Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues

2011-10-07 Thread James Sharp 

On 10/07/2011 04:42 PM, Kevin P. Fleming wrote:


You shouldn't be *receiving* CNG, as you are the calling endpoint.


You're right.  Hadn't even thought about that.



If you are seeing UDPTL packets containing T.38 CED, V.21 preamble, DIS,
etc. then something is badly wrong.

... and, that thing is probably the sequence number. Once Asterisk sees
a packet with sequence number 0, any subsequent packets received with
the same sequence number will be dropped (because according to the T.38
recommendation, they must be retransmissions... new packets would have
higher sequence numbers). So these UDPTL packets are never making their
way up to the FAX stack, and the FAX transaction never gets started.

I guess it must be common for UDPTL stacks out there to just not care
about repeated sequence numbers, although the one in Asterisk sure does
(and it's based on the same code as the one in CallWeaver, FreeSwitch
and maybe other packages too).

If you'd like to experiment, you can comment out lines 495 and 511 of
main/udptl.c, which will make Asterisk's UDPTL stack just not care at
all about the incoming sequence number.


HEAD out of SVN + the sequence number change still gets no fax transmit. 
 I do get a few addition fax status messages on the console, though. 
I'm getting


-- FAX handle 0: [ 000.062327 ], STAT_EVT_RX_IMG_STRT   st: WT_DIS 
 rt: UNEXPECT


4-9 times (with changing timestamps, of course), then nothing until 
disconnect.


It sounds like that Gafachi's T38 implementation is horribly, horribly 
broken.  I'm not tied to them at all, so if their stuff is broken, I'll 
go somewhere else.



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Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues

2011-10-07 Thread Kevin P. Fleming 

On 10/07/2011 03:29 PM, James Sharp wrote:

On 10/07/2011 04:04 PM, Kevin P. Fleming wrote:


First, we can see that Gafachi's T.38 implementation still has some
breakage in it (although the problems are ones that Asterisk has been
taught to deal with). In its 200 OK to the T.38 re-INVITE, it has
"a=T38FaxRateManagement:transferredTCFlocalTCF"; this is not valid (the
valid values for this are 'transferredTCF' and 'localTCF'). In addition,
even though Asterisk sent "a=T38FaxUdpEC:t38UDPRedundancy", Gafachi
replied with "a=T38FaxUdpEC:t38UDPFEC". For T.38 version 0 (which is in
use here), the only valid response was either what Asterisk sent, or no
T38FaxUdpEC value at all.

However, it is unlikely those are causing the call failure here. It's
hard to say for sure without seeing the contents of the UDPTL packets,
but based on their sizes, they are very likely "t38-nosignal" packets,
and if that's all the FAX stack in Asterisk ever received, it would not
trigger a FAX transaction to begin. Another possible problem is the
repeated 'seq 0' in all the UDPTL packets, but this could be a problem
with the UDPTL stack debugging messages themselves (this was just fixed
in the Subversion branches for Asterisk 1.8 and later a couple of days
ago).


Theres a few t30-nosignal packets at the beginning, but then they
transition to other t30 packets, including CNG, CED, preambles, training
and data. Wireshark says the sequence number is always 0, so it appears
that Asterisk is not mis-displaying


You shouldn't be *receiving* CNG, as you are the calling endpoint.

If you are seeing UDPTL packets containing T.38 CED, V.21 preamble, DIS, 
etc. then something is badly wrong.


... and, that thing is probably the sequence number. Once Asterisk sees 
a packet with sequence number 0, any subsequent packets received with 
the same sequence number will be dropped (because according to the T.38 
recommendation, they must be retransmissions... new packets would have 
higher sequence numbers). So these UDPTL packets are never making their 
way up to the FAX stack, and the FAX transaction never gets started.


I guess it must be common for UDPTL stacks out there to just not care 
about repeated sequence numbers, although the one in Asterisk sure does 
(and it's based on the same code as the one in CallWeaver, FreeSwitch 
and maybe other packages too).


If you'd like to experiment, you can comment out lines 495 and 511 of 
main/udptl.c, which will make Asterisk's UDPTL stack just not care at 
all about the incoming sequence number.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues

2011-10-07 Thread James Sharp 

On 10/07/2011 04:04 PM, Kevin P. Fleming wrote:


First, we can see that Gafachi's T.38 implementation still has some
breakage in it (although the problems are ones that Asterisk has been
taught to deal with). In its 200 OK to the T.38 re-INVITE, it has
"a=T38FaxRateManagement:transferredTCFlocalTCF"; this is not valid (the
valid values for this are 'transferredTCF' and 'localTCF'). In addition,
even though Asterisk sent "a=T38FaxUdpEC:t38UDPRedundancy", Gafachi
replied with "a=T38FaxUdpEC:t38UDPFEC". For T.38 version 0 (which is in
use here), the only valid response was either what Asterisk sent, or no
T38FaxUdpEC value at all.

