Re: [asterisk-users] How to play different different hold music.

[akhilesh chand]

hi,

Server A

extentsion.conf

exten => N,n,Set(Service_name="Test")
exten => N,n,Dial(IAX2/
server2:server2@192.168.14.112/${result},${Service_name})

but Server B doesn't identify service_name.






Server B

iax.conf

[general]
register => server1:server1@192.168.14.110


[server2]
type=friend
user=server2
secret=server2
host=dynamic
context=outgoing
auth=md5
trunk=yes




extentsion.conf

[outgoing]

exten => _X.,1,Set(_CALLTIME=${STRFTIME(,Asia/Calcutta,%d-%b-%y-%H-%M-%S)})
exten => _X.,n,Set(CHANNEL(musicclass)=${Service_name})
exten =>
_X.,n,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})
exten => _X.,n,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})
exten => _X.,n,Set(FILENAME=${EXTEN}_${CALLERID(num)}_${CALLTIME}.WAV)
exten => _X.,n,Set(RECORDFILENAME=${RECSUBDIR}/${FILENAME})
exten => _X.,n,MixMonitor(${RECORDING_PATH}${RECORDFILENAME})
exten => _X.,n,Dial(SIP/${EXTEN},120);EXTEN=4004,4005,4006
exten => _X.,n,Hangup()



sip.conf



[4004]
type=friend
context=outbound
defaultuser=4004
secret=4004
callerid="EXT4"
host=dynamic
nat=no
dtfmode=rfc2833
subscribecontext=outbound
canreinvite=no

[4005]
type=friend
context=outbound
defaultuser=4005
secret=4005
callerid="EXT5"
host=dynamic
nat=no
dtfmode=rfc2833
subscribecontext=outbound
canreinvite=no

[4006]
type=friend
context=outbound
defaultuser=4006
secret=4006
callerid="EXT6"
host=dynamic
nat=no
dtfmode=rfc2833
subscribecontext=outbound
canreinvite=no


[ccm100]
type = friend
context = outgoing
host = 192.168.14.91
disallow = all
allow = ulaw
allow = alaw
nat=yes
canreinvite = yes
qualify = yes

On Tue, Jul 3, 2012 at 7:46 PM, Danny Nicholas  wrote:

>  Since you’re using IAX2 to contact Server B, you can use channel
> variables to control the moh class.  There was a good thread in June on
> this.  An “easier” way however would be to have each service dial a
> different IAX number, then have each IAX number on server B select it’s MOH
> Class.
>
> Server A
>
> [service1]
>
> Exten => N,1,Set(Service_name=service1)
>
> Exten => N,n,Dial(IAX2,server2:1234)
>
> [service2]
>
> Exten => N,1,Set(Service_name=service2)
>
> Exten => N,n,Dial(IAX2,server2:3456)
>
> ** **
>
> Server B
>
> [default]
>
> Exten => N,1,Verbose(start)
>
> Exten => N,1234,answer()
>
> Exten => N,n,Set(MOHClass=1)
>
> Exten => N,3456,answer()
>
> Exten => N,n,Set(MOHClass=2)
>
> ** **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *akhilesh chand
> *Sent:* Tuesday, July 03, 2012 9:11 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] How to play different different hold
> music.
>
> ** **
>
>  
>
> hi,
>
>  
>
> Server A > extentsion.conf
>
>  
>
> exten => N,n,Set(Service_name="Test")
>
> exten => N,n,Dial(IAX2/
> server2:server2@192.168.14.112/${result},${Service_name}
> )
>
>  
>
> but Server B doesn't identify service_name.
>
>  
>
>  
>
> extentsion.conf
>
>  
>
> [outgoing]
>
> exten => _X.,1,Set(_CALLTIME=${STRFTIME(,Asia/Calcutta,%d-%b-%y-%H-%M-%S)})
> 
>
> exten => _X.,1,Set(CHANNEL(musicclass)=${Service_name})
>
> exten =>
> _X.,n,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})*
> ***
>
> exten => _X.,n,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})**
> **
>
> exten => _X.,n,Set(FILENAME=${EXTEN}_${CALLERID(num)}_${CALLTIME}.WAV)
>
> exten => _X.,n,Set(RECORDFILENAME=${RECSUBDIR}/${FILENAME})
>
> exten => _X.,n,MixMonitor(${RECORDING_PATH}${RECORDFILENAME})
>
> exten => _X.,n,Dial(SIP/${EXTEN},120)
>
> exten => _X.,n,Hangup()
>
>  
>
>  
>
>  
>
>  
>
>  
>
>  
>
> Regards
>
> Akhilesh
>
> ** **
>
> On Tue, Jul 3, 2012 at 6:00 PM, akhilesh chand 
> wrote:
>
> Dear All,
>
>  
>
> I have two server 'A' and 'B' . In Server 'A', five different
> ivr (Sevices) is playing and call is *forwarding *into Server 'B'. Server
> 'B' basically use for agent login(Extension).
>
> I want to play different hold music(Server 'B') bases on the corresponding
> services which is running into server 'A'.
>
>  
>
> A single agent takes the call from different different services but hold
> music is play astrisk own by default.
>
>  
>
> Is there any way to   play  different hold music bases on  services which
> run into server A.
>
>  
>
> I have some changes into musiconhold.conf (server B) but problem is no
> solve.
>
>  
>
> please help me.
>
>  
>
>  
>
> Regards
>
> Akhilesh
>
> ** **
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thur

