date:20110223

Re: [asterisk-users] AMI FullyBooted issue

2011-02-23 Thread Marcin Szymański

Hi,

I have this same behaviour on version 1.8.2.3 build from source. We are using 
AMI to originate call from our CRM software, but we ignore that message.

Regards,

Marcin

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Re: [asterisk-users] DIAL through Specific number in PRI

2011-02-23 Thread Faisal Hanif

If your PRI provider permit you to associate any ANI to any Circuit-ID you
can do this.

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of DHAVAL
INDRODIYA
Sent: Thursday, February 24, 2011 12:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] DIAL through Specific number in PRI

Hi ALL,

I have PRI line everything is fine , but my customer having a requirement
that they want to DIAL a number from PRI which gives callerid as 
Specific number.

i.e

PRI start from 3055 to 30550100  i have purchased a 100 number from
telco and our pilot number is 3055, now if some caller want to dial any
number but caller should shown is 30550008 like this.

is there any solution from asterisk side.

regards
Dhaval

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[asterisk-users] DIAL through Specific number in PRI

2011-02-23 Thread DHAVAL INDRODIYA

Hi ALL,

I have PRI line everything is fine , but my customer having a requirement
that they want to DIAL a number from PRI which gives callerid as
Specific number.

i.e

PRI start from 3055 to 30550100  i have purchased a 100 number from
telco and our pilot number is 3055, now if some caller want to dial any
number but caller should shown is 30550008 like this.

is there any solution from asterisk side.

regards
Dhaval
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Re: [asterisk-users] alarm POTS lines

2011-02-23 Thread Andrew Joakimsen

On Thu, Dec 2, 2010 at 11:58, Jeff LaCoursiere  wrote:
> we have a low-cost Atom based PBX and a "fax relay" setup locally with
> hylafax/iaxmodem to solve that issue, and it is working very well.  We
> don't however, have a solution for their alarm lines.

You would desire the entire path to be UL listed if you are doing
anything other than facilitating the phone call to the central
station. There is app_alarmreciever in Asterisk, and furthermore the
ContactID protocol is pure DTMF so that should work without issues.
But why use phone lines at all? Recently I installed a DSC T-Link
TL260GS which uses internet and GSM, there is no phone line plugged
into the alarm panel at all.

> The problem is of course that modem calls over VoIP are flaky at best.
> Even though these alarm calls are low baud rate, when we test with the
> alarm company we only pass about 30% of the time (ulaw from customer site
> to our central switch, then out a T1).  To be fair there is no QoS on
> their Internet links yet, and that certainly plays a role.

SIA format is 110 or 300 baud, ContactID is (rapid) DTMF.

-- 
Med Vennlig Hilsen,

A. Helge Joakimsen

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Re: [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1

2011-02-23 Thread Terry Wilson


On Feb 23, 2011, at 7:11 PM, Jose P. Espinal wrote:

> On 02/23/2011 08:56 PM, Leif Madsen wrote:
>> 
>> Actually I was wrong!
>> 
>> See here. It is being resolved.
>> 
>> https://reviewboard.asterisk.org/r/1107/
>> 
>> Leif.
> 
> Thanks for the feedback, Leif!
> 
> I will follow that incident closely, as I was starting to doubt about my 
> understanding of English (jk)

I had forgotten that I got a "Ship It!" on that patch. I went ahead and 
committed the fix to 1.6.2, 1.8, and trunk.

Terry
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Re: [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card

2011-02-23 Thread C F

This worked.
Thank you all for your help.

On Wed, Feb 23, 2011 at 1:42 PM, Greg Woods  wrote:
> On Wed, 2011-02-23 at 09:56 -0500, C F wrote:
>> This is the closest thing I was able to find in my wctdm.c file:
>>         if ((blah & 0xf) == 2) {
>>                 /* ProSLIC 3215, not a 3210 */
>>                 wc->flags[card] |= FLAG_3215;
>>         }
>> If I take out the 2 first lines I get errors when compling.
>
>
> Maybe you need to remove the closing brace too?
>
> --Greg
>
>
>
>
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Re: [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1

2011-02-23 Thread Jose P. Espinal


On 02/23/2011 08:56 PM, Leif Madsen wrote:


Actually I was wrong!

See here. It is being resolved.

https://reviewboard.asterisk.org/r/1107/

Leif.


Thanks for the feedback, Leif!

I will follow that incident closely, as I was starting to doubt about my 
understanding of English (jk)




--
Jose P. Espinal
http://www.eslackware.com
IRC: [OFTC|FreeNode]
Khratos @ #slackware | #asterisk/-doc/-bugs

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[asterisk-users] One way dialing over a SIP trunk

2011-02-23 Thread Mitch Johnson

I have a SIP trunk built between a Cisco CallManager version 8.  I can dial the 
phones registered to the Asterisk PBX from a phone registered to the Call 
Manager.

I've tried to keep the config as small as possible to help the troubleshooting 
process.  Attached is he most recent debug.

My Callmanager IP address is 10.169.169.250, Asterisk server is 10.169.169.251

SIP.CONF

[6001]
type=friend
secret=cisco2003
callerid="Dave" <6001>
host=dynamic
canreinvite=no
context=myphones
regexten=6001

[CM8]
type=friend
host=10.169.169.250
canreinvite=yes
;disallow=all
allow=ulaw
allow=alaw
qualify=yes
context=myphones
 
 
Extensions.conf
 
myphones]
exten => 6001,1,Dial(SIP/6001)
exten => 6001,2,Hangup()

exten => _X.,1,Dial(SIP/CM8/${EXTEN:0},30,rt)


Thanks for any help.

Mitch


{\rtf1\ansi\ansicpg1252\cocoartf1038\cocoasubrtf350
{\fonttbl\f0\fswiss\fcharset0 Helvetica;}
{\colortbl;\red255\green255\blue255;\red20\green54\blue165;}
\margl1440\margr1440\vieww12000\viewh8400\viewkind0
\deftab720
\pard\pardeftab720\ql\qnatural

\f0\fs24 \cf0 tunafish*CLI> \
\
<--- SIP read from UDP:{\field{\*\fldinst{HYPERLINK "http://10.169.169.138:2048/"}}{\fldrslt \cf2 \ul \ulc2 10.169.169.138:2048}} --->\
INVITE {\field{\*\fldinst{HYPERLINK "mailto:sip%3A6500@10.169.169.251"}}{\fldrslt \cf2 \ul \ulc2 sip:6500@10.169.169.251}};user=phone SIP/2.0\
Via: SIP/2.0/UDP 10.169.169.138:2048;branch=z9hG4bK-k3a2um4e9a01;rport\
From: "test 6002" <{\field{\*\fldinst{HYPERLINK "mailto:sip%3A6002@10.169.169.251"}}{\fldrslt \cf2 \ul \ulc2 sip:6002@10.169.169.251}}>;tag=zv4a0j6k6y\
To: <{\field{\*\fldinst{HYPERLINK "mailto:sip%3A6500@10.169.169.251"}}{\fldrslt \cf2 \ul \ulc2 sip:6500@10.169.169.251}};user=phone>\
Call-ID: 3c26ed7edf41-m2sk5pf5ralb\
CSeq: 1 INVITE\
Max-Forwards: 70\
Contact: <{\field{\*\fldinst{HYPERLINK "sip:6002@10.169.169.138:2048;line=vtmynxjj"}}{\fldrslt \cf2 \ul \ulc2 sip:6002@10.169.169.138:2048;line=vtmynxjj}}>;reg-id=1\
X-Serialnumber: 000413347AE4\
P-Key-Flags: keys="3"\
User-Agent: snom300/8.4.18\
Accept: application/sdp\
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE\
Allow-Events: talk, hold, refer, call-info\
Supported: timer, 100rel, replaces, from-change\
Session-Expires: 3600;refresher=uas\
Min-SE: 90\
Content-Type: application/sdp\
Content-Length: 395\
\
v=0\
o=root 1038861251 1038861251 IN IP4 10.169.169.138\
s=call\
c=IN IP4 10.169.169.138\
t=0 0\
m=audio 59248 RTP/AVP 0 8 9 99 3 18 4 101\
a=rtpmap:0 pcmu/8000\
a=rtpmap:8 pcma/8000\
a=rtpmap:9 g722/8000\
a=rtpmap:99 g726-32/8000\
a=rtpmap:3 gsm/8000\
a=rtpmap:18 g729/8000\
a=fmtp:18 annexb=no\
a=rtpmap:4 g723/8000\
a=rtpmap:101 telephone-event/8000\
a=fmtp:101 0-16\
a=ptime:20\
a=sendrecv\
<->\
--- (19 headers 18 lines) ---\
Sending to {\field{\*\fldinst{HYPERLINK "http://10.169.169.138:2048/"}}{\fldrslt \cf2 \ul \ulc2 10.169.169.138:2048}} (no NAT)\
Using INVITE request as basis request - 3c26ed7edf41-m2sk5pf5ralb\
Found peer '6002' for '6002' from {\field{\*\fldinst{HYPERLINK "http://10.169.169.138:2048/"}}{\fldrslt \cf2 \ul \ulc2 10.169.169.138:2048}}\
\
<--- Reliably Transmitting (no NAT) to {\field{\*\fldinst{HYPERLINK "http://10.169.169.138:2048/"}}{\fldrslt \cf2 \ul \ulc2 10.169.169.138:2048}} --->\
SIP/2.0 401 Unauthorized\
Via: SIP/2.0/UDP 10.169.169.138:2048;branch=z9hG4bK-k3a2um4e9a01;received=10.169.169.138;rport=2048\
From: "test 6002" <{\field{\*\fldinst{HYPERLINK "mailto:sip%3A6002@10.169.169.251"}}{\fldrslt \cf2 \ul \ulc2 sip:6002@10.169.169.251}}>;tag=zv4a0j6k6y\
To: <{\field{\*\fldinst{HYPERLINK "mailto:sip%3A6500@10.169.169.251"}}{\fldrslt \cf2 \ul \ulc2 sip:6500@10.169.169.251}};user=phone>;tag=as2244e25f\
Call-ID: 3c26ed7edf41-m2sk5pf5ralb\
CSeq: 1 INVITE\
Server: Asterisk PBX 1.8.2.4\
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH\
Supported: replaces, timer\
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a1c370b"\
Content-Length: 0\
\
\
<>\
Scheduling destruction of SIP dialog '3c26ed7edf41-m2sk5pf5ralb' in 32000 ms (Method: INVITE)\
\
<--- SIP read from UDP:{\field{\*\fldinst{HYPERLINK "http://10.169.169.138:2048/"}}{\fldrslt \cf2 \ul \ulc2 10.169.169.138:2048}} --->\
ACK {\field{\*\fldinst{HYPERLINK "mailto:sip%3A6500@10.169.169.251"}}{\fldrslt \cf2 \ul \ulc2 sip:6500@10.169.169.251}};user=phone SIP/2.0\
Via: SIP/2.0/UDP 10.169.169.138:2048;branch=z9hG4bK-k3a2um4e9a01;rport\
From: "test 6002" <{\field{\*\fldinst{HYPERLINK "mailto:sip%3A6002@10.169.169.251"}}{\fldrslt \cf2 \ul \ulc2 sip:6002@10.169.169.251}}>;tag=zv4a0j6k6y\
To: <{\field{\*\fldinst{HYPERLINK "mailto:sip%3A6500@10.169.169.251"}}{\fldrslt \cf2 \ul \ulc2 sip:6500@10.169.169.251}};user=phone>;tag=as2244e25f\
Call-ID: 3c26ed7edf41-m2sk5pf5ralb\
CSeq: 1 ACK\
Max-Forwards: 70\
Contact: <{\field{\*\fldinst{HYPERLINK "sip:6002@10.169.169.138:2048;line=vtmynxjj"}}{\fldrslt \cf2 \ul \ulc2 sip:6002@10.169.169.138:2048;line=vtmynxjj}}>;reg-id=1\
Content-L

