Re: [asterisk-users] Cisco IP 8841 asterisk integration

[Gopalakrishnan N]

Thank you for the information. Actually the phone came with sip firmware. I
tried with TFTP with SEPMAC.CNF.XML and other relevant xml files. But the
phone stuck with blank screen while bootup.

And Cisco TAC support says, the phones part number is with enterprise
firmware and it can't work with Asterisk.

Regards.

On Tue, 6 Dec 2016, 1:40 p.m. Toshaan Bharvani | VanTosh, <
tosh...@vantosh.com> wrote:



On 05/12/16 17:57, Gopalakrishnan N wrote:
> True agree, problem is somehow the people purchased am supporting to
> overcome that. Trying level best... around 20 phones has been
> purchased
I have been able to use the following Cisco IP Phone with Asterisk.
- Cisco SPA303, Cisco SPA504, Cisco 7941 and Cisco 8941
The SPA version just work as they are real SIP clients, however the IP
range, 7xxx, 8xxx, 9xxx are SCCP clients, converting them as you did to
SIP clients, is the Cisco SIP client and runs over TCP, not UDP as a
normal SIP client.
Coming back to your question, you will need to setup a DHCP with BOOTP
and TFTP wit XML distribution, as you have done for the firmware
distribution and Asterisk needs to have SIP wit TCP support.
Additionnaly you need to generate a Cisco SIP XML configuration file and
place it on your TFTP server with the MAC address.
The directory and video I have never been able to get working, however
as a SIP client, it works fine.
I have these phones running with a number of customers in production and
they all work, we actually mix types of phones, as not to require
customers to purchase new IP phones, however sometimes the effort and
time, which translates into cost, is too high.


>
> 
>
>
> On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal, <mefhigos...@gmail.com
> <mailto:mefhigos...@gmail.com>> wrote:
>
> With all the money you plan to invest in firmware, licenses, etc.,
> you have bought a Grandstream IP phone or Yealink...
Better still use a Raspberry Pi as a IP Softphone, or your Android phone
with SIP client.


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> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>

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Re: [asterisk-users] Cisco IP 8841 asterisk integration

[Gopalakrishnan N]

TrueAgree.

:)


On Mon, Dec 5, 2016 at 11:37 PM  wrote:

> > True agree, problem is somehow the people purchased am
> > supporting to overcome that. Trying level best... around 20
> > phones has been purchased
>
> Ah, yes, the "we purchased these without consulting you, but it is up to
> you to make them work" school of thought. It often goes with, "Well, what
> are we paying you for?" and "It's a phone, it shouldn't take you long to
> make it work."
>
> I have to say, unless I am working with a Cisco phone system, Cisco phones
> are not my favorite beasts to work with.
> __
> This email has been scanned by the Symantec Email Security.cloud service.
> For more information please visit http://www.symanteccloud.com
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> Check out the new Asterisk community forum at:
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>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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Re: [asterisk-users] Cisco IP 8841 asterisk integration

[Gopalakrishnan N]

True agree, problem is somehow the people purchased am supporting to
overcome that. Trying level best... around 20 phones has been
purchased



On Mon, 5 Dec 2016, 8:55 p.m. Victor Villarreal, 
wrote:

> With all the money you plan to invest in firmware, licenses, etc., you
> have bought a Grandstream IP phone or Yealink...
> --
> _
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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Cisco IP 8841 asterisk integration

[Gopalakrishnan N]

Actually now I have the phones with SIP firmware.  I will try with 3pcc
firmware along with XML files.

Or any idea if we have CUCM application can we change the firmware. am
ready to buy the developer edition.

Regards .

On Mon, 5 Dec 2016, 3:34 p.m. Steve Davies, <davies...@gmail.com> wrote:

> I tried... repeatedly... I failed. I bought some 3PCC phones, and they
> just worked.
>
> If you have the relevant Cisco telephony server product you might be able
> to trick it into doing what you want, as that has the proper upgrader for
> that model of phone.
>
> I previously had experience of upgrading the Cisco build to the SIP build
> on Cisco 7641 handsets, which have 2 similar builds, but none of the
> techniques seemed to apply this time around.
>
> Cheers,
> Steve
>
>
> On Sun, 4 Dec 2016 at 16:03 Gopalakrishnan N <gopalakrishnan...@gmail.com>
> wrote:
>
> Can't I upload the 3PCC firmware ? available from the Cisco website?
>
> Actually it came with sip88xx firmware.
>
> Regards .
>
>
> On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, <davies...@gmail.com> wrote:
>
> Hi,
>
> You have to buy the 3PCC version for this to work. Once you have this,
> they work very much like the Cisco SPA handsets.
>
> I also ended up with a non-3PCC handset and it is useless, and as far as I
> can tell they cannot be re-flashed.
>
> Cheers,
> Steve
>
>
>
> On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan N <gopalakrishnan...@gmail.com>
> wrote:
>
> Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have
> the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to
> upload woth TFTP due to some reason it's getting failed. Do I need to load
> 3pcc firmware  or anyway to Configure from the phone itself or from the
> GUI?
>
> I have the SEPMAC.cnf.xml as well.
>
> Any suggestions would be appreciated.
>
> Regards .
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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Re: [asterisk-users] Cisco IP 8841 asterisk integration

[Gopalakrishnan N]

Can't I upload the 3PCC firmware ? available from the Cisco website?

Actually it came with sip88xx firmware.

Regards .

On Fri, 2 Dec 2016, 10:38 p.m. Steve Davies, <davies...@gmail.com> wrote:

> Hi,
>
> You have to buy the 3PCC version for this to work. Once you have this,
> they work very much like the Cisco SPA handsets.
>
> I also ended up with a non-3PCC handset and it is useless, and as far as I
> can tell they cannot be re-flashed.
>
> Cheers,
> Steve
>
>
>
> On Fri, 2 Dec 2016 at 16:16 Gopalakrishnan N <gopalakrishnan...@gmail.com>
> wrote:
>
> Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have
> the phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to
> upload woth TFTP due to some reason it's getting failed. Do I need to load
> 3pcc firmware  or anyway to Configure from the phone itself or from the
> GUI?
>
> I have the SEPMAC.cnf.xml as well.
>
> Any suggestions would be appreciated.
>
> Regards .
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
> --
> _
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
> New to Asterisk? Start here:
>   https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>http://lists.digium.com/mailman/listinfo/asterisk-users
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[asterisk-users] Cisco IP 8841 asterisk integration

[Gopalakrishnan N]

Anyone tried integrating Cisco IP 8841 phone with Asterisk 11.x. I have the
phone with sip firmware came along with sip88xx-11.0.1SR xx. I tried to
upload woth TFTP due to some reason it's getting failed. Do I need to load
3pcc firmware  or anyway to Configure from the phone itself or from the
GUI?

I have the SEPMAC.cnf.xml as well.

Any suggestions would be appreciated.

Regards .
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[asterisk-users] Cisco IP phone serup

[Gopalakrishnan N]

Hi,

I have cisco 8841 IP phone. could someone light up how to configure with
Asterisk.

Thanks in advance.

Regards,
Gopal .
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Re: [asterisk-users] NAT on IPsec Tunnel

[Gopalakrishnan N]

Finally got it worked, the issue was E164 callerid format, where i set it
up, after removing the E164 format its was thru.

Regards

On Fri, Feb 12, 2016 at 9:31 PM Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Now incoming works fine, this is because of my SonicWALL firmware issue,
> tried with different SonicWALL inbound works.
>
> But for outbound am getting 408 request time out error in the NAT on VPN
> tunnel.
>
> On Fri, Feb 12, 2016 at 3:50 AM Gopalakrishnan N <
> gopalakrishnan...@gmail.com> wrote:
>
>> Hi all,
>>
>> Am using Asterisk 11.2.1. And for site testing, Verizon is doing Interop
>> testing with site to site IPsec tunnel and with public IP over the tunnel.
>>
>> Problem is when I do an inbound call, only a IVR message plays, whereas
>> am not able to transfer a call to extension, or dtmf not even works and
>> even outbound getting 408 request timeout.
>>
>> By all means I have configured externaddr and localnet in my sip.conf.
>>
>> Verizon says still the contact information shows my private IP, even
>> though I configured externaddr.
>>
>> When I route the incoming call directly to an hardphone extension, am not
>> able to answer the call in the hardphone, the ring LED keeps in blinking
>> even though i pickup the receiver, which is strange i haven;t seen.
>>
>> Can someone had any of this issue or throwing out any information would
>> help me.
>>
>> *Attached PCAP File:*
>> InboundCall_Direct_Extension - not able to answer in the hardphone
>> InboundCall_DTMF - Inbound call plays a message and wait for DTMF, where
>> its not recognized
>>
>> Tks.
>>
>
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Re: [asterisk-users] NAT on IPsec Tunnel

[Gopalakrishnan N]

Now incoming works fine, this is because of my SonicWALL firmware issue,
tried with different SonicWALL inbound works.

But for outbound am getting 408 request time out error in the NAT on VPN
tunnel.

On Fri, Feb 12, 2016 at 3:50 AM Gopalakrishnan N <
gopalakrishnan...@gmail.com> wrote:

> Hi all,
>
> Am using Asterisk 11.2.1. And for site testing, Verizon is doing Interop
> testing with site to site IPsec tunnel and with public IP over the tunnel.
>
> Problem is when I do an inbound call, only a IVR message plays, whereas am
> not able to transfer a call to extension, or dtmf not even works and even
> outbound getting 408 request timeout.
>
> By all means I have configured externaddr and localnet in my sip.conf.
>
> Verizon says still the contact information shows my private IP, even
> though I configured externaddr.
>
> When I route the incoming call directly to an hardphone extension, am not
> able to answer the call in the hardphone, the ring LED keeps in blinking
> even though i pickup the receiver, which is strange i haven;t seen.
>
> Can someone had any of this issue or throwing out any information would
> help me.
>
> *Attached PCAP File:*
> InboundCall_Direct_Extension - not able to answer in the hardphone
> InboundCall_DTMF - Inbound call plays a message and wait for DTMF, where
> its not recognized
>
> Tks.
>
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[asterisk-users] Asterisk SIP UUI Protocol

[Gopalakrishnan N]

Hi,

I came thru ISDN UUI (User-User Information) protocol which is defined in
this RFC - http://www.ietf.org/id/draft-ietf-cuss-sip-uui-17.txt

But I don't understand how to use this with Asterisk. Any idea would be
much appreciated.

Thanks.
Gopal.
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Re: [asterisk-users] log caller hangup events

[Gopalakrishnan N]

Logically yes, once the call hangup, the hangup handler will execute.

Regards,


On Mon, Aug 18, 2014 at 7:04 PM, Paul Greenberg p...@greenberg.pro wrote:

  Hi,


  I am mostly concerned with inbound calls.

 Would it work the same?


  Regards,

 Paul
  --
 *From:* asterisk-users-boun...@lists.digium.com 
 asterisk-users-boun...@lists.digium.com on behalf of Gopalakrishnan N 
 gopalakrishnan...@gmail.com
 *Sent:* Monday, August 18, 2014 4:13 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] log caller hangup events

  Hi,

  You can use Hangup handler. May be this post can you help you,
 http://gblades.blogspot.in/2013/07/how-to-get-sip-response-code-in.html

  Regards


 On Mon, Aug 18, 2014 at 9:45 AM, Paul Greenberg p...@greenberg.pro
 wrote:

  All,


  I would like to log a message whenever a party hangs up a call or
 session, i.e. no Dial(), user drops off a menu. The message should include
 the length of the user's session, the session's start time, and called ID.


  Theoretically, I could set up a channel variable and then ...


  Any advice would be most welcome!


  Regards,

 Paul

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Re: [asterisk-users] Question about SIP Dial

[Gopalakrishnan N]

It supposed to be like this Dial(SIP/${EXTEN}#ip.add.re.ss)

Regards


On Fri, Aug 15, 2014 at 6:20 AM, CDR vene...@gmail.com wrote:

 In channel PJSIP I use this format
 Dial(PJSIP/endpoint/sip:${EXTEN}@ip.add.re.ss)
 what would be the equivalent of this format in old SIP?
 I tried
 Dial(SIP/peer/${EXTEN}@ip.add.re.ss)
 but it does not work. I just cannot embed the IP address in the peer's
 definition, but I need to use some other configuration features that
 are unique to each peer.

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Re: [asterisk-users] log caller hangup events

[Gopalakrishnan N]

Hi,

You can use Hangup handler. May be this post can you help you,
http://gblades.blogspot.in/2013/07/how-to-get-sip-response-code-in.html

Regards


On Mon, Aug 18, 2014 at 9:45 AM, Paul Greenberg p...@greenberg.pro wrote:

  All,


  I would like to log a message whenever a party hangs up a call or
 session, i.e. no Dial(), user drops off a menu. The message should include
 the length of the user's session, the session's start time, and called ID.


  Theoretically, I could set up a channel variable and then ...


  Any advice would be most welcome!


  Regards,

 Paul

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[asterisk-users] Concurrent Calls via Manager Originate

[Gopalakrishnan N]

Can we have concurrent calls via asterisk manager interface, lets say
around 1000 or 1000+ concurrent calls.

Regards
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Re: [asterisk-users] PAGI

[Gopalakrishnan N]

Thanks Johan. Are you using this application for any credit card processing?


On Fri, Apr 4, 2014 at 5:29 PM, Johan Wilfer li...@jttech.se wrote:

 2014-04-03 18:58, Gopalakrishnan N skrev:

  Hi,

 Anybody using PAGI scripts,
 http://marcelog.github.io/articles/pagi_tutorial_create_
 voip_telephony_application_for_asterisk_with_agi_and_php.html

 Would like to know the feasibility to build a IVR solutions.

 Regards



 I use PAMI, and it works great. PAGI seems to be a sister-project for AGI.

 https://github.com/marcelog/PAMI
 https://github.com/marcelog/PAGI


 --
 Johan Wilfer


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[asterisk-users] PAGI

[Gopalakrishnan N]

Hi,

Anybody using PAGI scripts,
http://marcelog.github.io/articles/pagi_tutorial_create_voip_telephony_application_for_asterisk_with_agi_and_php.html

Would like to know the feasibility to build a IVR solutions.

Regards
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[asterisk-users] Enterprise VoIP Trunk

[Gopalakrishnan N]

Am looking for a service provider who can provide enterprise SIP trunk with
100 channels concurrent sessions.

I see some like Inphonex, Broadvoice... and etc

Is there any suggestions for the service providers.

Regards
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Re: [asterisk-users] SIP OPTIONS storm?

[Gopalakrishnan N]

SIP options message is due to check the peer registration is keepalive. As
per my understanding it might be because of network flap may be wireshark
trace can give you any clue.

Regards
On 13 Feb 2014 23:41, Tim Nelson tnel...@rockbochs.com wrote:

 Greetings-

 I recently experienced an odd situation. I have an Asterisk 11.5.0 system
 (Box A) with a SIP peering to another Asterisk 1.8.23.0 system (Box B). At
 some point, Box A started sending over 65Mbps of SIP OPTIONS packets to Box
 A. I do have qualify=yes for the peer on both sides, and the qualifyfreq is
 not set (aka default of 60secs).

 Of course, logs on Box A were not set to show debug info, so there is no
 indication of a problem. Logs on Box B show no issues, only at a very
 specific start time, there are suddenly tons of:

 [2014-02-13 00:12:50] DEBUG[31516] chan_sip.c: Allocating new SIP dialog
 for 2a338cf5518531e31190bd4b7826d137@x.y.z.166:5060 - OPTIONS (No RTP)

 I've done quite a bit of searching, but am not finding anything of
 consequence. Also, the Asterisk changelogs are not providing anything that
 would indicate this is known and fixed, at least for the 11.x branch.

 Thoughts/suggestions? Thanks!

 --Tim

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Re: [asterisk-users] Asterisk Crash in asterisk 10.0.0.

[Gopalakrishnan N]

Enable debugging module and backtrace and re-compile so that you will
bactrace of the crash logs.

Regards
On 14 Feb 2014 10:29, Arun Ram arunram@gmail.com wrote:

 Hi guys,
  I need a desperate help from you regarding this asterisk crash issue.