Re: [asterisk-users] Outbound Asterisk calls default directmedia specifications

[SamyGo]

I don't think you can set SIP properties in some variables anywhere in
asterisk dialplan or call file. What you can do is change the directmedia
options of the SIP or any other channel you're using. i.e if your call file
has

CHANNEL=SIP/12345@latestgateway

Then change the properties of the [latestgateway] in sip.conf. Also if
you're using an IP address directly

CHANNEL=SIP/12345@10.10.4.4

Then you can change the directmedia directive in sip.conf [general]
settings.

Hope it helped.

BR
Sammy Go.

On Wed, Jul 4, 2012 at 2:08 AM, sathiish kumar wrote:

> I am using call files to make calls to a remote machine but can't seem to
> quite understand the directmedia options that are set by default in
> Asterisk.Is there any way i can specify the directmedia options using call
> files?
>
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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Re: [asterisk-users] AMR - Segmentation Fault

[Chandrakant Solanki]

So, is http://sourceforge.net/projects/aterisk-amr/files/ same patch also
works in 1.8.13.0??

On Wed, Jul 4, 2012 at 3:18 AM, Hans Witvliet  wrote:

> On Tue, 2012-07-03 at 17:13 +0530, Chandrakant Solanki wrote:
> > Hi All,
> >
> > OS : Cent OS 5 64Bit
> > Asterisk : 1.8.0-rc2
> >
> > AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/
> >
> > When I tried to call or start asterisk, I found "Segmentation Fault".
>
> Without trying to be pedantic, but "1.8.0-rc2"
> Ever considered upgrading? To 1.8.13.0 or so..
>
> hans
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>



-- 
Regards,

Chandrakant Solanki
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Re: [asterisk-users] How to play different different hold music.

[SamyGo]

Hi,

The method Danny suggested is simple except I guess he swapped the priority
and exten field.
The idea is to dial a different extension on B server if you need to use
some other MOH class. If you don't want to change the dialled extension you
can always add a single digit prefix in Server-A and Server-B which
signifies the MOH class and use rest of the dialled extension as your real
destination number !

BR
Sammy



On Tue, Jul 3, 2012 at 7:16 PM, Danny Nicholas  wrote:

> Since you’re using IAX2 to contact Server B, you can use channel variables
> to control the moh class.  There was a good thread in June on this.  An
> “easier” way however would be to have each service dial a different IAX
> number, then have each IAX number on server B select it’s MOH Class.
>
> Server A
>
> [service1]
>
> Exten => N,1,Set(Service_name=service1)
>
> Exten => N,n,Dial(IAX2,server2:1234)
>
> [service2]
>
> Exten => N,1,Set(Service_name=service2)
>
> Exten => N,n,Dial(IAX2,server2:3456)
>
> ** **
>
> Server B
>
> [default]
>
> Exten => N,1,Verbose(start)
>
> Exten => N,1234,answer()
>
> Exten => N,n,Set(MOHClass=1)
>
> Exten => N,3456,answer()
>
> Exten => N,n,Set(MOHClass=2)
>
> ** **
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *akhilesh chand
> *Sent:* Tuesday, July 03, 2012 9:11 AM
> *To:* asterisk-users@lists.digium.com
> *Subject:* Re: [asterisk-users] How to play different different hold
> music.
>
> ** **
>
>  
>
> hi,
>
>  
>
> Server A > extentsion.conf
>
>  
>
> exten => N,n,Set(Service_name="Test")
>
> exten => N,n,Dial(IAX2/
> server2:server2@192.168.14.112/${result},${Service_name}
> )
>
>  
>
> but Server B doesn't identify service_name.
>
>  
>
>  
>
> extentsion.conf
>
>  
>
> [outgoing]
>
> exten => _X.,1,Set(_CALLTIME=${STRFTIME(,Asia/Calcutta,%d-%b-%y-%H-%M-%S)})
> 
>
> exten => _X.,1,Set(CHANNEL(musicclass)=${Service_name})
>
> exten =>
> _X.,n,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})*
> ***
>
> exten => _X.,n,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})**
> **
>
> exten => _X.,n,Set(FILENAME=${EXTEN}_${CALLERID(num)}_${CALLTIME}.WAV)
>
> exten => _X.,n,Set(RECORDFILENAME=${RECSUBDIR}/${FILENAME})
>
> exten => _X.,n,MixMonitor(${RECORDING_PATH}${RECORDFILENAME})
>
> exten => _X.,n,Dial(SIP/${EXTEN},120)
>
> exten => _X.,n,Hangup()
>
>  
>
>  
>
>  
>
>  
>
>  
>
>  
>
> Regards
>
> Akhilesh
>
> ** **
>
> On Tue, Jul 3, 2012 at 6:00 PM, akhilesh chand 
> wrote:
>
> Dear All,
>
>  
>
> I have two server 'A' and 'B' . In Server 'A', five different
> ivr (Sevices) is playing and call is *forwarding *into Server 'B'. Server
> 'B' basically use for agent login(Extension).
>
> I want to play different hold music(Server 'B') bases on the corresponding
> services which is running into server 'A'.
>
>  
>
> A single agent takes the call from different different services but hold
> music is play astrisk own by default.
>
>  
>
> Is there any way to   play  different hold music bases on  services which
> run into server A.
>
>  
>
> I have some changes into musiconhold.conf (server B) but problem is no
> solve.
>
>  
>
> please help me.
>
>  
>
>  
>
> Regards
>
> Akhilesh
>
> ** **
>
> --
> _
> -- 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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[asterisk-users] QUEUEMEMBER_STATUS incorrect?