Re: [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1

2011-02-23 Thread Leif Madsen


On 11-02-23 10:31 AM, Jose P. Espinal wrote:

Hello List,

I have a little issue with calls placed to a provider declared on sip.conf,
because of a not clear (*for me*) behavior of 'remotesecret' parameter.


Actually I was wrong!

See here. It is being resolved.

https://reviewboard.asterisk.org/r/1107/

Leif.

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Re: [asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1

2011-02-23 Thread Leif Madsen


On 11-02-23 10:31 AM, Jose P. Espinal wrote:

-
Added a new configuration option "remotesecret" for authentication to
remote services. For backwards compatibility, "secret" still has the
same function as before, but now you can configure both a remote secret
and a local secret for mutual authentication.
-
I thought that 'remotesecret' is used to authenticate myself when placing a call
to the remote network, as I used to do with 'secret' parameter.


I may be mistaken, because I don't use remotesecret, but I think the purpose of 
that was to allow different authentication depending on the direction. My guess 
is remotesecret is used to authenticate the "remote" end when a call is placed 
into Asterisk, and secret is used when you're placing a call to the remote server.


Or it's possible the feature has a bug and an issue should probably be opened on 
the issue tracker ;)


Leif.

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Re: [asterisk-users] REFER and dialplan broken(asdocumentedinchan_sip.c on line 11951)

2011-02-23 Thread Cary Fitch

It is free if you can use it.  You can pay for all the help you want to or
have the money to pay for.

 

The "Asterisk Software Charity Society" went bankrupt about 2500 years ago. 

 

You can pick some name from the mail list and demand they fix the issue you
perceive.  But you probably won't be able to wait for results.

 

There are 3 legs to any transaction.  Speed, Quality, Price.   You get to
pick any two.  The other party gets to set the 3rd one.

 

You can't set all 3.

 

Cary

 

Except in Wisconsin.

 

Even there.

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Re: [asterisk-users] REFER and dialplan broken (asdocumentedinchan_sip.c on line 11951)

2011-02-23 Thread Danny Nicholas

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch
Sent: Wednesday, February 23, 2011 4:04 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] REFER and dialplan broken
(asdocumentedinchan_sip.c on line 11951)

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 3:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as
documentedinchan_sip.c on line 11951)

I've exhausted every option without paying someone to fix this, so asterisk
might as well be commercial software.

It is free if you can use it.  You can pay for all the help you want to or
have the money to pay for.

The "Asterisk Software Charity Society" went bankrupt about 2500 years ago. 

You can pick some name from the mail list and demand they fix the issue you
perceive.  But you probably won't be able to wait for results.

There are 3 legs to any transaction.  Speed, Quality, Price.   You get to
pick any two.  The other party gets to set the 3rd one.

You can't set all 3.

Cary

Except in Wisconsin.

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Re: [asterisk-users] REFER and dialplan broken (as documentedinchan_sip.c on line 11951)

2011-02-23 Thread Cary Fitch

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 3:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as
documentedinchan_sip.c on line 11951)

I've exhausted every option without paying someone to fix this, so asterisk
might as well be commercial software.

It is free if you can use it.  You can pay for all the help you want to or
have the money to pay for.

The "Asterisk Software Charity Society" went bankrupt about 2500 years ago. 

You can pick some name from the mail list and demand they fix the issue you
perceive.  But you probably won't be able to wait for results.

There are 3 legs to any transaction.  Speed, Quality, Price.   You get to
pick any two.  The other party gets to set the 3rd one.

You can't set all 3.

Cary

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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Warren Selby

On Wed, Feb 23, 2011 at 3:43 PM, vip killa  wrote:

> I've exhausted every option without paying someone to fix this, so asterisk
> might as well be commercial software.
>
>
If you're really interested in trying to resolve your issue, as opposed to
just complaining about it, perhaps you can post the requested debug
information[1] from earlier.

[1] -
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

With the requested information, you may be surprised at the type and level
of help you may get from this mailing list.  If all else fails, you can
always open a new issue on the bug tracker and it will get looked at.  It's
a pretty painless procedure.


-- 
Thanks,
--Warren Selby, dCAP
http://www.selbytech.com
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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Steve Edwards


On Wed, 23 Feb 2011, vip killa wrote:

I've exhausted every option without paying someone to fix this, so 
asterisk might as well be commercial software.


You 'effing' kill me :)

You have to be a troll. You can't be this stupid.

--
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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread vip killa

I've exhausted every option without paying someone to fix this, so asterisk
might as well be commercial software.

On Wed, Feb 23, 2011 at 2:21 PM, Richard Kenner  wrote:

> > I recognize all the options given yet as I explained before they are not
> > viable. I do not have the resources to pay someone, I do not have the
> > expertise to fix this issue because according to an asterisk developer
> > "any fix in that area would be deeply architectural in nature"... what
> > other options are there?
>
> In a commercial product, you have two options when you find a bug:
>
> (1) Pay for it to be fixed.
> (2) Live with the bug.
>
> In an open-sourced product, you have those same two options, plus an
> additional one:
>
> (3) Fix it yourself.
>
> Those are the only three you have to choose from.
>
>
> --
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[asterisk-users] SIP friend name

2011-02-23 Thread Paul Dugas

Is there a way to configure a friend in sip.conf that allows a station
to register using a username other than the [name]?

I want to have something like this in sip.conf:

[1234]
username=something_really_long_and_random
secret=something_else_really_long_and_random
...

Then allow a SIP REGISTER like so:

REGISTER sip:10.0.0.200:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.201;branch=z9hG4bK9238ea1821361589
From: "Joe User"
;tag=CA8D62BF-9750393E
To: 
CSeq: 2 REGISTER
Call-ID: fbc4bb4b-91001d55-8c6da4c4@10.0.0.201
Contact: ;methods="INVITE,
ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY,
PRACK, UPDATE, REFER"
User-Agent: PolycomSoundPointIP-SPIP_501-UA/3.1.6.0017
Accept-Language: en
Authorization: Digest username="something_really_long_and_random",
realm="asterisk", nonce="35d57376", uri="sip:10.0.0.201:5060",
response="5a0596db3c3f1823b783ae195074cc5c", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0

And end up being able to reference the device via SIP/1234.

