Re: [asterisk-users] No extension found ?

2012-04-20 Thread Michel Verbraak 

On 21-04-12 08:19, Olivier CALVANO wrote:

Hi

I have a small problems with incoming call.

I have a peer actually configured for outcall :


sip.conf:

[Trunk-Telco]
type=peer
host=domaineofmysupplier.net
outboundproxy=domaineofmysupplier.net
session-timers=originate
session-expires=7200
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
dtmfmode=rfc2833
disallow=all
allow=alaw
insecure=port,invite
context=incoming

This SIP account work for outgoing call. when i want receive call from
this sipplier, i have a "extension not found".

In extensions.conf for incoming:

[incoming]
 exten =>  _X.,1,Dial(IAX2/VoIP/${EXTEN},180,rt)

in dialplan show incoming, no problems i see the dialplan.

when i call, i have:

<--- SIP read from UDP://84.xx.xx.72:5060 --->
INVITE sip:331NUMNOFOUND@78.IPOFMYSERVER:5060 SIP/2.0
Record-Route:
Record-Route:
Record-Route:
Record-Route:
Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0
Via: SIP/2.0/UDP 172.16.21.67;branch=z9hG4bK10e4.bbf4c444.0
Via: SIP/2.0/UDP 172.16.20.119;branch=z9hG4bK10e4.9fe53c91.0
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=z9hG4bK00151747E2606DB6CA39464AF542
From: "+331MYCLID"
;tag=2RUVP51HBW3E1D1u0K4NFQC0QNAN31
To:
Call-ID: 60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
CSeq: 20114 INVITE
Contact:
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
Content-Type: application/sdp
Max-Forwards: 67
P-Asserted-Identity:
Supported: timer, replaces
Content-Length: 369
Min-SE: 90
Session-Expires: 300
P-Charging-Vector: icid-value="4f924d2c1e20abe1d@172.16.20.119"
X-PSN-Trunk: ME

v=0
o=- 18406958643964291255 1 IN IP4 172.16.21.11
s=session
c=IN IP4 84.xx.xx.34
t=0 0
m=audio 64296 RTP/AVP 8 18 4 0 105 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=fmtp:4 bitrate=6.3
a=rtpmap:0 PCMU/8000
a=rtpmap:105 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
a=nortpproxy:yes

<->
--- (25 headers 17 lines) ---
   == Using SIP RTP CoS mark 5
Sending to 84.xx.xx.72 : 5060 (no NAT)
Using INVITE request as basis request -
60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 101
Peer audio RTP is at port 84.xx.xx.34:64296
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format PCMU for ID 0
Found unknown media description format X-CCD for ID 105
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined
- 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 84.xx.xx.34:64296
Looking for 331NUMNOFOUND in default (domain 78.IPOFMYSERVER)

It is looking for the 331NUMNOFOUND in context named "default".
Do you have this context? Does the extension exists in the context?

Do you have a register line in your sip.conf for this external provider? 
In the register line you can specify the extensions/device to use in the 
sip.conf so it knows the right context to start in extensions.conf 
instead of the default context.


For example: register => username:passw...@sip.voipbuster.com/Trunk-Telco


<--- Reliably Transmitting (no NAT) to 84.xx.xx.72:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0;received=84.xx.xx.72



<>
[Apr 21 08:01:16] NOTICE[11906]: chan_sip.c:18527
handle_request_invite: Call from '' to extension '331NUMNOFOUND'
rejected because extension not found.






Regards,
Michel.

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[asterisk-users] No extension found ?

2012-04-20 Thread Olivier CALVANO 
Hi

I have a small problems with incoming call.

I have a peer actually configured for outcall :


sip.conf:

[Trunk-Telco]
type=peer
host=domaineofmysupplier.net
outboundproxy=domaineofmysupplier.net
session-timers=originate
session-expires=7200
qualify=yes
dtmf=rfc2833
nat=no
canreinvite=no
canredirect=yes
dtmfmode=rfc2833
disallow=all
allow=alaw
insecure=port,invite
context=incoming

This SIP account work for outgoing call. when i want receive call from
this sipplier, i have a "extension not found".

