Re: [asterisk-users] SIP/GSM-gateway recommendation?

[Michelle Konzack]

Hello Stefan Gofferje,

Am 2012-07-25 22:35:28, hacktest Du folgendes herunter:
> can anybody recommend a priceworthy SIP/GSM-gateway that's known to work
> flawlessly with asterisk?

Normaly, an USB Stick Huawei K3765-HV would be enough...  However, under
Linux I had problems with the stability and now use the same stick  with
the Vodafone EasyBox 803A in which I stick the Huawei  and  then  use  a
multiport ISDN card in my Server to connect it.

> Should especially support CLIP/CLIR in both directions and it would be
> perfect if it would send notifications e.g. if the incoming call is
> diverted or if the remote party puts me on hold.

How do you wan to detect, if the remote party put you on hold?

AFAIK there is nothing in the GSM Protocol/Specs

> I don't favor GSM-PCI-cards because I'm just building a new asterisk
> based an an Atom board in a small casing.

If you use more then one USB-Stick, you will have to use an  USB  TT-Hub
or seperated USB ports on your board.

Thanks, Greetings and nice Day/Evening
Michelle Konzack

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Re: [asterisk-users] Asterisk

[Satish Barot]

Hi Herve,
Asterisk is legal in India and using it for Fax shouldn't create any issues
as far as legality is concerned.
Look at following link to get some  idea on VoIP regulation in India.
http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/8_5_1/ccmfeat/fslopar.html#wp1114625

--Satish Barot

On Wed, Jul 25, 2012 at 2:47 PM, Herve Prime  wrote:

> Is it legal to use Asterisk in India?
> We are planning to use Asterisk for Fax server and auto attendant.
>
> Thanks
>
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Re: [asterisk-users] Confbridge examples for Asterisk 10?

[Doug Lytle]

cjwstudios wrote:

Does anyone have any application examples for Confbridge in Asterisk
10?


I'm looking for such examples as well.  But just a note, meetme is still 
available if you're compiling from source, you just have to enable it.


Doug


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[asterisk-users] What TTS to use?

[Ishfaq Malik]

Hi

I'm thinking about deploying TTS onto our asterisk servers and was just
wondering which ones people use and like...

Thanks

Ish
-- 
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Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552


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[asterisk-users] callback - disa

[Федорчук Олег]

Hi/
I am newbe in asterisk.
I try to setup callback with Disa on my home server
Anybody help me, pls

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Re: [asterisk-users] What TTS to use?

[virendra bhati]

There are lot of TTS it's depends on you which one you like,

flite
festival
google
swift

main things of TTS is it's Voice accent.


On Thu, Jul 26, 2012 at 3:38 PM, Ishfaq Malik  wrote:

> Hi
>
> I'm thinking about deploying TTS onto our asterisk servers and was just
> wondering which ones people use and like...
>
> Thanks
>
> Ish
> --
> Ishfaq Malik 
> Department: VOIP Support
> Company: Packnet Limited
> t: +44 (0)845 004 4994
> f: +44 (0)161 660 9825
> e: i...@pack-net.co.uk
> w: http://www.pack-net.co.uk
>
> Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
> NORTH, MANCHESTER
> SCIENCE PARK, MANCHESTER, M156SE
> COMPANY REG NO. 04920552
>
>
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>



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Thanks and regards

 Virendra Bhati
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Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
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Re: [asterisk-users] What TTS to use?

[Chris Bagnall]

On 26/7/12 11:08 am, Ishfaq Malik wrote:

I'm thinking about deploying TTS onto our asterisk servers and was just
wondering which ones people use and like...


We've tried Festival, Cepstral and Ivona.

Ivona was by far and away the best.

If you need free (or very low cost) then your only real option is 
Festival. Just make sure to pick one of the newer voices.


This was all based on UK English. If you're after something else, you 
may find different results.



Kind regards,

Chris
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Re: [asterisk-users] What TTS to use?

[Ishfaq Malik]

On Thu, 2012-07-26 at 11:40 +0100, Chris Bagnall wrote:
> On 26/7/12 11:08 am, Ishfaq Malik wrote:
> > I'm thinking about deploying TTS onto our asterisk servers and was just
> > wondering which ones people use and like...
> 
> We've tried Festival, Cepstral and Ivona.
> 
> Ivona was by far and away the best.
> 
> If you need free (or very low cost) then your only real option is 
> Festival. Just make sure to pick one of the newer voices.
> 
> This was all based on UK English. If you're after something else, you 
> may find different results.
> 
> 
> Kind regards,
> 
> Chris

Hi Chris

UK English is exactly what we're after. Did you try flite at all?

Thanks

Ish

-- 
Ishfaq Malik 
Department: VOIP Support
Company: Packnet Limited
t: +44 (0)845 004 4994
f: +44 (0)161 660 9825
e: i...@pack-net.co.uk
w: http://www.pack-net.co.uk

Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET
NORTH, MANCHESTER
SCIENCE PARK, MANCHESTER, M156SE
COMPANY REG NO. 04920552


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Re: [asterisk-users] What TTS to use?

