[asterisk-users] no audio using g729A for Cisco AS5300 sip peer

[Roi Stork]

Hi,

We need help in enabling g729a codec for our SIP peer that's using Cisco
AS5300.
Our codec is purchased from Digium.

We are able to dial out the numbers and answer the call, but there's no
audio. This is when only g729a is allowed.

We noticed when they also allow ulaw codec on their side, the codec used
falls back to ulaw and the problem is gone.
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Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

[Joseph]

On 01/05/12 22:12, Bruce B wrote:

but not it is not working again.
I wish they stop screwing up with that Asterisk, they keep
introducing new version and more bugs :-/

  Wish not granted !!! :-) You will be the guinea pig to new features !!!
  Same issue with A2Billing connecting to Asterisk. With older version
  this problem is not there.
  -Bruce


Solved it again :-/
This time in sip.conf the pstn line must have:
insecure=very

I was happy with Asterisk 1.4.39 till Gentoo gurus removed it from portage under the 
"vulnerability" umbrella.
I try to stay with portage when it comes to package upgrade it is easier to manage package but whey they are upgrading new packages and introducing more bugs I don't like 
it.  I have to waste my time hunting for solution and if I'll be lucky I'll find one :-/


I think I will compile version outside portage and don't have to worry about upgrades version that don't work correctly. 
The biggest problem with Linux is frequent updates and screw-ups that comes with it!


--
Joseph

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Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

[John Novack]

Joseph wrote:


but not it is not working again.
I wish they stop screwing up with that Asterisk, they keep introducing 
new version and more bugs :-/
I just keep waisting my time hunting and fixing features that stop 
working.



Easy fix.
STOP using the latest and "greatest" or even near latest.

Revert to a version that works for your specific application, and leave 
it alone.


I am in a community that uses mostly 1.4, some 1.2, even a couple that 
still use 1.0, with a very few using 1.6 or 1.8
I remember 1.4 went through many versions with certain technologies 
broken, then fixed, then broken again


John Novack


--

Dog is my Co-pilot


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Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

[Bruce B]

>
> but not it is not working again.
> I wish they stop screwing up with that Asterisk, they keep introducing new
> version and more bugs :-/
>

Wish not granted !!! :-) You will be the guinea pig to new features !!!

Same issue with A2Billing connecting to Asterisk. With older version this
problem is not there.

-Bruce
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Re: [asterisk-users] calling specific 1800-number not going through.

[Joseph]

Solved,
It seems to me the vendor is blocking the 1800 number in Western Canada.
Our second line I'm not sure where it is terminated: Toronto or USA or it could 
be they are blocking the 1800 line to home users and not for business lines.

--
Joseph

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Re: [asterisk-users] calling specific 1800-number not going through.

[Jim Dickenson]

It took 36 seconds for that number to answer when I called it and it looks like 
the call hung up after 32000 ms when you dialed via asterisk.
-- 
Jim Dickenson
mailto:dicken...@cfmc.com

CfMC
http://www.cfmc.com/



On Jan 5, 2012, at 5:45 PM, Joseph wrote:

> I have a strange problem.
> I'm using the same dialplan to call 1800-number:
> 
> [toll-free]
> ;second "7" audiocodes strips
> exten => _71800XXX,1,Dial(SIP/7${EXTEN:1}@pstn-5665,60,tr)
> 
> When I call this number (through pstn-5665) 18005000347 the phone always 
> rings busy.
> When I call any other 1800-number the calls goes through.
> 
> When I call the same phone number 18005000347 through a different line the 
> calls goes through every time.
> 
> Here is call (busy) trace to that 18005000347 with sip debug ON:
> 
> Can anybody decipher why I'm getting busy signal to that particular 
> 1800-number but not others?
> 
> 
> <--- SIP read from UDP:10.0.0.110:5060 --->
> OPTIONS sip:gateway@10.0.0.110 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1457834404
> Max-Forwards: 70
> From: ;tag=1c1457828994
> To: 
> Call-ID: 1457828497512012183855@10.0.0.110
> CSeq: 1 OPTIONS
> Contact: 
> Allow: 
> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
> User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
> Accept: application/sdp, application/simple-message-summary, message/sipfrag
> Content-Length: 0
> 
> <->
> --- (12 headers 0 lines) ---
> Looking for gateway in default (domain 10.0.0.110)
> 
> <--- Transmitting (NAT) to 10.0.0.110:5060 --->
> SIP/2.0 404 Not Found
> Via: SIP/2.0/UDP 
> 10.0.0.110;branch=z9hG4bKac1457834404;received=10.0.0.110;rport=5060
> From: ;tag=1c1457828994
> To: ;tag=as7091ae01
> Call-ID: 1457828497512012183855@10.0.0.110
> CSeq: 1 OPTIONS
> Server: Centrala
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
> PUBLISH
> Supported: replaces, timer
> Accept: application/sdp
> Content-Length: 0
> 
> 
> <>
> Scheduling destruction of SIP dialog '1457828497512012183855@10.0.0.110' in 
> 32000 ms (Method: OPTIONS)
> Reliably Transmitting (no NAT) to 81.15.150.20:5060:
> OPTIONS sip:sip.actio.pl SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK03484db5
> Max-Forwards: 70
> From: "asterisk" ;tag=as64f6417c
> To: 
> Contact: 
> Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
> CSeq: 102 OPTIONS
> User-Agent: Centrala
> Date: Fri, 06 Jan 2012 01:39:07 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
> PUBLISH
> Supported: replaces, timer
> Content-Length: 0
> 
> 
> ---
> 
> <--- SIP read from UDP:81.15.150.20:5060 --->
> SIP/2.0 501 Unsupported Method
> Via: SIP/2.0/UDP 
> 10.0.0.100:5060;branch=z9hG4bK03484db5;received=68.148.245.78;rport=48715
> To: ;tag=4fc8ac12
> From: "asterisk";tag=as64f6417c
> Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
> CSeq: 102 OPTIONS
> Content-Length: 0
> 
> <->
> --- (7 headers 0 lines) ---
> Really destroying SIP dialog 
> '66070317301f64861df62d20769ba385@10.0.0.100:5060' Method: OPTIONS
>-- Accepted AUTHENTICATED TBD call from 10.0.0.108
> 
> <--- SIP read from UDP:10.0.0.110:5060 --->
> REGISTER sip:10.0.0.100 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360
> Max-Forwards: 70
> From: ;tag=1c1472330741
> To: 
> Call-ID: 809487713120129287@10.0.0.110
> CSeq: 245 REGISTER
> Contact: ;expires=3600
> Allow: 
> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
> Expires: 3600
> User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
> Content-Length: 0
> 
> <->
> --- (12 headers 0 lines) ---
> Sending to 10.0.0.110:5060 (NAT)
> 
> <--- Transmitting (no NAT) to 10.0.0.110:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360;received=10.0.0.110
> From: ;tag=1c1472330741
> To: ;tag=as21c548bd
> Call-ID: 809487713120129287@10.0.0.110
> CSeq: 245 REGISTER
> Server: Centrala
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
> PUBLISH
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a451a5b"
> Content-Length: 0
> 
> 
> <>
> Scheduling destruction of SIP dialog '809487713120129287@10.0.0.110' in 32000 
> ms (Method: REGISTER)
> 
> <--- SIP read from UDP:10.0.0.110:5060 --->
> REGISTER sip:10.0.0.100 SIP/2.0
> Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472407428
> Max-Forwards: 70
> From: ;tag=1c1472330741
> To: 
> Call-ID: 809487713120129287@10.0.0.110
> CSeq: 246 REGISTER
> Authorization: Digest 
> username="11",realm="asterisk",nonce="3a451a5b",uri="sip:10.0.0.100",algorithm=MD5,response="5dd6df18064f3d23cb86ca306820e596"
> Contact: ;expires=3600
> Allow: 
> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
> Expires: 3600
> User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
> Content-Length: 0
> 
> <->
> --- (13 headers 0 l