In extensions.conf for incoming:

[incoming]
exten => _X.,1,Dial(IAX2/VoIP/${EXTEN},180,rt)

in dialplan show incoming, no problems i see the dialplan.

when i call, i have:

<--- SIP read from UDP://84.xx.xx.72:5060 --->
INVITE sip:331NUMNOFOUND@78.IPOFMYSERVER:5060 SIP/2.0
Record-Route: 
Record-Route: 
Record-Route: 
Record-Route: 
Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0
Via: SIP/2.0/UDP 172.16.21.67;branch=z9hG4bK10e4.bbf4c444.0
Via: SIP/2.0/UDP 172.16.20.119;branch=z9hG4bK10e4.9fe53c91.0
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=z9hG4bK00151747E2606DB6CA39464AF542
From: "+331MYCLID"
;tag=2RUVP51HBW3E1D1u0K4NFQC0QNAN31
To: 
Call-ID: 60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
CSeq: 20114 INVITE
Contact: 
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY,
SUBSCRIBE, UPDATE
Content-Type: application/sdp
Max-Forwards: 67
P-Asserted-Identity: 
Supported: timer, replaces
Content-Length: 369
Min-SE: 90
Session-Expires: 300
P-Charging-Vector: icid-value="4f924d2c1e20abe1d@172.16.20.119"
X-PSN-Trunk: ME

v=0
o=- 18406958643964291255 1 IN IP4 172.16.21.11
s=session
c=IN IP4 84.xx.xx.34
t=0 0
m=audio 64296 RTP/AVP 8 18 4 0 105 101
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=fmtp:4 bitrate=6.3
a=rtpmap:0 PCMU/8000
a=rtpmap:105 X-CCD/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
a=nortpproxy:yes

<->
--- (25 headers 17 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 84.xx.xx.72 : 5060 (no NAT)
Using INVITE request as basis request -
60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
No matching peer for '+331MYCLID' from '84.xx.xx.72:5060'
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 0
Found RTP audio format 105
Found RTP audio format 101
Peer audio RTP is at port 84.xx.xx.34:64296
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found audio description format PCMU for ID 0
Found unknown media description format X-CCD for ID 105
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10d
(g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined
- 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 84.xx.xx.34:64296
Looking for 331NUMNOFOUND in default (domain 78.IPOFMYSERVER)

<--- Reliably Transmitting (no NAT) to 84.xx.xx.72:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 84.xx.xx.72;branch=z9hG4bK10e4.e7f23f11.0;received=84.xx.xx.72
Via: SIP/2.0/UDP 172.16.21.67;branch=z9hG4bK10e4.bbf4c444.0
Via: SIP/2.0/UDP 172.16.20.119;branch=z9hG4bK10e4.9fe53c91.0
Via: SIP/2.0/UDP 172.16.21.11:5060;branch=z9hG4bK00151747E2606DB6CA39464AF542
From: "+331MYCLID"
;tag=2RUVP51HBW3E1D1u0K4NFQC0QNAN31
To: ;tag=as53fc96aa
Call-ID: 60471500e217-4f924d2c-477df10c-66ea6f8-140732f@127.0.0.1
CSeq: 20114 INVITE
Server: Asterisk PBX 1.6.1.8
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0

<>
[Apr 21 08:01:16] NOTICE[11906]: chan_sip.c:18527
handle_request_invite: Call from '' to extension '331NUMNOFOUND'
rejected because extension not found.




a idea of the problems ?

My supplier use a lot of server, i thinkss that my asterisk don't link
IP of the incoming server to the extensions


thanks for your help
olivier

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Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6

2012-04-20 Thread Chad Wallace 
On Fri, 20 Apr 2012 13:17:01 -0700 (PDT)
bilal ghayyad  wrote:

> Well, I did make menuselect and I really found the XXX and did not
> get the ability to select the channel. So what could be the reason?

As Kevin said, you need to check the out put when you run ./configure.  
You could pipe it through less or copy-and-paste it into a text editor
to search it for anything about dahdi.  It should tell you what's wrong.


-- 

C. Chad Wallace, B.Sc.
The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0


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Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6

2012-04-20 Thread Kevin P. Fleming 

On 04/20/2012 03:17 PM, bilal ghayyad wrote:

Dear;

Well, I did make menuselect and I really found the XXX and did not get the 
ability to select the channel. So what could be the reason?


When you are in menuselect, looking at the 'channels' page, scroll the 
cursor down to chan_dahdi (marked with 'XXX'), and look at the bottom of 
the window/screen. In that area there will be information about the 
chan_dahdi dependencies that were or were not found by the the configure 
script. If you can copy and paste that information here, we can try to 
help you figure out what is going on. It's quite strange that 
codec_dahdi successfully built but chan_dahdi did not; the problem is 
likely not related to DAHDI, but due to some other dependency that 
chan_dahdi has. As I said before, what we really should be looking at is 
the configure script output that indicates what it was able to find and 
what it was not able to find, but the menuselect information is a 
reasonable next step.