[Chet W. Stevens]

I have been doing some testing with queues. I have been experiencing some 
strange behavior and I wanted to see if anyone else sees this. I am using 
1.8.11-cert2.

It is my understanding that I cannot directly tell from the dial plan that a 
member is dynamic or static unless I check RQMSTATUS from RemoveQueueMember for 
NOTDYNAMIC. Is this true? (If I try something like using AST_CONFIG then I can 
only get back the
first static member for a queue) The problem that I am seeing is that if I try 
to remove a queue member that is static, the values from then on that I see 
from ${QUEUE_MEMBER(,free) ${QUEUE_MEMBER(,ready) are incorrect. 
They are always one
less than the actual value for that queue. Can anyone else confirm this? Thank 
you.

Chet Stevens

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Re: [asterisk-users] AMR - Segmentation Fault

[Hans Witvliet]

On Tue, 2012-07-03 at 17:13 +0530, Chandrakant Solanki wrote:
> Hi All,
> 
> OS : Cent OS 5 64Bit
> Asterisk : 1.8.0-rc2
> 
> AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/
> 
> When I tried to call or start asterisk, I found "Segmentation Fault". 

Without trying to be pedantic, but "1.8.0-rc2"
Ever considered upgrading? To 1.8.13.0 or so..

hans


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[asterisk-users] Outbound Asterisk calls default directmedia specifications

[sathiish kumar]

I am using call files to make calls to a remote machine but can't seem to
quite understand the directmedia options that are set by default in
Asterisk.Is there any way i can specify the directmedia options using call
files?
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[asterisk-users] Free PBX: hangup even if did not dial # in the voicemail

[bilal ghayyad]

Dear;

What is the setting to be done on freepbx to let the voicemail go for hangup 
after while (or after leaving the message) even if the caller did not dial #. 
It is very important for me to be sure of the hangup status.

Regards
Bilal



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Re: [asterisk-users] IAX trunking stopped working

[Noah Engelberth]

> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Matthew Jordan
> Sent: Tuesday, July 03, 2012 2:27 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] IAX trunking stopped working
> 
> 
> - Original Message -
> 
> > From: "Noah Engelberth" 
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> > 
> > Sent: Tuesday, July 3, 2012 10:56:10 AM
> > Subject: [asterisk-users] IAX trunking stopped working
> 
> > I administer a group of Asterisk servers running a mix of 10.3, 10.4,
> > and 1.8.8.1 (mostly 10.4). One of those servers is a call
> > concentrator/relay for E911 service. All of the other servers make an
> > IAX connection to the relay server, which then hands off to a SIP
> > trunk to my E911 provider. It all worked as recently as 2 weeks ago,
> > but I discovered that sometime between then and now it stopped working
> > without any explanation. Last modified time on the config files is
> > over 2 months ago.
> 
> > The setup is as follows:
> 
> > On the call relay (IAX “receiver”)
> > [my-remote-server]
> > type=user
> > host=dynamic
> > username=my-remote-username
> > encryption=yes
> > secret=my-remote-secret
> > context=my-call-context
> > deny=0.0.0.0/0.0.0.0
> > permit=remote.server.ip.address/255.255.255.255
> 
> > On the VoIP servers (IAX “sender”)
> > - One of the servers is set to register: register =>
> > my-remote-username:my-remote-sec...@call.relay.server.ip
> > - Another is set to just use the peer definition as below without
> > trying to register [my-remote-server] type=peer
> > host=call.relay.server.ip username=my-remote-username
> > secret=my-remote-secret qualify=no
> 
> > Dialplan on the VoIP servers:
> > exten => 911,1,Verbose()
> > same => n,Dial(IAX2/my-remote-server/911)
> 
> > Dialplan on the relay server:
> > [my-call-context]
> > exten => 911,1,Verbose()
> > same => n,Dial(Relay to E911)
> 
> > The issue I’m seeing is this:
> > - On the servers that are set to register, the relay server is
> > rejecting the registration (I’ve confirmed the
> > username/peername/secrets are an exact match on both sides, and
> > nothing has changed from when they were working). IAX debug on the
> > relay server shows the auths come in and the relay server send REGREJ
> > – Registration Refused, Cause Code 29. IAX debug on the server
> > attempting to register shows sending the REGAUTH packets and receiving
> > the REGREJ packets. The IP address shown in the IAX debug packets
> > matches the IP address in the permit rule for each peer that’s
> > supposed to register.
> 
> Can you set authdebug and iaxdebug to true in your iax2 configuration -
> preferably on both ends, but in particular on the server that is servicing the
> registration attempt - and post the portion of the DEBUG logs that shows it
> rejecting the registration?
> 
> Something is failing authentication, and authdebug should tell us at least why
> the server thinks it should reject the attempt.
> 

