Re: [asterisk-users] Asterisk 11- Answer with [m=image 0 udptl t38] and Call Drop

2013-01-16 Thread Salman Zafar 
Thanks Jordan, for having a look at this matter.

Yes, that is what Asterisk 11 is sending. Here are complete sip debugs from
Asterisk attached. Please refer to IP mapping from OP to have a better
understanding.

Is there any way of getting it off from SIP parser on compile time as I am
not using this feature and do not intend to use in future.



On Wed, Jan 16, 2013 at 7:01 PM, Matthew Jordan  wrote:

> On 01/16/2013 07:28 AM, Salman Zafar wrote:
> > Hello All,
> >I am having a bit peculiar problem with Asterisk 11 for a
> > carrier. This carrier shares quite some information in SDP header, which
> > should not be the problem, however what happen is as follow:
> >
> >
> > Carrier> (INVITE) -> *SIP Proxy -> Asterisk 11 -> Answer()* -> right
> > after answering call drops... Carrier send a BYE with (cause 79: service
> > or option not implemented).
> >
> > *NOTE: Please refer to complete SIP traces attached. *
> > *
> > *
> > *Also Note:*
> > _Carrier_: 62.61.147.214
> > _Proxy_: 77.X.X.X:5060
> > _Asterisk11_: 77.X.X.X:5080
> >
> > *_Here is Invite SDP  from Carrier -> Proxy -> Asterisk 11_*
> >
> > INVITE sip:69609000@77.X.X.X SIP/2.0
> > v=0
> > o=AudiocodesGW 1638819008 1638818710 IN IP4 62.61.147.214
> > s=Phone-Call
> > c=IN IP4 77.X.X.X
> > t=0 0
> > m=audio 53372 RTP/AVP 8 118 18
> > a=rtpmap:8 PCMA/8000
> > a=rtpmap:118 PCMA/8000
> > a=gpmd:118 vbd=yes
> > a=rtpmap:18 G729/8000
> > a=fmtp:18 annexb=no
> > a=ptime:20
> > a=sendrecv
> > a=rtcp:53373 IN IP4 77.X.X.X
> > m=image 56854 udptl t38
> > a=T38FaxVersion:0
> > a=T38MaxBitRate:14400
> > a=T38FaxMaxBuffer:1024
> > a=T38FaxMaxDatagram:122
> > a=T38FaxRateManagement:transferredTCF
> > a=T38FaxUdpEC:t38UDPRedundancy
> >
> > /*_SDP:After Answered by Asterisk 11_*/
> > v=0
> > o=root 164966782 164966782 IN IP4 77.X.X.X
> > s=Asterisk v11.0.1
> > c=IN IP4 77.X.X.X
> > t=0 0
> > m=audio 12636 RTP/AVP 18 8
> > a=rtpmap:18 G729/8000
> > a=fmtp:18 annexb=no
> > a=rtpmap:8 PCMA/8000
> > a=ptime:20
> > a=sendrecv
> > *_m=image 0 udptl t38_*
>
>
> The appropriate way for Asterisk to indicate that it does not support a
> media stream is to set the port number to 0. We have to inform the
> offerer that we don't support the media stream; removing it from the SDP
> completely is not allowed.
>
> Per RFC 3264, section 6:
>
> "   An offered stream MAY be rejected in the answer, for any reason.  If
>a stream is rejected, the offerer and answerer MUST NOT generate
>media (or RTCP packets) for that stream.  To reject an offered
>stream, the port number in the corresponding stream in the answer
>MUST be set to zero. "
>
> > I have tired by disabling/unloading fax modules as *I am not using* them
> > but no results. Secondly, also tried tweaking of udptl ever-odd nothing
> > worked.
>
> You've configured your system to not support fax correctly. Asterisk is
> rejecting the offered image stream accordingly.
>
> > The same carrier works for Asterisk 1.6.X and the only difference I have
> > notice so far is the above underlined line in Answered SDP -> m=image 0
> > udptl t38. I think if I some how do not advertise udptl here i would be
> > able to avoid this scenario. I have tried multiple ways to strip off SDP
> > from incoming INVITE at SIP Proxy level but it is not SDP wise enough.
> >
>
> I'm not sure what 1.6.x is sending. It's possible that it just
> completely removed the stream from the SDP answer, which is wrong.
>
> Section 6 again:
>
> "For each "m=" line in the offer, there MUST be a corresponding "m="
>line in the answer."
>
> > *Note:*
> >
> > In Asterisk 1.6 =>  WARNING[32671]: chan_sip.c:8833 process_sdp:
> > Unsupported SDP media type in offer: image 59978 udptl t38
> > In Asterisk 11 => WARNING[18748][C-002f]: chan_sip.c:10277
> > process_sdp: Failed to initialize UDPTL, declining image stream
> >
> >
>
> An initial glance at this makes me think your carrier is doing something
> wrong. Just to check, however, is the SDP answer you pasted the entire
> SDP that Asterisk 11 responds with? Specifically, are there no format
> attributes for the image stream in the SDP that Asterisk responds with?
>
> --
> Matthew Jordan
> Digium, Inc. | Engineering Manager
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at: http://digium.com & http://asterisk.org
>
>
>
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-- 
Regards

**
Muhammad Salman
***
; *** INVITE sip:69609000@77.X.X.X 
SIP/2.0

Record-Route: 
Via: SIP/2.0/UDP 77.X.X.X;branch=z9hG4bKbf17.c3dd9193.0

INVI

Re: [asterisk-users] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable

2013-01-16 Thread Pete Mundy 
On 17/01/2013, at 4:35 AM, A J Stiles  wrote:

> Unplug "10.3.22.6", and try pinging it.  If something answers, then you 
> indeed 
> have a clash.  Check your DHCP server configuration, and make sure any 
> manually-assigned addresses are outside its pool of addresses.

If you do this test, remember to make sure to keep pinging with the host 
disconnected for minimum 30 seconds so as to give your local OS's arp table a 
chance to time out (or manually delete the original ARP entry before starting 
the ping).

