Re: [asterisk-users] Add PinCode on my dialplan

[Sam Govind]

Though its in Spanish - Hope it is exactly what OP requires !

http://www.voztovoice.org/?q=node/477

Found it via : http://www.voip-info.org/ News

Regards,
-Sammy

On Thu, Sep 22, 2011 at 6:50 PM, bakko  wrote:

> Hi look at option a. This option put on accountcode field the name on the
> left your password file.
>
> Regards
>
> Enviado desde mi iPad
>
> El 22/09/2011, a las 5:49, Malvin Rito 
> escribió:
>
> Hi,
>
> I tried Authenticate where pass codes are stored on the file pass.conf and
> it works.
>
> exten => _,1,Authenticate(/etc/asterisk/pass.conf)
>
> Since I have my CDR, I want to have a report wherein I can check which pass
> code did the call. How can I achieve it?
> Using authenticate through file does not replace ACCOUNT_CODE field with
> the pass code entered, it only show *ast_h323 *under the Account_Code
> field.
>
> Regards,
> Malvin
>
> On 9/21/2011 1:01 PM, Sam Govind wrote:
>
> See "core show application authe"
> If passwords are already the same as those of voicemail.conf go for
> application VMAuthenticate() - DIA generates a dial-tone which I don't think
> is suitable for dialling out from users(insiders)
>
>-= Info about application 'Authenticate' =-
>
>  [Synopsis]
> Authenticate a user
>
>  [Description]
> This application asks the caller to enter a given password in order to
> continue
> dialplan execution.
> If the password begins with the '/' character,  it is interpreted as a file
> which contains a list of valid passwords, listed 1 password per line in the
> file.
> When using a database key, the value associated with the key can be
> anything.
> Users have three attempts to authenticate before the channel is hung
> up.
>
>  [Syntax]
> Authenticate(password[,options[,maxdigits[,prompt]]])
>
>  [Arguments]
> password
> Password the user should know
> options
> a: Set the channels' account code to the password that is entered
> d: Interpret the given path as database key, not a literal file
> m: Interpret the given path as a file which contains a list of account
> codes and password hashes delimited with ':', listed one per line in
> the
> file. When one of the passwords is matched, the channel will have its
> account code set to the corresponding account code in the file.
> r: Remove the database key upon successful entry (valid with 'd'
> only)
> maxdigits
> maximum acceptable number of digits. Stops reading after maxdigits
> have been entered (without requiring the user to press the '#' key).
> Defaults to 0 - no limit - wait for the user press the '#' key.
> prompt
> Override the agent-pass prompt file.
>
>  [See Also]
> VMAuthenticate(), DISA()
>
>
>  On Wed, Sep 21, 2011 at 9:47 AM, Malvin Rito <
> mr...@mail.altcladding.com.ph> wrote:
>
>>  Thanks. ?If I want to use unique PIN for every user that dials out how
>> can I implement it using Authenticate app?
>>
>> Regards,
>>  Malvin
>>
>>
>> On 9/21/2011 12:42 PM, Sam Govind wrote:
>>
>> DISA and DB based Auth could be an overkill.
>>
>> Kyle showed the very simplistic dial plan if Dial-out pin is common for
>> the whole system.
>> See application *Authenticate(password[,options[,maxdigits[,prompt]]] *and
>> if Voicemail PIN are required to be used use application 
>> *MAuthenticate([mailbox][@context][,options]
>> *
>>
>>  Regards,
>>
>>  - Sammy
>>
>> On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton < 
>> k...@mocker.org> wrote:
>>
>>> Something like this should work:
>>>
>>>  exten => _011.,1,Answer
>>> exten => _011.,n,Wait(1)
>>> exten => _011.,n,Read(password,enter-password,5)
>>> exten => _011.,n,GotoIf($[${password} = 12345]?5:9)
>>>
>>>  exten => _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
>>> exten => _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
>>>
>>>  exten => _011.,n,Hangup
>>> exten => _011.,n,Playback(invalid)
>>> exten => _011.,n,Hangup
>>>
>>>  Could be cleaned up (the GotoIf isn't very descriptive about where it's
>>> going), but it's a starting point.
>>>
>>>
>>>  On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:
>>>
>>>  Hi List,
>>> I currently have a asterisk server running used for dialing-out for IDD
>>> but I want to Put a pincode wherein only users with the right pin code will
>>> be allowed to 

Re: [asterisk-users] Make asterisk cluster appear and operate as a single server?

[Sam Govind]

Hey,
Why do you think using OpenSIPs is not going to work for you ? You can
always add SIP trunks on openSips and based upon which trunks getting the
call you can LB or/and FO to as many asterisk servers as you want !

Regards,
-Sammy

On Sun, Oct 2, 2011 at 7:12 AM, Michelle Dupuis  wrote:

>  If one server is supposed to carry the full load of the other during
> failure, then you have to size each server to handle  100% load - so load
> balancing is pointless.
>
> Checkout haast at www.generationd.com and read  the docs on how it does
> failover...certainly good for ideas.
>
>  --
> *From:* asterisk-users-boun...@lists.digium.com [
> asterisk-users-boun...@lists.digium.com] On Behalf Of Tobias Steen [
> tobias.st...@s2.se]
> *Sent:* Saturday, October 01, 2011 6:30 PM
> *To:* Asterisk Users List
> *Subject:* [asterisk-users] Make asterisk cluster appear and operate as a
> single server?
>
>   Hi,
>
>
>
> I'm trying to plan a system of clustered asterisk machines where a number
> of SIP trunks will be hosted on the platform. Each trunk will be hosted for
> a specific customer who owns it and therefore payment is handled directly
> between the customers and their trunk-providers, each trunk will have about
> 50-200 simultaneous calls.
>
>
>
> No SIP phones will be directly connected to the platform, my thought is
> that the asterisk machines should only receive incoming and make outgoing
> calls through the trunks, and then connect the calls with each other.
>
>
>
> To make this scalable and have the option of running an infinite number of
> sip-trunks, I need a good way to load-balance my asterisk servers and
> implement failover support and also be able to add / replace the machines in
> the cluster in a safe and reliable way.
>
>
>
> I'm have some experience building single asterisk solutions but I have
> never worked with load balancing of multiple asterisk machines.
>
>
>
> Is it possible to configure all trunks on a single asterisk setup which is
> then reflected over a cluster of asterisk machines? If I have a cluster of
> machines, I guess I need some kind of front-end application / system? I will
> then also need to be able to connect calls between the machines, the calls
> to be connected with each other will always be incoming and outgoing on the
> same trunk.
>
>
>
> In other words, I want to create a large cluster of asterisk machines to
> appear and operate as a single asterisk server.
>
>
>
> I've looked at projects like OpenSIP but it feels like this is not really
> what I need?
>
>
>
> I really appreciate if someone can help me get on the correct path here, I
> need all the feedback I can get.
>
>
>
>
>
> Thanks in advance!
>
>
>
>
>
> Best regards
>
> Tobias
>
>
>
> --
> _
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Re: [asterisk-users] invite authentication error !?

[Sam Govind]

Please post some configurations. YOU CAN REMOVE username / secret from the
sip configs but posting remaining configs will only help to your issue
resolution, Alex first words were "This is just a speculative shot in the
dark" .

Fine if you were just Trolling here !


2011/9/30 cnasterisk 

> **
> asterisk can register successfully on the remote party. so i think username
> & password must be ok
>
>
> 2011-09-30
> --
>  cnasterisk
> ------
> *>发件人：* Sam Govind
> *>发送时间：* 2011-09-30  16:05:21
> *>收件人：* Asterisk Users Mailing List - Non-Commercial Discussion
> *>抄送：*
> *>主题：* Re: [asterisk-users] invite authentication error !?
>  >Whatever that remote party is, you are most definitely using
> a username/secret declaration for that. So the sip attributes set for that
> proxy define the behaviour for this.
>
> On Fri, Sep 30, 2011 at 12:55 PM, cnasterisk  wrote:
>
>> **
>>  hi, Sam
>>  thanks for your kindly reply.
>> The remote proxy is not asterisk
>>
>> 2011-09-30
>> --
>>  cnasterisk
>> --
>> *>发件人：* Sam Govind
>> *>发送时间：* 2011-09-30  15:36:41
>>  *>收件人：* Asterisk Users Mailing List - Non-Commercial Discussion
>> *>抄送：*
>>  *>主题：* Re: [asterisk-users] invite authentication error !?
>>   >What Sip declaration are you using for the remote sip proxy in
>> sip.conf?
>>
>> On Fri, Sep 30, 2011 at 12:30 PM, Alex Balashov <
>> abalas...@evaristesys.com> wrote:
>>
>>>  This is just a speculative shot in the dark, but remember that the
>>> domain of the From URI is important, and that the authentication "realm"
>>> (domain) is part of the authentication credentials.  So, what you have in
>>> your 'fromdomain' and 'host' settings on the peer does matter.
>>>
>>> --
>>> This message was painstakingly thumbed out on my mobile, so apologies for
>>> brevity, errors, and general sloppiness.
>>>
>>> Alex Balashov - Principal
>>> Evariste Systems LLC
>>> 260 Peachtree Street NW
>>> Suite 2200
>>> Atlanta, GA 30303
>>> Tel: +1-678-954-0670
>>> Fax: +1-404-961-1892
>>> Web: <http://www.evaristesys.com/>http://www.evaristesys.com/
>>>
>>> On Sep 30, 2011, at 3:16 AM, "cnasterisk"  wrote:
>>>
>>>   hi,
>>>Dear all.
>>>I setted a sip account on a sip trunk. when a  client call via this
>>> sip trunk, asterisk call failed on this trunk.
>>> I have captured the sip messages on the host where asterisk located, and
>>> found that:
>>>
>>> 1. asterisk send a INVITE message to remote sip proxy without
>>> "proxy-authorization" field.
>>> 2. the remote sip proxy send back a
>>> " SIP/2.0 407 Proxy Authentication Required" message.
>>> 3. asterisk send a INVITE message with  "proxy-authorization" field.
>>> 4. remote proxy send back a "403(Forbidden)" message， that is mean "wrong
>>> password"
>>>
>>> I also tested the sip account on a softphone, it works normal!
>>>
>>> why this happed？ and how can i solve it?
>>>
>>>
>>>
>>> 2011-09-30
>>> --
>>> kevin
>>>
>>>  --
>>> _
>>> -- 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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>>>
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>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>>
>>
>>
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>> __

Re: [asterisk-users] invite authentication error !?

[Sam Govind]

Whatever that remote party is, you are most definitely using
a username/secret declaration for that. So the sip attributes set for that
proxy define the behaviour for this.

On Fri, Sep 30, 2011 at 12:55 PM, cnasterisk  wrote:

> **
>  hi, Sam
> thanks for your kindly reply.
> The remote proxy is not asterisk
>
> 2011-09-30
> --
>  cnasterisk
> ----------
> *>发件人：* Sam Govind
> *>发送时间：* 2011-09-30  15:36:41
> *>收件人：* Asterisk Users Mailing List - Non-Commercial Discussion
> *>抄送：*
> *>主题：* Re: [asterisk-users] invite authentication error !?
>  >What Sip declaration are you using for the remote sip proxy in sip.conf?
>
> On Fri, Sep 30, 2011 at 12:30 PM, Alex Balashov  > wrote:
>
>>  This is just a speculative shot in the dark, but remember that the
>> domain of the From URI is important, and that the authentication "realm"
>> (domain) is part of the authentication credentials.  So, what you have in
>> your 'fromdomain' and 'host' settings on the peer does matter.
>>
>> --
>> This message was painstakingly thumbed out on my mobile, so apologies for
>> brevity, errors, and general sloppiness.
>>
>> Alex Balashov - Principal
>> Evariste Systems LLC
>> 260 Peachtree Street NW
>> Suite 2200
>> Atlanta, GA 30303
>> Tel: +1-678-954-0670
>> Fax: +1-404-961-1892
>> Web: <http://www.evaristesys.com/>http://www.evaristesys.com/
>>
>> On Sep 30, 2011, at 3:16 AM, "cnasterisk"  wrote:
>>
>>   hi,
>>Dear all.
>>I setted a sip account on a sip trunk. when a  client call via this sip
>> trunk, asterisk call failed on this trunk.
>> I have captured the sip messages on the host where asterisk located, and
>> found that:
>>
>> 1. asterisk send a INVITE message to remote sip proxy without
>> "proxy-authorization" field.
>> 2. the remote sip proxy send back a
>> " SIP/2.0 407 Proxy Authentication Required" message.
>> 3. asterisk send a INVITE message with  "proxy-authorization" field.
>> 4. remote proxy send back a "403(Forbidden)" message， that is mean "wrong
>> password"
>>
>> I also tested the sip account on a softphone, it works normal!
>>
>> why this happed？ and how can i solve it?
>>
>>
>>
>> 2011-09-30
>> --
>> kevin
>>
>>  --
>> _
>> -- 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>
>> http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>><http://lists.digium.com/mailman/listinfo/asterisk-users>
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>>
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>> _
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
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Re: [asterisk-users] invite authentication error !?

[Sam Govind]

What Sip declaration are you using for the remote sip proxy in sip.conf?

On Fri, Sep 30, 2011 at 12:30 PM, Alex Balashov
wrote:

> This is just a speculative shot in the dark, but remember that the domain
> of the From URI is important, and that the authentication "realm" (domain)
> is part of the authentication credentials.  So, what you have in your
> 'fromdomain' and 'host' settings on the peer does matter.
>
> --
> This message was painstakingly thumbed out on my mobile, so apologies for
> brevity, errors, and general sloppiness.
>
> Alex Balashov - Principal
> Evariste Systems LLC
> 260 Peachtree Street NW
> Suite 2200
> Atlanta, GA 30303
> Tel: +1-678-954-0670
> Fax: +1-404-961-1892
> Web: http://www.evaristesys.com/
>
> On Sep 30, 2011, at 3:16 AM, "cnasterisk"  wrote:
>
> hi,
>Dear all.
>I setted a sip account on a sip trunk. when a  client call via this sip
> trunk, asterisk call failed on this trunk.
> I have captured the sip messages on the host where asterisk located, and
> found that:
>
> 1. asterisk send a INVITE message to remote sip proxy without
> "proxy-authorization" field.
> 2. the remote sip proxy send back a
> " SIP/2.0 407 Proxy Authentication Required" message.
> 3. asterisk send a INVITE message with  "proxy-authorization" field.
> 4. remote proxy send back a "403(Forbidden)" message， that is mean "wrong
> password"
>
> I also tested the sip account on a softphone, it works normal!
>
> why this happed？ and how can i solve it?
>
>
>
> 2011-09-30
> --
> kevin
>
> --
> _
> -- 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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> _
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>   http://www.asterisk.org/hello
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Re: [asterisk-users] Features not working

[Sam Govind]

Hey,
Whats the output of command "features show" ? on CLI ?


On Fri, Sep 30, 2011 at 1:51 AM, Mike Diehl  wrote:

> Hi all.
>
> I could have sworn this working at one time...
>
> But it doesn't look like any of the functions provided by features.so is
> working for me.  (one-touch monitoring, attended/blind transfer, etc)
>
> I've (re)loaded features.so, as well as bridge_builtin_features.so.
>
> The config file looks sane.
>
> What else should I try?
>
> TIA,
>
> --
>
> Take care and have fun,
> Mike Diehl.
>
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Re: [asterisk-users] record calls of specific agnets

[Sam Govind]

I guess that was this variable like SPYGROUP which needs to be set for
specific extensions and then ask Chanspy to spy on that group. !!

On Thu, Sep 29, 2011 at 9:37 PM, A J Stiles
wrote:

> On Thursday 29 September 2011, Lyle McKarns wrote:
> > Hello Asterisk List!
> >
> > I have been asked to record calls from specific agents, and I am having
> > difficulty finding if this is possible, and if so, how exactly to do it.
>
> Have a "recorded" context and an "unrecorded" context in your dialplan,
> identical save for the lines that start the recording and cleanup processes
> being absent from the latter.  Then set contexts per extension in sip.conf.
>
> --
> AJS
>
> Answers come *after* questions.
>
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Re: [asterisk-users] PSTN connectivity

[Sam Govind]

Hey Warren I thought that these are the complete CLI logs for one call. It
started like "  == Using SIP RTP CoS mark 5" and from-internal priority-1
..So that seemed legit to me. Yeah I too suspect that dialing rules are not
being matched and thats why Gotoif's are failing.