However, it is unlikely those are causing the call failure here. It's
hard to say for sure without seeing the contents of the UDPTL packets,
but based on their sizes, they are very likely "t38-nosignal" packets,
and if that's all the FAX stack in Asterisk ever received, it would not
trigger a FAX transaction to begin. Another possible problem is the
repeated 'seq 0' in all the UDPTL packets, but this could be a problem
with the UDPTL stack debugging messages themselves (this was just fixed
in the Subversion branches for Asterisk 1.8 and later a couple of days
ago).


Theres a few t30-nosignal packets at the beginning, but then they 
transition to other t30 packets, including CNG, CED, preambles, training 
and data.  Wireshark says the sequence number is always 0, so it appears 
that Asterisk is not mis-displaying


http://pastebin.ca/2087784

I can provide the raw tcpdump file if needed.



If you would, please retry this with the HEAD of the Asterisk 10 branch
instead of 10.0.0-beta1, and also capture the UDPTL packets themselves
so we can see what they contained.




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Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues

2011-10-07 Thread Kevin P. Fleming 

On 10/07/2011 03:06 PM, Nasir Iqbal wrote:

for which user/number sip reinvite is for? ooh! you are running a direct
application without any dialplan or user, may be that is the cause!  I
think you should first write fax dialplan, reload asterisk and test
again with originate but this time with extension not application.


No, none of that is relevant. It's perfectly acceptable to call 
SendFAX() on a CLI/AMI/spool-originated channel.


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Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues

2011-10-07 Thread Nasir Iqbal 
for which user/number sip reinvite is for? ooh! you are running a direct
application without any dialplan or user, may be that is the cause!  I think
you should first write fax dialplan, reload asterisk and test again with
originate but this time with extension not application.

Nasir Iqbal

ICT Innovations
http://www.ictinnovations.com/



On Sat, Oct 8, 2011 at 12:20 AM, James Sharp  wrote:

> On 10/07/2011 12:27 AM, Nasir Iqbal wrote:
>
>> Check firewall and NAT settings!
>>
>> Monitoring sip and media flow from asterisk cli can help, use "sip set
>> debug on", "rtp set debug on" and "udptl set debug on"
>>
>>
> No NAT involved and I shut off IPTables.  Still no luck.  Debug shows the
> SIP invite, RTP frames going in & out, the SIP reinvite, and then UDPTL
> frames coming in until timeout.
>
> See the entire transaction at http://pastebin.ca/2087758
>
>
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Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues

2011-10-07 Thread Kevin P. Fleming 

On 10/07/2011 02:20 PM, James Sharp wrote:

On 10/07/2011 12:27 AM, Nasir Iqbal wrote:

Check firewall and NAT settings!

Monitoring sip and media flow from asterisk cli can help, use "sip set
debug on", "rtp set debug on" and "udptl set debug on"



No NAT involved and I shut off IPTables. Still no luck. Debug shows the
SIP invite, RTP frames going in & out, the SIP reinvite, and then UDPTL
frames coming in until timeout.

See the entire transaction at http://pastebin.ca/2087758


Thanks for that; it helps.

First, we can see that Gafachi's T.38 implementation still has some 
breakage in it (although the problems are ones that Asterisk has been 
taught to deal with). In its 200 OK to the T.38 re-INVITE, it has 
"a=T38FaxRateManagement:transferredTCFlocalTCF"; this is not valid (the 
valid values for this are 'transferredTCF' and 'localTCF'). In addition, 
even though Asterisk sent "a=T38FaxUdpEC:t38UDPRedundancy", Gafachi 
replied with "a=T38FaxUdpEC:t38UDPFEC". For T.38 version 0 (which is in 
use here), the only valid response was either what Asterisk sent, or no 
T38FaxUdpEC value at all.


However, it is unlikely those are causing the call failure here. It's 
hard to say for sure without seeing the contents of the UDPTL packets, 
but based on their sizes, they are very likely "t38-nosignal" packets, 
and if that's all the FAX stack in Asterisk ever received, it would not 
trigger a FAX transaction to begin. Another possible problem is the 
repeated 'seq 0' in all the UDPTL packets, but this could be a problem 
with the UDPTL stack debugging messages themselves (this was just fixed 
in the Subversion branches for Asterisk 1.8 and later a couple of days ago).


If you would, please retry this with the HEAD of the Asterisk 10 branch 
instead of 10.0.0-beta1, and also capture the UDPTL packets themselves 
so we can see what they contained.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org

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Re: [asterisk-users] Digium FFA + Gafachi T38 outgoing issues

2011-10-07 Thread James Sharp 

On 10/07/2011 12:27 AM, Nasir Iqbal wrote:

Check firewall and NAT settings!

Monitoring sip and media flow from asterisk cli can help, use "sip set
debug on", "rtp set debug on" and "udptl set debug on"



No NAT involved and I shut off IPTables.  Still no luck.  Debug shows 
the SIP invite, RTP frames going in & out, the SIP reinvite, and then 
UDPTL frames coming in until timeout.