Pete




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Description: S/MIME cryptographic signature
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Re: [asterisk-users] special conference room

2013-01-16 Thread Danny Nicholas 
You could set up the caller meetme where the user presses 1 to 

Exit the conference

Whisper to the moderator

Rejoin the conference

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Wednesday, January 16, 2013 4:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] special conference room

 

ok,

now i have got some very valuable information to start off with. thank you
all.
i´ll be back to report success or further questions...

just one thing, that i think might be a showstopper that i may have not
explained clear enough...:
muting and unmuting a caller should have the effect, that the caller can
talk 
to the moderator or not... any caller should NEVER hear what other callers
are talking... may he be muted or not...

yves

Am 16.01.2013 23:01, schrieb Danny Nicholas:

>From what I read, neither confbridge or meetme have the whisper feature
built-in;  This doesn’t matter because the moderator would have to use
meetmeadmin or the confbridge equivalent to control the other functions.
The moderator would either need two phones or a phone and a web interface.
Let’s say Yves’ “special conference” is .  The moderator would start
using this command

Exten => s,1,meetme()

The participants would do

Exten => s,1,meetme(,m) – muted so they can listen but not talk

- there is one admin / moderator and several "normal" callers.
- the callers must not hear any other caller, only the moderator

 

The moderator would need to be able to enumerate the conference by doing

Asterisk –rx “core show channels verbose”|grep meetme

This is supposed to be doable from the dialplan but my google-fu failed me
on it.
- the moderator must be able to mute and unmute any caller at any time

 

Establish a maximum number of users and set this up for each one

Exten => 99,1,meetmeadmin(,M,1) let user 1 talk

Exten => 199,1,meetmeadmin(,m,1) turn user 1 back off
- the moderator must be able to talk to all callers or to a specific caller.

Exten => 901,1,chanspy(SIP/XXX,w)
- the modetator must be able to kick off any caller at any time...

Exten => 299,1,meetmeadmin(,k,1) kick out user 1

Exten => 666,1,meetmeadmin(,K) shut it down




 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 16, 2013 3:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] special conference room

 

Sounds like a conference with all attendees permanently muted  (except the
“moderator”).

 

The moderator uses “whisper” to communicate with individuals.

--Don

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Wednesday, January 16, 2013 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] special conference room

 

barat and danny,

thank you for your input...
I am using asterisk 11.2 and i read about meetme. Yes, it has many switches
and options and
can help me a lot... but as you already said... "does _almost_ all
features..."... unfortunately I
need ALL the constraints fulfilled... therefore i admit I have not tried it
in deep, because just
from reading the doc I realized, that it wont fit all my needs...
btw.: I understood the "mute" switch to disable the callers to talk to the
conference.. (so to say
it mutes the callers microphone, not his earphones am I wrong? 
nevertheless... any more hints for my original feature-request?

thank you all,
yves


Am 16.01.2013 19:03, schrieb Bharat Lalcheta:

Please study meetme application's options. You will get almost all feature
you ask for in it

On Jan 16, 2013 5:37 AM, "Yves A."  wrote:

Hi list,

I am in need of a "special" asterisk conference room with the following
constraints:

- there is one admin / moderator and several "normal" callers.
- the callers must not hear any other caller, only the moderator
- the moderator must be able to mute and unmute any caller at any time
- the moderator must be able to talk to all callers or to a specific caller.
- the modetator must be able to kick off any caller at any time...

Any hints on how to realize that are highly appreciated..

Thanx in advance,
yves


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Re: [asterisk-users] special conference room

2013-01-16 Thread Yves A. 

ok,

now i have got some very valuable information to start off with. thank 
you all.

i´ll be back to report success or further questions...

just one thing, that i think might be a showstopper that i may have not
explained clear enough...:
muting and unmuting a caller should have the effect, that the caller can 
talk

to the moderator or not... any caller should NEVER hear what other callers
are talking... may he be muted or not...

yves

Am 16.01.2013 23:01, schrieb Danny Nicholas:


From what I read, neither confbridge or meetme have the whisper 
feature built-in;  This doesn't matter because the moderator would 
have to use meetmeadmin or the confbridge equivalent to control the 
other functions.  The moderator would either need two phones or a 
phone and a web interface.  Let's say Yves' "special conference" is 
.  The moderator would start using this command


Exten => s,1,meetme()

The participants would do

Exten => s,1,meetme(,m) -- muted so they can listen but not talk

- there is one admin / moderator and several "normal" callers.
- the callers must not hear any other caller, only the moderator

The moderator would need to be able to enumerate the conference by doing

Asterisk --rx "core show channels verbose"|grep meetme

This is supposed to be doable from the dialplan but my google-fu 
failed me on it.

- the moderator must be able to mute and unmute any caller at any time

Establish a maximum number of users and set this up for each one

Exten => 99,1,meetmeadmin(,M,1) let user 1 talk

Exten => 199,1,meetmeadmin(,m,1) turn user 1 back off
- the moderator must be able to talk to all callers or to a specific 
caller.


Exten => 901,1,chanspy(SIP/XXX,w)
- the modetator must be able to kick off any caller at any time...

Exten => 299,1,meetmeadmin(,k,1) kick out user 1

Exten => 666,1,meetmeadmin(,K) shut it down

*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Don Kelly

*Sent:* Wednesday, January 16, 2013 3:34 PM
*To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
*Subject:* Re: [asterisk-users] special conference room

Sounds like a conference with all attendees permanently muted  (except 
the "moderator").


The moderator uses "whisper" to communicate with individuals.

--Don

*From:*asterisk-users-boun...@lists.digium.com 
 
[mailto:asterisk-users-boun...@lists.digium.com] 
 *On Behalf 
Of *Yves A.

*Sent:* Wednesday, January 16, 2013 3:11 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] special conference room

barat and danny,

thank you for your input...
I am using asterisk 11.2 and i read about meetme. Yes, it has many 
switches and options and
can help me a lot... but as you already said... "does _almost_ all 
features..."... unfortunately I
need ALL the constraints fulfilled... therefore i admit I have not 
tried it in deep, because just

from reading the doc I realized, that it wont fit all my needs...
btw.: I understood the "mute" switch to disable the callers to talk to 
the conference.. (so to say

it mutes the callers microphone, not his earphones am I wrong?
nevertheless... any more hints for my original feature-request?

thank you all,
yves


Am 16.01.2013 19:03, schrieb Bharat Lalcheta:

Please study meetme application's options. You will get almost all
feature you ask for in it

On Jan 16, 2013 5:37 AM, "Yves A." mailto:yves...@gmx.de>> wrote:

Hi list,

I am in need of a "special" asterisk conference room with the
following constraints:

- there is one admin / moderator and several "normal" callers.
- the callers must not hear any other caller, only the moderator
- the moderator must be able to mute and unmute any caller at any time
- the moderator must be able to talk to all callers or to a
specific caller.
- the modetator must be able to kick off any caller at any time...

Any hints on how to realize that are highly appreciated..