On Thu, Sep 29, 2011 at 8:15 PM, Warren Selby  wrote:

>
> On Thu, Sep 29, 2011 at 7:51 AM, michael k  wrote:
>
>> Thanks for the update. but how do i resolve this issue ? can you help me
>> please ?
>>
>
> You didn't provide a full CLI trace of the outgoing call, you only supplied
> the hangup portion of the call.  Please try again.
>
> Also, what are the dialing rules like in your country?  You only have
> outbound dial patterns setup to handle North American numbers (8+ NXXNXX
> or 8+ NXX).
> The Dial Pattern box in the Outbound Rules box is where you define what
> numbers you want to go out over this trunk.  If you dial a number that
> doesn't match one of these
> patterns, FreePBX is going to look internally for a dial pattern to match
> against, and if it doesn't find one there, it will end the call.
>
>
> --
> Thanks,
> --Warren Selby, dCAP
> http://www.SelbyTech.com 
>
>
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Re: [asterisk-users] PSTN connectivity

[Sam Govind]

Actually its easier. I haven't worked on FreePBX lately so what I remember
is here: You've created a Zap trunk. Focus on the Dial Rule - You can keep
it empty as well. Then you've created an outbound route its dial-rule is
important.

But the funny thing which I didn't mention before is that you've ZAP defined
in FreePBX but actually its DAHDI so I remember they've this cute parameter
in amportal.conf which tells FreePBX to convert ZAP into DAHDI.



On Thu, Sep 29, 2011 at 11:57 AM, michael k  wrote:

> Can you please figure out the configuration issue in my freepbx ?
>
>
>
>
>
> On Thu, Sep 29, 2011 at 11:35 AM, Sam Govind  wrote:
>
>> The Call at this point is not even looking for FXO/Dahdi/Zap.. See the
>> CLI. there is some misconfiguration in FreePBX and your dialled number is
>> not hitting any dial-able rule.  See your FreePBX guide.
>>
>>
>> On Thu, Sep 29, 2011 at 11:01 AM, michael k  wrote:
>>
>>> Hi,
>>>
>>>   Please see the sample.
>>>
>>> A ) Analog HardwareType Ports Action   FXO Ports 1 
>>> Edit<http://192.168.1.134/admin/config.php?type=setup&display=dahdi&dahdi_form=analog_signalling&ports=fxo>
>>>   FXS
>>> Ports --
>>>
>>> B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: *from-analog
>>> *
>>>
>>> *
>>> C ) ZAP Trunk (DAHDI compatibility Mode)*
>>>
>>>
>>> Trunk Description:
>>> Outbound Caller ID:CID Options:
>>>   Maximum Channels:   Disable Trunk:  Disable  Monitor Trunk Failures:
>>> Enable   Outgoing Dial Rules   Dial Rules: 0471+NXX
>>>   Dial Rules Wizards:
>>>   Outbound Dial Prefix:Outgoing Settings   Zap Identifier (trunk
>>> name):
>>>
>>>
>>> *D ) INBOUND route *
>>>
>>>  Description:
>>> Extensions: 199
>>> *
>>>
>>> E ) **OUTBOUND Route*
>>>
>>> Route Name:  9_outside  Route CID:  Override Extension CID  Route
>>> Password:  PIN Set:
>>>  Emergency Dialing:  Intra Company Route:  Music On Hold?
>>>   Dial Patterns
>>> 8|NXXNXX 8|NXX
>>>   Dial patterns wizards*: *
>>>   Trunk SequenceZAP/g0  0
>>> *
>>> F ) In command Line I can see the following things *
>>>
>>>
>>> [root@astrisks ~]# *dahdi_cfg -vv*
>>>
>>>
>>> DAHDI Tools Version - 2.3.0
>>>
>>> DAHDI Version: 2.3.0.1
>>> Echo Canceller(s):
>>> Configuration
>>> ==
>>>
>>>
>>> Channel map:
>>>
>>> Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01)
>>>
>>> 1 channels to configure.
>>>
>>> Setting echocan for channel 1 to none
>>>
>>>
>>> [root@astrisks ~]# *dahdi_scan*
>>>
>>> [1]
>>> active=yes
>>> alarms=OK
>>> description=Wildcard X100P Board 1
>>> name=WCFXO/0
>>> manufacturer=Digium
>>> devicetype=Wildcard X100P
>>> location=PCI Bus 02 Slot 02
>>> basechan=1
>>> totchans=1
>>> irq=193
>>> type=analog
>>> port=1,FXO
>>>
>>>
>>>
>>> *Asterisk CLI*
>>>
>>>
>>> *astrisks*CLI> dahdi show status*
>>>
>>> Description  Alarms  IRQbpviol CRC4   Fra
>>> Codi Options  LBO
>>> Wildcard X100P Board 1   OK  0  0  0  CAS
>>> Unk   0 db (CSU)/0-133 feet (DSX-1)
>>>
>>> *
>>> output when i dialing to a local number*
>>>
>>> Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890)
>>> Verbosity is at least 3
>>>   == Using SIP RTP TOS bits 184
>>>   == Using SIP RTP CoS mark 5
>>> -- Executing [s@from-internal:1] Macro("SIP/199-003a",
>>> "hangupcall") in new stack
>>> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-003a",
>>> "1?skiprg") in new stack
>>> -- Goto (macro-hangupcall,s,4)
>>> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-003a",
>>> "1?skipblkvm") in new stack
>>> -- Goto (macro-hangupcall,s,7)
>>> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-003a",
>>> "1?theend") in new stack
>>> -- Goto (macro-hangupcall,s,9)
>>> -- 

Re: [asterisk-users] PSTN connectivity

[Sam Govind]

The Call at this point is not even looking for FXO/Dahdi/Zap.. See the CLI.
there is some misconfiguration in FreePBX and your dialled number is not
hitting any dial-able rule.  See your FreePBX guide.


On Thu, Sep 29, 2011 at 11:01 AM, michael k  wrote:

> Hi,
>
>   Please see the sample.
>
> A ) Analog HardwareType Ports Action   FXO Ports 1 
> Edit<http://192.168.1.134/admin/config.php?type=setup&display=dahdi&dahdi_form=analog_signalling&ports=fxo>
>   FXS
> Ports --
>
> B) Analog FXO PortsPort 1: *Loop Start* Group: *0* Context: * from-analog*
>
> *
> C ) ZAP Trunk (DAHDI compatibility Mode)*
>
>
> Trunk Description:
> Outbound Caller ID:CID Options:
>   Maximum Channels:   Disable Trunk:  Disable  Monitor Trunk Failures:
> Enable   Outgoing Dial Rules   Dial Rules: 0471+NXX
>   Dial Rules Wizards:
>   Outbound Dial Prefix:Outgoing Settings   Zap Identifier (trunk name):
>
>
>
> *D ) INBOUND route *
>
>  Description:
> Extensions: 199
> *
>
> E ) **OUTBOUND Route*
>
> Route Name:  9_outside  Route CID:  Override Extension CID  Route
> Password:  PIN Set:
>  Emergency Dialing:  Intra Company Route:  Music On Hold?
>   Dial Patterns
> 8|NXXNXX 8|NXX
>   Dial patterns wizards*: *
>   Trunk SequenceZAP/g0  0
> *
> F ) In command Line I can see the following things *
>
>
> [root@astrisks ~]# *dahdi_cfg -vv*
>
>
> DAHDI Tools Version - 2.3.0
>
> DAHDI Version: 2.3.0.1
> Echo Canceller(s):
> Configuration
> ==
>
>
> Channel map:
>
> Channel 01: FXS Loopstart (Default) (Echo Canceler: none) (Slaves: 01)
>
> 1 channels to configure.
>
> Setting echocan for channel 1 to none
>
>
> [root@astrisks ~]# *dahdi_scan*
>
> [1]
> active=yes
> alarms=OK
> description=Wildcard X100P Board 1
> name=WCFXO/0
> manufacturer=Digium
> devicetype=Wildcard X100P
> location=PCI Bus 02 Slot 02
> basechan=1
> totchans=1
> irq=193
> type=analog
> port=1,FXO
>
>
>
> *Asterisk CLI*
>
>
> *astrisks*CLI> dahdi show status*
>
> Description  Alarms  IRQbpviol CRC4   Fra
> Codi Options  LBO
> Wildcard X100P Board 1   OK  0  0  0  CAS
> Unk   0 db (CSU)/0-133 feet (DSX-1)
>
> *
> output when i dialing to a local number*
>
> Connected to Asterisk 1.6.2.11 currently running on astrisks (pid = 2890)
> Verbosity is at least 3
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
> -- Executing [s@from-internal:1] Macro("SIP/199-003a",
> "hangupcall") in new stack
> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-003a",
> "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,4)
> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-003a",
> "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,7)
> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-003a",
> "1?theend") in new stack
> -- Goto (macro-hangupcall,s,9)
> -- Executing [s@macro-hangupcall:9] Hangup("SIP/199-003a", "") in
> new stack
>   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
> 'SIP/199-003a' in macro 'hangupcall'
>   == Spawn extension (from-internal, s, 1) exited non-zero on
> 'SIP/199-003a'
> -- Executing [h@from-internal:1] Macro("SIP/199-003a",
> "hangupcall") in new stack
> -- Executing [s@macro-hangupcall:1] GotoIf("SIP/199-003a",
> "1?skiprg") in new stack
> -- Goto (macro-hangupcall,s,4)
> -- Executing [s@macro-hangupcall:4] GotoIf("SIP/199-0000003a",
> "1?skipblkvm") in new stack
> -- Goto (macro-hangupcall,s,7)
> -- Executing [s@macro-hangupcall:7] GotoIf("SIP/199-003a",
> "1?theend") in new stack
> -- Goto (macro-hangupcall,s,9)
>-- Executing [s@macro-hangupcall:9] Hangup("SIP/199-003a", "") in
> new stack
>   == Spawn extension (macro-hangupcall, s, 9) exited non-zero on
> 'SIP/199-003a' in macro 'hangupcall'
>   == Spawn extension (from-internal, h, 1) exited non-zero on
> 'SIP/199-003a'
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> On Wed, Sep 28, 2011 at 2:50 PM, Sam Govind  wrote:
>
>> Some CLI logs will get you better help on the issue ! also paste the FXO
>> configurations and how you configured it !
>>
>> On Wed, Sep 28, 2011 at 2:11 PM, michael k  wrote

Re: [asterisk-users] PSTN connectivity

[Sam Govind]

Some CLI logs will get you better help on the issue ! also paste the FXO
configurations and how you configured it !

On Wed, Sep 28, 2011 at 2:11 PM, michael k  wrote:

> Hi All,
>
>   I am trying to connect my asterisk box with freepbx to PSTN. I
> have purchased x100p FXO card and installed in my asterisk server. My
> freepbx detected the x100p FXO card and i can see the card specific details
> in command line. I have configured the following things.
>
> 1. OUTBOUND caller id and Dialing rules in Freepbx.
>
> 2. INBOUND route
>
> When i call to the PSTN number before connecting to the FXO card, i am
> getting a ringing. But i get a message like the "number is out of order"
> when i just connect the line to FXO card.
>
> Please some one help me to resolve his issue
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Call does not pass through

[Sam Govind]

terisk -r
> Asterisk 1.6.2.7, Copyright (C) 1999 - 2010 Digium, Inc. and others.
> Created by Mark Spencer  
> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
> details.
> This is free software, with components licensed under the GNU General
> Public
> License version 2 and other licenses; you are welcome to redistribute it
> under
> certain conditions. Type 'core show license' for details.
> =
>   == Parsing '/etc/asterisk/asterisk.conf':   == Found
> Connected to Asterisk 1.6.2.7 currently running on localhost (pid = 11533)
> Verbosity is at least 4
> -- Remote UNIX connection
> localhost*CLI> sip set debug peer cordia
> SIP Debugging Enabled for IP: 66.148.120.167:5060
> localhost*CLI> core set verbose 0
> Verbosity is now OFF
> Audio is at 192.168.254.15 port 19144
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Reliably Transmitting (no NAT) to 66.148.120.167:5060:
> INVITE sip:15707088788@66.148.120.167 SIP/2.0
> Via: SIP/2.0/UDP 192.168.254.15:5060;branch=z9hG4bK1e0698f8;rport
> Max-Forwards: 70
> From: "10.1.129.247" ;tag=as4f38e456
> To: 
> Contact: 
> Call-ID: 1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.2.7
> Date: Wed, 28 Sep 2011 02:43:24 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 235
>
> v=0
> o=root 2082067001 2082067001 IN IP4 192.168.254.15
> s=Asterisk PBX 1.6.2.7
> c=IN IP4 192.168.254.15
> t=0 0
> m=audio 19144 RTP/AVP 0 8
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ---
> localhost*CLI>
> <--- SIP read from UDP:66.148.120.167:5060 --->
> SIP/2.0 100 Trying
> CSeq: 102 INVITE
> Via: SIP/2.0/UDP 222.127.178.113:5060;branch=z9hG4bK1e0698f8;rport
> From: "10.1.129.247" ;tag=as4f38e456
> Call-ID: 1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15
> To: ;tag=2809191014129494936357
> Contact: 
> Content-Length: 0
>
>
> <->
> --- (8 headers 0 lines) ---
> localhost*CLI>
> <--- SIP read from UDP:66.148.120.167:5060 --->
> SIP/2.0 200 OK
> CSeq: 102 INVITE
> Via: SIP/2.0/UDP 222.127.178.113:5060;branch=z9hG4bK1e0698f8;rport
> From: "10.1.129.247" ;tag=as4f38e456
> Call-ID: 1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15
> To: ;tag=2809191014129494936357
> Contact: 
> Content-Type: application/sdp
> Content-Length: 225
>
> v=0
> o=VoipSwitch 6356 7356 IN IP4 66.148.120.167
> s=VoipSIP
> i=Audio Session
> c=IN IP4 66.148.120.167
> t=0 0
> m=audio 6356 RTP/AVP 0 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
>
> <--------->
> --- (9 headers 11 lines) ---
> Found RTP audio format 0
> Found RTP audio format 101
> Found audio description format PCMU for ID 0
> Found audio description format telephone-event for ID 101
> Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0
> (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
> Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x1
> (telephone-event), combined - 0x0 (nothing)
> Peer audio RTP is at port 66.148.120.167:6356
> list_route: hop: 
> set_destination: Parsing  for
> address/port to send to
> set_destination: set destination to 66.148.120.167, port 5060
> Transmitting (no NAT) to 66.148.120.167:5060:
> ACK sip:66.148.120.167:5060;transport=udp SIP/2.0
> Via: SIP/2.0/UDP 192.168.254.15:5060;branch=z9hG4bK28aa746f;rport
> Max-Forwards: 70
> From: "10.1.129.247" ;tag=as4f38e456
> To: ;tag=2809191014129494936357
> Contact: 
> Call-ID: 1a2d18961fc1c50a50ecad427e9f350c@192.168.254.15
> CSeq: 102 ACK
> User-Agent: Asterisk PBX 1.6.2.7
> Content-Length: 0
>
>
> ---
> localhost*CLI>*
>
> Regards,
> Malvin
>
>
> On 9/28/2011 1:46 PM, Sam Govind wrote:
>
> I see a couple of conflicting extensions as well as something I assume
> copy-paste malfunction. Please paste the CLI logs of the call.
>
> On Wed, Sep 28, 2011 at 8:26 AM, Malvin Rito <
> mr...@mail.altcladding.com.ph> wrote:
>
>>  Thanks All. Here is my config:
>>
>> *On my Firewall NAT:*
>>
>> *I allowed the following ports: 4569,5004-5082, 1-2*
>> *
>> On Asterisk Box:*
>>
>> Here is the extensions.conf:
>> *[general]
>> static=yes
>> autofallthrough=yes
>>
>> [avaya-i

Re: [asterisk-users] Call does not pass through

[Sam Govind]

I see a couple of conflicting extensions as well as something I assume
copy-paste malfunction. Please paste the CLI logs of the call.