[Chris Bagnall]

UK English is exactly what we're after. Did you try flite at all?


No, I wasn't aware of flite when we ran these tests.



Kind regards,

Chris
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Re: [asterisk-users] What TTS to use?

[Patrick Lists]

On 26-07-12 12:40, Chris Bagnall wrote:

On 26/7/12 11:08 am, Ishfaq Malik wrote:

I'm thinking about deploying TTS onto our asterisk servers and was just
wondering which ones people use and like...


We've tried Festival, Cepstral and Ivona.


I didn't know about Ivona so thanks for mentioning it. Their UK English 
voice sounds very good. And with Cepstral's price hike with their 
release of version 6 (and killing of 5) it's good to know there's an 
alternative.


Apparently Ivona has an GPL licensed Asterisk plug-in 
(http://www.ivona.com/en/telecom/asterisk/). Do you know where that 
plug-in can be downloaded? I could not find it.


Regards,
Patrick


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[asterisk-users] callback on busy

[pepesz]

Dear all,

I know the topic comes back like boomerang, but I did not find a nice
solution.
Does someone has/knows how to achieve "call back on busy" otherwise called
camping?
If one is calling the extension and it is busy, then caller should get
something like "Press 5 to request call back" and after the previous call
is finished the system should:
1) call caller
2) dial callee

or something similar ;)

This topic comes back so many times - I'm wonder if there is already a
function for that implemented in asterisk (my current one is 10.5)
Thanks in advance.

pepesz
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[asterisk-users] Realtime Queue and Queue_members

[Jonas Kellens]

Hello,

is there a way to order call queue members in the database table?

When defining the table for realtime queue_members, I notice there is no 
ID-column.


Can I add an ID-column, or will this fail realtime Queues ?


Kind regards,
Jonas.
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asterisk-users@lists.digium.com

[Jorge Mendoza]

Thank you again Mitul. 
Ok, the we ill use E&M. 
Regards 
Jorge Mendoza 

- Original Message -

From: "Mitul Limbani"  
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
 
Sent: Wednesday, 25 July, 2012 11:35:08 PM 
Subject: Re: [asterisk-users] Dahdi+Redfone+Channel Bank+E&M 


Same for E1 as well unless your operator is giving mfcr2 on cas. 
Mitul 
On Jul 26, 2012 9:17 AM, "Jorge Mendoza" < jmendo...@tcc.com.pe > wrote: 




Thank you Mitul for your answer. 
Yes, we have tested e&m before and it works. But I don't know why. That is what 
I don't understand. You said that E&M signalling does not have separate 
signalling channel, that is true for the T1 but not for E1. My understanding is 
that E1 CAS pass the E&M information in the bits abcd of channel 16, the 
signalling channel. 
Regards 
-- 
Jorge Mendoza 



From: "Mitul Limbani" < mi...@enterux.in > 
To: "Asterisk Users Mailing List - Non-Commercial Discussion" < 
asterisk-users@lists.digium.com > 
Sent: Wednesday, 25 July, 2012 8:15:25 PM 
Subject: Re: [asterisk-users] Dahdi+Redfone+Channel Bank+E&M 


E&M signalling do not have seperate signalling channel. 
Configure as e&m=1-31 
Mitul 
On Jul 26, 2012 6:40 AM, "Jorge Mendoza" < jmendo...@tcc.com.pe > wrote: 


Hi, 

We are trying to connect an Asterisk server with a Channel Bank with E&M 
interfaces using a RedFone TDMoE device. 
The CB have a E1 CAS interface. 
OS: Ubuntu Server 11.10 64 bits 
dahdi: dahdi-linux-complete-2.6.1+2.6.1 

Redfone configuration: 
/etc/redfone.conf 

[span1] 
framing=cas 
encoding=hdb3 

System configuration: 
/etc/dahdi/system.conf 

dynamic=ethmf,eth0/00:50:c2:65:e0:64/0,31,1 
dynamic=ethmf,eth0/00:50:c2:65:e0:64/1,31,0 

e&me1=1-15,17-31 
dchan=16 

alaw=1-31 

loadzone = fr 
defaultzone = fr 

Error message: 
 
# dahdi_cfg -v 

Changing signalling on channel 1 from Unused to E & M E1 
Changing law on channel 1 from Mu-law to A-law 
DAHDI_CHANCONFIG failed on channel 1: Invalid argument (22) 
Selected signaling not supported 
Possible causes: 
e&me1 signaling is being used on a T1 line (use e&m) 
RBS signaling is being used on a E1 CCS span 
Signaling is being assigned to channel 16 of an E1 CAS span 
= 

I don't understand the last possible cause mentioned: "Signaling is being 
assigned to channel 16 of an E1 CAS span", because the dchan is channel 16. 
Where is the error? 
Thank you. 
-- 
Jorge Mendoza 

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Re: [asterisk-users] callback on busy

[Danny Nicholas]