Thanx in advance,
yves


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Re: [asterisk-users] special conference room

2013-01-16 Thread Danny Nicholas 
>From what I read, neither confbridge or meetme have the whisper feature
built-in;  This doesn't matter because the moderator would have to use
meetmeadmin or the confbridge equivalent to control the other functions.
The moderator would either need two phones or a phone and a web interface.
Let's say Yves' "special conference" is .  The moderator would start
using this command

Exten => s,1,meetme()

The participants would do

Exten => s,1,meetme(,m) - muted so they can listen but not talk

- there is one admin / moderator and several "normal" callers.
- the callers must not hear any other caller, only the moderator

 

The moderator would need to be able to enumerate the conference by doing

Asterisk -rx "core show channels verbose"|grep meetme

This is supposed to be doable from the dialplan but my google-fu failed me
on it.
- the moderator must be able to mute and unmute any caller at any time

 

Establish a maximum number of users and set this up for each one

Exten => 99,1,meetmeadmin(,M,1) let user 1 talk

Exten => 199,1,meetmeadmin(,m,1) turn user 1 back off
- the moderator must be able to talk to all callers or to a specific caller.

Exten => 901,1,chanspy(SIP/XXX,w)
- the modetator must be able to kick off any caller at any time...

Exten => 299,1,meetmeadmin(,k,1) kick out user 1

Exten => 666,1,meetmeadmin(,K) shut it down



 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Don Kelly
Sent: Wednesday, January 16, 2013 3:34 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] special conference room

 

Sounds like a conference with all attendees permanently muted  (except the
"moderator").

 

The moderator uses "whisper" to communicate with individuals.

--Don

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Wednesday, January 16, 2013 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] special conference room

 

barat and danny,

thank you for your input...
I am using asterisk 11.2 and i read about meetme. Yes, it has many switches
and options and
can help me a lot... but as you already said... "does _almost_ all
features..."... unfortunately I
need ALL the constraints fulfilled... therefore i admit I have not tried it
in deep, because just
from reading the doc I realized, that it wont fit all my needs...
btw.: I understood the "mute" switch to disable the callers to talk to the
conference.. (so to say
it mutes the callers microphone, not his earphones am I wrong? 
nevertheless... any more hints for my original feature-request?

thank you all,
yves


Am 16.01.2013 19:03, schrieb Bharat Lalcheta:

Please study meetme application's options. You will get almost all feature
you ask for in it

On Jan 16, 2013 5:37 AM, "Yves A."  wrote:

Hi list,

I am in need of a "special" asterisk conference room with the following
constraints:

- there is one admin / moderator and several "normal" callers.
- the callers must not hear any other caller, only the moderator
- the moderator must be able to mute and unmute any caller at any time
- the moderator must be able to talk to all callers or to a specific caller.
- the modetator must be able to kick off any caller at any time...

Any hints on how to realize that are highly appreciated..

Thanx in advance,
yves


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Re: [asterisk-users] N Priority in Mysql

2013-01-16 Thread Danny Nicholas 
"n" priority is a runtime value set by the dialplan.  To use it in a
database, you would have to update the database using something like
"dialplan show context".

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Roy Abshire
Sent: Wednesday, January 16, 2013 3:44 PM
To: Asterisk Users
Subject: [asterisk-users] N Priority in Mysql

Why doesn't the "n" priority work in a mysql database??
This way I don't have to re-number everything when I insert a new line...

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[asterisk-users] N Priority in Mysql

2013-01-16 Thread Roy Abshire 

Why doesn't the "n" priority work in a mysql database??
This way I don't have to re-number everything when I insert a new line...

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Re: [asterisk-users] special conference room

2013-01-16 Thread Don Kelly 
Sounds like a conference with all attendees permanently muted  (except the
"moderator").

 

The moderator uses "whisper" to communicate with individuals.

--Don

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Wednesday, January 16, 2013 3:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] special conference room

 

barat and danny,

thank you for your input...
I am using asterisk 11.2 and i read about meetme. Yes, it has many switches
and options and
can help me a lot... but as you already said... "does _almost_ all
features..."... unfortunately I
need ALL the constraints fulfilled... therefore i admit I have not tried it
in deep, because just
from reading the doc I realized, that it wont fit all my needs...
btw.: I understood the "mute" switch to disable the callers to talk to the
conference.. (so to say
it mutes the callers microphone, not his earphones am I wrong? 
nevertheless... any more hints for my original feature-request?

thank you all,
yves


Am 16.01.2013 19:03, schrieb Bharat Lalcheta:

Please study meetme application's options. You will get almost all feature
you ask for in it

On Jan 16, 2013 5:37 AM, "Yves A."  wrote:

Hi list,

I am in need of a "special" asterisk conference room with the following
constraints:

- there is one admin / moderator and several "normal" callers.
- the callers must not hear any other caller, only the moderator
- the moderator must be able to mute and unmute any caller at any time
- the moderator must be able to talk to all callers or to a specific caller.
- the modetator must be able to kick off any caller at any time...

Any hints on how to realize that are highly appreciated..

Thanx in advance,
yves


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Re: [asterisk-users] special conference room

2013-01-16 Thread Johan Wilfer 
2013-01-16 22:10, Yves A. skrev:
> barat and danny,
> 
> thank you for your input...
> I am using asterisk 11.2 and i read about meetme. Yes, it has many
> switches and options and
> can help me a lot... but as you already said... "does _almost_ all
> features..."... unfortunately I
> need ALL the constraints fulfilled... therefore i admit I have not tried
> it in deep, because just
> from reading the doc I realized, that it wont fit all my needs...
> btw.: I understood the "mute" switch to disable the callers to talk to
> the conference.. (so to say
> it mutes the callers microphone, not his earphones am I wrong?
> nevertheless... any more hints for my original feature-request?
> 
> thank you all,
> yves
> 
>> I am in need of a "special" asterisk conference room with the
>> following constraints:
>>
>> - there is one admin / moderator and several "normal" callers.
>> - the callers must not hear any other caller, only the moderator
>> - the moderator must be able to mute and unmute any caller at any time
>> - the moderator must be able to talk to all callers or to a
>> specific caller.
>> - the modetator must be able to kick off any caller at any time...
>>
>> Any hints on how to realize that are highly appreciated..
>>

You can do all this with Meetme, ChanSpy and ChannelRedirect. You could
also use the AMI variants of the above commands. Maybe you could use
Confbridge that is intended to replace Meetme.

So the simple answer is yes, this can be done.