 On Thu, Feb 13, 2014 at 5:48 PM, Arun Ram arunram@gmail.com wrote:

 Hi,

 I  am facing asterisk crash issue  in my  Asterisk 10.0.0. safe 
 asteriskgenerated a core dump in  /tmp path . I  viewed the core dump using
 viewcore in linux.

  *can anyone tell the reason for the crash .  waiting eagerly for an
 answer from asterisk support guys*.* please the find the core dump
 attachment too* ..


 *Below is the information in core dump *

 --


 *Thanks  RegardsArunram.c*


 *The Power of someone has the power to do something.. anything !!*




 --


 *Thanks  RegardsArunram.c*


 *The Power of someone has the power to do something.. anything !!*

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[asterisk-users] Voice XML Asterisk Integration

[Gopalakrishnan N]

Which is the best way around to integrate Asterisk with VoiceXML like
VoiceGlue...! Am using Asterisk 11.2.1.

Regards.
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Re: [asterisk-users] Integration of OpenVXI

[Gopalakrishnan N]

Anyone using Voiceglue with latest Asterisk 11.6 certified version?


On Mon, Jun 20, 2011 at 10:00 PM, Jean-Denis Girard jd.gir...@sysnux.pfwrote:

 -BEGIN PGP SIGNED MESSAGE-
 Hash: SHA1

 Hi,

 Le 20/06/2011 04:40, Gopal krishnan a écrit :
  Have anybody integrated
  OpenVXI http://www.speech.cs.cmu.edu/openvxi/ with Asterisk?

 Voiceglue works for me: http://www.voiceglue.org/


 Thanks,
 - --
 Jean-Denis Girard

 SysNux  Systèmes  Linux  en Polynésie française
 http://www.sysnux.pf/   Tél: +689 50 10 40 / GSM: +689 79 75 27
 -BEGIN PGP SIGNATURE-

 iEYEARECAAYFAk3/dbgACgkQuu7Rv+oOo/hemACdEN4qLhxLl9LJGpdGIfd8zZ0B
 PAsAnRxitrzwt5RhWPeo/iwVuYqfeKNh
 =LpwD
 -END PGP SIGNATURE-

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Re: [asterisk-users] Change the preferred audio playback format

[Gopalakrishnan N]

Hope basically depends on the codec Asterisk will playback the file
automatically
On 23 Jan 2014 19:25, Gareth Blades mailinglist+aster...@dns99.co.uk
wrote:

 On 23/01/14 13:38, Ishfaq Malik wrote:

 Hi

 Is there any way to change the preferred audio playback format in
 asterisk (I'm using 1.8.25.0)
 i.e. first check for gsm, if doesn't exits then check for slin?


 It should pick whichever source format requires the least cpu to transcode
 into the desired output format.
 So generally that means if there is a source available in the same format
 as the output then it will use it otherwise it will use slin etc...


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Re: [asterisk-users] AGI Script not working

[Gopalakrishnan N]

Library is Asterisk Perl library and module DBI. The same script working in
different machine with same Asterisk version and same Perl version. Am able
to see Tx and Rx from script.




On Mon, Dec 2, 2013 at 8:08 AM, Eric Wieling ewiel...@nyigc.com wrote:

 Sounds like you are violating the AGI protocol.   Which Perl AGI library
 are you using?


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N
 Sent: Saturday, November 30, 2013 1:27 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] AGI Script not working

 I have a Perl AGI script updating some values to database like recorded
 file path, unique ID and callerid. When I run the script with test
 dialplan, its not updating to database.

 Whereas database connection is fine, when I run agi debug I see only Tx
 packets not Rx packets, firewall is also OFF.

 Any other specific reason why there is no Rx.

 The same script working in one more Asterisk machine.

 Regards

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Re: [asterisk-users] AGI Script not working

[Gopalakrishnan N]

Thanks... I got it working actually I found with this command /usr/bin/perl
-d agi file name from this I got to know that my library is missing and
installed Asterisk-perl module and now its fine.

Once again thank you.


On Mon, Dec 2, 2013 at 3:05 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 Library is Asterisk Perl library and module DBI. The same script working
 in different machine with same Asterisk version and same Perl version. Am
 able to see Tx and Rx from script.




 On Mon, Dec 2, 2013 at 8:08 AM, Eric Wieling ewiel...@nyigc.com wrote:

 Sounds like you are violating the AGI protocol.   Which Perl AGI library
 are you using?


 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Gopalakrishnan N
 Sent: Saturday, November 30, 2013 1:27 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] AGI Script not working

 I have a Perl AGI script updating some values to database like recorded
 file path, unique ID and callerid. When I run the script with test
 dialplan, its not updating to database.

 Whereas database connection is fine, when I run agi debug I see only Tx
 packets not Rx packets, firewall is also OFF.

 Any other specific reason why there is no Rx.

 The same script working in one more Asterisk machine.

 Regards

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Re: [asterisk-users] Answering agent

[Gopalakrishnan N]

Alao enable cel table that will have all the information
On 29 Nov 2013 23:25, Todd R. tjrl...@live.com wrote:

 I do this by writing custom CDR. I write the agents extension write into
 the CDR records. This makes is easy to just parse through the CDR and get
 all the info you need about the call.

 Google something like asterisk custom CDR




  On Nov 29, 2013, at 11:36 AM, Leandro Dardini ldard...@gmail.com
 wrote:
 
  Hello friends,
  when a call arrives in the queue, a CDR record is created, but there is
 no info about which agent has picked up the call. I can find that info only
 in queue_log.
 
  Is there a way to have that info in the CDR or maybe in a variable in
 the h context, when the call is ended?
 
  Leandro
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[asterisk-users] AGI Script not working

[Gopalakrishnan N]

I have a Perl AGI script updating some values to database like recorded
file path, unique ID and callerid. When I run the script with test
dialplan, its not updating to database.

Whereas database connection is fine, when I run agi debug I see only Tx
packets not Rx packets, firewall is also OFF.

Any other specific reason why there is no Rx.

The same script working in one more Asterisk machine.

Regards
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[asterisk-users] Channel not releasing immediately for Attended Transfer

[Gopalakrishnan N]

I have a situation where Asterisk is not releasing the channel for Attended
transfer immediately once I transferred and hangup from my side. The call
is still ongoing and disconnecting after the third party disconnected.

I see that its bug in the Asterisk, but not sure its fixed in version
11.2.1.

Any one facing this issue?

Regards.
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Re: [asterisk-users] Sangoma transcoding card bug - drops audio samples

[Gopalakrishnan N]

If you are getting like this dropped packets then nothing to worry.. thisis
just an cli message in my case I face this but there is no voice delay
in actual call.
On 22 Nov 2013 21:11, Eric Wieling ewiel...@nyigc.com wrote:

 Are you getting errors like this?



 [Nov 22 10:39:36] WARNING[6307][C-09a1]: codec_sangoma.c:969
 sangoma_frameout: [2724][ulawtog729] Got Seq 7400 but expecting 2154 (time
 since last read = 0ms), dropped 5246 packets





 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Grzegorz Garlewicz
 *Sent:* Friday, November 22, 2013 2:55 AM
 *To:* asterisk-users@lists.digium.com
 *Subject:* [asterisk-users] Sangoma transcoding card bug - drops audio
 samples




 There is a serious bug in Sangoma transcoding cards. The card has an
 internal, small jitter buffer and it drops samples

 from the audio stream when there is high jitter in the network. The
 bandwidth is cheap now so for me the only reason

 to use transcoding is where I have low-bandwidth-high-jitter links.
 Sangoma said they will not fix it and we had to go back

 to software transconding.


 Do you have any experience with using Digium cards in such scenario?

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Re: [asterisk-users] Multi-Voicemail Message?

[Gopalakrishnan N]

You can have something like this,
exten = _,1, Answer
exten = _, 2, voicemail ($EXTEN)
 On 25 Sep 2013 05:04, Tim Nelson tnel...@rockbochs.com wrote:

 Greetings-

 I have an odd scenario where I need to dial an extension (lets call it
 555), the system prompts for a list of voicemail boxes, then once complete,
 allows the caller to leave a voicemail that is sent to all voicemail boxes
 previously specified.

 How would you do this? Obviously calling Voicemail(), but how to get input
 for multiple extensions/voicemails, and delimit them properly for passing
 to Voicemail()?

 All ideas welcome. Thanks!

 --Tim

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[asterisk-users] Bad Magic Internal Error

[Gopalakrishnan N]

What does this mean of bad magic internal error, SIP to SIP calling is
fine, when I use SIP via GSM I have this, and asterisk restarts
automatically. Asterisk version which am using is 11.1.2.



Regards
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[asterisk-users] G729 CPU Utilization

[Gopalakrishnan N]

Hi,

How much CPU utilization will it take when I use G729 transcoding via
hardware based transcoder.

Regards
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Re: [asterisk-users] Kepress while on Queue

[Gopalakrishnan N]

oh great thanks...


On Wed, Aug 28, 2013 at 1:12 PM, Satish Barot satish4aster...@gmail.comwrote:

 Yes you can. Check the 'context' parameter in queues.conf. When caller
 presses a single digit extension while waiting in a queue, (s)he'll be
 taken out of queue to this context. Then you can send caller to different
 queue from this context.

 --Satish Barot
 Ahmedabad, India.
 +919978599700


 On Wed, Aug 28, 2013 at 12:59 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi,

 Will Keypress option will work when am in the queue and hearing MoH?

 Lets say a caller is waiting in queue and while he is hearing MoH, can he
 key in some DTMF and go to some other queue? is that possible?

 Regards

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Re: [asterisk-users] Kepress while on Queue

[Gopalakrishnan N]

also if am not wrong queue timeout will also applicable for this.. !


On Wed, Aug 28, 2013 at 11:37 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 oh great thanks...


 On Wed, Aug 28, 2013 at 1:12 PM, Satish Barot 
 satish4aster...@gmail.comwrote:

 Yes you can. Check the 'context' parameter in queues.conf. When caller
 presses a single digit extension while waiting in a queue, (s)he'll be
 taken out of queue to this context. Then you can send caller to
 different queue from this context.

 --Satish Barot
 Ahmedabad, India.
 +919978599700


 On Wed, Aug 28, 2013 at 12:59 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi,

 Will Keypress option will work when am in the queue and hearing MoH?

 Lets say a caller is waiting in queue and while he is hearing MoH, can
 he key in some DTMF and go to some other queue? is that possible?

 Regards

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[asterisk-users] Kepress while on Queue

[Gopalakrishnan N]

Hi,

Will Keypress option will work when am in the queue and hearing MoH?

Lets say a caller is waiting in queue and while he is hearing MoH, can he
key in some DTMF and go to some other queue? is that possible?

Regards
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Re: [asterisk-users] How to get the original SIP result code

[Gopalakrishnan N]

You can use AMI Commands and run sip set debug from that you have to
capture the response code.

http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Command

Regards,


On Thu, Aug 22, 2013 at 10:43 PM, Mordechay Kaganer mkaga...@gmail.comwrote:

 B.H.

 Hello, i'm using AMI Originate action (with async=true) to send outgoing
 calls to a SIP trunk (using asterisk-java library to connect to AMI).

 The problem is that in case of failed originate, OriginateResponse event
 is returning only the reason code which is sometimes not sufficient to
 determine the real cause of failure. Also, there's no way to link between
 the channel that was trying to dial and failed and the original Originate
 request, because OriginateResponse is issued only after the failed channel
 was hang up. Only successful OriginateResponse will contain the unique id
 of the established channel.

 Is there any way that my AMI application can get the original SIP response
 of the failed Originate action?

 I'm using Asterisk 1.8.22 and slightly tweaked asterisk-java (
 https://blogs.reucon.com/asterisk-java/) 1.0.0.


 --
 כתיבה וחתימה טובה לשנה טובה ומתוקה בגשמיות וברוחניות!
 משיח NOW!
 Moshiach is coming very soon, prepare yourself!
 יחי אדוננו מורינו ורבינו מלך המשיח לעולם ועד!

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Re: [asterisk-users] Ingress and Egress

[Gopalakrishnan N]

Basically I have some background noise like keyboard stoke or clicking
sound in random basis, I need to measure that, when I check my IPLC its
fine, and with my Telco service provider its fine...

So am trying to conclude with some solution... trying to identify the root
cause.

Any advice would be appreciated.

Thanks.


On Wed, Aug 21, 2013 at 4:46 PM, jg webaccou...@jgoettgens.de wrote:

 You do not need to calculate the jitter values yourself. For a quick check
 you can use the CLI cmd sip show channelstats. For external monitoring
 you could capture the RTCP AMI events.

 jg

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[asterisk-users] Ingress and Egress

[Gopalakrishnan N]

Hi,

Can Ingress and Egress can be used in Asterisk, so that Jitter can be
calculated...!

Regards
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[asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

[Gopalakrishnan N]

Hi,

Am making a simple SIP trunk between two Asterisk server,

Server 1
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.30.2.58
context=man02-trunk
port=5060
qualify=yes
disallow=all
;allow=g729
allow=g729
;allow=alaw
nat=force_rport,comedia
dtmfmode=rfc2833
relaxdtmf=yes
insecure=invite,port

extensions.conf
[man02-trunk]
exten = _1X.,1,Dial(SIP/usman02/${EXTEN})
exten = _1X.,n,Hangup


Server2
sip.conf
[usman02]
type=peer
username=usman02
secret=usman02
host=10.10.10.81
context=us02-trunk-inbound
port=5060
qualify=yes
disallow=all
allow=g729
;allow=ulaw
;allow=alaw
nat=force_rport,comedia
dtmfmode=rfc2833
relaxdtmf=yes
insecure=port,invite

extensions.conf
[us02-trunk-inbound]
exten = _X.,Dial(SIP/${EXTEN},60)


Now when I dial from server1, in the server 2 am getting the error as,
[Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth:
username mismatch, have 2001, digest has usman02

things are fine.. but I dont know where the mistake is...!

Can you some one advise me... !

Thanks.
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Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

[Gopalakrishnan N]

Even I tried the type as friend.. but no use...


On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 Hi,

 Am making a simple SIP trunk between two Asterisk server,

 Server 1
 sip.conf
 [usman02]
 type=peer
 username=usman02
 secret=usman02
 host=10.30.2.58
 context=man02-trunk
 port=5060
 qualify=yes
 disallow=all
 ;allow=g729
 allow=g729
 ;allow=alaw
 nat=force_rport,comedia
 dtmfmode=rfc2833
 relaxdtmf=yes
 insecure=invite,port

 extensions.conf
 [man02-trunk]
 exten = _1X.,1,Dial(SIP/usman02/${EXTEN})
 exten = _1X.,n,Hangup


 Server2
 sip.conf
 [usman02]
 type=peer
 username=usman02
 secret=usman02
 host=10.10.10.81
 context=us02-trunk-inbound
 port=5060
 qualify=yes
 disallow=all
 allow=g729
 ;allow=ulaw
 ;allow=alaw
 nat=force_rport,comedia
 dtmfmode=rfc2833
 relaxdtmf=yes
 insecure=port,invite

 extensions.conf
 [us02-trunk-inbound]
 exten = _X.,Dial(SIP/${EXTEN},60)


 Now when I dial from server1, in the server 2 am getting the error as,
 [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth:
 username mismatch, have 2001, digest has usman02

 things are fine.. but I dont know where the mistake is...!

 Can you some one advise me... !

 Thanks.

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Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

[Gopalakrishnan N]

Thanks for the comments.

Without changing anything, adding fromuser=usman02 in both side worked for
me..

Thanks.


On Mon, Aug 19, 2013 at 1:01 AM, Andrew Colin and...@vsave.co.za wrote:

  change server two to host = dynamic

 then add register = on server 1

 On 8/18/2013 6:29 PM, Gopalakrishnan N wrote:

 Even I tried the type as friend.. but no use...