On Wed, Sep 28, 2011 at 8:26 AM, Malvin Rito
wrote:

>  Thanks All. Here is my config:
>
> *On my Firewall NAT:*
>
> *I allowed the following ports: 4569,5004-5082, 1-2*
> *
> On Asterisk Box:*
>
> Here is the extensions.conf:
> *[general]
> static=yes
> autofallthrough=yes
>
> [avaya-internal]
> exten => s,1,Answer()
> exten => s,2,background(pls-entr-num-uwish2-call)
> exten => s,3,WaitExten()
> exten => s,4,Dial(SIP/${EXTEN})
> exten => s,5,Dial(H323/${EXTEN})
> exten => s,6,PlayBack(vm-nobodyavail)
> exten => s,7,HangUp()
>
> exten => 1000,1,Dial(SIP/1000)
> exten => 1000,1,Answer()
>
> exten => 1000,2,PlayBack(vm-goodbye)
> exten => 1000,3,HangUp()
>
> #Extension for recording
> exten => 9000,1,Answer()
> exten => 9000,2,Background(pm-to-record-phrase)
> exten => 9000,3,Hangup()
> #exten => 9000,3,Wait(2)
> exten => 9000,4,Record(alt_recording%d:ulaw)
> exten => 9000,5,Wait(2)
> exten => 9000,6,Playback(${RECORDED_FILE})
> exten => 9000,7,Wait(2)
> exten => 9000,8,Hangup
>
> exten => _,1,Dial(SIP/${EXTEN}@Avaya)
> exten => _11XX,1,Dial(H323/${EXTEN}@Avaya)
>
> exten => _X,1,Authenticate(/etc/asterisk/passcode.txt,a)
> exten => _X,2,Dial(SIP/${EXTEN}@cordia)
>
> exten => _,1,Authenticate(/etc/asterisk/passcode.txt,a)
> exten => _,2,Dial(SIP/${EXTEN}@cordia)*
>
>
>
> Regards,
> Malvin
>
>
> On 9/26/2011 9:56 PM, Ruben Rögels wrote:
>
> Am 26.09.2011 13:18, schrieb Malvin Rito:
>
>  Hi list,
> My call does not pass through on the first dial, I have to redial again
> the number for the call to pass through. I'm not sure if the problem is
> on my voip proovider or my asterisk server setup. Any thoughts pls?
>
> Regards,
> Malvin
>
>  Hi,
>
> could be a NAT related issue.
>
> Please be more specific about your setup.
>
> best regards,
> Ruben
>
> --
> _
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>
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Re: [asterisk-users] Receiving musinc on hold instead of ring

[Sam Govind]

Very strange indeed! post the dialplan lines as well. Seems like a very
normal Dial command execution. Also complete SIP packets for this particular
behaviour can show some insider. Which version of Asterisk you are using?

On Wed, Sep 28, 2011 at 6:44 AM, Alejandro Recarey wrote:

> Hi all and thanks for reading.
>
> I am having a very strange issue. When dialing out with a certain
> carrier, asterisk 1.6.20 will play music on hold instead of a ring
> tone, although this behaviour is NOT what I want.
>
> Example dialplan execution:
>
> -- Executing [0021266xxx@test:13] Progress("SIP/100-1e04", "") in new
> stack
> -- Executing [0021266xxx@test:14]
> Dial("SIP/100-1e04","SIP/21266xxx@x.x.x.x") in new stack
> -- Called 21266xxx@x.x.x.x
> -- Call on SIP/x.x.x.x-1e05 placed on hold
> -- Started music on hold, class 'default', on SIP/100-1e04
> -- SIP/x.x.x.x-1e05 is making progress passing it to SIP/100-1e04
>
> Now, a SIP packet capture shows no trace of the call being put on hold!
>
> Sample wireshark capture for the same call:
>
> x.x.x.x -> y.y.y.y SIP/SDP Request: INVITE sip:21266xxx@x.x.x.x, with
> session description
> y.y.y.y -> x.x.x.x SIP Status: 100 Giving a try
> y.y.y.y -> x.x.x.x SIP/SDP Status: 180 Ringing, with session description
>
> And I get the music on hold instead of the ringtone. I have tried
> placing Progress() in front of Dial() but to no avail. I do not want
> to use the "r" option in Dial() because then I lose the destination
> ringtone in early media which is important to my customers.
>
> Anybody had a similar issue? Any idea of what parameters I can try to
> tweak, as I am stumped...
>
> Thanks!
>
> Alex
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Set (MONITOR_FILENAME=.................) for queuing recording calls

[Sam Govind]

:P I'd this very similar situation/ project Carl - and guess what. The
filename is created before the call actually hits QUEUE application so these
Queue variables are not populated by then so filename won't contain the
Agent Number.
UNLESS you move the file after queue to a new filename containing the Agent
Number.

like ;

exten => whatever,n,SET(MONITOR_FILENAME=blah-blah)
exten => whatever,n,Queue(${params}); Queue should contain option "c" to
continue in dialplan when callee hangup. Caller hangup case needs special
attention too
exten => whatever,n,System(mv ${old-Filename} ${old-Filename}-
${MEMBERINTERFACE})

I guess this should do the job.

On Tue, Sep 27, 2011 at 8:30 PM, Carlos Chavez wrote:

> On Tue, 2011-09-27 at 03:47 -0700, bilal ghayyad wrote:
> > Dears;
> >
> > I am facing now a problem in the recording the calls that coming via the
> queue, the problem that I am not able to make the filename contains the
> agent (for example its extension) who received the call.
> >
> > Actually by looking to the below settings, it is clear that the agent
> name (it the phone extension or it is sip username .. etc) will not be
> included in the filename.
> >
> > How can I include the agent name in the filename? Because in outboud it
> is easy as the ${CHANNEL} will contain the sip username of the IP Phone but
> in the outbound it will contain the DAHDI channel that the call came via it
> .. so How to inlude the sip username for the IP Phone of the agent that is
> going to get the call from the queue?
> >
> > exten =>
> s,1,Set(MONITOR_FILENAME=${CHANNEL}${CALLERID(num)}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)})
> > exten => s,2,Queue(OrangeCMG,t,,,180)
> > exten => s,3,Macro(voicemail,SIP/reception)
> >
> > Regards
> > Bilal
> >
> >
> ; If set to yes, just prior to the caller being bridged with a queue
> member
> ; the following variables will be set
> ; MEMBERINTERFACE is the interface name (eg. Agent/1234)
> ; MEMBERNAME is the member name (eg. Joe Soap)
> ; MEMBERCALLS is the number of calls that interface has taken,
> ; MEMBERLASTCALL is the last time the member took a call.
> ; MEMBERPENALTY is the penalty of the member
> ; MEMBERDYNAMIC indicates if a member is dynamic or not
> ; MEMBERREALTIME indicates if a member is realtime or not
> ;
> ;setinterfacevar=no
>
> Basically the variable ${MEMBERINTERFACE} will have the extension (if
> using dynamic members) or the agent number.
>
> --
> Telecomunicaciones Abiertas de México S.A. de C.V.
> Carlos Chávez Prats
> Director de Tecnología
> +52-55-91169161 ext 2001
>
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Sam Govind]

Correct me if I'm wrong or don't know anything other than AMI Originate
Event or a call file to kick start a call from asterisk ! So making a new or
modifying asterisk call-file cron job/poller seems like a nice idea but why
put on extra load on Asterisk. (See pbx_spool.c if still want to modify).
The simple idea is create a MySQL trigger for your Table insertion, the data
in the table at insertion time becomes parameters for a simple script that
triggers an AMI event (or call file) whichever is easier for you.

On Tue, Sep 27, 2011 at 6:54 PM, Nick Khamis  wrote:

> Hello David,
>
> At first I assumed asterisk used call files out of the box for
> normal-initiated/instantiated calls however,
> this is incorrect. I think call files was the easy approach for client
> just to place a file with call details
> in some location. I am trying to do the same with a db record. My
> first question is, how does asterisk
> initiate calls, i.e. what part of the source code is responsible for
> that. Are there any threads involved etc.
>
> Cheers,
>
> Nick.
>
> On Tue, Sep 27, 2011 at 9:35 AM, David Moring 
> wrote:
> > Hi Nick,
> >
> > Understand your reasoning - though as Matt points out sql db isn't in the
> > core so compiling it there would preclude seemless upgrades.  Also, I
> > personally would be concerned putting the calls right into the call-file
> > thread might create an issue if you hung on a db query or insert.
>  Finally
> > (and I'd love to hear the answer not knowing), but I believe
> > "normally-initiated/instantiated" calls are handled with direct calls via
> > either SIP requests and/or AMI - thus even using the proposed method, I
> > *think* the db/file-drop method is going to create some overhead that
> might
> > not scale well...
> >
> > Best,
> >
> > David
> >
> > -Original Message-
> > From: Nick Khamis 
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > 
> > Date: Mon, 26 Sep 2011 18:49:07 -0400
> > Subject: Re: [asterisk-users] Asterisk Realtime Time Dial App
> >
> > Hello David,
> >
> > Thank you so much for your response. I am sure it can be easily done
> > using AGI. The reason I am leaning more
> > towards storing the call information in a database record, is because
> > our existing client applications can be easily
> > modified to write to MySQL. The asterisk cron/thread that would
> > querying the DB should be no different than existing implementation
> > used process the call files?
> > For those of you that may be interested in what we are doing. We are
> > developing an application that will apply NLP
> > services on text generated using the speech to text module, and
> > generate the response that will then be forwarded to
> > the text to speech.
> >
> > Cheers,
> >
> > Nick
> > .
> >
> >
> >
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Re: [asterisk-users] number of calls simultaneous from AMI

[Sam Govind]

If you can post any relevant code sections and CLI output for this then
it'll be lot better to determine whats causing this. I never got any problem
initiating as many call as u can say from AMI !

On Tue, Sep 27, 2011 at 5:36 PM, Jerry Geis  wrote:

>  I am starting up 4 calls over the AMI.
> It "appears" as though the first 3 start up and go out right away.
> The 4th call is delayed like 15 seconds.
>
> Any thoughts on why this fourth call might be getting delayed...
>
> Everything is working its just delayed.
>
> Jerry
>
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Re: [asterisk-users] Asterisk Realtime Time Dial App

[Sam Govind]

Hmmm..interesting..I haven't came across anything like this so far..How
about making a new table for the insertion of a new call data..and trigger
some script to activate AMI/Call file according to new call data.

http://dev.mysql.com/doc/refman/5.0/en/faqs-triggers.html#qandaitem-B-5-1-10

On Mon, Sep 26, 2011 at 3:53 AM, Nick Khamis  wrote:

> Hello Everyone,
>
> I have MOH, Sip Friends/Peers, Voice Mail all working realtime. I was
> wondering if it Is possible to have Asterisk make a calls based on a
> record inserted in a table realtime? If I have to develop something using
> AGI
> or AMI, I can do this  with a little direction?
>
> Thanks in Advance,
>
> Nick
>
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Re: [asterisk-users] Asynchronous AGI Problems (Asterisk 1.8.7.0), ubuntu-server

[Sam Govind]

Oh! I was informed that Async:AGI is an AGI that is called in from AMI. Do
tell more about it.

On Sun, Sep 25, 2011 at 5:26 PM, Mehmet Avcioglu wrote:

>
> Actually it doesn't say "AGI(async:script)" it says "AGI(async:agi)" and
> than continues further to setting up an AMI user so the script is executed
> through the manager interface?? Than it says "AGI(agi:async)".?? Well most
> importantly it says "Cons of async AGI: It is the most complex method of
> using AGI to implement." ..:) I have been interested in Async AGI as well
> and after reading your post looked into the link you provided, seems
> different than what we immediately think, a background process.
>
> Perhaps just start the script normally "AGI(script.sh)" and than inside it
> run your background process "background-script.sh > /dev/null 2>&1 <
> /dev/null &" or fork a new process, detach, run in background, etc...
>
> Hopefully somebody else can point us towards the right direction in setting
> up a real asterisk asynchronous AGI application.
>
> --
> Mehmet
>
> On Sep 25, 2011, at 2:00 AM, Randall Degges wrote:
>
> > Hi Everyone,
> >
> > I've been trying to get asynchronous AGIs working in some Asterisk code I
> have. I'm using Asterisk 1.8.7.0, and I'm very familiar with dialplan and
> AGI scripting overall. Here's my problem: I can't get Asterisk to execute
> *any* AGIs asynchronously.
> >
> > Firstly, I discovered asynchronous AGIs via "Asterisk: The Definitive
> Guide". The asynchronous AGI information I read can be found online, here:
> http://ofps.oreilly.com/titles/9780596517342/AGI.html (scroll down to the
> section titled "Async AGI--AMI Controlled AGI").
> >
> > According to the book, since Asterisk 1.6.0 the AGI dialplan application
> has been able to execute AGI scripts asynchronously, via the syntax:
> >
> > exten => s,1,AGI(async:script)
> >
> > According to the book, using the "async:" prefix should have Asterisk run
> the AGI script in the background and instantly continue executing dialplan
> code.
> >
> > So here's my Asterisk dialplan code that's being run:
> >
> > [hangup]
> > exten => s,1,AGI(async:/etc/asterisk/scripts/hangup.py)
> > exten => s,n,Return()
> >
> > Pretty simple context--essentially my AGI script just does some call
> clean up logic before a caller hangs up, talking to a few web servers and
> generating statistics for later usage. What happens when Asterisk executes
> this context, is:
> >
> > WARNING[7911]: res_agi.c:1622 launch_script: Failed to execute
> '/var/lib/asterisk/agi-bin/async:/etc/asterisk/scripts/hangup.py': File does
> not exist.
> >
> > As you can see, Asterisk is ignoring the async: directive, and treating
> it as part of the AGI script path.
> >
> > Is there anyway for me to make asynchronous AGIs work? I've tried
> searching online to no avail.
> >
> > I'd greatly appreciate any responses, thanks for your time.
> >
> > -Randall
> >
> > --
> > Randall Degges
> > http://rdegges.com/
> >
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Re: [asterisk-users] AGI Problem

[Sam Govind]

Should we consider it ignorable or you are still interested in resolving !
Best of Luck in any case !

On Sat, Sep 24, 2011 at 6:16 PM, Mehmet Avcioglu wrote:

>
> Out of about 18000 instances this problem occurred 48 times yesterday. So
> it has been pretty much impossible for me to replicate it under test
> conditions.
>
> Thanks
>
> --
> Mehmet
>
> On Sep 24, 2011, at 4:00 PM, Sam Govind wrote:
>
> Thats wicked !! hmmm stop your asterisk (if u can afford) and run it like
> "asterisk -cvg" and then make a call.. see whats your AGI doing in there
> !!
>
> On Sat, Sep 24, 2011 at 5:43 PM, Mehmet Avcioglu wrote:
>
>>
>> Thanks for the response.
>>
>> Asterisk logs the execution of the AGI and script completed messages
>> within the same second, so less than a second.
>>
>> --
>> Mehmet
>>
>> On Sep 24, 2011, at 3:34 PM, Sam Govind wrote:
>>
>> How much time your AGI is taking? Check if it is completing its task and
>> not killed by asterisk. I guess we've 6~7 seconds before asterisk kills all
>> call channel and related tasks.
>>
>> On Sat, Sep 24, 2011 at 3:21 PM, Mehmet Avcioglu wrote:
>>
>>>
>>> On Sep 23, 2011, at 8:01 PM, Mehmet Avcioglu wrote:
>>> > I have an AGI script that occasionally disappears without completing
>>> its action and asterisk logs the following.
>>> >
>>> >  AGI Script script.php completed,
>>> returning 4
>>> >  Spawn extension (context, 0123456, 2) exited non-zero on
>>> 'Local/0123456@context-f46e;1'
>>>
>>> I also changed the dialplan and added a line to print AGISTATUS, but when
>>> this "returning 4" happens, asterisk stops there and doesn't execute any
>>> further dialplan actions, so I don't even see AGISTATUS value.
>>>
>>> exten => h,1,AGI(script.php,${ANSWEREDTIME},,,)
>>> exten => h,n,NoOp(${AGISTATUS})
>>>
>>> Executing [h@context:1] AGI("Local/0123456@context-4b79;1",
>>> "script.php,13,,,") in new stack
>>> AGI Script script.php completed, returning
>>> 4
>>> Spawn extension (context, h, 1) exited non-zero on
>>> 'Local/0123456@context-4b79;1'
>>
>>
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Re: [asterisk-users] AGI Problem

[Sam Govind]

Thats wicked !! hmmm stop your asterisk (if u can afford) and run it like
"asterisk -cvg" and then make a call.. see whats your AGI doing in there
!!