-- 
Johan Wilfer


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Re: [asterisk-users] special conference room

2013-01-16 Thread Yves A. 

barat and danny,

thank you for your input...
I am using asterisk 11.2 and i read about meetme. Yes, it has many 
switches and options and
can help me a lot... but as you already said... "does _almost_ all 
features..."... unfortunately I
need ALL the constraints fulfilled... therefore i admit I have not tried 
it in deep, because just

from reading the doc I realized, that it wont fit all my needs...
btw.: I understood the "mute" switch to disable the callers to talk to 
the conference.. (so to say

it mutes the callers microphone, not his earphones am I wrong?
nevertheless... any more hints for my original feature-request?

thank you all,
yves


Am 16.01.2013 19:03, schrieb Bharat Lalcheta:


Please study meetme application's options. You will get almost all 
feature you ask for in it


On Jan 16, 2013 5:37 AM, "Yves A." > wrote:


Hi list,

I am in need of a "special" asterisk conference room with the
following constraints:

- there is one admin / moderator and several "normal" callers.
- the callers must not hear any other caller, only the moderator
- the moderator must be able to mute and unmute any caller at any time
- the moderator must be able to talk to all callers or to a
specific caller.
- the modetator must be able to kick off any caller at any time...

Any hints on how to realize that are highly appreciated..

Thanx in advance,
yves


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Re: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start

2013-01-16 Thread Warren Selby 
On Wed, Jan 16, 2013 at 1:37 PM, Danny Nicholas  wrote:

> Same issue exists with 11.2
>
>
I've created issue 20945 to track this, at least for 1.8.20.0.

https://issues.asterisk.org/jira/browse/ASTERISK-20945

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Re: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start

2013-01-16 Thread Danny Nicholas 
Same issue exists with 11.2

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bakko
Sent: Wednesday, January 16, 2013 1:27 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to
connect to remote asterisk message on service asterisk start

me too.

regards

El 16/01/2013 13:25, Eric Wieling escribió:
> I am also experiencing this issue.  Asterisk is in fact running, you can
verify by running "asterisk -rvvv" (-r connects to an EXISTING asterisk
process) or using ps.
>
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren 
> Selby
> Sent: Wednesday, January 16, 2013 1:21 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to 
> connect to remote asterisk message on service asterisk start
>
> I'm trying to decide if I need to open an issue for this or if it's just a
misconfiguration issue of some sort.  Here's the situation - yesterday
morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS
5.8 installation and got a shell of a basic asterisk install setup (minimum
required configuration files, etc, with no dialplan or sip peers setup yet).
In the afternoon, I got the notification that asterisk 1.8.20.0 had been
released, so today, I downloaded the latest 1.8-current.tar.gz and compiled
and installed it (./configure, make menuselect and choose all the same
options as my previous install, make, make install).
>
>
> Now, when I start the asterisk service using "service asterisk start" from
the command line, this is the output:
>
> [root@pbx ~]# service asterisk start
> Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?) Starting asterisk:
>
>
> However, the /var/run/asterisk/asterisk.ctl file is being created and the
process is starting:
>
> [root@pbx ~]# ls -lh /var/run/asterisk/ total 4.0K srwxr-xr-x 1 root 
> root 0 Jan 16 12:07 asterisk.ctl
> -rw-r--r-- 1 root root 6 Jan 16 12:07 asterisk.pid
>
>
> However, I'm no longer getting the usual splash message when I connect to
the asterisk console...this is what I get:
>
> [root@pbx ~]# asterisk -r
> Verbosity is at least 3
> pbx*CLI>
>
>
> I don't have any peers setup yet, or even any dialplan configured to test,
but when I go through the logs, I don't find any errors or warnings that I'm
not expecting.
>
>
> I've gone back to the asterisk 1.8.19.1 install and everything works as
expected (no error messages, full splash about license / version on
connection to console, etc).  I performed make clean in my 1.8.20 source
directory, then ./configure, make menuselect, make, make install, and even
make config, and I'm still seeing this message pop up when restarting /
starting the service.
>
>
> I went through the CHANGELOG.TXT for 1.8.20.0 and it appears there are
some items talking about changing the way the process starts up (commit
r376428), but I'm not enough of a coder to understand if those would cause
what I'm seeing.
>
>
> Is anyone else seeing this issue?  Should I open an issue on the tracker?
Anyone see something obvious I missed?
>
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>


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Re: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start

2013-01-16 Thread Bakko 

me too.

regards

El 16/01/2013 13:25, Eric Wieling escribió:

I am also experiencing this issue.  Asterisk is in fact running, you can verify by 
running "asterisk -rvvv" (-r connects to an EXISTING asterisk process) or using 
ps.



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Wednesday, January 16, 2013 1:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to 
remote asterisk message on service asterisk start

I'm trying to decide if I need to open an issue for this or if it's just a 
misconfiguration issue of some sort.  Here's the situation - yesterday morning, 
I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 
installation and got a shell of a basic asterisk install setup (minimum 
required configuration files, etc, with no dialplan or sip peers setup yet).  
In the afternoon, I got the notification that asterisk 1.8.20.0 had been 
released, so today, I downloaded the latest 1.8-current.tar.gz and compiled and 
installed it (./configure, make menuselect and choose all the same options as 
my previous install, make, make install).


Now, when I start the asterisk service using "service asterisk start" from the 
command line, this is the output:

[root@pbx ~]# service asterisk start
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl 
exist?) Starting asterisk:


However, the /var/run/asterisk/asterisk.ctl file is being created and the 
process is starting:

[root@pbx ~]# ls -lh /var/run/asterisk/
total 4.0K
srwxr-xr-x 1 root root 0 Jan 16 12:07 asterisk.ctl
-rw-r--r-- 1 root root 6 Jan 16 12:07 asterisk.pid


However, I'm no longer getting the usual splash message when I connect to the 
asterisk console...this is what I get:

[root@pbx ~]# asterisk -r
Verbosity is at least 3
pbx*CLI>


I don't have any peers setup yet, or even any dialplan configured to test, but 
when I go through the logs, I don't find any errors or warnings that I'm not 
expecting.


I've gone back to the asterisk 1.8.19.1 install and everything works as 
expected (no error messages, full splash about license / version on connection 
to console, etc).  I performed make clean in my 1.8.20 source directory, then 
./configure, make menuselect, make, make install, and even make config, and I'm 
still seeing this message pop up when restarting / starting the service.


I went through the CHANGELOG.TXT for 1.8.20.0 and it appears there are some 
items talking about changing the way the process starts up (commit r376428), 
but I'm not enough of a coder to understand if those would cause what I'm 
seeing.