 On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi,

  Am making a simple SIP trunk between two Asterisk server,

  Server 1
 sip.conf
  [usman02]
 type=peer
 username=usman02
 secret=usman02
 host=10.30.2.58
 context=man02-trunk
 port=5060
 qualify=yes
 disallow=all
 ;allow=g729
 allow=g729
 ;allow=alaw
 nat=force_rport,comedia
 dtmfmode=rfc2833
 relaxdtmf=yes
 insecure=invite,port

  extensions.conf
  [man02-trunk]
 exten = _1X.,1,Dial(SIP/usman02/${EXTEN})
 exten = _1X.,n,Hangup


  Server2
 sip.conf
  [usman02]
 type=peer
 username=usman02
 secret=usman02
 host=10.10.10.81
 context=us02-trunk-inbound
 port=5060
 qualify=yes
 disallow=all
 allow=g729
 ;allow=ulaw
 ;allow=alaw
 nat=force_rport,comedia
 dtmfmode=rfc2833
 relaxdtmf=yes
  insecure=port,invite

  extensions.conf
  [us02-trunk-inbound]
 exten = _X.,Dial(SIP/${EXTEN},60)


  Now when I dial from server1, in the server 2 am getting the error as,
 [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266 check_auth:
 username mismatch, have 2001, digest has usman02

  things are fine.. but I dont know where the mistake is...!

  Can you some one advise me... !

  Thanks.




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Re: [asterisk-users] Asterisk SIP Trunk between two Asterisk Servers

[Gopalakrishnan N]

ok thanks Asghar Mohammad


On Mon, Aug 19, 2013 at 1:05 AM, Asghar Mohammad asghar...@gmail.comwrote:

 just remove username.
 type peer authenticate by ip


 On Sun, Aug 18, 2013 at 7:01 PM, Andrew Colin and...@vsave.co.za wrote:

  change server two to host = dynamic

 then add register = on server 1

 On 8/18/2013 6:29 PM, Gopalakrishnan N wrote:

 Even I tried the type as friend.. but no use...


 On Mon, Aug 19, 2013 at 12:27 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi,

  Am making a simple SIP trunk between two Asterisk server,

  Server 1
 sip.conf
  [usman02]
 type=peer
 username=usman02
 secret=usman02
 host=10.30.2.58
 context=man02-trunk
 port=5060
 qualify=yes
 disallow=all
 ;allow=g729
 allow=g729
 ;allow=alaw
 nat=force_rport,comedia
 dtmfmode=rfc2833
 relaxdtmf=yes
 insecure=invite,port

  extensions.conf
  [man02-trunk]
 exten = _1X.,1,Dial(SIP/usman02/${EXTEN})
 exten = _1X.,n,Hangup


  Server2
 sip.conf
  [usman02]
 type=peer
 username=usman02
 secret=usman02
 host=10.10.10.81
 context=us02-trunk-inbound
 port=5060
 qualify=yes
 disallow=all
 allow=g729
 ;allow=ulaw
 ;allow=alaw
 nat=force_rport,comedia
 dtmfmode=rfc2833
 relaxdtmf=yes
  insecure=port,invite

  extensions.conf
  [us02-trunk-inbound]
 exten = _X.,Dial(SIP/${EXTEN},60)


  Now when I dial from server1, in the server 2 am getting the error as,
 [Aug 18 09:22:49] WARNING[2779][C-08db]: chan_sip.c:16266
 check_auth: username mismatch, have 2001, digest has usman02

  things are fine.. but I dont know where the mistake is...!

  Can you some one advise me... !

  Thanks.




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[asterisk-users] Random dead calls

[Gopalakrishnan N]

Hi,

Am getting dead or silence calls at sometimes for my agents, when I checked
my CDR the caller-id shows my vendor's name and some shows as real customer
name.

When I call back again the real customer's number its reaching, the
answering machine owned by customer.

I have a confusion, or how to find out are these numbers are from any auto
dialer or from real customers.

Thanks.
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Re: [asterisk-users] auto answer

[Gopalakrishnan N]

yes its not asterisk configuration, its phone feature and phone
configuration.


On Wed, Jul 17, 2013 at 3:27 PM, bilal ghayyad bilmar...@yahoo.com wrote:

 So it is not at asterisk configuration?

 Regards
 Bilal

   --
  *From:* A J Stiles asterisk_l...@earthshod.co.uk

 *To:* bilal ghayyad bilmar...@yahoo.com; Asterisk Users Mailing List -
 Non-Commercial Discussion asterisk-users@lists.digium.com
 *Sent:* Wednesday, July 17, 2013 12:57 PM

 *Subject:* Re: [asterisk-users] auto answer

 On Wednesday 17 July 2013, bilal ghayyad wrote:
  But this not in the sip.conf, this in the extensions.conf, right?
 
  Regards
  Bilal

 No.  This would be set up in the phone's own configuration file, which in
 turn
 depends on the make and model of phone  (and its location depends on your
 site
 setup).

 --
 AJS

 Answers come *after* questions.



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Re: [asterisk-users] auto answer

[Gopalakrishnan N]

If am not wrong even without doing any setting in asterisk side, if the
phone has Auto Answer it works.. !

Correct me if am wrong.


On Wed, Jul 17, 2013 at 9:14 PM, Steve Edwards asterisk@sedwards.comwrote:

 Please don't top post.


 On Wed, 17 Jul 2013, bilal ghayyad wrote:

  So it is not at asterisk configuration?


 1) The phone has to be configured to allow it.

 2) Asterisk has to set the appropriate SIP header for your specific model
 phone prior to 'dialing' the phone for each call. I.e. the added SIP header
 for a Cisco is different than for a Polycom.

 --
 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] FLAC script to convert from wav to FLAC and also with other 3 to 4 formats

[Gopalakrishnan N]

Hi,

Below link is the script which i found while surfing, this script basically
converts your voice file to flac format, where the file is reduced to 50%.

http://legroom.net/files/software/convtoflac.sh

The quality is really good, I tested. this...

In large production environment this script can be used, only challenging
part, please make sure the CPU usage is within the limit while conversion.

Can be used like this,
exten =
_4X.,n,MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,b,/root/
flac.sh ${MIXMONITOR_FILENAME}.wav)

Regards,
Gopal.
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Re: [asterisk-users] External Recording Server for Asterisk Voicemail

[Gopalakrishnan N]

If you want to store in external, why can't you have a NAS device and mount
to Asterisk server, let the mounted be a part in asterisk.conf, so that
voicemail will get recorded in external server...

Will it makes sense... !

Thanks.


On Mon, Jul 15, 2013 at 4:19 PM, Amit Salunkhe amitsalunkh...@gmail.comwrote:

 Hello All,

 I'm planning to use Asterisk only for voicemail Application and Recording
 will be done at different server.

 When user changing his personal greeting or leaving voicemail Call need to
 throw to external Voicemnail recording server over SIP til the time
 recording complete.

 While throwing Cal from Asterisk to application box i have to use SIP
 request which having some string in R-URI. Please let me know is this
 possible with configuration example.



 Regards
 Amit

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Re: [asterisk-users] Executing Script after MixMonitor is called

[Gopalakrishnan N]

I tried with the ^ symbol but still there is no success.

And regards to the path, actually my file is in path /root, is that to be
in /usr/sbin or somewhere?

Basically am able to see the application executed in the CLI, like the
below,

 Executing [4090@test:1] Set(SIP/4092-003b,
START_TIME=2013-07-05_14:43:11) in new stack
-- Executing [4090@test:2] Set(SIP/4092-003b,
MIXMONITOR_FILENAME=4090-2013-07-05_14:43:11-OUT) in new stack
-- Executing [4090@test:3] MixMonitor(SIP/4092-003b,
IND_PRI/4092/OUT/2013-07/05/4090-2013-07-05_14:43:11-OUT.wav,b,/root/flac.sh
4090-2013-07-05_14:43:11-OUT) in new stack
-- Executing [4090@test:4] Set(SIP/4092-003b,
CDR(userfield)=/var/spool/asterisk/monitor/IND_PRI/4092/OUT/2013-07/05/4090-2013-07-05_14:43:11-OUT.wav)
in new stack
-- Executing [4090@test:5] Dial(SIP/4092-003b, SIP/4090,30) in
new stack
  == Using SIP RTP CoS mark 5
  == Begin MixMonitor Recording SIP/4092-003b
-- Called SIP/4090
-- SIP/4090-003c is ringing
-- SIP/4090-003c answered SIP/4092-003b
-- fixed jitterbuffer created on channel SIP/4090-003c
-- fixed jitterbuffer created on channel SIP/4092-003b
-- Executing [h@test:1] MYSQL(SIP/4092-003b, Connect connid
localhost root Iopex1063 Logs) in new stack
-- Executing [h@test:2] MYSQL(SIP/4092-003b, Query resultid 1
insert into
call_log(accountcode,start,end,src,dst,uniqueid,userfield,hangupcause)
values(4092,2013-07-05
14:43:11,now(),30993091,4090,1373049791.59,/var/spool/asterisk/monitor/IND_PRI/4092/OUT/2013-07/05/4090-2013-07-05_14:43:11-OUT.wav,16))
in new stack
-- Executing [h@test:3] MYSQL(SIP/4092-003b, Disconnect 1) in
new stack
-- fixed jitterbuffer destroyed on channel SIP/4090-003c
  == Spawn extension (test, 4090, 5) exited non-zero on 'SIP/4092-003b'
-- fixed jitterbuffer destroyed on channel SIP/4092-003b
  == MixMonitor close filestream
  == *Executing [/root/flac.sh 4090-2013-07-05_14:43:11-OUT]*
  == End MixMonitor Recording SIP/4092-003b

But the file is not converted, I suspect it could be a path issue.



Regards


On Fri, Jul 5, 2013 at 10:59 AM, Satish Barot satish4aster...@gmail.comwrote:

 On Fri, Jul 5, 2013 at 1:45 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 exten = _4X.,1,Set(START_TIME=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
 exten = _4X.,n,Set(MIXMONITOR_FILENAME=${EXTEN}-${START_TIME}-OUT)
 ;exten =
 _4X.,n,MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,m,/root/flac.sh
 ${MIXMONITOR_FILENAME}.wav)
 exten =
 _4X.,n,MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,b,/root/flac.sh
 ${MIXMONITOR_FILENAME}.wav)
 exten =
 _4X.,n,Set(CDR(userfield)=IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME})
 exten = _4X.,n,Dial(SIP/${EXTEN},30)
 exten = _4X.,n,Hangup

 Regards
 On 4 Jul 2013 11:18, Satish Barot satish4aster...@gmail.com wrote:

 On Thu, Jul 4, 2013 at 1:30 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 I tried with hangup cause but my script is not executed... also I tried
 the same script with mix monitor itself no sucess.

 The script what I have is, am converting wav file to flac format..
 On 11 Jun 2013 11:17, Satish Barot satish4aster...@gmail.com wrote:

 And yes if you want to use System application in your dialplan then
 have System in your h extension

 System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav 
 /PathToMp3FileToBE Stored/filename.mp3)



 On Tue, Jun 11, 2013 at 10:38 AM, Satish Barot 
 satish4aster...@gmail.com wrote:

 Hi Gopamkrishnan,

 Check the 'command' argument for Mixmonitor. Mixmonitor itself has a
 facility to execute a command when recording is over.

 *In my case, 'wav2mp3' is a script which gets executed and converts 
 recorded wav audio file to mp3. I pass ${FILENAME} as an argument to my 
 script.*

 *You should have something like 
 *MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav)
  in your dialplan.


 Hope this helps.

 --Satish Barot


 Ahmedabad, India


 On Tue, Jun 11, 2013 at 9:31 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi Satish,

 I tried with sox, without any parameter, just sox filename.wav to
 filename.mp3, in linux shell prompt... the file is been converted...

 Now If i want to run that command using dialplan,

 MixMonitor(filename.wav,m)
 Monitor_Exec(sox filename.wav filename.mp3)

 Or to use System command?

 Regards..


 On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot 
 satish4aster...@gmail.com wrote:

 This is how I use a wav to mp3 script on Mixmonitor in my dialplan
 (Asterisk 1.8.7.0).
 ...
 same = n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
 ^{FILENAME})
 ...
 and my script is...

 #!/bin/bash

 WAV=/var/spool/asterisk/monitor/$1

Re: [asterisk-users] Executing Script after MixMonitor is called

[Gopalakrishnan N]

exten = _4X.,1,Set(START_TIME=${STRFTIME(${EPOCH},,%Y-%m-%d_%H:%M:%S)})
exten = _4X.,n,Set(MIXMONITOR_FILENAME=${EXTEN}-${START_TIME}-OUT)
;exten =
_4X.,n,MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,m,/root/flac.sh
${MIXMONITOR_FILENAME}.wav)
exten =
_4X.,n,MixMonitor(IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME}.wav,b,/root/flac.sh
${MIXMONITOR_FILENAME}.wav)
exten =
_4X.,n,Set(CDR(userfield)=IND_PRI/${CDR(accountcode)}/OUT/${STRFTIME(${EPOCH},,%Y-%m)}/${STRFTIME(${EPOCH},,%d)}/${MIXMONITOR_FILENAME})
exten = _4X.,n,Dial(SIP/${EXTEN},30)
exten = _4X.,n,Hangup

Regards
On 4 Jul 2013 11:18, Satish Barot satish4aster...@gmail.com wrote:

 On Thu, Jul 4, 2013 at 1:30 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 I tried with hangup cause but my script is not executed... also I tried
 the same script with mix monitor itself no sucess.

 The script what I have is, am converting wav file to flac format..
 On 11 Jun 2013 11:17, Satish Barot satish4aster...@gmail.com wrote:

 And yes if you want to use System application in your dialplan then have
 System in your h extension

 System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav 
 /PathToMp3FileToBE Stored/filename.mp3)

 On Tue, Jun 11, 2013 at 10:38 AM, Satish Barot 
 satish4aster...@gmail.com wrote:

 Hi Gopamkrishnan,

 Check the 'command' argument for Mixmonitor. Mixmonitor itself has a
 facility to execute a command when recording is over.

 *In my case, 'wav2mp3' is a script which gets executed and converts 
 recorded wav audio file to mp3. I pass ${FILENAME} as an argument to my 
 script.*

 *You should have something like 
 *MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav) 
 in your dialplan.

 Hope this helps.

 --Satish Barot


 Ahmedabad, India


 On Tue, Jun 11, 2013 at 9:31 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi Satish,

 I tried with sox, without any parameter, just sox filename.wav to
 filename.mp3, in linux shell prompt... the file is been converted...

 Now If i want to run that command using dialplan,

 MixMonitor(filename.wav,m)
 Monitor_Exec(sox filename.wav filename.mp3)

 Or to use System command?

 Regards..


 On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot 
 satish4aster...@gmail.com wrote:

 This is how I use a wav to mp3 script on Mixmonitor in my dialplan
 (Asterisk 1.8.7.0).
 ...
 same = n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
 ^{FILENAME})
 ...
 and my script is...

 #!/bin/bash

 WAV=/var/spool/asterisk/monitor/$1
 MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
 MP3DEST=/var/spool/asterisk/mp3/$MP3
 /usr/bin/lame ${WAV} ${MP3DEST} --silent -b 16 -s 9.6 -m m
 --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1

 --SATISH BAROT
 Ahmedabad,India.


 On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib fkha...@iconnecths.com
  wrote:

 Hello Guys,
 I am trying to convert files that are .wac to mp3 after mixmonitor
 command is called but it doesnt execute the command, I tried the 
 command in
 terminal it worked, any help please ... below is my dial plan
 exten=6500,n,Set(MIXMONITOR_EXEC= nice -n 19 /usr/local/bin/lame
 -b 8 -t -F -m m --bitwidth 8 --quiet
 /var/spool/asterisk/monitor/${CALLFILENAME}.wav
 /var/spool/asterisk/monitor/${CALLFILENAME}.mp3  rm -f
 /var/spool/asterisk/monitor/${CALLFILENAME}.wav)
 exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)



 Show your latest dialplan and script.

 --Satish Barot
 Ahmedabad, India

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[asterisk-users] Asterisk crash

[Gopalakrishnan N]

Suddenly my asterisk restarted automatically and came up in seven seconds,

While checking core dump I see some message related to snmp.

