIP video support problem
 
 Hi Jacky
 
 Try using Eye Beam from X-Ten for vidio with Asterisk.
 
 www.Xten.com
 
 Doug Reid
 
 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Jacky
 Sent: Tuesday, October 19, 2004 8:20 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] SIP video support problem
 
 
 Hi, List
 
 I have used Windows Messenger for video call via Asterisk Server.
 But Windows Messenger function can't match our requirement.
 We are looking more SIP Video Phone can use under Asterisk.
 
 Any suggestion for video Phone(Software or Hardware)?
 Also I still have an question about video/audio codec?
 Does Asterisk only bypass the codec frame when call is not softswitch?
 Can * handle mpeg4 or other codec when video client use this codec?
 
 Thanks,
 
 
 --
 Jacky
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RE: [Asterisk-Users] AGI RECORD FILE BUG!

[Scott Stingel]

If you have a solidly re-produceable bug, suggest that you go to
http://bugs.digium.com/login_page.php
Sign up, and post the bug.

Regards, 


Scott M. Stingel
President,
Emerging Voice Technology, Inc.
Palo Alto California  London England
www.evtmedia.com 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simon Smith
Sent: Tuesday, October 19, 2004 3:49 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] AGI RECORD FILE BUG!
Importance: High

 
I am experiencing a problem with the  RECORD FILE functionality in AGI when
I am doing a Record_file. 
After approx 20 mins + the Record_file ceases to accept escape digits and
therefore records for ever or until my timeout I set. It acts like a dead
application, just recording without the ability to stop.
 
It basically does not allow you to use the escape with the DTMF string you
give and for some reason it works perfectly fine at the beginning of the
call and on small recordings. Please help
 
It is consuming me, we have tried everything and read all the forums. Any
ideas?
 
Simon



-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ronan de
Kermadec
Sent: Tuesday, 19 October 2004 8:18 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] About Supervised Call Transfert on GS BT100

Hi,

I have a Grandstream Budge Tone 100 and i wanted to use the supervised call
transfert feature but i don't find any tips for that. So there is my
question : Is this feature is implemented on GS BT100 and if it is not, it
is possible to implement it directly on Asterisk. Juts for your infomation,
blind transfert work fine with the transfert key.

Thanks a lot !

Ronan

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#== gPopper Menu ===#
Delete from Gmail inbox:   mailto:del|[EMAIL PROTECTED]
Mark message as unread:mailto:unr|[EMAIL PROTECTED]
Mark message as read:  mailto:rea|[EMAIL PROTECTED]

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RE: [Asterisk-Users] AGI RECORD FILE BUG!

[Simon Smith]

Oh ok, so there are other threads with Recording...Where are they? 

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Tuesday, 19 October 2004 11:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] AGI RECORD FILE BUG!

You also are having a problem realizing that we have now seen your message
SEVERAL times and shoved into other threads that are irrelavent to recording
or AGI. You are not helping yourself by doing this. 

--
Steven Critchfield [EMAIL PROTECTED]

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#== gPopper Menu ===#
Delete from Gmail inbox:   mailto:del|[EMAIL PROTECTED]
Mark message as unread:mailto:unr|[EMAIL PROTECTED]
Mark message as read:  mailto:rea|[EMAIL PROTECTED]

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Re: [Asterisk-Users] mISDN, CAPI, ISDN ???

[Dave Cotton]

On Tue, 2004-10-19 at 14:58 +0200, [EMAIL PROTECTED] wrote:
 I've just used chan_capi it's very easy to use with Fritz!Cards and
 therefore I like it ;-)

Worked straight out of the box on an AVM C2, hope it does the same with
2 Fritz!Cards in the same machine.


-- 
Dave Cotton [EMAIL PROTECTED]

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RE: [Asterisk-Users] GSM to g729 Conversion

[Kanuri, Seshu (Company IT)]

You are mixing oranges and apples here i guess. 
G729is a MediaTransmission Protocol Codec the other is a Compressed 
AudioFile format.

There are no .g729 audio files as far as I 
know.

Seshu Kanuri




From: [EMAIL PROTECTED] 
[mailto:[EMAIL PROTECTED] On Behalf Of Victor 
CartesSent: Monday, October 18, 2004 3:39 PMTo: 
[EMAIL PROTECTED]Subject: [Asterisk-Users] GSM to g729 
Conversion

Hi!

Does anybody know how to convert .gsm file format 
to .g729 in order to use it for an IVR system?

Thanks in advance.

Vïctor




NOTICE: If received in error, please destroy and notify sender.  Sender does not waive confidentiality or privilege, and use is prohibited.

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[Asterisk-Users] Transparent SIP Server

[Andreas Anderson]

Hi Guys,
i need to do some kind of CDR for all clients inside my network, but they do 
not register/use the same
sip-server, some of them use iptel, others fwd and various other services.

Can i somehow put asterisk in the (control-)path between my clients and the 
other services
(iptable-redirect like with a squid-proxy), so the clients don't have to 
change their settings and
still register with their respective service, but asterisk does a complete 
CDR on every call?

If thats not possible, anyone knows a software that supports this? SER?
Regards,
Andreas
_
Need more speed? Get Xtra JetStream  @ http://xtra.co.nz/jetstream
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Re: [Asterisk-Users] Transparent SIP Server

[Darren Sessions]

SER most definitely does CDR archiving via MySql database. It's a 
hellaciously fast and stable proxy - sounds like it'd be a good choice 
for the core of your network with all the different components.