--
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Digium, Inc. | Director of Software Technologies
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Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6

2012-04-20 Thread bilal ghayyad 
Dear;

Well, I did make menuselect and I really found the XXX and did not get the 
ability to select the channel. So what could be the reason?

>From the other side, I find the following when I type for lspci

02:01.0 Ethernet controller: Digium, Inc. Wildcard TDM410 4-port analog card 
(rev 11) 

By the way, when I am installing asterisk 1.8, I do not find this problem at 
all. Only with asterisk 1.4

Any advise?
Regards
Bilal


> 
> > Yes, first thing I do is the make all and make install
> for dahdi,
> > then I do ./configure and make and make install for
> asterisk. But I
> > do not find the chan_dahdi under the
> /usr/lib/asterisk/modules. WHY?
> 
> You probably need to run make menuselect after ./configure
> and before
> make to select dahdi for building & installation.
> 



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Re: [asterisk-users] Advice on Asterisk Conference

2012-04-20 Thread Leandro Dardini 
1. No, asterisk can act as pbx and as conference server

2. No, just bought a powerful server

3. Not me, sorry

4. You are limited only by the CPU of your server
Il giorno 20/apr/2012 19:21, "Mitchell Johnson" 
ha scritto:

> We're looking into using Asterisk to do our conferencing.  Currently we do
> all our conferencing using Cisco, we have a router with PVDM modules so we
> can offload the hardware resources.
>
> I'm looking for some best practices on how to set it up.
>
> 1.  DO I need a separate server for the conference server?
> 2.  Do I need to offload the actual conference to a router with PVDM
> modules.
> 3.  Does anyone have experience with transitioning from Cisco conferencing
> to Asterisk?
> 4.  How many participants can participate in a conference?
>
> Thanks,
>
> Mitch
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[asterisk-users] DAHDI-Linux 2.6.1, 2.5.1 and DAHDI-Tools 2.6.1, 2.5.1 Now Available

2012-04-20 Thread Asterisk Development Team 
The Asterisk Development Team has announced the releases of:
  DAHDI-Linux 2.6.1
  DAHDI-Linux 2.5.1
  DAHDI-Tools 2.6.1
  DAHDI-Tools 2.5.1
  DAHDI-Linux-Complete 2.6.1+2.6.1
  DAHDI-Linux-Complete 2.5.1+2.5.1

These releases are available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

2.6.1 and 2.5.1 are bugfix releases of which the most noteable changes are:

- Fix for Digium dual and quadspan cards in E1 mode when used with a hardware
  echocanceler that was introduced in 2.6.0.
- Fix for intermittent failure to decode FSK caller ID on Digium voicebus
  analog cards introduced in 2.6.0.
- Support for Linux kernel versions up to 3.4.

Issues closed in these releases:
DAHLIN-275: E1 spans have noise on some alternative channels when VPM is 
active
DAHLIN-274: dahdi_dummy failes to compile
DAHLIN-283: Disable Active State Power Management on PCIe links for DAHDI 
devices.
DAHLIN-280: dahdi_dynamic_eth(ethmf,loc)
DAHLIN-286: DAHDI driver wctdm24xxp does not compile with GCC 3.4.4
DAHLIN-279: dahdi will not compile with CONFIG_DAHDI_ECHOCAN_PROCESS_TX
DAHLIN-278: dahdi will not compile with CONFIG_DAHDI_NET
DAHLIN-185: Dahdi dummy includes time.h, should be timer.h for low-res 
timer
DAHLIN-288: compilation error when CONFIG_DAHDI_WATCHDOG is defined
  And in the 2.5.1 release only:
DAHLIN-272: No PCM on a TDM410 FXS module since r10167

The DAHDI-Linux shortlog of changes since 2.6.0:

  Mike Sinkovsky (1):
dahdi: Fix compilation when CONFIG_DAHDI_WATCHDOG is defined.
  
  Oron Peled (9):
xpp: bugfix: fix bad refcount
xpp: Don't deactivate XPDs on unregistration
xpp: handle failures during dahdi_register_device()
xpp: reset Astribank SPI busses
xpp: FXS: better power-down to lower noise
A parent-less device should not crash dahdi
remove a duplicate dev_set_name()
xpp: FXS: atomic vbat_h power handling
xpp: FXS: added a 'lower_ringing_noise' parameter
  