[asterisk-users] calling specific 1800-number not going through.

[Joseph]

I have a strange problem.
I'm using the same dialplan to call 1800-number:

[toll-free]
;second "7" audiocodes strips
exten => _71800XXX,1,Dial(SIP/7${EXTEN:1}@pstn-5665,60,tr)

When I call this number (through pstn-5665) 18005000347 the phone always rings 
busy.
When I call any other 1800-number the calls goes through.

When I call the same phone number 18005000347 through a different line the 
calls goes through every time.

Here is call (busy) trace to that 18005000347 with sip debug ON:

Can anybody decipher why I'm getting busy signal to that particular 1800-number 
but not others?


<--- SIP read from UDP:10.0.0.110:5060 --->
OPTIONS sip:gateway@10.0.0.110 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1457834404
Max-Forwards: 70
From: ;tag=1c1457828994
To: 
Call-ID: 1457828497512012183855@10.0.0.110
CSeq: 1 OPTIONS
Contact: 
Allow: 
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<->
--- (12 headers 0 lines) ---
Looking for gateway in default (domain 10.0.0.110)

<--- Transmitting (NAT) to 10.0.0.110:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 
10.0.0.110;branch=z9hG4bKac1457834404;received=10.0.0.110;rport=5060
From: ;tag=1c1457828994
To: ;tag=as7091ae01
Call-ID: 1457828497512012183855@10.0.0.110
CSeq: 1 OPTIONS
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<>
Scheduling destruction of SIP dialog '1457828497512012183855@10.0.0.110' in 
32000 ms (Method: OPTIONS)
Reliably Transmitting (no NAT) to 81.15.150.20:5060:
OPTIONS sip:sip.actio.pl SIP/2.0
Via: SIP/2.0/UDP 10.0.0.100:5060;branch=z9hG4bK03484db5
Max-Forwards: 70
From: "asterisk" ;tag=as64f6417c
To: 
Contact: 
Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
CSeq: 102 OPTIONS
User-Agent: Centrala
Date: Fri, 06 Jan 2012 01:39:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:81.15.150.20:5060 --->
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP 
10.0.0.100:5060;branch=z9hG4bK03484db5;received=68.148.245.78;rport=48715
To: ;tag=4fc8ac12
From: "asterisk";tag=as64f6417c
Call-ID: 66070317301f64861df62d20769ba385@10.0.0.100:5060
CSeq: 102 OPTIONS
Content-Length: 0

<->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '66070317301f64861df62d20769ba385@10.0.0.100:5060' 
Method: OPTIONS
-- Accepted AUTHENTICATED TBD call from 10.0.0.108

<--- SIP read from UDP:10.0.0.110:5060 --->
REGISTER sip:10.0.0.100 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360
Max-Forwards: 70
From: ;tag=1c1472330741
To: 
Call-ID: 809487713120129287@10.0.0.110
CSeq: 245 REGISTER
Contact: ;expires=3600
Allow: 
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 3600
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Content-Length: 0

<->
--- (12 headers 0 lines) ---
Sending to 10.0.0.110:5060 (NAT)

<--- Transmitting (no NAT) to 10.0.0.110:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472335360;received=10.0.0.110
From: ;tag=1c1472330741
To: ;tag=as21c548bd
Call-ID: 809487713120129287@10.0.0.110
CSeq: 245 REGISTER
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3a451a5b"
Content-Length: 0


<>
Scheduling destruction of SIP dialog '809487713120129287@10.0.0.110' in 32000 
ms (Method: REGISTER)

<--- SIP read from UDP:10.0.0.110:5060 --->
REGISTER sip:10.0.0.100 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472407428
Max-Forwards: 70
From: ;tag=1c1472330741
To: 
Call-ID: 809487713120129287@10.0.0.110
CSeq: 246 REGISTER
Authorization: Digest 
username="11",realm="asterisk",nonce="3a451a5b",uri="sip:10.0.0.100",algorithm=MD5,response="5dd6df18064f3d23cb86ca306820e596"
Contact: ;expires=3600
Allow: 
REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Expires: 3600
User-Agent: Audiocodes-Sip-Gateway-/v.5.80A.032.003
Content-Length: 0

<->
--- (13 headers 0 lines) ---
Sending to 10.0.0.110:5060 (no NAT)

<--- Transmitting (no NAT) to 10.0.0.110:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.110;branch=z9hG4bKac1472407428;received=10.0.0.110
From: ;tag=1c1472330741
To: ;tag=as21c548bd
Call-ID: 809487713120129287@10.0.0.110
CSeq: 246 REGISTER
Server: Centrala
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, 
PUBLISH
Supported: replaces, timer
Expires: 3600
Contact: ;expires=3600
Date: Fri, 06 Jan 2012 01:39:11 GMT
Content-Length: 0


<>
Scheduling destruction of SIP dia

Re: [asterisk-users] DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0 Released

[David Backeberg]

On Wed, Jan 4, 2012 at 4:45 PM, Asterisk Development Team
 wrote:
>The Asterisk Development Team is pleased to announce the first
>release of DAHDI-Linux 2.6.0 and DAHDI-Tools 2.6.0.