On Sat, Sep 24, 2011 at 5:43 PM, Mehmet Avcioglu wrote:

>
> Thanks for the response.
>
> Asterisk logs the execution of the AGI and script completed messages within
> the same second, so less than a second.
>
> --
> Mehmet
>
> On Sep 24, 2011, at 3:34 PM, Sam Govind wrote:
>
> How much time your AGI is taking? Check if it is completing its task and
> not killed by asterisk. I guess we've 6~7 seconds before asterisk kills all
> call channel and related tasks.
>
> On Sat, Sep 24, 2011 at 3:21 PM, Mehmet Avcioglu wrote:
>
>>
>> On Sep 23, 2011, at 8:01 PM, Mehmet Avcioglu wrote:
>> > I have an AGI script that occasionally disappears without completing its
>> action and asterisk logs the following.
>> >
>> >  AGI Script script.php completed,
>> returning 4
>> >  Spawn extension (context, 0123456, 2) exited non-zero on
>> 'Local/0123456@context-f46e;1'
>>
>> I also changed the dialplan and added a line to print AGISTATUS, but when
>> this "returning 4" happens, asterisk stops there and doesn't execute any
>> further dialplan actions, so I don't even see AGISTATUS value.
>>
>> exten => h,1,AGI(script.php,${ANSWEREDTIME},,,)
>> exten => h,n,NoOp(${AGISTATUS})
>>
>> Executing [h@context:1] AGI("Local/0123456@context-4b79;1",
>> "script.php,13,,,") in new stack
>> AGI Script script.php completed, returning
>> 4
>> Spawn extension (context, h, 1) exited non-zero on
>> 'Local/0123456@context-4b79;1'
>
>
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Re: [asterisk-users] Set (MONITOR_FILENAME=.................) for queuing recording calls

[Sam Govind]

yes you are right. You should put it before calling in the queue. Set
monitor filename as you want. Also you can set directory in the filename
here too. So when the queu is triggered and mixmonitor starts it'll use that
filename and record inot the file.


On Sat, Sep 24, 2011 at 2:08 AM, bilal ghayyad  wrote:

> Hi All;
>
> I noticed in the queues.conf the configuration for recording the calls in
> the queuing, and regarding to the filename (or any other parameter), it is
> written that I can determine the filename using the command:
>
> Set(MONITOR_FILENAME=foo)
>
>
> But it should be called from the dialing plan, but really i did not
> understand how to call it from the dialing plan.
>
> Well, for example this is my dialing plan to route for the queuing, how I
> can set the filename:
>
> [CustomerSupport]
>
> include => Internal
>
> exten => s,1,Queue(CustomerSupport,t,,,120)
> exten => s,2,Macro(voicemail,SIP/reception)
>
> By the way, I need in the filename to appear the following:
> The SIP username for the IP Phone that the call is routed for it
> The calling number
> The Time of the call
>
> Actually for the outbound recording, I am using the below command (I
> mentioned it to declare the time format I am using and to declare how the
> filename to be named):
>
> exten =>
> _9Z,1,MixMonitor(${CHANNEL}${EXTEN:1}${STRFTIME(${EPOCH},,%Y%m%d%H%M%S)}.wav)
>
> So I hope if someone can help me to write the Set(MONITOR_FILENAME=foo) in
> a way to acheive same format the filename of the recorded outgoing calls (in
> addition that until now I am not able to know where I have to place the
> Set(MONITOR_FILENAME=foo).
>
>
> For example, should I place it as following:
> exten => s,1,Set(MONITOR_FILENAME=.)
> exten => s,2,Queue(CustomerSupport,t,,,120)
> exten => s,3,Macro(voicemail,SIP/reception)
>
> Appreciate if someone help me plz.
> Regards
> Bilal
>
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Re: [asterisk-users] AGI Problem

[Sam Govind]

How much time your AGI is taking? Check if it is completing its task and not
killed by asterisk. I guess we've 6~7 seconds before asterisk kills all call
channel and related tasks.

On Sat, Sep 24, 2011 at 3:21 PM, Mehmet Avcioglu wrote:

>
> On Sep 23, 2011, at 8:01 PM, Mehmet Avcioglu wrote:
> > I have an AGI script that occasionally disappears without completing its
> action and asterisk logs the following.
> >
> >  AGI Script script.php completed,
> returning 4
> >  Spawn extension (context, 0123456, 2) exited non-zero on
> 'Local/0123456@context-f46e;1'
>
>
> I also changed the dialplan and added a line to print AGISTATUS, but when
> this "returning 4" happens, asterisk stops there and doesn't execute any
> further dialplan actions, so I don't even see AGISTATUS value.
>
> exten => h,1,AGI(script.php,${ANSWEREDTIME},,,)
> exten => h,n,NoOp(${AGISTATUS})
>
> Executing [h@context:1] AGI("Local/0123456@context-4b79;1",
> "script.php,13,,,") in new stack
> AGI Script script.php completed, returning 4
> Spawn extension (context, h, 1) exited non-zero on
> 'Local/0123456@context-4b79;1'
>
> --
> Mehmet
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Re: [asterisk-users] Change of default IVR prompt for meetme conference bridge.

[Sam Govind]

UmmmWhen I was a child I replaced the prompts to do that, Now I'd
suggest you to try creating a new directory in /sounds folder like "/en" i.e
"/meetme" and put in corresponding prompts there.
Then just before going into the meetme application change the Language for
the current call in dial plan like this.

exten => le-going-to-meetme,n,SET(CHANNEL(language)=meetme)
same =>  n,Meetme(${some-conf-room},${itsoptions})


Try this and let me know the results. Make sure that the actual sound files
are placed on equal directory level as they are currently in
sounds/en/.wav or w/e

I hope this be of some help to you.

Thanks and regards,
- Sammy


On Thu, Sep 22, 2011 at 7:21 AM, NaJIm  wrote:

> Hi,
>
> Is it possible to change the default voice prompt for Asterisk meet me
> conference bridge. We have our own customized recordings for Welcome and PIN
> request and would like to use that instead of the default "Please enter
> your.. " .
>
> If I replace the default sound file with my custom file by using the
> same filename as the default message, will it affect any other
> applications..??
>
> Thanks,
> Najim
>
>
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Re: [asterisk-users] Add PinCode on my dialplan

[Sam Govind]

See "core show application authe"
If passwords are already the same as those of voicemail.conf go for
application VMAuthenticate() - DIA generates a dial-tone which I don't think
is suitable for dialling out from users(insiders)

  -= Info about application 'Authenticate' =-

[Synopsis]
Authenticate a user

[Description]
This application asks the caller to enter a given password in order to
continue
dialplan execution.
If the password begins with the '/' character,  it is interpreted as a file
which contains a list of valid passwords, listed 1 password per line in the
file.
When using a database key, the value associated with the key can be
anything.
Users have three attempts to authenticate before the channel is hung
up.

[Syntax]
Authenticate(password[,options[,maxdigits[,prompt]]])

[Arguments]
password
Password the user should know
options
a: Set the channels' account code to the password that is entered
d: Interpret the given path as database key, not a literal file
m: Interpret the given path as a file which contains a list of account
codes and password hashes delimited with ':', listed one per line in the
file. When one of the passwords is matched, the channel will have its
account code set to the corresponding account code in the file.
r: Remove the database key upon successful entry (valid with 'd'
only)
maxdigits
maximum acceptable number of digits. Stops reading after maxdigits
have been entered (without requiring the user to press the '#' key).
Defaults to 0 - no limit - wait for the user press the '#' key.
prompt
Override the agent-pass prompt file.

[See Also]
VMAuthenticate(), DISA()


On Wed, Sep 21, 2011 at 9:47 AM, Malvin Rito
wrote:

>  Thanks. ?If I want to use unique PIN for every user that dials out how can
> I implement it using Authenticate app?
>
> Regards,
> Malvin
>
>
> On 9/21/2011 12:42 PM, Sam Govind wrote:
>
> DISA and DB based Auth could be an overkill.
>
> Kyle showed the very simplistic dial plan if Dial-out pin is common for the
> whole system.
> See application *Authenticate(password[,options[,maxdigits[,prompt]]] *and
> if Voicemail PIN are required to be used use application 
> *MAuthenticate([mailbox][@context][,options]
> *
>
>  Regards,
>
>  - Sammy
>
> On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton  wrote:
>
>> Something like this should work:
>>
>>  exten => _011.,1,Answer
>> exten => _011.,n,Wait(1)
>> exten => _011.,n,Read(password,enter-password,5)
>> exten => _011.,n,GotoIf($[${password} = 12345]?5:9)
>>
>>  exten => _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
>> exten => _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
>>
>>  exten => _011.,n,Hangup
>> exten => _011.,n,Playback(invalid)
>> exten => _011.,n,Hangup
>>
>>  Could be cleaned up (the GotoIf isn't very descriptive about where it's
>> going), but it's a starting point.
>>
>>
>>  On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:
>>
>>  Hi List,
>> I currently have a asterisk server running used for dialing-out for IDD
>> but I want to Put a pincode wherein only users with the right pin code will
>> be allowed to dial IDD. Any sample dialplan you can suggest pls?
>>
>> Thanks,
>> Malvin
>> --
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>
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Re: [asterisk-users] RESEND: Mixmonitor command parameter problem on Asterisk 1.8.4

[Sam Govind]

+1 Dale
Alternatively I'd troubles using the MixMonitor() command execution, so what
I did is used System(my commands here) just after the StopMixMonitor().
Using StopMixMonitor() is always recommended to guarantee save the recorded
file and using any commands via System() is easy.

On Wed, Sep 21, 2011 at 6:57 AM, Dale Noll  wrote:

> **
> I am not real familiar with the size of MixMonitor parameters, but just
> looking at the output, I would suggest you change the logic to call a script
> with a single argument.
> something like this,
>
> MixMonitor(${FILENAME},bW(2),/usr/local/bin/convert_to_mp3 ^{FILENAME})
>
> --- /usr/local/bin/convert_to_mp3 --
> #!/bin/bash
>
> WAV=$1
> MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
> /usr/bin/lame "${WAV}" "${MP3}" -b 16 -s 9.6 -m m --bitwidth 8 --lowpass
> 9.6 --resample 8 --lowpass-width 1 && rm -f "${WAV}"
>
> --- end of script ---
> Set the permissions so it is executable by the asterisk owner.
>
> Note:  This has not been tested and is intended as a starting point.
>
>
> Dale
>
>
>
> On 09/20/2011 07:53 PM, Ikka - Mitra Kreasindo wrote:
>
>  Is anyone can help me with this ? I’m really desperate…
>
> ** **
>
> Thx in ad.
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [
> mailto:asterisk-users-boun...@lists.digium.com]
> *On Behalf Of *Ikka - Mitra Kreasindo
> *Sent:* Wednesday, September 14, 2011 5:02 PM
> *To:* 'Asterisk Users Mailing List - Non-Commercial Discussion'
> *Subject:* [asterisk-users] Mixmonitor command parameter problem on
> Asterisk 1.8.4
>
> ** **
>
> Dear all…
>
> ** **
>
> I’m using MixMonitor command in my dialplan, and I used the “command”
> parameter to execute some thing after recording the file.
>
> ** **
>
> I used the command parameter to convert the wav file that created earlier
> to MP3 and than deleted the WAV file.
>
> ** **
>
> It worked fine with asterisk 1.4.21.2. and 1.6x
>
> But than I have a new asterisk server with asterisk 1.8.4. The command
> parameter doesn’t work. It’s trimed for about 297 character only. The rest
> was gone. 
>
> ** **
>
> This is part of the log with Asterisk 1.4.21.2
>
> ** **
>
>   -- Executing [08129981925@speedy:7] MixMonitor("SIP/10001-b7d71bd0",
> "/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav|bW(2)|/usr/bin/lame
> "/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav"
> "/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.mp3"
> -b 16 -s 9.6 -m m --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1
> && rm -f
> "/var/spool/asterisk/recording/speedy/2011/09/14/-08129981925-Ikka_Testing-10001-20110914-163803.wav"")
> in new stack
>
> ** **
>
> This is part of the log with Asterisk 1.8.4
>
> ** **
>
>   -- Executing [08129981925@speedy:7] MixMonitor("SIP/10001-001a",
> "/var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_IT-10001-20110914-165248.wav,bW(2),/usr/bin/lame
> "/var/spool/asterisk/recording/speedy/2011/09/14/ACCOUNT-08129981925-Admin_IT-10001-20110914-165248.wav"
> "/var/spool/asterisk") in new stack
>
> ** **
>
> ** **
>
> As you can see, with 1.8.4 the command paramater is trimed… 
>
> ** **
>
> Is there some changes / bug with MixMonitor in Asterisk 1.8.4 ? Is there a
> quick workaround for this problem ? 
>
> ** **
>
>
> --
> "The truth speaks for itself. I'm just the messenger."
>  Lyta Alexander - Babylon 5
>
>
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Re: [asterisk-users] Add PinCode on my dialplan

[Sam Govind]

DISA and DB based Auth could be an overkill.

Kyle showed the very simplistic dial plan if Dial-out pin is common for the
whole system.
See application *Authenticate(password[,options[,maxdigits[,prompt]]] *and
if Voicemail PIN are required to be used use application
*MAuthenticate([mailbox][@context][,options]
*

Regards,

- Sammy

On Wed, Sep 21, 2011 at 8:32 AM, Kyle Sexton  wrote:

> Something like this should work:
>
> exten => _011.,1,Answer
> exten => _011.,n,Wait(1)
> exten => _011.,n,Read(password,enter-password,5)
> exten => _011.,n,GotoIf($[${password} = 12345]?5:9)
>
> exten => _011.,n,NoOp(Matched _9011 - CheckRec-InternationalCall)
> exten => _011.,n,Dial(SIP/+${EXTEN:3}@outbound)
>
> exten => _011.,n,Hangup
> exten => _011.,n,Playback(invalid)
> exten => _011.,n,Hangup
>
> Could be cleaned up (the GotoIf isn't very descriptive about where it's
> going), but it's a starting point.
>
>
> On Sep 20, 2011, at 8:34 AM, Malvin Rito wrote:
>
> Hi List,
> I currently have a asterisk server running used for dialing-out for IDD but
> I want to Put a pincode wherein only users with the right pin code will be
> allowed to dial IDD. Any sample dialplan you can suggest pls?
>
> Thanks,
> Malvin
> --
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>
>
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Re: [asterisk-users] [Asterisk-Users]Using same extension number for outgoing and incoming both internal and PSTN

[Sam Govind]

Hey,

I don;t think asterisk-guru could've been wrong on this one - possibly
different scenario than your's. Anyway I see what you did there ! There is
no need for separate context for  incoming or outgoing if you don't want.
What you are doing is *exten=>_NXXN,1,Dial(SIP/${EXTEN}@1401**) *
*
*
When you defined the SIp user/peer [1401] you stated context for handling
dial request as "my-office" and when you tried dialling out you told
asterisk to dial the requested number located at 1401 which should've been
@ if calls need to be dialed to gateway and If your
gateway just accepts SIP based (w/o auth) calls.

*exten=>_NXXN,1,Dial(SIP/${EXTEN}@192.168.14.???**)  *
*
*
If your gateway shows attitude in serving direct request you may need to
create user in gateway and telling asterisk to register on Grandstream as a
user and dial-out using that user like.

*exten=>_NXXN,1,Dial(SIP/${EXTEN}@gstream-user**)  *
*
*
There could be more possible alternatives to successfully dial-out using one
context for handling incoming an out going/ preferred is you create separate
contexts.