The "inherent" problem with this is that it either requires a "brute force"
solution or a "queue and call" solution.  The "brute Force" solution would
be something like this:

[callee-is-busy]

Exten => s,1,playback(callbackmsg)

Exten => s,n,wait(3)

Exten => s,n,playback(vm-goodbye)

Exten => s,n,hangup()

Exten => 5,1,agi(writecallback.sh,$(CALLERID(num)},${EXTEN})

Exten => 5,n,playback(vm-goodbye)

Exten => 5,n,hangup()

Exten => I,1,playback(invalid-selection)

Exten => I,n,goto(callee-is-busy,s,1)

 

Writecallback.sh

#!/bin/sh

Echo "Channel: DAHDI/g1/$1" > retcall.txt

Echo "CallerID: $2" >> retcall.txt

Echo "Maxtries: 50" >> retcall.txt

Echo "WaitTime: 60" >> retcall.txt

Echo "retryTime: 15" >> retcall.txt

Echo "Context: default" >> retcall.txt

Mv retcall.txt /var/spool/asterisk/outgoing

 

The "queue and call" solution would involve using a shell to monitor the
extension using AMI or asterisk -rx "core show channels verbose" to see when
the line became available, then launching the call using the
writecallback.sh above.

 

Here is a link to a "wait for available" solution -
http://www.voip-info.org/wiki/view/Asterisk+tips+campon

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of pepesz
Sent: Thursday, July 26, 2012 8:03 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] callback on busy

 

Dear all,

I know the topic comes back like boomerang, but I did not find a nice
solution.
Does someone has/knows how to achieve "call back on busy" otherwise called
camping?
If one is calling the extension and it is busy, then caller should get
something like "Press 5 to request call back" and after the previous call is
finished the system should:
1) call caller
2) dial callee

or something similar ;)

This topic comes back so many times - I'm wonder if there is already a
function for that implemented in asterisk (my current one is 10.5)
Thanks in advance. 

pepesz

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Re: [asterisk-users] callback - disa

[Danny Nicholas]

Try google-ing "asterisk disa" - help is offered more readily to those who
have tried first.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of  
Sent: Thursday, July 26, 2012 5:37 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] callback - disa

Hi/
I am newbe in asterisk.
I try to setup callback with Disa on my home server Anybody help me, pls

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Re: [asterisk-users] callback on busy

[Richard Mudgett]

> I know the topic comes back like boomerang , but I did not find a
> nice solution.
> Does someone has/knows how to achieve "call back on busy" otherwise
> called camping?
> If one is calling the extension and it is busy, then caller should
> get something like "Press 5 to request call back" and after the
> previous call is finished the system should:
> 1) call caller
> 2) dial callee
> 
> or something similar ;)
> 
> This topic comes back so many times - I'm wonder if there is already
> a function for that implemented in asterisk (my current one is 10.5)
> Thanks in advance.

You want call completion which has been in Asterisk since v1.8:

https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29

Richard

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Re: [asterisk-users] Confbridge examples for Asterisk 10?

[cjwstudios]

Hi Doug,

I did find the following on voip-info.

http://www.voip-info.org/wiki/view/Asterisk+cmd+ConfBridge

It's somewhat rudimentary but it does work.

Thanks,
C

On Thu, Jul 26, 2012 at 2:36 AM, Doug Lytle  wrote:
> cjwstudios wrote:
>>
>> Does anyone have any application examples for Confbridge in Asterisk
>> 10?
>
>
> I'm looking for such examples as well.  But just a note, meetme is still
> available if you're compiling from source, you just have to enable it.
>
> Doug
>
>
> --
> Ben Franklin quote:
>
> "Those who would give up Essential Liberty to purchase a little Temporary
> Safety, deserve neither Liberty nor Safety."
>
>
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[asterisk-users] Asterisk Realtime issue after registering with x-lite

[virendra bhati]

Hi All,

I have an small issue, which is not creating any problem on working syatem
but not sure about the problem that is why eager to know about it. I had
installed Asterisk realtime with Asterisk 1.4.41. Every thing is working
good but getting warning at Asterisk CLI.

[Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 1000
Really destroying SIP dialog
'0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method:
SUBSCRIBE
[Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 1000
Really destroying SIP dialog
'9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method:
SUBSCRIBE


If anyone have any suggestion please reply to me.

-- 

Thanks and regards

 Virendra Bhati
+91-9718300881
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
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Re: [asterisk-users] Asterisk Realtime issue after registering withx-lite

[motty.cruz]

can you post your sip.conf for  Exten. 1000?
it does not seem like you have 
[1000]
 
mailbox=1000@default
 
 
Thanks, 
-motty

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Thursday, July 26, 2012 10:35 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk Realtime issue after registering
withx-lite


Hi All,

I have an small issue, which is not creating any problem on working syatem
but not sure about the problem that is why eager to know about it. I had
installed Asterisk realtime with Asterisk 1.4.41. Every thing is working
good but getting warning at Asterisk CLI.

[Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 1000
Really destroying SIP dialog
'0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method:
SUBSCRIBE
[Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
valid host
[Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 1000
Really destroying SIP dialog
'9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method:
SUBSCRIBE


If anyone have any suggestion please reply to me. 

-- 


Thanks and regards

 Virendra Bhati
+91-9718300881
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)


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Re: [asterisk-users] Asterisk Realtime issue after registering withx-lite

[virendra bhati]

My sip.conf don't have any entry related to sip pees. I have everything
into database.

for more details please check below url, which have good example of
asterisk realtime

http://bahjons.com/stuff/asterisk-realtime-installation-guide

On Thu, Jul 26, 2012 at 11:14 PM, motty.cruz  wrote:

> **
> can you post your sip.conf for  Exten. 1000?
> it does not seem like you have
> [1000]
>
> mailbox=1000@default
>
> **
> *Thanks, *
> *-motty*
>
>  --
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
> *Sent:* Thursday, July 26, 2012 10:35 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] Asterisk Realtime issue after registering
> withx-lite
>
>  Hi All,
>
> I have an small issue, which is not creating any problem on working syatem
> but not sure about the problem that is why eager to know about it. I had
> installed Asterisk realtime with Asterisk 1.4.41. Every thing is working
> good but getting warning at Asterisk CLI.
>
> [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
> valid host
> [Jul 26 21:17:36] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
> valid host
> [Jul 26 21:17:36] NOTICE[17811]: chan_sip.c:16897
> handle_request_subscribe: Received SIP subscribe for peer without mailbox:
> 1000
> Really destroying SIP dialog
> '0415a756e3185f77MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method:
> SUBSCRIBE
> [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
> valid host
> [Jul 26 21:20:37] WARNING[17811]: chan_sip.c:10571 check_via: '[' is not a
> valid host
> [Jul 26 21:20:37] NOTICE[17811]: chan_sip.c:16897
> handle_request_subscribe: Received SIP subscribe for peer without mailbox:
> 1000
> Really destroying SIP dialog
> '9e6fd45fdb070a15MTIwZmNmOWZmNmVlZjJjYmM1ZWJlMjk1NGEzYTdkOTg.' Method:
> SUBSCRIBE
>
>
> If anyone have any suggestion please reply to me.
>
> --
>
> Thanks and regards
>
>  Virendra Bhati
> +91-9718300881
> Asterisk Developer
> E-mail-: virbh...@gmail.com
> Skype id:- virbhati2
> New Delhi(India)
>
>
> --
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>
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> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 

Thanks and regards

 Virendra Bhati
+91-9718300881
Asterisk Developer
E-mail-: virbh...@gmail.com
Skype id:- virbhati2
New Delhi(India)
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[asterisk-users] Video conferencing?

[Ken D'Ambrosio]

Hi, all.  I see that, with Asterisk 10, there've been some additions with an
eye toward conferencing, and, apparently, hooks for video conferencing. 
Googling like crazy, however, has given me little to go on.  I've been tasked
with bringing a video conferencing solution in-house, and have yet to find a
decent standalone OSS solution, but I begin to wonder if Asterisk (perhaps in
conjunction with some application?) could do the trick.

If anyone's had any good experiences with (preferably) Asterisk video
conferencing, or (failing that) OSS video conferencing, please let me know.

Thanks much!

-Ken






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[asterisk-users] Video conferencing?

[Ken D'Ambrosio]

Hi, all.  I'm 99% sure that Asterisk technically *supports* 
videoconferencing -- at least, as a conduit -- but are there products 
out there that leverage that?  I've been tasked with bringing 
videoconferencing internal to my company, and had been coming up empty 
looking for standalone solutions, when I suddenly realized that my 
favorite PBX software might be able to help out.


Thanks much for any pointers you might be able to give me,

-Ken




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Re: [asterisk-users] Video conferencing (and SMTP server hiccups)?

[Ken D'Ambrosio]

Apologies for the multiple sends -- I'd been having some outbound SMTP issues,
and thought the first one had fallen into the ether.  Turned out, it was the
upstream host that was the issue.  Once kicked, lo!

-Ken


On Wed, 25 Jul 2012 14:24:50 -0400 Ken D'Ambrosio  wrote

> Hi, all.  I'm 99% sure that Asterisk technically *supports* 
> videoconferencing -- at least, as a conduit -- but are there products 
> out there that leverage that?  I've been tasked with bringing 
> videoconferencing internal to my company, and had been coming up empty 
> looking for standalone solutions, when I suddenly realized that my 
> favorite PBX software might be able to help out.
> 
> Thanks much for any pointers you might be able to give me,
> 
> -Ken
> 
> 
> 
> 
> --
> _
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Re: [asterisk-users] Video conferencing?