Is anyone else seeing this issue?  Should I open an issue on the tracker?  
Anyone see something obvious I missed?


--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com


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Re: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start

2013-01-16 Thread Eric Wieling 
I am also experiencing this issue.  Asterisk is in fact running, you can verify 
by running "asterisk -rvvv" (-r connects to an EXISTING asterisk process) or 
using ps.



-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent: Wednesday, January 16, 2013 1:21 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to 
remote asterisk message on service asterisk start

I'm trying to decide if I need to open an issue for this or if it's just a 
misconfiguration issue of some sort.  Here's the situation - yesterday morning, 
I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS 5.8 
installation and got a shell of a basic asterisk install setup (minimum 
required configuration files, etc, with no dialplan or sip peers setup yet).  
In the afternoon, I got the notification that asterisk 1.8.20.0 had been 
released, so today, I downloaded the latest 1.8-current.tar.gz and compiled and 
installed it (./configure, make menuselect and choose all the same options as 
my previous install, make, make install).  


Now, when I start the asterisk service using "service asterisk start" from the 
command line, this is the output:

[root@pbx ~]# service asterisk start
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl 
exist?) Starting asterisk:


However, the /var/run/asterisk/asterisk.ctl file is being created and the 
process is starting:

[root@pbx ~]# ls -lh /var/run/asterisk/
total 4.0K
srwxr-xr-x 1 root root 0 Jan 16 12:07 asterisk.ctl
-rw-r--r-- 1 root root 6 Jan 16 12:07 asterisk.pid


However, I'm no longer getting the usual splash message when I connect to the 
asterisk console...this is what I get:

[root@pbx ~]# asterisk -r
Verbosity is at least 3
pbx*CLI>


I don't have any peers setup yet, or even any dialplan configured to test, but 
when I go through the logs, I don't find any errors or warnings that I'm not 
expecting.


I've gone back to the asterisk 1.8.19.1 install and everything works as 
expected (no error messages, full splash about license / version on connection 
to console, etc).  I performed make clean in my 1.8.20 source directory, then 
./configure, make menuselect, make, make install, and even make config, and I'm 
still seeing this message pop up when restarting / starting the service.  


I went through the CHANGELOG.TXT for 1.8.20.0 and it appears there are some 
items talking about changing the way the process starts up (commit r376428), 
but I'm not enough of a coder to understand if those would cause what I'm 
seeing.  


Is anyone else seeing this issue?  Should I open an issue on the tracker?  
Anyone see something obvious I missed?


--
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com


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[asterisk-users] Issue after upgrade to 1.8.20 - Unable to connect to remote asterisk message on service asterisk start

2013-01-16 Thread Warren Selby 
I'm trying to decide if I need to open an issue for this or if it's just a
misconfiguration issue of some sort.  Here's the situation - yesterday
morning, I downloaded asterisk 1.8.19.1 and installed it on a fresh CentOS
5.8 installation and got a shell of a basic asterisk install setup (minimum
required configuration files, etc, with no dialplan or sip peers setup
yet).  In the afternoon, I got the notification that asterisk 1.8.20.0 had
been released, so today, I downloaded the latest 1.8-current.tar.gz and
compiled and installed it (./configure, make menuselect and choose all the
same options as my previous install, make, make install).

Now, when I start the asterisk service using "service asterisk start" from
the command line, this is the output:

[root@pbx ~]# service asterisk start
Unable to connect to remote asterisk (does /var/run/asterisk/asterisk.ctl
exist?)
Starting asterisk:

However, the /var/run/asterisk/asterisk.ctl file is being created and the
process is starting:

[root@pbx ~]# ls -lh /var/run/asterisk/
total 4.0K
srwxr-xr-x 1 root root 0 Jan 16 12:07 asterisk.ctl
-rw-r--r-- 1 root root 6 Jan 16 12:07 asterisk.pid

However, I'm no longer getting the usual splash message when I connect to
the asterisk console...this is what I get:

[root@pbx ~]# asterisk -r
Verbosity is at least 3
pbx*CLI>

I don't have any peers setup yet, or even any dialplan configured to test,
but when I go through the logs, I don't find any errors or warnings that
I'm not expecting.

I've gone back to the asterisk 1.8.19.1 install and everything works as
expected (no error messages, full splash about license / version on
connection to console, etc).  I performed make clean in my 1.8.20 source
directory, then ./configure, make menuselect, make, make install, and even
make config, and I'm still seeing this message pop up when restarting /
starting the service.

I went through the CHANGELOG.TXT for 1.8.20.0 and it appears there are some
items talking about changing the way the process starts up (commit
r376428), but I'm not enough of a coder to understand if those would cause
what I'm seeing.

Is anyone else seeing this issue?  Should I open an issue on the tracker?
Anyone see something obvious I missed?

-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com 
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Re: [asterisk-users] special conference room

2013-01-16 Thread Bharat Lalcheta 
Please study meetme application's options. You will get almost all feature
you ask for in it
On Jan 16, 2013 5:37 AM, "Yves A."  wrote:

> Hi list,
>
> I am in need of a "special" asterisk conference room with the following
> constraints:
>
> - there is one admin / moderator and several "normal" callers.
> - the callers must not hear any other caller, only the moderator
> - the moderator must be able to mute and unmute any caller at any time
> - the moderator must be able to talk to all callers or to a specific
> caller.
> - the modetator must be able to kick off any caller at any time...
>
> Any hints on how to realize that are highly appreciated..
>
> Thanx in advance,
> yves
>
>
> --
> __**__**_
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>   http://www.asterisk.org/hello
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>   
> http://lists.digium.com/**mailman/listinfo/asterisk-**users
>
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Re: [asterisk-users] special conference room

2013-01-16 Thread Danny Nicholas 
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Yves A.
Sent: Tuesday, January 15, 2013 6:07 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] special conference room

Hi list,

I am in need of a "special" asterisk conference room with the following
constraints:

- there is one admin / moderator and several "normal" callers.
- the callers must not hear any other caller, only the moderator
- the moderator must be able to mute and unmute any caller at any time
- the moderator must be able to talk to all callers or to a specific caller.
- the modetator must be able to kick off any caller at any time...

Any hints on how to realize that are highly appreciated..

Thanx in advance,
yves

Hint 1 - specify your Asterisk version since these capabilities change
availabilities between releases.
Hint 2 - you can have the moderator enter the conference normally and
everyone else enter muted
Hint 3 - the moderator should probably have a web interface to accomplish
the other tasks
Hint 4 - refer to this link -
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_ConfBridg
e



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Re: [asterisk-users] AGI command

2013-01-16 Thread Steve Edwards 

On Wed, 16 Jan 2013, Muhammad wrote:

**When you say 'doesn't work' do you mean 'doesn't do what I want' or 
'does not execute?'


I mean I do all steps in Mr. Nir presentation documents and not works.