  Shaun Ruffell (30):
wctdm24xxp: FXS on-hook transmission timer incorrect.
wct4xxp: VPM module creates noise on alternate channels on E1 spans.
wctdm24xxp: Shorten RINGOFF debounce interval from 512ms to 128ms.
xpp: Use 'bool' type for boolean module parameters on kernel versions 
>= 2.6.31.
xpp: '%d' -> '%lu' when displaying module_refcount on kernel versions 
>= 3.3
dahdi_dummy: Fix compilation since dahdi-linux 2.6.0.
dahdi: Add dahdi_pci_disable_link_state for kernel < 2.6.25.
wct4xxp: __t4_frame_in and __t4_framer_out slowdowns.
wct4xxp: Add compile-time option to disable ASPM for PCIe devices.
wcte12xp, wctdm24xxp: Add compile-time option to disable ASPM for PCIe 
devices.
dahdi: Update dev_set_name / dev_name for RHEL 5.6+.
dahdi_dynamic_eth: Move tx packet flushing to process context.
dahdi_dynamic: Since dynamic devices are 'parentless' we must name them.
dahdi_dynamic_eth: Prevent crash is packet arrives before span is fully 
configured.
dahdi_dynamic_eth: Fix compilation on kernels < 2.6.22.
wct4xxp: Disable all interrupts explicitly in interrupt handler.
wct4xxp: Trivial formatting changes around request_irq.
wctdm24xxp: Remove forward declaration of inline for GCC 3.4.4
wctdm24xxp, wcte12xp: Allow VPMOCT032 firmware to be compiled into 
driver.
dahdi_dynamic: Do not call into dahdi_dynamic without holding reference.
dahdi_dynamic: Remove calls to __module_get().
dahdi_dynamic: Close race on unload if red alarm timer was running when 
unloaded.
dahdi_dynamic_eth: Make ztdeth_exit() symetrical with ztdeth_init() and 
fix race on unload.
dahdi_dynamic_loc: Change and check the dyn->pvt pointer under lock.
dahdi: Fix compilation when CONFIG_DAHDI_ECHOCAN_PROCESS_TX is defined.
dahdi: Fix compilation when CONFIG_DAHDI_NET is defined.
dahdi_dummy: Include timer.h instead of time.h
wcb4xxp: Remove asm/system.h include.
wcte12xp, wctdm24xxp, wct4xxp: Print warning about potential GPL 
violation w/HOTPLUG_FIRMWARE=no.
xpp: Fix compilation when CONFIG_DAHDI_WATCHDOG is defined.
  
  Tzafrir Cohen (8):
Build OSLEC EC if in the tree
Astribank I firmwares rev. 7107
USB_RECOV.hex: recovering from xpp hardware issues
xpp: USB_FW rev 10401: minor 6FXS/2FXO caps issue
xpp: firmwares to support E-Main 4
xpp: firmwares: useless 0x1A at EOF
FPGA_1161.201.hex rev 10532: fix reset of XR1000
FPGA_1161.201.hex rev 10545: fix reset of XR1000

The DAHDI-Linux diffstat from the 2.6.0 release:

  README|   11 +-
  drivers/dahdi/Kbuild

[asterisk-users] Advice on Asterisk Conference

2012-04-20 Thread Mitchell Johnson 
We're looking into using Asterisk to do our conferencing.  Currently we do all 
our conferencing using Cisco, we have a router with PVDM modules so we can 
offload the hardware resources.

I'm looking for some best practices on how to set it up.

1.  DO I need a separate server for the conference server?
2.  Do I need to offload the actual conference to a router with PVDM modules.
3.  Does anyone have experience with transitioning from Cisco conferencing to 
Asterisk?
4.  How many participants can participate in a conference?  

Thanks,

Mitch
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Re: [asterisk-users] asterisk 1.4.39 and dahdi 2.6 on Ubuntu

2012-04-20 Thread Kevin P. Fleming 

On 04/19/2012 05:59 PM, bilal ghayyad wrote:

Dears;

I see this at the /var/log/asterisk/messages:

[Apr 20 01:49:48] ERROR[1657] codec_dahdi.c: Failed to open 
/dev/dahdi/transcode: No such file or directory


If you aren't using a DAHDI transcoding card, then you don't need to 
load the codec_dahdi module in Asterisk. Since it was built, though, you 
clearly have DAHDI built and installed properly, and the Asterisk build 
process was aware of that.




Again, I am installing asterisk and dahdi at Ubuntu (uname -a
Linux House 3.0.0-17-server #30-Ubuntu SMP Thu Mar 8 22:15:30 UTC 2012 x86_64 
x86_64 x86_64 GNU/Linux).

I do not know if you were talking about the messages logs or about someting 
else?