>2.6.0 is a feature release which:
>
>        wct4xxp: Expose serial number in dahdi_device and kernel log.

This is VERY exciting news. Thank you very much!

So much easier than having to schedule time to open a box when you
forgot to write down the serial number when doing the rack-and-stack.

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[asterisk-users] which choice: asterisk-gui or freepbx?

[Tom Poe]

Anyone point me to discussion as to which is better choice for new 
asterisknow user?

Thanks, Tom


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Re: [asterisk-users] asterisk 1.8.8 - caller ID not working.

[Joseph]

On 01/05/12 16:42, Joseph wrote:

I just noticed after upgrade from Asterisk 1.4.39 to 1.8.8
my caller ID is not working

WARNING[1671]: chan_sip.c:13956 check_auth: username mismatch, have <11>, digest has 

NOTICE[1671]: chan_sip.c:22048 handle_request_invite: Failed to authenticate device "KMIEC 
Z" ;tag=1c976040515

--
Joseph

--


I had this problem before and was able to solve it:

---copy--
Extending delay to 6sec. didn't help.  
It is a SIP bug I believe and some temporary work around I was able to find on 
the net are:
insecure=invite   
insecure=very

insecure=port,invite

Previously I had insecure=port but it stop working; changing it back to:
insecure=invite solved the problem.
end copy--

but not it is not working again.
I wish they stop screwing up with that Asterisk, they keep introducing new 
version and more bugs :-/
I just keep waisting my time hunting and fixing features that stop working.

--
Joseph

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Re: [asterisk-users] Best non polycom SIP conference room phone

[Carlos Alvarez]

On Thu, Jan 5, 2012 at 5:10 PM, C F  wrote:

> On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman 
> wrote:
> > I am looking for a really good SIP conference room phone for use with
> > asterisk. I do not like Polycom at all.
>
> You have a really bad taste.
>

There was an interesting flamewar one day in the Asterisk IRC channel over
Polycom love/hate.  We fall into the hate category here, and hope to never
have to deal with them.  If there was an SPA-series conference phone, we'd
all rejoice.

-- 
Carlos Alvarez
TelEvolve
602-889-3003
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Re: [asterisk-users] Anyone have a reliable T.38 Solution

[David Backeberg]

On Thu, Jan 5, 2012 at 8:05 AM, Steve Underwood  wrote:
> No PAP2 or PAP2T supports T.38, even though many people will swear that they
> do. For a little while there was some beta code for the PAP2T with badly
> broken T.38 support. Perhaps this is where the "legend of T.38 on a PAP2T"
> started. Of course, on the internet, when someone posts an incorrect message
> many people would like to believe is right, a 1000 people cite it as proven
> fact.

Thanks for clearing that up. I was getting all excited that I could
flash the PAP2T; I've always used regular voice tones over SIP with
the PAP2Ts.

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Re: [asterisk-users] Best non polycom SIP conference room phone

[C F]

On Thu, Jan 5, 2012 at 12:19 PM, Bryant Zimmerman  wrote:
> I am looking for a really good SIP conference room phone for use with
> asterisk. I do not like Polycom at all.

You have a really bad taste.

> What would you all recommend? I have
> to be able to get them in the US. I found several that looked good but could
> not get them. And yes cost does matter but quality is the most important
> thing.

Then go with Polycom.



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

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[asterisk-users] asterisk 1.8.8 - caller ID not working.

[Joseph]

I just noticed after upgrade from Asterisk 1.4.39 to 1.8.8
my caller ID is not working

WARNING[1671]: chan_sip.c:13956 check_auth: username mismatch, have <11>, digest has 

NOTICE[1671]: chan_sip.c:22048 handle_request_invite: Failed to authenticate device "KMIEC 
Z" ;tag=1c976040515

--
Joseph

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Re: [asterisk-users] STOP loading extensions.ael

[Joseph]

On 01/05/12 16:52, Danny Nicholas wrote:

Modify /etc/asterisk/modules.conf with the line
Noload = pbx_ael.so

And restart asterisk
Noload should be noload - stupid Outlook



Thank you, I did:
noload => pbx_ael.so
noload => pbx_lua.so

and it is working I think.

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Re: [asterisk-users] STOP loading extensions.ael

[Danny Nicholas]

Modify /etc/asterisk/modules.conf with the line
Noload = pbx_ael.so

And restart asterisk
Noload should be noload - stupid Outlook


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Thursday, January 05, 2012 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] STOP loading extensions.ael

On Thu, 5 Jan 2012, Joseph wrote:

> How do I stop loading extensions.ael dial plan?
> I'm only using extension.conf.

1) rm /etc/asterisk/extensions.ael

2) Don't load pbx_ael.so

--
Thanks in advance,
-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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Re: [asterisk-users] STOP loading extensions.ael

[Steve Edwards]

On Thu, 5 Jan 2012, Joseph wrote:


How do I stop loading extensions.ael dial plan?
I'm only using extension.conf.


1) rm /etc/asterisk/extensions.ael

2) Don't load pbx_ael.so

--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] STOP loading extensions.ael

[Joseph]

How do I stop loading extensions.ael dial plan?
I'm only using extension.conf.

--
Joseph

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Re: [asterisk-users] Best non polycom SIP conference room phone

[Kevin P. Fleming]

On 01/05/2012 04:19 PM, Jamie A. Stapleton wrote:


http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_duo.html


This doesn't qualify as 'non-Polycom' and it appears to be more 
expensive than the SoundStation IP5000, since it is basically an IP5000 
plus analog connectivity. Although in my personal opinion, it's really 
hard to beat the IP5000.


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Re: [asterisk-users] Best non polycom SIP conference room phone

[Jamie A. Stapleton]

Some ideas:
* http://www.clearone.com/voip-conference-phones.html
* http://www.konftel.com/Products/Konftel300IP
* 
http://www.polycom.com/products/voice/conferencing_solutions/conference_phones/soundstation/soundstation_duo.html

We have tested all of these in our lab but I would prefer not to be too verbose 
about my preferences on a mailing list.