Your PHP script executes correctly on my dev box, but I would change the 
log file path to something absolute like '/tmp/agi_log.log' so we know 
where it is.


1) What does the Asterisk console log look like when you try to execute 
the AGI?


2) If you enter '/usr/bin/php -v', what do you get? I get:

PHP 5.1.6 (cli) (built: Feb 22 2012 19:34:21)
Copyright (c) 1997-2006 The PHP Group
Zend Engine v2.1.0, Copyright (c) 1998-2006 Zend Technologies

3) If you enter the following command in a shell, what do you get?

sudo -u asterisk /var/lib/asterisk/agi-bin/testAGI.php (If you run Asterisk as another user, use that user name in the command 
above.)


I get:

PHP Notice:  Undefined offset:  1 in /var/lib/asterisk/agi-bin/testAGI.php 
on line 31

STREAM FILE demo-congrats #
SAY NUMBER 123456 #

The 'notice' is because we passed an empty AGI environment.

--
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] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable

2013-01-16 Thread A J Stiles 
On Tuesday 15 January 2013, Ahmed Munir wrote:
> Hi,
> 
> I configured Asterisk 10 for inbound fax, for couple of weeks I didn't see
> any issues until today. The setup  I configured for inbound fax is quite
> simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38
> protocol and later Asterisk stores/forwards the fax to specific end user.
> 
> The configuration I made in sip.conf for enabling T38 is listed below;
> 
> t38pt_udptl = yes,fec,maxdatagram=400
> faxdetect = t38
> 
> And in udptl.conf, I just uncommented 'use_even_ports = yes
> ;' and rest of it set as default.
> 
> 
> Here is the error I'm usually seeing in Asterisk side;
> 
> [Jan 15 14:13:28] NOTICE[20514] udptl.c: UDPTL (SIP/10.3.22.6-0ad6):
> Transmission error to 10.3.22.6:18428: Resource temporarily unavailable
> 
> If this notice comes, it occurs repeatedly unless I need to restart the
> asterisk service. For some reason it also effect the V-GW.
> 
> Please advise what is the reason that I'm getting this message and how can
> I avoid it?

That looks a bit like the result of an IP address clash, especially as it is 
affecting more than one device.  If there are two "10.3.22.6"s on your network, 
then exactly this sort of thing can happen.

Unplug "10.3.22.6", and try pinging it.  If something answers, then you indeed 
have a clash.  Check your DHCP server configuration, and make sure any 
manually-assigned addresses are outside its pool of addresses.



-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] OT - Which Call Center class wireless headet with bluetooth connectivity ?

2013-01-16 Thread Olivier 
2013/1/16 Administrator TOOTAI 

> Le 16/01/2013 12:29, Olivier a écrit :
>
>> Hi,
>>
>
> Hello
>
>
>>
>> I'm usually working with GN Netcom 9120 Flex and have been very satisfied
>> with it but for Call Center agents needed to wear and work with headset all
>> day long in potentially  noisy offices, I'm wondering if wireless headets
>> with bluetooth connectivity exist ?
>> I'm refering to bluetooth as I've seen desktop SIP phone (such as Linksys
>> SPA525G and more) with embedded bluetooth connectivity and desktop SIP
>> phone is a requirement (ie no softphone).
>>
>
> I perhaps understand you wrong, but what would be the difference by using
> headset bluetooth connectivity?
>
> We install GN9330e to our customers (as well as use them in the office),
> they are happy with. Can be used as headset or on the ear.
>
> Other: with bluetooth connection you have a 5 m max range, with DECT it's
> much more.
>

... and depending on agents localization, a 5m range can be a limitation or
a feature.


>
> --
> Daniel
>
> --
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Re: [asterisk-users] OT - Which Call Center class wireless headet with bluetooth connectivity ?

2013-01-16 Thread Olivier 
2013/1/16 Administrator TOOTAI 

> Le 16/01/2013 12:29, Olivier a écrit :
>
>> Hi,
>>
>
> Hello
>
>
>>
>> I'm usually working with GN Netcom 9120 Flex and have been very satisfied
>> with it but for Call Center agents needed to wear and work with headset all
>> day long in potentially  noisy offices, I'm wondering if wireless headets
>> with bluetooth connectivity exist ?
>> I'm refering to bluetooth as I've seen desktop SIP phone (such as Linksys
>> SPA525G and more) with embedded bluetooth connectivity and desktop SIP
>> phone is a requirement (ie no softphone).
>>
>
> I perhaps understand you wrong, but what would be the difference by using
> headset bluetooth connectivity?
>

That's the point !
I've read here and there some bluetooth headsets are able to be paired with
several devices at the same time.
What I have in mind is to pair a single (bluetooth) headset with both a
deskphone and a DECT handset, giving agents more mobility (with a DECT
headset alone, you can answer/hangup but cannot transfer or proceed with
other telephony operations).


>
> We install GN9330e to our customers (as well as use them in the office),
> they are happy with. Can be used as headset or on the ear.
>
> Other: with bluetooth connection you have a 5 m max range, with DECT it's
> much more.
>
> --
> Daniel
>
> --
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Re: [asterisk-users] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable

2013-01-16 Thread Ahmed Munir 
Hi Christopher,

I'm using Asterisk 10.4.2. Do I need to install updated version to resolve
this issue? Please advise.