On Oct 19, 2004, at 10:01 AM, Andreas Anderson wrote:
Hi Guys,
i need to do some kind of CDR for all clients inside my network, but 
they do not register/use the same
sip-server, some of them use iptel, others fwd and various other 
services.

Can i somehow put asterisk in the (control-)path between my clients 
and the other services
(iptable-redirect like with a squid-proxy), so the clients don't have 
to change their settings and
still register with their respective service, but asterisk does a 
complete CDR on every call?

If thats not possible, anyone knows a software that supports this? SER?
Regards,
Andreas
_
Need more speed? Get Xtra JetStream  @ http://xtra.co.nz/jetstream
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Re: [Asterisk-Users] Re: IAX2 Nat issue, Any help greatly appreciated

[Benjamin on Asterisk Mailing Lists]

On Tue, 19 Oct 2004 09:44:12 -0400, Gene Willingham
[EMAIL PROTECTED] wrote:
 What I think is happening is:  If I receive an inbound call on IAX during an
 IAX registration, the call does not get setup.  I appear to be unavailable
 to the other server. When a call fails I noticed using tcpdump that the
 inbound packets are destined for port 13081.  When the call succeeds the
 inbound packets are destined for port 4569.  Port 13081 seems to make sense
 when looking at iax2 show registry.  But it does not match the output from
 tcpdump when compared to calls that succeed.
 
 gw1*CLI iax2 show registry
 Host  UsernamePerceived Refresh  State
 66.234.228.170:4569   QSa55JPy58  x.x.x.50:13081   60  Registered
 
 [IAX2 debug enabled]
 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass:
 REGREQ
 
Timestamp: 00017ms  SCall: 2  DCall: 0 [66.234.228.170:4569]
USERNAME: QSa55JPy58
REFRESH : 60
 
 gw1*CLI
 Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
 REGACK
Timestamp: 00015ms  SCall: 00186  DCall: 2 [66.234.228.170:4569]
USERNAME: QSa55JPy58
DATE TIME   : 156437288
REFRESH : 60
APPARENT ADDRES : IPV4 x.x.x.50:13081
 
 gw1*CLI
 Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
 
Timestamp: 00015ms  SCall: 2  DCall: 00186 [66.234.228.170:4569]
 Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass:
 HANGUP
Timestamp: 09779ms  SCall: 00518  DCall: 0 [66.234.228.170:4569]
 
 Output from tcpdump:
 22:02:48.246092 x.x.com.4569  170-228-234-66.cosmoweb.net.4569: udp 12 (DF)
 [tos 0x10]
 22:03:18.597719 170-228-234-66.cosmoweb.net.4569  x.x.com.13081: udp 84
 (DF)
 22:03:20.601668 170-228-234-66.cosmoweb.net.4569  x.x.com.13081: udp 84
 (DF)
 22:03:28.406522 170-228-234-66.cosmoweb.net.4569  x.X.com.13081: udp 12
 (DF)
 22:03:30.406566 170-228-234-66.cosmoweb.net.4569  x.x.com.13081: udp 12
 (DF)
 22:03:30.601889 170-228-234-66.cosmoweb.net.4569  X.X.com.13081: udp 84
 (DF)
 22:03:38.236056 X.x.com.4569  170-228-234-66.cosmoweb.net.4569: udp 28 (DF)
 [tos 0x10]
 22:03:38.246584 170-228-234-66.cosmoweb.net.4569  x.x.com.4569: udp 52 (DF)

The last two look like an outgoing call which you initiated.

You have to distinguish between incoming and outgoing calls, they are
two entirely different scenarios, especially when you traverse NAT.

An incoming call is -- from the viewpoint of your NAT router -- a
response to your earlier registration. So, as far as the NAT router is
concerned, you are the one who originated the incoming call by calling
out first making the registration request.

NAT uses different ports in order to map different streams going to
the same target port. In your example, the registration request is
mapped to 13081 but it will nevertheless reach Voicepulse's server on
port 4569. When you get an incoming call from Voicepulse, then as a
result of that registration having arrived from 13081, Voicepulse will
come in at your NAT router on 13081 even though it will have used 4569
on its own outbound interface. This is what allows your NAT router to
figure out that this is a response to your earlier registration and
that the stream is to be sent to your Asterisk server.

There could still be another IAX device on your network also having
registered with Voicepulse. Your NAT router would have mapped that to
another port, for example 1. Then Voicepulse would send a call for
that device to port 1 and your NAT router would then know that the
call is meant for the other IAX device and not your Asterisk server
whose registration was mapped to 13081. This is how NAT works.

Now when you make an outgoing calls, then that call may well use port
4569 on the WAN side of your NAT router and response traffic would
then come in on port 4569.

What I believe may be the problem you are facing is that your NAT
router may perceive the NAT mapping and the fact that your Asterisk
server is in a DMZ as a conflict. When you get the incoming call on
port 13081, the NAT router may not properly map it according to the
NAT mapping table but it may give the DMZ rule priority and simply
pass the traffic on to your Asterisk server in the DMZ that is to say
it will not be mapped back to 4569 but it will come in on port 13081
which your Asterisk server isn't configured to recognise as an IAX
port. It doesn't know anything about the mapping table of your NAT
router, so it won't map it back to 4569. After all, this is the job of
your NAT router, but it seems that for some reason it didn't do that
job.

