Re: [asterisk-users] Inquiry:How to convert *.wav files ?
Hello Hadi In beginning i also face this problem . I solved it by converting to SLN format. You also try to convert it to sln format. this link might help you http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk On Sat, Sep 26, 2009 at 10:44 AM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR3.wav FR3.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is reported as WAVE audio mono 8000 Hz) into 32 bit 44 KHz for better voice quality . Can you please help me . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to convert *.wav files ?
yeah it can :) On Sat, Sep 26, 2009 at 11:30 AM, hadi motamedi motamed...@gmail.comwrote: Thank you for your reply . Excuse me , you mean the Asterisk can play SLN files ? Can you please confirm ? On Sat, Sep 26, 2009 at 6:57 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello Hadi In beginning i also face this problem . I solved it by converting to SLN format. You also try to convert it to sln format. this link might help you http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk On Sat, Sep 26, 2009 at 10:44 AM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR3.wav FR3.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is reported as WAVE audio mono 8000 Hz) into 32 bit 44 KHz for better voice quality . Can you please help me . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Inquiry:Asterisk server remote access
Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to convert *.wav files ?
2009/9/26 hadi motamedi motamed...@gmail.com: I need to convert the original *.wav sound files (their file attribute is reported as WAVE audio mono 8000 Hz) into 32 bit 44 KHz for better voice quality . That's useless. You can do that of course, but even if you reencode the file, the quality of sound itself will not change. Also, unless you use a wideband codec, you won't be able to send more than 8000Hz over the line for most standard codecs (u/alaw, gsm, etc.). Just play the .wav you already have - you cannot get a better quality than that in the original file. -- KTHXBYE, Stanisław Pitucha, Gradwell Voip Engineer ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to convert *.wav files ?
Use Audocity Software Ravindra kumar On Sat, Sep 26, 2009 at 11:14 AM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR3.wav FR3.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is reported as WAVE audio mono 8000 Hz) into 32 bit 44 KHz for better voice quality . Can you please help me . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk server remote access
use Asterisk now software. You can access by IP. On Sat, Sep 26, 2009 at 2:11 PM, hadi motamedi motamed...@gmail.com wrote: Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk server remote access
hadi motamedi wrote: Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance SSH to the server. DC ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
Sorry My Question was not very clear. Asterisk System that is placed some where on local LAN (suppose in office A) A sip(or any other whose softphone is available) phone Client that is out side this local network (suppose at office B). now if I want the asterisk server to be avaiable for this sip phone. As Asterisk Server is also behind NAT. SIP phone is also in any other network. How can I make them communicate. As in LAN i can easily by giving asterisk server IP. On Sat, Sep 26, 2009 at 7:57 PM, Philipp Kempgen philipp.kemp...@amooma.dewrote: Abbas Shakeel wrote: I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. Depends on what your goals are. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configuration at router level and all things like that. http://www.voip-info.org/wiki/view/NAT+and+VOIP I also heard that in Pakistan VOIP is not allowed. We need to buy a liscense that is very expensive and so on ... What exactly is your question? http://catb.org/~esr/faqs/smart-questions.html#explicit http://catb.org/~esr/faqs/smart-questions.html#homework http://catb.org/~esr/faqs/smart-questions.html#keepcool *SCNR* Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to convert *.wav files ?
Hello Hadi While playing files extension is not specified. Remove the extension and Enjoy On Sat, Sep 26, 2009 at 3:13 PM, ravi kumar ravi...@gmail.com wrote: Use Audocity Software Ravindra kumar On Sat, Sep 26, 2009 at 11:14 AM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR3.wav FR3.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is reported as WAVE audio mono 8000 Hz) into 32 bit 44 KHz for better voice quality . Can you please help me . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk server remote access
Thank you for your reply . But I am seeking for PPPoE remote access that fits my case here . Can you please let me know if there is any solution in this regard ? (like PPPD) On Sat, Sep 26, 2009 at 12:16 PM, Michiel van Baak mich...@vanbaak.infowrote: On 09:41, Sat 26 Sep 09, hadi motamedi wrote: Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance ssh usern...@server -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk server remote access
On Sat, 26 Sep 2009, hadi motamedi wrote: Thank you for your reply . But I am seeking for PPPoE remote access that fits my case here . Can you please let me know if there is any solution in this regard ? (like PPPD) It would be really cool if iaxmodem would actually answer an incoming modem call and pass traffic to something like pppd. For those times when the pstn link is up, but something is wrong with the 'net connection... I think Sangoma lets you split a trunk into voice and data, but I suspect you don't want to lose channels other than dynamically... So the short answer is no, you will need a modem or a real net connection connected to your asterisk box for remote maintenance. j On Sat, Sep 26, 2009 at 12:16 PM, Michiel van Baak mich...@vanbaak.infowrote: On 09:41, Sat 26 Sep 09, hadi motamedi wrote: Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance ssh usern...@server -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