> --
>
> Date: Tue, 15 Jan 2013 15:45:31 -0600
> From: Christopher Harrington 
> Subject: Re: [asterisk-users] Getting UDPTL (SIP): Transmission error:
> Resource temporarily unavailable
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Message-ID:
>  ckcfqh2ntzbzxuhu+vsgqvbn+nqc5gytdk...@mail.gmail.com>
> Content-Type: text/plain; charset="utf-8"
>
> Can you be more specific about your Asterisk version? 10.xx.yy ?
>
> Sounds like some sort of resource leak.
>
>
> On Tue, Jan 15, 2013 at 3:02 PM, Ahmed Munir  >wrote:
>
> > Hi,
> >
> > I configured Asterisk 10 for inbound fax, for couple of weeks I didn't
> see
> > any issues until today. The setup  I configured for inbound fax is quite
> > simple i.e. Cisco Voice GW sends the fax calls to Asterisk using T.38
> > protocol and later Asterisk stores/forwards the fax to specific end user.
> >
> > The configuration I made in sip.conf for enabling T38 is listed below;
> >
> > t38pt_udptl = yes,fec,maxdatagram=400
> > faxdetect = t38
> >
> > And in udptl.conf, I just uncommented 'use_even_ports = yes
> > ;' and rest of it set as default.
> >
> >
> > Here is the error I'm usually seeing in Asterisk side;
> >
> > [Jan 15 14:13:28] NOTICE[20514] udptl.c: UDPTL (SIP/10.3.22.6-0ad6):
> > Transmission error to 10.3.22.6:18428: Resource temporarily unavailable
> >
> > If this notice comes, it occurs repeatedly unless I need to restart the
> > asterisk service. For some reason it also effect the V-GW.
> >
> > Please advise what is the reason that I'm getting this message and how
> can
> > I avoid it?
> >
> >
> > --
> > Regards,
> >
> > Ahmed Munir Chohan
> >
> >
> > --
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> >
>
>
>
> --
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> ACSDi Office: 763.559.5800
> Mobile Phone: 612.326.4248
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> >
>
> -
>


-- 
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Re: [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)

2013-01-16 Thread Matthew Jordan 
On 01/16/2013 05:31 AM, Ishfaq Malik wrote:
> On Thu, 2012-01-12 at 11:51 +, Ishfaq Malik wrote:
> 
> Hi Everyone
> 
> This issue has reared it's ugly head again for us. If a call comes into
> a queue and the caller abandons the call, the call does not show in the
> CDR.
> 
> This is also the case for asterisk version 1.8.18
> 
> Does anyone have any ideas, or try to replicate it?
> 
> Thanks in advance
> 
> Ish
> 

Do you have unanswered=yes set in cdr.conf?

CDRs in Queues can depend heavily on your dialplan, whether or not the
call is Answered prior to it going into the Queue, etc. What is the
state of the inbound channel when it goes into the Queue?

-- 
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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



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Re: [asterisk-users] Asterisk 11- Answer with [m=image 0 udptl t38] and Call Drop

2013-01-16 Thread Matthew Jordan 
On 01/16/2013 07:28 AM, Salman Zafar wrote:
> Hello All,
>I am having a bit peculiar problem with Asterisk 11 for a
> carrier. This carrier shares quite some information in SDP header, which
> should not be the problem, however what happen is as follow:
> 
> 
> Carrier> (INVITE) -> *SIP Proxy -> Asterisk 11 -> Answer()* -> right
> after answering call drops... Carrier send a BYE with (cause 79: service
> or option not implemented).
> 
> *NOTE: Please refer to complete SIP traces attached. *
> *
> *
> *Also Note:*
> _Carrier_: 62.61.147.214
> _Proxy_: 77.X.X.X:5060
> _Asterisk11_: 77.X.X.X:5080
> 
> *_Here is Invite SDP  from Carrier -> Proxy -> Asterisk 11_*
> 
> INVITE sip:69609000@77.X.X.X SIP/2.0
> v=0
> o=AudiocodesGW 1638819008 1638818710 IN IP4 62.61.147.214
> s=Phone-Call
> c=IN IP4 77.X.X.X
> t=0 0
> m=audio 53372 RTP/AVP 8 118 18
> a=rtpmap:8 PCMA/8000
> a=rtpmap:118 PCMA/8000
> a=gpmd:118 vbd=yes
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=ptime:20
> a=sendrecv
> a=rtcp:53373 IN IP4 77.X.X.X
> m=image 56854 udptl t38
> a=T38FaxVersion:0
> a=T38MaxBitRate:14400
> a=T38FaxMaxBuffer:1024
> a=T38FaxMaxDatagram:122
> a=T38FaxRateManagement:transferredTCF
> a=T38FaxUdpEC:t38UDPRedundancy
> 
> /*_SDP:After Answered by Asterisk 11_*/
> v=0
> o=root 164966782 164966782 IN IP4 77.X.X.X
> s=Asterisk v11.0.1
> c=IN IP4 77.X.X.X
> t=0 0
> m=audio 12636 RTP/AVP 18 8
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:8 PCMA/8000
> a=ptime:20
> a=sendrecv
> *_m=image 0 udptl t38_*


The appropriate way for Asterisk to indicate that it does not support a
media stream is to set the port number to 0. We have to inform the
offerer that we don't support the media stream; removing it from the SDP
completely is not allowed.

Per RFC 3264, section 6:

"   An offered stream MAY be rejected in the answer, for any reason.  If
   a stream is rejected, the offerer and answerer MUST NOT generate
   media (or RTCP packets) for that stream.  To reject an offered
   stream, the port number in the corresponding stream in the answer
   MUST be set to zero. "

> I have tired by disabling/unloading fax modules as *I am not using* them
> but no results. Secondly, also tried tweaking of udptl ever-odd nothing
> worked.

You've configured your system to not support fax correctly. Asterisk is
rejecting the offered image stream accordingly.

> The same carrier works for Asterisk 1.6.X and the only difference I have
> notice so far is the above underlined line in Answered SDP -> m=image 0
> udptl t38. I think if I some how do not advertise udptl here i would be
> able to avoid this scenario. I have tried multiple ways to strip off SDP
> from incoming INVITE at SIP Proxy level but it is not SDP wise enough. 
> 

I'm not sure what 1.6.x is sending. It's possible that it just
completely removed the stream from the SDP answer, which is wrong.

Section 6 again:

"For each "m=" line in the offer, there MUST be a corresponding "m="
   line in the answer."

> *Note:*
> 
> In Asterisk 1.6 =>  WARNING[32671]: chan_sip.c:8833 process_sdp:
> Unsupported SDP media type in offer: image 59978 udptl t38
> In Asterisk 11 => WARNING[18748][C-002f]: chan_sip.c:10277
> process_sdp: Failed to initialize UDPTL, declining image stream
> 
> 

An initial glance at this makes me think your carrier is doing something
wrong. Just to check, however, is the SDP answer you pasted the entire
SDP that Asterisk 11 responds with? Specifically, are there no format
attributes for the image stream in the SDP that Asterisk responds with?

-- 
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org



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Re: [asterisk-users] OT - Which Call Center class wireless headet with bluetooth connectivity ?

2013-01-16 Thread Administrator TOOTAI 

Le 16/01/2013 12:29, Olivier a écrit :

Hi,


Hello



I'm usually working with GN Netcom 9120 Flex and have been very 
satisfied with it but for Call Center agents needed to wear and work 
with headset all day long in potentially  noisy offices, I'm wondering 
if wireless headets with bluetooth connectivity exist ?
I'm refering to bluetooth as I've seen desktop SIP phone (such as 
Linksys SPA525G and more) with embedded bluetooth connectivity and 
desktop SIP phone is a requirement (ie no softphone).