After some more messing around, I think it's my configuration error on the 
registration.  I have the peers on the "receiving" server set up as users, not 
friends, since the calls will only be one-way from the VoIP servers to the 
relay server.  With the debugs on, the error on registration is [Jul  3 
14:52:40] NOTICE[1447]: chan_iax2.c:8100 register_verify: No registration for 
peer 'my-remote-peer' (from remote.peer.ip.address).  Changing the "receiving" 
server's definition to a friend and putting in a host=dynamic line makes the 
errors go away and registration work as expected.  So I guess that was legacy 
cruft configuration causing me confusion there, sorry.

> > - On the server that is set to just send the calls, an attempt to dial
> > 911 just hangs for 60 seconds and eventually times out without sending
> > the call. IAX debug on the relay server shows the call start frame get
> > RX’d, shows the relay server try to TX a CTOKEN frame, and nothing
> > further (other than retransmissions of the call start frame). IAX
> > debug on the server trying to send the call to the relay server shows
> > the TX for the call start frame, but no RX for the CTOKEN frames.
> 
> Is this the same server that failed to register?  If so, it may be best to 
> focus
> on that problem first.
> 

No, different server.  Also, after doing some further investigation, it looks 
like my problem of no-calls may be network related -- if I change what vlan the 
"receiving" server uses to connect to the network so it is internal to my LAN, 
the IAX connections start mostly working (though I break my trunk connection to 
my E911 provider, and the one server that connects has a NAT issue, but that's 
definitely a network problem).   When the server is connected to my Public IP 
VLAN, the SIP trunk to the E911 provider works, but the IAX connections don't.

> > Ultimately, this has gone from working 

Re: [asterisk-users] IAX trunking stopped working

[Matthew Jordan]

- Original Message - 

> From: "Noah Engelberth" 
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> 
> Sent: Tuesday, July 3, 2012 10:56:10 AM
> Subject: [asterisk-users] IAX trunking stopped working

> I administer a group of Asterisk servers running a mix of 10.3, 10.4,
> and 1.8.8.1 (mostly 10.4). One of those servers is a call
> concentrator/relay for E911 service. All of the other servers make
> an IAX connection to the relay server, which then hands off to a SIP
> trunk to my E911 provider. It all worked as recently as 2 weeks ago,
> but I discovered that sometime between then and now it stopped
> working without any explanation. Last modified time on the config
> files is over 2 months ago.

> The setup is as follows:

> On the call relay (IAX “receiver”)
> [my-remote-server]
> type=user
> host=dynamic
> username=my-remote-username
> encryption=yes
> secret=my-remote-secret
> context=my-call-context
> deny=0.0.0.0/0.0.0.0
> permit=remote.server.ip.address/255.255.255.255

> On the VoIP servers (IAX “sender”)
> - One of the servers is set to register: register =>
> my-remote-username:my-remote-sec...@call.relay.server.ip
> - Another is set to just use the peer definition as below without
> trying to register
> [my-remote-server]
> type=peer
> host=call.relay.server.ip
> username=my-remote-username
> secret=my-remote-secret
> qualify=no

> Dialplan on the VoIP servers:
> exten => 911,1,Verbose()
> same => n,Dial(IAX2/my-remote-server/911)

> Dialplan on the relay server:
> [my-call-context]
> exten => 911,1,Verbose()
> same => n,Dial(Relay to E911)

> The issue I’m seeing is this:
> - On the servers that are set to register, the relay server is
> rejecting the registration (I’ve confirmed the
> username/peername/secrets are an exact match on both sides, and
> nothing has changed from when they were working). IAX debug on the
> relay server shows the auths come in and the relay server send
> REGREJ – Registration Refused, Cause Code 29. IAX debug on the
> server attempting to register shows sending the REGAUTH packets and
> receiving the REGREJ packets. The IP address shown in the IAX debug
> packets matches the IP address in the permit rule for each peer
> that’s supposed to register.