Please feel free to call me if you want more detail,
-jamie
(804) 412-1601
sip: ja...@cbsiva.com
Skype:  cbsi_jamie

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of José Pablo Méndez 
Soto
Sent: Thursday, January 05, 2012 1:06 PM
To: brya...@zktech.com; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Best non polycom SIP conference room phone

Hello Bryant,

Have you seen the snom meetingpoint?
http://www.snom.com/en/products/sip-conference-phone/snom-meetingpoint/

I don't own one, but it looks like a fine piece of hardware. And snom is 
manufacturer of supported phones for Microsoft's Lync server (must say 
something their quality right?)

http://technet.microsoft.com/en-us/lync/gg278172.aspx

Doubles Polycom price though...

 José Pablo Méndez


On Thu, Jan 5, 2012 at 11:19 AM, Bryant Zimmerman 
mailto:brya...@zktech.com>> wrote:
I am looking for a really good SIP conference room phone for use with asterisk. 
I do not like Polycom at all. What would you all recommend? I have to be able 
to get them in the US. I found several that looked good but could not get them. 
And yes cost does matter but quality is the most important thing.
Thanks

Bryant

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Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)

[Ron Bergin]

Ahmed Munir wrote:
> Hi,
>
> I installed the modules in asterisk user home directory with read and
> excitable permissions for asterisk but still my AGI not working.

IMO, it would have been better to install it in it's normal location.

Is your script using the warnings and strict pragmas?

What error message do you receive when running the script from the command
line?

Did you add the proper "use lib ''" statement to add the install
directory to the @INC array?

Ron Bergin

>
> Please provide me other advise to resolve this issue.
>
>
>> Date: Wed, 4 Jan 2012 11:30:34 -0600
>> From: "Danny Nicholas" 
>> Subject: Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus
>>(oracle)
>> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>>
>> Message-ID: <00ca01cccb06$911e8300$b35b8900$@debsinc.com>
>> Content-Type: text/plain; charset="us-ascii"
>>
>> The module probably isn't readable/executeable from Asterisk
>>
>>
>>
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed
>> Munir
>> Sent: Wednesday, January 04, 2012 10:45 AM
>> To: asterisk-users@lists.digium.com
>> Subject: [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)
>>
>>
>>
>> Hi all,
>>
>> I'm trying to run an AGI in PERL which uses the module DBD-Oracle.
>> Currently
>> my AGI is working fine in my two servers but not in my other four
>> servers.
>> When  I tried execute an AGI (as a user asterisk) in command line it
>> works
>> fine (even I also declare environmental variables in user profile and in
>> my
>> AGI), but when I tried to call my AGI (perl) in dial plan, it don't get
>> executed.
>>
>> Please advise me to resolve this issue.
>>
>> --
>> Regards,
>>
>> Ahmed Munir Chohan
>>
>>
>> -
> Regards,
>
> Ahmed Munir Chohan
> --
> _
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[asterisk-users] Blind transfers being cancelled by asterisk & hanging up on remote caller

[Douglas Mortensen]

Hello all,

I have a system running AsteriskNOW with asterisk asterisk-1.8.8.1-1_centos5 
from AsteriskNOW repository. I just changed our Polycom 335 sip.conf so that 
blindpreferred=1 (all transfers default as blind transfers). If a customer 
calls in & we answer & transfer, everything works fine. But if we call out to a 
customer & then transfer to another internal extension, that extension quickly 
rings & then the call is immediately gone & hung up. We are using Polycom 
firmware 3.3.3.

In troubleshooting this & analyzing the asterisk logs (& asterisk SIP debug), I 
am seeing a few interesting items. Any help would be appreciated.

For the sake of simplicity I am going to say that extension 20 is the original 
internal extension & extension 21 is the extension we are trying to transfer 
to. Here's what I've learned so far:


1.   Asterisk SIP debugs are clearly showing that the transfer starts & SIP 
INVITE is sent to ext 21. Ext 21 then sends a SIP message back to asterisk 
indicating that it is ringing. Asterisk then sends a SIP CANCEL to ext 21. Call 
is abruptly terminated for all parties.

2.   Asterisk logs are displaying WARNING messages during the transfer 
phase as:
WARNING[25423] chan_sip.c: Asked to transmit frame type slin, while native 
formats is 0x4 (ulaw) read/write = 0x4 (ulaw)/0x4 (ulaw)
This error repeats 10-11 times quickly (within 1 second) prior to the SIP 
CANCEL. (see sample of logs below)

3.   Other interesting lines from the asterisk full log:
VERBOSE[25592] app_dial.c: -- Connected line update to 
SIP/fpbx-1-b0c4ceff-2104 prevented.
[Jan  5 20:51:30] VERBOSE[25592] app_dial.c: -- 
SIP/fpbx-1-b0c4ceff-2104 requested special control 20, passing it to 
SIP/21-2105
[Jan  5 20:51:30] VERBOSE[25592] app_macro.c:   == Spawn extension 
(macro-dial-one, s, 42) exited non-zero on 'SIP/fpbx-1-b0c4ceff-2104' in 
macro 'dial-one'
[Jan  5 20:51:30] VERBOSE[25592] app_macro.c:   == Spawn extension 
(macro-exten-vm, s, 14) exited non-zero on 'SIP/fpbx-1-b0c4ceff-2104' in 
macro 'exten-vm'

4.   In looking for the slin codec's use in our system, it is not permitted 
in our system.

a.   We use an upstream SIP trunk provider for PSTN connectivity. That 
trunk's codes are configured as disallow=all followed by allow=ulaw&g729

b.  Our normal sip.conf settings for all internal phones are: disallow=all 
allow=g722 allow=ulaw allow=alaw allow=gsm

5.   So it would seem that slin is not even in use on our system. So why 
then does it seem like a sip device is asking to use slin? The only thing I can 
find if I grep /var/log/asterisk/full for slin is:
[Jan  5 19:50:40] VERBOSE[24133] file.c: --  
Playing '/var/spool/asterisk/voicemail/default/21/unavail.slin' (language 'en')

So it appears that our audio recordings for voicemail are in slin format?