I have seen this with quite a few software DMZs. In most cases this
can be solved by taking the Asterisk server out of the DMZ and use
individual port forwarding for those ports that you want to be
exposed, like 5038 for remote management for example, and not forward
4569. This will allow the NAT router and the IAX register 

RE: [Asterisk-Users] windows messenger

[Whisker, Peter]

The SIP client in Windows Messenger 5.0 seems to work fine with Asterisk
though.

Peter

-Original Message-
From: Robert Rozman [mailto:[EMAIL PROTECTED]
Sent: 11 October 2004 22:08
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] windows messenger


I'm not sure if I remember right, but I think that 4.7v of Windows Messenger
did work in presence sense, but as I told I'm not sure.

Regards,

Robert.
- Original Message - 
From: Bill Seddon [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Monday, October 11, 2004 6:13 PM
Subject: RE: [Asterisk-Users] windows messenger


 Asterisk doesn't support MSN9 the protocol Windows Messenger (and MSN
 Messenger) uses to communicate with a messenger server such as MSN or
 Windows 2003 running the Live Conferencing server.

 It should be possible to write an MSN9 server independently of Asterisk
 since the information needed by such a server is available via the Manager
 API.

 Bill Seddon

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of shabanip
 Sent: October 11, 2004 4:55 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] windows messenger

 is it possible to windows messenger clients of an asterisk server to chat
 (text chat) with each other?
 what about the status presence? is it possible to each windows messenger
 client of an asterisk server to see the presence on other clients?
 if not, what is missing in asterisk?



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Re: [Asterisk-Users] mISDN, CAPI, ISDN ???

[Massimo De Nadal]

Dave Cotton wrote:
On Tue, 2004-10-19 at 14:58 +0200, [EMAIL PROTECTED] wrote:
 

I've just used chan_capi it's very easy to use with Fritz!Cards and
therefore I like it ;-)
   

Worked straight out of the box on an AVM C2, hope it does the same with
2 Fritz!Cards in the same machine.
 

Sadly no.
If you want to use 2 fritz! in the same box you have to do a little hack 
with the drivers.
http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO

maxx
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[Asterisk-Users] ExtensionState

[Joseph]

I would like to find a way to list active extensions
with either the manager api or an agi script.

Using ExtensionState in the manager api I can't seem to get the syntax
right.

I tried the show channels with the exec command and it did not seem to
work.

And I tried Channel Status with the agi and get a -1 answer.

Really, what I want is a way to determine what phones are not use,
than make a call file and call them in preparation to do a broadcast
message.

Thanks for any tips.

-- 
respectfully, Joseph ===
-= **  =

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RE: [Asterisk-Users] mISDN, CAPI, ISDN ???

[Erwan DESVERGNES]

I don't know for the C2 but for the USB one it doesn't. 
AVM says it's normal.


-Message d'origine-
De : Dave Cotton [mailto:[EMAIL PROTECTED] 
Envoyé : mardi 19 octobre 2004 15:53
À : Asterisk List
Objet : Re: [Asterisk-Users] mISDN, CAPI, ISDN ???

On Tue, 2004-10-19 at 14:58 +0200, [EMAIL PROTECTED] wrote:
 I've just used chan_capi it's very easy to use with Fritz!Cards and
 therefore I like it ;-)

Worked straight out of the box on an AVM C2, hope it does the same with
2 Fritz!Cards in the same machine.


-- 
Dave Cotton [EMAIL PROTECTED]

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RE: [Asterisk-Users] Working Asterisk With Vonage

[Jay Milk]

Looks like you're dialing on a zap channel, no?

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] 
 Sent: Tuesday, October 19, 2004 6:15 AM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Working Asterisk With Vonage
 
 
 Hi ! 
 I have been working on making my asterisk server work with Vonage 
 services. I have been able to recieve calls on my asterisk 
 machine but i 
 couldnt call through that account to other people. Means if i 
 call a zap 
 channel and then dial 1 314 652 ... then i get an error like 
 
 Executing Dial(Zap/3-1, SIP/dialled 
 number@sphone.vopr.vonage.net:5061) 
 in new stack
 -- Called dialled number@sphone.vopr.vonage.net:5061
 -- Got SIP response 404 Not Found back from 216.115.25.198
 -- SIP/sphone.vopr.vonage.net-ec6e is circuit-busy
   == Everyone is busy at this time
 -- Executing Hangup(Zap/3-1, ) in new stack
   == Spawn extension (local, 192512100488, 2) exited non-zero 
 on 'Zap/3-1'
 -- Hungup 'Zap/3-1'
 
 
 whether i dial any number ... i get the same response... and 
 always ... 
 Can anyone guess what might be the problem ? 
 in sip .conf my settings are :
 
 register = username:password@sphone.vopr.vonage.net:5061
 
 [sphone.vopr.vonage.net]
 type = peer
 fromuser = username
 secret = password
 host = asterisk machine ip:5070 
 fromdomain=sphone.vopr.vonage.net dtmfmode=rfc2833 nat = yes 
 canreinvite=no
 
 In extensions.conf i have done : 
 
 exten = _1.,1,Dial,SIP/[EMAIL PROTECTED]:5061,tr
 exten = _1.,2,Hangup
 
 Please help me in this reagard.
 
 Regards ,
 
 Usman.
 
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RE: [Asterisk-Users] mISDN, CAPI, ISDN ???

[Erwan Desvergnes]

Seem It doesn't work for the USB one. And for the pci one, the current
drivers it's not then same.

-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Massimo De
Nadal
Envoyé : mardi 19 octobre 2004 16:24
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] mISDN, CAPI, ISDN ???