Can you set authdebug and iaxdebug to true in your iax2 configuration - 
preferably
on both ends, but in particular on the server that is servicing the registration
attempt - and post the portion of the DEBUG logs that shows it rejecting the
registration?

Something is failing authentication, and authdebug should tell us at least
why the server thinks it should reject the attempt.

> - On the server that is set to just send the calls, an attempt to
> dial 911 just hangs for 60 seconds and eventually times out without
> sending the call. IAX debug on the relay server shows the call start
> frame get RX’d, shows the relay server try to TX a CTOKEN frame, and
> nothing further (other than retransmissions of the call start
> frame). IAX debug on the server trying to send the call to the relay
> server shows the TX for the call start frame, but no RX for the
> CTOKEN frames.

Is this the same server that failed to register?  If so, it may be best
to focus on that problem first.

> Ultimately, this has gone from working to totally broken without any
> apparent change to my configuration. I need help to try to
> troubleshoot it further, I’ve tried everything I can think of
> (including transferring the backed-up working config files to a
> brand new clean-load server, upgrading Asterisk, and recreating the
> configurations by hand), and nothing seems to be helping.

Since you have a number of different servers running different versions
of Asterisk, can you provide which server in your scenario is running
which version?

> Thank you,

> Noah Engelberth
> MetaLINK Technologies



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Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
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[asterisk-users] IAX trunking stopped working

[Noah Engelberth]

I administer a group of Asterisk servers running a mix of 10.3, 10.4, and 
1.8.8.1 (mostly 10.4).  One of those servers is a call concentrator/relay for 
E911 service.  All of the other servers make an IAX connection to the relay 
server, which then hands off to a SIP trunk to my E911 provider.  It all worked 
as recently as 2 weeks ago, but I discovered that sometime between then and now 
it stopped working without any explanation.  Last modified time on the config 
files is over 2 months ago.

The setup is as follows:

On the call relay (IAX "receiver")
[my-remote-server]
type=user
host=dynamic
username=my-remote-username
encryption=yes
secret=my-remote-secret
context=my-call-context
deny=0.0.0.0/0.0.0.0
permit=remote.server.ip.address/255.255.255.255

On the VoIP servers (IAX "sender")

-  One of the servers is set to register: register => 
my-remote-username:my-remote-sec...@call.relay.server.ip

-  Another is set to just use the peer definition as below without 
trying to register
[my-remote-server]
type=peer
host=call.relay.server.ip
username=my-remote-username
secret=my-remote-secret
qualify=no

Dialplan on the VoIP servers:
exten => 911,1,Verbose()
same => n,Dial(IAX2/my-remote-server/911)

Dialplan on the relay server:
[my-call-context]
exten => 911,1,Verbose()
same => n,Dial(Relay to E911)

The issue I'm seeing is this:

-  On the servers that are set to register, the relay server is 
rejecting the registration (I've confirmed the username/peername/secrets are an 
exact match on both sides, and nothing has changed from when they were 
working).  IAX debug on the relay server shows the auths come in and the relay 
server send REGREJ - Registration Refused, Cause Code 29.  IAX debug on the 
server attempting to register shows sending the REGAUTH packets and receiving 
the REGREJ packets.  The IP address shown in the IAX debug packets matches the 
IP address in the permit rule for each peer that's supposed to register.

-  On the server that is set to just send the calls, an attempt to dial 
911 just hangs for 60 seconds and eventually times out without sending the 
call.  IAX debug on the relay server shows the call start frame get RX'd, shows 
the relay server try to TX a CTOKEN frame, and nothing further (other than 
retransmissions of the call start frame).  IAX debug on the server trying to 
send the call to the relay server shows the TX for the call start frame, but no 
RX for the CTOKEN frames.

Ultimately, this has gone from working to totally broken without any apparent 
change to my configuration.  I need help to try to troubleshoot it further, 
I've tried everything I can think of (including transferring the backed-up 
working config files to a brand new clean-load server, upgrading Asterisk, and 
recreating the configurations by hand), and nothing seems to be helping.


Thank you,

Noah Engelberth
MetaLINK Technologies

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Re: [asterisk-users] Please dont tell me this is impossible

[Thorsten Göllner]

Sorry, but I am using a self developed PHP-Library where I parse STDIN 
myself. So I have no problem on this side. You are using a Perl-API? 
There should be a method available for getting the AGI-Result-String?! I 
never used Perl myself ...


Am 03.07.2012 16:13, schrieb CDR:

Yes, ai saw that information on the debug, but how do you bring it
inside a variable, so you may use it? I could not find a way. Maybe I
am missing something?

On Tue, Jul 3, 2012 at 9:20 AM, Thorsten Göllner  wrote:

I just tried it on asterisk 1.8.13 with "agi set debug on". The last log
line reveals it - streamfile return the endpos.