Anyway, these are the logs that I see at the messages after running 
/etc/init.d/asterisk restart:


[Apr 20 01:49:48] NOTICE[1657] cdr.c: CDR simple logging enabled.
[Apr 20 01:49:48] NOTICE[1657] loader.c: 142 modules will be loaded.
[Apr 20 01:49:48] WARNING[1657] res_smdi.c: No SMDI interfaces are available to 
listen on, not starting SMDI listener.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: Starting AEL load process.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: calculated config 
file name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: parsed config file 
name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: checked config file 
name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: compiled config 
file name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: merged config file 
name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] NOTICE[1657] pbx_ael.c: AEL load process: verified config 
file name '/etc/asterisk/extensions.ael'.
[Apr 20 01:49:48] ERROR[1657] codec_dahdi.c: Failed to open 
/dev/dahdi/transcode: No such file or directory


All of that is perfectly normal. If you want that ERROR message to go 
away, add 'noload => codec_dahdi' to your /etc/asterisk/modules.conf file.


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Re: [asterisk-users] E & M signalling and Dahdi

2012-04-20 Thread Steve Underwood 

On 04/20/2012 11:30 PM, Eduardo Pimenta wrote:


Hello all,


Does anyone know if E&M over E1 signalling works on top of R2, ISDN 
and where can I find a sample Dahdi configuration? Have done a lot of 
google and cannot find a proper E1 configuration.


No it doesn't. E&M signalling is the same layer as R2 and ISDN. It is an 
alternative to them, not another layer.


Steve


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[asterisk-users] E & M signalling and Dahdi

2012-04-20 Thread Eduardo Pimenta 
Hello all,


Does anyone know if E&M over E1 signalling works on top of R2, ISDN and
where can I find a sample Dahdi configuration? Have done a lot of google
and cannot find a proper E1 configuration.

Thanks,

Eduardo
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[asterisk-users] dahdi cannot make simaltaneous calls

2012-04-20 Thread Mc GRATH Ricardo 
: As indicated in following 
octets<   Ext: 1  Coding: 0  Number Specified  Channel 
Type: 3<   Ext: 1  Channel: 1 Type: CPE]Received message 
for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0, call->pri is 0x8ca59a0 TEI/SAPI 
0/0-- Processing IE 24 (cs0, Channel Identification)q931.c:7104 
post_handle_q931_message: Call 32771 enters state 3 (Outgoing Call Proceeding). 
 Hold state: Idle
< Protocol Discriminator: Q.931 (8)  len=9< TEI=0 Call Ref: len= 2 (reference 
4/0x4) (Sent to originator)< Message Type: RELEASE COMPLETE (90)< [08 02 82 
a2]< Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  
Location: Public network serving the local user (2)<  Ext: 1  
Cause: Circuit/channel congestion (34), class = Network Congestion (resource 
unavailable) (2) ]Received message for call 0x8cc6f80 on 0x8ca59a0 TEI/SAPI 
0/0, call->pri is 0x8ca59a0 TEI/SAPI 0/0-- Processing IE 8 (cs0, 
Cause)q931.c:7197 post_handle_q931_message: Call 32772 enters state 0 (Null).  
Hold state: Idleq931_hangup: other hangupNEW_HANGUP DEBUG: Calling q931_hangup, 
ourstate Null, peerstate Null, hold-state IdleNEW_HANGUP DEBUG: Destroying the 
call, ourstate Null, peerstate Null, hold-state Idle
< Protocol Discriminator: Q.931 (8)  len=9< TEI=0 Call Ref: len= 2 (reference 
3/0x3) (Sent to originator)< Message Type: ALERTING (1)< [1e 02 82 88]< 
Progress Indicator (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  0: 0  
Location: Public network serving the local user (2)<
   Ext: 1  Progress Description: Inband information or appropriate pattern now 
available. (8) ]Received message for call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0, 
call->pri is 0x8ca59a0 TEI/SAPI 0/0-- Processing IE 30 (cs0, Progress 
Indicator)q931.c:6983 post_handle_q931_message: Call 32771 enters state 4 (Call 
Delivered).  Hold state: Idleq931_hangup: other hangupNEW_HANGUP DEBUG: Calling 
q931_hangup, ourstate Call Delivered, peerstate Call Received, hold-state 
Idleq931.c:4845 q931_disconnect: Call 32771 enters state 11 (Disconnect 
Request).  Hold state: Idle
> DL-DATA request> Protocol Discriminator: Q.931 (8)  len=9> TEI=0 Call Ref: 
> len= 2 (reference 3/0x3) (Sent from originator)> Message Type: DISCONNECT 
> (69)TEI=0 Transmitting N(S)=7, window is open V(A)=7 K=7
> Protocol Discriminator: Q.931 (8)  len=9> TEI=0 Call Ref: len= 2 (reference 
> 3/0x3) (Sent from originator)> Message Type: DISCONNECT (69)> [08 02 81 90]> 
> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  
> Location: Private network serving the local user (1)>  Ext: 1 
>  Cause: Normal Clearing (16), class = Normal Event (1) ]
< Protocol Discriminator: Q.931 (8)  len=5< TEI=0 Call Ref: len= 2 (reference 
3/0x3) (Sent to originator)< Message Type: RELEASE (77)Received message for 
call 0x8cc4a70 on 0x8ca59a0 TEI/SAPI 0/0, call->pri is 0x8ca59a0 TEI/SAPI 
0/0q931.c:7237 post_handle_q931_message: Call 32771 enters state 0 (Null).  
Hold state: Idleq931_hangup: other hangupNEW_HANGUP DEBUG: Calling q931_hangup, 
ourstate Null, peerstate Release Request, hold-state Idle
> DL-DATA request> Protocol Discriminator: Q.931 (8)  len=9> TEI=0 Call Ref: 
> len= 2 (reference 3/0x3) (Sent from originator)> Message Type: RELEASE 
> COMPLETE (90)TEI=0 Transmitting N(S)=8, window is open V(A)=8 K=7
> Protocol Discriminator: Q.931 (8)  len=9> TEI=0 Call Ref: len= 2 (reference 
> 3/0x3) (Sent from originator)> Message Type: RELEASE COMPLETE (90)> [08 02 81 
> 90]> Cause (len= 4) [ Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  
> Location: Private network serving the local user (1)>  Ext: 1 
>  Cause: Normal Clearing (16), class = Normal Event (1) ]q931_hangup: other 
> hangupNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null, 
> hold-state IdleNEW_HANGUP DEBUG: Destroying the call, ourstate Null, 
> peerstate Null, hold-state Idle