Dave Cotton wrote:

On Tue, 2004-10-19 at 14:58 +0200, [EMAIL PROTECTED] wrote:
  

I've just used chan_capi it's very easy to use with Fritz!Cards and
therefore I like it ;-)



Worked straight out of the box on an AVM C2, hope it does the same with
2 Fritz!Cards in the same machine.
  

Sadly no.
If you want to use 2 fritz! in the same box you have to do a little hack 
with the drivers.
http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO

maxx


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Re: [Asterisk-Users] mISDN, CAPI, ISDN ???

[Massimo De Nadal]

only avm active cards permit multiple installation straight forward (b1, 
c2 and c4)
with fritz! pci you can do the hack mentioned above, for  fritz! usb you 
can't  install more then one.
This limitiation is due to avm drivers design, they choose to allow 
multiple installation only on hi-end boards.

Erwan DESVERGNES wrote:
I don't know for the C2 but for the USB one it doesn't. 
AVM says it's normal.

-Message d'origine-
De : Dave Cotton [mailto:[EMAIL PROTECTED] 
Envoyé : mardi 19 octobre 2004 15:53
À : Asterisk List
Objet : Re: [Asterisk-Users] mISDN, CAPI, ISDN ???

On Tue, 2004-10-19 at 14:58 +0200, [EMAIL PROTECTED] wrote:
 

I've just used chan_capi it's very easy to use with Fritz!Cards and
therefore I like it ;-)
   

Worked straight out of the box on an AVM C2, hope it does the same with
2 Fritz!Cards in the same machine.
 

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[Asterisk-Users] test-driving G.729?

[Roy Sigurd Karlsbakk]

hi all
we're setting up a rather large end-user VoIP system, and due to 
pressure from norwegian telephony authorities, we consider choosing 
something instead of G.711A, possibly G.729.

does anyone know if it is possible to test-drive G.729 without paying 
Digium for it?

roy
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Re: [Asterisk-Users] mISDN, CAPI, ISDN ???

[Massimo De Nadal]

yes, It's not the same,
but applying the same hack to newer drivers it's not so difficult, 
almost for pci fritz!

Erwan Desvergnes wrote:
Seem It doesn't work for the USB one. And for the pci one, the current
drivers it's not then same.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Massimo De
Nadal
Envoyé : mardi 19 octobre 2004 16:24
À : Asterisk Users Mailing List - Non-Commercial Discussion
Objet : Re: [Asterisk-Users] mISDN, CAPI, ISDN ???
 

Sadly no.
If you want to use 2 fritz! in the same box you have to do a little hack 
with the drivers.
http://www.quiss.org/caiviar/Two-Fritzcards-HOWTO

maxx
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[Asterisk-Users] Spandsp debug log question

[Wilson Pickett]

I have been using this testing various fax machines. I have one source
that always works, Jfax. When I send my self a fax, it always gets
converted and sent.

Today I was passing the basement sysconsole and saw a bunch of debug
stuff slide by for a fax that seems to give trouble. I know l'm
supposed to get a log of this and send it in. How do I get spandsp to
record a log somewhere, or is it already? I can't see where.
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RE: [Asterisk-Users] test-driving G.729?

[Deon Rodden]

I personally think for a codec that's almost 1/3 the size of ULaw, the
quality is great. I consider ULaw above telephone quality, and g729 to be at
telephone quality.

But just 5 minutes ago I moved a user over to g729a. Changed the
SIP000.cnf file for the Cisco phone, but forgot to change the dtmfmode
in sip.conf from inband to rfc2833 and Asterisk wigged out with 300,000
messages about dtmf and such. Once I fixed that, he said the quality was
horrible. He could definitely hear the difference and hated it. Although I
could hear him fine, sounded good enough to me. Not sure what that was
about. I guess mileage will vary.

Have you looked into that open-source implementation of G729? There was
something on the WIKI about 3 different implementations of it. One being
where you paid license per channel fees, one that was free/open source, and
another I can't remember. Check the WIKI.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roy Sigurd
Karlsbakk
Sent: Tuesday, October 19, 2004 10:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] test-driving G.729?

hi all

we're setting up a rather large end-user VoIP system, and due to 
pressure from norwegian telephony authorities, we consider choosing 
something instead of G.711A, possibly G.729.

does anyone know if it is possible to test-drive G.729 without paying 
Digium for it?

roy

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Re: [Asterisk-Users] test-driving G.729?

[Andrew Kohlsmith]

On October 19, 2004 10:47 am, Roy Sigurd Karlsbakk wrote:
 does anyone know if it is possible to test-drive G.729 without paying
 Digium for it?

You're too cheap to blow $20 (I assume you need 2 licenses) on a test?  
Seriously it's not that expensive and if it doesn't work it doesn't work -- I 
piss away $20 on beer and wings the odd time, or a few hours of pool...

-A.
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[Asterisk-Users] Problem with NFAS trunkgroups

[Tony Mountifield]

Anyone here know about NFAS trunkgroups?

I have a TE405P card with spans 1, 3 and 4 connected to T1s. I have
happily had NI2 PRI running on them with each trunk having its own
D-channel. Using v1-0 from CVS.