I've tried adding name, username, auth, authname, defaultname, and
fromname among others and none seem to do what I'm looking for.  Tried
reading through chan_sip.c but have not figured it out yet.  Figured
I'd ask.

Thanks in advance,

Paul

PS: I had a SIP entry with too easy a secret get exploited to the tune
of $500 worth of calls to Liberia...

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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Jeff LaCoursiere




On Wed, 23 Feb 2011, Danny Nicholas wrote:






From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
inchan_sip.c on line 11951)

 

I recognize all the options given yet as I explained before they are not 
viable. I do not have the resources to pay someone, I do not have the expertise 
to
fix this issue because according to an asterisk developer "any fix in that area 
would be deeply architectural in nature"... what other options are there?

 



From what I see, the “source fix” on the Asterisk level would indeed be a major 
undertaking.  But since you are using an AGI to control the Queue command
instead of using it from the dialplan, you have more control over this problem 
than you realize.  For simplicity of illustration, let’s say your AGI simply
wants to take a call and send it to the next agent in the queue. Your Agents 
are Agent007, AgentQ and AgentM.  Because you did the Polycom transfer from
Agent007 to pussygalore, Agent007 is marked as busy in the queue although the 
call is no longer active for 007. One possible workaround would be to have a
duplicate “bail queue” set up the same way.  If my AGI does a “core show 
channels” and sees that 007 is not on the phone, I can do queue(bail) instead of
queue(normal).

 



Watch out for race conditions doing things like this...

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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Richard Kenner

> I recognize all the options given yet as I explained before they are not
> viable. I do not have the resources to pay someone, I do not have the
> expertise to fix this issue because according to an asterisk developer
> "any fix in that area would be deeply architectural in nature"... what
> other options are there?

In a commercial product, you have two options when you find a bug:

(1) Pay for it to be fixed.
(2) Live with the bug.

In an open-sourced product, you have those same two options, plus an
additional one:

(3) Fix it yourself.

Those are the only three you have to choose from.


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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Danny Nicholas

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 12:44 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

I recognize all the options given yet as I explained before they are not
viable. I do not have the resources to pay someone, I do not have the
expertise to fix this issue because according to an asterisk developer "any
fix in that area would be deeply architectural in nature"... what other
options are there?

 



>From what I see, the "source fix" on the Asterisk level would indeed be a
major undertaking.  But since you are using an AGI to control the Queue
command instead of using it from the dialplan, you have more control over
this problem than you realize.  For simplicity of illustration, let's say
your AGI simply wants to take a call and send it to the next agent in the
queue. Your Agents are Agent007, AgentQ and AgentM.  Because you did the
Polycom transfer from Agent007 to pussygalore, Agent007 is marked as busy in
the queue although the call is no longer active for 007. One possible
workaround would be to have a duplicate "bail queue" set up the same way.
If my AGI does a "core show channels" and sees that 007 is not on the phone,
I can do queue(bail) instead of queue(normal).

 

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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Jason Parker


On 02/23/2011 12:43 PM, vip killa wrote:

I recognize all the options given yet as I explained before they are not viable.
I do not have the resources to pay someone, I do not have the expertise to fix
this issue because according to an asterisk developer "any fix in that area
would be deeply architectural in nature"... what other options are there?



Option 3 was "wait for someone else with the skills and/or money necessary to 
fix it".  Demanding that somebody fix an issue will not work in any community, 
open source or otherwise.  You'll only be labeled a nuisance and ignored.


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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Warren Selby

Sorry for the top post - this is from my phone. 

Sounds like the issue may actually be with the AGI that is handling your ACD 
queue. I've used the built-in Queue() command to handle situations like you 
describe without running into the issues you detailed. And that's with Polycom 
phones, too. 

Without more details, I'm not sure how much help you're going to get. Show us 
some console output of the issue, capture the proper debug logs, etc, and 
perhaps you'll find more help. 

Thanks,
--Warren Selby, dCAP

On Feb 23, 2011, at 11:57 AM, vip killa  wrote:

> I'm sorry i don't know what you mean by natively. I'm almost certain the 
> queue is handled via AGI and not using asterisk's queue. 
> 
> On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas  wrote:
> Do you use the Queue command “natively” or from the AGI?  In the example you 
> gave, if you did a “core show channels”, I assume that Agent007 would be 
> idle, but ineligible for Queue activity.
> 
>  
> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
> Sent: Wednesday, February 23, 2011 11:37 AM
> 
> 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
> inchan_sip.c on line 11951)
>  
> 
> Sure, it really manifests itself whenever using AGI for call flow, but this 
> is how it affects us...
> 
> incoming call -> queue -> agent007 -> xfer -> pussygalore
> 
> now the AGI/dialplan thinks agent007 is on phone with pussygalore until that 
> xfered call terminates so if another call comes into queue while pussygalore 
> is on the phone w/ that xfered call, agent007 will not even be attempted by 
> queue
> 
>  
> 
> I'm sure there could be other scenarios in which this REFER issue could pose 
> a problem but this is the most consequential scenario which we have to deal 
> with everyday.
> 
>  
> 
>  
> 
> On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas  wrote:
> 
> I use Polycom 501’s and use the Transfer Key to send inbound calls to other 
> extensions.  Can you give me an A-B-C example of how this problem manifests 
> itself?
> 
>  
> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
> Sent: Wednesday, February 23, 2011 11:11 AM
> 
> 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
> inchan_sip.c on line 11951)
> 
>  
> 
> Interesting but the issue I'm having relates to Inbound and Outbound REFERs 
> since I'm using Polycom's Transfer softkey (which allows for both Inbound and 
> Outbound Transfers). I know this is not an issue when using Asterisk's 
> built-in transfer (only allows Inbound transfers).
> 
>  
> 
> On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas  wrote:
> 
> Have you read this thread?
> 
> http://forums.digium.com/viewtopic.php?t=74418
> 
>  
> 
>  
> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
> Sent: Wednesday, February 23, 2011 10:36 AM
> 
> 
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> 
> Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
> inchan_sip.c on line 11951)
> 
>  
> 
> I did not see this issue anywhere on issues.asterisk.org
> 
> Can you give me a reference number to the issue? Also, it is a problem with 
> all releases of asterisk.
> 
> On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas  wrote:
> 
> From: asterisk-users-boun...@lists.digium.com 
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
> Sent: Wednesday, February 23, 2011 10:11 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [asterisk-users] REFER and dialplan broken (as documented 
> inchan_sip.c on line 11951)
> 
>  
> 
> There is a problem when transferring calls using REFER, asterisk does not 
> notify dialplan. I've been told to use AMI as a workaround to notify my 
> dialplan/routing program but that would require a huge change to our 
> software. I was wondering if there is any intention of fixing this problem.
> 
> Here is issue as stated in chan_sip.c
> 
> "this is currently broken as we have no way of telling the dialplan engine 
> whether a transfer succeeds or fails."
> 
> Thanks.
> 
>  
> 
> I’m quite certain that this problem is being considered (for reference, this 
> is a 1.8.X issue).  If you aren’t satisfied with the progress being made, you 
> should research your own solution and/or offer a bounty.
> 
> 
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> _
> -- 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.di

Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread vip killa

I recognize all the options given yet as I explained before they are not
viable. I do not have the resources to pay someone, I do not have the
expertise to fix this issue because according to an asterisk developer "any
fix in that area would be deeply architectural in nature"... what other
options are there?

On Wed, Feb 23, 2011 at 1:38 PM, Watkins, Bradley <
bradley.watk...@compuware.com> wrote:

> You are still focusing on ONE of the choices given when that isn’t your
> only option.  It is simply untrue to say that the answer to “it’s broken”
> was “pay us”.  You were (now on multiple occasions) told how it would come
> to pass that a resolution will come about.  You choose to ignore precisely
> two-thirds of the options available to you in order to continue to grind
> your axe.
>
>
>
> I am convinced you are either trolling or simply myopic.  You have choices,
> they are yours to make.  Stop trying to say that the entire Asterisk
> development community is simply in it for money, because that is patently
> false.
>
>
>
> - Brad
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 1:28 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> It's simple, if a product is broken shouldn't it be fixed? In this case the
> answer is "for a price" which is absurd because it is an open source
> product. If there was a decent community of developers surrounding this
> "open source project", it would be fixed simply because it's broken, no
> questions asked.
>
> On Wed, Feb 23, 2011 at 1:19 PM, Watkins, Bradley <
> bradley.watk...@compuware.com> wrote:
>
> Implying that the Asterisk developers (which is itself a fairly nebulous
> statement since those who contribute to Asterisk are many and come from
> different companies/countries/etc.) are “not in it to make a good product”
> but to make a “profit” is not only highly insulting but a complete
> mischaracterization of what you were told on IRC.
>
>
>
> What you were told was that there are essentially three choices (and this
> goes for pretty much any open source software, as already stated).
>
>
>
> You may either fix it yourself (if you have the skills), pay someone to fix
> it for you (if you can or must trade money for expediency), or wait for
> someone else with the skills and/or money necessary to fix it.
>
>
>
> Regards,
>
> - Brad
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 1:05 PM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> Yes, they want money, they've told me that several times...it's unfortunate
> that asterisk's dev community is not in it to make a good product but a
> profit
>
> On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas  wrote:
>
> My bad – “natively” means using the Queue command from the dialplan.  Since
> the “powers that be” are aware of this problem,  I suppose it will get fixed
> when somebody either has some spare time or a sufficient bounty is offered.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 11:57 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> I'm sorry i don't know what you mean by natively. I'm almost certain the
> queue is handled via AGI and not using asterisk's queue.
>
> On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas 
> wrote:
>
> Do you use the Queue command “natively” or from the AGI?  In the example
> you gave, if you did a “core show channels”, I assume that Agent007 would be
> idle, but ineligible for Queue activity.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 11:37 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> Sure, it really manifests itself whenever using AGI for call flow, but this
> is how it affects us...
>
> incoming call -> queue -> agent007 -> xfer -> pussygalore
>
> now the AGI/dialplan thinks agent007 is on phone with pussygalore until
> that xfered call terminates so if another call comes into queue while
> pussygalore is on the phone w/ that xfered call, agent007 will not even be
> attempted by queue
>
>
>
> I'm sure the