I am not an expert with all of this, but I am doing my best to try to put the 
pieces together. So again, just to recap, the problem I need to solve is why 
are blind transfers calls getting terminated when the call being transferred 
originated as an outbound call from one of our internal Polycom phones.

Any help would be GREATLY appreciated! :)

Just a greater excerpt from the logs:


[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Goto (macro-dial-one,s,30)
[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Executing [s@macro-dial-one:30] 
Set("SIP/fpbx-1-b0c4ceff-2104", "D_OPTIONS=
tr") in new stack
[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Executing [s@macro-dial-one:31] 
ExecIf("SIP/fpbx-1-b0c4ceff-2104", "0?SIPAd
dHeader(Alert-Info: )") in new stack
[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Executing [s@macro-dial-one:32] 
ExecIf("SIP/fpbx-1-b0c4ceff-2104", "0?SIPAd
dHeader()") in new stack
[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Executing [s@macro-dial-one:33] 
ExecIf("SIP/fpbx-1-b0c4ceff-2104", "0?Set(C
HANNEL(musicclass)=)") in new stack
[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Executing [s@macro-dial-one:34] 
GosubIf("SIP/fpbx-1-b0c4ceff-2104", "0?qwai
t,1") in new stack
[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Executing [s@macro-dial-one:35] 
Set("SIP/fpbx-1-b0c4ceff-2104", "__CWIGNORE
=") in new stack
[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Executing [s@macro-dial-one:36] 
Set("SIP/fpbx-1-b0c4ceff-2104", "__KEEPCID=
TRUE") in new stack
[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Executing [s@macro-dial-one:37] 
GotoIf("SIP/fpbx-1-b0c4ceff-2104", "0?usego
to,1") in new stack
[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Executing [s@macro-dial-one:38] 
GotoIf("SIP/fpbx-1-b0c4ceff-2104", "0?godia
l") in new stack
[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Executing [s@macro-dial-one:39] 
Set("SIP/fpbx-1-b0c4ceff-2104", "CONNECTEDL
INE(name,i)=Kyle") in new stack
[Jan  5 20:51:30] VERBOSE[25592] pbx.c: -- Executing [s@macro-dial-one:40] 
Set("SIP/fpbx-1-b0c4ceff-2104", "CONNECTEDL
INE(num)=21") in

Re: [asterisk-users] question on CDR

[Danny Nicholas]

The information should be available in the ${EXTEN} variable, but you will
have to probably record it somewhere other than the default 1.4 CDR.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, January 05, 2012 12:22 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question on CDR

I used my cell to call in and create a CDR record here from asterisk 1.4.43:

"","317XXX","s","default","""GEIS JERRY "" 
<317XXX>","DAHDI/23-1","","BackGround","SM_ATTENDANT","2012-01-05
18:12:09","2012-01-05 18:12:10","2012-01-05
18:12:18",9,8,"ANSWERED","DOCUMENTATION","1325787129.624",""

I am surprised that the number DIALED or DNIS is not present in the CDR.
How do I get that as part of the CDR?

I am using a PRI so that information should be delivered and available.

Thanks,

jerry


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[asterisk-users] question on CDR

[Jerry Geis]

I used my cell to call in and create a CDR record here from asterisk 1.4.43:

"","317XXX","s","default","""GEIS JERRY "" 
<317XXX>","DAHDI/23-1","","BackGround","SM_ATTENDANT","2012-01-05 
18:12:09","2012-01-05 18:12:10","2012-01-05 
18:12:18",9,8,"ANSWERED","DOCUMENTATION","1325787129.624",""


I am surprised that the number DIALED or DNIS is not present in the CDR.
How do I get that as part of the CDR?

I am using a PRI so that information should be delivered and available.

Thanks,

jerry


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Re: [asterisk-users] Best non polycom SIP conference room phone

[Carlos Alvarez]

I guess on a similar vein, anyone have a lower-cost option instead of
Polycom?  We have customers asking for something about half the price for
small conference rooms and light usage.

2012/1/5 José Pablo Méndez Soto 

> Hello Bryant,
>
> Have you seen the snom meetingpoint?
> http://www.snom.com/en/products/sip-conference-phone/snom-meetingpoint/
>
> I don't own one, but it looks like a fine piece of hardware. And snom is
> manufacturer of supported phones for Microsoft's Lync server (must say
> something their quality right?)
>
> http://technet.microsoft.com/en-us/lync/gg278172.aspx
>
> Doubles Polycom price though...
>
>  *José Pablo Méndez
> *
>
>
> On Thu, Jan 5, 2012 at 11:19 AM, Bryant Zimmerman wrote:
>
>> I am looking for a really good SIP conference room phone for use with
>> asterisk. I do not like Polycom at all. What would you all recommend? I
>> have to be able to get them in the US. I found several that looked good but
>> could not get them. And yes cost does matter but quality is the most
>> important thing.
>>
>> Thanks
>>
>> Bryant
>>
>> --
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>>
>
>
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602-889-3003
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Re: [asterisk-users] Best non polycom SIP conference room phone

[José Pablo Méndez Soto]

Hello Bryant,

Have you seen the snom meetingpoint?
http://www.snom.com/en/products/sip-conference-phone/snom-meetingpoint/

I don't own one, but it looks like a fine piece of hardware. And snom is
manufacturer of supported phones for Microsoft's Lync server (must say
something their quality right?)

http://technet.microsoft.com/en-us/lync/gg278172.aspx

Doubles Polycom price though...

 *José Pablo Méndez
*


On Thu, Jan 5, 2012 at 11:19 AM, Bryant Zimmerman wrote:

> I am looking for a really good SIP conference room phone for use with
> asterisk. I do not like Polycom at all. What would you all recommend? I
> have to be able to get them in the US. I found several that looked good but
> could not get them. And yes cost does matter but quality is the most
> important thing.
>
> Thanks
>
> Bryant
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Where are the fax instructions?

[José Pablo Méndez Soto]

Thanks from the bottom of my heart sir!

Asterisk 1.8.7.1 is my build, I couldn't chose res_fax_spandsp from
menuselect due to dependencies, so I installed libspandsp-dev:

:~# apt-cache policy libspandsp*
libspandsp2:
  Installed: 0.0.6~pre12-1
  Candidate: 0.0.6~pre12-1
  Version table:
 *** 0.0.6~pre12-1 0

libspandsp-dev:
  Installed: 0.0.6~pre12-1
  Candidate: 0.0.6~pre12-1
  Version table:
 *** 0.0.6~pre12-1 0

So, my dialplan snippet looks good, but I need to correct the lacking
gateway capability?