[2012-07-03 15:16:39] VERBOSE[7046] res_agi.c: AGI
Rx << STREAM FILE /audio1/dtmf_detector/2.0 1234567890*#

[2012-07-03 15:16:39] VERBOSE[7046] res_agi.c: -- Playing
'/audio1/dtmf_detector/2.0' (escape_digits=1234567890*#) (sample_offset 0)

[2012-07-03 15:16:40] VERBOSE[7046] res_agi.c: AGI
Tx >> 200 result=0 endpos=4800

So please doublecheck your result.

Am 03.07.2012 00:47, schrieb CDR:


1.8 is my version, until the new one is stable.

On Mon, Jul 2, 2012 at 4:57 PM, Thorsten Göllner  wrote:


What Asterisk version?

Am 02.07.2012 15:14, schrieb CDR:


Thanks. I already solved it using this command. The only issue was
that it gives you as return the ASCII code of the digit pressed
instead of the digit itself. For some reason my brain did not process
that detail. But it does work. However, the offset played is not
returned. Has anybody tested this and has a coding sample in perl?
Philip

On Mon, Jul 2, 2012 at 8:52 AM, Thorsten Göllner 
wrote:



So take a look here:
http://www.voip-info.org/wiki/view/stream+file

Am 29.06.2012 16:06, schrieb CDR:


This is from the documentation of Perl-AGI
"$AGI->stream_file($filename, $digits, $offset)
Executes AGI Command "STREAM FILE $filename $digits [$offset]"
This command instructs Asterisk to play the given sound file and
listen for the given dtmf digits. The fileextension must not be used
in the filename because Asterisk will find the most appropriate file
type. $filename can be an array of files or a single filename.
Example: $AGI->stream_file('demo-echotest', '0123');
$AGI->stream_file(['demo-echotest', 'demo-welcome'], '0123');
Returns: -1 on error or hangup, 0 if playback completes without a
digit being pressed, or the ASCII numerical value of the digit if a
digit was pressed"

It does not mention that it returns the offset at which the file
stopped playing. Also, if you could get that number, then restarting
the stream would result, I guess, in an audible interruption. Please
advise how to get the offset on the result and I will try.
Yours
Philip



On Fri, Jun 29, 2012 at 6:27 AM, Thorsten Göllner 
wrote:




Am 29.06.2012 11:38, schrieb CDR:


I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI->command. I need to play
a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has expired. So far I found impossible to
achieve
this functionality. Am I missing something?
Philip






The Playcommand will be interrupted by the key but the agi result
contains
the offset. So you can play this file from offset again until you
$maxdigits
has been pressed. Take a look here:
https://wiki.asterisk.org/wiki/display/AST/AGICommand_STREAM+FILE



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Re: [asterisk-users] How to play different different hold music.

[Danny Nicholas]

Since you're using IAX2 to contact Server B, you can use channel variables
to control the moh class.  There was a good thread in June on this.  An
"easier" way however would be to have each service dial a different IAX
number, then have each IAX number on server B select it's MOH Class.

Server A

[service1]

Exten => N,1,Set(Service_name=service1)

Exten => N,n,Dial(IAX2,server2:1234)

[service2]

Exten => N,1,Set(Service_name=service2)

Exten => N,n,Dial(IAX2,server2:3456)

 

Server B

[default]

Exten => N,1,Verbose(start)

Exten => N,1234,answer()

Exten => N,n,Set(MOHClass=1)

Exten => N,3456,answer()

Exten => N,n,Set(MOHClass=2)

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of akhilesh chand
Sent: Tuesday, July 03, 2012 9:11 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] How to play different different hold music.

 

 

hi,

 

Server A > extentsion.conf

 

exten => N,n,Set(Service_name="Test")

exten =>
N,n,Dial(IAX2/server2:server2@192.168.14.112/${result},${Service_name}
 )

 

but Server B doesn't identify service_name.

 

 

extentsion.conf

 

[outgoing]

exten => _X.,1,Set(_CALLTIME=${STRFTIME(,Asia/Calcutta,%d-%b-%y-%H-%M-%S)})

exten => _X.,1,Set(CHANNEL(musicclass)=${Service_name})

exten =>
_X.,n,Set(CALLTIME=${STRFTIME(${EPOCH},Asia/Calcutta,%d-%b-%Y-%H-%M-%S)})

exten => _X.,n,Set(RECSUBDIR=${STRFTIME(${EPOCH},Asia/Calcutta,%d%b%Y)})

exten => _X.,n,Set(FILENAME=${EXTEN}_${CALLERID(num)}_${CALLTIME}.WAV)

exten => _X.,n,Set(RECORDFILENAME=${RECSUBDIR}/${FILENAME})

exten => _X.,n,MixMonitor(${RECORDING_PATH}${RECORDFILENAME})

exten => _X.,n,Dial(SIP/${EXTEN},120)

exten => _X.,n,Hangup()

 

 

 

 

 

 

Regards

Akhilesh

 

On Tue, Jul 3, 2012 at 6:00 PM, akhilesh chand 
wrote:

Dear All,

 

I have two server 'A' and 'B' . In Server 'A', five different ivr (Sevices)
is playing and call is forwarding into Server 'B'. Server 'B' basically use
for agent login(Extension).