No symbol table info available.
#5 0x7fc7e6249faa in agent_thread (arg=value optimized out) at
snmp/agent.c:206
__PRETTY_FUNCTION__ = agent_thread
#6 0x0056dd0b in dummy_start (data=value optimized out) at
utils.c:1028
__cancel_buf = {__cancel_jmp_buf = {{__cancel_jmp_buf = {89647040,
7553562169405615537, 140735377460432, 140496194722240, 4, 7,
-7540143656030687823,
7553561768520461745}, __mask_was_saved = 0}}, __pad = {0x7fc7d1c74e90, 0x0,
0x0, 0x0}}
__cancel_arg = 0x7fc7d1c75700
not_first_call = value optimized out
ret = value optimized out
a = {start_routine = 0x7fc7e6249eb0 agent_thread, data = 0x0, name =
0x7fc7d1c74d70 \300\347W\005}
#7 0x7fc830e54851 in start_thread () from /lib64/libpthread.so.0
No symbol table info available.
#8 0x7fc8323c611d in clone () from /lib64/libc.so.6
No symbol table info available.
(gdb) quit

Will this be related to snmp?

Regards
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Re: [asterisk-users] Asterisk crash

[Gopalakrishnan N]

Ok thanks posting now
On 5 Jul 2013 03:09, Matthew Jordan mjor...@digium.com wrote:


 On Thu, Jul 4, 2013 at 3:30 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Suddenly my asterisk restarted automatically and came up in seven seconds,

 While checking core dump I see some message related to snmp.

 No symbol table info available.
 #5 0x7fc7e6249faa in agent_thread (arg=value optimized out) at
 snmp/agent.c:206
 __PRETTY_FUNCTION__ = agent_thread
 #6 0x0056dd0b in dummy_start (data=value optimized out) at
 utils.c:1028
 __cancel_buf = {__cancel_jmp_buf = {{__cancel_jmp_buf = {89647040,
 7553562169405615537, 140735377460432, 140496194722240, 4, 7,
 -7540143656030687823,
 7553561768520461745}, __mask_was_saved = 0}}, __pad = {0x7fc7d1c74e90,
 0x0, 0x0, 0x0}}
 __cancel_arg = 0x7fc7d1c75700
 not_first_call = value optimized out
 ret = value optimized out
 a = {start_routine = 0x7fc7e6249eb0 agent_thread, data = 0x0, name =
 0x7fc7d1c74d70 \300\347W\005}
 #7 0x7fc830e54851 in start_thread () from /lib64/libpthread.so.0
 No symbol table info available.
 #8 0x7fc8323c611d in clone () from /lib64/libc.so.6
 No symbol table info available.
 (gdb) quit

 Will this be related to snmp?


 Possibly, but not necessarily. Without seeing the whole backtrace it's
 hard to say for certain.

 The Asterisk wiki has instructions on how to properly get a backtrace from
 a core dump created by Asterisk:

 https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace

 Please do file an issue in the issue tracker - https://issues.asterisk.org- 
 crashes are always bugs.

 Thanks!

 Matt

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

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Re: [asterisk-users] Executing Script after MixMonitor is called

[Gopalakrishnan N]

I tried with hangup cause but my script is not executed... also I tried the
same script with mix monitor itself no sucess.

The script what I have is, am converting wav file to flac format..
On 11 Jun 2013 11:17, Satish Barot satish4aster...@gmail.com wrote:

 And yes if you want to use System application in your dialplan then have
 System in your h extension

 System(/PathToSox/sox -r 8000 -c 1 /PathToWavFile/filename.wav 
 /PathToMp3FileToBE Stored/filename.mp3)






 On Tue, Jun 11, 2013 at 10:38 AM, Satish Barot 
 satish4aster...@gmail.comwrote:

 Hi Gopamkrishnan,

 Check the 'command' argument for Mixmonitor. Mixmonitor itself has a
 facility to execute a command when recording is over.

 *In my case, 'wav2mp3' is a script which gets executed and converts recorded 
 wav audio file to mp3. I pass ${FILENAME} as an argument to my script.
 *

 *You should have something like 
 *MixMonitor(filename.wav,m,/PathToYourScript/YourScriptName^filename.wav) in 
 your dialplan.

 Hope this helps.

 --Satish Barot


 Ahmedabad, India





 On Tue, Jun 11, 2013 at 9:31 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi Satish,

 I tried with sox, without any parameter, just sox filename.wav to
 filename.mp3, in linux shell prompt... the file is been converted...

 Now If i want to run that command using dialplan,

 MixMonitor(filename.wav,m)
 Monitor_Exec(sox filename.wav filename.mp3)

 Or to use System command?

 Regards..


 On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot 
 satish4aster...@gmail.com wrote:

 This is how I use a wav to mp3 script on Mixmonitor in my dialplan
 (Asterisk 1.8.7.0).
 ...
 same = n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
 ^{FILENAME})
 ...
 and my script is...

 #!/bin/bash

 WAV=/var/spool/asterisk/monitor/$1
 MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
 MP3DEST=/var/spool/asterisk/mp3/$MP3
 /usr/bin/lame ${WAV} ${MP3DEST} --silent -b 16 -s 9.6 -m m
 --bitwidth 8 --lowpass 9.6 --resample 8 --lowpass-width 1

 --SATISH BAROT
 Ahmedabad,India.


 On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib 
 fkha...@iconnecths.comwrote:

 Hello Guys,
 I am trying to convert files that are .wac to mp3 after mixmonitor
 command is called but it doesnt execute the command, I tried the command 
 in
 terminal it worked, any help please ... below is my dial plan
 exten=6500,n,Set(MIXMONITOR_EXEC= nice -n 19 /usr/local/bin/lame -b
 8 -t -F -m m --bitwidth 8 --quiet
 /var/spool/asterisk/monitor/${CALLFILENAME}.wav
 /var/spool/asterisk/monitor/${CALLFILENAME}.mp3  rm -f
 /var/spool/asterisk/monitor/${CALLFILENAME}.wav)
 exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)

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[asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

Am using Asterisk 11.2 in one location and 11.1 in another location.

when I trunk between two servers, the status is unreachable.

But with different server with 11.2 and 11.2 it works fine.

I tried both IAX and SIP.

the trunk in sip.conf what i have is,
[serverb]
type=friend
username=serverb
secret=serverb
host=10.10.10.5
port=5060
context=default
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=3000
nat=force_rport,comedia
disallow=all
allow=g729
allow=ulaw
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=10.10.10.5/255.255.255.0

Is there any issue with 11.1?
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Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

Also tried one more scenario, particularly from one IP to other IP not
registering.

For example like 10.10.10.5 to 10.20.10.5

If it is 10.10.10.5 to 10.30.2.5 - working
If it is 10.30.2.5 to 10.20.10.4 works fine.

really strange... I suspect some issue on the network side...

Problem is there is no packet loss.. with mtr it is fine, tracepath is
fine, ping is fine... :(


On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 Am using Asterisk 11.2 in one location and 11.1 in another location.

 when I trunk between two servers, the status is unreachable.

 But with different server with 11.2 and 11.2 it works fine.

 I tried both IAX and SIP.

 the trunk in sip.conf what i have is,
 [serverb]
 type=friend
 username=serverb
 secret=serverb
 host=10.10.10.5
 port=5060
 context=default
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=3000
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.10.10.5/255.255.255.0

 Is there any issue with 11.1?

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Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

[servera]
type=friend
username=servera
secret=servera
host=10.30.2.5
port=5060
context=Manila
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
disallow=all
allow=g729
allow=ulaw
allow=alaw
deny=0.0.0.0/0.0.0.0
permit=10.30.2.5/255.255.255.0

If i use host=dynamic, it wont communicate each other and will result to
unmonitored


and the IP segment is two different segment. where am able to ping each
other.



On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad asghar...@gmail.com wrote:

 hi,
 paste server a trunk also, if you want register why you are not using
 host=dynamic?
 both servers are on 10.10.10.0 ? if no then check your deny permit seting.


 On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Also tried one more scenario, particularly from one IP to other IP not
 registering.

 For example like 10.10.10.5 to 10.20.10.5

 If it is 10.10.10.5 to 10.30.2.5 - working
 If it is 10.30.2.5 to 10.20.10.4 works fine.

 really strange... I suspect some issue on the network side...

 Problem is there is no packet loss.. with mtr it is fine, tracepath is
 fine, ping is fine... :(


 On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Am using Asterisk 11.2 in one location and 11.1 in another location.

 when I trunk between two servers, the status is unreachable.

 But with different server with 11.2 and 11.2 it works fine.

 I tried both IAX and SIP.

 the trunk in sip.conf what i have is,
 [serverb]
 type=friend
 username=serverb
 secret=serverb
 host=10.10.10.5
 port=5060
 context=default
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=3000
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.10.10.5/255.255.255.0

 Is there any issue with 11.1?



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Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

can't we use without register command both way as peer to peer?


On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad asghar...@gmail.com wrote:

 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b and
 10.10.10.0 on a.
 2. use host=dynamic type=friend on  side A and host=ip type=peer on side B.
 3. general section in sip.conf of side B register with server A.

 please see comments in sip.conf
 ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
 registering
 ; as any IP address used for staticly
 defined
 ; hosts.  This helps avoid the
 configuration
 ; error of allowing your users to register
 at
 ; the same address as a SIP provider.



 On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 [servera]
 type=friend
 username=servera
 secret=servera
 host=10.30.2.5
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.30.2.5/255.255.255.0

 If i use host=dynamic, it wont communicate each other and will result to
 unmonitored


 and the IP segment is two different segment. where am able to ping each
 other.



 On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad asghar...@gmail.comwrote:

 hi,
 paste server a trunk also, if you want register why you are not using
 host=dynamic?
 both servers are on 10.10.10.0 ? if no then check your deny permit
 seting.


 On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Also tried one more scenario, particularly from one IP to other IP not
 registering.

 For example like 10.10.10.5 to 10.20.10.5

 If it is 10.10.10.5 to 10.30.2.5 - working
 If it is 10.30.2.5 to 10.20.10.4 works fine.

 really strange... I suspect some issue on the network side...

 Problem is there is no packet loss.. with mtr it is fine, tracepath is
 fine, ping is fine... :(


 On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Am using Asterisk 11.2 in one location and 11.1 in another location.

 when I trunk between two servers, the status is unreachable.

 But with different server with 11.2 and 11.2 it works fine.

 I tried both IAX and SIP.

 the trunk in sip.conf what i have is,
 [serverb]
 type=friend
 username=serverb
 secret=serverb
 host=10.10.10.5
 port=5060
 context=default
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=3000
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.10.10.5/255.255.255.0

 Is there any issue with 11.1?



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Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

I tried creating two trunks with following,
*1st Location*
[10.30.2.5]
type=friend
username=indman01
secret=indman01
host=dynamic
port=5060
context=Manila
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
disallow=all
allow=g729
allow=ulaw

*2nd Location*
[10.20.111.48]
type=friend
username=manind01
secret=manind01
host=dynamic
port=5060
context=india
insecure=port,invite
dtmfmode=rfc2833
relaxdtmf=yes
directmedia=no
qualify=yes
nat=force_rport,comedia
disallow=all
allow=g729
allow=ulaw
allow=alaw

My dialplan is like this
exten = _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN})
exten = _2XXX,n,Hangup

And the output I get is
 Executing [2001@Test:1] Dial(SIP/3081-27d2, SIP/10.30.2.5/2001) in
new stack
[Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)
-- Executing [2001@Test:2] Hangup(SIP/3081-27d2, ) in new stack
  == Spawn extension (Test, 2001, 2) exited non-zero on 'SIP/3081-27d2'

Actually the trunk which i mentioned in my first email, it was working...
and from today it is not

Still breaking... what could be the reason... !



On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad asghar...@gmail.com wrote:

 yes you can. just create trunks on both side with static ip and in dial
 use trunk name.
 exten = _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =
 _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
 make a call from a to b and one from b to and post cli log here or upload
 anyware else.


 On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 can't we use without register command both way as peer to peer?


 On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad asghar...@gmail.comwrote:

 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b
 and 10.10.10.0 on a.
 2. use host=dynamic type=friend on  side A and host=ip type=peer on side
 B.
 3. general section in sip.conf of side B register with server A.

 please see comments in sip.conf
 ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
 registering
 ; as any IP address used for staticly
 defined
 ; hosts.  This helps avoid the
 configuration
 ; error of allowing your users to
 register at
 ; the same address as a SIP provider.



 On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 [servera]
 type=friend
 username=servera
 secret=servera
 host=10.30.2.5
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.30.2.5/255.255.255.0

 If i use host=dynamic, it wont communicate each other and will result
 to unmonitored


 and the IP segment is two different segment. where am able to ping each
 other.



 On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad asghar...@gmail.comwrote:

 hi,
 paste server a trunk also, if you want register why you are not using
 host=dynamic?
 both servers are on 10.10.10.0 ? if no then check your deny permit
 seting.


 On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Also tried one more scenario, particularly from one IP to other IP
 not registering.

 For example like 10.10.10.5 to 10.20.10.5

 If it is 10.10.10.5 to 10.30.2.5 - working
 If it is 10.30.2.5 to 10.20.10.4 works fine.

 really strange... I suspect some issue on the network side...

 Problem is there is no packet loss.. with mtr it is fine, tracepath
 is fine, ping is fine... :(


 On Wed, Jul 3, 2013 at 1:05 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Am using Asterisk 11.2 in one location and 11.1 in another location.

 when I trunk between two servers, the status is unreachable.

 But with different server with 11.2 and 11.2 it works fine.

 I tried both IAX and SIP.

 the trunk in sip.conf what i have is,
 [serverb]
 type=friend
 username=serverb
 secret=serverb
 host=10.10.10.5
 port=5060
 context=default
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=3000
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.10.10.5/255.255.255.0

 Is there any issue with 11.1?



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Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

still the peer shows unreachable let me restart and give a try...


On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad asghar...@gmail.com wrote:

 *1st Location*
 [manila]
 type=peer
 username=indman01
 secret=indman01
 host=10.30.2.5 -- ip of 2nd location
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw

 1st location dialplan
 exten = _2XXX,1,Dial(SIP/manila/${EXTEN} http://10.30.2.5/$%7BEXTEN%7D)
 exten = _2XXX,n,Hangup

 *2nd Location*
 [india]
 type=friend
 username=manind01
 secret=manind01
 host=dynamic
 port=5060
 context=10.20.111.48 - ip of 1st location
  insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw

 2st location dialplan
 exten = _2XXX,1,Dial(SIP/india/${EXTEN} http://10.30.2.5/$%7BEXTEN%7D)
 exten = _2XXX,n,Hangup

 then you should handle the call when it arrive in any server
 let me know if it work.


 On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 I tried creating two trunks with following,
 *1st Location*
 [10.30.2.5]
 type=friend
 username=indman01
 secret=indman01
 host=dynamic
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw

 *2nd Location*
 [10.20.111.48]
 type=friend
 username=manind01
 secret=manind01
 host=dynamic
 port=5060
 context=india
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw

 My dialplan is like this
 exten = _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN}http://10.30.2.5/$%7BEXTEN%7D
 )
 exten = _2XXX,n,Hangup

 And the output I get is
  Executing [2001@Test:1] Dial(SIP/3081-27d2, SIP/10.30.2.5/2001)
 in new stack
 [Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
 Subscriber absent)
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Executing [2001@Test:2] Hangup(SIP/3081-27d2, ) in new
 stack
   == Spawn extension (Test, 2001, 2) exited non-zero on
 'SIP/3081-27d2'

 Actually the trunk which i mentioned in my first email, it was working...
 and from today it is not

 Still breaking... what could be the reason... !



 On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad asghar...@gmail.comwrote:

 yes you can. just create trunks on both side with static ip and in dial
 use trunk name.
 exten = _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =
 _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
 make a call from a to b and one from b to and post cli log here or
 upload anyware else.


 On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 can't we use without register command both way as peer to peer?


 On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad asghar...@gmail.comwrote:

 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on b
 and 10.10.10.0 on a.
 2. use host=dynamic type=friend on  side A and host=ip type=peer on
 side B.
 3. general section in sip.conf of side B register with server A.

 please see comments in sip.conf
 ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
 registering
 ; as any IP address used for staticly
 defined
 ; hosts.  This helps avoid the
 configuration
 ; error of allowing your users to
 register at
 ; the same address as a SIP provider.



