Re: [asterisk-users] G.729 codec in pass-thru mode

2013-06-05 Thread Kamlesh Kumar
Matthew,
 
allow=all is defined in sip.conf for the ITSP's SIP peer. Additionally, ITSP 
supports g729 codec as we are able to send the traffic from other soft switch. 
In case g729 on asterisk box, as I mentioned earlier, call even doesn't go out 
of the asterisk box. Below extracts from log also indicate the same thing. 
 
[Jun  5 12:46:49] -- AGI Script Executing Application: (Dial) Options: 
(SIP/yyy.yyy.yyy.yyy/12127773456)
[Jun  5 12:46:49]   == Using SIP RTP CoS mark 5
[Jun  5 12:46:49] -- Couldn't call yyy.yyy.yyy.yyy/12127773456
[Jun  5 12:46:49]   == Everyone is busy/congested at this time (0:0/0/0)

Regards,
Kamlesh 
 
 Date: Tue, 4 Jun 2013 10:27:11 -0500
 From: mr...@imminc.com
 To: asterisk-users@lists.digium.com
 Subject: Re: [asterisk-users] G.729 codec in pass-thru mode
 
 Kamlesh Kumar wrote:
  
  SIP.conf
  [100]
  username=100
  secret=password
  type=friend
  host=dynamic
  nat=yes
  canreinvite=no
  insecure=port
  disallow=all
  allow=ulaw
  allow=alaw
  allow=g729
  context=asterisk
  qualify=no
 
 Is there also an allow=g729 line in sip.conf for the ITSP's SIP peer?
 
  SIP Trace: 
  201.xxx.xxx.xxx = SIP Softphone which originates the call 
  xxx.xxx.xxx.xxx = Asterisk server 
  yyy.yyy.yyy.yyy = ITSP 
  
  ...
  
  --- SIP read from UDP:yyy.yyy.yyy.yyy:5060 ---
  SIP/2.0 183 Session Progress
  Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK15380659;rport=5060
  From: 100 sip:1...@xxx.xxx.xxx.xxx;tag=as643c20b1
  To: sip:12127773...@yyy.yyy.yyy.yyy;tag=gK029aaa8c
  Call-ID: 07714ae4593feb5c3e42b3a01cf4a...@xxx.xxx.xxx.xxx
  CSeq: 102 INVITE
  Contact: sip:12127773...@yyy.yyy.yyy.yyy:5060
  Allow: 
  INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,PRACK,UPDATE,OPTIONS
  Content-Length:  234
  Content-Disposition: session; handling=required
  Content-Type: application/sdp
  v=0
  o=Sonus_UAC 24592 17457 IN IP4 yyy.yyy.yyy.yyy
  s=SIP Media Capabilities
  c=IN IP4 zzz.zzz.zzz.zzz
  t=0 0
  m=audio 21996 RTP/AVP 0 101
  a=rtpmap:0 PCMU/8000
  a=rtpmap:101 telephone-event/8000
  a=fmtp:101 0-15
  a=sendrecv
  a=maxptime:20
  -
  [Jun  3 13:11:31] --- (11 headers 11 lines) ---
  [Jun  3 13:11:31] Found RTP audio format 0
  [Jun  3 13:11:31] Found RTP audio format 101
  [Jun  3 13:11:31] Found audio description format PCMU for ID 0
  [Jun  3 13:11:31] Found audio description format telephone-event for ID 101
  [Jun  3 13:11:31] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 
  (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
  [Jun  3 13:11:31] Non-codec capabilities (dtmf): us - 0x1 
  (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 
  (telephone-event)
  [Jun  3 13:11:31] Peer audio RTP is at port zzz.zzz.zzz.zzz:21996
  [Jun  3 13:11:31] -- SIP/yyy.yyy.yyy.yyy-34d9 is making progress 
  passing it to SIP/100-34d8
  [Jun  3 13:11:31] Audio is at xxx.xxx.xxx.xxx port 26042
  [Jun  3 13:11:31] Adding codec 0x4 (ulaw) to SDP
  [Jun  3 13:11:31] Adding non-codec 0x1 (telephone-event) to SDP
 
 This response from the ITSP says that only u-law may be used for the call.
 Please contact the ITSP and confirm that they actually support the G.729 
 codec.
 
 Regards,
 
 Matthew Roth
 InterMedia Marketing Solutions
 Software Engineer and Systems Developer
 
 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
 
 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
  --
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

[asterisk-users] incoming DAHDI Channel explained

2013-06-05 Thread Thorsten Göllner

Hi,

I use a Sangoma A104d-Card (with 4 x germany E1). I process some calls 
via an AGI-Script. When parsing the AGI-Variables I can see one that 
look like that:


[agi_channel] = DAHDI/i3/211123456-89c

What hat do the values mean in detail, please?

DAHDI : this is clear
i3 : does it mean, that the call comes in via E1-Port 3?
211123456 : Incoming-Call Caller-ID
-89c : ?

WANPIPE Release: 7.0.1
DAHDI Version: 2.6.2 Echo Canceller: HWEC
libpri version: 1.4.12

Best regards
-Thorsten-


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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


[asterisk-users] sendmail when no response

2013-06-05 Thread Salaheddine Elharit
hello list,

i need  your help please regarding send mail i use astreisk 1.4;

i try to send mail when no response like below


exten = 5xx,1,Dial(SIP/223, 10)
exten = 5xx,n,system(echo test ${DNIS} Email| mail -s 'Call failed'
myadresseem...@gmail.com)

when i launch the CLI i found :

You have new mail in /var/spool/mail/root

i check the root and i found :

Return-Path: root
Received: (from root@localhost)
by localhost.localdomain (8.13.1/8.13.1/Submit) id r55B3Deh023821;
Wed, 5 Jun 2013 11:03:13 GMT
Date: Wed, 5 Jun 2013 11:03:13 GMT
From: root root
Message-Id: 201306051103.r55B3Deh023821@localhost.localdomain
To: failed, myadresseem...@gmail.com
Subject: Call

test Email

--r55B3Dei023821.1370430193/localhost.localdomain--

could you please tell me how to do in order to send email to my address
gmail for example

thanks and regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] sendmail when no response

2013-06-05 Thread jg
Your problem looks more like an MTA configuration problem. You need at 
least a valid relay host.


jg

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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


Re: [asterisk-users] incoming DAHDI Channel explained

2013-06-05 Thread jg

Sangoma's tech support is probably the better source of information.