/etc/zaptel.conf has the following:

span=1,1,0,esf,b8zs
span=2,0,0,esf,b8zs
span=3,0,0,esf,b8zs
span=4,0,0,esf,b8zs
bchan=1-23
dchan=24
bchan=25-47
dchan=48
bchan=49-71
dchan=72
bchan=73-95
dchan=96
loadzone = us
defaultzone=us

For non-NFAS, /etc/asterisk/zapata.conf has the following:

[trunkgroups]
#nothing
[channels]
switchtype = national
signalling = pri_cpe
group = 1
channel = 49-71,73-95
group = 2
channel = 1-23

With this config, I could successfully make calls on all B-channels.

Now I need, if possible, to use NFAS. I will be using it on four trunks,
with one primary and one secondary D-channel, and so 94 B-channels. To
test this, the telco has set up NFAS across two of the current trunks,
spans 1 and 3, with primary D-channel on span 1 (chan 24) and secondary
on span 3 (chan 72). Span 2 is still not connected, and Span 4 is a
standalone trunk.

I have changed /etc/asterisk/zapata.conf as follows:

[trunkgroups]
trunkgroup = 1,24,72
spanmap = 1,1,1
spanmap = 3,1,3
[channels]
switchtype = national
signalling = pri_cpe
group = 1
channel = 1-23,49-71
group = 2
channel = 73-95

What I find now is that I can make a call on channel 1, but no audio is
passed. I cannot make a call at all on channel 49. I can still make
calls on channels 73 to 95 and audio IS passed correctly.

Showing the first span gives me the following:

host1*CLI pri show span 1
Primary D-channel: 24
Status: Provisioned, Up, Active
Secondary D-channel: 72
Status: Provisioned, Up, Standby
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0
Switchtype: National ISDN
Type: CPE
Window Length: 0/7
Sentrej: 0
SolicitFbit: 0
Retrans: 0
Busy: 0
Overlap Dial: 0

which looks correct. If the telco takes the primary D-channel down, the
secondary changes from Standby to Active and I get a message to say it
has switched.

But this doesn't look right:

host1*CLI pri show span 3
No PRI running on span 3
host1*CLI 

Can anyone shed some light on all this?

Thanks in advance!
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] test-driving G.729?

[Steve Underwood]

Deon Rodden wrote:
I personally think for a codec that's almost 1/3 the size of ULaw, the
quality is great. I consider ULaw above telephone quality, and g729 to be at
telephone quality.
 

uLaw *is* telephone quality. Its what the PSTN uses. G.729 is much 
inferior, but at 1/8th the size (not 1/3rd unless you count all the RTP 
overhead). Its a trade-off. If you can't hear the difference, see a 
nurse and get the wax out of your ears :-)

G.729 is OK in a quite room. If there is a lot of background noise it is 
awful. Might that explain the difference in perception?

Regards,
Steve
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RE: [Asterisk-Users] test-driving G.729?

[Kanuri, Seshu (Company IT)]

/SNIP/
 Have you looked into that open-source implementation of G729? There
was something on the WIKI about 3 different implementations of it. One
being where you paid license per channel fees, one that was 
free/open source, and another I can't remember. Check the WIKI.
/SNIP/

There is no such thing as Open Source G729. Any such implementation is
called a Hack, if that is what one would like to call it. There are
sevral such hacks available but none of them really work. I tried them.
The Audio quality is miserable with such implementations.

Seshu Kanuri 

 
NOTICE: If received in error, please destroy and notify sender.  Sender does not waive 
confidentiality or privilege, and use is prohibited. 
 
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[Asterisk-Users] Re: Problem with NFAS trunkgroups

[Tony Mountifield]

I wrote:
 Anyone here know about NFAS trunkgroups?

Just a little more info (please see original message for main details):

 I have changed /etc/asterisk/zapata.conf as follows:
 
 [trunkgroups]
 trunkgroup = 1,24,72
 spanmap = 1,1,1
 spanmap = 3,1,3
 [channels]
 switchtype = national
 signalling = pri_cpe
 group = 1
 channel = 1-23,49-71
 group = 2
 channel = 73-95
 
 What I find now is that I can make a call on channel 1, but no audio is
 passed. I cannot make a call at all on channel 49. I can still make
 calls on channels 73 to 95 and audio IS passed correctly.

I have now taken the logical span off the spanmap lines:

spanmap = 1,1
spanmap = 3,1

That seems to have fixed the lack of audio when calling on channel 1, but
Asterisk still tells me that channel 49 is not available.

Some insight would be greatly appreciated!

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] Almost there--Remote connection

[Ferguson, Michael]

G'Day All;

Greetings and best wishes. I need some help as follows:

My Grandstream 100 is at a remote location on broadband and connects to
my * server else where.
From a POST line I dial the 3 to the * server and selects the ext # of
the remote GS100 IP phone.
The GS100 rings. When answered I can clearly hear everything coming from
the phone that's calling in.
The caller cannot hear anything coming from the GS100 IP phone.

If I make a call out from the GS100 to a POTS #, the POTS number rings.
Upon answering, the GS100 can also hear everything from the POTS phone
but the POTS phone is not hearing anything from the GS100.

I believe the phone is setup right.

The * server is behind a firewall and I have opened ports 
1-10100
5060
5004
4569

So it seems that my something is not allowing signal from the GS100 IP
phone out but is allowing signal in.

Any thoughts one where/what I should be modifying?

Thanks much.  
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Re: [Asterisk-Users] GSM to g729 Conversion

[Matthew Boehm]

States right here:

http://www.voip-info.org/tiki-index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk

Asterisk can play anything it has a format and codec for. Including wav,
gsm, g729, g726, wav49 all of which can be used for Playback and Background.