I want to play different hold music(Server 'B') bases on the corresponding
services which is running into server 'A'.

 

A single agent takes the call from different different services but hold
music is play astrisk own by default.

 

Is there any way to   play  different hold music bases on  services which
run into server A.

 

I have some changes into musiconhold.conf (server B) but problem is no
solve.

 

please help me.

 

 

Regards

Akhilesh

 

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Re: [asterisk-users] How to play different different hold music.

[A J Stiles]

That is all.  Thank you.

FAQ.
welcome to read the
believe me, you are
If you do not

to *bottom*.
we read from *top*
In this mailing list,

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] Please dont tell me this is impossible

[Thorsten Göllner]

I just tried it on asterisk 1.8.13 with "agi set debug on". The last log 
line reveals it - streamfile return the endpos.


[2012-07-03 15:16:39] VERBOSE[7046] res_agi.c: 
AGI Rx << STREAM FILE /audio1/dtmf_detector/2.0 
1234567890*#


[2012-07-03 15:16:39] VERBOSE[7046] res_agi.c: -- Playing 
'/audio1/dtmf_detector/2.0' (escape_digits=1234567890*#) (sample_offset 0)


[2012-07-03 15:16:40] VERBOSE[7046] res_agi.c: 
AGI Tx >> 200 result=0 endpos=4800


So please doublecheck your result.

Am 03.07.2012 00:47, schrieb CDR:

1.8 is my version, until the new one is stable.

On Mon, Jul 2, 2012 at 4:57 PM, Thorsten Göllner  wrote:

What Asterisk version?

Am 02.07.2012 15:14, schrieb CDR:


Thanks. I already solved it using this command. The only issue was
that it gives you as return the ASCII code of the digit pressed
instead of the digit itself. For some reason my brain did not process
that detail. But it does work. However, the offset played is not
returned. Has anybody tested this and has a coding sample in perl?
Philip

On Mon, Jul 2, 2012 at 8:52 AM, Thorsten Göllner  wrote:


So take a look here:
http://www.voip-info.org/wiki/view/stream+file

Am 29.06.2012 16:06, schrieb CDR:


This is from the documentation of Perl-AGI
"$AGI->stream_file($filename, $digits, $offset)
Executes AGI Command "STREAM FILE $filename $digits [$offset]"
This command instructs Asterisk to play the given sound file and
listen for the given dtmf digits. The fileextension must not be used
in the filename because Asterisk will find the most appropriate file
type. $filename can be an array of files or a single filename.
Example: $AGI->stream_file('demo-echotest', '0123');
$AGI->stream_file(['demo-echotest', 'demo-welcome'], '0123');
Returns: -1 on error or hangup, 0 if playback completes without a
digit being pressed, or the ASCII numerical value of the digit if a
digit was pressed"

It does not mention that it returns the offset at which the file
stopped playing. Also, if you could get that number, then restarting
the stream would result, I guess, in an audible interruption. Please
advise how to get the offset on the result and I will try.
Yours
Philip



On Fri, Jun 29, 2012 at 6:27 AM, Thorsten Göllner 
wrote:



Am 29.06.2012 11:38, schrieb CDR:


I have been fighting all night with version 1.8 and have not found a
way to do this with any command or Perl AGI->command. I need to play a
file and wait until the customer presses at least $maxdigits to
return, BUT, the file must continue playing until $maxdigits is
received or $timeout has expired. So far I found impossible to achieve
this functionality. Am I missing something?
Philip





The Playcommand will be interrupted by the key but the agi result
contains
the offset. So you can play this file from offset again until you
$maxdigits
has been pressed. Take a look here:
https://wiki.asterisk.org/wiki/display/AST/AGICommand_STREAM+FILE



--
Thorsten Göllner

OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf

Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54




--
Thorsten Göllner

OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf

Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54




--
Thorsten Göllner

OVM Office Voice Media GmbH
Herderstrasse 68
40237 Düsseldorf

Tel.: +49(0)211 / 618 57 53
Fax: +49(0)211 / 618 57 54



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Re: [asterisk-users] How to play different different hold music.

[SamyGo]

Hi,
if possible for you put some header in SIP which mentions the music on hold
flag on Server-A. The Dial the call to Server-B. On Server-B extract the
value of that header and change the music on hold class based on the value.