 On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 [servera]
 type=friend
 username=servera
 secret=servera
 host=10.30.2.5
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.30.2.5/255.255.255.0

 If i use host=dynamic, it wont communicate each other and will result
 to unmonitored


 and the IP segment is two different segment. where am able to ping
 each other.



 On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad 
 asghar...@gmail.comwrote:

 hi,
 paste server a trunk also, if you want register why you are not
 using host=dynamic?
 both servers are on 10.10.10.0 ? if no then check your deny permit
 seting.


 On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Also tried one more scenario, particularly from one IP to other IP
 not registering.

 For example like 10.10.10.5 to 10.20.10.5

 If it is 10.10.10.5 to 10.30.2.5 - working
 If it is 10.30.2.5 to 10.20.10.4 works fine.

 really strange... I suspect some issue on the network side...

 Problem is there is no packet loss.. with mtr it is fine, tracepath
 is fine, ping is fine

Re: [asterisk-users] Asterisk trunking between two location

[Gopalakrishnan N]

By having different server, i made it work. I suspect some network issue...


On Wed, Jul 3, 2013 at 3:27 AM, Asghar Mohammad asghar...@gmail.com wrote:

 make a call and post cli log


 On Tue, Jul 2, 2013 at 11:54 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 still the peer shows unreachable let me restart and give a try...


 On Wed, Jul 3, 2013 at 2:49 AM, Asghar Mohammad asghar...@gmail.comwrote:

 *1st Location*
 [manila]
 type=peer
 username=indman01
 secret=indman01
 host=10.30.2.5 -- ip of 2nd location
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw

 1st location dialplan
 exten = _2XXX,1,Dial(SIP/manila/${EXTEN}http://10.30.2.5/$%7BEXTEN%7D
 )
 exten = _2XXX,n,Hangup

 *2nd Location*
 [india]
 type=friend
 username=manind01
 secret=manind01
 host=dynamic
 port=5060
 context=10.20.111.48 - ip of 1st location
  insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw

 2st location dialplan
 exten = _2XXX,1,Dial(SIP/india/${EXTEN} http://10.30.2.5/$%7BEXTEN%7D
 )
 exten = _2XXX,n,Hangup

 then you should handle the call when it arrive in any server
 let me know if it work.


 On Tue, Jul 2, 2013 at 10:56 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 I tried creating two trunks with following,
 *1st Location*
 [10.30.2.5]
 type=friend
 username=indman01
 secret=indman01
 host=dynamic
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw

 *2nd Location*
 [10.20.111.48]
 type=friend
 username=manind01
 secret=manind01
 host=dynamic
 port=5060
 context=india
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 nat=force_rport,comedia
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw

 My dialplan is like this
 exten = _2XXX,1,Dial(SIP/10.30.2.5/${EXTEN}http://10.30.2.5/$%7BEXTEN%7D
 )
 exten = _2XXX,n,Hangup

 And the output I get is
  Executing [2001@Test:1] Dial(SIP/3081-27d2, SIP/10.30.2.5/2001)
 in new stack
 [Jul  2 16:49:57] WARNING[15766][C-2b94]: app_dial.c:2437
 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
 Subscriber absent)
   == Everyone is busy/congested at this time (1:0/0/1)
 -- Executing [2001@Test:2] Hangup(SIP/3081-27d2, ) in new
 stack
   == Spawn extension (Test, 2001, 2) exited non-zero on
 'SIP/3081-27d2'

 Actually the trunk which i mentioned in my first email, it was
 working... and from today it is not

 Still breaking... what could be the reason... !



 On Wed, Jul 3, 2013 at 2:05 AM, Asghar Mohammad asghar...@gmail.comwrote:

 yes you can. just create trunks on both side with static ip and in
 dial use trunk name.
 exten = _X.,1,Dial(SIP/trunka/${EXTEN}) on side b and exten =
 _X.,1,Dial(SIP/trunkb/${EXTEN}) on side a.
 make a call from a to b and one from b to and post cli log here or
 upload anyware else.


 On Tue, Jul 2, 2013 at 10:25 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 can't we use without register command both way as peer to peer?


 On Wed, Jul 3, 2013 at 1:45 AM, Asghar Mohammad 
 asghar...@gmail.comwrote:

 1. you permiting 10.10.10.0 on b but you should permit 10.30.2.0 on
 b and 10.10.10.0 on a.
 2. use host=dynamic type=friend on  side A and host=ip type=peer on
 side B.
 3. general section in sip.conf of side B register with server A.

 please see comments in sip.conf
 ;dynamic_exclude_static = yes   ; Disallow all dynamic hosts from
 registering
 ; as any IP address used for
 staticly defined
 ; hosts.  This helps avoid the
 configuration
 ; error of allowing your users to
 register at
 ; the same address as a SIP provider.



 On Tue, Jul 2, 2013 at 10:04 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 [servera]
 type=friend
 username=servera
 secret=servera
 host=10.30.2.5
 port=5060
 context=Manila
 insecure=port,invite
 dtmfmode=rfc2833
 relaxdtmf=yes
 directmedia=no
 qualify=yes
 disallow=all
 allow=g729
 allow=ulaw
 allow=alaw
 deny=0.0.0.0/0.0.0.0
 permit=10.30.2.5/255.255.255.0

 If i use host=dynamic, it wont communicate each other and will
 result to unmonitored


 and the IP segment is two different segment. where am able to ping
 each other.



 On Wed, Jul 3, 2013 at 1:29 AM, Asghar Mohammad 
 asghar...@gmail.com wrote:

 hi,
 paste server a trunk also, if you want register why you are not
 using host=dynamic?
 both servers are on 10.10.10.0 ? if no then check your deny permit
 seting.


 On Tue, Jul 2, 2013 at 9:53 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Also tried one more scenario, particularly from one IP to other
 IP not registering.

 For example like

[asterisk-users] IAX2 netsock error with name resolution

[Gopalakrishnan N]

Am getting netsock error like this when using IAX2,

Connected to Asterisk 11.2.1 currently running on indiaprimaryast01 (pid =
4270)
  == Using SIP RTP CoS mark 5
-- Executing [2001@Test:1] Dial(SIP/4090-0005,
SIP/2001@IAX2/IND-MAN,30)
in new stack
[Jun 23 06:31:36] NOTICE[4383][C-0005]: chan_sip.c:29491
sip_request_call: Conflicting extension values given. Using '2001' and not
'IND-MAN'
  == Using SIP RTP CoS mark 5
[Jun 23 06:31:36] ERROR[4383][C-0005]: netsock2.c:269
ast_sockaddr_resolve: getaddrinfo(IAX2, (null), ...): Temporary failure
in name resolution
[Jun 23 06:31:36] WARNING[4383][C-0005]: chan_sip.c:6191 create_addr:
No such host: IAX2
[Jun 23 06:31:36] WARNING[4383][C-0005]: app_dial.c:2437
dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
Subscriber absent)
  == Everyone is busy/congested at this time (1:0/0/1)

My hostname are proper,
in /etc/hostname and /etc/sysconfig/network

Even then am not able to find why am getting this error. Also am able to
ping with my own hostname.

Regards,
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Re: [asterisk-users] IAX2 netsock error with name resolution

[Gopalakrishnan N]

After changing my dialplan as suggested, there is no socket error, but
getting Busy/Congested, and the call is hanging up, let me check that
part...

Earlier my dialplan was,
;exten = _2XXX,1,Dial(SIP/${EXTEN}@${MANIAX},30)

and I changed like this exten = _2XXX,1,Dial(${MANIAX}/${EXTEN},30)

whether the SIP matters?

And now since its a SIP extension in other side, am getting failed because
the extension is not able to find.


Regards.


On Sun, Jun 23, 2013 at 5:22 PM, Alec Davis siva...@paradise.net.nz wrote:

 snip
  -- Executing [2001@Test:1] Dial(SIP/4090-0005,
 SIP/2001@IAX2/IND-MAN,30) in new stack
  [Jun 23 06:31:36] NOTICE[4383][C-0005]: chan_sip.c:29491
 sip_request_call: Conflicting extension values given. Using '2001' and not
 'IND-MAN'
== Using SIP RTP CoS mark 5
  [Jun 23 06:31:36] ERROR[4383][C-0005]: netsock2.c:269
 ast_sockaddr_resolve: getaddrinfo(IAX2, (null), ...): Temporary failure
 in name resolution
  [Jun 23 06:31:36] WARNING[4383][C-0005]: chan_sip.c:6191 create_addr:
 No such host: IAX2
  [Jun 23 06:31:36] WARNING[4383][C-0005]: app_dial.c:2437
 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 -
 Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)

 Try this syntax Dial(IAX2/IND-MAN/2001,30)
 Where IND-MAN is the name of a peer/friend [IND-MAN] defined in iax.conf
 and 2001 is the extension on the remote system 'IND-MAN' where 2001 dials
 SIP/2001

 Alec Davis


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[asterisk-users] Asterisk Queue Frame

[Gopalakrishnan N]

What happens when we increase the queue frame size in channels.c

if ((queued_frames + new_frames  128 || queued_voice_frames +
new_voice_frames  96)) {

Be default it is 128 and 96 if i increase it to 256 and 192 what will
happen? will it impact to default behavior?


Regards,
Gopal.
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Re: [asterisk-users] Asterisk Queue Frame

[Gopalakrishnan N]

actually when i get the message my call volume is around 180 to 200
calls will that matter... and some calls being transferred to other
location and some are making outbound calls, some are inbound...

Is there any way for optimization?


On Fri, Jun 21, 2013 at 5:57 AM, Richard Mudgett rmudg...@digium.comwrote:

 On Thu, Jun 20, 2013 at 6:55 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 What happens when we increase the queue frame size in channels.c

 if ((queued_frames + new_frames  128 || queued_voice_frames +
 new_voice_frames  96)) {

 Be default it is 128 and 96 if i increase it to 256 and 192 what will
 happen? will it impact to default behavior?


 It looks like you are getting the Exceptionally long queue length
 warning message.  The
 change you mention will just increase the allowed size of the queue.
 Doing that won't really
 help much as it will just delay getting the message.  That warning message
 means Asterisk
 is falling behind in processing frames on the channel.

 Richard

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Re: [asterisk-users] Executing Script after MixMonitor is called

[Gopalakrishnan N]

Hi Satish,

I tried with sox, without any parameter, just sox filename.wav to
filename.mp3, in linux shell prompt... the file is been converted...

Now If i want to run that command using dialplan,

MixMonitor(filename.wav,m)
Monitor_Exec(sox filename.wav filename.mp3)

Or to use System command?

Regards..


On Fri, Jan 27, 2012 at 11:29 AM, Satish Barot satish4aster...@gmail.comwrote:

 This is how I use a wav to mp3 script on Mixmonitor in my dialplan
 (Asterisk 1.8.7.0).
 ...
 same = n,MixMonitor(${FILENAME},W(4),/var/spool/asterisk/wav2mp3
 ^{FILENAME})
 ...
 and my script is...

 #!/bin/bash

 WAV=/var/spool/asterisk/monitor/$1
 MP3=$(echo $1 | sed 's/\.wav$/.mp3/')
 MP3DEST=/var/spool/asterisk/mp3/$MP3
 /usr/bin/lame ${WAV} ${MP3DEST} --silent -b 16 -s 9.6 -m m --bitwidth
 8 --lowpass 9.6 --resample 8 --lowpass-width 1

 --SATISH BAROT
 Ahmedabad,India.


 On Wed, Jan 25, 2012 at 8:59 PM, Faraj Khasib fkha...@iconnecths.comwrote:

 Hello Guys,
 I am trying to convert files that are .wac to mp3 after mixmonitor
 command is called but it doesnt execute the command, I tried the command in
 terminal it worked, any help please ... below is my dial plan
 exten=6500,n,Set(MIXMONITOR_EXEC= nice -n 19 /usr/local/bin/lame -b 8
 -t -F -m m --bitwidth 8 --quiet
 /var/spool/asterisk/monitor/${CALLFILENAME}.wav
 /var/spool/asterisk/monitor/${CALLFILENAME}.mp3  rm -f
 /var/spool/asterisk/monitor/${CALLFILENAME}.wav)
 exten=6500,n,MixMonitor(${CALLFILENAME}.wav,b)

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[asterisk-users] Asterisk HA

[Gopalakrishnan N]

I was go through'ing the following links for HA,

https://wiki.asterisk.org/wiki/display/TOP/Failover+-+Linux - which doesn't
have file syncing.

https://www.johncahill.net/wiki/index.php/2_Node_Active/Passive_cluster -
this one has file syncing with pacemaker

Any other HA applications available or the lsyncd with pacemaker is good?

Regards
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[asterisk-users] Codec Mismatch

[Gopalakrishnan N]

Sometimes in huge call volume am facing this type of error,

[Jun  4 08:42:46] WARNING[8459][C-79fa]: channel.c:5075 ast_write:
Codec mismatch on channel Local/8038@xss-call-out-4774;1 setting write
format to slin from ulaw native formats (ulaw)
[Jun  4 08:43:04] WARNING[8285][C-79da]: channel.c:5075 ast_write:
Codec mismatch on channel Local/6513@xss-call-out-4775;1 setting write
format to slin from ulaw native formats (ulaw)
[Jun  4 08:43:10] WARNING[8790][C-7a2c]: channel.c:5075 ast_write:
Codec mismatch on channel Local/18002662279@xss-call-out-4778;1 setting
write format to slin from ulaw native formats (ulaw)
[Jun  4 08:43:23] WARNING[8355][C-79e6]: channel.c:5075 ast_write:
Codec mismatch on channel Local/2896@xss-call-out-4779;1 setting write
format to slin from ulaw native formats (ulaw)
[Jun  4 08:43:25] WARNING[7577][C-798a]: channel.c:5075 ast_write:
Codec mismatch on channel Local/2896@xss-call-out-477a;1 setting write
format to slin from ulaw native formats (ulaw)


basically Asterisk will do the slin to ulaw, hope there should not be any
problem...

But am not sure why am getting this error? will this affect my call?
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Re: [asterisk-users] Most suitable version for Production ENV

[Gopalakrishnan N]

Asterisk 1.8 is stable
On 1 Jun 2013 16:40, luke devon luke_de...@yahoo.com wrote:

 Hi

 As I seen on the Asterisk web site , there is packages called ,

 AsteriskLatest Version - 11.4.0

 asterisk-11-current.tar.gzhttp://downloads.asterisk.org/pub/telephony/asterisk/asterisk-11-current.tar.gz
  and

 asterisk-1.8-current.tar.gzhttp://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8-current.tar.gz

 May I now which one is the most suitable for a production environment ?

 Thanks in advance
 Luke


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[asterisk-users] Queue Periodic Announce not working...

[Gopalakrishnan N]

I am having a queue where included periodic announce like the below,

[test]
context = default
member = Agent/1001
member = Agent/1002
music = default
strategy = rrmemory
ringinuse = no
timeout = 15
retry = 1
maxlen = 0
joinempty = yes
leavewhenempty = no
periodic-announce = /var/lib/asterisk/sounds/en/test/AVG-15.wav
periodic-announce-frequency=30
random-periodic-announce=no
relative-periodic-announce=yes
wrapuptime = 10

When am entering into the queue, the CLI shows playing periodic announce
but actually its not playing. Even I do have the file in the proper
directory.

Below is the CLI log,


-- SIP/1001-0010 Playing 'avgtest/AVG-13.slin' (language 'en')
[May 30 05:50:07] NOTICE[2711]: chan_sip.c:24257 handle_request_subscribe:
Received SIP subscribe for peer without mailbox: 1001
-- Executing [500@default:10] Queue(SIP/1001-0010, avg) in new
stack
-- Started music on hold, class 'avgtest', on SIP/1001-0010
-- Stopped music on hold on SIP/1001-0010
-- Playing periodic announcement
-- Started music on hold, class 'avgtest', on SIP/1001-0010


Regards
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Re: [asterisk-users] Queue Periodic Announce not working...

[Gopalakrishnan N]

It works.

Thanks
 On 30 May 2013 19:39, Doug Lytle supp...@drdos.info wrote:

  periodic-announce = /var/lib/asterisk/sounds/en/test/AVG-15.wav

 Try it without the .wav

 Doug

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[asterisk-users] DTMF recognized after call establishment

[Gopalakrishnan N]

Hi,

I am receiving DTMF without any reason after call establishment.