[Jonathan Rose]

Ken D'Ambrosio wrote:
> From: "Ken D'Ambrosio" 
> To: asterisk-users@lists.digium.com
> Sent: Wednesday, July 25, 2012 1:24:50 PM
> Subject: [asterisk-users] Video conferencing?
> 
> Hi, all.  I'm 99% sure that Asterisk technically *supports*
> videoconferencing -- at least, as a conduit -- but are there products
> out there that leverage that?  I've been tasked with bringing
> videoconferencing internal to my company, and had been coming up
> empty
> looking for standalone solutions, when I suddenly realized that my
> favorite PBX software might be able to help out.
> 
> Thanks much for any pointers you might be able to give me,
> 
> -Ken

I haven't used much in the way of video myself, but Asterisk 10 did some
major overhauls to the confbridge application and I believe you can perform
video teleconferencing with basically any SIP device/program that allows
for a video stream, at least as long as it supports one of the video formats
that Asterisk uses (h263 or h264). You could do some basic testing for this
by running a few softphones connected to your Asterisk box. Jitsi does video
calling as does Linphone... though I've personally had some trouble setting
up h264 to work with Linphone. Good luck.

--
Jonathan R. Rose
Digium, Inc. | Software Engineer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
direct +1 256 428 6139 

Check us out at: http://digium.com & http://asterisk.org

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Re: [asterisk-users] Video conferencing?

[Carlos Chavez]

On Wed, 2012-07-25 at 16:05 -0400, Ken D'Ambrosio wrote:
> Hi, all.  I see that, with Asterisk 10, there've been some additions with an
> eye toward conferencing, and, apparently, hooks for video conferencing. 
> Googling like crazy, however, has given me little to go on.  I've been tasked
> with bringing a video conferencing solution in-house, and have yet to find a
> decent standalone OSS solution, but I begin to wonder if Asterisk (perhaps in
> conjunction with some application?) could do the trick.
> 
> If anyone's had any good experiences with (preferably) Asterisk video
> conferencing, or (failing that) OSS video conferencing, please let me know.
> 
Asterisk 10 now supports basic video conference functions with
Confbridge.  I have not personally used this as we are still using 1.8
on all production servers at the moment but I saw it working last year
at Astricon.  From what I saw you can have all participants call in to
the same conference but they will only see the video from the active
speaker, not all participants at the same time.  Video source will
change to the active speaker.

There was another product shown at Astricon that allowed full video
conference integration with Asterisk: http://www.projectdiastar.org/

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001


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Re: [asterisk-users] Video call using Asterisk

[Julio Araujo]

Hello guys,

Because I'm using AsteriskNOW and the FREEPBX was automatically installed the 
/etc/asterisk changed a little bit, so after read some .conf files I made a 
little modification on sip_general_custom.conf inserting the following lines:
videosupport=yes
allow=h263
and then video call between two extensions works as expected.

/Julio



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julio Araujo
Sent: terça-feira, 24 de julho de 2012 11:23
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Video call using Asterisk

I'm using the Asterisk 2.0.2 the latest released from the asterisk.org.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julio Araujo
Sent: terça-feira, 24 de julho de 2012 11:20
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Video call using Asterisk

Hello,

What is the set of configuration that should be done in the Asterisk 1.0.8 
using FreePBX that can allow a simple video call between two extensions?


Thanks in advance.
[http://www.ericsson.com/shared/images/Email_line.gif]
JULIO ARAUJO
TE ENGINEER MS

Ericsson
ITTE & Test Environment
São Jose dos Campos, Brazil
Phone +551239084121
SMS/MMS +551281150089
julio.ara...@ericsson.com
www.ericsson.com


[http://www.ericsson.com/shared/images/Email_campaigns.gif]

This Communication is Confidential. We only send and receive email on the basis 
of the terms set out at 
www.ericsson.com/email_disclaimer




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Re: [asterisk-users] IAX2 Registered OK without IP

[Alejandro Imass]

On Tue, Jul 24, 2012 at 3:54 PM, Alejandro Imass  wrote:
> On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass  wrote:
> we upgraded to 1.8.13.1 and we have much the same problem although after
> the upgrade I don't seem to find any cases where the qualify value is
> OK (xx ms) and the IP is gone (like we had in 1.4.29) but the effect
> is the same: the extension becomes non-reachable pretty quickly.
>

I stand corrected. It's EXACTLY the same behavior as 1.4.29, The
Status shows OK (XX ms) and the IP is "(null)"

> IMHO this is indicating that the qualify settings are being ignored,

I've been experimenting with qualifyfreqok and qualifyfreqnotok and
just by specifiying _any_ values for these parameters it makes matters
much worse.

> and the only workaround has been lowering the re-register time to
> sometimes as low as 3 seconds. Even though several docs say that the

A re-registration every 3 seconds seems to do the trick but why can't
qualify keep the connection alive??

> qualify values can also act as a keep alive, it's not working for us
> and I still have to reduce the register time to very low values
> because qualifyfreqok and qualifyfreqnotok don't see to be doing
> anything...
>

Stand corrected: they just make the problem worse.

> Any clues??

I get the feeling that IAX extensions are definitively not very
popular even though most of the older concerns about IAX should be
gone by now. We really want to support and keep investing into making
IAX work for us but the little support we get here is really
discouraging.