Regards,
- Sammy

On Tue, Sep 20, 2011 at 8:13 PM, Samuel Sappa  wrote:

> Sorry if this question already asked.
> I'm implementing Voip with asterisk and grandstream gxw4108, according
> from the manual, for connecting with PSTN I must configure one SIP
> account and assign that for dialing the PSTN so in my sip.conf I
> configure SIP account(extension) :
>
> [1401]
> type=friend
> username=1401
> secret=1401
> host=dynamic
> context=my-office
> insecure=port
>
> in my extension.con
> [my-office]
> exten=>1401,1,Dial(SIP/1401,60)
> exten=>_NXXN,1,Dial(SIP/${EXTEN}@1401)
>
> but the problem is when I dial the number for the PSTN it's run/dial
> on internal extension, from the asterisk guru website it's wrote to
> separate the incoming and out going
> in sip.conf
> [1401]
> type=friend
> username=1401
> secret=1401
> host=dynamic
> context=my-office-in
> insecure=port
>
> [1401]
> type=friend
> username=1401
> secret=1401
> host=dynamic
> context=my-office-out
> insecure=port
>
> in extension.conf
> [my-office-in]
> exten=>1401,1,Dial(SIP/1001,60)
> [my-office-out]
> exten=>_NXXN,1,Dial(SIP/${EXTEN}@1401)
>
> but still with this won't work too
> My question it's
> Is it my configuration true/correct or if there any other way for my
> problem
> I'm using 1 Stage Dialing and the asterisk server and Grandstream
> using different IP Address 192.168.101.xxx (for asterisk server) and
> 192.168.14.xxx (for grandstream gateway)
> thank you for helping
> --
> Regards
> Samuel Sappa,
>
> --
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Re: [asterisk-users] oddity with CISCO CCM and Asterisk

[Sam Govind]

Hi Danny,
If you explain some more about this phantom process !! I've never seen
asterisks doing this before. This initiation of a new call is always
dependent upon arrival of an INVITE. I doubt its CCM that is doing some
re-INVITES or sort of keepalive for this call and thus a phantom call is
created !



On Mon, Sep 19, 2011 at 10:58 PM, Danny Nicholas  wrote:

> Hi List,
>
>  I have a system that connects into Asterisk 1.4.41 using CISCO
> CCM 7.  Everything works great except when a call is transferred to the
> operator.  The call goes to the operator via a native bridge and is
> completed, then a “phantom process” starts and tries to launch a new call
> every 15 minutes.  I modified the dialplan to hangup these phantom calls,
> but no still joy.  I get this message:
>
> [Sep 19 10:32:47] NOTICE[14249] chan_sip.c: Call from 'XXX' to extension
> 'X' rejected because extension not found.  
>
> ** **
>
> 14249 is not showing as running, but it is accessible vi pmap 14249
>
> ** **
>
> Could this be a problem with pbx_loopback?
>
> ** **
>
> Thanks
>
> Danny Nicholas
>
> ** **
>
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Re: [asterisk-users] Sip re-register / delay problem.

[Sam Govind]

Hey,

I don't think there could be any solution to this. Even a SIP Proxy...I
don't think so you'll get enough control there to get re-registers from
lagging users only. SIP Timers adjustment on each user level is something
atleast I haven't cam across so far. SIP Timers are global params for all.

btw,
registerattempts = 5
registertimeout = 5

are used when asterisk is registering out to some other server and not for
asterisk clients.

Regards,
-Sammy

On Sun, Sep 18, 2011 at 1:15 PM, Catalin S.  wrote:

> Hello,
>
> Can someone help me with some tips on this?
>
> many thanks
>
>
> On Wed, Sep 14, 2011 at 5:03 PM, Catalin S. wrote:
>
>> Hello,
>>
>> For the moment I have the following settings in my sip.conf. I want to
>> optimize them to archive the following things:
>>
>> - for the moment all my users will re-register too often. I want that only
>> lagged users to re-register quickly.
>> - check from time to time all users but no too often to see if is logged
>> and can be called.
>>
>> Overall i want only lagged users to reregister and users with good
>> response time to be check from time to time.
>>
>> defaultexpiry = 900
>> defaultexpirey = 900
>> maxexpiry = 300
>> maxexpirey = 300
>> minexpiry = 60
>> registerattempts = 5
>> registertimeout = 5
>>  rtpholdtimeout = 900
>> rtptimeout = 60
>> jbmaxsize = 60
>> jbresyncthreshold = 200
>> qualify = yes
>> qualify = 600
>> qualifyfreq = 60
>>
>> Thank you.
>>
>> P.S. If you consider that i use too much options you can tell me what to
>> drop. I use asterisk 1.8.6.0.
>>
>>
>
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Re: [asterisk-users] single registration per user

[Sam Govind]

Hmmm..this could be a complex solution - Use OpenSIPS to handle
registration. On each new register attempt see if a user AOR or other
records exists already - if yes deny registration.

On Mon, Sep 19, 2011 at 1:23 AM, Catalin S.  wrote:

> Hello Eric,
>
> Is about outgoing calls from multiple devices with the same username at
> aprox same time. The overwritten is for incomming calls. I want to prevent
> using the same account in multiple devices at same time. The solution with
> IP will not apply because users may be behind nat or will change everytime
> multiple access points. Do you have any other clues?
>
> Thank you for answers,
> Best regards.
>
>
> On Sun, Sep 18, 2011 at 8:37 PM, Eric Wieling  wrote:
>
>> Asterisk only allows one device per peer to register.  If a 2nd device
>> registers, the first registration is overwritten.
>>
>> You can use permit/deny to limit which IPs a device can register from.
>>
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S.
>> Sent: Sunday, September 18, 2011 4:07 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>> Subject: [asterisk-users] single registration per user
>>
>> Hello,
>>
>> I use asterisk 1.8.6.0 and I have aprox 100 extensions. I want to lock
>> every extension to a single registration per device. Many of users tried to
>> log on my asterisk from 2, 3 devices and I want allow only one.
>> Is there any solution for fix this?
>>
>> Thank you.
>>
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>
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Re: [asterisk-users] DTMF problem

[Sam Govind]

Hey there,

I don't think that its DTMF mode issue ! OP say pressing 9 asterisk ignores
while pressing 6 is OK. Using expensive PBX solutions should be never be the
first priority.

So I'd a similar experience in some asterisk version when I used to enter 2
asterisk always took 3-4 seconds to do anything wheras all other DTMF digits
were working as quickly as DTMF entered.
Since pressing 6 key works fine means this could be more possibly issue with
handset.
Also there is an option in background application "m" where background will
only accept the DTMF whose extens are created in the same context..so if
you've something like this in your dialplan

[test-BKGRND]
exten => s,1,Background(som-sound-file,m)
exten => s,n,Waitexten(20)

exten => 1,1,NOOP(User presssed 1)
exten => 3,1,NOOP(User presssed 1)
exten => 5,1,NOOP(User presssed 1)

Background will act as only recognizing DTMF 1,3,and 5.

even if its DTmfmode issue ...you can one more trick to fix this as well..
use application sipdtmfmode(inband|rfc2833|info)  if call is coming from a
particular caller/UA.

I hope this could be of some help.

Regards,

- Sammy

On Mon, Sep 19, 2011 at 4:51 AM, Zeeshan A Zakaria wrote:

> This DTMF problem has always been there and there is no real solution for
> it, other than using those expensive PBX systems like that from Avaya,
> Cisco, etc. This problem happens when you are sending inband DTMF tones. Via
> softphone you are sending out-of-band DTMF which is basically SIP messages.
>
> --
> Zeeshan A Zakaria
>
> IT Consultant
> www.zeeshanz.com
> 855-ZEESHAN
>
> "asterisk asterisk"  wrote:
>
> >From time to time, I have a DTMF problem. The asterisk cannot recognize
> >my
> >handset key pressed if I press 9 to start with. However, if I press
> >with 6,
> >it is ok.
> >
> >On the other hand, if DMTF key is generated from softphone, it is ok.
> >
> >My dialplan is as follow
> >
> >exten => 1002,1,Answer
> >exten => 1002,n,Wait(2)
> >exten => 1002,n,Background(thank-you-for-calling)
> >exten => 1002,n,Background(vm-enter-num-to-call)
> >exten => 1002,n,WaitExten()
> >exten => 1002,n,Hangup
> >exten => i,1,Background(pbx-invalid)
> >exten => i,2,Goto(1002,1)
> >exten => t,1,Background(vm-goodbye)
> >exten => t,2,Hangup
> >
> >Thanks for the help in advance.
> >--
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>
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Re: [asterisk-users] Message recorder

[Sam Govind]

Hey Steve,

I very much appreciate your reply, I created that dialplan the same day but
after 8~10 hours of working. Yes, I use combination of MixMonitor and
Monitor.
I'll share my logical dialplan here.

It works like: MixMonitor() is used with option a (append) to record all
user recordings in one file. Monitor is only for most recent recording,
overwrites on each retry. This had to be Gosub for user to press * or #. On
* I jump back to Mixmonitor and monitor and wait for user input at
Waitexten(300,m(silence)). on # just concatenate the required Audio with
previously required audio file.

its something like this (psuedo-code)

[recorder-gosub]
exten => s,1,NOOP(I'm trying to record your voice - Previously recorded
final messages are in file ${GOODAUDIO})
same =>  n,SET(ALLAUDIO=bigfilename)
same =>  n,SET(THISAUDIO=small-file)
same =>  n(rerecord),Mixmonitor(${ALLAUDIO},a)
same =>  n,Monitor(${THISAUDIO})
same =>  n,Waitexten(300,m(silence))

exten => i,1,GOTO(*,1)
exten => t,1,GOTO(*,1)

exten => *,1,GOTO(s,rerecord) ; user wants to retry recording message

exten => #,1,StopMonitor()   ; user thinks his message is god enough and
save message now
same => n,StopMixmonitor()
same => n,System(sox ${THISAUDIO} ${GOODAUDIO} ${GOODAUDIO})  ;This will
concatenate this required recording to previously recorded good responses.
same => n,Return()

P.S: Macros are useless ! Use Gosub instead. Calling a Gosub inside a macro
(just to get user DTMF) clears macro stack and leaves you look like
stupified.

Thanks,
- Sammy

On Sun, Sep 18, 2011 at 2:13 AM, Steve Edwards wrote:

> On Sat, 17 Sep 2011, Sam Govind wrote:
>
>  Requirement: Two copies of the recorded message are required.
>>
>
>  [Recorder-A] One will contain only the last message recorded(final)
>>
>
>  [Recorder-B] second one will record all the previous retries of the
>> recording.
>>
>> Once the instruction file is played sound recording will start.
>>
>> Meanwhile recording if user press * Instruction file is played again and
>> message recording: continues for Recorder-B and restarted for Recorder-A.
>>
>> If user presses # meanwhile recording..Save both files and continue to
>> next extensions.
>>
>> I've done sort of combination with Record() application, Mixmonitor,
>> Monitor etc but nothing successful so far !
>>
>
> I don't think you can have a single call recording an 'attempt' while you
> record the entire call at the same time. How about recording each attempt
> separately and concatenating after the fact?
>
> I did something similar a few years ago using an AGI with the 'record file'
> AGI command and exec'ing the 'monitor' and 'stopmonitor' applications.
>
> At the completion of the call, an AGI executed 'normalize' to adjust
> different caller's 'volume' on the individual files, 'sox' to concatenate
> the files, 'ffmpeg' to encode to WMA (client requirement) and 'curl' to
> upload to the client's web site.
>
> FYI, if you anticipate concatenating more than 32 files, you'll need sox
> 14.x instead of sox 12.x included with some distributions.
>
> --
> Thanks in advance,
> --**--**
> -
> Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
> Newline  Fax: +1-760-731-3000
>
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[asterisk-users] Message recorder

[Sam Govind]

Hello List,

Good day. I'm trying to create a message recorder. User will be prompted
with some soundfile (instructions on how to use recorder * to restart and #
to submit).

*Requirement:* Two copies of the recorded message are required.
[Recorder-A] One will contain only the last message recorded(final)
[Recorder-B] second one will record all the previous retries of the
recording.

Once the instruction file is played sound recording will start.

Meanwhile recording if user press * Instruction file is played again and
message recording:  continues for Recorder-B and restarted for Recorder-A.

If user presses # meanwhile recording..Save both files and continue to next
extensions.

I've done sort of combination with Record() application, Mixmonitor, Monitor
etc but nothing successful so far !

Please help.

Thanks.

- Sammy
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Re: [asterisk-users] Temporarily disable DTMF on a call

[Sam Govind]

Hey :)
Smiles on your reply but Its complicated :P
Anyways  I was actually using SayUNixTime() application and found out that
if a digit is pressed it breaks and go to that extension. So I wanted user
to listen to the time but key presses don't do any harm as well.
I've successfully done it now using application SIPdtmfmode(info) whereas it
was rfc2833 just before hitting the sayunixtime() and then after that reset
sipdtmfmode(rf2833)

Like this.

...
exten => s,n,SIPDtmfMode(info)
exten => s,n,SayUnixTime(${params})
exten => s,n,SIPDtmfMode(rfc2833)
...

Now your x-girlfriend will never know if its her phone's fault or you've
done some trick :P

On Fri, Sep 16, 2011 at 11:39 PM, Danny Nicholas  wrote:

> I would do this with ex-girlfriend logic
>
> ** **
>
> [mycontext]
>
> Exten => s,1,playback(instructions)
>
> Exten => s/5551212,n,goto(end)
>
> Exten => s,n,read(var,prompt, …)
>
> Exten => s,n,process..
>
> Exten =>s(end),n,hangup
>
> ** **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Sam Govind
> *Sent:* Friday, September 16, 2011 1:03 PM
> *To:* asterisk-users@lists.digium.com
> *Subject:* [asterisk-users] Temporarily disable DTMF on a call
>
> ** **
>
> Hello List,
>
> ** **
>
> I need help on disabling DTMF from a caller for a specific set of dialplan
> commands and enable DTMF for some other dialplan part. This is not a SIP
> peer - just live incoming call on SIP. 
>
> ** **
>
> Please help.
>
> ** **
>
> Thanks
>
> ** **
>
> -Sammy
>
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[asterisk-users] Temporarily disable DTMF on a call

[Sam Govind]

Hello List,

I need help on disabling DTMF from a caller for a specific set of dialplan
commands and enable DTMF for some other dialplan part. This is not a SIP
peer - just live incoming call on SIP.

Please help.

Thanks

-Sammy
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Re: [asterisk-users] Monitoring second leg being dialed?

[Sam Govind]

The image just don't open for me, a wild from appears and tells me "Domain
blocked bla bla". Try attaching image in this mail.


> Can Dahdi/Asterisk do that? Has anyone used a small Asterisk box at
> home connected to their ADSL modem so that they can make free calls
> from overseas?
>

LOL- Its like asking an army "Have you guys ever worked with guns!! " :P

Please try producing SIP traces so your problem could be identified. Which
asterisk and DAHDI version you are using btw?

On Fri, Sep 16, 2011 at 2:51 PM, Gilles  wrote:

> On Fri, 16 Sep 2011 11:13:16 +0500, Sam Govind 
> wrote:
> >The image you provided didn't open so I'm not sure about the design.
>
> Sorry about that. It's a PNG file and it opens in the two browsers I
> tried.
>
> The reason I don't simply get a subscription with a VoIP provider and
> must go through an Asterisk server + connection to the FXS port is
> that outgoing calls are free, which is nice when calling cellphones,
> especially when travelling.
>
> >  If you can send some SIP flow diagram and Asterisk CLI logs maybe it'll
> help
> >understand the problem.
>
> I haven't done it yet, so have no logs to show.
>
> I'd simply like to hear what's going on channel #2 while Dahdi is
> still dialing, instead of simply being kept waiting.
>
>

>
> Thank you.
>
>
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Re: [asterisk-users] Inter-astersik dialling encounteres no audio

[Sam Govind]

This obviously is pointing to NAT issue. see if you've configured both
asterisk servers with externip= PUBLICIPOFAsterisks.

Studying SIP traces on each console and specially looking at the SDPs in
INVITE will help you find out exact problem. I expect that one of the
asterisk box is sending the audio to its LAN/Private IP whereas it should be
sending RTPs to Public IP of other Asterisk.