The log as follows, and I suspect something related to directmedia,
[May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is
making progress passing it to SIP/MAN-000a4b48
[May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
answered SIP/MAN-000a4b48
[May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on
SIP/MyTrunk-000a4b49, duration 0 ms
[May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin '*'
on SIP/MyTrunk-000a4b49
[May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on
SIP/MyTrunk-000a4b49
[May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on
SIP/MyTrunk-000a4b49, duration 0 ms
[May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin '8'
on SIP/MyTrunk-000a4b49
[May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on
SIP/MyTrunk-000a4b49
[May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on
SIP/MAN-000a4af0, duration 100 ms
[May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with
duration 100 queued on SIP/MAN-000a4af0
[May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued on
SIP/MAN-000a4af0
[May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on
SIP/MAN-000a4b41, duration 100 ms
[May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with
duration 100 queued on SIP/MAN-000a4b41
[May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued on
SIP/MAN-000a4b41
[May 17 00:33:55] VERBOSE[4106] pbx.c:   == Spawn extension
(sip-trunk-inbound, 2127773456, 1) exited non-zero on 'SIP/MyTrunk-000a4af3'
[May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1]
NoOp(SIP/MAN-000a4b09, 16) in new stack
[May 17 00:33:56] VERBOSE[4136] pbx.c:   == Spawn extension
(trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09'

Is this some thing related to SIP RE-INVITE?

Thanks.
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Re: [asterisk-users] DTMF recognized after call establishment

[Gopalakrishnan N]

So any resolution for this?

I suspect it could be related to RE INVITE


On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad asghar...@gmail.comwrote:

 i had this in past there was an ATA configured to send 9 at the end of
 dialing in my case.


 On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi,

 I am receiving DTMF without any reason after call establishment.

 The log as follows, and I suspect something related to directmedia,
 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
 is making progress passing it to SIP/MAN-000a4b48
 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
 answered SIP/MAN-000a4b48
 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on
 SIP/MyTrunk-000a4b49, duration 0 ms
 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin
 '*' on SIP/MyTrunk-000a4b49
 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on
 SIP/MyTrunk-000a4b49
 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on
 SIP/MyTrunk-000a4b49, duration 0 ms
 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin
 '8' on SIP/MyTrunk-000a4b49
 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on
 SIP/MyTrunk-000a4b49
 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on
 SIP/MAN-000a4af0, duration 100 ms
 [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with
 duration 100 queued on SIP/MAN-000a4af0
 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued
 on SIP/MAN-000a4af0
 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on
 SIP/MAN-000a4b41, duration 100 ms
 [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with
 duration 100 queued on SIP/MAN-000a4b41
 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued
 on SIP/MAN-000a4b41
 [May 17 00:33:55] VERBOSE[4106] pbx.c:   == Spawn extension
 (sip-trunk-inbound, 2127773456, 1) exited non-zero on
 'SIP/MyTrunk-000a4af3'
 [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1]
 NoOp(SIP/MAN-000a4b09, 16) in new stack
 [May 17 00:33:56] VERBOSE[4136] pbx.c:   == Spawn extension
 (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09'

 Is this some thing related to SIP RE-INVITE?

 Thanks.


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Re: [asterisk-users] DTMF recognized after call establishment

[Gopalakrishnan N]

Let me try with dtmfmode as auto...
On 28 May 2013 19:32, Asghar Mohammad asghar...@gmail.com wrote:

 work around was block dtmf.
 set wrong type of dtmf in incoming trunk.


 On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 So any resolution for this?

 I suspect it could be related to RE INVITE


 On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad asghar...@gmail.comwrote:

 i had this in past there was an ATA configured to send 9 at the end of
 dialing in my case.


 On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi,

 I am receiving DTMF without any reason after call establishment.

 The log as follows, and I suspect something related to directmedia,
 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
 is making progress passing it to SIP/MAN-000a4b48
 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49
 answered SIP/MAN-000a4b48
 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on
 SIP/MyTrunk-000a4b49, duration 0 ms
 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin
 '*' on SIP/MyTrunk-000a4b49
 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on
 SIP/MyTrunk-000a4b49
 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on
 SIP/MyTrunk-000a4b49, duration 0 ms
 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin
 '8' on SIP/MyTrunk-000a4b49
 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on
 SIP/MyTrunk-000a4b49
 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on
 SIP/MAN-000a4af0, duration 100 ms
 [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8'
 with duration 100 queued on SIP/MAN-000a4af0
 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8'
 queued on SIP/MAN-000a4af0
 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on
 SIP/MAN-000a4b41, duration 100 ms
 [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1'
 with duration 100 queued on SIP/MAN-000a4b41
 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1'
 queued on SIP/MAN-000a4b41
 [May 17 00:33:55] VERBOSE[4106] pbx.c:   == Spawn extension
 (sip-trunk-inbound, 2127773456, 1) exited non-zero on
 'SIP/MyTrunk-000a4af3'
 [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing
 [h@trunk-outbound:1] NoOp(SIP/MAN-000a4b09, 16) in new stack
 [May 17 00:33:56] VERBOSE[4136] pbx.c:   == Spawn extension
 (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09'

 Is this some thing related to SIP RE-INVITE?

 Thanks.


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Re: [asterisk-users] Asterisk 11 dtmf not recognised

[Gopalakrishnan N]

With Asterisk 1.8 I got it working.

Regards


On Sat, May 25, 2013 at 2:37 AM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 Tried info, rfc2833, inband and finally kept as auto.
 On 25 May 2013 02:20, Doug Lytle supp...@drdos.info wrote:

   dtmfmode=auto

 dtmfmode=info

 or

 dtmfmode=rfc2833

 Doug


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Re: [asterisk-users] Asterisk 11 dtmf not recognised

[Gopalakrishnan N]

Am using Read application to get the digit, since its recognizing... I
would like to get for 3 attempts and then after 3rd attempt it has to
playback some different message like entries exceeded.

My dialplan as,
exten = 100,1(begin),Playback(letters/a)
exten = 100,n,Set(rightPIN=1)
exten = 100,n,Read(inPIN,,1,skip,3,3) ; Attempts for 5 times with 3 seconds
of timeout
exten = 100,n,GotoIf($[${inPIN} = ${rightPIN}]?pin-accepted,1)
exten = 100,n,Playback(letters/c) ; Didn't go to pin-accepted, so play
badPIN and hangup
exten = pin-accepted,1,Playback(letters/b) ; correct pin, play


what happens its keep on asking to enter digit If my DTMF didnt match. Do i
need to use any return function... ?

Actually my goal is to ask for 3 times and if not matched then return to
some other application.

Thanks in advance.


On Sat, May 25, 2013 at 3:19 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 With Asterisk 1.8 I got it working.

 Regards


 On Sat, May 25, 2013 at 2:37 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Tried info, rfc2833, inband and finally kept as auto.
 On 25 May 2013 02:20, Doug Lytle supp...@drdos.info wrote:

   dtmfmode=auto

 dtmfmode=info

 or

 dtmfmode=rfc2833

 Doug


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Re: [asterisk-users] Asterisk 11 dtmf not recognised

[Gopalakrishnan N]

Finally got it working with 3 attempts by the fialplan,

exten = 300,1,Playback(letters/a)
exten = 300,n,Set(gottries=0)
exten = 300,n(getmore),Set(rightPIN=1)
exten = 300,n,Read(inPIN,,1,skip,3,3) ; Attempts for 3 times with 3 seconds
of timeout
exten = 300,n(gotdigit),GotoIf($[${inPIN} = ${rightPIN}]?pin-accepted,1)
exten = 300,n,Set(gottries=$[${gottries}+1];
exten = 300,n,GotoIf($[${LEN(${inPIN})} == 0]?reallynothing:gotdigit)
exten = 300,n(reallynothing),GotoIf($[${gottries}3]?done:getmore) ;
Attempts for 3 tries if greater than 3 then it will come out or else
getmore will called
exten = 300,n(done),Playback(letters/c) ; Didn't go to pin-accepted, so
play badPIN and hangup
exten = pin-accepted,1,Playback(letters/b) ; correct pin, play

Thanks
 On 25 May 2013 15:38, Gopalakrishnan N gopalakrishnan...@gmail.com
wrote:

 Am using Read application to get the digit, since its recognizing... I
 would like to get for 3 attempts and then after 3rd attempt it has to
 playback some different message like entries exceeded.

 My dialplan as,
 exten = 100,1(begin),Playback(letters/a)
 exten = 100,n,Set(rightPIN=1)
 exten = 100,n,Read(inPIN,,1,skip,3,3) ; Attempts for 5 times with 3
 seconds of timeout
 exten = 100,n,GotoIf($[${inPIN} = ${rightPIN}]?pin-accepted,1)
 exten = 100,n,Playback(letters/c) ; Didn't go to pin-accepted, so play
 badPIN and hangup
 exten = pin-accepted,1,Playback(letters/b) ; correct pin, play


 what happens its keep on asking to enter digit If my DTMF didnt match. Do
 i need to use any return function... ?

 Actually my goal is to ask for 3 times and if not matched then return to
 some other application.

 Thanks in advance.


 On Sat, May 25, 2013 at 3:19 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 With Asterisk 1.8 I got it working.

 Regards


 On Sat, May 25, 2013 at 2:37 AM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Tried info, rfc2833, inband and finally kept as auto.
 On 25 May 2013 02:20, Doug Lytle supp...@drdos.info wrote:

   dtmfmode=auto

 dtmfmode=info

 or

 dtmfmode=rfc2833

 Doug


 --
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 Those who would give up Essential Liberty to purchase a little
 Temporary Safety, deserve neither Liberty nor Safety.

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Re: [asterisk-users] Asterisk 11 dtmf not recognised

[Gopalakrishnan N]

Tried info, rfc2833, inband and finally kept as auto.
On 25 May 2013 02:20, Doug Lytle supp...@drdos.info wrote:

  dtmfmode=auto

 dtmfmode=info

 or

 dtmfmode=rfc2833

 Doug


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[asterisk-users] GotoIf function

[Gopalakrishnan N]

Hi,

Actually i would like to get the input from the user and he should not try
more than 3 times, he can try more than 3 times, if yes it will get routed
to the next priority and if not it goes to the loopback again from the
beginning.

And following is the one I created, I just want to know whether this will
validate the input and will allow for 3 times

exten = s,1,GotoIfTime(08:00-09:00,mon-fri,*,*?2:avgtech,1)
exten = s,n,Background(voicemessage_1)
exten = s,n(voicemessage2),Background(voicemessage_2)

exten = s,n(begin),Set(wait=2)
exten = s,n,Set(gottries=0)
exten = s,n,Read(get,silence/1${wait})

exten = s,n(gotnothing),Set(gottries=$[${gottries}+1]
exten = s,n,GotoIf($[${LEN(${get})} == 0]?reallynothing:gotdigit)
exten = s,n(reallynothing),GotoIf($[${gottries}3]?done:voicemessage5)
exten = s,n(done),Background(voicemessage3)
exten = s,n,Background(voicemessage4)
exten = s,n,Playback(moh)
exten = s,n, ; Addittional messageing
exten = s,n,Queue(general technical team)

exten = s,n(voicemessage5),Goto(voicemessage2)

exten = s,n(gotdigit),Set(got=${get})
exten = s,n,GotoIf( $[ ${got} = 1]?doneinstall)
exten = s,n(doneinstall),Background(voicemessage3)
exten = s,n,Background(voicemessage4)
exten = s,n,Playback(moh)
exten = s,n, ; Addittional messageing
exten = s,n,Queue(installation technical skill)

exten = s,n,GotoIf( $[ ${got} = 2]?done2)
exten = s,n(done2),Background(voicemessage6)
exten = s,n,Goto(begin2)
exten = s,n(begin2),Set(wait=2)
exten = s,n,Set(gottries=0)
exten = s,n,Read(get,silence/1${wait})
exten = s,n(gotnothing),Set(gottries=$[${gottries}+1]
exten = s,n,GotoIf($[${LEN(${get})} == 0]?reallynothing:gotdigit2)
exten = s,n(reallynothing),GotoIf($[${gottries}3]?done:option2)
exten = s,n(done),Background(voicemessage3)
exten = s,n,Background(voicemessage4)
exten = s,n,Playback(moh)
exten = s,n, ; Addittional messageing
exten = s,n,Queue(general technical skill)

exten = s,n(option2),Background(voicemessage5)
exten = s,n,Goto(done2)

and so on... for digit 3...

Thanks in advance...

Regards.
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Re: [asterisk-users] GotoIf function

[Gopalakrishnan N]

I just want to make some increment... to 3 and yes go to the desired option
not to one more option.




On Thu, May 23, 2013 at 7:19 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 Hi,

 Actually i would like to get the input from the user and he should not try
 more than 3 times, he can try more than 3 times, if yes it will get routed
 to the next priority and if not it goes to the loopback again from the
 beginning.

 And following is the one I created, I just want to know whether this will
 validate the input and will allow for 3 times

 exten = s,1,GotoIfTime(08:00-09:00,mon-fri,*,*?2:avgtech,1)
 exten = s,n,Background(voicemessage_1)
 exten = s,n(voicemessage2),Background(voicemessage_2)

 exten = s,n(begin),Set(wait=2)
 exten = s,n,Set(gottries=0)
 exten = s,n,Read(get,silence/1${wait})

 exten = s,n(gotnothing),Set(gottries=$[${gottries}+1]
 exten = s,n,GotoIf($[${LEN(${get})} == 0]?reallynothing:gotdigit)
 exten = s,n(reallynothing),GotoIf($[${gottries}3]?done:voicemessage5)
 exten = s,n(done),Background(voicemessage3)
 exten = s,n,Background(voicemessage4)
 exten = s,n,Playback(moh)
 exten = s,n, ; Addittional messageing
 exten = s,n,Queue(general technical team)

 exten = s,n(voicemessage5),Goto(voicemessage2)

 exten = s,n(gotdigit),Set(got=${get})
 exten = s,n,GotoIf( $[ ${got} = 1]?doneinstall)
 exten = s,n(doneinstall),Background(voicemessage3)
 exten = s,n,Background(voicemessage4)
 exten = s,n,Playback(moh)
 exten = s,n, ; Addittional messageing
 exten = s,n,Queue(installation technical skill)

 exten = s,n,GotoIf( $[ ${got} = 2]?done2)
  exten = s,n(done2),Background(voicemessage6)
 exten = s,n,Goto(begin2)
 exten = s,n(begin2),Set(wait=2)
 exten = s,n,Set(gottries=0)
 exten = s,n,Read(get,silence/1${wait})
 exten = s,n(gotnothing),Set(gottries=$[${gottries}+1]
 exten = s,n,GotoIf($[${LEN(${get})} == 0]?reallynothing:gotdigit2)
 exten = s,n(reallynothing),GotoIf($[${gottries}3]?done:option2)
 exten = s,n(done),Background(voicemessage3)
 exten = s,n,Background(voicemessage4)
 exten = s,n,Playback(moh)
 exten = s,n, ; Addittional messageing
 exten = s,n,Queue(general technical skill)

 exten = s,n(option2),Background(voicemessage5)
 exten = s,n,Goto(done2)

 and so on... for digit 3...

 Thanks in advance...

 Regards.

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Re: [asterisk-users] Error 488 Not Acceptable Here

[Gopalakrishnan N]

488 not acceptable is due to codec error. Make sure you have right codec in
place between the end points.


On Fri, May 24, 2013 at 12:18 AM, Maximilian Grobecker 
m.grobec...@portunity.de wrote:

 Hi,

 Maybe you have not allowed T.38 as acceptable codec ;-)
 You can try with allow=all in your sip.conf.


 Am 22.05.2013 16:39, schrieb Andrew Colin:
  Hi guys,
 
  Any idea why I am getting this error when someone tries to send me a T38
  Fax?
 
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[asterisk-users] VoIP Incoming Issue

[Gopalakrishnan N]

I have made the SIP bind port to 5070, and already I have one VoIP trunk
configured in my Asterisk 1.6.

Now the problem is after changing the bind port at some point of time, am
not able to dial in the DID number of the VoIP trunk!

Changing the bind port matters for this?