Best,

-- 
Alejandro Imass

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[asterisk-users] Polycom Presence with Asterisk 1.8.12.0

[Tim Nelson]

Greetings-

I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0 
system. Everything is running smoothly with few problems. However, I have an 
issue that maybe someone could shed light on...

Many of the phones have 'buddy watch' enabled for the other phones, basically 
Polycom's version of BLF. This works fine when watched extensions are on the 
phone, ringing, etc, as the LED lights/flashes appropriately for the status. 
However, the phones also offer various presence states such as 'Out to Lunch' 
or 'Away from Desk' etc. When a phone is set to one of these presence states, 
the other phones watching never show that status. Does that make sense? Is 
there any reason why those states would not propagate between the phones 
(through Asterisk?) ?

And, on a side note, if anyone knows how to remove the 'thistle' background 
from a Polycom phone I'd be especially delighted. It was set by a user on a 
device, and there is no option to remove it, or replace it with the blank 
background which is the default. :/

--Tim

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Re: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0

[Danny Nicholas]

Question 1 - I think asterisk only supports a limited set of statuses
Question 2 - you could reset the phone and re-provision it or possibly just
tweak the config file and update it.  I have 501's so the 550 is just a WAG.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, July 26, 2012 3:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0

Greetings-

I've got a handful of Polycom IP 550 handsets connected to an Asterisk
1.8.12.0 system. Everything is running smoothly with few problems. However,
I have an issue that maybe someone could shed light on...

Many of the phones have 'buddy watch' enabled for the other phones,
basically Polycom's version of BLF. This works fine when watched extensions
are on the phone, ringing, etc, as the LED lights/flashes appropriately for
the status. However, the phones also offer various presence states such as
'Out to Lunch' or 'Away from Desk' etc. When a phone is set to one of these
presence states, the other phones watching never show that status. Does that
make sense? Is there any reason why those states would not propagate between
the phones (through Asterisk?) ?

And, on a side note, if anyone knows how to remove the 'thistle' background
from a Polycom phone I'd be especially delighted. It was set by a user on a
device, and there is no option to remove it, or replace it with the blank
background which is the default. :/

--Tim

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Re: [asterisk-users] Polycom Presence with Asterisk 1.8.12.0

[Kevin P. Fleming]

On 07/26/2012 03:32 PM, Danny Nicholas wrote:

Question 1 - I think asterisk only supports a limited set of statuses


Asterisk does not *receive* presence updates from Polycom phones (or 
really, non-Digium phones) at all. Instead, the presence (status) 
updates you are seeing appear on your phones are the statuses that 
Asterisk itself generates based on the phones' activity.


--
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] Polycom Presence with Asterisk 1.8.12.0

[Dave Fullerton]

On 07/26/2012 04:28 PM, Tim Nelson wrote:

Greetings-

I've got a handful of Polycom IP 550 handsets connected to an Asterisk 1.8.12.0 
system. Everything is running smoothly with few problems. However, I have an 
issue that maybe someone could shed light on...

Many of the phones have 'buddy watch' enabled for the other phones, basically 
Polycom's version of BLF. This works fine when watched extensions are on the 
phone, ringing, etc, as the LED lights/flashes appropriately for the status. 
However, the phones also offer various presence states such as 'Out to Lunch' 
or 'Away from Desk' etc. When a phone is set to one of these presence states, 
the other phones watching never show that status. Does that make sense? Is 
there any reason why those states would not propagate between the phones 
(through Asterisk?) ?

And, on a side note, if anyone knows how to remove the 'thistle' background 
from a Polycom phone I'd be especially delighted. It was set by a user on a 
device, and there is no option to remove it, or replace it with the blank 
background which is the default. :/


If you just want to reset the background on that phone then you want:
Menu, 3, 1, 1, 4, 2, 2, Select (Seems like you should get god mode for 
that too, but alas, no).


If you want to prohibit anyone from setting that particular background 
you could always remove the jpg from the provisioning server.


As for the buddy status, I don't think it works (or ever will work) with 
asterisk. I always turn that button off when I set up my site sip.conf 
to avoid any questions.


-Dave



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[asterisk-users] Call ID of the second call leg

[Leandro Dardini]

Hello friends,
I am trying to deeply integrate asterisk cdr with voipmonitor cdr. I can
access the caller Call ID (fbasename field in voipmonitor cdr) looking at
the SIPCALLID variable in asterisk, but how can I access from within
asterisk the Call ID of the second leg of the call (the one originating
from asterisk to the destination peer)? is there a variable holding this
value?