Re: [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card

2011-02-23 Thread Greg Woods

On Wed, 2011-02-23 at 09:56 -0500, C F wrote:
> This is the closest thing I was able to find in my wctdm.c file:
> if ((blah & 0xf) == 2) {
> /* ProSLIC 3215, not a 3210 */
> wc->flags[card] |= FLAG_3215;
> }
> If I take out the 2 first lines I get errors when compling.


Maybe you need to remove the closing brace too?

--Greg




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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Watkins, Bradley

You are still focusing on ONE of the choices given when that isn't your only 
option.  It is simply untrue to say that the answer to "it's broken" was "pay 
us".  You were (now on multiple occasions) told how it would come to pass that 
a resolution will come about.  You choose to ignore precisely two-thirds of the 
options available to you in order to continue to grind your axe.

I am convinced you are either trolling or simply myopic.  You have choices, 
they are yours to make.  Stop trying to say that the entire Asterisk 
development community is simply in it for money, because that is patently false.

- Brad

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
inchan_sip.c on line 11951)

It's simple, if a product is broken shouldn't it be fixed? In this case the 
answer is "for a price" which is absurd because it is an open source product. 
If there was a decent community of developers surrounding this "open source 
project", it would be fixed simply because it's broken, no questions asked.
On Wed, Feb 23, 2011 at 1:19 PM, Watkins, Bradley 
mailto:bradley.watk...@compuware.com>> wrote:
Implying that the Asterisk developers (which is itself a fairly nebulous 
statement since those who contribute to Asterisk are many and come from 
different companies/countries/etc.) are "not in it to make a good product" but 
to make a "profit" is not only highly insulting but a complete 
mischaracterization of what you were told on IRC.

What you were told was that there are essentially three choices (and this goes 
for pretty much any open source software, as already stated).

You may either fix it yourself (if you have the skills), pay someone to fix it 
for you (if you can or must trade money for expediency), or wait for someone 
else with the skills and/or money necessary to fix it.

Regards,
- Brad

From: 
asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 1:05 PM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
inchan_sip.c on line 11951)

Yes, they want money, they've told me that several times...it's unfortunate 
that asterisk's dev community is not in it to make a good product but a profit
On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas 
mailto:da...@debsinc.com>> wrote:
My bad - "natively" means using the Queue command from the dialplan.  Since the 
"powers that be" are aware of this problem,  I suppose it will get fixed when 
somebody either has some spare time or a sufficient bounty is offered.


From: 
asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 11:57 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
inchan_sip.c on line 11951)

I'm sorry i don't know what you mean by natively. I'm almost certain the queue 
is handled via AGI and not using asterisk's queue.
On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas 
mailto:da...@debsinc.com>> wrote:
Do you use the Queue command "natively" or from the AGI?  In the example you 
gave, if you did a "core show channels", I assume that Agent007 would be idle, 
but ineligible for Queue activity.


From: 
asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 11:37 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
inchan_sip.c on line 11951)

Sure, it really manifests itself whenever using AGI for call flow, but this is 
how it affects us...
incoming call -> queue -> agent007 -> xfer -> pussygalore
now the AGI/dialplan thinks agent007 is on phone with pussygalore until that 
xfered call terminates so if another call comes into queue while pussygalore is 
on the phone w/ that xfered call, agent007 will not even be attempted by queue

I'm sure there could be other scenarios in which this REFER issue could pose a 
problem but this is the most consequential scenario which we have to deal with 
everyday.


On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas 
mailto:da...@debsinc.com>> wrote:
I use Polycom 501's and use the Transfer Key to send inbound c

Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread vip killa

Actually from what I understand Asterisk is the only product that has this
REFER problem. I know for a fact FreeSWITCH (open-source) handles REFERs
fine.

On Wed, Feb 23, 2011 at 1:28 PM, Danny Nicholas  wrote:

>  
>
> Asterisk is (IMO) a very good product.  It is NOT a perfect product, but
> I’m sure that most if not all of the Commercial PBX products available are
> not either.  You get what you pay for;  In this case, you pay in time
> instead of actual cash (unless you use the commercial flavor of Asterisk).
> It all boils down to what you need and what you are willing to do/pay to get
> that.
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Danny Nicholas



Asterisk is (IMO) a very good product.  It is NOT a perfect product, but I'm
sure that most if not all of the Commercial PBX products available are not
either.  You get what you pay for;  In this case, you pay in time instead of
actual cash (unless you use the commercial flavor of Asterisk). It all boils
down to what you need and what you are willing to do/pay to get that.

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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread vip killa

It's simple, if a product is broken shouldn't it be fixed? In this case the
answer is "for a price" which is absurd because it is an open source
product. If there was a decent community of developers surrounding this
"open source project", it would be fixed simply because it's broken, no
questions asked.

On Wed, Feb 23, 2011 at 1:19 PM, Watkins, Bradley <
bradley.watk...@compuware.com> wrote:

> Implying that the Asterisk developers (which is itself a fairly nebulous
> statement since those who contribute to Asterisk are many and come from
> different companies/countries/etc.) are “not in it to make a good product”
> but to make a “profit” is not only highly insulting but a complete
> mischaracterization of what you were told on IRC.
>
>
>
> What you were told was that there are essentially three choices (and this
> goes for pretty much any open source software, as already stated).
>
>
>
> You may either fix it yourself (if you have the skills), pay someone to fix
> it for you (if you can or must trade money for expediency), or wait for
> someone else with the skills and/or money necessary to fix it.
>
>
>
> Regards,
>
> - Brad
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 1:05 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> Yes, they want money, they've told me that several times...it's unfortunate
> that asterisk's dev community is not in it to make a good product but a
> profit
>
> On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas  wrote:
>
> My bad – “natively” means using the Queue command from the dialplan.  Since
> the “powers that be” are aware of this problem,  I suppose it will get fixed
> when somebody either has some spare time or a sufficient bounty is offered.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 11:57 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> I'm sorry i don't know what you mean by natively. I'm almost certain the
> queue is handled via AGI and not using asterisk's queue.
>
> On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas 
> wrote:
>
> Do you use the Queue command “natively” or from the AGI?  In the example
> you gave, if you did a “core show channels”, I assume that Agent007 would be
> idle, but ineligible for Queue activity.
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 11:37 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> Sure, it really manifests itself whenever using AGI for call flow, but this
> is how it affects us...
>
> incoming call -> queue -> agent007 -> xfer -> pussygalore
>
> now the AGI/dialplan thinks agent007 is on phone with pussygalore until
> that xfered call terminates so if another call comes into queue while
> pussygalore is on the phone w/ that xfered call, agent007 will not even be
> attempted by queue
>
>
>
> I'm sure there could be other scenarios in which this REFER issue could
> pose a problem but this is the most consequential scenario which we have to
> deal with everyday.
>
>
>
>
>
> On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas 
> wrote:
>
> I use Polycom 501’s and use the Transfer Key to send inbound calls to other
> extensions.  Can you give me an A-B-C example of how this problem manifests
> itself?
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 11:11 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> Interesting but the issue I'm having relates to Inbound and Outbound REFERs
> since I'm using Polycom's Transfer softkey (which allows for both Inbound
> and Outbound Transfers). I know this is not an issue when using Asterisk's
> built-in transfer (only allows Inbound transfers).
>
>
>
> On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas 
> wrote:
>
> Have you read this thread?
>
> http://forums.digium.com/viewtopic.php?t=74418
>
>
>
>
> --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, Febru

Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Watkins, Bradley

Implying that the Asterisk developers (which is itself a fairly nebulous 
statement since those who contribute to Asterisk are many and come from 
different companies/countries/etc.) are "not in it to make a good product" but 
to make a "profit" is not only highly insulting but a complete 
mischaracterization of what you were told on IRC.

What you were told was that there are essentially three choices (and this goes 
for pretty much any open source software, as already stated).

You may either fix it yourself (if you have the skills), pay someone to fix it 
for you (if you can or must trade money for expediency), or wait for someone 
else with the skills and/or money necessary to fix it.