Regards.
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Re: [asterisk-users] debug strategy for one-way audio calls

[Gopalakrishnan N]

@Marrie For one way audio as a debug strategy you can enable RTP debug and
see whether you have both way packets flow SENT and GOT.

Regards


On Thu, May 2, 2013 at 6:05 PM, Johan Wilfer li...@jttech.se wrote:

 2013-05-02 13:19, Marie Fischer skrev:
  Hello everybody,
 
  from time to time, we get so-called simplex / one-way audio calls, where
 one party cannot hear the other. The only thing in common is that is does
 happen with calls via SIP trunk, not ISDN and not internal calls. Nothing
 strange in verbose and SIP logs. Could even be some weird intermittent
 firewall issue I guess.
 
  Apart from logging all traffic 24/7 via tcpdump (not really convenient),
 can you give me some ideas how to debug this kind of issue?
 
  Asterisk 1.8.11-cert10 on CentOS 6.3, if that matters.
 

 Voipmonitor.org is great for debugging voip. You can either use only the
 sniffer (opensource) and use mysql + the pcap files or you can also buy
 the commercial webgui. Either way, it's a great product.

 /Johan

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[asterisk-users] Asterisk with R2D configuration

[Gopalakrishnan N]

Hi,

Has anybody worked on R2D Brazillian setup. I have configured R2 using
OpenR2 with Asterisk.

While doing some analysis I found R2D is already included in libopenr2.

Have anyone tested the same.

Regards,
Gopal.
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Re: [asterisk-users] SIP Retransmitting REGISTER message

[Gopalakrishnan N]

I have registered in sip.conf and in my network i am not using any port
forwarding kind of stuff (NAT), Asterisk server is directly connected to
Internet and the Internet router doesn't have any firewall.

And attached is asterisk log, that SIP REGISTER messages keep on sending
and no response from the server.

I am sure that this is some network issue, because the same account i
tested in different network (Network B) in some other place and it got
registered, even i am able to make call.

One thing which i don't understand is in same network (Network A) in xlite
phone the account is getting registered and not in Asterisk server.

I just want to isolate things why I am not getting any response, or
somewhere the response is getting lost! :(


Regards,
Gopal.

On Wed, Sep 26, 2012 at 6:32 PM, SamyGo govoi...@gmail.com wrote:

 Hi,
 How are you connected to server ? How have you configured your asterisk
 server to register to other side ? What about any NAT involved in your
 scenario ?Turn on sip debug and share your registrations.

 BR
 Sammy
 On Sep 26, 2012 5:54 PM, Danny Nicholas da...@debsinc.com wrote:

 Another possibility – you registered from the softphone first and the
 provider took the IP address from your PC and “locked out” the IP address
 of your Asterisk server.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Wednesday, September 26, 2012 7:51 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] SIP Retransmitting REGISTER message

 ** **

 there is no firewall, its just the router gave by the service provider.
 May be the SIP port issue?

 ** **

 Regards.

 ** **

 On Wed, Sep 26, 2012 at 6:17 PM, Danny Nicholas da...@debsinc.com
 wrote:

 The Asterisk server and softphone are hitting the firewall from two
 different points.  Start there.

  

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Wednesday, September 26, 2012 7:45 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] SIP Retransmitting REGISTER message

  

 Hi,

  

 I was trying to register a VoIP trunk in Asterisk , where its keep on
 sending Register message to the server, where I am not getting any response
 from server. 

  

 But whereas if i register in Xlite softphone the account is getting
 registered. 

  

 I suspect it could be network related issue, but since in softphone it is
 getting registered from the same network. 

  

 Any ideas to isolate things would be appreciated. 

  

 Regards,

 Gopal. 


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Nov 29 16:10:17 VERBOSE[5249] logger.c: Retransmitting #1 (NAT) to 
202.85.243.105:5060:
REGISTER sip:sip.pennytel.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK566dab4d;rport
From: sip:8889191...@sip.pennytel.com;tag=as107f0d7d
To: sip:8889191...@sip.pennytel.com
Call-ID: 7163c7526c3e0a70262c658d7d527...@1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: sip:Pennytel@117.223.64.95
Event: registration
Content-Length: 0


---
Nov 29 16:10:17 VERBOSE[5249] logger.c: Retransmitting #1 (NAT) to 
204.74.213.5:5061:
REGISTER sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 117.223.64.95:5060;branch=z9hG4bK27a09fca;rport
From: sip:905871...@sip.callwithus.com;tag=as614f747c
To: sip:905871...@sip.callwithus.com
Call-ID: 63649a661edcb5861eb8188844452...@1234.com
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Max-Forwards: 70
Expires: 120
Contact: sip:905871851@117.223.64.95
Event: registration
Content-Length: 0


---
Nov 29 16:10:17

Re: [asterisk-users] SIP Retransmitting REGISTER message

[Gopalakrishnan N]

yes this is the link http://www.callwithus.com/configuration am following,
and using the same, except type=friend i am using type=peer,

[general]
register = username:passw...@sip.callwithus.com

[callwithus]
type=peer
host=sip.callwithus.com
username=username
secret=password
qualify=no
insecure=invite
nat=yes

Also Asterisk server has access to Internet. I can able to ping
sip.callwithus.com.

The same account working in different network.

Regards.


On Thu, Sep 27, 2012 at 12:56 PM, SamyGo govoi...@gmail.com wrote:



 I have registered in sip.conf


 wow that was very detailed. I think I asked How have you configured this
 to register ? I'm pretty much sure you've nat related string mis-configured
 in your sip.conf.

 Can you tell if your asterisk server has access to internet !! I can see
 the same situation happening with callwithus register attempts !!

 See this page from callwithus and configure your asterisk accordingly for
 both accounts.

 http://www.callwithus.com/configuration

 BR
 Sammy



 On Thu, Sep 27, 2012 at 12:09 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 I have registered in sip.conf and in my network i am not using any port
 forwarding kind of stuff (NAT), Asterisk server is directly connected to
 Internet and the Internet router doesn't have any firewall.

 And attached is asterisk log, that SIP REGISTER messages keep on sending
 and no response from the server.

 I am sure that this is some network issue, because the same account i
 tested in different network (Network B) in some other place and it got
 registered, even i am able to make call.

 One thing which i don't understand is in same network (Network A) in
 xlite phone the account is getting registered and not in Asterisk server.

 I just want to isolate things why I am not getting any response, or
 somewhere the response is getting lost! :(


 Regards,
 Gopal.


 On Wed, Sep 26, 2012 at 6:32 PM, SamyGo govoi...@gmail.com wrote:

 Hi,
 How are you connected to server ? How have you configured your asterisk
 server to register to other side ? What about any NAT involved in your
 scenario ?Turn on sip debug and share your registrations.

 BR
 Sammy
 On Sep 26, 2012 5:54 PM, Danny Nicholas da...@debsinc.com wrote:

 Another possibility – you registered from the softphone first and the
 provider took the IP address from your PC and “locked out” the IP address
 of your Asterisk server.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan
 N
 *Sent:* Wednesday, September 26, 2012 7:51 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] SIP Retransmitting REGISTER message

 ** **

 there is no firewall, its just the router gave by the service provider.
 May be the SIP port issue?

 ** **

 Regards.

 ** **

 On Wed, Sep 26, 2012 at 6:17 PM, Danny Nicholas da...@debsinc.com
 wrote:

 The Asterisk server and softphone are hitting the firewall from two
 different points.  Start there.

  

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan
 N
 *Sent:* Wednesday, September 26, 2012 7:45 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] SIP Retransmitting REGISTER message

  

 Hi,

  

 I was trying to register a VoIP trunk in Asterisk , where its keep on
 sending Register message to the server, where I am not getting any response
 from server. 

  

 But whereas if i register in Xlite softphone the account is getting
 registered. 

  

 I suspect it could be network related issue, but since in softphone it
 is getting registered from the same network. 

  

 Any ideas to isolate things would be appreciated. 

  

 Regards,

 Gopal. 


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[asterisk-users] SIP Retransmitting REGISTER message

[Gopalakrishnan N]

Hi,

I was trying to register a VoIP trunk in Asterisk , where its keep on
sending Register message to the server, where I am not getting any response
from server.

But whereas if i register in Xlite softphone the account is getting
registered.

I suspect it could be network related issue, but since in softphone it is
getting registered from the same network.

Any ideas to isolate things would be appreciated.

Regards,
Gopal.
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Re: [asterisk-users] SIP Retransmitting REGISTER message

[Gopalakrishnan N]

there is no firewall, its just the router gave by the service provider. May
be the SIP port issue?

Regards.

On Wed, Sep 26, 2012 at 6:17 PM, Danny Nicholas da...@debsinc.com wrote:

 The Asterisk server and softphone are hitting the firewall from two
 different points.  Start there.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Wednesday, September 26, 2012 7:45 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] SIP Retransmitting REGISTER message

 ** **

 Hi,

 ** **

 I was trying to register a VoIP trunk in Asterisk , where its keep on
 sending Register message to the server, where I am not getting any response
 from server. 

 ** **

 But whereas if i register in Xlite softphone the account is getting
 registered. 

 ** **

 I suspect it could be network related issue, but since in softphone it is
 getting registered from the same network. 

 ** **

 Any ideas to isolate things would be appreciated. 

 ** **

 Regards,

 Gopal. 

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Re: [asterisk-users] SIP Retransmitting REGISTER message

[Gopalakrishnan N]

But even then all the IP go via router, so when it goes to service provider
it will go as the same IP address, since its coming from the same network.

Because the softphone and asterisk machine are local network which is
commonly connected to a router.

Regards.

On Wed, Sep 26, 2012 at 6:31 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 ahh... ! OK.. I though of this...



 On Wed, Sep 26, 2012 at 6:24 PM, Danny Nicholas da...@debsinc.com wrote:

 Another possibility – you registered from the softphone first and the
 provider took the IP address from your PC and “locked out” the IP address
 of your Asterisk server.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Wednesday, September 26, 2012 7:51 AM

 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] SIP Retransmitting REGISTER message

 ** **

 there is no firewall, its just the router gave by the service provider.
 May be the SIP port issue?

 ** **

 Regards.

 ** **

 On Wed, Sep 26, 2012 at 6:17 PM, Danny Nicholas da...@debsinc.com
 wrote:

 The Asterisk server and softphone are hitting the firewall from two
 different points.  Start there.

  

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Wednesday, September 26, 2012 7:45 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] SIP Retransmitting REGISTER message

  

 Hi,

  

 I was trying to register a VoIP trunk in Asterisk , where its keep on
 sending Register message to the server, where I am not getting any response
 from server. 

  

 But whereas if i register in Xlite softphone the account is getting
 registered. 

  

 I suspect it could be network related issue, but since in softphone it is
 getting registered from the same network. 

  

 Any ideas to isolate things would be appreciated. 

  

 Regards,

 Gopal. 


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Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi,

I have started asterisk using strace, and the log is listed in pastebin
http://pastebin.com/ry2Q1e6x

Moreover, for some peoples Asterisk is properly installed in OpenSuse 12.1
(i586), can you please correct me with the installation steps what I did,
my steps as follows,

   1. OpenSuse fresh installation
   2. Login to root in terminal (sudo -i)
   3. Download libpri, dahdi and Asterisk
   4. Install libpri and dahdi (even though I am not using any dahdi
   hardware) - make and make install
   5. Installation of Asterisk (./configure, make menuconfig, make, make
   install and make samples)
   6. Start Asterisk (asterisk -c) - here hangs while loading modules.

any other packages has to be installed or the installation is fine! please
advice!

Regards.


On Thu, Aug 30, 2012 at 7:03 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:

 On Thu, Aug 30, 2012 at 01:42:06PM +0200, Patrick Lists wrote:
  On 08/30/2012 09:45 AM, Gopalakrishnan N wrote:
  Hi,
  
  I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host,
  I am not using any virtualbox, still i struck in loading the modules.
 
  Please do not top post.
 
  Install strace and then start asterisk with the command:
  # strace asterisk

 Asterisk will fork into the background and the process you trace will
 exit.

   strace -f asterisk #?
   strace asterisk -f #?

 Just in case you wonder, 'asterisk -f strace' will not work as you might
 have expected from the above examples. Nither will '-f strace asterisk'.

 '-U asterisk ' may also come in handy.

 --
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] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi,

I have installed libpri, Dahdi 2.6 and Asterisk 1.8.15, in normal host, I
am not using any virtualbox, still i struck in loading the modules.

Regards.


On Tue, Aug 28, 2012 at 10:47 PM, Bryant Zimmerman brya...@zktech.comwrote:

 I would install both dahdi and libpri. I brought up a 12.2 RC-2 VM on
 hyper-v Windows 8 and followed our standard asterisk build and have no
 issues yet but we have not run full testing to confirm.  Also a point of
 not 12.2 is RC for the next 8 days or so.


 Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003


 --
 *From*: Gopalakrishnan N gopalakrishnan...@gmail.com
 *Sent*: Tuesday, August 28, 2012 1:13 PM
 *To*: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com

 *Subject*: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
 12.2


 If I don't need to install dahdi hardware, is it really I need to have
 libpri installed?

 Regards.
 On Aug 28, 2012 10:26 PM, Danny Nicholas da...@debsinc.com wrote:

  Check Jason Parker’s post from today and see if you skipped any of the
 preliminary build steps.  It is possible that something like libpri is
 biting you.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Tuesday, August 28, 2012 11:52 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk hangs while starting in
 OpenSuse 12.2



 I tried that too, what happens is asterisk is loading but after that if I
 try to start any one module for example chan_sip.so, asterisk hangs.

 Regards.

 On Aug 28, 2012 6:44 PM, Danny Nicholas da...@debsinc.com wrote:

 Extensions/trunks.  Another thought is that you might make your
 modules.conf not load anything to start with so you can eliminate a rogue
 module as the problem.  Just change autoload=yes to autoload=no.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Monday, August 27, 2012 11:47 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk hangs while starting in
 OpenSuse 12.2



 Hi danny,

 Are you talking about modules or sip extensions and dahdi extensions
 because its a fresh installation also it doesn't have dahdi interface, I am
 just trying to have only ip side.

 Regards

 On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote:

 I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and
 10 SP2).  My advice would be to try to start the box with as few SIP/DAHDI
 channels as possible to begin with and add as you get things stable.



 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Monday, August 27, 2012 8:52 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk hangs while starting in
 OpenSuse 12.2



 Hi Patrick,



 With other OS it works like charm. Only with OpenSuse, I am facing this
 issue, since I have a situation to stick with OpenSuse, I am struggling in
 this.



 Regards.

 On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists 
 asterisk-l...@puzzled.xs4all.nl wrote:

 On 27-08-12 08:25, Gopalakrishnan N wrote:

 This is really tuff working with OpenSuse. I am clueless how to sort our
 this.



 Maybe switch to a different distribution? I have used CentOS and RHEL for
 years without any problems and as far as I know both debian and ubuntu
 should work without problems too.

 Regards,
 Patrick




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Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

I tried that too, what happens is asterisk is loading but after that if I
try to start any one module for example chan_sip.so, asterisk hangs.

Regards.
On Aug 28, 2012 6:44 PM, Danny Nicholas da...@debsinc.com wrote:

 Extensions/trunks.  Another thought is that you might make your
 modules.conf not load anything to start with so you can eliminate a rogue
 module as the problem.  Just change autoload=yes to autoload=no.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Monday, August 27, 2012 11:47 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
 12.2

 ** **

 Hi danny,

 Are you talking about modules or sip extensions and dahdi extensions
 because its a fresh installation also it doesn't have dahdi interface, I am
 just trying to have only ip side. 

 Regards

 On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote:

 I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and
 10 SP2).  My advice would be to try to start the box with as few SIP/DAHDI
 channels as possible to begin with and add as you get things stable.

  

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Monday, August 27, 2012 8:52 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
 12.2

  

 Hi Patrick,

  

 With other OS it works like charm. Only with OpenSuse, I am facing this
 issue, since I have a situation to stick with OpenSuse, I am struggling in
 this. 

  

 Regards. 

 On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists 
 asterisk-l...@puzzled.xs4all.nl wrote:

 On 27-08-12 08:25, Gopalakrishnan N wrote:

 This is really tuff working with OpenSuse. I am clueless how to sort our
 this.