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Message: 4
Date: Thu, 19 Apr 2012 17:11:57 -0300
From: Josu? Conti 
Subject: [asterisk-users] Company info
To: Asterisk Users Mailing List - Non-Commercial Discussion

Message-ID:

Content-Type: text/plain; charset="iso-8859-1"

Dear all,
Please let me know if anybody have informations about a company called
Convergia, like your products, ASR/ACD or more details.

With Best Regards

Josue
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-

Re: [asterisk-users] Experience with virtual servers?

2012-04-20 Thread Bruce Komito 
We also run asterisk in a virtual environment, VMWare specifically, along side 
of web, database, email and DNS (virtual) servers.  As far as I'm concerned, it 
runs as well as it ever did in a real environment.   We are using HP Proliant 
DL360 G5's (3gz Xeon 5160 dual core processors).  In our case, the VM hosts 
that run asterisk are only running Linux guests, and so we require relatively 
little memory...only 4gb.  We also have a larger VM host, similarly configured, 
but running windows guests and that one has 18gb.

Before settling on VMWare, we tried some of the open source solutions, and 
those did not work as well for us, but VMWare is TOPS.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant Zimmerman
Sent: Friday, April 20, 2012 7:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Experience with virtual servers?

We run many of our asterisk servers on Hyper-V Clusters with openSuse 12.1. 
They work great All of our PRI &PSTN conversions are done with gateway 
appliances and the bulk of our traffic comes in SIP trunk from providers. We 
have 16 switches on virtual and 10 on dedicated. We are add all new asterisk 
switches as virtual, and removing old physical installs as space is needed in 
the racks to accommodate new servers to support the virtual deployments.
Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003


From: "Brynjolfur Thorvardsson" mailto:bi...@itanet.nu>>
Sent: Friday, April 20, 2012 8:54 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
mailto:asterisk-users@lists.digium.com>>
Subject: [asterisk-users] Experience with virtual servers?
Hi All

Does anybody have experience with running Asterisk on virtual servers? I have 
been experimenting with two suppliers and I am not altogether happy with sound 
quality etc.

Is it perhaps foolish to try and install a "production" Asterisk server on a 
virtual machine? With dedicated servers being comparatively cheap (although 
still several times more expensive than virtual servers), perhaps that is the 
way  I should be going? I have heard someone mention "Asterisk friendly" VPS 
providers, how can you tell if they are or aren't friendly?

We currently have our Asterisk server running on a five year old single AMD CPU 
32 bit machine with 512Mb and that works fine. Even the cheapest virtual server 
vendors offer servers that seem much more powerful but after testing I am not 
so sure any more!

Any info would be very welcome!