See the entire transaction at http://pastebin.ca/2087758

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Re: [asterisk-users] Cisco AS5400XM

2011-10-07 Thread Richard Zheng 
On Thu, Oct 6, 2011 at 11:25 AM, Kyle Sexton  wrote:

> I'm looking at the Cisco AS5400XM to convert some incoming T1s to SIP
> signaling.  Has anyone had any experience with these devices?  The
> feature cards that Cisco sells can be a little confusing.  I'm
> thinking something like below is what I need.
>
> (1) AS5400XM, AS5400XM Starter Kit (inc Chassis, MB, Def Mem)
> (1) AS54-AC-RPS-PWR, AS5400 AC Redundant Power Supply
> (1) AS54-DFC-8CT1, AS5400 OCTAL T1/PRI DFC Card
> (2) AS54-DFC-108NP, AS5400 108 Voice/Universal Port Feature Card
>
> Any thoughts would be appreciated.  Thanks.
>

We also looked it one time. One of the model is no longer supported by
cisco. The replacement model with all DSP loaded was quite expensive, about
$20k per box even on ebay.
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Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX

2011-10-07 Thread Kevin P. Fleming 

On 10/07/2011 07:46 AM, Administrator TOOTAI wrote:

Hi,

I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken
from deb http://packages.asterisk.org/deb lucid main) including dahdi
from this same repository. No FFA involved.

On incoming calls (only SIP, no telephony card), fax detection is
working but reception failed with

-- Executing [fax@from-TOOTAiAudio:19]
ReceiveFAX("SIP/tootaiAUDIO-0564", "/tmp/1317991071.1614.tiff,c") in
new stack
[Oct 7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec:
ReceiveFAX does not support polling
== Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on
'SIP/tootaiAUDIO-0564'

What can be the problem?


You included the 'c' option to ReceiveFAX, telling it to act as the 
'caller', even though it isn't the caller. This argument is parsed by 
ReceiveFAX in spite of it not being supported because the older app_fax 
version did support it, and we didn't want dialplans that included it to 
silently ignore the 'c' option. The same is true for the 'a' option; 
you'll note that neither of them are included in the documentation for 
the ReceiveFAX and SendFAX applications, and shouldn't be used.


Why did you specify the 'c' option?

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Re: [asterisk-users] dahdi show status command not avilable in CLI

2011-10-07 Thread Sammy Govind 
Please paste the configurations in the #included files as well.

On Fri, Oct 7, 2011 at 7:07 PM, michael k  wrote:

> Hi,
>
>
> This is my /etc/asterisk/chan_dahdi.conf file.
>
>
> [root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf
> ; Copied from DAHDI Module of FreePBX
>
> [general]
>
> #include chan_dahdi_general.conf
>
> [channels]
>
> ; include dahdi groups defined by DAHDI module of FreePBX
> #include chan_dahdi_groups.conf
>
> ;added by mic 06-oct-20011
> #include /etc/asterisk/dahdi-channels.conf
>
> ; include dahdi extensions defined in FreePBX
> #include chan_dahdi_additional.conf
>
>
> Any issues in this ?
>
>  Michael.k
>
>
>
> On Fri, Oct 7, 2011 at 7:19 PM, Eric Wieling  wrote:
>
>> It is likely you have an error in your /etc/asterisk/chan_dahdi.conf
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
>> Sent: Friday, October 07, 2011 9:24 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: Re: [asterisk-users] dahdi show status command not avilable in
>> CLI
>>
>> Hi,
>>
>>I am getting this error message while executing the  "module load
>> chan_dahdi.so".
>>
>> astrisks*CLI> module load chan_dahdi.so
>>
>> Unable to load module chan_dahdi.so
>> Command 'module load chan_dahdi.so' failed.
>>  == Parsing '/etc/asterisk/chan_dahdi.conf':   == Found
>>  == Parsing '/etc/asterisk/chan_dahdi_general.conf':   == Found
>>  == Parsing '/etc/asterisk/chan_dahdi_groups.conf':   == Found
>>  == Parsing '/etc/asterisk/dahdi-channels.conf':   == Found
>>  == Parsing '/etc/asterisk/chan_dahdi_additional.conf':   == Found
>>
>>
>> Thanks,
>>
>> Michael.k
>>
>>
>>
>> On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling  wrote:
>>
>>
>>What happens when you do the module load chan_dahdi.so command?
>>
>>
>>-Original Message-
>>From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
>>
>>Sent: Thursday, October 06, 2011 12:10 PM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: Re: [asterisk-users] dahdi show status command not
>> avilable in CLI
>>
>>Hi,
>>
>>
>>astrisks*CLI> module unload chan_dahdi.so
>>
>>Unable to unload resource chan_dahdi.so
>>Command 'module unload chan_dahdi.so ' failed.
>>
>>Producing some other error messages !
>>
>>
>>On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling 
>> wrote:
>>
>>
>>   In the Asterisk CLI run the commands "module unload
>> chan_dahdi.so" and "module load chan_dahdi.so".
>>
>>
>>
>>
>>   -Original Message-
>>   From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
>>   Sent: Thursday, October 06, 2011 11:40 AM
>>   To: Asterisk Users Mailing List - Non-Commercial Discussion
>>   Subject: Re: [asterisk-users] dahdi show status command not
>> avilable in CLI
>>
>>   Hi,
>>
>>I was run the commands "dahdi_genconf" and dahdi_cfg
>> outside the CLI as the part of x100p card installation. Before issuing this
>> command the dahdi show status command was available. There may any issues ?
>>
>>
>>   Michael.k
>>
>>
>>
>>   On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo <
>> gincantal...@fgasoftware.com> wrote:
>>
>>
>>
>>  Hi Michael,
>>
>>  what if you reload the module chan_dahdi from within
>> the * CLI? It should give some hints.
>>
>>  Giorgio
>>
>>
>>
>>  On 10/06/2011 05:22 PM, michael k wrote:
>>
>>  Hi Giorgio,
>>
>>  Thanks for your reply. I will produce some
>> output for your reference.
>>
>>  # lsmod | grep dahdi
>>
>>  dahdi_echocan_mg2  39688  1
>>  dahdi_transcode42372  1 wctc4xxp
>>  dahdi_voicebus 79424  2
>> wctdm24xxp,wcte12xp
>>  dahdi 238384  14
>> dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp
>>  crc_ccitt  35265  2
>> wctdm24xxp,dahdi
>>
>>
>>  # service dahdi status
>>
>>  ### Span  1: WCFXO/0 "Wildcard X100P Board 1"
>> (MASTER)
>>1 FXOFXSKS   (SWEC: MG2)
>> (battery)
>>
>>
>>  # dahdi_cfg -vv
>>
>>  DAHDI Tools Version - 2.3.0
>>
>>  DAHDI Version: 2.3.0.1
>>  Echo Canceller(s): MG2
>>