Thank you

Leandro
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Re: [asterisk-users] IAX2 Registered OK without IP

[Duncan Turnbull]

On 27/07/2012, at 8:16 AM, Alejandro Imass  wrote:

> On Tue, Jul 24, 2012 at 3:54 PM, Alejandro Imass  wrote:
>> On Tue, Jun 12, 2012 at 4:04 PM, Alejandro Imass  wrote:
>> we upgraded to 1.8.13.1 and we have much the same problem although after
>> the upgrade I don't seem to find any cases where the qualify value is
>> OK (xx ms) and the IP is gone (like we had in 1.4.29) but the effect
>> is the same: the extension becomes non-reachable pretty quickly.
>> 
> 
Can you confirm whether you have a firewall between the phones or not? And also 
using tcpdump and IAX debug what packets you are seeing

This is a network problem and something is disrupting the return packets so you 
need to see where it's occurring 

>From the cli use the iax2 set debug command and watch what's happening. Are 
>the packets being returned? If they aren't check with tcpdump to see if they 
>are at least getting to your interface

Do the same at the other end to work out whats missing


> I stand corrected. It's EXACTLY the same behavior as 1.4.29, The
> Status shows OK (XX ms) and the IP is "(null)"
> 
>> IMHO this is indicating that the qualify settings are being ignored,
> 
> I've been experimenting with qualifyfreqok and qualifyfreqnotok and
> just by specifiying _any_ values for these parameters it makes matters
> much worse.
> 
>> and the only workaround has been lowering the re-register time to
>> sometimes as low as 3 seconds. Even though several docs say that the
> 
> A re-registration every 3 seconds seems to do the trick but why can't
> qualify keep the connection alive??
> 
>> qualify values can also act as a keep alive, it's not working for us
>> and I still have to reduce the register time to very low values
>> because qualifyfreqok and qualifyfreqnotok don't see to be doing
>> anything...
>> 
> 
> Stand corrected: they just make the problem worse.
> 
>> Any clues??
> 
> I get the feeling that IAX extensions are definitively not very
> popular even though most of the older concerns about IAX should be
> gone by now. We really want to support and keep investing into making
> IAX work for us but the little support we get here is really
> discouraging.
> 
> Best,
> 
> -- 
> Alejandro Imass
> 
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[asterisk-users] asterisk crash

[Brandon B.]

Hi, one of our asterisk servers recently crashed. Any direction as to how I
can provide helpful information about this issue would be appreciated:

asterisk*CLI> core show version
Asterisk 1.8.13.0 built by root @ asterisk on a i686 running Linux on
2012-06-10 22:22:13 UTC

Jul 26 16:39:39 asterisk kernel: [525204.496674] asterisk[5766]: segfault
at 0 ip b7521c86 sp b57a8f48 error 4 in libc-2.11.3.so[b74ad000+14]

The asterisk server is running on an up to date Debian 6 system, but
without the -g option as per
https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace, so I
beleive I cannot provide any more information.

Is this a libc6 problem?
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Re: [asterisk-users] asterisk crash

[Rusty Newton]

On 7/26/2012 6:21 PM, Brandon B. wrote:
Hi, one of our asterisk servers recently crashed. Any direction as to 
how I can provide helpful information about this issue would be 
appreciated:
If the seg fault happens again, make sure to get a non-optimized 
backtrace per the instructions you linked, and follow 
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines 
before posting it to the tracker.


--
Rusty Newton
Digium, Inc | Open Source Community Support Manager
Check us out at: www.digium.com www.asterisk.org


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Re: [asterisk-users] callback on busy

[Duncan Turnbull]

On 27/07/2012, at 3:42 AM, Richard Mudgett  wrote:

>> I know the topic comes back like boomerang , but I did not find a
>> nice solution.
>> Does someone has/knows how to achieve "call back on busy" otherwise
>> called camping?
>> If one is calling the extension and it is busy, then caller should
>> get something like "Press 5 to request call back" and after the
>> previous call is finished the system should:
>> 1) call caller
>> 2) dial callee
>> 
>> or something similar ;)
>> 
>> This topic comes back so many times - I'm wonder if there is already
>> a function for that implemented in asterisk (my current one is 10.5)
>> Thanks in advance.
> 
> You want call completion which has been in Asterisk since v1.8:
> 
> https://wiki.asterisk.org/wiki/display/AST/Call+Completion+Supplementary+Services+%28CCSS%29
> 

This is what I am guessing FreePBX uses to do it too

http://www.freepbx.org/trac/ticket/778


> Richard
> 
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[asterisk-users] still got ReceiveFax() problem, how to properly setup asterisk fax?

[Roi Stork]

I've posted my problem with ReceiveFax() a long time ago.
Majority of the incoming faxes still end up with a T2 timeout or
hangup (fax session hangup) errors.

Our Setup:
- we're using the Digium Free Fax module for Asterisk, all settings are default
- incoming/outgoing faxes go through an E1 line
- faxes are outgoing/received via Sangoma A104DE Card
- fax .tiff image is converted to pdf and sent to email
- clock source has already been set to NORMAL (from the E1 line), and
hardware/software echo cancellation already disabled

It's strange to have timeout errors since we're using an E1 line,
which we have tested to have no problems in voice quality.
I have read that it may be the fault of the sending fax machine or the
pstn line it's connected to, but no one in the office believes it.

They believe it's due to a poorly configured setup. But I don't
know/can't confirm if it's the cause.
Can anyone help me check the config files to see If I missed something?