So, how can you make g729 files for Playback and Background?

Thanks,
Matthew

- Original Message - 
From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Tuesday, October 19, 2004 8:54 AM
Subject: RE: [Asterisk-Users] GSM to g729 Conversion


You are mixing oranges and apples here i guess. G729 is a Media Transmission
Protocol Codec the other is a Compressed Audio File format.

There are no .g729 audio files as far as I know.

Seshu Kanuri



  _

From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Victor Cartes
Sent: Monday, October 18, 2004 3:39 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] GSM to g729 Conversion


Hi!

Does anybody know how to convert .gsm file format to .g729 in order to use
it for an IVR system?

Thanks in advance.

Vïctor


NOTICE: If received in error, please destroy and notify sender.  Sender does
not waive confidentiality or privilege, and use is prohibited.








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RE: [Asterisk-Users] Almost there--Remote connection

[Karl Dyson]

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 16:18
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Almost there--Remote connection
 

[snip]

 
 The * server is behind a firewall and I have opened ports 
 1-10100 5060
 5004
 4569
 

IIRC, SIP uses 1-2 by default. Have you changed this to
1-10100?

Cheers,

Karl


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RE: [Asterisk-Users] GSM to g729 Conversion

[Brian West]

Record them from a phone that speaks g729 right to raw .g729 files.

bkw

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Matthew Boehm
 Sent: Tuesday, October 19, 2004 10:21 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] GSM to g729 Conversion
 
 States right here:
 
 http://www.voip-info.org/tiki-
 index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk
 
 Asterisk can play anything it has a format and codec for. Including wav,
 gsm, g729, g726, wav49 all of which can be used for Playback and
 Background.
 
 So, how can you make g729 files for Playback and Background?
 
 Thanks,
 Matthew
 
 - Original Message -
 From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 [EMAIL PROTECTED]
 Sent: Tuesday, October 19, 2004 8:54 AM
 Subject: RE: [Asterisk-Users] GSM to g729 Conversion
 
 
 You are mixing oranges and apples here i guess. G729 is a Media
 Transmission
 Protocol Codec the other is a Compressed Audio File format.
 
 There are no .g729 audio files as far as I know.
 
 Seshu Kanuri
 
 
 
   _
 
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Victor
 Cartes
 Sent: Monday, October 18, 2004 3:39 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] GSM to g729 Conversion
 
 
 Hi!
 
 Does anybody know how to convert .gsm file format to .g729 in order to use
 it for an IVR system?
 
 Thanks in advance.
 
 Vïctor
 
 
 NOTICE: If received in error, please destroy and notify sender.  Sender
 does
 not waive confidentiality or privilege, and use is prohibited.
 
 
 
 
 --
 --
 
 
 
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RE: [Asterisk-Users] Almost there--Remote connection

[Ferguson, Michael]

The 1-10100 was given to me by a prior post so I really do not know.
I will change the forewall to allow 1-2 and see if it works.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Dyson
Sent: Tuesday, October 19, 2004 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection


 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 16:18
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Almost there--Remote connection
 

[snip]

 
 The * server is behind a firewall and I have opened ports 
 1-10100 5060
 5004
 4569
 

IIRC, SIP uses 1-2 by default. Have you changed this to
1-10100?

Cheers,

Karl


This e-mail has been scanned for all viruses by Star. The
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RE: [Asterisk-Users] Almost there--Remote connection

[Deon Rodden]

My firewall script has something to the effect of:

# Allow Existing traffic through
-A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT

# Incoming VOIP Ports
-A INPUT -m state --state NEW -m tcp -p tcp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 2727:2727 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 4569:4569 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5060:5061 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 1:2 -j ACCEPT

That's for IAX2 and SIP.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Dyson
Sent: Tuesday, October 19, 2004 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection

 -Original Message-
 From: [EMAIL PROTECTED] 
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 16:18
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Almost there--Remote connection
 

[snip]

 
 The * server is behind a firewall and I have opened ports 
 1-10100 5060
 5004
 4569
 

IIRC, SIP uses 1-2 by default. Have you changed this to
1-10100?

Cheers,

Karl


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[Asterisk-Users] Fax over IP doesn't works

[Miroslav Nachev]

   Hi,

   We try to send Fax through IP Network but without success. The
other party use NetCentrex SoftSwitch and our communication protocol
between us is H.323 (OpenH323). The error that the other party receive
is: bearer capability not imoplemented.

   Is it possible to send Fax using Asterisk to the other party
through IP network? What T.38 and Asterisk?


   Regards,
   Miro.

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Re: [Asterisk-Users] Problem with NFAS trunkgroups

[Andrew Kohlsmith]

On October 19, 2004 11:01 am, Tony Mountifield wrote:
 Anyone here know about NFAS trunkgroups?

Yes, I worked with them in the dialup world on AS5248s and MaxTNTs.

 I have a TE405P card with spans 1, 3 and 4 connected to T1s. I have
 happily had NI2 PRI running on them with each trunk having its own
 D-channel. Using v1-0 from CVS.
 Now I need, if possible, to use NFAS. I will be using it on four trunks,
 with one primary and one secondary D-channel, and so 94 B-channels. To
 test this, the telco has set up NFAS across two of the current trunks,
 spans 1 and 3, with primary D-channel on span 1 (chan 24) and secondary
 on span 3 (chan 72). Span 2 is still not connected, and Span 4 is a
 standalone trunk.

I am not aware of NFAS working on any Digium equipment.