Regards,
- Brad

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 1:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
inchan_sip.c on line 11951)

Yes, they want money, they've told me that several times...it's unfortunate 
that asterisk's dev community is not in it to make a good product but a profit
On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas 
mailto:da...@debsinc.com>> wrote:
My bad - "natively" means using the Queue command from the dialplan.  Since the 
"powers that be" are aware of this problem,  I suppose it will get fixed when 
somebody either has some spare time or a sufficient bounty is offered.


From: 
asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 11:57 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
inchan_sip.c on line 11951)

I'm sorry i don't know what you mean by natively. I'm almost certain the queue 
is handled via AGI and not using asterisk's queue.
On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas 
mailto:da...@debsinc.com>> wrote:
Do you use the Queue command "natively" or from the AGI?  In the example you 
gave, if you did a "core show channels", I assume that Agent007 would be idle, 
but ineligible for Queue activity.


From: 
asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 11:37 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
inchan_sip.c on line 11951)

Sure, it really manifests itself whenever using AGI for call flow, but this is 
how it affects us...
incoming call -> queue -> agent007 -> xfer -> pussygalore
now the AGI/dialplan thinks agent007 is on phone with pussygalore until that 
xfered call terminates so if another call comes into queue while pussygalore is 
on the phone w/ that xfered call, agent007 will not even be attempted by queue

I'm sure there could be other scenarios in which this REFER issue could pose a 
problem but this is the most consequential scenario which we have to deal with 
everyday.


On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas 
mailto:da...@debsinc.com>> wrote:
I use Polycom 501's and use the Transfer Key to send inbound calls to other 
extensions.  Can you give me an A-B-C example of how this problem manifests 
itself?


From: 
asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 11:11 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
inchan_sip.c on line 11951)

Interesting but the issue I'm having relates to Inbound and Outbound REFERs 
since I'm using Polycom's Transfer softkey (which allows for both Inbound and 
Outbound Transfers). I know this is not an issue when using Asterisk's built-in 
transfer (only allows Inbound transfers).

On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas 
mailto:da...@debsinc.com>> wrote:
Have you read this thread?
http://forums.digium.com/viewtopic.php?t=74418



From: 
asterisk-users-boun...@lists.digium.com
 
[mailto:asterisk-users-boun...@lists.digium.com]
 On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:36 AM

To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented 
inchan_sip.c on line 119

Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Steve Edwards


Un-top-posting...

On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas  
wrote:


My bad – “natively” means using the Queue command from the dialplan.  
Since the “powers that be” are aware of this problem,  I suppose it will 
get fixed when somebody either has some spare time or a sufficient 
bounty is offered.


On Wed, 23 Feb 2011, vip killa wrote:

Yes, they want money, they've told me that several times...it's 
unfortunate that asterisk's dev community is not in it to make a good 
product but a profit


And what are you 'in it' for?

The developer community is populated by many kinds of people. Some do it 
because it's their job, some do it for the challenge, some do it 'for the 
greater good' and some do it as 'a gun for hire.'


Whatever their motivation, are you receiving more than you give? My guess 
is 'yes' which makes it 'fortunate' for you and me.


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

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

Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread vip killa

Yes, they want money, they've told me that several times...it's unfortunate
that asterisk's dev community is not in it to make a good product but a
profit

On Wed, Feb 23, 2011 at 1:01 PM, Danny Nicholas  wrote:

>  My bad – “natively” means using the Queue command from the dialplan.
> Since the “powers that be” are aware of this problem,  I suppose it will get
> fixed when somebody either has some spare time or a sufficient bounty is
> offered.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 11:57 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> I'm sorry i don't know what you mean by natively. I'm almost certain the
> queue is handled via AGI and not using asterisk's queue.
>
> On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas 
> wrote:
>
> Do you use the Queue command “natively” or from the AGI?  In the example
> you gave, if you did a “core show channels”, I assume that Agent007 would be
> idle, but ineligible for Queue activity.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 11:37 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> Sure, it really manifests itself whenever using AGI for call flow, but this
> is how it affects us...
>
> incoming call -> queue -> agent007 -> xfer -> pussygalore
>
> now the AGI/dialplan thinks agent007 is on phone with pussygalore until
> that xfered call terminates so if another call comes into queue while
> pussygalore is on the phone w/ that xfered call, agent007 will not even be
> attempted by queue
>
>
>
> I'm sure there could be other scenarios in which this REFER issue could
> pose a problem but this is the most consequential scenario which we have to
> deal with everyday.
>
>
>
>
>
> On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas 
> wrote:
>
> I use Polycom 501’s and use the Transfer Key to send inbound calls to other
> extensions.  Can you give me an A-B-C example of how this problem manifests
> itself?
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 11:11 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> Interesting but the issue I'm having relates to Inbound and Outbound REFERs
> since I'm using Polycom's Transfer softkey (which allows for both Inbound
> and Outbound Transfers). I know this is not an issue when using Asterisk's
> built-in transfer (only allows Inbound transfers).
>
>
>
> On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas 
> wrote:
>
> Have you read this thread?
>
> http://forums.digium.com/viewtopic.php?t=74418
>
>
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 10:36 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> I did not see this issue anywhere on issues.asterisk.org
>
> Can you give me a reference number to the issue? Also, it is a problem with
> all releases of asterisk.
>
> On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas 
> wrote:
>   --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 10:11 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> There is a problem when transferring calls using REFER, asterisk does not
> notify dialplan. I've been told to use AMI as a workaround to notify my
> dialplan/routing program but that would require a huge change to our
> software. I was wondering if there is any intention of fixing this problem.
>
> Here is issue as stated in chan_sip.c
>
> "this is currently broken as we have no way of telling the dialplan engine
> whether a transfer succeeds or fails."
>
> Thanks.
>
>
>
> I’m quite certain that this problem is being considered (for reference,
> this is a 1.8.X issue).  If you aren’t satisfied with the progress being
> made, you should research your own solution and/or offer a bount

Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Danny Nicholas

My bad - "natively" means using the Queue command from the dialplan.  Since
the "powers that be" are aware of this problem,  I suppose it will get fixed
when somebody either has some spare time or a sufficient bounty is offered.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 11:57 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

I'm sorry i don't know what you mean by natively. I'm almost certain the
queue is handled via AGI and not using asterisk's queue. 

On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas  wrote:

Do you use the Queue command "natively" or from the AGI?  In the example you
gave, if you did a "core show channels", I assume that Agent007 would be
idle, but ineligible for Queue activity.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 11:37 AM


To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

Sure, it really manifests itself whenever using AGI for call flow, but this
is how it affects us...

incoming call -> queue -> agent007 -> xfer -> pussygalore

now the AGI/dialplan thinks agent007 is on phone with pussygalore until that
xfered call terminates so if another call comes into queue while pussygalore
is on the phone w/ that xfered call, agent007 will not even be attempted by
queue

 

I'm sure there could be other scenarios in which this REFER issue could pose
a problem but this is the most consequential scenario which we have to deal
with everyday.

 

 

On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas  wrote:

I use Polycom 501's and use the Transfer Key to send inbound calls to other
extensions.  Can you give me an A-B-C example of how this problem manifests
itself?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 11:11 AM


To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

Interesting but the issue I'm having relates to Inbound and Outbound REFERs
since I'm using Polycom's Transfer softkey (which allows for both Inbound
and Outbound Transfers). I know this is not an issue when using Asterisk's
built-in transfer (only allows Inbound transfers).

 

On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas  wrote:

Have you read this thread?

http://forums.digium.com/viewtopic.php?t=74418

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:36 AM


To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

I did not see this issue anywhere on issues.asterisk.org

Can you give me a reference number to the issue? Also, it is a problem with
all releases of asterisk.

On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas  wrote:

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

There is a problem when transferring calls using REFER, asterisk does not
notify dialplan. I've been told to use AMI as a workaround to notify my
dialplan/routing program but that would require a huge change to our
software. I was wondering if there is any intention of fixing this problem.

Here is issue as stated in chan_sip.c

"this is currently broken as we have no way of telling the dialplan engine
whether a transfer succeeds or fails."

Thanks. 

 

I'm quite certain that this problem is being considered (for reference, this
is a 1.8.X issue).  If you aren't satisfied with the progress being made,
you should research your own solution and/or offer a bounty.


--
_
-- 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

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

 


--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar

Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread vip killa

I'm sorry i don't know what you mean by natively. I'm almost certain the
queue is handled via AGI and not using asterisk's queue.