---wanpipe1.conf--
[devices]
wanpipe1 = WAN_AFT_TE1, Comment

[interfaces]
w1g1 = wanpipe1, , TDM_VOICE, Comment

[wanpipe1]
CARD_TYPE   = AFT
S514CPU = A
CommPort= PRI
AUTO_PCISLOT= NO
PCISLOT = 4
PCIBUS  = 5
FE_MEDIA= E1
FE_LCODE= HDB3
FE_FRAME= CRC4
FE_LINE = 4
TE_CLOCK= NORMAL
TE_REF_CLOCK= 0
TE_SIG_MODE = CCS
TE_HIGHIMPEDANCE= NO
TE_RX_SLEVEL= 430
HW_RJ45_PORT_MAP = DEFAULT
LBO = 120OH
FE_TXTRISTATE   = NO
MTU = 1500
UDPPORT = 9000

TTL = 255
IGNORE_FRONT_END= NO
TDMV_SPAN   = 1
TDMV_DCHAN  = 16
TE_AIS_MAINTENANCE = NO #NO: defualt  YES: Start port in AIS
Blue Alarm and keep line down
#wanpipemon -i w1g1 -c Ttx_ais_off to
disable AIS maintenance mode

#wanpipemon -i w1g1 -c Ttx_ais_on to enable AIS maintenance mode
TDMV_HW_DTMF= YES   # YES: receive dtmf events from hardware
TDMV_HW_FAX_DETECT  = YES   # YES: receive fax
1100hz events from hardware
HWEC_OPERATION_MODE = OCT_NO_ECHO# OCT_NORMAL: echo
cancelation enabled with nlp (default)

 # OCT_SPEECH: improves software tone detection by disabling
NLP (echo possible)

 # OCT_NO_ECHO:disables echo cancelation but allows VQE/tone
functions.
HWEC_DTMF_REMOVAL   = NO# NO: default  YES: remove dtmf out of
incoming media (must have hwdtmf enabled)
HWEC_NOISE_REDUCTION= NO# NO: default  YES: reduces noise on
the line - could break fax
HWEC_ACUSTIC_ECHO   = NO# NO: default  YES: enables acustic
echo cancelation
HWEC_NLP_DISABLE= NO# NO: default  YES: guarantees
software tone detection (possible echo)
HWEC_TX_AUTO_GAIN   = 0 # 0: disable   -40-0: default tx audio
level to be maintained (-20 default)
HWEC_RX_AUTO_GAIN   = 0 # 0: disable   -40-0: default tx audio
level to be maintained (-20 default)
HWEC_TX_GAIN= 0 # 0: disable   -24-24: db
values to be applied to tx signal
HWEC_RX_GAIN= 0 # 0: disable   -24-24: db
values to be applied to tx signal

[w1g1]
ACTIVE_CH   = ALL
TDMV_HWEC   = NO
MTU = 8


--dahdi/system.conf
#autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
#autogenrated on 2012-06-29
#Dahdi Channels Configurations
#For detailed Dahdi options, view /etc/dahdi/system.conf.bak
loadzone=us
defaultzone=us

#Sangoma A104 port 4 [slot:4 bus:5 span:1] 
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
#echocanceller=none,1-15,17-31
hardhdlc=16


---chan_dahdi.conf
;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
;autogenrated on 2012-06-29
;Dahdi Channels Configurations
;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak

[trunkgroups]

[channels]
context=default
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
;echocancel=yes
;echocancelwhenbridged=yes
echocancel=no
echocancelwhenbridged=no
relaxdtmf=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no

;Sangoma A104 port 4 [slot:4 bus:5 span:1] 
switchtype=euroisdn
context=from-pstn
group=0
echocancel=no
faxdetect=both
signalling=pri_cpe
channel =>1-15,17-31



---extensions.conf:

exten => 6512,1,Answer()
exten => 6512,n,NoOp(${EXTEN})
exten => 6512,n,Goto(fax-rx,receive,1)

[fax-rx]
exten => receive,1,NoOp( FAX RECEIVE ${FROM_DID} )
exten => receive,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
exten => receive,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten => receive,n,Set(FAXFILE=fax-${FROM_DID}-${EPOCH}-rx.tif)
exten => receive,n,Set(GLOBAL(LASTFAXCALLERNUM)=${CALLERID(num)})
exten => receive,n,Set(GLOBAL(LASTFAXCALLERNAME)=${CALLERID(name)})
exten => receive,n,NoOp( SETTING FAXOPT )
exten => receive,n,Set(FAXOPT(

Re: [asterisk-users] Video conferencing?

[Dmitry Melekhov]

25.07.2012 22:24, Ken D'Ambrosio пишет:
Hi, all.  I'm 99% sure that Asterisk technically *supports* 
videoconferencing


well, confbridge supports sort of videoconferences , but our users 
refused to use them because asterisk switches video in the middle of 
stream and this leads to broken picture. developers refuse to fix this, 
i.e. add switching on i-frame ...



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