[asterisk-users] DIDs in Singapore

2011-10-07 Thread Jeff LaCoursiere 
Can anyone suggest an ITSP with Singapore DIDs and local Singapore
termination?

Cheers,

j


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Re: [asterisk-users] Add SIP diversion header in originate from AMI?

2011-10-07 Thread Jim Dickenson 
You can dial a local channel which executes a dial plan that does what you want.

Channel: Local/dial_number@cfmc_cdi_private

This will use exten dial_number in the cfmc_cdi_private context.

If you add something like this to the originate packet

Variable: CfMC_Use_CID=5419712513

You can use ${CfMC_Use_CID} to get the value.

-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Oct 7, 2011, at 8:03 AM, Tobias Steen wrote:

> Hello!
> 
> I want to thank everyone who helped me out with tips for load balancing 
> asterisk machines in a cluster.
> 
> I have encountered a new problem that is related to SIP diversion headers in 
> the INVITE.
> 
> I make calls through the manager interface and now want to add a 
> SIP-Diversion header that changes the CallerID of a number that is not 
> available on the trunk, the CallerID to be visible externally is connected to 
> an external customer service hired by another company.
> 
> My question:
> How can I add this header in a originateaction call via AMI?
> 
> Does the originated calls go through any context where I can add this header 
> with dialplan functions like "AddSipHeader()" or is it possible to dothis 
> directly in the OriginateAction through AMI?
>  
>  
> Example from voip-info:
>  
> [macro-diversion-header]
> exten => s,1,SIPAddHeader(Diversion: 
> \;reason=user=busy\;screen=no\;privacy=off)
>  
>  
> Best regards
> Tobias
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Re: [asterisk-users] Add SIP diversion header in originate from AMI?

2011-10-07 Thread Alex Balashov 
Try run your outbound leg through a Local channel.

--
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brevity, errors, and general sloppiness.

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Fax: +1-404-961-1892
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On Oct 7, 2011, at 11:03 AM, "Tobias Steen"  wrote:

> Hello!
> 
> I want to thank everyone who helped me out with tips for load balancing 
> asterisk machines in a cluster.
> 
> I have encountered a new problem that is related to SIP diversion headers in 
> the INVITE.
> 
> I make calls through the manager interface and now want to add a 
> SIP-Diversion header that changes the CallerID of a number that is not 
> available on the trunk, the CallerID to be visible externally is connected to 
> an external customer service hired by another company.
> 
> My question:
> How can I add this header in a originateaction call via AMI?
> 
> 
> Does the originated calls go through any context where I can add this header 
> with dialplan functions like "AddSipHeader()" or is it possible to do this 
> directly in the OriginateAction through AMI?
> 
>  
> 
>  
> 
> Example from voip-info:
> 
>  
> 
> [macro-diversion-header]
> exten => s,1,SIPAddHeader(Diversion: 
> \;reason=user=busy\;screen=no\;privacy=off)
> 
>  
> 
>  
> 
> Best regards
> 
> Tobias
> 
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Add SIP diversion header in originate from AMI?