Don't put a SIP server behind destination NAT. Just don't. ABBAS SHAKEEL wrote: Sorry My Question was not very clear. Asterisk System that is placed some where on local LAN (suppose in office A) A sip(or any other whose softphone is available) phone Client that is out side this local network (suppose at office B). now if I want the asterisk server to be avaiable for this sip phone. As Asterisk Server is also behind NAT. SIP phone is also in any other network. How can I make them communicate. As in LAN i can easily by giving asterisk server IP. On Sat, Sep 26, 2009 at 7:57 PM, Philipp Kempgen philipp.kemp...@amooma.de mailto:philipp.kemp...@amooma.de wrote: Abbas Shakeel wrote: I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. Depends on what your goals are. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configuration at router level and all things like that. http://www.voip-info.org/wiki/view/NAT+and+VOIP I also heard that in Pakistan VOIP is not allowed. We need to buy a liscense that is very expensive and so on ... What exactly is your question? http://catb.org/~esr/faqs/smart-questions.html#explicit http://catb.org/~esr/faqs/smart-questions.html#homework http://catb.org/~esr/faqs/smart-questions.html#keepcool *SCNR* Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
are using some kind of router? On Sat, Sep 26, 2009 at 8:20 AM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: Hello I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configuration at router level and all things like that. I also heard that in Pakistan VOIP is not allowed. We need to buy a liscense that is very expensive and so on ... -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bayardo Sánchez García Web Developer - Internet Portals - Asterisk Support - Windows Server Support - Proxy Support - Linux Server E-mail: bayardo.sanc...@gmail.com Linux User: #418392 America Central - Managua, NI (505) 2249-2853 - 84886876 IM msn messenger: bjsanch...@hotmail.com Skype: bayardo.sanchez This email is intended solely for the person or organization to which it is addressed. It may contain privileged and confidential information. If you are not the intended recipient, you are prohibited from copying, disclosing or distributing this email or its contents (as it may be unlawful for you to do so) or taking any action in reliance on it. If you have received this email by mistake, please delete it. All e-mail sent to this address will be received by B.S. Solution e-mail system and is subject to archiving and review by someone other than the recipient. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk server remote access
On 09:41, Sat 26 Sep 09, hadi motamedi wrote: Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance app_dahdiras or app_zapras? --Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
Abbas Shakeel wrote: I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. Depends on what your goals are. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configuration at router level and all things like that. http://www.voip-info.org/wiki/view/NAT+and+VOIP I also heard that in Pakistan VOIP is not allowed. We need to buy a liscense that is very expensive and so on ... What exactly is your question? http://catb.org/~esr/faqs/smart-questions.html#explicit http://catb.org/~esr/faqs/smart-questions.html#homework http://catb.org/~esr/faqs/smart-questions.html#keepcool *SCNR* Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voiced E-mail
Does anyone have info or starter points on how to take emails from an external POP3 or IMAP server and cause them to be voiced by Asterisk? It is our e-mail server, so we can do anything to it. My question is concept or products required to get asterisk to do the job. Text-to-voice converter? Program to strip email down to just to, from, text, special mail box, have it call user, or have user call in? Whatever anyone that has done something like this would suggest. Thank you. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoiceMail.conf reading variables
Hi Is it possible to read the full name of the Voice Mail extension from voicemail.conf using VMauthenticate command ? as everytime I call VMauthenticate and try to feed in my password - it always returns VM_NAME as empty string . Alternatively let me know if there is any other way to read the VM_NAME for a particular extension Thanks Sriram ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk server remote access
Thank you for your reply . Can you please let me know if there is an facility to provide PPP over E1 as my Asterisk has ISDN PRI link outwards ? I mean if any facility inside Asterisk can provide PPP over E1 for remote access via ISDN PRI link ? On Sat, Sep 26, 2009 at 11:18 AM, ravi kumar ravi...@gmail.com wrote: use Asterisk now software. You can access by IP. On Sat, Sep 26, 2009 at 2:11 PM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOIP solutions
Hello I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configuration at router level and all things like that. I also heard that in Pakistan VOIP is not allowed. We need to buy a liscense that is very expensive and so on ... -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
Thanks Alex By just avoiding this will solve this problem? On Sat, Sep 26, 2009 at 9:47 PM, Alex Balashov abalas...@evaristesys.comwrote: Don't put a SIP server behind destination NAT. Just don't. ABBAS SHAKEEL wrote: Sorry My Question was not very clear. Asterisk System that is placed some where on local LAN (suppose in office A) A sip(or any other whose softphone is available) phone Client that is out side this local network (suppose at office B). now if I want the asterisk server to be avaiable for this sip phone. As Asterisk Server is also behind NAT. SIP phone is also in any other network. How can I make them communicate. As in LAN i can easily by giving asterisk server IP. On Sat, Sep 26, 2009 at 7:57 PM, Philipp Kempgen philipp.kemp...@amooma.de mailto:philipp.kemp...@amooma.de wrote: Abbas Shakeel wrote: I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. Depends on what your goals are. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configuration at router level and all things like that. http://www.voip-info.org/wiki/view/NAT+and+VOIP I also heard that in Pakistan VOIP is not allowed. We need to buy a liscense that is very expensive and so on ... What exactly is your question? http://catb.org/~esr/faqs/smart-questions.html#explicit http://catb.org/~esr/faqs/smart-questions.html#homework http://catb.org/~esr/faqs/smart-questions.html#keepcool *SCNR* Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - In which countries are ISDN subaddresses used ?