 *José Pablo Méndez
*


On Thu, Jan 5, 2012 at 10:04 AM, Kevin P. Fleming wrote:

> On 01/05/2012 01:03 AM, José Pablo Méndez Soto wrote:
>
>> Hello,
>>
>> Trying to set up res_fax_spandsp. Based on
>> https://wiki.asterisk.org/**wiki/display/AST/T.38+Fax+**GatewayI
>>  wrote this
>> in my extensions.conf:
>>
>> exten => 306,1,NoOp(Fax transmission)
>> same => n,Set(FAXOPT(gateway)=yes)
>> same => n,Dial(DAHDI/3)->FXS port to fax machine
>> same => n,Hangup()
>>
>> Call flow Im trying to pull out is as follows:
>>
>> Zoiper  -->  Asterisk with TDM410 --> FXS --> Analog fax machine
>>
>> I am totally lost about the use of this new gateway module in the
>> dialplan. I think it loads ok:
>>
>> CLI> fax show capabilities
>>
>> Registered FAX Technology Modules:
>>
>> Type: Spandsp
>> Description : Spandsp FAX Driver
>> Capabilities: SEND RECEIVE T.38 G.711
>>
>
> Your res_fax_spandsp module was built without gateway support; what
> version of Asterisk are you using?
>
>
>
>> 1 registered modules
>>
>> Also I have the FFA manual, which I couldn't understand. I think FAXOPT
>> is common to both, but still not sure how to put them together. Where
>> can I find documentation about configuring the call flow described?
>>
>
> Fax for Asterisk is totally unrelated to T.38 gateway support.
>
>
>  Or some insight will also be appreciated.
>>
>> Here is my sip peer config:
>>
>> [105](headquarters) ;zoiper phone
>> type=friend
>> secret=
>> mailbox=105@default
>> t38pt_udptl = yes
>>
>> Dahdi:
>> ;FXS Modules
>> group = 2
>> signalling = fxo_ks
>> context = interno
>> channel = 3-4
>> faxdetect = both
>>
>> Finally, a verbose output:
>>
>>   == Using SIP RTP CoS mark 5
>> -- Executing [606@intern:1] NoOp("SIP/105-0002", "Fax
>> Transmission") in new stack
>> -- Executing [606@intern:2] Set("SIP/105-0002",
>> "FAXOPT(gateway)=yes") in new stack
>> [Jan  5 00:59:57] WARNING[1831]: res_fax.c:2783 acf_faxopt_write:
>> channel 'SIP/605-0002' set FAXOPT(gateway) to 'yes' is unhandled!
>> -- Executing [606@intern:3] Dial("SIP/605-0002", "DAHDI/3") in
>> new stack
>> -- Called DAHDI/3
>> -- DAHDI/3-1 is ringing
>> -- DAHDI/3-1 is ringing
>> -- DAHDI/3-1 is ringing
>> -- DAHDI/3-1 answered SIP/605-0002
>> -- Hanging up on 'DAHDI/3-1'
>> -- Hungup 'DAHDI/3-1'
>>   == Spawn extension (intern, 606, 3) exited non-zero on
>> 'SIP/105-0002'
>>
>
> Since 'fax show capabilities' did not show that you have gateway support,
> you won't be able to enable it. That problem must be corrected first.
>
> --
> 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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[asterisk-users] Best non polycom SIP conference room phone

[Bryant Zimmerman]

I am looking for a really good SIP conference room phone for use with 
asterisk. I do not like Polycom at all. What would you all recommend? I 
have to be able to get them in the US. I found several that looked good but 
could not get them. And yes cost does matter but quality is the most 
important thing.


Thanks


Bryant
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Re: [asterisk-users] asterisk 1.8 codec negotiation

[Kevin P. Fleming]

On 01/04/2012 07:06 PM, cov...@ccs.covici.com wrote:

Kevin P. Fleming  wrote:


On 01/01/2012 04:17 PM, cov...@ccs.covici.com wrote:

Hi.  I am using asterisk 1.8 and everything was working fine when I was
at svn  342661.  I then upgraded to vrsion 349339 and discovered the
following problem -- one of the end points is a freeswitch box which
offers a number of codecs, including PCMU.  However, when I tried to
make a call I got a 488 response and  a message "multiple audio streams
not supported" in the log.


"multiple audio streams" != "multiple audio codecs". For some reason
Asterisk is receiving an INVITE with an offer for more than one audio
stream (m=audio), and that is not supported.

OK, but if I have a phone or in my case a server which offers a choice
of codecs, why can't asterisk just pick the ones it has rather than
reject the call?  Is there a way to do this correctly as far as asterisk
is concerned?


Did you read the message I sent? It seems that every time you post a 
question here, you ascribe behavior to Asterisk that isn't supported by 
the logs you have posted, or really, anything at all.


Asterisk does *not* reject calls just because more codecs were offered 
than it can support (or has been configured to allow). It never has, and 
never will. It has *always* acted in exactly the fashion you have 
requested: if an offer is received that contains at least one codec that 
Asterisk can support, and has been configured to allow, then the offer 
will be accepted with all supported-and-enabled codecs.


The error message you posted above has *nothing* to do with codecs being 
offered, it is caused by multiple audio streams being offered.


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Re: [asterisk-users] Where are the fax instructions?

[Kevin P. Fleming]

On 01/05/2012 01:03 AM, José Pablo Méndez Soto wrote:

Hello,

Trying to set up res_fax_spandsp. Based on
https://wiki.asterisk.org/wiki/display/AST/T.38+Fax+Gateway I wrote this
in my extensions.conf:

exten => 306,1,NoOp(Fax transmission)
 same => n,Set(FAXOPT(gateway)=yes)
 same => n,Dial(DAHDI/3)->FXS port to fax machine
 same => n,Hangup()

Call flow Im trying to pull out is as follows:

Zoiper  -->  Asterisk with TDM410 --> FXS --> Analog fax machine

I am totally lost about the use of this new gateway module in the
dialplan. I think it loads ok:

CLI> fax show capabilities

Registered FAX Technology Modules:

Type: Spandsp
Description : Spandsp FAX Driver
Capabilities: SEND RECEIVE T.38 G.711


Your res_fax_spandsp module was built without gateway support; what 
version of Asterisk are you using?