  

 Maybe switch to a different distribution? I have used CentOS and RHEL for
 years without any problems and as far as I know both debian and ubuntu
 should work without problems too.

 Regards,
 Patrick





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Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

If I don't need to install dahdi hardware, is it really I need to have
libpri installed?

Regards.
 On Aug 28, 2012 10:26 PM, Danny Nicholas da...@debsinc.com wrote:

 Check Jason Parker’s post from today and see if you skipped any of the
 preliminary build steps.  It is possible that something like libpri is
 biting you.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Tuesday, August 28, 2012 11:52 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
 12.2

 ** **

 I tried that too, what happens is asterisk is loading but after that if I
 try to start any one module for example chan_sip.so, asterisk hangs.

 Regards.

 On Aug 28, 2012 6:44 PM, Danny Nicholas da...@debsinc.com wrote:

 Extensions/trunks.  Another thought is that you might make your
 modules.conf not load anything to start with so you can eliminate a rogue
 module as the problem.  Just change autoload=yes to autoload=no.

  

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Monday, August 27, 2012 11:47 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
 12.2

  

 Hi danny,

 Are you talking about modules or sip extensions and dahdi extensions
 because its a fresh installation also it doesn't have dahdi interface, I am
 just trying to have only ip side. 

 Regards

 On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote:

 I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and
 10 SP2).  My advice would be to try to start the box with as few SIP/DAHDI
 channels as possible to begin with and add as you get things stable.

  

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Monday, August 27, 2012 8:52 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
 12.2

  

 Hi Patrick,

  

 With other OS it works like charm. Only with OpenSuse, I am facing this
 issue, since I have a situation to stick with OpenSuse, I am struggling in
 this. 

  

 Regards. 

 On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists 
 asterisk-l...@puzzled.xs4all.nl wrote:

 On 27-08-12 08:25, Gopalakrishnan N wrote:

 This is really tuff working with OpenSuse. I am clueless how to sort our
 this.

  

 Maybe switch to a different distribution? I have used CentOS and RHEL for
 years without any problems and as far as I know both debian and ubuntu
 should work without problems too.

 Regards,
 Patrick





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Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi Bryant,

As you said, I dont have Hyper-V, I avoided virtualbox and tested in normal
host directly, even then it hangs while loading modules.
 *Asterisk Dynamic Loader Starting:*
*  == Parsing '/etc/asterisk/modules.conf':   == Found*
*[Aug 27 11:52:21] NOTICE[22886]: loader.c:1133 load_modules: 186 modules
will be loaded.*

This is really tuff working with OpenSuse. I am clueless how to sort our
this.

Regards.

On Fri, Aug 24, 2012 at 3:55 AM, Hans Witvliet aster...@a-domani.nl wrote:

 On Thu, 2012-08-23 at 15:01 +0530, Gopalakrishnan N wrote:
  Hi,
 
 
  Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1
  (32bit) version in virtualbox. Downloaded Asterisk 1.8.15. Installed,
  installation went fine.
 
 

 Have you tried the versions from the OBS?

 Or perhaps a virtualbox issue? Its notorious for vapourizing
 cpu-cycles...

 hw



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Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi Patrick,

With other OS it works like charm. Only with OpenSuse, I am facing this
issue, since I have a situation to stick with OpenSuse, I am struggling in
this.

Regards.

On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists 
asterisk-l...@puzzled.xs4all.nl wrote:

 On 27-08-12 08:25, Gopalakrishnan N wrote:

 This is really tuff working with OpenSuse. I am clueless how to sort our
 this.


 Maybe switch to a different distribution? I have used CentOS and RHEL for
 years without any problems and as far as I know both debian and ubuntu
 should work without problems too.

 Regards,
 Patrick




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

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Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi danny,

Are you talking about modules or sip extensions and dahdi extensions
because its a fresh installation also it doesn't have dahdi interface, I am
just trying to have only ip side.

Regards
On Aug 27, 2012 7:27 PM, Danny Nicholas da...@debsinc.com wrote:

 I use OpenSuse here (our production boxes are Suse Enterprise 11 SP1 and
 10 SP2).  My advice would be to try to start the box with as few SIP/DAHDI
 channels as possible to begin with and add as you get things stable.

 ** **

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Gopalakrishnan N
 *Sent:* Monday, August 27, 2012 8:52 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
 12.2

 ** **

 Hi Patrick,

 ** **

 With other OS it works like charm. Only with OpenSuse, I am facing this
 issue, since I have a situation to stick with OpenSuse, I am struggling in
 this. 

 ** **

 Regards. 

 On Mon, Aug 27, 2012 at 4:24 PM, Patrick Lists 
 asterisk-l...@puzzled.xs4all.nl wrote:

 On 27-08-12 08:25, Gopalakrishnan N wrote:

 This is really tuff working with OpenSuse. I am clueless how to sort our
 this.

 ** **

 Maybe switch to a different distribution? I have used CentOS and RHEL for
 years without any problems and as far as I know both debian and ubuntu
 should work without problems too.

 Regards,
 Patrick





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 ** **

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Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi,

Again I stuck up with OpenSuse 12.1 also, Installed OpenSuse 12.1 (32bit)
version in virtualbox. Downloaded Asterisk 1.8.15. Installed, installation
went fine.

While starting Asterisk, it hangs here,
*Asterisk Dynamic Loader Starting:*
*  == Parsing '/etc/asterisk/modules.conf':   == Found*
*[Aug 23 14:56:14] NOTICE[19340]: loader.c:1133 load_modules: 186 modules
will be loaded.*

any my linux machine uname -a output is below,
*Linux linux-w6le.site 3.1.0-1.2-default #1 SMP Thu Nov 3 14:45:45 UTC 2011
(187dde0) i686 i686 i386 GNU/Linux*
*
*
Any suggestion would be much appreciated.

Regards,
Gopal.

On Tue, Aug 21, 2012 at 11:24 AM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 Ok Thanks Bryant, let me try with OpenSuse 12.1.

 Regards.


 On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman brya...@zktech.comwrote:

 I have the current version of 8.x and 10.x on systems. I am using
 OpenSuse 12.1, We are working on getting a 12.2 boxs up just running out of
 time. Asterisk on all of our boxes are complied from source.

 Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003


 --
 *From*: Gopalakrishnan N gopalakrishnan...@gmail.com
 *Sent*: Monday, August 20, 2012 10:11 AM
 *To*: Bryant Zimmerman brya...@zktech.com
 *Subject*: Re: [asterisk-users] Asterisk hangs while starting in
 OpenSuse 12.2


 It's really glad that asterisk is installed at your machine in open suse.
 Can you let me know which version you are using and the architecture.

 Regards.
 On Aug 20, 2012 6:22 PM, Bryant Zimmerman brya...@zktech.com wrote:

 I compile from source..

 Sent from my Verizon Wireless Phone

 - Reply message -
 From: Gopalakrishnan N gopalakrishnan...@gmail.com
 Date: Mon, Aug 20, 2012 8:15 am
 Subject: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com

  From the forum I understand OpenSuse 12.2 is pre-relase and better to
 use OpenSuse 12.1. Lets check with OpenSuse 12.1.

  Regards.


 On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Its really weird working with OpenSuse. I am not sure how others are
 using with OpenSuse. Through Yast also I tried to install Asterisk package,
 it didn't find.

  Now I am clueless to work with OpenSuse.



  Regards.


 On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi Patrick,

  Thanks for your suggestion, even though I added my hostname in the
 /etc/hosts, still the problem persists. Also I tried to install in 
 OpenSuse
 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
 hanging at modules while starting Asterisk.

  Regards,
 Gopal.



  Please do not top post and properly trim your replies.

 Have you made sure that on the OpenSuse box your DNS is configured
 properly? You should be able to lookup your IP address/FQDN both ways. So
 for example 192.168.1.1 (replace with your IP adres) should resolve in
 your.box.com (replace with your FQDN) and vice versa your.box.comshould 
 resolve into 192.168.1.1. See man dig or man nslookup for commands
 that can do DNS lookups.

 Regards,
 Patrick




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 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Its really weird working with OpenSuse. I am not sure how others are using
with OpenSuse. Through Yast also I tried to install Asterisk package, it
didn't find.

Now I am clueless to work with OpenSuse.



Regards.


On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 Hi Patrick,

 Thanks for your suggestion, even though I added my hostname in the
 /etc/hosts, still the problem persists. Also I tried to install in OpenSuse
 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
 hanging at modules while starting Asterisk.

 Regards,
 Gopal.



 Please do not top post and properly trim your replies.

 Have you made sure that on the OpenSuse box your DNS is configured
 properly? You should be able to lookup your IP address/FQDN both ways. So
 for example 192.168.1.1 (replace with your IP adres) should resolve in
 your.box.com (replace with your FQDN) and vice versa your.box.com should
 resolve into 192.168.1.1. See man dig or man nslookup for commands that can
 do DNS lookups.

 Regards,
 Patrick




 --
 __**__**_
 -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users



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_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
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To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

From the forum I understand OpenSuse 12.2 is pre-relase and better to use
OpenSuse 12.1. Lets check with OpenSuse 12.1.

Regards.


On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N 
gopalakrishnan...@gmail.com wrote:

 Its really weird working with OpenSuse. I am not sure how others are using
 with OpenSuse. Through Yast also I tried to install Asterisk package, it
 didn't find.

 Now I am clueless to work with OpenSuse.



 Regards.


 On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi Patrick,

 Thanks for your suggestion, even though I added my hostname in the
 /etc/hosts, still the problem persists. Also I tried to install in OpenSuse
 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
 hanging at modules while starting Asterisk.

 Regards,
 Gopal.



 Please do not top post and properly trim your replies.

 Have you made sure that on the OpenSuse box your DNS is configured
 properly? You should be able to lookup your IP address/FQDN both ways. So
 for example 192.168.1.1 (replace with your IP adres) should resolve in
 your.box.com (replace with your FQDN) and vice versa your.box.comshould 
 resolve into 192.168.1.1. See man dig or man nslookup for commands
 that can do DNS lookups.

 Regards,
 Patrick




 --
 __**__**
 _
 -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users




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

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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Ok Thanks Bryant, let me try with OpenSuse 12.1.

Regards.

On Mon, Aug 20, 2012 at 7:46 PM, Bryant Zimmerman brya...@zktech.comwrote:

 I have the current version of 8.x and 10.x on systems. I am using OpenSuse
 12.1, We are working on getting a 12.2 boxs up just running out of time.
 Asterisk on all of our boxes are complied from source.

 Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003


 --
 *From*: Gopalakrishnan N gopalakrishnan...@gmail.com
 *Sent*: Monday, August 20, 2012 10:11 AM
 *To*: Bryant Zimmerman brya...@zktech.com
 *Subject*: Re: [asterisk-users] Asterisk hangs while starting in OpenSuse
 12.2


 It's really glad that asterisk is installed at your machine in open suse.
 Can you let me know which version you are using and the architecture.

 Regards.
 On Aug 20, 2012 6:22 PM, Bryant Zimmerman brya...@zktech.com wrote:

 I compile from source..

 Sent from my Verizon Wireless Phone

 - Reply message -
 From: Gopalakrishnan N gopalakrishnan...@gmail.com
 Date: Mon, Aug 20, 2012 8:15 am
 Subject: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2
 To: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com

  From the forum I understand OpenSuse 12.2 is pre-relase and better to
 use OpenSuse 12.1. Lets check with OpenSuse 12.1.

  Regards.


 On Mon, Aug 20, 2012 at 5:34 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Its really weird working with OpenSuse. I am not sure how others are
 using with OpenSuse. Through Yast also I tried to install Asterisk package,
 it didn't find.

  Now I am clueless to work with OpenSuse.



  Regards.


 On Fri, Aug 17, 2012 at 2:38 PM, Gopalakrishnan N 
 gopalakrishnan...@gmail.com wrote:

 Hi Patrick,

  Thanks for your suggestion, even though I added my hostname in the
 /etc/hosts, still the problem persists. Also I tried to install in OpenSuse
 12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
 hanging at modules while starting Asterisk.

  Regards,
 Gopal.



  Please do not top post and properly trim your replies.

 Have you made sure that on the OpenSuse box your DNS is configured
 properly? You should be able to lookup your IP address/FQDN both ways. So
 for example 192.168.1.1 (replace with your IP adres) should resolve in
 your.box.com (replace with your FQDN) and vice versa your.box.comshould 
 resolve into 192.168.1.1. See man dig or man nslookup for commands
 that can do DNS lookups.

 Regards,
 Patrick




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

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






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

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

Re: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi Patrick,

Thanks for your suggestion, even though I added my hostname in the
/etc/hosts, still the problem persists. Also I tried to install in OpenSuse
12.2 (32bit) in virtualbox (like vmware) even there I faced problem like
hanging at modules while starting Asterisk.

Regards,
Gopal.


 Please do not top post and properly trim your replies.

 Have you made sure that on the OpenSuse box your DNS is configured
 properly? You should be able to lookup your IP address/FQDN both ways. So
 for example 192.168.1.1 (replace with your IP adres) should resolve in
 your.box.com (replace with your FQDN) and vice versa your.box.com should
 resolve into 192.168.1.1. See man dig or man nslookup for commands that can
 do DNS lookups.

 Regards,
 Patrick




 --
 __**__**_
 -- 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-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users

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

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

[asterisk-users] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi,

I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and
downloaded Asterisk 1.8 current version, after installing Asterisk, while
starting Asterisk it hangs at the stage of loading modules.conf, I checked
the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but still
after updating through yast also i am facing the issue.

Have anybody faced this type of issue with OpenSuse 12.2, its really wired
working with OpenSuse 12.2, even i tried with OpenSuse 12.1 as well which
results to same failure.


Regards,
Gopal.
--
_
-- 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] Asterisk hangs while starting in OpenSuse 12.2

[Gopalakrishnan N]

Hi,

Thanks for your comments. Even I tried with 12.1 also, its the same issue,
I am not sure whether it may be hardware related. Logs below,

OS details - uname -a
Linux laptop-prasad 3.3.0-2-desktop #1 SMP PREEMPT Sat Mar 24 00:11:53 UTC
2012 (7e9dd21) x86_64 x86_64 x86_64 GNU/Linux

while executing asterisk -c from the root prompt, its stuck as below
and the CPU usage is fully utilized,

  == Manager registered action DBPut
  == Manager registered action DBDel
  == Manager registered action DBDelTree
  == Parsing '/etc/asterisk/enum.conf':   == Found
  == Registered application 'CallCompletionRequest'
  == Registered application 'CallCompletionCancel'
  == Parsing '/etc/asterisk/ccss.conf':   == Found
 Asterisk Dynamic Loader Starting:
  == Parsing '/etc/asterisk/modules.conf':   == Found
[Aug 14 10:48:36] NOTICE[3805]: loader.c:1133 load_modules: 184 modules
will be loaded.


Any advice would be much appreciated.

Regards,
Gopal.


On Tue, Aug 14, 2012 at 3:37 AM, Bryant Zimmerman brya...@zktech.comwrote:

 I am running OpenSuse 12.1 with no issues. I have not tried 12.2 beta yet.

 Thanks

 Bryant Zimmerman (ZK Tech Inc.)
 616-855-1030 Ext. 2003


 --
 *From*: Gopalakrishnan N gopalakrishnan...@gmail.com
 *Sent*: Monday, August 13, 2012 8:19 AM
 *To*: Asterisk Users Mailing List - Non-Commercial Discussion 
 asterisk-users@lists.digium.com
 *Subject*: [asterisk-users] Asterisk hangs while starting in OpenSuse 12.2


 Hi,

  I am using OpenSuse 12.2 64bit OS which uses Kernel 3.3.x version and
 downloaded Asterisk 1.8 current version, after installing Asterisk, while
 starting Asterisk it hangs at the stage of loading modules.conf, I checked
 the forum https://issues.asterisk.org/jira/browse/ASTERISK-19245 but
 still after updating through yast also i am facing the issue.

  Have anybody faced this type of issue with OpenSuse 12.2, its really
 wired working with OpenSuse 12.2, even i tried with OpenSuse 12.1 as well
 which results to same failure.


  Regards,
 Gopal.


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

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

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

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