-A.
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Re: [Asterisk-Users] Setting CallerID on UK BRI line

[Linus Surguy]

yep, my mobile displays caller id for other numbers - and it even works
perfectly displaying caller id information set by a cheap ISDN pbx on the
*same* ISDN line as the Asterisk box. Curious. Even without setting a
callerid on the outgoing calls I get No Caller ID on my mobile (or other
phones  - including other BT lines). BT are not withholding a number and I
can change the callerID presented on the other phone system and it works
perfectly.
Strange, I will investigate more later.
Remembering that you can only set caller ID to numbers that have been issued 
to you by BT, assuming you have a valid telephone number of, for example, 
0118 321 1234

Try:
SetCallerID(4)
SetCallerID(34)
SetCallerID(234)
SetCallerID(11234)
SetCallerID(211234)
Normally you find it's either the 6 digit version or the single digit 
version that works with BT.

Linus
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RE: [Asterisk-Users] AGI RECORD FILE BUG!

[Steven Critchfield]

On Tue, 2004-10-19 at 23:14 +1000, Simon Smith wrote:
 Oh ok, so there are other threads with Recording...Where are they? 

You put this message into a thread about AGI Get Data', You put this in
a thread about video door phones. You put it in a thread in -dev with
subject line of Unusual problem. You put it in the -dev thread on
skype. 

So you have posted 2 new threads about the problem, one here and one in
-dev. You have also improperly injected your problem into 4 other
threads. 

In the past, all users on -dev where also subscribed here. It isn't so
much the case anymore. 

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Steven
 Critchfield
 Sent: Tuesday, 19 October 2004 11:06 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: RE: [Asterisk-Users] AGI RECORD FILE BUG!
 
 You also are having a problem realizing that we have now seen your message
 SEVERAL times and shoved into other threads that are irrelavent to recording
 or AGI. You are not helping yourself by doing this. 
 

-- 
Steven Critchfield [EMAIL PROTECTED]

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RE: [Asterisk-Users] Almost there--Remote connection

[Ferguson, Michael]

Thanks. I think that's Iptables. No?
I have a hardware firewall.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Deon
Rodden
Sent: Tuesday, October 19, 2004 11:35 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Almost there--Remote connection


My firewall script has something to the effect of:

# Allow Existing traffic through
-A INPUT -m state --state ESTABLISHED,RELATED -j ACCEPT

# Incoming VOIP Ports
-A INPUT -m state --state NEW -m tcp -p tcp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5036:5045 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 2727:2727 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 4569:4569 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 5060:5061 -j ACCEPT
-A INPUT -m state --state NEW -m udp -p udp --dport 1:2 -j
ACCEPT

That's for IAX2 and SIP.

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl Dyson
Sent: Tuesday, October 19, 2004 11:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Almost there--Remote connection

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of 
 Ferguson, Michael
 Sent: 19 October 2004 16:18
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Almost there--Remote connection
 

[snip]

 
 The * server is behind a firewall and I have opened ports 
 1-10100 5060
 5004
 4569
 

IIRC, SIP uses 1-2 by default. Have you changed this to
1-10100?

Cheers,

Karl


This e-mail has been scanned for all viruses by Star. The
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Re: [Asterisk-Users] FYI - Zoom X5v built-in VoIP DSL router

[Scott Wolf]

We just put in for a demo of one of these today. Do you know if it does
any sort of QOS or traffic shapping. The specs don't seem to mention it.

Scott Wolf

Ben Merrills wrote:

  
  
  
  
  
  
  Just thought I
would let the
list know, as we got our pre release versions today of the new Zoom X5
that
supports VoIP. The device comes with an RJ11 phone socket on the back
and lets
you configure your ADSL router to become a SIP phone (using your
existing PSTN
phone). Better still, it also allows you to switch the phone between
landline
and SIP, and does it automatically for incoming calls.
  
  No idea what the
price of
these devices will be when they hit the shops, but setting one up
today, if
anyone thinks it would be helpful I dont mind doing a little review of
the hardware once its tested.
  
  Model Number is
5565
  
  Cheers,
  Ben Merrills
  
  
  Griffin
Internet
  T: 0870
8040862
  F: 0870
8040805
  W: www.griffin.com
  
  
  

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Re: [Asterisk-Users] Fax over IP doesn't works

[Pedro Howat Rodrigues]

Hi ,
I tried this a lot, but with no sucess , even in a local network , there 
is always some loss and you receive only chunks of the original file .

Pedro.
Miroslav Nachev wrote:
  Hi,
  We try to send Fax through IP Network but without success. The
other party use NetCentrex SoftSwitch and our communication protocol
between us is H.323 (OpenH323). The error that the other party receive
is: bearer capability not imoplemented.
  Is it possible to send Fax using Asterisk to the other party
through IP network? What T.38 and Asterisk?
  Regards,
  Miro.
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[Asterisk-Users] chan_mISDN

[Erwan DESVERGNES]

Did someone have succeed to compile
chan_misdn???



Ive got an error when in try to compile



chan_misdn.c:68: error:
`__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared
here (not in a function)





thanks



_

Erwan
 Desvergnes
- ANDIUM -

82/86 rue Château Gaillard

69100 Villeurbanne



Tel. 04 3743 44
45 / Fax 04 37 43 44 44

E-mail: [EMAIL PROTECTED]








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[Asterisk-Users] Re: Problem with NFAS trunkgroups

[Tony Mountifield]

In article [EMAIL PROTECTED],
Andrew Kohlsmith [EMAIL PROTECTED] wrote:
 On October 19, 2004 11:01 am, Tony Mountifield wrote:
  Anyone here know about NFAS trunkgroups?
 