1 registered modules

Also I have the FFA manual, which I couldn't understand. I think FAXOPT
is common to both, but still not sure how to put them together. Where
can I find documentation about configuring the call flow described?


Fax for Asterisk is totally unrelated to T.38 gateway support.


Or some insight will also be appreciated.

Here is my sip peer config:

[105](headquarters) ;zoiper phone
type=friend
secret=
mailbox=105@default
t38pt_udptl = yes

Dahdi:
;FXS Modules
group = 2
signalling = fxo_ks
context = interno
channel = 3-4
faxdetect = both

Finally, a verbose output:

   == Using SIP RTP CoS mark 5
 -- Executing [606@intern:1] NoOp("SIP/105-0002", "Fax
Transmission") in new stack
 -- Executing [606@intern:2] Set("SIP/105-0002",
"FAXOPT(gateway)=yes") in new stack
[Jan  5 00:59:57] WARNING[1831]: res_fax.c:2783 acf_faxopt_write:
channel 'SIP/605-0002' set FAXOPT(gateway) to 'yes' is unhandled!
 -- Executing [606@intern:3] Dial("SIP/605-0002", "DAHDI/3") in
new stack
 -- Called DAHDI/3
 -- DAHDI/3-1 is ringing
 -- DAHDI/3-1 is ringing
 -- DAHDI/3-1 is ringing
 -- DAHDI/3-1 answered SIP/605-0002
 -- Hanging up on 'DAHDI/3-1'
 -- Hungup 'DAHDI/3-1'
   == Spawn extension (intern, 606, 3) exited non-zero on 'SIP/105-0002'


Since 'fax show capabilities' did not show that you have gateway 
support, you won't be able to enable it. That problem must be corrected 
first.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
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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] Asterisk1.8 support video trancoding ?

[Kevin P. Fleming]

On 01/05/2012 08:47 AM, Paul Belanger wrote:

On 12-01-05 05:12 AM, Durgesh Mishra wrote:

Hi friend,

Is asterisk 1.8 support video trascoding ?


No version of asterisk supports it.


Technically, not quite correct. Asterisk *supports* video transcoding, 
it just doesn't have any implementations available (in other words, it 
is possible, but not yet available). Yes, this is an extremely pedantic 
correction, so sue me :-)


--
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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] Video trancoding not done.

[Kevin P. Fleming]

On 01/05/2012 03:35 AM, Durgesh Mishra wrote:

Hi,

I deposit video mail in H263 and now i want to retreive it with H264
codecs support softphone. I am using asterisk-1.8.7.1.


There is no way to do that at this time. Asterisk is capable of 
transcoding video streams if the appropriate codec modules are loaded, 
but no such codec modules exist as far as I know.


--
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] asterisk -> AGI (perl) -> sqlplus(oracle)

[Danny Nicholas]

Did you "install" DBD::Oracle or is it just in your /home/asterisk
directory?  Put up the part of the PERL module that has your include
statements.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Thursday, January 05, 2012 8:54 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)

 

The thing is, my AGI is working fine if I don't include DBD::Oracle library
in my script. If I include DBD::Oracle library my AGI script gets aborted. I
installed DBD::Oracle module in asterisk application home directory as its'
permissions are listed below;

[asterisk@klpi062 ~]$ ls -lh
/home/asterisk/perl-lib/lib/lib64/perl5/site_perl/5.8.8/x86_64-linux-thread-
multi/
total 40K
drwxr-xr-x 3 asterisk asterisk 4.0K Jan  3 14:16 auto
drwxr-xr-x 3 asterisk asterisk 4.0K Jan  3 14:16 DBD
-rwxr-xr-x 1 asterisk asterisk 1.3K Aug 26 14:09 oraperl.ph
-rwxr-xr-x 1 asterisk asterisk  28K Oct 12 12:43 Oraperl.pm

I also included the library path for locating DBD::Oracle module in my AGI.
But still unable to understand even though asterisk has permissions to
access DBD module but still AGI don't work when I include DBD library in it.

 

Date: Wed, 4 Jan 2012 12:18:24 -0600
From: "Danny Nicholas" 
Subject: Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus
   (oracle)
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
   
Message-ID: <010101cccb0d$3fa6a140$bef3e3c0$@debsinc.com>
Content-Type: text/plain; charset="us-ascii"

What are the permissions on the module you are trying to run? (ls -l
/var/lib/asterisk/agi-bin/module)



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Wednesday, January 04, 2012 12:15 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)



Hi,

I installed the modules in asterisk user home directory with read and
excitable permissions for asterisk but still my AGI not working.

Please provide me other advise to resolve this issue.


Date: Wed, 4 Jan 2012 11:30:34 -0600
From: "Danny Nicholas" 
Subject: Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus
  (oracle)
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
  
Message-ID: <00ca01cccb06$911e8300$b35b8900$@debsinc.com>
Content-Type: text/plain; charset="us-ascii"

The module probably isn't readable/executeable from Asterisk



From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
Sent: Wednesday, January 04, 2012 10:45 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)



Hi all,

I'm trying to run an AGI in PERL which uses the module DBD-Oracle. Currently
my AGI is working fine in my two servers but not in my other four servers.
When  I tried execute an AGI (as a user asterisk) in command line it works
fine (even I also declare environmental variables in user profile and in my
AGI), but when I tried to call my AGI (perl) in dial plan, it don't get
executed.

Please advise me to resolve this issue.

--
Regards,

Ahmed Munir Chohan





-- 
Regards,

Ahmed Munir Chohan



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Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus(oracle)

[Ahmed Munir]

The thing is, my AGI is working fine if I don't include DBD::Oracle library
in my script. If I include DBD::Oracle library my AGI script gets aborted.
I installed DBD::Oracle module in asterisk application home directory as
its' permissions are listed below;

[asterisk@klpi062 ~]$ ls -lh
/home/asterisk/perl-lib/lib/lib64/perl5/site_perl/5.8.8/x86_64-linux-thread-multi/
total 40K
drwxr-xr-x 3 asterisk asterisk 4.0K Jan  3 14:16 auto
drwxr-xr-x 3 asterisk asterisk 4.0K Jan  3 14:16 DBD
-rwxr-xr-x 1 asterisk asterisk 1.3K Aug 26 14:09 oraperl.ph
-rwxr-xr-x 1 asterisk asterisk  28K Oct 12 12:43 Oraperl.pm

I also included the library path for locating DBD::Oracle module in my AGI.
But still unable to understand even though asterisk has permissions to
access DBD module but still AGI don't work when I include DBD library in it.