On Fri, Sep 16, 2011 at 12:50 PM, Lee, John (Sydney)  wrote:

> **
>
> I have been deploying Asterisk (open source PABX) in the company which I
> work.
>
> So far, all the Asterisk servers do not really talk to each other.
> Recently, I am experimenting to dial from one Asterisk server to another
> through the WAN and I encountered a no-audio problem although the callee's
> phone can ring.
>
> I understand that the no-audio means that SIP traffic (TCP/UDP 5060) is
> allowed to go through but not RTP (UDP 16384-32767).
>
>
>
> Case A
>
> ==
>
> This is a simplified diagram of how I am testing the dialling between 2
> subnets.
>
> In this case, phone A is registered in Asterisk A and phone B is
> registered in Asterisk B.
>
> Phone A <--> Asterisk A <--> Router A <<==>> WAN <<==>> Router B <-->
> Asterisk B <--> Phone B
>
>
>
> Case B
>
> ==
>
> However, before I have tested successfully using this kind of connection.
>
> In this case, phone B1 and B2 are registered in Asterisk B although they
> are on different subnets.
>
> Both phone B1 and B2 can ring and audio is allowed to pass through.
>
> Phone B1 <--> Router A <<==>> WAN <<==>> Router B <--> Asterisk B <-->
> Phone B2
>
>
>
> I am mystified why audio is allowed go through in case B but not case A.
>
>
>
> Can someone be kind enough to help me to understand why I have this
> problem?
>
> If the router is blocking RTP traffic, then why is that I have no audio
> problem in case B?
>
> Thanks in advance.
>
>
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Re: [asterisk-users] Monitoring second leg being dialed?

[Sam Govind]

The image you provided didn't open so I'm not sure about the design. If you
can send some SIP flow diagram and Asterisk CLI logs maybe it'll help
understand the problem.

On Fri, Sep 16, 2011 at 1:28 AM, Gilles  wrote:

> Hello
>
>My ISP provides an FXS port to plug a handset, which can be used to
> make free calls to (GSM) cellphones, similar to the Billion ADSL
> modems:
>
> http://au.billion.com/product/voip.php
>
> My plan is to install an SIP client on a smartphone, so that when I'm
> travelling, I can connect to a good wifi hotspot, register with an
> Asterisk server at home which has an FXO card, tell Asterisk the
> number I wish to dial, and have it dial out through the FXO card and
> the FXS port on the ADSL modem.
>
> Here's the diagram:
>
> http://img844.imageshack.us/img844/3308/asterisksippstncallback.png
>
> Problem is, Dahdi/Zaptel doesn't provide call progression, so that 1)
> when Asterisk passes the call to Dahdi/Zaptel, Asterisk considers the
> call "answered" although there's no actual phone connection yet, and
> 2) Dahdi/Zaptel doesn't trigger an event so we know if the call was
> answered (and if yes, by a live human being rather than an answering
> machine) or if the line is still ringing.
>
> A so-so solution is to simply tell Asterisk to loop through a voice
> message ("This is a call from Joe Allen. Please hit any key and you
> will be connected"), so we know that a human being has answered the
> call, but I was wondering if there were a better solution.
>
> Is it possible for Asterisk to somehow play on channel #1 what's
> happening on channel #2 while Dahdi/Zaptel is actually still dialing,
> so that I handle call progression manually from my cellphone and the
> callee doesn't end up hearing that odd recorded message?
>
> Thank you.
>
>
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Re: [asterisk-users] testing simultaneous calls

[Sam Govind]

A little  look at the dialplan which rings your extension, or get dtmf, and
plays DTMF will help better understand. btw you can set the
context/extension/priority in a call file to skip some priorities of a
particular extension set.

On Fri, Sep 16, 2011 at 12:18 AM, ERIC HERRON  wrote:

> ** **
>
> Asterisk 1.4.26 keeps randomly crashing then restarting itself on my live
> production.
>
> ** **
>
> I cannot run valgrind and I do not have the right flags set in menuselect.
> 
>
> ** **
>
> I can however at the dead of the night run stress tests.
>
> ** **
>
> I want to simulate x-amount of concurrent calls to both a dtmf dialplan,
> which is working, as well as MoH dialplan to see if this could be the cause
> of crashing.
>
> ** **
>
> How do I test this?
>
> Is it a call file that can handle this without ringing my extension first,
> like internal system calling?
>
> ** **
>
> Thanks,
>
> --Eric
>
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Re: [asterisk-users] Mysql dialplan statement not executed

[Sam Govind]

I expect that your same query when executed directly on MySQL console
executes successfully ! Normally errors in DB queries are printed on CLI but
apparently there is nothing wrong.

On Wed, Sep 14, 2011 at 5:51 PM, Jonas Kellens wrote:

> **
> Hello,
>
> I do the following in a macro in the dialplan :
>
> exten => s,n,MYSQL(Connect connid localhost user password AsteriskDB)
> exten => s,n,MYSQL(Query resultid ${connid} UPDATE custDB SET active=1
> WHERE routeID=${ARG1} AND nr=1)
> exten => s,n,MYSQL(Disconnect ${connid})
>
> But nothing changes in my database...
>
> This is the CLI :
>
> [Sep 14 16:41:04] -- Executing [s@macro-bal:15]
> MYSQL("SIP/vc5-000b", "Connect connid localhost user password
> AsteriskDB") in new stack
> [Sep 14 16:41:04] -- Executing [s@macro-bal:16]
> MYSQL("SIP/vc5-000b", "Query resultid 15 UPDATE custDB SET active=1
> WHERE routeID=195 AND nr=1") in new stack
> [Sep 14 16:41:04] -- Executing [s@macro-bal:17]
> MYSQL("SIP/vc5-000b", "Disconnect 15") in new stack
>
> Seems OK, no warnings. But the update has not taken place.
>
>
> Kind regards,
> Jonas.
>
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Re: [asterisk-users] secret=pw in sip.conf affecting inter-asterisk sip call

[Sam Govind]

Hey,

The callee server is complaining too loud "Call from '2765' to extension '*
1166:password*' rejected because *extension not found*."
Try changing the Dial string as DIAL(SIP/asterisk-callee/${EXTEN}) or w/e
extension you require in place of ${EXTEN}
Let me know what changes.

Also this is a good read:
http://www.panoramisk.com/90/sip-trunk-with-asterisk/en/

Wed, Sep 14, 2011 at 12:37 PM, Lee, John (Sydney) wrote:

> I was trying to do a SIP call between two Asterisk servers (1.4.21.2)
>
> ** **
>
> 1) On the caller server, I coded the following in extensions.conf
>
> Dial(SIP/1166:password@asterisk-callee);
>
> ** **
>
> 2) On the callee server, I coded the following in sip.conf
>
> [1166]
>
> type=friend; Friends place calls and receive calls
>
> context=incoming   ; Context for incoming calls from this user
> 
>
> host=dynamic   ; This peer register with us
>
> dtmfmode=rfc2833   ; Choices are inband, rfc2833, or info
>
> qualify=yes; Monitor latency between Asterisk server
> and phone
>
> call-limit=99
>
> username=1166  ; Username to use in INVITE until peer
> registers
>
> secret=password; Normally you do NOT need to set this
> parameter
>
> mailbox=1166@default   ; mailbox 5100 in voicemail context
> .default.
>
> callgroup=1
>
> pickupgroup=1
>
> ** **
>
> The call was unsuccessful as follows.
>
> 
>
> 1) On the caller machine, this is what we got from the console
>
> -- Executing [1166@incoming:1] Dial("SIP/1166-09d81668",
> "SIP/1166:password@asterisk-callee") in new stack
>
> -- Called 1166:password@asterisk-callee
>
> -- SIP/asterisk-callee is circuit-busy
>
>   == Everyone is busy/congested at this time (1:0/1/0)
>
> ** **
>
> 2) On the callee machine, this is what we got from the console,
>
> [Sep 14 14:34:12] NOTICE[11991]: chan_sip.c:14035 handle_request_invite:
> Call from '2765' to extension '1166:password' rejected because extension not
> found.
>
> ** **
>
> However, I found out that if I remove “secret=..” from the SIP entry and
> call without the password, then I will be able to call.
>
> ** **
>
> Any thoughts?
>
> ** **
>
> ** **
>
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Re: [asterisk-users] broadcast

[Sam Govind]

I don't know why you are running into problems.
Once a call file is executed it creates two legs (according to call file
structure) A leg is "Channel: Local/1234@conference" and once it Answers the
call file the second leg is bridged which should be
Context->Extension->priority. So what I'm asking is make your conference
A-leg and your Playback/messages dial plan B-leg.

take a look at the changes I made to your dial-plan

[conference]
exten => 1234,1,Answer()
exten => 1234,n,Gotoif($[${FIRST-CALLER} > 1]?startmsg:pass)
exten => 1234,n(startmsg),System(echo -e
"Channel:local/1234@conference\\nContext: contest-call\\nExtension:
23\\nPriority: 1" > /tmp/${UNIQUEID}.call)
exten => 1234,n,system(mv /tmp/${UNIQUEID}.call
/var/spool/asterisk/outgoing)
exten => 1234,n(pass),Konference(43689956,ADMRSTVL)
exten => 1234,n,Hangup()

[contest-call]
exten => 23,1,Answer()
exten => 23,n,Set(p="/var/spool/asterisk/monitor/")
exten => 23,n,playback(${p}/LQA/12/Biology/Que3)
exten => 23,n,playback(${p}/LQA/12/Biology/Que4)
exten => 23,n,playback(${p}/LQA/12/Biology/Que5)
exten => 23,n,playback(${p}/LQA/12/Biology/Que6)
exten => 23,n,playback(${p}/LQA/12/Biology/Que7)
exten => 23,n,Wait(10)
exten => 23,n,Hangup()

Here, changed your script to what I'm thinking. use the above tweak
accordingly. make sure to find out FIRST-CALLER so your tapes start playing
into conference just for once.

-Sammy

On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati  wrote:

> Hi List,
>
> I make a script for .call file and then I started playback on local channel
> but nothing was hearing at another channles.
>
> exten => 1234,1,Answer()
> exten => 1234,n,System(echo -e "Channel: Channel: 
> local/23@contest-call\\nContext:
> contest-call\\nExtension: 23\\nPriority: 1" > /tmp/${UNIQUEID}.call)
> exten => 1234,n,Konference(43689956,ADMRSTVL)
>
> [contest-call]
>
> exten => _X!,1,Answer()
> exten => _X!,n,Set(p="/var/spool/asterisk/monitor/")
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que3)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que4)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que5)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que6)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que7)
> exten => _X!,n,Konference(43689956,ADMRSTV)
> exten => _X!,n,Wait(10)
> exten => _X!,n,Hangup()
>
> in it I am dialing 1234 from softphone then join to conf in mute mode,
> after it .call file start playback at it's own channels but I am not able to
> hear anything into conf.
>
> As i know localdial is not joining into the conf. but how I will do it so
> that I will be able to hear any played file into conference ?
>
>
>
> On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind  wrote:
>
>> Good to know,
>>
>> I think it'll be a feedback score or a poll from members of the
>> conference. So if you use the R option and collect DTMF from members, and an
>> AMI script listening to that particular DTMF event collects all. This way
>> your AMI listener script should be able to tell you at the end of poll what
>> user inserted with DTMF.
>>
>> So overall insertion of a broadcast message using Ahmed's method of .call
>> file and later on collecting DTMF events from AMI script
>> should theoretically work for you.
>>
>> On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati wrote:
>>
>>> Hi Sam,
>>>
>>> You are right. I am looking for the same
>>>
>>> On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind  wrote:
>>>
>>>> IMHO, I think Bhaati is trying to get feedback from
>>>> multiple conference users. See DTMF options in Konference module.
>>>>  'R' : enable DTMF relay: DTMF tones generate a manager event
>>>>  If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
>>>> members in the conference
>>>>
>>>> While some file is played and users press any DTMF collect the AMI
>>>> events from each user and use them as you require.
>>>>
>>>> Ref:
>>>> http://main.voiptoday.org/index.php?option=com_content&view=article&id=566:asterisk-conferencing-module-appkonference-16-is-now-available&catid=35:general&Itemid=173
>>>>
>>>>
>>>> On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati wrote:
>>>>
>>>>> Hi Ahmed,
>>>>>
>>>>> Konference is also an conferencing application.
>>>>>
>>>>> On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed 
>>>>> wrote:
>>>>

Re: [asterisk-users] broadcast

[Sam Govind]

Virendra,
you need to change your logic just a bit. in call file a Channel is one
which needs to be dialled fires (See
link<http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out>).
this will be an extension where your Konference is Hosted for all the other
callers to join. i.e *Channel: local/s@Konference*

[Konference]
exten => s,1,ANSWER()
exten => s,n,if(conference is already started//do nothing else: trigger the
system command to make a call file...don't forget to move it to outgoing
directory)
exten => s,n,SET(some thing else you need to set for each incoming call i.e
save CallerID etc)
exten => s,n(message),Konference(43689956,ADMRSTV)
exten => s,n,Hangup()

Note that the call file should be triggered only for the first caller and
not every time a participant joins in. That'll case overlap message
broadcasts.

Next thing in call file is the destination which will be playing broadcast
message once Konference is called.

*Context:*broadcast-message
*Extension: *s
*Priority: *1
*
*
[broadcast-message]
exten => s,1,Answer()
exten => s,n,Set(p="/var/spool/asterisk/monitor/")
exten => s,n,playback(${p}/LQA/12/Biology/Que3)
exten => s,n,playback(${p}/LQA/12/Biology/Que4)
exten => s,n,playback(${p}/LQA/12/Biology/Que5)
exten => s,n,playback(${p}/LQA/12/Biology/Que6)
exten => s,n,playback(${p}/LQA/12/Biology/Que7)
exten => s,n,Wait(10)
exten => s,n,Hangup()

This should work and konference should listen to the playbacks.

Regards,
Sammy.

On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati  wrote:

> Hi List,
>
> I make a script for .call file and then I started playback on local channel
> but nothing was hearing at another channles.
>
> exten => 1234,1,Answer()
> exten => 1234,n,System(echo -e "Channel: Channel: 
> local/23@contest-call\\nContext:
> contest-call\\nExtension: 23\\nPriority: 1" > /tmp/${UNIQUEID}.call)
> exten => 1234,n,Konference(43689956,ADMRSTVL)
>
> [contest-call]
>
> exten => _X!,1,Answer()
> exten => _X!,n,Set(p="/var/spool/asterisk/monitor/")
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que3)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que4)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que5)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que6)
> exten => _X!,n,playback(${p}/LQA/12/Biology/Que7)
> exten => _X!,n,Konference(43689956,ADMRSTV)
> exten => _X!,n,Wait(10)
> exten => _X!,n,Hangup()
>
> in it I am dialing 1234 from softphone then join to conf in mute mode,
> after it .call file start playback at it's own channels but I am not able to
> hear anything into conf.
>
> As i know localdial is not joining into the conf. but how I will do it so
> that I will be able to hear any played file into conference ?
>
>
>
> On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind  wrote:
>
>> Good to know,
>>
>> I think it'll be a feedback score or a poll from members of the
>> conference. So if you use the R option and collect DTMF from members, and an
>> AMI script listening to that particular DTMF event collects all. This way
>> your AMI listener script should be able to tell you at the end of poll what
>> user inserted with DTMF.
>>
>> So overall insertion of a broadcast message using Ahmed's method of .call
>> file and later on collecting DTMF events from AMI script
>> should theoretically work for you.
>>
>> On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati wrote:
>>
>>> Hi Sam,
>>>
>>> You are right. I am looking for the same
>>>
>>> On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind  wrote:
>>>
>>>> IMHO, I think Bhaati is trying to get feedback from
>>>> multiple conference users. See DTMF options in Konference module.
>>>>  'R' : enable DTMF relay: DTMF tones generate a manager event
>>>>  If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
>>>> members in the conference
>>>>
>>>> While some file is played and users press any DTMF collect the AMI
>>>> events from each user and use them as you require.
>>>>
>>>> Ref:
>>>> http://main.voiptoday.org/index.php?option=com_content&view=article&id=566:asterisk-conferencing-module-appkonference-16-is-now-available&catid=35:general&Itemid=173
>>>>
>>>>
>>>> On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati wrote:
>>>>
>>>>> Hi Ahmed,
>>>>>
>>>>> Konference is also an conferencing application.
>>>>>
>>>>> On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed 
>>>

Re: [asterisk-users] Asterisk is keep on sending Register request

[Sam Govind]

Hey krishnan,

Everything happens for a reason. The most intuitive cause of this issue
seems to be network change. Can you confirm that no change in networking
happened! because your server is sending register requests but not getting
responses. Meanwhile the same server replying to scenarios2 can imply either
at your or the server end is blocking. There could be NAT issue if its not
firewall.