When did that happen? Added to libpri, someone beat me to it. What you may have seen is my recent minimal implementation as a patch to 1.6.1 - 1.6.2 https://issues.asterisk.org/view.php?id=15604, which is working, but deprecated. The task list to get it done properly for trunk Asterisk 1.6.3 is documented by Richard Mudgett by his note dated 25/09/09, at the link mentioned above. Regarding ISDN subadrress and associated costs, I can't speak for other countries, but here in New Zealand, it's enabled by default, and doesn't incur any extra costs that I'm aware of. Alec Davis _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier Sent: Saturday, 26 September 2009 9:06 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] OT - In which countries are ISDN subaddresses used ? Hi, I've seen this ISDN subaddress feature added to libpri. Which countries are using it ? How is this billed ? Do you have to pay an extra to your telco to benefit from this subaddresses ? Cheers ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to convert *.wav files ?
Thank you very much for your confirmation . Excuse me , the format needs to be like the followings ? exten = s-NOANSWER,n,playback(FR1.sln) Can you please do me favor and confirm if the above is correct ? On Sat, Sep 26, 2009 at 7:42 AM, ABBAS SHAKEEL shakeel.abbas@gmail.comwrote: A good way is to give try On Sat, Sep 26, 2009 at 11:41 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: yeah it can :) On Sat, Sep 26, 2009 at 11:30 AM, hadi motamedi motamed...@gmail.comwrote: Thank you for your reply . Excuse me , you mean the Asterisk can play SLN files ? Can you please confirm ? On Sat, Sep 26, 2009 at 6:57 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello Hadi In beginning i also face this problem . I solved it by converting to SLN format. You also try to convert it to sln format. this link might help you http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk On Sat, Sep 26, 2009 at 10:44 AM, hadi motamedi motamed...@gmail.com wrote: Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR3.wav FR3.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is reported as WAVE audio mono 8000 Hz) into 32 bit 44 KHz for better voice quality . Can you please help me . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:Asterisk server remote access
On 09:41, Sat 26 Sep 09, hadi motamedi wrote: Dear All Can you please do me favor and let me know if there is an facility in Asterisk server that can be used to have remote access to the server ? Please be informed that we have installed commissioned our Asterisk server at remote site with DECT telephony service provisioning for our subscribers . Can you please let me know if there is an facility in Asterisk pbx that can be used to provide remote access to the server for our maintenance duties ? Thank you in advance ssh usern...@server -- Michiel van Baak mich...@vanbaak.eu http://michiel.vanbaak.eu GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD Why is it drug addicts and computer aficionados are both called users? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Inquiry:How to convert *.wav files ?
A good way is to give try On Sat, Sep 26, 2009 at 11:41 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: yeah it can :) On Sat, Sep 26, 2009 at 11:30 AM, hadi motamedi motamed...@gmail.comwrote: Thank you for your reply . Excuse me , you mean the Asterisk can play SLN files ? Can you please confirm ? On Sat, Sep 26, 2009 at 6:57 AM, ABBAS SHAKEEL shakeel.abbas@gmail.com wrote: Hello Hadi In beginning i also face this problem . I solved it by converting to SLN format. You also try to convert it to sln format. this link might help you http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk On Sat, Sep 26, 2009 at 10:44 AM, hadi motamedi motamed...@gmail.comwrote: Dear All Can you please do me favor and let me know how can I convert *.wav files into 32 bit 44 KHz ? Please be informed that I have specific sound files in *.wav format that I converted them into *.gsm format with the aid of the following command : #sox FR3.wav FR3.gsm It got through but the voice quality is poor . I need to convert the original *.wav sound files (their file attribute is reported as WAVE audio mono 8000 Hz) into 32 bit 44 KHz for better voice quality . Can you please help me . Thank you in advance ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DAHDI disconnect supervision timing
On Fri, Sep 25, 2009 at 01:47:04PM -0400, Stephen Brown wrote: Sure thing, this is if I hang up before it hits voicemail: This does not include debug-level information . In the CLI, set: core set debug 5 Then in logger.conf make sure you have a log file that also gets verbose and debug. If you changed anyything, use 'logger reload' in the CLI to reload the settings. Now try an incoming call. -- 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] rtp.conf dtmftimeout
Brian Camp wrote: What unit is dtmftimeout measured in? In samples, 1/8000 of a second each, or 125 us if you prefer. The sample configuration is provided below. Does it mean... ; The amount of time a DTMF digit with no 'end' marker should be ; allowed to continue (in 'samples', 1/8000 of a second) ; ;dtmftimeout=3000 This means that the timeout is 3000/8000 of a second = 3/8 = 0.375 s. -kkm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Where are phone registrations kept?