2011-10-07 Thread Tobias Steen 
Hello!

I want to thank everyone who helped me out with tips for load balancing
asterisk machines in a cluster.

I have encountered a new problem that is related to SIP diversion headers in
the INVITE.

I make calls through the manager interface and now want to add a
SIP-Diversion header that changes the CallerID of a number that is not
available on the trunk, the CallerID to be visible externally is connected
to an external customer service hired by another company.

My question:
How can I add this header in a originateaction call via AMI?



Does the originated calls go through any context where I can add this header
with dialplan functions like "AddSipHeader()" or is it possible to do this
directly in the OriginateAction through AMI?

 

 

Example from voip-info:

 

[macro-diversion-header]
exten => s,1,SIPAddHeader(Diversion:
\;reason=user=busy\;screen=no\;privacy=off)

 

 

Best regards

Tobias

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Re: [asterisk-users] Asterisk 1.8.7 and ReceiveFAX

2011-10-07 Thread Kristijan Vrban 
remove the "c" argument

Kristijan

2011/10/7 Administrator TOOTAI :
> Hi,
>
> I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from
> deb http://packages.asterisk.org/deb lucid main) including dahdi from this
> same repository. No FFA involved.
>
> On incoming calls (only SIP, no telephony card), fax detection is working
> but reception failed with
>
>  -- Executing [fax@from-TOOTAiAudio:19]
> ReceiveFAX("SIP/tootaiAUDIO-0564", "/tmp/1317991071.1614.tiff,c") in new
> stack
> [Oct  7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: ReceiveFAX
> does not support polling
>  == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on
> 'SIP/tootaiAUDIO-0564'
>
> What can be the problem?
>
> Thanks for any hint.
>
> --
> Daniel
>
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Re: [asterisk-users] dahdi show status command not avilable in CLI

2011-10-07 Thread michael k 
Hi,


This is my /etc/asterisk/chan_dahdi.conf file.


[root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf
; Copied from DAHDI Module of FreePBX

[general]

#include chan_dahdi_general.conf

[channels]

; include dahdi groups defined by DAHDI module of FreePBX
#include chan_dahdi_groups.conf

;added by mic 06-oct-20011
#include /etc/asterisk/dahdi-channels.conf

; include dahdi extensions defined in FreePBX
#include chan_dahdi_additional.conf


Any issues in this ?

 Michael.k


On Fri, Oct 7, 2011 at 7:19 PM, Eric Wieling  wrote:

> It is likely you have an error in your /etc/asterisk/chan_dahdi.conf
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
> Sent: Friday, October 07, 2011 9:24 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI
>
> Hi,
>
>I am getting this error message while executing the  "module load
> chan_dahdi.so".
>
> astrisks*CLI> module load chan_dahdi.so
>
> Unable to load module chan_dahdi.so
> Command 'module load chan_dahdi.so' failed.
>  == Parsing '/etc/asterisk/chan_dahdi.conf':   == Found
>  == Parsing '/etc/asterisk/chan_dahdi_general.conf':   == Found
>  == Parsing '/etc/asterisk/chan_dahdi_groups.conf':   == Found
>  == Parsing '/etc/asterisk/dahdi-channels.conf':   == Found
>  == Parsing '/etc/asterisk/chan_dahdi_additional.conf':   == Found
>
>
> Thanks,
>
> Michael.k
>
>
>
> On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling  wrote:
>
>
>What happens when you do the module load chan_dahdi.so command?
>
>
>-Original Message-
>From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
>
>Sent: Thursday, October 06, 2011 12:10 PM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] dahdi show status command not avilable
> in CLI
>
>Hi,
>
>
>astrisks*CLI> module unload chan_dahdi.so
>
>Unable to unload resource chan_dahdi.so
>Command 'module unload chan_dahdi.so ' failed.
>
>Producing some other error messages !
>
>
>On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling 
> wrote:
>
>
>   In the Asterisk CLI run the commands "module unload
> chan_dahdi.so" and "module load chan_dahdi.so".
>
>
>
>
>   -Original Message-
>   From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
>   Sent: Thursday, October 06, 2011 11:40 AM
>   To: Asterisk Users Mailing List - Non-Commercial Discussion
>   Subject: Re: [asterisk-users] dahdi show status command not
> avilable in CLI
>
>   Hi,
>
>I was run the commands "dahdi_genconf" and dahdi_cfg
> outside the CLI as the part of x100p card installation. Before issuing this
> command the dahdi show status command was available. There may any issues ?
>
>
>   Michael.k
>
>
>
>   On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo <
> gincantal...@fgasoftware.com> wrote:
>
>
>
>  Hi Michael,
>
>  what if you reload the module chan_dahdi from within
> the * CLI? It should give some hints.
>
>  Giorgio
>
>
>
>  On 10/06/2011 05:22 PM, michael k wrote:
>
>  Hi Giorgio,
>
>  Thanks for your reply. I will produce some
> output for your reference.
>
>  # lsmod | grep dahdi
>
>  dahdi_echocan_mg2  39688  1
>  dahdi_transcode42372  1 wctc4xxp
>  dahdi_voicebus 79424  2
> wctdm24xxp,wcte12xp
>  dahdi 238384  14
> dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp
>  crc_ccitt  35265  2
> wctdm24xxp,dahdi
>
>
>  # service dahdi status
>
>  ### Span  1: WCFXO/0 "Wildcard X100P Board 1"
> (MASTER)
>1 FXOFXSKS   (SWEC: MG2)
> (battery)
>
>
>  # dahdi_cfg -vv
>
>  DAHDI Tools Version - 2.3.0
>
>  DAHDI Version: 2.3.0.1
>  Echo Canceller(s): MG2
>  Configuration
>  ==
>
>  Channel map:
>
>  Channel 01: FXS Kewlstart (Default) (Echo
> Canceler: mg2) (Slaves: 01)
>
>  1 channels to configure.
>
>