1- Make sure you can ping from your asterisk server to Registrar server. Do
a traceroute as well.
2- Check for any firewalls in between (could be fail2ban/ iptables)
3- Verify that no network changes occured.
4- Call your service provider and tell them that their server is not talking
to your server any more. :P

best of luck.

-Sammy

On Mon, Sep 12, 2011 at 11:06 PM, Gopal krishnan <
gopalakrishnan...@gmail.com> wrote:

> Hi,
>
> *Scenario 1*
> I am trying to register a VoIP trunk, which is keep on sending the register
> request and I am not getting any response from the SIP Server, this I am
> trying from one network.
>
> *Scenario 2*
> From another network if I try the same VoIP trunk, the account got
> registered.
>
> One thing here to notice is the same account has already been worked in 
> *Scenario
> 1* and now which is not working without any reason.
>
> Any comments would be much appreciated.
>
> Regards
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3

[Sam Govind]

1- *-bash: obd-demo.ulaw: No such file or directory* // try use absolute
file path i.e /usr/src/mymp3.mp3 . I guess that's why its saying no such
file or directory.
2-  http://lists.digium.com/pipermail/asterisk-users/2006-March/144689.html Go
through this thread.
3-  When everything fails from sox - libraries dependencies issues  I use
http://www.nch.com.au/switch/index.html this converter. This can help you
for some time for free.

On Tue, Sep 13, 2011 at 5:12 AM, Kaushal Shriyan
wrote:

> Hi,
>
> Can someone please comment about the below issue
>
> [root@host0040 kaushal]# file obd-demo.mp3
> obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural
> [root@host0040 kaushal]# sox obd-demo.mp3 -e stat
> sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
>
> [root@host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w
> vm-intro.ulaw
> sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
>
> [root@host0040 kaushal]# sox -v 0.125 -V  -t au -r 8000
> -U -b -c 1  resample -ql
> -bash: obd-demo.ulaw: No such file or directory
> [root@host0040 kaushal]# sox -V obd-demo.mp3 -t au -r 8000 -U -b -c 1
> obd-demo.ulaw resample -ql
> sox: Failed reading obd-demo.mp3: Do not understand foReply
> rmat type: mp3
>
> [root@host0040 kaushal]#
>
> When i invoke the same obd-demo.mp3 it works perfectly fine
>
> host0040*CLI> channel originate DAHDI/g0/xx Application
> MP3Player /home/kaushal/obd-demo.mp3
> [Sep  9 16:44:56] DEBUG[12691]: sig_pri.c:985 sig_pri_request:
> sig_pri_request 1
> [Sep  9 16:44:56] DEBUG[12691]: sig_pri.c:6427 sig_pri_call: CALLER NAME:
>  NUM:
>  -- Requested transfer capability: 0x00 - SPEECH
>  -- Launching MP3Player(/home/kaushal/obd-demo.mp3) on
> DAHDI/i1/9833754756-1
>
> [root@host0040 ~]# rpm -qa | grep sox
> sox-12.18.1-1.el5_5.1
> [root@host0040 ~]# rpm -qa | grep lame
> lame-3.98.4-1.el5.rf
> lame-devel-3.98.4-1.el5.rf
> [root@host0040 ~]#
>
>
> MP3 support in  SoX  is  optional
>and requires access to either or both the external
> libmad and libmp3lame libraries.  To see if there is support for Mp3
> run sox -h and
>look for it under the list of supported file formats as
> "mp3".
>
> [root@host0040 ~]# sox -h
> sox: Version 12.18.1
>
> Usage: [ gopts ] [ fopts ] ifile [ fopts ] ofile [ effect [ effopts ] ]
>
> gopts: -e -h -p -q -S -V
>
> fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -v volume -x
>
> effect: avg band bandpass bandreject chorus compand copy dcshift
> deemph earwax echo echos fade filter flanger highp highpass lowp
> lowpass mask mcompand noiseprof noisered pan phaser pick pitch
> polyphase rate repeat resample reverb reverse silence speed stat
> stretch swap synth trim vibro vol
>
> effopts: depends on effect
>
> Supported file formats: aiff al alsa au auto avr cdr cvs dat vms gsm
> hcom la lu maud nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw
> ub ul uw voc vorbis vox wav wve
>
> Which package contains libmad and libmp3lame libraries available on CentOS
> 5.6
>
> Regards,
>
> Kaushal
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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Re: [asterisk-users] Asterisk Manager Interface (AMI)

[Sam Govind]

Hey,

I think I remember the same post before. previously I heard someone telling
to use vicidial or some other thing  like that.But I don't think that those
are totally AMI based call-generators.

What I'd recently done is make a php page which connects to Asterisk's AMI
port. I send page request with destination number as parameter and depending
upon the HTTP arguments it send an ORIGINATE event to Asterisk with the
destination number to be dialled out via DAHDI(PRI) and once the call is
answered bridge it to a local dial plan extension which in term played a
sound-file/message to the connecting number.

So whenever I want Asterisk to initiate a call I send a HTTP request to my
Web-Server(hosting Asterisk) a call originated and played a message. You can
choose your design and directly connect to AMI and keep on sending ORIGINATE
events until you've all 200 channels occupied.

Hope it will help.

On Tue, Sep 13, 2011 at 6:26 AM, Kaushal Shriyan
wrote:

> Hi,
>
> I have 8 E1 PRI Lines and i have 200 phone numbers and 200 channels
> (25 channels per PRI). Can someone please help me understand using
> Asterisk Manager Interface (AMI) available in Asterisk to dial out 200
> numbers and run a campaign for 200 numbers concurrently and play a mp3
> file ?
>
> Please suggest/guide.
>
> Regards,
>
> Kaushal
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] broadcast

[Sam Govind]

Good to know,

I think it'll be a feedback score or a poll from members of the conference.
So if you use the R option and collect DTMF from members, and an AMI script
listening to that particular DTMF event collects all. This way your AMI
listener script should be able to tell you at the end of poll what user
inserted with DTMF.

So overall insertion of a broadcast message using Ahmed's method of .call
file and later on collecting DTMF events from AMI script
should theoretically work for you.

On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati  wrote:

> Hi Sam,
>
> You are right. I am looking for the same
>
> On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind  wrote:
>
>> IMHO, I think Bhaati is trying to get feedback from
>> multiple conference users. See DTMF options in Konference module.
>>  'R' : enable DTMF relay: DTMF tones generate a manager event
>>  If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
>> members in the conference
>>
>> While some file is played and users press any DTMF collect the AMI events
>> from each user and use them as you require.
>>
>> Ref:
>> http://main.voiptoday.org/index.php?option=com_content&view=article&id=566:asterisk-conferencing-module-appkonference-16-is-now-available&catid=35:general&Itemid=173
>>
>>
>> On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati wrote:
>>
>>> Hi Ahmed,
>>>
>>> Konference is also an conferencing application.
>>>
>>> On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed wrote:
>>>
>>>> Hhhmmm..I dunt have any experience with module Konference. Maybe anyone
>>>> else can help you on that. 
>>>>
>>>> ** **
>>>>
>>>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>>>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
>>>> *Sent:* Monday, September 12, 2011 1:28 PM
>>>>
>>>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>>> *Subject:* Re: [asterisk-users] broadcast
>>>>
>>>>  ** **
>>>>
>>>> Hi Ahmed,
>>>>
>>>> I did the same thing earlier to test the load of Digium card. But this
>>>> time I want to play file and want to get some DTMF from all the members of
>>>> conference.
>>>>
>>>> So in this case I need more control into Konference module. But when I
>>>> use .call files then control will not go longer with all events.
>>>>
>>>> Is there any alternate way to do it?
>>>>
>>>> I appreciate your suggestion and will doing in parallel at higher
>>>> priority
>>>>
>>>> On Mon, Sep 12, 2011 at 12:33 PM, Gohar Ahmed 
>>>> wrote:
>>>>
>>>> Make a .call file..join one leg to local extension which plays the file
>>>> and the other leg to conference. The local extension will be like a
>>>> conference member.
>>>>
>>>>  
>>>>
>>>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>>>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
>>>> *Sent:* Monday, September 12, 2011 11:44 AM
>>>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>>>> *Subject:* [asterisk-users] broadcast
>>>>
>>>>  
>>>>
>>>> Hi List,
>>>>
>>>> Is there any way by which I can broadcast any audio file to all members
>>>> into the conference ?
>>>> I don't want to play file individual channels.
>>>>
>>>> -- 
>>>>
>>>>
>>>>
>>>>
>>>> -
>>>> Thanks and regards
>>>>
>>>>  Virendra Bhati
>>>> +91-9172341457
>>>> Software Engineer
>>>>
>>>>  
>>>>
>>>>
>>>> --
>>>> _
>>>> -- 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] broadcast

[Sam Govind]

IMHO, I think Bhaati is trying to get feedback from
multiple conference users. See DTMF options in Konference module.
 'R' : enable DTMF relay: DTMF tones generate a manager event
 If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
members in the conference

While some file is played and users press any DTMF collect the AMI events
from each user and use them as you require.

Ref:
http://main.voiptoday.org/index.php?option=com_content&view=article&id=566:asterisk-conferencing-module-appkonference-16-is-now-available&catid=35:general&Itemid=173


On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati  wrote:

> Hi Ahmed,
>
> Konference is also an conferencing application.
>
> On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed wrote:
>
>> Hhhmmm..I dunt have any experience with module Konference. Maybe anyone
>> else can help you on that. 
>>
>> ** **
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
>> *Sent:* Monday, September 12, 2011 1:28 PM
>>
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* Re: [asterisk-users] broadcast
>>
>> ** **
>>
>> Hi Ahmed,
>>
>> I did the same thing earlier to test the load of Digium card. But this
>> time I want to play file and want to get some DTMF from all the members of
>> conference.
>>
>> So in this case I need more control into Konference module. But when I use
>> .call files then control will not go longer with all events.
>>
>> Is there any alternate way to do it?
>>
>> I appreciate your suggestion and will doing in parallel at higher priority
>> 
>>
>> On Mon, Sep 12, 2011 at 12:33 PM, Gohar Ahmed 
>> wrote:
>>
>> Make a .call file..join one leg to local extension which plays the file
>> and the other leg to conference. The local extension will be like a
>> conference member.
>>
>>  
>>
>> *From:* asterisk-users-boun...@lists.digium.com [mailto:
>> asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
>> *Sent:* Monday, September 12, 2011 11:44 AM
>> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
>> *Subject:* [asterisk-users] broadcast
>>
>>  
>>
>> Hi List,
>>
>> Is there any way by which I can broadcast any audio file to all members
>> into the conference ?
>> I don't want to play file individual channels.
>>
>> -- 
>>
>>
>>
>>
>> -
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-9172341457
>> Software Engineer
>>
>>  
>>
>>
>> --
>> _
>> -- 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
>>
>>
>>
>>
>> -- 
>>
>>
>>
>>
>> -
>> Thanks and regards
>>
>>  Virendra Bhati
>> +91-9172341457
>> Software Engineer
>>
>> ** **
>>
>> --
>> _
>> -- 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
>>
>
>
>
> --
>
>
>
> -
> Thanks and regards
>
>  Virendra Bhati
> +91-9172341457
> Software Engineer
>
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>   http://www.asterisk.org/hello
>
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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Re: [asterisk-users] Call drop in 10 seconds without disconnecting a-party call

[Sam Govind]

Thats goood ! :) thanks for updating.

On Fri, Sep 9, 2011 at 2:16 PM, Ishfaq Malik  wrote:

> Hi
>
> Can you please provide an excerpt of your logs when this happens?
>
> Regards
>
> Ish
>
> On Fri, 2011-09-09 at 09:05 +, Vinod Dharashive wrote:
> > Hi sam,
> >
> > Have solved the problem with your advice. Call drop in 10 seconds without
> disconnecting a-party call. Thank you very much.
> >
> > [TB]
> >
> > exten =>_X.,1,Wait(${INCOMING_WAIT})
> >
> > exten =>_X.,2,Verbose(TB)
> >
> > exten =>_X.,3,Answer()
> >
> > exten =>_X.,4,Set(mainLoop=0)
> >
> > ;exten =>_X.,5,Set(TIMEOUT(absolute)=5)
> >
> > exten =>_X.,5,Playback(/var/callagent/prompts/monitor/thanks)
> >
> > exten => _X.,6,Dial(DAHDI/7/
> >
> > 09501032209,100,L(3[:1][:3000])g)
> >
> > exten =>_X.,7,Noop(${DIALEDTIME})
> >
> > exten =>_X.,8,Goto(TB,_X.,1)
> >
> > exten =>_X.,n,Hangup()
> >
> > Cheers
> > Vinod Dharashive
> > Sent from BlackBerry® on Airtel
> >
> > -Original Message-
> > From: Sam Govind 
> > Sender: asterisk-users-boun...@lists.digium.com
> > Date: Wed, 7 Sep 2011 11:53:33
> > To: Asterisk Users Mailing List - Non-Commercial Discussion<
> asterisk-users@lists.digium.com>
> > Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
> >   
> > Subject: Re: [asterisk-users] (no subject)
> >
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Re: [asterisk-users] Queue agent login notification

[Sam Govind]

you definitely need to create the file extconfig - take sample from
internet. the DB tables need to be created on your own, take help from
internet pages.

On Wed, Sep 7, 2011 at 6:19 PM, Michael  wrote:

> I couldn't find the extconfig.conf file in /etc/asterisk and queue_log
> doesn't exist either (as a file or as a db table). We're using AsteriskNOW,
> so maybe these files/tables were not created.
>
> How should we add them?
>
> Thanks.
>
> On Wed, Sep 7, 2011 at 8:54 AM, Sam Govind  wrote:
>
>> See this link:
>> http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL
>>
>> You'll find similar pages where you can setup to store queue
>> logs/events(as Alex mentioned) in MySQL DB and further do your triggers or
>> functions on them.
>>
>
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Re: [asterisk-users] (no subject)

[Sam Govind]

See absolute timeout. I think yours' a complex thing to achieve I guess
absolute timeout may be the thing that can help. In older versions
absoluteTimeoute(n) could take you to exten T when time n elapsed. now I
guess funtion Timeout() is used as replacement.

here's an excerpt from somewhere:

 ; limit calls to ex-girlfriend to 300 seconds
exten => 123,1,Set(TIMEOUT(absolute)=300)
exten => 123,2,Dial(${EX-GIRLFRIEND})
exten => T,1,Playback(im-sorry)
exten => T,2,Playback(vm-goodbye)
exten => T,3,Hangup(  )


Also see if Dial() command options L(x:y:z), or S(x) work out for you when
combined with option g.

On Wed, Sep 7, 2011 at 7:42 AM, Vinod Dharashive wrote:

> Hi team,
>
> I am trying to find solution to hangup b-party call after 1 min with out
> disconnecting the call of a-party but following dial plan which is
> disconnect both the calls.
>
>
> Please suggest me the solution.
>
> [TB]
>
>
>
> exten => _X.,1,Wait(${INCOMING_WAIT})
>
> exten =>_X.,2,Verbose(TB)
>
> exten =>_X.,3,Answer()
>
> exten => _X.,4,Set(mainLoop=0)
>
> exten => _X.,5,Set(TIMEOUT(absolute)=10); set time in  milliseconds
>
> exten => _X.,6,Playback(/var/callagent/prompts/monitor/thanks)
>
> exten => _X.,7,Dial(DAHDI/7/
>
> 09501032209,10,S(60))
>
>
>
> exten => _X.,8,Noop(${DIALEDTIME})
>
> exten =>_X.,9,Goto(TB,_X.,1)
>
> exten =>_X.,n,Hangup()
>
> Thanks
> Vinod Dharashive
> Sent from BlackBerry® on Airtel
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Re: [asterisk-users] Queue agent login notification

[Sam Govind]

See this link:
http://www.voip-info.org/wiki/view/Asterisk+queue_log+on+MySQL

You'll find similar pages where you can setup to store queue logs/events(as
Alex mentioned) in MySQL DB and further do your triggers or functions on
them.