> Date: Wed, 4 Jan 2012 12:18:24 -0600
> From: "Danny Nicholas" 
> Subject: Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus
>(oracle)
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>
> Message-ID: <010101cccb0d$3fa6a140$bef3e3c0$@debsinc.com>
> Content-Type: text/plain; charset="us-ascii"
>
> What are the permissions on the module you are trying to run? (ls -l
> /var/lib/asterisk/agi-bin/module)
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
> Sent: Wednesday, January 04, 2012 12:15 PM
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)
>
>
>
> Hi,
>
> I installed the modules in asterisk user home directory with read and
> excitable permissions for asterisk but still my AGI not working.
>
> Please provide me other advise to resolve this issue.
>
>
> Date: Wed, 4 Jan 2012 11:30:34 -0600
> From: "Danny Nicholas" 
> Subject: Re: [asterisk-users] asterisk -> AGI (perl) -> sqlplus
>   (oracle)
> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
>   
> Message-ID: <00ca01cccb06$911e8300$b35b8900$@debsinc.com>
> Content-Type: text/plain; charset="us-ascii"
>
> The module probably isn't readable/executeable from Asterisk
>
>
>
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ahmed Munir
> Sent: Wednesday, January 04, 2012 10:45 AM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] asterisk -> AGI (perl) -> sqlplus (oracle)
>
>
>
> Hi all,
>
> I'm trying to run an AGI in PERL which uses the module DBD-Oracle.
> Currently
> my AGI is working fine in my two servers but not in my other four servers.
> When  I tried execute an AGI (as a user asterisk) in command line it works
> fine (even I also declare environmental variables in user profile and in my
> AGI), but when I tried to call my AGI (perl) in dial plan, it don't get
> executed.
>
> Please advise me to resolve this issue.
>
> --
> Regards,
>
> Ahmed Munir Chohan
>
>
>
>
> --
Regards,

Ahmed Munir Chohan
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Re: [asterisk-users] Asterisk1.8 support video trancoding ?

[Paul Belanger]

On 12-01-05 05:12 AM, Durgesh Mishra wrote:

Hi friend,

Is asterisk 1.8  support video trascoding ?


No version of asterisk supports it.

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Re: [asterisk-users] Anyone have a reliable T.38 Solution

[Steve Underwood]

On 01/05/2012 07:45 PM, Michael Keuter wrote:

Am 05.01.2012 um 04:55 schrieb Matt Darnell:


On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn
  wrote:

I'm using  the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and 
rx_fax on multiple installations with no problems.

David,

Are you running 10.0 or 1.8?

Glad to know that the PAP2T has a solid T.38 implementation!

-Matt

There seem to be at least 2 versions of the PAP2T. The one I have (in Germany) 
does NOT support T.38.

Michael

http://www.mksolutions.info
No PAP2 or PAP2T supports T.38, even though many people will swear that 
they do. For a little while there was some beta code for the PAP2T with 
badly broken T.38 support. Perhaps this is where the "legend of T.38 on 
a PAP2T" started. Of course, on the internet, when someone posts an 
incorrect message many people would like to believe is right, a 1000 
people cite it as proven fact.


Steve



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Re: [asterisk-users] Anyone have a reliable T.38 Solution

[Michael Keuter]

Am 05.01.2012 um 04:55 schrieb Matt Darnell:

> On Wed, Jan 4, 2012 at 1:02 AM, David Klaverstyn
>  wrote:
>> I'm using  the Linksys PAP2T and the Grandstream with SpanDSP and tx_fax and 
>> rx_fax on multiple installations with no problems.
> 
> David,
> 
> Are you running 10.0 or 1.8?
> 
> Glad to know that the PAP2T has a solid T.38 implementation!
> 
> -Matt

There seem to be at least 2 versions of the PAP2T. The one I have (in Germany) 
does NOT support T.38.

Michael

http://www.mksolutions.info





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[asterisk-users] Asterisk1.8 support video trancoding ?

[Durgesh Mishra]

Hi friend, 

Is asterisk 1.8  support video trascoding ? 



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Re: [asterisk-users] Anyone have a reliable T.38 Solution

[Frank Sautter]

On 04.01.2012 07:25, Matt Darnell wrote:
> We are looking to roll a solution that will have the following network layout:
> 
> ISDN-PRI <--> Asterisk <--> T.38 <--> ATA <--> Fax
> 
> Does version 1.8 with the Digium fax driver have this capability?  I
> like 1.8 because it is a long term support version.
> 
> What ATA's are people using?
> 
> Any working solutions would be great!

we are using such a setup:


Telco <-PRI-> bero*fixBox <-BRI-> Fritz!BoxFon <-analog-> FaxMachine
  |
  +--<-SIP-> Asterisk <-SIP-> Phones
  |
  +-- t38modem <-tty-> Hylafax


The bero*fixBox http://www.beronet.com/product/berofix-gateways/ is has
one ISDN-PRI and 4 ISDN-BRI ports. The PRI-port is connected to the
telco, one of the ISDN-BRI ports is connected to an old Fritz!Box Fon
http://www.avm.de/en/Produkte/FRITZBox/FRITZ_Box_Fon/index.php which has
3 (newer ones only 2) analog ports.

The calls are routed by the bero*fixBox to either the asterisk server or
the hylafax server based on the DID number.
The bero*fix box has ~30 softmodems that translate to T.38-SIP which is
then connected to the t38modem on the hylafax server which handles
around 70,000 faxpages a month.
Outgoing (incoming only for test purposes) faxes from our analog fax
machines are routed by our asterisk server back to the bero*fix box so
we achive bit-slip-free (and therefore reliable) connection.

Beronet also has PCI- and PCIe-cards to be directly installed in your
asterisk server.

Frank

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[asterisk-users] Video trancoding not done.

[Durgesh Mishra]

Hi, 

I deposit video mail in H263 and now i want to retreive it with H264 codecs 
support softphone.  I am using asterisk-1.8.7.1. 

How i can do it? 



Thanks&Regards 

Durgesh Mishra--
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