Re: [asterisk-users] dahdi show status command not avilable in CLI

2011-10-07 Thread Eric Wieling 
It is likely you have an error in your /etc/asterisk/chan_dahdi.conf

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
Sent: Friday, October 07, 2011 9:24 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI

Hi,

I am getting this error message while executing the  "module load 
chan_dahdi.so". 

astrisks*CLI> module load chan_dahdi.so

Unable to load module chan_dahdi.so
Command 'module load chan_dahdi.so' failed.
  == Parsing '/etc/asterisk/chan_dahdi.conf':   == Found
  == Parsing '/etc/asterisk/chan_dahdi_general.conf':   == Found
  == Parsing '/etc/asterisk/chan_dahdi_groups.conf':   == Found
  == Parsing '/etc/asterisk/dahdi-channels.conf':   == Found
  == Parsing '/etc/asterisk/chan_dahdi_additional.conf':   == Found


Thanks,

Michael.k



On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling  wrote:


What happens when you do the module load chan_dahdi.so command?


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k

Sent: Thursday, October 06, 2011 12:10 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] dahdi show status command not avilable in 
CLI

Hi,


astrisks*CLI> module unload chan_dahdi.so

Unable to unload resource chan_dahdi.so
Command 'module unload chan_dahdi.so ' failed.

Producing some other error messages !


On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling  wrote:


   In the Asterisk CLI run the commands "module unload 
chan_dahdi.so" and "module load chan_dahdi.so".




   -Original Message-
   From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
   Sent: Thursday, October 06, 2011 11:40 AM
   To: Asterisk Users Mailing List - Non-Commercial Discussion
   Subject: Re: [asterisk-users] dahdi show status command not 
avilable in CLI

   Hi,

I was run the commands "dahdi_genconf" and dahdi_cfg 
outside the CLI as the part of x100p card installation. Before issuing this 
command the dahdi show status command was available. There may any issues ?


   Michael.k



   On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo 
 wrote:



  Hi Michael,

  what if you reload the module chan_dahdi from within the 
* CLI? It should give some hints.

  Giorgio



  On 10/06/2011 05:22 PM, michael k wrote:

  Hi Giorgio,

  Thanks for your reply. I will produce some output 
for your reference.

  # lsmod | grep dahdi

  dahdi_echocan_mg2  39688  1
  dahdi_transcode42372  1 wctc4xxp
  dahdi_voicebus 79424  2 
wctdm24xxp,wcte12xp
  dahdi 238384  14 
dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp
  crc_ccitt  35265  2 wctdm24xxp,dahdi


  # service dahdi status

  ### Span  1: WCFXO/0 "Wildcard X100P Board 1" 
(MASTER)
1 FXOFXSKS   (SWEC: MG2) (battery)


  # dahdi_cfg -vv

  DAHDI Tools Version - 2.3.0

  DAHDI Version: 2.3.0.1
  Echo Canceller(s): MG2
  Configuration
  ==

  Channel map:

  Channel 01: FXS Kewlstart (Default) (Echo 
Canceler: mg2) (Slaves: 01)

  1 channels to configure.

  Setting echocan for channel 1 to mg2


  # dahdi_scan

  [1]
  active=yes
  alarms=OK
  description=Wildcard X100P Board 1
  name=WCFXO/0

Re: [asterisk-users] dahdi show status command not avilable in CLI

2011-10-07 Thread michael k 
Hi,

I am getting this error message while executing the  "module load
chan_dahdi.so".

astrisks*CLI> module load chan_dahdi.so

Unable to load module chan_dahdi.so
Command 'module load chan_dahdi.so' failed.
  == Parsing '/etc/asterisk/chan_dahdi.conf':   == Found
  == Parsing '/etc/asterisk/chan_dahdi_general.conf':   == Found
  == Parsing '/etc/asterisk/chan_dahdi_groups.conf':   == Found
  == Parsing '/etc/asterisk/dahdi-channels.conf':   == Found
  == Parsing '/etc/asterisk/chan_dahdi_additional.conf':   == Found