Hi, I've built an Asterisk HA cluster by means of heartbeat and drbd. The following folders are stored on shared storage and referred to by means of symbolic links: /etc/asterisk /var/lib/asterisk /usr/lib/asterisk /var/spool/asterisk /var/log/asterisk I was under the impression that phone registrations were stored in /var/lib/asterisk/astdb and as such preserved when failing over. But when failing over I need to restart the phones in order to have them work with the newly actived asterisk node. This seems to point to the fact that phone registrations are stored elsewhere or are forgotten when Asterisk is restarted, but the latter seems not really true anyway. So, what is going wrong here? Were are the registrations stored? Or should I build in something to have the phones rebooted when I failover? Thank you, Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
If not, it will solve many other problems you would otherwise have. ABBAS SHAKEEL wrote: Thanks Alex By just avoiding this will solve this problem? On Sat, Sep 26, 2009 at 9:47 PM, Alex Balashov abalas...@evaristesys.com mailto:abalas...@evaristesys.com wrote: Don't put a SIP server behind destination NAT. Just don't. ABBAS SHAKEEL wrote: Sorry My Question was not very clear. Asterisk System that is placed some where on local LAN (suppose in office A) A sip(or any other whose softphone is available) phone Client that is out side this local network (suppose at office B). now if I want the asterisk server to be avaiable for this sip phone. As Asterisk Server is also behind NAT. SIP phone is also in any other network. How can I make them communicate. As in LAN i can easily by giving asterisk server IP. On Sat, Sep 26, 2009 at 7:57 PM, Philipp Kempgen philipp.kemp...@amooma.de mailto:philipp.kemp...@amooma.de mailto:philipp.kemp...@amooma.de mailto:philipp.kemp...@amooma.de wrote: Abbas Shakeel wrote: I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. Depends on what your goals are. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configuration at router level and all things like that. http://www.voip-info.org/wiki/view/NAT+and+VOIP I also heard that in Pakistan VOIP is not allowed. We need to buy a liscense that is very expensive and so on ... What exactly is your question? http://catb.org/~esr/faqs/smart-questions.html#explicit http://catb.org/~esr/faqs/smart-questions.html#homework http://catb.org/~esr/faqs/smart-questions.html#keepcool *SCNR* Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now:
Re: [asterisk-users] Where are phone registrations kept?
Contact bindings for AORs/registrations are stored in AstDB, but the state of a peer as being registered is stored in runtime memory. I agree that this is kind of silly. Bart Coninckx wrote: Hi, I've built an Asterisk HA cluster by means of heartbeat and drbd. The following folders are stored on shared storage and referred to by means of symbolic links: /etc/asterisk /var/lib/asterisk /usr/lib/asterisk /var/spool/asterisk /var/log/asterisk I was under the impression that phone registrations were stored in /var/lib/asterisk/astdb and as such preserved when failing over. But when failing over I need to restart the phones in order to have them work with the newly actived asterisk node. This seems to point to the fact that phone registrations are stored elsewhere or are forgotten when Asterisk is restarted, but the latter seems not really true anyway. So, what is going wrong here? Were are the registrations stored? Or should I build in something to have the phones rebooted when I failover? Thank you, Bart ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
On Sat, 2009-09-26 at 21:54 +0500, ABBAS SHAKEEL wrote: Thanks Alex By just avoiding this will solve this problem? No, Just moving the asterisk-server before the firewall won;t do any good. because in that situation the firewall is in between asterisk and your LOCAL sip-clients: you just solved one problem by introducing another eg, the just moved the problem to another place What you can do (perhaps not the best solution...) is having one asterisk server behind your firewall, serving all your local sip-clients. And another at the other side of the firewall, only for serving remote clients. And have both systems talking to each other with IAX instead of SIP In that case you _only_ have to allow port 4569 for IAX instead of 5060 and 1...2 for SIP hw ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
On 26/09/09 19:42, Hans Witvliet wrote: snip / What you can do (perhaps not the best solution...) is having one asterisk server behind your firewall, serving all your local sip-clients. And another at the other side of the firewall, only for serving remote clients. And have both systems talking to each other with IAX instead of SIP In that case you _only_ have to allow port 4569 for IAX instead of 5060 and 1...2 for SIP Hmmm, has anyone tried SIP over a VPN? We are thinking of testing this but haven't yet... Al ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New thread - SIP over VPN
On Sat, 26 Sep 2009, Alan Lord (News) wrote: Hmmm, has anyone tried SIP over a VPN? We are thinking of testing this but haven't yet... Al I have a client with Sonicwall VPNs. Asterisk is at head office on internal LAN, six external locations all have Linksys 2102 ATAs and Polycom IP501 phones registering and placing calls through the tunnels. It seems to work fine, but there is plenty of bandwidth between the offices, and they use G729. I think wrapping up the UDP stream into a TCP based tunnel might cause havoc if there is any packet loss or delay. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New thread - SIP over VPN