I perhaps understand you wrong, but what would be the difference by 
using headset bluetooth connectivity?


We install GN9330e to our customers (as well as use them in the office), 
they are happy with. Can be used as headset or on the ear.


Other: with bluetooth connection you have a 5 m max range, with DECT 
it's much more.


--
Daniel

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[asterisk-users] Asterisk 11- Answer with [m=image 0 udptl t38] and Call Drop

2013-01-16 Thread Salman Zafar 
Hello All,
   I am having a bit peculiar problem with Asterisk 11 for a
carrier. This carrier shares quite some information in SDP header, which
should not be the problem, however what happen is as follow:


Carrier> (INVITE) -> *SIP Proxy -> Asterisk 11 -> Answer()* -> right
after answering call drops... Carrier send a BYE with (cause 79: service or
option not implemented).

*NOTE: Please refer to complete SIP traces attached. *
*
*
*Also Note:*
*Carrier*: 62.61.147.214
*Proxy*: 77.X.X.X:5060
*Asterisk11*: 77.X.X.X:5080

*Here is Invite SDP  from Carrier -> Proxy -> Asterisk 11*

INVITE sip:69609000@77.X.X.X SIP/2.0
v=0
o=AudiocodesGW 1638819008 1638818710 IN IP4 62.61.147.214
s=Phone-Call
c=IN IP4 77.X.X.X
t=0 0
m=audio 53372 RTP/AVP 8 118 18
a=rtpmap:8 PCMA/8000
a=rtpmap:118 PCMA/8000
a=gpmd:118 vbd=yes
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtcp:53373 IN IP4 77.X.X.X
m=image 56854 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:122
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy

*SDP:After Answered by Asterisk 11*
v=0
o=root 164966782 164966782 IN IP4 77.X.X.X
s=Asterisk v11.0.1
c=IN IP4 77.X.X.X
t=0 0
m=audio 12636 RTP/AVP 18 8
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv
*m=image 0 udptl t38*

I have tired by disabling/unloading fax modules as *I am not using* them
but no results. Secondly, also tried tweaking of udptl ever-odd nothing
worked.

The same carrier works for Asterisk 1.6.X and the only difference I have
notice so far is the above underlined line in Answered SDP -> m=image 0
udptl t38. I think if I some how do not advertise udptl here i would be
able to avoid this scenario. I have tried multiple ways to strip off SDP
from incoming INVITE at SIP Proxy level but it is not SDP wise enough.


*Note:*

In Asterisk 1.6 =>  WARNING[32671]: chan_sip.c:8833 process_sdp:
Unsupported SDP media type in offer: image 59978 udptl t38
In Asterisk 11 => WARNING[18748][C-002f]: chan_sip.c:10277 process_sdp:
Failed to initialize UDPTL, declining image stream


-- 
Regards

*Muhammad Salman Zafar*
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Re: [asterisk-users] Call abandoned from queue not showing in CDR (possible bug)

2013-01-16 Thread Ishfaq Malik 
On Thu, 2012-01-12 at 11:51 +, Ishfaq Malik wrote:
> Hi
> 
> I'm using 1.8.7.0 with the RealTime architecture.
> 
> If a call goes into application Queue and is abandoned by the caller, no
> entry is made in the CDR. Entries are made into the queue log. 
> 
> This cannot be correct behaviour, all calls should show in the CDR.
> 
> Could anyone else try to reproduce this and if others get the same
> thing, I'll raise a bug on it.
> 
> Thanks
> 
> Ish

Hi Everyone

This issue has reared it's ugly head again for us. If a call comes into
a queue and the caller abandons the call, the call does not show in the
CDR.

This is also the case for asterisk version 1.8.18

Does anyone have any ideas, or try to replicate it?

Thanks in advance

Ish

-- 
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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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[asterisk-users] OT - Which Call Center class wireless headet with bluetooth connectivity ?

2013-01-16 Thread Olivier 
Hi,


I'm usually working with GN Netcom 9120 Flex and have been very satisfied
with it but for Call Center agents needed to wear and work with headset all
day long in potentially  noisy offices, I'm wondering if wireless headets
with bluetooth connectivity exist ?
I'm refering to bluetooth as I've seen desktop SIP phone (such as Linksys
SPA525G and more) with embedded bluetooth connectivity and desktop SIP
phone is a requirement (ie no softphone).

Suggestions welcome.

Regards
--
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Re: [asterisk-users] AGI command

2013-01-16 Thread Muhammad 
***When you say 'doesn't work' do you mean 'doesn't do what I want' or
'does not execute?'*

I mean I do all steps in Mr. Nir presentation documents and not works.

Here is my php code:

#!/usr/bin/php -q
 testAGI

[testAGI]
exten => 147,1,Answer
exten => 147,2,AGI(testAGI.php)
exten => 147,3,Hangup


147 is my extension.
operators can call a number via their extension(each extension may use for
one or more users?)

I register extension 147 in my client softphone and call a number. what
happen after that?

On Wed, Jan 16, 2013 at 11:19 AM, Steve Edwards
wrote:

> On Wed, 16 Jan 2013, Muhammad wrote:
>
>  I wrote some php code to working with AGI, but it dosen't work.
>>
>
> When you say 'doesn't work' do you mean 'doesn't do what I want' or 'does
> not execute?'
>
> If you enable AGI debugging, what does the Asterisk console log look like?
>
> Did you use an established PHP library or 'roll your own?'
>
> A good way to test an AGI is to create a text file containing all the
> cruft (the AGI 'environment') Asterisk sends to the AGI along with the
> expected responses. Then you can execute your AGI completely external from
> Asterisk with a shell command line like:
>
> /var/lib/asterisk/agi-bin/my-**firs-agi 
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: 
> +1-760-468-3867PST
> Newline  Fax: 
> +1-760-731-3000
>
>
> --
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Re: [asterisk-users] AGI command

2013-01-16 Thread Steve Edwards 

On Wed, 16 Jan 2013, Zohair Raza wrote:

Make sure Asterisk has access to your AGI script, and make it executable 
(chmod u+x agi.php). Also make sure it has shebang (!#/usr/bin/php)


Make sure that the user executing the Asterisk process can execute your 
script. Ownership (user and group), directory permissions, and file 
permissions all play a part.


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

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