Thanks,

Michael.k


On Thu, Oct 6, 2011 at 9:44 PM, Eric Wieling  wrote:

> What happens when you do the module load chan_dahdi.so command?
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
> Sent: Thursday, October 06, 2011 12:10 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] dahdi show status command not avilable in CLI
>
> Hi,
>
>
> astrisks*CLI> module unload chan_dahdi.so
>
> Unable to unload resource chan_dahdi.so
> Command 'module unload chan_dahdi.so ' failed.
>
> Producing some other error messages !
>
>
> On Thu, Oct 6, 2011 at 9:17 PM, Eric Wieling  wrote:
>
>
>In the Asterisk CLI run the commands "module unload chan_dahdi.so"
> and "module load chan_dahdi.so".
>
>
>
>
>-Original Message-
>From: asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
>Sent: Thursday, October 06, 2011 11:40 AM
>To: Asterisk Users Mailing List - Non-Commercial Discussion
>Subject: Re: [asterisk-users] dahdi show status command not avilable
> in CLI
>
>Hi,
>
> I was run the commands "dahdi_genconf" and dahdi_cfg outside
> the CLI as the part of x100p card installation. Before issuing this command
> the dahdi show status command was available. There may any issues ?
>
>
>Michael.k
>
>
>
>On Thu, Oct 6, 2011 at 8:59 PM, gincantalupo <
> gincantal...@fgasoftware.com> wrote:
>
>
>
>   Hi Michael,
>
>   what if you reload the module chan_dahdi from within the *
> CLI? It should give some hints.
>
>   Giorgio
>
>
>
>   On 10/06/2011 05:22 PM, michael k wrote:
>
>   Hi Giorgio,
>
>   Thanks for your reply. I will produce some output for
> your reference.
>
>   # lsmod | grep dahdi
>
>   dahdi_echocan_mg2  39688  1
>   dahdi_transcode42372  1 wctc4xxp
>   dahdi_voicebus 79424  2 wctdm24xxp,wcte12xp
>   dahdi 238384  14
> dahdi_echocan_mg2,xpp,dahdi_transcode,wcb4xxp,wctdm,wcfxo,wctdm24xxp,wcte11xp,wct1xxp,wcte12xp,dahdi_voicebus,wct4xxp
>   crc_ccitt  35265  2 wctdm24xxp,dahdi
>
>
>   # service dahdi status
>
>   ### Span  1: WCFXO/0 "Wildcard X100P Board 1"
> (MASTER)
> 1 FXOFXSKS   (SWEC: MG2) (battery)
>
>
>   # dahdi_cfg -vv
>
>   DAHDI Tools Version - 2.3.0
>
>   DAHDI Version: 2.3.0.1
>   Echo Canceller(s): MG2
>   Configuration
>   ==
>
>   Channel map:
>
>   Channel 01: FXS Kewlstart (Default) (Echo Canceler:
> mg2) (Slaves: 01)
>
>   1 channels to configure.
>
>   Setting echocan for channel 1 to mg2
>
>
>   # dahdi_scan
>
>   [1]
>   active=yes
>   alarms=OK
>   description=Wildcard X100P Board 1
>   name=WCFXO/0
>   manufacturer=Digium
>   devicetype=Wildcard X100P
>   location=PCI Bus 02 Slot 02
>   basechan=1
>   totchans=1
>   irq=193
>   type=analog
>   port=1,FXO
>
>
>   These outputs shows that the modules are loaded
> correctly. Any other clues ?
>
>   Michael.k
>
>
>
>   On Thu, Oct 6, 2011 at 8:43 PM, gincantalupo <
> gincantal...@fgasoftware.com> wrote:
>
>
>   Hi Michael,
>
>   if you type "dah" followed by TAB and nothing
> appears, it means you do not have dahdi module loaded or dahdi_cfg
> application not launched before starting asterisk.
>
>   Giorgio
>
>
>   On 10/06/2011 04:57 PM, michael k wrote:
>
>   Hi All,
>
>

[asterisk-users] Asterisk 1.8.7 and client outside network

2011-10-07 Thread Administrator TOOTAI 

Hi,

my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and 
GrandStream) connected from the lan


I now want to connect a snom320 from outside but it failed, having always

[Oct  7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt: 
getnameinfo(): ai_family not supported
[Oct  7 14:48:04] WARNING[3870]: chan_sip.c:13597 
parse_register_contact: Domain 'XX.XXX.XXX.XX:2048' disallowed by 
contact ACL (violating IP )
[Oct  7 14:48:04] WARNING[3870]: chan_sip.c:14306 register_verify: 
Registration denied because of contact ACL


doesn't matter if I connect through a VPN or to the public IP using STUN.