DAHDI: obviously DAHDI channel
i: incoming call
3: span 3 (not the port)
211123456: CLID, probably subject to filtering (see 
national/international prefix settings)

89c: internal counter (i.e. 2204 calls so far)

jg

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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


Re: [asterisk-users] G.729 codec in pass-thru mode

2013-06-05 Thread Matthew J. Roth
Kamlesh Kumar wrote:
 
 allow=all is defined in sip.conf for the ITSP's SIP peer. Additionally, ITSP
 supports g729 codec as we are able to send the traffic from other soft switch.


There must be some difference between your Asterisk servers.  Please set them
up for calling the ITSP with G.729 and provide the following CLI output from
both of them.  Be sure to preserve any differences when obscuring IP addresses
and label the output clearly as G.729 working and G.729 fails.

  CLI core show version
  CLI sip show settings
  CLI sip show peer 100
  CLI sip show peer ITSP's SIP peer
  CLI g729 show licenses

Regards,

Matthew Roth
InterMedia Marketing Solutions
Software Engineer and Systems Developer

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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


Re: [asterisk-users] incoming DAHDI Channel explained

2013-06-05 Thread Richard Mudgett
 Sangoma's tech support is probably the better source of information.
 
 DAHDI: obviously DAHDI channel
 i: incoming call

The 'i' is for ISDN not incoming call since it will be this way for outgoing 
calls as well.

 3: span 3 (not the port)
 211123456: CLID, probably subject to filtering (see
 national/international prefix settings)
 89c: internal counter (i.e. 2204 calls so far)

The other fields are pretty much as described by jg.

Richard

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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


Re: [asterisk-users] incoming DAHDI Channel explained

2013-06-05 Thread jg
Yes, my assumption was wrong and to make things worse, my CDR data 
clearly show that i cannot denote incoming calls.


Maybe it's time that I learn the rules as well:

Analog channels  do not seem to have a special identifier. The 1st call 
for analog channel 13 would be s.th. like DAHDI/13-1.


Outside calls via an ISDN connection with s.th. like 
DIAL(DAHDI/r2/08932168,..) would dial the number using DAHDI group 2 in 
a round robin fashion, but internally the channel would be s.th. like 
DAHDI/iX/08932168-abcd. The span X is not related to the dial group and 
depends on the configuration.


For a BRI device a single span has 2 channels, a PRI device up to 30. As 
far as channel variables go the actual channel does not seem to get 
reported, but this is not really necessary.


jg

--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
  http://www.asterisk.org/hello

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


Re: [asterisk-users] incoming DAHDI Channel explained

2013-06-05 Thread Mordechay Kaganer
B.H.

Hi!

On Wed, Jun 5, 2013 at 7:26 PM, jg webaccou...@jgoettgens.de wrote:

For a BRI device a single span has 2 channels, a PRI device up to 30. As
 far as channel variables go the actual channel does not seem to get
 reported, but this is not really necessary.


AFAIK, at least for AMI listeners, the real channel/span is reported
by DAHDIChannelEvent attributes:

'dahdichannel' reports the actual DAHDI channel number
'dahdispan' is a span number




 jg



-- 
משיח NOW!
Moshiach is coming very soon, prepare yourself!
יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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

Re: [asterisk-users] problem to install asterisk on vps digitalocean

2013-06-05 Thread Johan Wilfer
 2013/6/4 James Cloos cl...@jhcloos.com mailto:cl...@jhcloos.com
  t == troxlinux  xserverli...@gmail.com
 mailto:xserverli...@gmail.com writes:
 t I try to install asterisk on vps server , but fails when I want to
 t install dahdi
 
 There is no hardware for dahdi to use; you shouldn't need to install it.

2013-06-04 22:28, troxlinux skrev:
 if it is true I have not any hardware but I need help to solve it and
 I think it could serve other future

Depending on the type of VPS this can work or not. My general experience
of normal stock-VPS:es are that they are often oversold. And it will
create a lot of issues for you with voip.

On OpenVZ you can't load kernel-modules in the VE anyway, so it's no
idea to compile Dahdi. But if you control the HN, you can install it
there and give the VE access to it. In that case you should compile it
in the VE as well. The same applies to LXC.

On XEN/KVM I think it will work, but be sure to test the timing with
dahdi_test - you don't want the results to drop below 99.9. Try create
some i/o (for example compile asterisk) and run the test in another window.

Dahdi in VE's is a pain, but can work if you control the HN. You should
do development and testing on a dedicated server. After that, try move
to a VPS and try to fix the issues. The other way around is very hard.


-- 
Johan Wilfer


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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


[asterisk-users] Asterisk HA

2013-06-05 Thread Gopalakrishnan N
I was go through'ing the following links for HA,

https://wiki.asterisk.org/wiki/display/TOP/Failover+-+Linux - which doesn't
have file syncing.

https://www.johncahill.net/wiki/index.php/2_Node_Active/Passive_cluster -
this one has file syncing with pacemaker

Any other HA applications available or the lsyncd with pacemaker is good?

Regards
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

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