I use SIP over OpenVPN incessantly. Works great. Jeff LaCoursiere wrote: On Sat, 26 Sep 2009, Alan Lord (News) wrote: Hmmm, has anyone tried SIP over a VPN? We are thinking of testing this but haven't yet... Al I have a client with Sonicwall VPNs. Asterisk is at head office on internal LAN, six external locations all have Linksys 2102 ATAs and Polycom IP501 phones registering and placing calls through the tunnels. It seems to work fine, but there is plenty of bandwidth between the offices, and they use G729. I think wrapping up the UDP stream into a TCP based tunnel might cause havoc if there is any packet loss or delay. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New thread - SIP over VPN
Depending on the latency, wrapping the UDP stream into a TCP-based tunnel can be good -- if the VPN tunnel occasionally drops a packet, the tunnel will re-transmit the UDP packet. Of course, if the (one-way) latency is too high, the re-transmitted payload will arrive outside the jitter buffer and be dropped by the SIP CPE. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Saturday, September 26, 2009 2:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] New thread - SIP over VPN On Sat, 26 Sep 2009, Alan Lord (News) wrote: Hmmm, has anyone tried SIP over a VPN? We are thinking of testing this but haven't yet... Al I have a client with Sonicwall VPNs. Asterisk is at head office on internal LAN, six external locations all have Linksys 2102 ATAs and Polycom IP501 phones registering and placing calls through the tunnels. It seems to work fine, but there is plenty of bandwidth between the offices, and they use G729. I think wrapping up the UDP stream into a TCP based tunnel might cause havoc if there is any packet loss or delay. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New thread - SIP over VPN
On Sat, 2009-09-26 at 19:32 +, Jeff LaCoursiere wrote: On Sat, 26 Sep 2009, Alan Lord (News) wrote: Hmmm, has anyone tried SIP over a VPN? We are thinking of testing this but haven't yet... Al I have a client with Sonicwall VPNs. Asterisk is at head office on internal LAN, six external locations all have Linksys 2102 ATAs and Polycom IP501 phones registering and placing calls through the tunnels. It seems to work fine, but there is plenty of bandwidth between the offices, and they use G729. I think wrapping up the UDP stream into a TCP based tunnel might cause havoc if there is any packet loss or delay. snip We are using SIP over both IPSec and SSL VPNs very successfully with access controls in the tunnel ingress via the ISCS network security management project (http://iscs.sourceforge.net). There are a couple of issues. I'm not sure what you mean by a TCP tunnel unless you are referring to something like using OpenVPN over TCP rather than the default UDP. IPSec tunnels (which we use for LAN-to-LAN connections) are an IP level protocol and not TCP. OpenVPN (which we use for remote access) defaults to UDP port 1194 but can use any UDP or TCP socket. There has been some discussion that using it over TCP for VoIP can produce better results because the packets are less likely to be delivered out of order although perhaps with greater latency. All VPN processes will introduce additional latency. We have not found that to be a problem but several rounds of encryption / decryption over long distance connections in complex environments might introduce enough latency to be problematic. We have not found that yet. Depending on your VPN protocol implementation, there may or may not be an option to pass the ToS bits from the original packet into the IP header of the VPN packet. This is very important. Even though the Internet will not honor the ToS bits, you will want the gateways on both ends to do so, especially the one placing the packets onto your last mile. Since the VPN gateways cannot look inside the packet until it is decrypted, they have no way of distinguishing a large FTP packet from an RTP packet. Passing the ToS bits through may help. However, be careful. Most VoIP implementations seem to be setting DSCP bits instead of explicitly the ToS bits. DSCP uses the ToS bits but in a way different from the way ToS is set up to interpret them. If I remember correctly, setting DSCP to Expedited Forwarding sets the bits which coincide with ToS in such a way that Linux based gateways will place the packets into the band 1 which is the default processing band and not band 0 which is the high priority band. For example, on Asterisk, we did not set our RTP QoS to b8 but rather to b0 (if I recall correctly). We have one case using OpenVPN where the sound quality is occasionally problematic. In our case it's a little easy. The remote desktops are based upon our soon to be released SimplicITy model (http://www.ssiservices.biz) and accessed via NX or X2Go technology. Usually, the only traffic passing through the OpenVPN tunnel is the VoIP traffic. We have thus changed the gateway itself to treat all UDP packets on port 1194 as high priority. We'll see if that makes the problem go away. Hope this helps - John -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New thread - SIP over VPN
On Sat, 26 Sep 2009, John A. Sullivan III wrote: snip We are using SIP over both IPSec and SSL VPNs very successfully with access controls in the tunnel ingress via the ISCS network security management project (http://iscs.sourceforge.net). There are a couple of issues. I'm not sure what you mean by a TCP tunnel unless you are referring to something like using OpenVPN over TCP rather than the default UDP. Isn't an SSL based tunnel all TCP? [snip] to UDP port 1194 but can use any UDP or TCP socket. There has been some discussion that using it over TCP for VoIP can produce better results because the packets are less likely to be delivered out of order although perhaps with greater latency. The resends would have to happen within the jitter buffer period, as someone else pointed out, or I would think large chunks would be missing in the audio (the missing packet plus all the ones queued up after it that missed the jitter window). Total speculation on my part. [snipped excellent tips on ToS!] Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New thread - SIP over VPN