My sip.conf:

localnet=172.24.0.0/12
localnet=169.254.0.0/255.255.0.0 ; Zero conf local network
localnet=
autodomain=yes
;allowexternaldomains=yes
domain=172.24.30.250 ;Asterisk Server IP
domain=
domain=

[309](snom320,ulaw-phone,callgroup1)
type=friend
insecure=port,invite
secret=VoIP2auDIo
contactdeny=0.0.0.0/0.0.0.0
contactpermit=XX.XXX.XXX.XX/32 ; External IP from phone, same as 
disallowed by contact ACL

deny=0.0.0.0/0.0.0.0
permit=XX.XXX.XXX.XX/32
nat=yes

Any clue? Why violating IP is empty?

Thanks for your help

--
Daniel

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[asterisk-users] Asterisk 1.8.7 and ReceiveFAX

2011-10-07 Thread Administrator TOOTAI 

Hi,

I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken 
from deb http://packages.asterisk.org/deb lucid main) including dahdi 
from this same repository. No FFA involved.


On incoming calls (only SIP, no telephony card), fax detection is 
working but reception failed with


 -- Executing [fax@from-TOOTAiAudio:19] 
ReceiveFAX("SIP/tootaiAUDIO-0564", "/tmp/1317991071.1614.tiff,c") in 
new stack
[Oct  7 14:37:52] WARNING[6961]: res_fax.c:1651 receivefax_exec: 
ReceiveFAX does not support polling
  == Spawn extension (from-TOOTAiAudio, fax, 19) exited non-zero on 
'SIP/tootaiAUDIO-0564'


What can be the problem?

Thanks for any hint.

--
Daniel

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[asterisk-users] Problem With Playing Busy Tone

2011-10-07 Thread Jon Farmer 
Hi

Since upgrading to 1.8.4.3 my callers no longer hear busy tone when I
use playtones().


Here is the CLI output on such a case

http://pastebin.com/TMBFhngh

Any ideas anyone?

Regards

Jon

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Re: [asterisk-users] call pickup

2011-10-07 Thread isrlgb 
Search for dialog-info pickup
-Original Message-
From: Marek Cervenka 
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 07 Oct 2011 09:47:45 
To: Asterisk Users Mailing List - Non-Commercial 
Discussion
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] call pickup

On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote:
> Am 05.10.2011 20:42, schrieb Marek Cervenka:
>> hello,
>>
>> is there some way to notify people in the same pickup group about call
>> from caller to callee?
>>
>> i.e. i have call from 111 to 222
>> there are 222,333,444 in the same pickup group
>>
>> 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup
>> the call with *8
>>
>> siemens have this on their sip openstage phones. how they do this?
>
> You can have that with subscriptions/hints, for example Snom phones
> can display not only a call to one of the peers but also the caller
> and callee
> identification.
>

can you point me to some doc/examples?
how this is implemented in SIP?
i think about sending some notify from dialplan (i have incoming call, i
know who is in pickup group, i can send call to callee and before send
some "NOTIFY" to other phones in the pickupgroup)
i found only one app like this - jabbersend. but i need this
notification on phone screen

> This works jaw to cheek with BLF (busy lamp field) which allows to
> monitor
> other extensions' status (in_use, ringing...).
>
> Of course you can be member of a pickup group without "monitoring" the
> status of any of the peers, and you can monitor a peer's status without
> being in the same pickup group (although not pickup the call then,
> obviously :-)
>


-- 
---
Marek Cervenka
Centrum Vypocetni Techniky
jabber  - cerv...@njs.netlab.cz
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA   - http://www.fpf.slu.cz
RHCE 100-175-678
===


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Re: [asterisk-users] call pickup

2011-10-07 Thread Marek Cervenka 
On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote:
> Am 05.10.2011 20:42, schrieb Marek Cervenka:
>> hello,
>>
>> is there some way to notify people in the same pickup group about call
>> from caller to callee?
>>
>> i.e. i have call from 111 to 222
>> there are 222,333,444 in the same pickup group
>>
>> 333,444 see on the phone (aastra) that 111 calling to 222 and can pickup
>> the call with *8
>>
>> siemens have this on their sip openstage phones. how they do this?
>
> You can have that with subscriptions/hints, for example Snom phones
> can display not only a call to one of the peers but also the caller
> and callee
> identification.
>

can you point me to some doc/examples?
how this is implemented in SIP?
i think about sending some notify from dialplan (i have incoming call, i
know who is in pickup group, i can send call to callee and before send
some "NOTIFY" to other phones in the pickupgroup)
i found only one app like this - jabbersend. but i need this
notification on phone screen

> This works jaw to cheek with BLF (busy lamp field) which allows to
> monitor
> other extensions' status (in_use, ringing...).
>
> Of course you can be member of a pickup group without "monitoring" the
> status of any of the peers, and you can monitor a peer's status without
> being in the same pickup group (although not pickup the call then,
> obviously :-)
>


-- 
---
Marek Cervenka
Centrum Vypocetni Techniky
jabber  - cerv...@njs.netlab.cz
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA   - http://www.fpf.slu.cz
RHCE 100-175-678
===


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