On Wed, Feb 23, 2011 at 12:52 PM, Danny Nicholas  wrote:

>  Do you use the Queue command “natively” or from the AGI?  In the example
> you gave, if you did a “core show channels”, I assume that Agent007 would be
> idle, but ineligible for Queue activity.
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 11:37 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> Sure, it really manifests itself whenever using AGI for call flow, but this
> is how it affects us...
>
> incoming call -> queue -> agent007 -> xfer -> pussygalore
>
> now the AGI/dialplan thinks agent007 is on phone with pussygalore until
> that xfered call terminates so if another call comes into queue while
> pussygalore is on the phone w/ that xfered call, agent007 will not even be
> attempted by queue
>
>
>
> I'm sure there could be other scenarios in which this REFER issue could
> pose a problem but this is the most consequential scenario which we have to
> deal with everyday.
>
>
>
>
>
> On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas 
> wrote:
>
> I use Polycom 501’s and use the Transfer Key to send inbound calls to other
> extensions.  Can you give me an A-B-C example of how this problem manifests
> itself?
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 11:11 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> Interesting but the issue I'm having relates to Inbound and Outbound REFERs
> since I'm using Polycom's Transfer softkey (which allows for both Inbound
> and Outbound Transfers). I know this is not an issue when using Asterisk's
> built-in transfer (only allows Inbound transfers).
>
>
>
> On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas 
> wrote:
>
> Have you read this thread?
>
> http://forums.digium.com/viewtopic.php?t=74418
>
>
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 10:36 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> I did not see this issue anywhere on issues.asterisk.org
>
> Can you give me a reference number to the issue? Also, it is a problem with
> all releases of asterisk.
>
> On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas 
> wrote:
>   --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 10:11 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> There is a problem when transferring calls using REFER, asterisk does not
> notify dialplan. I've been told to use AMI as a workaround to notify my
> dialplan/routing program but that would require a huge change to our
> software. I was wondering if there is any intention of fixing this problem.
>
> Here is issue as stated in chan_sip.c
>
> "this is currently broken as we have no way of telling the dialplan engine
> whether a transfer succeeds or fails."
>
> Thanks.
>
>
>
> I’m quite certain that this problem is being considered (for reference,
> this is a 1.8.X issue).  If you aren’t satisfied with the progress being
> made, you should research your own solution and/or offer a bounty.
>
>
> --
> _
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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Danny Nicholas

Do you use the Queue command "natively" or from the AGI?  In the example you
gave, if you did a "core show channels", I assume that Agent007 would be
idle, but ineligible for Queue activity.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 11:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

Sure, it really manifests itself whenever using AGI for call flow, but this
is how it affects us...

incoming call -> queue -> agent007 -> xfer -> pussygalore

now the AGI/dialplan thinks agent007 is on phone with pussygalore until that
xfered call terminates so if another call comes into queue while pussygalore
is on the phone w/ that xfered call, agent007 will not even be attempted by
queue

 

I'm sure there could be other scenarios in which this REFER issue could pose
a problem but this is the most consequential scenario which we have to deal
with everyday.

 

 

On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas  wrote:

I use Polycom 501's and use the Transfer Key to send inbound calls to other
extensions.  Can you give me an A-B-C example of how this problem manifests
itself?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 11:11 AM


To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

Interesting but the issue I'm having relates to Inbound and Outbound REFERs
since I'm using Polycom's Transfer softkey (which allows for both Inbound
and Outbound Transfers). I know this is not an issue when using Asterisk's
built-in transfer (only allows Inbound transfers).

 

On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas  wrote:

Have you read this thread?

http://forums.digium.com/viewtopic.php?t=74418

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:36 AM


To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

I did not see this issue anywhere on issues.asterisk.org

Can you give me a reference number to the issue? Also, it is a problem with
all releases of asterisk.

On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas  wrote:

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

There is a problem when transferring calls using REFER, asterisk does not
notify dialplan. I've been told to use AMI as a workaround to notify my
dialplan/routing program but that would require a huge change to our
software. I was wondering if there is any intention of fixing this problem.

Here is issue as stated in chan_sip.c

"this is currently broken as we have no way of telling the dialplan engine
whether a transfer succeeds or fails."

Thanks. 

 

I'm quite certain that this problem is being considered (for reference, this
is a 1.8.X issue).  If you aren't satisfied with the progress being made,
you should research your own solution and/or offer a bounty.


--
_
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  http://lists.digium.com/mailman/listinfo/asterisk-users

 


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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread vip killa

Sure, it really manifests itself whenever using AGI for call flow, but this
is how it affects us...
incoming call -> queue -> agent007 -> xfer -> pussygalore
now the AGI/dialplan thinks agent007 is on phone with pussygalore until that
xfered call terminates so if another call comes into queue while pussygalore
is on the phone w/ that xfered call, agent007 will not even be attempted by
queue

I'm sure there could be other scenarios in which this REFER issue could pose
a problem but this is the most consequential scenario which we have to deal
with everyday.


On Wed, Feb 23, 2011 at 12:23 PM, Danny Nicholas  wrote:

>  I use Polycom 501’s and use the Transfer Key to send inbound calls to
> other extensions.  Can you give me an A-B-C example of how this problem
> manifests itself?
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 11:11 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> Interesting but the issue I'm having relates to Inbound and Outbound REFERs
> since I'm using Polycom's Transfer softkey (which allows for both Inbound
> and Outbound Transfers). I know this is not an issue when using Asterisk's
> built-in transfer (only allows Inbound transfers).
>
>
>
> On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas 
> wrote:
>
> Have you read this thread?
>
> http://forums.digium.com/viewtopic.php?t=74418
>
>
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 10:36 AM
>
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> I did not see this issue anywhere on issues.asterisk.org
>
> Can you give me a reference number to the issue? Also, it is a problem with
> all releases of asterisk.
>
> On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas 
> wrote:
>   --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 10:11 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> There is a problem when transferring calls using REFER, asterisk does not
> notify dialplan. I've been told to use AMI as a workaround to notify my
> dialplan/routing program but that would require a huge change to our
> software. I was wondering if there is any intention of fixing this problem.
>
> Here is issue as stated in chan_sip.c
>
> "this is currently broken as we have no way of telling the dialplan engine
> whether a transfer succeeds or fails."
>
> Thanks.
>
>
>
> I’m quite certain that this problem is being considered (for reference,
> this is a 1.8.X issue).  If you aren’t satisfied with the progress being
> made, you should research your own solution and/or offer a bounty.
>
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
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>
>
>
> --
> _
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>
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Re: [asterisk-users] calls between iax and sip

2011-02-23 Thread Steve Edwards


On Wed, 23 Feb 2011, salaheddine elharit wrote:


  == Agent '1018' logged in (format ulaw/slin)


An agent is not the same as an extension.

but when i call from sip extension 106 to iax extension (1018) i got the 
message below


[Feb 23 09:55:49] NOTICE[25420]: chan_sip.c:13952 handle_request_invite: 
Call from '106' to extension '1018' rejected because extension not 
found.


This NOTICE is from the SIP channel driver, not the IAX channel driver.

What does the dial statement that generates the above NOTICE look like?

What does 'iax2 show peer 1018' display?

--
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] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Danny Nicholas

I use Polycom 501's and use the Transfer Key to send inbound calls to other
extensions.  Can you give me an A-B-C example of how this problem manifests
itself?

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 11:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

Interesting but the issue I'm having relates to Inbound and Outbound REFERs
since I'm using Polycom's Transfer softkey (which allows for both Inbound
and Outbound Transfers). I know this is not an issue when using Asterisk's
built-in transfer (only allows Inbound transfers).

 

On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas  wrote:

Have you read this thread?

http://forums.digium.com/viewtopic.php?t=74418

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:36 AM


To: Asterisk Users Mailing List - Non-Commercial Discussion

Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

I did not see this issue anywhere on issues.asterisk.org

Can you give me a reference number to the issue? Also, it is a problem with
all releases of asterisk.

On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas  wrote:

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

There is a problem when transferring calls using REFER, asterisk does not
notify dialplan. I've been told to use AMI as a workaround to notify my
dialplan/routing program but that would require a huge change to our
software. I was wondering if there is any intention of fixing this problem.

Here is issue as stated in chan_sip.c

"this is currently broken as we have no way of telling the dialplan engine
whether a transfer succeeds or fails."

Thanks. 

 

I'm quite certain that this problem is being considered (for reference, this
is a 1.8.X issue).  If you aren't satisfied with the progress being made,
you should research your own solution and/or offer a bounty.


--
_
-- 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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To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

 


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  http://www.asterisk.org/hello

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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread vip killa

Interesting but the issue I'm having relates to Inbound and Outbound REFERs
since I'm using Polycom's Transfer softkey (which allows for both Inbound
and Outbound Transfers). I know this is not an issue when using Asterisk's
built-in transfer (only allows Inbound transfers).


On Wed, Feb 23, 2011 at 12:01 PM, Danny Nicholas  wrote:

>  Have you read this thread?
>
> http://forums.digium.com/viewtopic.php?t=74418
>
>
>
>
>  --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 10:36 AM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> I did not see this issue anywhere on issues.asterisk.org
>
> Can you give me a reference number to the issue? Also, it is a problem with
> all releases of asterisk.
>
> On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas 
> wrote:
>   --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 10:11 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> There is a problem when transferring calls using REFER, asterisk does not
> notify dialplan. I've been told to use AMI as a workaround to notify my
> dialplan/routing program but that would require a huge change to our
> software. I was wondering if there is any intention of fixing this problem.
>
> Here is issue as stated in chan_sip.c
>
> "this is currently broken as we have no way of telling the dialplan engine
> whether a transfer succeeds or fails."
>
> Thanks.
>
>
>
> I’m quite certain that this problem is being considered (for reference,
> this is a 1.8.X issue).  If you aren’t satisfied with the progress being
> made, you should research your own solution and/or offer a bounty.
>
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
> --
> _
> -- 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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> To UNSUBSCRIBE or update options visit:
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>
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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Danny Nicholas

Have you read this thread?

http://forums.digium.com/viewtopic.php?t=74418

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

I did not see this issue anywhere on issues.asterisk.org

Can you give me a reference number to the issue? Also, it is a problem with
all releases of asterisk.