On Sat, 2009-09-26 at 22:09 +, Jeff LaCoursiere wrote: On Sat, 26 Sep 2009, John A. Sullivan III wrote: snip We are using SIP over both IPSec and SSL VPNs very successfully with access controls in the tunnel ingress via the ISCS network security management project (http://iscs.sourceforge.net). There are a couple of issues. I'm not sure what you mean by a TCP tunnel unless you are referring to something like using OpenVPN over TCP rather than the default UDP. Isn't an SSL based tunnel all TCP? Not in the case of OpenVPN. I'm not sure about the commercial offerings. That could very well be the case as I believe most of them developed out of the web proxy model. I was probably trapped by my own context! Thanks - John snip -- John A. Sullivan III Open Source Development Corporation +1 207-985-7880 jsulli...@opensourcedevel.com http://www.spiritualoutreach.com Making Christianity intelligible to secular society ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
On Sat, 2009-09-26 at 20:07 +0100, Alan Lord (News) wrote: On 26/09/09 19:42, Hans Witvliet wrote: snip / What you can do (perhaps not the best solution...) is having one asterisk server behind your firewall, serving all your local sip-clients. And another at the other side of the firewall, only for serving remote clients. And have both systems talking to each other with IAX instead of SIP In that case you _only_ have to allow port 4569 for IAX instead of 5060 and 1...2 for SIP Hmmm, has anyone tried SIP over a VPN? We are thinking of testing this but haven't yet... yes, even via satelite link. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New thread - SIP over VPN
On Sat, 2009-09-26 at 19:32 +, Jeff LaCoursiere wrote: On Sat, 26 Sep 2009, Alan Lord (News) wrote: Hmmm, has anyone tried SIP over a VPN? We are thinking of testing this but haven't yet... Al I have a client with Sonicwall VPNs. Asterisk is at head office on internal LAN, six external locations all have Linksys 2102 ATAs and Polycom IP501 phones registering and placing calls through the tunnels. It seems to work fine, but there is plenty of bandwidth between the offices, and they use G729. I think wrapping up the UDP stream into a TCP based tunnel might cause havoc if there is any packet loss or delay. Packet loss shouldn't be your major concern, delay, (latency) is a real pita ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New thread - SIP over VPN
On Sat, 2009-09-26 at 22:09 +, Jeff LaCoursiere wrote: On Sat, 26 Sep 2009, John A. Sullivan III wrote: snip We are using SIP over both IPSec and SSL VPNs very successfully with access controls in the tunnel ingress via the ISCS network security management project (http://iscs.sourceforge.net). There are a couple of issues. I'm not sure what you mean by a TCP tunnel unless you are referring to something like using OpenVPN over TCP rather than the default UDP. Isn't an SSL based tunnel all TCP? No, could be either. [snip] to UDP port 1194 but can use any UDP or TCP socket. There has been some discussion that using it over TCP for VoIP can produce better results because the packets are less likely to be delivered out of order although perhaps with greater latency. The resends would have to happen within the jitter buffer period, as someone else pointed out, or I would think large chunks would be missing in the audio (the missing packet plus all the ones queued up after it that missed the jitter window). Total speculation on my part. Re-sending audio packets is waste of resource, Better to have an audio-stream with an occasional missing packet, than the delay of waiting for re-transmission and re-ordering. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Know for how long an agent is talking?
Hi, Is there a way to know for how long an agent is talking on the queue call? (without keeping a timer myself... just asking asterisk) thanks, Gabriel ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] digium fax: failed to queue document
In my quest to actually send a fax, I'm now stuck trying to send the
confirm.
First I send the fax:
-- Executing [s...@outbound-fax:2] System(Console/dsp, env echo
-e Channel:DAHDI/g0/12036378447\\nContext:fax-tx\\nExtension:
s\\nPriority: 1\\n /var/spool/asterisk/outgoing/call-1254012878.0) in
new stack
-- Auto fallthrough, channel 'Console/dsp' status is 'UNKNOWN'
Hangup on console
-- Attempting call on DAHDI/g0/12036378447 for s...@fax-tx:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
Channel DAHDI/1-1 was answered.
-- Executing [...@fax-tx:1] SendFAX(DAHDI/1-1,
/var/spool/asterisk/fax/20090922_1301.tif) in new stack
-- Channel 'DAHDI/1-1' sending fax
'/var/spool/asterisk/fax/20090922_1301.tif'
-- Channel 'DAHDI/1-1' fax session '0' started
.
And that works.
Then I try to send the confirm:
'h' =1. Set(RID=${FAXOPT(remotestationid)})
[pbx_config]
2.
Set(DateTime=${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
[pbx_config]
3.
Set(GLOBAL(StatusFile)=/var/spool/asterisk/fax-tx-status-${DateTime})
[pbx_config]
4. System(env echo -e ${FAXOPT(pages)} Page Fax
sent to ${EXTEN}. Remote ID: ${RID} ${StatusFile}-l1) [pbx_config]
5. System(env echo -e Status: ${FAXOPT(status)}
${FAXOPT(statusstr)} ${StatusFile}-l2) [pbx_config]
6. System(convert -background white -fill black
-pointsize 18 text:${StatusFile}-l1 text:${StatusFile}-l2 -crop
600x100+1+1 -append ${StatusFile}.tif) [pbx_config]
7. Set(GLOBAL(StatusFile)=${StatusFile})
[pbx_config]
8.
Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config]
9. System(env echo -e
Channel:DAHDI/g0/abbbccc\\nContext:fax-confirm-tx\\nExtension:
s\\nPriority: 1\\n ${UniqueFile}) [pbx_config]
But that fails:
-- Executing [...@fax-tx:1] Set(DAHDI/1-1, RID=bbb-ccc-) in
new stack
-- Executing [...@fax-tx:2] Set(DAHDI/1-1,
DateTime=20090926_2055) in new stack
-- Executing [...@fax-tx:3] Set(DAHDI/1-1,
GLOBAL(StatusFile)=/var/spool/asterisk/fax-tx-status-20090926_2055) in
new stack
== Setting global variable 'StatusFile' to
'/var/spool/asterisk/fax-tx-status-20090926_2055'
-- Executing [...@fax-tx:4] System(DAHDI/1-1, env echo -e 1 Page
Fax sent to h. Remote ID: bbb-ccc-
/var/spool/asterisk/fax-tx-status-20090926_2055-l1) in new stack
-- Executing [...@fax-tx:5] System(DAHDI/1-1, env echo -e Status:
SUCCESS FAX_SUCCESS
/var/spool/asterisk/fax-tx-status-20090926_2055-l2) in new stack
-- Executing [...@fax-tx:6] System(DAHDI/1-1, convert -background
white -fill black -pointsize 18
text:/var/spool/asterisk/fax-tx-status-20090926_2055-l1
text:/var/spool/asterisk/fax-tx-status-20090926_2055-l2 -crop
600x100+1+1 -append
/var/spool/asterisk/fax-tx-status-20090926_2055.tif) in new stack
-- Executing [...@fax-tx:7] Set(DAHDI/1-1,
GLOBAL(StatusFile)=/var/spool/asterisk/fax-tx-status-20090926_2055) in
new stack
== Setting global variable 'StatusFile' to
'/var/spool/asterisk/fax-tx-status-20090926_2055'
-- Executing [...@fax-tx:8] Set(DAHDI/1-1,
UniqueFile=/var/spool/asterisk/outgoing/call-1254012879.1) in new stack
-- Executing [...@fax-tx:9] System(DAHDI/1-1, env echo -e
Channel:DAHDI/g0/abbbccc\\nContext:fax-confirm-tx\\nExtension:
s\\nPriority: 1\\n /var/spool/asterisk/outgoing/call-1254012879.1) in
new stack
-- Attempting call on DAHDI/g0/abbbccc for s...@fax-confirm-tx:1
(Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Hungup 'DAHDI/1-1'
[Sep 26 20:55:17] NOTICE[14894]: pbx_spool.c:357 attempt_thread: Call
completed to DAHDI/g0/abbbccc
Channel DAHDI/2-1 was answered.
-- Executing [...@fax-confirm-tx:1] SendFAX(DAHDI/2-1,
/var/spool/asterisk/fax-tx-status-20090926_2055.tif) in new stack
-- Channel 'DAHDI/2-1' sending fax
'/var/spool/asterisk/fax-tx-status-20090926_2055.tif'
[Sep 26 20:55:26] ERROR[14912]: res_fax_digium.c:1761 dgm_fax_start: fax
handle: 0 failed to queue document
'/var/spool/asterisk/fax-tx-status-20090926_2055.tif'
[Sep 26 20:55:26] ERROR[14912]: res_fax.c:811 generic_fax_exec: channel
'DAHDI/2-1' fax session '1' failure, reason: 'failed to start fax session'
The file does exist:
file /var/spool/asterisk/fax-tx-status-20090926_2055.tif
/var/spool/asterisk/fax-tx-status-20090926_2055.tif: TIFF image data,
little-endian
ls -l /var/spool/asterisk/fax-tx-status-20090926_2055.tif
-rw-r--r-- 1 root root 480892 2009-09-26 20:55
/var/spool/asterisk/fax-tx-status-20090926_2055.tif
Any help appreciated.
sean
___
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Register Now: http://www.astricon.net
Re: [asterisk-users] New thread - SIP over VPN
Last week I did a Microsoft VPM from one XP computer to another via Verizon broadband wireless. SIP worked ok, but BLF on a Grand Stream 2010 didn't work. In addition to the VPN the phone was behind a NAT router. The phone was already set up behind the NAT Router, the only difference was to get the connectivity via Wireless VPN. There could have been some missing ports in the VPN environment. The audio was good, but there were times it lost clarity, likely to wireless bandwidth/lag/jitter issues. I decided that couldn't be my main business phone. Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New thread - SIP over VPN
Resending is not a waste if the re-transmitted packet can arrive within the jitter buffer window. Practically speaking, though, since UDP packets are generally not retransmitted (unless it's within some kind of TCP-based tunnel), it's a moot point. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Saturday, September 26, 2009 6:06 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] New thread - SIP over VPN snip Re-sending audio packets is waste of resource, Better to have an audio-stream with an occasional missing packet, than the delay of waiting for re-transmission and re-ordering. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New thread - SIP over VPN
As with many applications using UDP as a transport, most UDP-based application-layer VPN schemes (such as OpenVPN) do have some sort of rudimentary backward acknowledgment and reliability layers implemented on top of UDP. They're just a lot more lightweight, primitive, and generally much faster and less exacting than TCP. Frank Bulk wrote: Resending is not a waste if the re-transmitted packet can arrive within the jitter buffer window. Practically speaking, though, since UDP packets are generally not retransmitted (unless it's within some kind of TCP-based tunnel), it's a moot point. Frank -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet Sent: Saturday, September 26, 2009 6:06 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] New thread - SIP over VPN snip Re-sending audio packets is waste of resource, Better to have an audio-stream with an occasional missing packet, than the delay of waiting for re-transmission and re-ordering. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] digium fax: failed to queue document
u don't change the ${uniquefile} for the second System/Originate
try to add a string to the ${uniquefile} ...
eg
${uniquefile}0
Martin
On Sat, Sep 26, 2009 at 8:05 PM, sean darcy seandar...@gmail.com wrote:
In my quest to actually send a fax, I'm now stuck trying to send the
confirm.