On Wed, Sep 7, 2011 at 10:46 AM, Michael  wrote:

> We're using FreePBX and I'm wondering if it's possible to add to the
> login/logout macros a command to execute an AGI/Command to launch an
> external process for that.
>
> Thanks.
>
> On Fri, Aug 12, 2011 at 2:30 PM, Alex Vishnev  wrote:
>
>> you can monitor queue_log file for ADDMEMBER or REMOVEMEMBER events. from
>> that point on, you can store them or take any other action.
>> the other way is to use AMI an monitor for Agent login/logoff events
>> On Aug 12, 2011, at 7:06 AM, Michael wrote:
>>
>> > Hello,
>> >
>> > Is there a way to either store login/logout agent information in a
>> database or at least send an email when an agent logs in or out of a queue?
>> >
>> > Thanks,
>> >
>> > Michael
>>
>
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Re: [asterisk-users] Beginner Question: Remote access

[Sam Govind]

There could be as easy solutions as using teamviewer or use tools like
Hamachi used in combination with dyn-dns etc. IP-tunneling I guess needs
static public IPs for the sake of completing the route.

On Wed, Sep 7, 2011 at 5:30 AM, A Dunor  wrote:

> Thanks for the speedy pointer Danny.
>
>
>
> On 9/6/2011 8:27 PM, Danny Nicholas wrote:
>
>> Google for IP-tunneling.
>>
>> -Original Message-
>> From: 
>> asterisk-users-bounces@lists.**digium.com
>> [mailto:asterisk-users-**boun...@lists.digium.com]
>> On Behalf Of A Dunor
>> Sent: Tuesday, September 06, 2011 7:17 PM
>> To: asterisk-users@lists.digium.**com 
>> Subject: [asterisk-users] Beginner Question: Remote access
>>
>> Hello list, I am a beginner at asterisk. I want to access my asterisk box
>> from my laptop, on a different network (mobile hotspot). The asterisk box
>> doesn't have a static ip, how do I connect with it using ssh or other such
>> programs?
>>
>> Thanks for your guidance guys. It is highly appreciated.
>>
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Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

[Sam Govind]

Are you talking about AstDB or MySQL as DB backend for asterisk?

On Mon, Sep 5, 2011 at 1:23 PM, virendra bhati  wrote:

> Hi Sammy,
>
> Thanks for share your experance and provide a new way of Asterisk
> communication with DB.
>
> Actually I am using this feature only for MOH feature of asterisk right
> now. But I will used it to all the configuration files too as per the needs.
>
>
> I am not too much aware abut the Asterisk DB handling that's why I used
> ODBC and mysql connection with asterisk.
>
> One more thing please tell me how to take backup of Asterisk DB ? It is my
> 1st and last things which hurt me.
>
>
>
> On Mon, Sep 5, 2011 at 12:48 PM, Sam Govind  wrote:
>
>> 1- Per my experience I've used DB with configuration files and I was
>> amazed that Asterisk was taking a union of DB + conf file configurations and
>> accepting both.So if you just make a simple script or DB function to do file
>> operation on some event/cronjob you'll be saved.
>>
>> Moreover, if that still may induce duplication into configurations then DB
>> replication and redundancy is the best way to cater your failure case. There
>> are hundreds of how-tos on DB redundancy and failure etc.
>>
>> 2- If you've to move forward with this approach I'll suggest you to read
>> only part of configuration file corresponding to one user i.e [user-1-area]
>> and over-write that part only. If a new user then just append. This way file
>> data loss will be minimized(may even avoided totally).
>>
>> Those were all my suggestions, if anyone else can add valuable comments to
>> this.
>>
>> -
>> sammy
>>
>>
>> On Mon, Sep 5, 2011 at 11:45 AM, virendra bhati wrote:
>>
>>> Hi Sammy,
>>>
>>> Ans of 1st question:-
>>>
>>> As per my experiance Asterisk realtime(DB) based data will lost when
>>> your server is creash and you may not take backup of your server's DB.
>>> If any one know  then plese guide me so that I will start working on it.
>>>
>>> Ans of 2nd question:-
>>>
>>> Your question is correct if more then one user will access these
>>> configuration files then might be some problem will come.
>>>
>>> For this issue I am just make a connection with server then close it
>>> after finishing the job. So problem will be avoided 
>>>
>>>
>>>
>>> On Mon, Sep 5, 2011 at 10:47 AM, Sam Govind  wrote:
>>>
>>>> Though this might have been resolved/accomplished already but I've
>>>> couple of questions for Virendra Bhati.
>>>>
>>>> 1- If you are doing this to make new accounts for new users, why
>>>> couldn't you use Asterisk realtime(DB) based configurations of
>>>> Voicemail/MoH/SIP/dialplan etc wouldn't it be much easier than doing lots
>>>> and lots of filing !?
>>>>
>>>> 2- Since its a web-based Filing operations and if multiple users are to
>>>> use the same page for appending/overwriting their configurations wouldn't 
>>>> it
>>>> lead to information being lost when multiple users applying their changes ?
>>>> I wonder how do you handle that ?
>>>>
>>>> I'm sure I'd more questions when I started writing this response mail
>>>> but now I've forgotten those :P
>>>>
>>>> Thanks,
>>>> Sammy.
>>>>
>>>> On Sun, Sep 4, 2011 at 6:00 PM, Tzafrir Cohen >>> > wrote:
>>>>
>>>>> On Fri, Sep 02, 2011 at 04:58:52PM +0530, virendra bhati wrote:
>>>>> > Hi list,
>>>>> >
>>>>> > I want ot do basic work (add-edit-delete) into asterisk configuration
>>>>> files,
>>>>> > like sip.conf, manager.conf,musiconhold.conf etc.
>>>>> >
>>>>> > Please guide me how to configure all these files from from AMI
>>>>> connection. I
>>>>> > am able to login into AMI from Login action but I want to do more
>>>>> task in to
>>>>> > it.
>>>>> >
>>>>> > *AMI login:- *
>>>>> >
>>>>> > *login.php*
>>>>> >
>>>>> > >>>> > $socket = fsockopen("127.0.0.1","5038", $errno, $errstr, 30);
>>>>> > fputs($socket, "Action: Login\r\n");
>>>>> > fputs($socket, "UserName: root\r\n");
>>>

Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

[Sam Govind]

1- Per my experience I've used DB with configuration files and I was amazed
that Asterisk was taking a union of DB + conf file configurations and
accepting both.So if you just make a simple script or DB function to do file
operation on some event/cronjob you'll be saved.

Moreover, if that still may induce duplication into configurations then DB
replication and redundancy is the best way to cater your failure case. There
are hundreds of how-tos on DB redundancy and failure etc.

2- If you've to move forward with this approach I'll suggest you to read
only part of configuration file corresponding to one user i.e [user-1-area]
and over-write that part only. If a new user then just append. This way file
data loss will be minimized(may even avoided totally).

Those were all my suggestions, if anyone else can add valuable comments to
this.

-
sammy


On Mon, Sep 5, 2011 at 11:45 AM, virendra bhati  wrote:

> Hi Sammy,
>
> Ans of 1st question:-
>
> As per my experiance Asterisk realtime(DB) based data will lost when your
> server is creash and you may not take backup of your server's DB.
> If any one know  then plese guide me so that I will start working on it.
>
> Ans of 2nd question:-
>
> Your question is correct if more then one user will access these
> configuration files then might be some problem will come.
>
> For this issue I am just make a connection with server then close it after
> finishing the job. So problem will be avoided 
>
>
>
> On Mon, Sep 5, 2011 at 10:47 AM, Sam Govind  wrote:
>
>> Though this might have been resolved/accomplished already but I've couple
>> of questions for Virendra Bhati.
>>
>> 1- If you are doing this to make new accounts for new users, why couldn't
>> you use Asterisk realtime(DB) based configurations of
>> Voicemail/MoH/SIP/dialplan etc wouldn't it be much easier than doing lots
>> and lots of filing !?
>>
>> 2- Since its a web-based Filing operations and if multiple users are to
>> use the same page for appending/overwriting their configurations wouldn't it
>> lead to information being lost when multiple users applying their changes ?
>> I wonder how do you handle that ?
>>
>> I'm sure I'd more questions when I started writing this response mail but
>> now I've forgotten those :P
>>
>> Thanks,
>> Sammy.
>>
>> On Sun, Sep 4, 2011 at 6:00 PM, Tzafrir Cohen 
>> wrote:
>>
>>> On Fri, Sep 02, 2011 at 04:58:52PM +0530, virendra bhati wrote:
>>> > Hi list,
>>> >
>>> > I want ot do basic work (add-edit-delete) into asterisk configuration
>>> files,
>>> > like sip.conf, manager.conf,musiconhold.conf etc.
>>> >
>>> > Please guide me how to configure all these files from from AMI
>>> connection. I
>>> > am able to login into AMI from Login action but I want to do more task
>>> in to
>>> > it.
>>> >
>>> > *AMI login:- *
>>> >
>>> > *login.php*
>>> >
>>> > >> > $socket = fsockopen("127.0.0.1","5038", $errno, $errstr, 30);
>>> > fputs($socket, "Action: Login\r\n");
>>> > fputs($socket, "UserName: root\r\n");
>>> > fputs($socket, "Secret: energy\r\n\r\n");
>>> > ?>
>>> > *AMI command:-*
>>> >
>>> > Below commands are for musiconhold.conf. I want to add new MOH context
>>> into
>>> > it.
>>> > >> > include("login.php");
>>> >   fputs($socket, "Action: UpdateConfig\r\n");
>>> >   fputs($socket, "Filename: musiconhold.conf\r\n");
>>> >   fputs($socket, "Srcfilename: musiconhold.conf\r\n");
>>> >   fputs($socket, "Dstfilename: musiconhold.conf\r\n");
>>> >   fputs($socket, "Action-00: newcat\r\n");
>>> >   fputs($socket, "Cat-00: bhavik\r\n");
>>> >   fputs($socket, "mode: files\r\n");
>>> >   fputs($socket, "directory: /var/lib/asterisk/moh\r\n");
>>> >   fputs($socket, "Reload: yes\r\n");
>>> >   fputs($socket, "ActionID: 9873497149817\r\n");
>>> >   fputs($socket, "Action: Logoff\r\n\r\n");
>>>
>>> You're not really editing. You're writing.
>>>
>>> Note the following:
>>>
>>> * It requires Asterisk to be running, and accessible through the manager
>>>  interface.
>>> * asterisk.conf may be in a path that i

Re: [asterisk-users] how to add-edit-delete entery into asterisk conf files

[Sam Govind]

Though this might have been resolved/accomplished already but I've couple of
questions for Virendra Bhati.

1- If you are doing this to make new accounts for new users, why couldn't
you use Asterisk realtime(DB) based configurations of
Voicemail/MoH/SIP/dialplan etc wouldn't it be much easier than doing lots
and lots of filing !?

2- Since its a web-based Filing operations and if multiple users are to use
the same page for appending/overwriting their configurations wouldn't it
lead to information being lost when multiple users applying their changes ?
I wonder how do you handle that ?

I'm sure I'd more questions when I started writing this response mail but
now I've forgotten those :P

Thanks,
Sammy.

On Sun, Sep 4, 2011 at 6:00 PM, Tzafrir Cohen wrote:

> On Fri, Sep 02, 2011 at 04:58:52PM +0530, virendra bhati wrote:
> > Hi list,
> >
> > I want ot do basic work (add-edit-delete) into asterisk configuration
> files,
> > like sip.conf, manager.conf,musiconhold.conf etc.
> >
> > Please guide me how to configure all these files from from AMI
> connection. I
> > am able to login into AMI from Login action but I want to do more task in
> to
> > it.
> >
> > *AMI login:- *
> >
> > *login.php*
> >
> >  > $socket = fsockopen("127.0.0.1","5038", $errno, $errstr, 30);
> > fputs($socket, "Action: Login\r\n");
> > fputs($socket, "UserName: root\r\n");
> > fputs($socket, "Secret: energy\r\n\r\n");
> > ?>
> > *AMI command:-*
> >
> > Below commands are for musiconhold.conf. I want to add new MOH context
> into
> > it.
> >  > include("login.php");
> >   fputs($socket, "Action: UpdateConfig\r\n");
> >   fputs($socket, "Filename: musiconhold.conf\r\n");
> >   fputs($socket, "Srcfilename: musiconhold.conf\r\n");
> >   fputs($socket, "Dstfilename: musiconhold.conf\r\n");
> >   fputs($socket, "Action-00: newcat\r\n");
> >   fputs($socket, "Cat-00: bhavik\r\n");
> >   fputs($socket, "mode: files\r\n");
> >   fputs($socket, "directory: /var/lib/asterisk/moh\r\n");
> >   fputs($socket, "Reload: yes\r\n");
> >   fputs($socket, "ActionID: 9873497149817\r\n");
> >   fputs($socket, "Action: Logoff\r\n\r\n");
>
> You're not really editing. You're writing.
>
> Note the following:
>
> * It requires Asterisk to be running, and accessible through the manager
>  interface.
> * asterisk.conf may be in a path that is not the configuration
>  directory. I'm not sure if this special case is handled.
> * #include are basically handled, but mostly for reading. IIRC the write
>  is back to a single file. No idea about #exec, which will probably
>  have odd interactions with UpdateConfig. Configuration templates
>  ('[section](template)') are also not handled gracefully.
>
> >
> > After doing all no success :((
>
> This is a report of the the thing that did not happen. Next time you ask
> a question, please report what actually does happen ("I got the following
> response: ...").
>
> --
>   Tzafrir Cohen
> icq#16849755  jabber:tzafrir.co...@xorcom.com
> +972-50-7952406   mailto:tzafrir.co...@xorcom.com
> http://www.xorcom.com  iax:gu...@local.xorcom.com/tzafrir
>
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Re: [asterisk-users] Dialing multiple endpoints and CallerID presentation

[Sam Govind]

Alternative work around to this could be:

1- Make two different dialplan extensions. One to dial DAHDI numbers with
setting for DAHDI and other extension for SIP dialing. Both extensions
setting different CallerID presentation
2- Create a queue with Local extensions as static members (strategy=ringall)

So whenever you want to dial to both B, and C location use the queue
dial-out.

I think it should work.


On Mon, Aug 29, 2011 at 12:15 PM, Olivier  wrote:

> Hi,
>
> I've got the following use case where I want to simultaneously dial 2
> endpoints that both need different CallerID presentation.
> How can I do that, from the dialplan preferably ?
>
> For instance, let say phone A needs to both dial B, an internal SIP phone
> and C, a cell phone reachable through a DAHDI span from a an Asterisk system
> where :
> 1. users can use 4-digits short numbers to reach other internal phones.
> 2. calls going out through the DAHDI span, must have CallerIDs presented
> without any prefix.
>
> Ideally, CallerID should be presented :
> 1- with 4-digits for internal phones
> 2- with 10-digits for external phones
> so that both phones can return the call without re-dialing.
>
>
> Suggestions ?
>
> A is 1234 alias DID 051234
> B is 5678
> C is 0123456789
> I was thinking of using something like this:
>
>
> Dial(SIP/5678&DAHDI/g1/0123456789)
>
> What could be  and
> 
>
> Regards
>
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Re: [asterisk-users] Wanted a modified SIP message body

[Sam Govind]

Use the  *SIPAddHeader(Header:Content)* application in dialplan. I don't
think Method specific SIP headers can be done via asterisk.

On Fri, Aug 26, 2011 at 3:05 PM, Jaime Lozano wrote:

> Hello everybody,
> I want Asterisk Server to send packets (SIP packets) to some 3Com
> telephones with the text "TZ: 7200\n" (ie Time Zone = two hours) in the
> message body because 3com PBX sends this variable. I would like to know if I
> it is possible to configure Asterisk to do it, and how.
>
> have a nice day!
>
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