On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas  wrote:

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

There is a problem when transferring calls using REFER, asterisk does not
notify dialplan. I've been told to use AMI as a workaround to notify my
dialplan/routing program but that would require a huge change to our
software. I was wondering if there is any intention of fixing this problem.

Here is issue as stated in chan_sip.c

"this is currently broken as we have no way of telling the dialplan engine
whether a transfer succeeds or fails."

Thanks. 

 

I'm quite certain that this problem is being considered (for reference, this
is a 1.8.X issue).  If you aren't satisfied with the progress being made,
you should research your own solution and/or offer a bounty.


--
_
-- 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

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

 

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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Danny Nicholas

Mea Culpa - I see this in my 1.4.37 source as well (line 8401 in this
release chan_sip.c).  Hopefully someone like Tilghman will address this;  a
"simple hack" would be to create a C daemon that did a "core show channels"
and transmit to appropriate results back for referral.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:36 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

I did not see this issue anywhere on issues.asterisk.org

Can you give me a reference number to the issue? Also, it is a problem with
all releases of asterisk.

On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas  wrote:

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

 

There is a problem when transferring calls using REFER, asterisk does not
notify dialplan. I've been told to use AMI as a workaround to notify my
dialplan/routing program but that would require a huge change to our
software. I was wondering if there is any intention of fixing this problem.

Here is issue as stated in chan_sip.c

"this is currently broken as we have no way of telling the dialplan engine
whether a transfer succeeds or fails."

Thanks. 

 

I'm quite certain that this problem is being considered (for reference, this
is a 1.8.X issue).  If you aren't satisfied with the progress being made,
you should research your own solution and/or offer a bounty.


--
_
-- 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

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

 

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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread vip killa

I did not see this issue anywhere on issues.asterisk.org
Can you give me a reference number to the issue? Also, it is a problem with
all releases of asterisk.

On Wed, Feb 23, 2011 at 11:28 AM, Danny Nicholas  wrote:

>   --
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa
> *Sent:* Wednesday, February 23, 2011 10:11 AM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] REFER and dialplan broken (as documented
> inchan_sip.c on line 11951)
>
>
>
> There is a problem when transferring calls using REFER, asterisk does not
> notify dialplan. I've been told to use AMI as a workaround to notify my
> dialplan/routing program but that would require a huge change to our
> software. I was wondering if there is any intention of fixing this problem.
>
> Here is issue as stated in chan_sip.c
>
> "this is currently broken as we have no way of telling the dialplan engine
> whether a transfer succeeds or fails."
>
> Thanks.
>
>
>
> I’m quite certain that this problem is being considered (for reference,
> this is a 1.8.X issue).  If you aren’t satisfied with the progress being
> made, you should research your own solution and/or offer a bounty.
>
> --
> _
> -- 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
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
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Re: [asterisk-users] REFER and dialplan broken (as documented inchan_sip.c on line 11951)

2011-02-23 Thread Danny Nicholas

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa
Sent: Wednesday, February 23, 2011 10:11 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] REFER and dialplan broken (as documented
inchan_sip.c on line 11951)

There is a problem when transferring calls using REFER, asterisk does not
notify dialplan. I've been told to use AMI as a workaround to notify my
dialplan/routing program but that would require a huge change to our
software. I was wondering if there is any intention of fixing this problem.

Here is issue as stated in chan_sip.c

"this is currently broken as we have no way of telling the dialplan engine
whether a transfer succeeds or fails."

Thanks. 

I'm quite certain that this problem is being considered (for reference, this
is a 1.8.X issue).  If you aren't satisfied with the progress being made,
you should research your own solution and/or offer a bounty.

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[asterisk-users] REFER and dialplan broken (as documented in chan_sip.c on line 11951)

2011-02-23 Thread vip killa

There is a problem when transferring calls using REFER, asterisk does not
notify dialplan. I've been told to use AMI as a workaround to notify my
dialplan/routing program but that would require a huge change to our
software. I was wondering if there is any intention of fixing this problem.
Here is issue as stated in chan_sip.c
"this is currently broken as we have no way of telling the dialplan engine
whether a transfer succeeds or fails."
Thanks.
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[asterisk-users] secret vs remotesecret on outgoing calls in Asterisk 1.6.2.16.1

2011-02-23 Thread Jose P. Espinal


Hello List,

I have a little issue with calls placed to a provider declared on 
sip.conf, because of a not clear (*for me*) behavior of 'remotesecret' 
parameter.


Before continuing, this is my environment:

Asterisk:  1.6.2.16.1
OS:CentOS release 5.5 (Final)
   2.6.18-194.32.1.el5


Details:

I have this block on sip.conf

- start 
...
register => john:j0nhp...@66.128.xx.xxx
...

[john-peer]
type=peer
defaultuser=john
remotesecret=j0nhp4ss
;secret=j0nhp4ss
host=66.128.XX.XXX
directmedia=no
dtmfmode=rfc2833
context=jonh-context

- end 


When I send a call to that block, I receive the following response 
unless (I explicitly indicate a 'secret' parameter, no matter if 
'remotesecret' parameter was indicated):


"Forbidden" from '"Test Account" ;tag=as749a7ced'



If I set the 'secret' parameter, everything goes smoothly as expected.


Maybe I'm obviating something 'basic', but the CHANGES file says:

-
Added a new configuration option "remotesecret" for authentication to
remote services. For backwards compatibility, "secret" still has the
same function as before, but now you can configure both a remote secret
and a local secret for mutual authentication.
-

and on sip.conf.sample

-
;remotesecret=guessit ; Our password to their service
-

I thought that 'remotesecret' is used to authenticate myself when 
placing a call to the remote network, as I used to do with 'secret' 
parameter.



Doing a: grep -ir 'remotesecret' . (inside the Asterisk source 
directory) indicates that only this files mention that parameter:


./ChangeLog
./channels/chan_sip.c
./CHANGES:
./configs/sip.conf.sample


Could someone please point me to documentation regarding this two 
parameters?



Thanks in advice.


--
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http://www.eslackware.com
IRC: Khratos @ #asterisk / -doc / -bugs

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Re: [asterisk-users] AMI FullyBooted issue

2011-02-23 Thread Ishfaq Malik

On Wed, 2011-02-23 at 09:37 -0500, Paul Belanger wrote:
> On 11-02-23 05:39 AM, Ishfaq Malik wrote:
> > Has anyone else experienced anything like this?
> > 
> There is a patch on the issue tracker[1], please test it out and report
> your feedback.
> 
> [1] https://issues.asterisk.org/view.php?id=18168
> 
Hi Paul

Thanks for that. Unfortunately I'm on strict instructions to use rpm
packages only so can't test your patch.

However, the eventfilter work around solves my problem so thanks for
bringing it to my attention.

Ish
-- 
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Office:   0161 660 3062


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Re: [asterisk-users] Problem installing FXS module in old digium 4 channel tdm card

2011-02-23 Thread C F

This is the closest thing I was able to find in my wctdm.c file:
if ((blah & 0xf) == 2) {
/* ProSLIC 3215, not a 3210 */
wc->flags[card] |= FLAG_3215;
}
If I take out the 2 first lines I get errors when compling.


On Tue, Feb 22, 2011 at 11:43 PM, Shaun Ruffell  wrote:
> On Tue, Feb 22, 2011 at 11:00:22PM -0500, C F wrote:
>> On Mon, Feb 21, 2011 at 11:23 PM, Shaun Ruffell  wrote:
>> > On 2/21/11 4:46 PM, C F wrote:
>> >> I just installed an FXS module onto a 4 channel tdm thats about 5
>> >> years old and it wont work. Running dmesg I can see the following
>> >> error:
>> >>
>
> [snip]
>
>> >>  ! Init Indirect Registers UNSUCCESSFULLY.
>> >> Indirect Registers failed verification.
>> >> Module 0: FAILED FXS (FCC)
>> >> Module 1: Installed -- AUTO FXO (FCC mode)
>> >> Module 2: Installed -- AUTO FXO (FCC mode)
>> >> Module 3: Installed -- AUTO FXO (FCC mode)
>> >> Found a Wildcard TDM: Wildcard TDM400P REV E/F (3 modules)
>> >>
>> >> Does this have to do with the fact that the module is way newer than the
>> >> card?
>> >>
>> >
>> > Not having much direct experience with the wctdm.c driver, that would be my
>> > guess. You might be able to edit the wctdm_proslic_insane() function to
>> > force the FLAG_3215 on for the card and see if that gives you a different
>> > result.
>> >
>>
>> How/Where would I do that?
>>
>
> Around line 1297 of drivers/dahdi/wctdm.c you could change:
>
>  if (wctdm_getreg(wc, card, 1) & 0x80)
>                /* ProSLIC 3215, not a 3210 */
>                wc->flags[card] |= FLAG_3215;
>
>  to
>
>  wc->flags[card] |= FLAG_3215;
>
> and just skip the read of register 1. I don't know if this will work in your
> case, but it's something to try.
>
> --
> Shaun Ruffell
> Digium, Inc. | Linux Kernel Developer
> 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] AMI FullyBooted issue

2011-02-23 Thread Paul Belanger

On 11-02-23 05:39 AM, Ishfaq Malik wrote:
> Has anyone else experienced anything like this?
> 
There is a patch on the issue tracker[1], please test it out and report
your feedback.