Regards

Binni


No virus found in this message.
Checked by AVG - www.avg.com
Version: 2012.0.1913 / Virus Database: 2411/4947 - Release Date: 04/19/12
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Re: [asterisk-users] Experience with virtual servers?

2012-04-20 Thread Bryant Zimmerman 
We run many of our asterisk servers on Hyper-V Clusters with openSuse 12.1. 
They work great All of our PRI &PSTN conversions are done with gateway 
appliances and the bulk of our traffic comes in SIP trunk from providers. We 
have 16 switches on virtual and 10 on dedicated. We are add all new asterisk 
switches as virtual, and removing old physical installs as space is needed in 
the racks to accommodate new servers to support the virtual deployments.

Thanks

Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003


 From: "Brynjolfur Thorvardsson" 
Sent: Friday, April 20, 2012 8:54 AM
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Subject: [asterisk-users] Experience with virtual servers?

  Hi All   Does anybody have experience with running Asterisk on virtual 
servers? I have been experimenting with two suppliers and I am not altogether 
happy with sound quality etc.Is it perhaps foolish to try and install a 
"production" Asterisk server on a virtual machine? With dedicated servers being 
comparatively cheap (although still several times more expensive than virtual 
servers), perhaps that is the way  I should be going? I have heard someone 
mention "Asterisk friendly" VPS providers, how can you tell if they are or 
aren't friendly? We currently have our Asterisk server running on a five 
year old single AMD CPU 32 bit machine with 512Mb and that works fine. Even the 
cheapest virtual server vendors offer servers that seem much more powerful but 
after testing I am not so sure any more!   Any info would be very welcome!   
Regards   Binni

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Re: [asterisk-users] Experience with virtual servers?

2012-04-20 Thread Stuart Elvish - IP Exchange Systems 
Hi Binni,

It often depends on how over-subscribed / over-sold the server is as
well as CPU scheduling. People often suggest KVM VPS' over OpenVZ etc.

There are a few companies that have VPS products specifically designed
for voice hosting (presumably a lower ratio of VM's per server and
upstream bandwidth better suited for voice) and this may be worth
investigating. Some even provide templates for provisioning your
container / VM with TrixBox or Elastix.

If you are looking for a particular geographical location or a have a
specific solution (conferencing / trunk lines etc) your options won't be
as many.

Kind Regards
Stuart

On 04/20/2012 07:51 PM, Brynjolfur Thorvardsson wrote:
> Hi All
> 
>  
> 
> Does anybody have experience with running Asterisk on virtual servers? I
> have been experimenting with two suppliers and I am not altogether happy
> with sound quality etc.
> 
>  
> 
> Is it perhaps foolish to try and install a “production” Asterisk server
> on a virtual machine? With dedicated servers being comparatively cheap
> (although still several times more expensive than virtual servers),
> perhaps that is the way  I should be going? I have heard someone mention
> “Asterisk friendly” VPS providers, how can you tell if they are or
> aren’t friendly?  
> 
>  
> 
> We currently have our Asterisk server running on a five year old single
> AMD CPU 32 bit machine with 512Mb and that works fine. Even the cheapest
> virtual server vendors offer servers that seem much more powerful but
> after testing I am not so sure any more!
> 
>  
> 
> Any info would be very welcome!
> 
>  
> 
> Regards
> 
>  
> 
> Binni
> 
> 
> 
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> _
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>http://www.asterisk.org/hello
> 
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Re: [asterisk-users] Experience with virtual servers?

2012-04-20 Thread Danny Nicholas 
As long as you are using SIP trunking, Asterisk will perform nicely.  If you 
want PRI or DAHDI trunks, that's a different bridge to cross.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arthur Stanfield
Sent: Friday, April 20, 2012 8:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Experience with virtual servers?

Hi Binni,

We run a number of Asterisk servers on virtual machines. I'm not heavily 
involved in the virtualisation side of the business so i'm afraid i can't give 
you much advice on it, Past saying it is possible to have an Asterisk System up 
and running reliably on virtual machines.

Our virtualisation platform is KVM based.

Hopefully someone with more knowledge than me will be able to help!.

Cheers,
AJ.

- Original Message -
From: "Brynjolfur Thorvardsson" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, 20 April, 2012 1:51:03 PM
Subject: [asterisk-users] Experience with virtual servers?





Hi All 



Does anybody have experience with running Asterisk on virtual servers? I have 
been experimenting with two suppliers and I am not altogether happy with sound 
quality etc. 



Is it perhaps foolish to try and install a “production” Asterisk server on a 
virtual machine? With dedicated servers being comparatively cheap (although 
still several times more expensive than virtual servers), perhaps that is the 
way I should be going? I have heard someone mention “Asterisk friendly” VPS 
providers, how can you tell if they are or aren’t friendly? 