Regards,
Sammy


On Tue, Jul 3, 2012 at 5:30 PM, akhilesh chand wrote:

> Dear All,
>
> I have two server 'A' and 'B' . In Server 'A', five different
> ivr (Sevices) is playing and call is *forwarding *into Server 'B'. Server
> 'B' basically use for agent login(Extension).
> I want to play different hold music(Server 'B') bases on the corresponding
> services which is running into server 'A'.
>
> A single agent takes the call from different different services but hold
> music is play astrisk own by default.
>
> Is there any way to   play  different hold music bases on  services which
> run into server A.
>
> I have some changes into musiconhold.conf (server B) but problem is no
> solve.
>
> please help me.
>
>
> Regards
> Akhilesh
>
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
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[asterisk-users] How to play different different hold music.

[akhilesh chand]

Dear All,

I have two server 'A' and 'B' . In Server 'A', five different ivr (Sevices)
is playing and call is *forwarding *into Server 'B'. Server 'B' basically
use for agent login(Extension).
I want to play different hold music(Server 'B') bases on the corresponding
services which is running into server 'A'.

A single agent takes the call from different different services but hold
music is play astrisk own by default.

Is there any way to   play  different hold music bases on  services which
run into server A.

I have some changes into musiconhold.conf (server B) but problem is no
solve.

please help me.


Regards
Akhilesh
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Re: [asterisk-users] SendFAX timestamp

[Steve Underwood]

Hi David,

The old app_fax code, which allowed spandsp to be used with Asterisk 
before Digium introduced the new modules supported the features you 
want. Maybe someone can go through that code and port the feature into 
the current res-fax code.


Steve

On 07/03/2012 09:57 AM, David Cunningham wrote:

Kevin,

Thanks for the reply.


On 29 June 2012 00:29, Kevin P. Fleming > wrote:


On 06/27/2012 09:30 PM, David Cunningham wrote:

Would anyone else know if Asterisk allows use of SpanDSP's
time zone
conversion?


No, SendFAX (in res_fax) doesn't currently offer the ability to do
what you are asking about.




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[asterisk-users] AMR - Segmentation Fault

[Chandrakant Solanki]

Hi All,

OS : Cent OS 5 64Bit
Asterisk : 1.8.0-rc2

AMR Source Link : http://sourceforge.net/projects/aterisk-amr/files/

When I tried to call or start asterisk, I found "Segmentation Fault". Below
I paste same for AMR


Loaded symbols for /usr/lib/asterisk/modules/app_db.so
Core was generated by `asterisk -qg'.
Program terminated with signal 11, Segmentation fault.
#0  D_plsf_3 (st=, mode=,
bfi=, indice=,
lsp1_q=0x7fff11d05df0)
at sp_dec.c:567
567   tmp = ( ( cos_table[ind+1]-cos_table[ind] )*offset ) << 1;
(gdb) br
Breakpoint 1 at 0x2aaab57093f1: file sp_dec.c, line 567.
(gdb) bt
#0  D_plsf_3 (st=, mode=,
bfi=, indice=,
lsp1_q=0x7fff11d05df0)
at sp_dec.c:567
#1  0x2aaab570df95 in Decoder_amr (st=0x2aaad6147d00, mode=MR515,
parm=0x7fff11d06a40, frame_type=,
synth=0x7fff11d060a0,
A_t=0x7fff11d06730) at sp_dec.c:4717
#2  0x2aaab5712e6a in Speech_Decode_Frame (st=0x2aaad613e200, mode=80,
parm=0x2aaab5725400, frame_type=4294949091, synth=0x2aaad6142ba0)
at sp_dec.c:5676
#3  0x2aaab56efb25 in Decoder_Interface_Decode (st=0x2aaad613e1e0,
bits=, synth=0x2aaad6142ba0, bfi=)
at interf_dec.c:816
#4  0x2aaab56ee6f9 in amrtolin_framein (pvt=0x2aaad613e5c0, f=) at codec_amr.c:263
#5  0x00528244 in framein (pvt=0x2aaad613e5c0, f=0x2aaab5942e40) at
translate.c:178
#6  0x00529538 in calc_cost (t=0x2aaab593ff40, seconds=1) at
translate.c:397
#7  0x0052a00c in __ast_register_translator (t=0x2aaab593ff40,
mod=) at translate.c:835
#8  0x2aaab56ee37b in load_module () at codec_amr.c:490
#9  0x004c29e3 in start_resource (mod=0xdf) at loader.c:785
#10 0x004c3308 in load_resource_list (load_order=0x7fff11d07000,
global_symbols=0, mod_count=0x7fff11d0701c) at loader.c:973
#11 0x004c3727 in load_modules (preload_only=0) at loader.c:1126
#12 0x0043c2c4 in main (argc=,
argv=0x7fff11d095e8) at asterisk.c:3794
(gdb) quit


-- 
Regards,

Chandrakant Solanki
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[asterisk-users] Centos 6 mISDN

[Andrew Colin]

Hi Guys

Has anyone got this working on Centos 6?
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