[1] https://issues.asterisk.org/view.php?id=18168

-- 
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Digium, Inc. | Software Developer
twitter: pabelanger | IRC: pabelanger (Freenode)
Check us out at: http://digium.com & http://asterisk.org

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[asterisk-users] Adhearsion 1.0.1 Released

2011-02-23 Thread Ben Klang

The Adhearsion team announces the release of Adhearsion version 1.0.1. 
Adhearsion is an open source Ruby-language framework for creating telephony 
applications.  This update primarily addresses compatibility with newer 
versions of other software but also adds native support for Bundler to newly 
created Adhearsion applications.

Here are some highlights from the changelog:

Handling of new Asterisk 1.6/1.8 events
Improved control of Asterisk Queues
Two new dialplan methods have been added: say_chars and say_phonetic
Ruby 1.9 is now an officially supported platform
Fix compatibility with Rails 3
Bundler now included by default for new Adhearsion applications
Not bad for a dot release!  You can read the full CHANGELOG here.

As always I'd like to thank the Adhearsion community for their contributions to 
this release.  Special thanks to contributors Ben Langfeld, Robert Jackson and 
Matthew Clark.

To install Adhearsion just type gem install adhearsion at your nearest command 
prompt.  For help getting started, checkout our Wiki and Getting Started pages. 
 As always, you can find us on irc.freenode.net #adhearsion or our Google 
Groups mailing list.  Contributors welcome!  Check out the sources on 
Adhearsion's Github.


/BAK/
-- 
Ben Klang
bkl...@mojolingo.com
404.475.4841

Mojo Lingo -- Voice applications that work like magic
http://mojolingo.com
Twitter: @mojolingo--
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[asterisk-users] AMI FullyBooted issue

2011-02-23 Thread Ishfaq Malik

Hi

We're still testing out asterisk 1.8 (using 1.8.2.2 from rpm package)
before putting it into production and I'm observing an odd issue when
using the AMI

Every request I send to the AMI just results in a FullyBooted response
rather than the expected response. Here are some examples from my logs

-- Call started: 22/02/2011 11:34:03 --
action: command
command: core show channels

Event: FullyBooted
Privilege: system,all
SequenceNumber: 1706
File: manager.c
Line: 2937
Func: action_login
Status: Fully Booted


-- Call started: 22/02/2011 10:28:15 --
action: command
command: sip show peers

Event: FullyBooted
Privilege: system,all
SequenceNumber: 1610
File: manager.c
Line: 2937
Func: action_login
Status: Fully Booted


Has anyone else experienced anything like this?


-- 
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Software Developer
PackNet Ltd

Office:   0161 660 3062


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[asterisk-users] extend the timout on ringing for pri or sip

2011-02-23 Thread Israel Gottlieb

Hi

Does anyone know how i could extend the timer for the ringing time on a pri
or sip trunk ?
Today the call gets a cancel request after a minute if not answerd yet
is it on asterisk or is a provider side setting?
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Re: [asterisk-users] calls between iax and sip

2011-02-23 Thread salaheddine elharit

Thanks steve for your response



the details is below



When i call from iax extension (1018) to sip extension there is no issue

  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4-r110474M-AheevaCCS-3.1.0_Build-11629 currently
running on srvradio (pid = 24818)
Verbosity is at least 3
-- Accepting UNAUTHENTICATED call from 192.168.5.131:
   > requested format = ulaw,
   > requested prefs = (),
   > actual format = ulaw,
   > host prefs = (alaw|ulaw),
   > priority = mine
-- Executing [MCALL106^1298455141.287500@agents:1] Set("IAX2/1018-6",
"AH_TEMP=106^1298455141.287500") in new stack
-- Executing [MCALL106^1298455141.287500@agents:2] NoOp("IAX2/1018-6",
"[106^1298455141.287500]") in new stack
-- Executing [MCALL106^1298455141.287500@agents:3] Set("IAX2/1018-6",
"AH_EXTEN=106") in new stack
-- Executing [MCALL106^1298455141.287500@agents:4] Set("IAX2/1018-6",
"AHEEVA_TRACKNUM=1298455141.287500") in new stack
-- Executing [MCALL106^1298455141.287500@agents:5] Goto("IAX2/1018-6",
"agents|106|1") in new stack
-- Goto (agents,106,1)
-- Executing [106@agents:1] Dial("IAX2/1018-6", "SIP/106") in new stack
-- Called 106
-- SIP/106-095133e8 is ringing
-- SIP/106-095133e8 answered IAX2/1018-6
  == Agent '1018' logged out
  == Spawn extension (agents, AH1018, 1) exited non-zero on 'IAX2/1018-4'
  == Spawn extension (agents, 106, 1) exited non-zero on 'IAX2/1018-6'
-- Executing [h@agents:1] GotoIf("IAX2/1018-4", "0?3:2") in new stack
-- Executing [h@agents:1] GotoIf("IAX2/1018-6", "1?3:2") in new stack
-- Goto (agents,h,2)
-- Executing [h@agents:2] AHEventsProxy("IAX2/1018-4",
"MSG_TYPE_TERMINATE_CALL1298455155") in new stack
 AHEventsProxy: Channel [IAX2/1018-4]. Data
[MSG_TYPE_TERMINATE_CALL1298455155]
-- chan is IAX2/1018-4
 AHEventsProxy: Send To CtiServer: socket:[67].
message:[41,1298455155Ipbx01^~]
-- Executing [h@agents:3] Hangup("IAX2/1018-4", "") in new stack
  == Spawn extension (agents, h, 3) exited non-zero on 'IAX2/1018-4'
-- Hungup 'IAX2/1018-4'
-- Goto (agents,h,3)
-- Executing [h@agents:3] Hangup("IAX2/1018-6", "") in new stack
  == Spawn extension (agents, h, 3) exited non-zero on 'IAX2/1018-6'
-- Hungup 'IAX2/1018-6'
-- Accepting UNAUTHENTICATED call from 192.168.5.131:
   > requested format = ulaw,
   > requested prefs = (),
   > actual format = ulaw,
   > host prefs = (alaw|ulaw),
   > priority = mine
-- Executing [AH1018@agents:1] AgentLogin("IAX2/1018-9", "1018|s") in
new stack
-- Started music on hold, class 'none', on channel 'IAX2/1018-9'
  == Agent '1018' logged in (format ulaw/slin)
-- Stopped music on hold on IAX2/1018-9
[Feb 23 09:59:22] NOTICE[25420]: chan_sip.c:15012 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 106
srvradio*CLI>
but when i call from sip extension 106 to iax extension (1018) i got the
message below

  == Parsing '/etc/asterisk/asterisk.conf': Found
  == Parsing '/etc/asterisk/extconfig.conf': Found
Connected to Asterisk 1.4-r110474M-AheevaCCS-3.1.0_Build-11629 currently
running on srvradio (pid = 24818)
Verbosity is at least 3
[Feb 23 09:55:49] NOTICE[25420]: chan_sip.c:13952 handle_request_invite:
Call from '106' to extension '1018' rejected because extension not found.
srvradio*CLI>

thank you for your help

2011/2/22 Danny Nicholas 

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve
> Edwards
> Sent: Tuesday, February 22, 2011 12:33 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] calls between iax and sip
>
> On Tue, 22 Feb 2011, salaheddine elharit wrote:
>
> > i have asterisk installed and i have configured a client iax and sip
> > without any issue, when i call a internal extension sip from iax there
> > is no problem
> >
> > but when i call a iax extension from sip extension the result is
> > KO(wrong number)
> >
> > any help please
>
> No details, no help.
>
> Crank up verbosity on the CLI and see if the messages yield a clue. If
> not, please post the console messages.
>
> Isn't Dionne Warrick a poster on this list? :)
>
>
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Re: [asterisk-users] [1.4.39.1/AGI] ast_carefulwrite: write() returned error: Broken pipe

2011-02-23 Thread Gilles

On Tue, 22 Feb 2011 16:33:05 -0800 (PST), Steve Edwards
 wrote:
>While the documentation on the protocol is clear, nobody gets it right the 
>first time -- which is why I always suggest using an established library 
>for the language of your choice.

Indeed, neither the 2nd nor the 3rd edition of the Asterisk book make
it clear that an AGI script _must_ read all data from stdin before
going ahead.

Since it's a pretty simple protocol, it doesn't look like there's a
Lua library to handle AGI scripts.

Thank you.

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