We currently have our Asterisk server running on a five year old single AMD CPU 
32 bit machine with 512Mb and that works fine. Even the cheapest virtual server 
vendors offer servers that seem much more powerful but after testing I am not 
so sure any more! 



Any info would be very welcome! 



Regards 



Binni
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Re: [asterisk-users] Experience with virtual servers?

2012-04-20 Thread Arthur Stanfield 
Hi Binni,

We run a number of Asterisk servers on virtual machines. I'm not heavily 
involved in the virtualisation side of the business so i'm afraid i can't give 
you much advice on it, Past saying it is possible to have an Asterisk System up 
and running reliably on virtual machines.

Our virtualisation platform is KVM based.

Hopefully someone with more knowledge than me will be able to help!.

Cheers,
AJ.

- Original Message -
From: "Brynjolfur Thorvardsson" 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 

Sent: Friday, 20 April, 2012 1:51:03 PM
Subject: [asterisk-users] Experience with virtual servers?





Hi All 



Does anybody have experience with running Asterisk on virtual servers? I have 
been experimenting with two suppliers and I am not altogether happy with sound 
quality etc. 



Is it perhaps foolish to try and install a “production” Asterisk server on a 
virtual machine? With dedicated servers being comparatively cheap (although 
still several times more expensive than virtual servers), perhaps that is the 
way I should be going? I have heard someone mention “Asterisk friendly” VPS 
providers, how can you tell if they are or aren’t friendly? 



We currently have our Asterisk server running on a five year old single AMD CPU 
32 bit machine with 512Mb and that works fine. Even the cheapest virtual server 
vendors offer servers that seem much more powerful but after testing I am not 
so sure any more! 



Any info would be very welcome! 



Regards 



Binni 
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[asterisk-users] Experience with virtual servers?

2012-04-20 Thread Brynjolfur Thorvardsson 
Hi All

Does anybody have experience with running Asterisk on virtual servers? I have 
been experimenting with two suppliers and I am not altogether happy with sound 
quality etc.

Is it perhaps foolish to try and install a "production" Asterisk server on a 
virtual machine? With dedicated servers being comparatively cheap (although 
still several times more expensive than virtual servers), perhaps that is the 
way  I should be going? I have heard someone mention "Asterisk friendly" VPS 
providers, how can you tell if they are or aren't friendly?

We currently have our Asterisk server running on a five year old single AMD CPU 
32 bit machine with 512Mb and that works fine. Even the cheapest virtual server 
vendors offer servers that seem much more powerful but after testing I am not 
so sure any more!

Any info would be very welcome!

Regards

Binni
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Re: [asterisk-users] Company info

2012-04-20 Thread Josué Conti 
Dear Steven, no is not.
I´m looking for sip connections and this company is a opportunity, but I
would like to know if anybody have informations or use your products, just
it.
Like, this company is confident?
My apologies if seemed this.

With Best Regards

Josue

Em 20 de abril de 2012 05:40, Steven Howes escreveu:

> Can't tell if this is a transparent attempt at advertising, or...?
>
> S
>
> On 19 Apr 2012, at 22:09, Josué Conti wrote:
>
> This is your website:
>
> http://www.convergia.com/
>
> Thanks in advanced for any informations.
>
> Best Regards
>
> Josue
>
> Em 19 de abril de 2012 17:11, Josué Conti  escreveu:
>
>> Dear all,
>> Please let me know if anybody have informations about a company called
>> Convergia, like your products, ASR/ACD or more details.
>>
>> With Best Regards
>>
>> Josue
>>
>
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>
>
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Re: [asterisk-users] Company info

2012-04-20 Thread Doug Lytle 

Steven Howes wrote:

Can't tell if this is a transparent attempt at advertising, or...?



If he doesn't asking any further questions, we'll know.

Doug


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


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Re: [asterisk-users] Company info

2012-04-20 Thread Steven Howes 
Can't tell if this is a transparent attempt at advertising, or...?

S

On 19 Apr 2012, at 22:09, Josué Conti wrote:

> This is your website:
> 
> http://www.convergia.com/
> 
> Thanks in advanced for any informations.
> 
> Best Regards
> 
> Josue
> 
> Em 19 de abril de 2012 17:11, Josué Conti  escreveu:
> Dear all, 
> Please let me know if anybody have informations about a company called 
> Convergia, like your products, ASR/ACD or more details.
> 
> With Best Regards
> 
> Josue
> 
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>   http://www.asterisk.org/hello
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