 Yes, I worked with them in the dialup world on AS5248s and MaxTNTs.

OK, I was too vague! What I really want is someone who has successfully
set up Asterisk on NSAF trunkgroups.

  I have a TE405P card with spans 1, 3 and 4 connected to T1s. I have
  happily had NI2 PRI running on them with each trunk having its own
  D-channel. Using v1-0 from CVS.
  Now I need, if possible, to use NFAS. I will be using it on four trunks,
  with one primary and one secondary D-channel, and so 94 B-channels. To
  test this, the telco has set up NFAS across two of the current trunks,
  spans 1 and 3, with primary D-channel on span 1 (chan 24) and secondary
  on span 3 (chan 72). Span 2 is still not connected, and Span 4 is a
  standalone trunk.
 
 I am not aware of NFAS working on any Digium equipment.

But that's what [trunkgroups] is all about!

The comment near the top of zapata.conf.sample is:

[trunkgroups]
;
; Trunk groups are used for NFAS or GR-303 connections.
;

Or are you saying that NFAS support in Asterisk is not complete yet?

Certainly it seems able to handle primary and secondary d-channels,
it just seems that there is something I haven't set up correctly,
because the documentation is a little thin.

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] Patch: Inbound-only busydetect

[yamamoto]

Hi Marconi,
I couldn't access URL.
I want to try your patch.
Marconi Rivello wrote:
Where to get it:
http://www.carcara.lncc.br/marconi/mr_busydetect.patch
tatsuya
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[Asterisk-Users] Wellgate SIP product users - voice your concern!

[Vahan Yerkanian]

After long email communication with May Lin, Wellgate's International 
Sales Dept./Project Manager, I was asked to supply them with a list of 
email addresses of people with the same FXO/FXS sip version hardware's 
bugs related to registration and anything else. They're trying to find 
out the number of people affected with the firmware bugs.

If you are one of those unfortunate to buy their buggy products, pls 
send an email to [EMAIL PROTECTED], May Lin, and voice your concerns!

regards,
Vahan Yerkanian
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RE: [Asterisk-Users] Fax over IP doesn't works

[Yiannis Costopoulos]

Well,

assuming that some of these CODECS do error correction and drop any
information that hasn't come through instead of doing error detection and
request to re-transmit the lost information, is somewhat expected. Are there
any Fax over IP protocols?

Yiannis.

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of Pedro Howat
 Rodrigues
 Sent: 19 October 2004 15:53
 To: Miroslav Nachev; Asterisk Users Mailing List - Non-Commercial
 Discussion
 Subject: Re: [Asterisk-Users] Fax over IP doesn't works


 Hi ,

 I tried this a lot, but with no sucess , even in a local network , there
 is always some loss and you receive only chunks of the original file .

 Pedro.

 Miroslav Nachev wrote:

Hi,
 
We try to send Fax through IP Network but without success. The
 other party use NetCentrex SoftSwitch and our communication protocol
 between us is H.323 (OpenH323). The error that the other party receive
 is: bearer capability not imoplemented.
 
Is it possible to send Fax using Asterisk to the other party
 through IP network? What T.38 and Asterisk?
 
 
Regards,
Miro.
 
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Re: [Asterisk-Users] GSM to g729 Conversion

[Matthew Boehm]

There is no way to convert existing files to g729? The only reason we need
the licenses is to access voicemail since they are in GSM.  All our phones
have g729 built in. But if you try and access VM, you get that No coversion
for GSM to g729 error. But if all the voicemail sounds where in g729, then
we don't need the licenses.

Matthew

- Original Message - 
From: Brian West [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Tuesday, October 19, 2004 10:25 AM
Subject: RE: [Asterisk-Users] GSM to g729 Conversion


Record them from a phone that speaks g729 right to raw .g729 files.

bkw

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Matthew Boehm
 Sent: Tuesday, October 19, 2004 10:21 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [Asterisk-Users] GSM to g729 Conversion

 States right here:

 http://www.voip-info.org/tiki-
 index.php?page=Convert%20WAV%20audio%20files%20for%20use%20in%20Asterisk

 Asterisk can play anything it has a format and codec for. Including wav,
 gsm, g729, g726, wav49 all of which can be used for Playback and
 Background.

 So, how can you make g729 files for Playback and Background?

 Thanks,
 Matthew

 - Original Message -
 From: Kanuri, Seshu (Company IT) [EMAIL PROTECTED]
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 [EMAIL PROTECTED]
 Sent: Tuesday, October 19, 2004 8:54 AM
 Subject: RE: [Asterisk-Users] GSM to g729 Conversion


 You are mixing oranges and apples here i guess. G729 is a Media
 Transmission
 Protocol Codec the other is a Compressed Audio File format.

 There are no .g729 audio files as far as I know.

 Seshu Kanuri



   _

 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] On Behalf Of Victor
 Cartes
 Sent: Monday, October 18, 2004 3:39 PM
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] GSM to g729 Conversion


 Hi!

 Does anybody know how to convert .gsm file format to .g729 in order to use
 it for an IVR system?

 Thanks in advance.

 Vïctor
 

 NOTICE: If received in error, please destroy and notify sender.  Sender
 does
 not waive confidentiality or privilege, and use is prohibited.




 --
 --
 


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