First I send the fax:
-- Executing [s...@outbound-fax:2] System(Console/dsp, env echo
-e Channel:DAHDI/g0/12036378447\\nContext:fax-tx\\nExtension:
s\\nPriority: 1\\n /var/spool/asterisk/outgoing/call-1254012878.0) in
new stack
-- Auto fallthrough, channel 'Console/dsp' status is 'UNKNOWN'
Hangup on console
-- Attempting call on DAHDI/g0/12036378447 for s...@fax-tx:1 (Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
Channel DAHDI/1-1 was answered.
-- Executing [...@fax-tx:1] SendFAX(DAHDI/1-1,
/var/spool/asterisk/fax/20090922_1301.tif) in new stack
-- Channel 'DAHDI/1-1' sending fax
'/var/spool/asterisk/fax/20090922_1301.tif'
-- Channel 'DAHDI/1-1' fax session '0' started
.
And that works.
Then I try to send the confirm:
'h' = 1. Set(RID=${FAXOPT(remotestationid)})
[pbx_config]
2.
Set(DateTime=${STRFTIME(${EPOCH},,%Y%m%d)}_${STRFTIME(${EPOCH},,%H%M)})
[pbx_config]
3.
Set(GLOBAL(StatusFile)=/var/spool/asterisk/fax-tx-status-${DateTime})
[pbx_config]
4. System(env echo -e ${FAXOPT(pages)} Page Fax
sent to ${EXTEN}. Remote ID: ${RID} ${StatusFile}-l1) [pbx_config]
5. System(env echo -e Status: ${FAXOPT(status)}
${FAXOPT(statusstr)} ${StatusFile}-l2) [pbx_config]
6. System(convert -background white -fill black
-pointsize 18 text:${StatusFile}-l1 text:${StatusFile}-l2 -crop
600x100+1+1 -append ${StatusFile}.tif) [pbx_config]
7. Set(GLOBAL(StatusFile)=${StatusFile})
[pbx_config]
8.
Set(UniqueFile=/var/spool/asterisk/outgoing/call-${UNIQUEID}) [pbx_config]
9. System(env echo -e
Channel:DAHDI/g0/abbbccc\\nContext:fax-confirm-tx\\nExtension:
s\\nPriority: 1\\n ${UniqueFile}) [pbx_config]
But that fails:
-- Executing [...@fax-tx:1] Set(DAHDI/1-1, RID=bbb-ccc-) in
new stack
-- Executing [...@fax-tx:2] Set(DAHDI/1-1,
DateTime=20090926_2055) in new stack
-- Executing [...@fax-tx:3] Set(DAHDI/1-1,
GLOBAL(StatusFile)=/var/spool/asterisk/fax-tx-status-20090926_2055) in
new stack
== Setting global variable 'StatusFile' to
'/var/spool/asterisk/fax-tx-status-20090926_2055'
-- Executing [...@fax-tx:4] System(DAHDI/1-1, env echo -e 1 Page
Fax sent to h. Remote ID: bbb-ccc-
/var/spool/asterisk/fax-tx-status-20090926_2055-l1) in new stack
-- Executing [...@fax-tx:5] System(DAHDI/1-1, env echo -e Status:
SUCCESS FAX_SUCCESS
/var/spool/asterisk/fax-tx-status-20090926_2055-l2) in new stack
-- Executing [...@fax-tx:6] System(DAHDI/1-1, convert -background
white -fill black -pointsize 18
text:/var/spool/asterisk/fax-tx-status-20090926_2055-l1
text:/var/spool/asterisk/fax-tx-status-20090926_2055-l2 -crop
600x100+1+1 -append
/var/spool/asterisk/fax-tx-status-20090926_2055.tif) in new stack
-- Executing [...@fax-tx:7] Set(DAHDI/1-1,
GLOBAL(StatusFile)=/var/spool/asterisk/fax-tx-status-20090926_2055) in
new stack
== Setting global variable 'StatusFile' to
'/var/spool/asterisk/fax-tx-status-20090926_2055'
-- Executing [...@fax-tx:8] Set(DAHDI/1-1,
UniqueFile=/var/spool/asterisk/outgoing/call-1254012879.1) in new stack
-- Executing [...@fax-tx:9] System(DAHDI/1-1, env echo -e
Channel:DAHDI/g0/abbbccc\\nContext:fax-confirm-tx\\nExtension:
s\\nPriority: 1\\n /var/spool/asterisk/outgoing/call-1254012879.1) in
new stack
-- Attempting call on DAHDI/g0/abbbccc for s...@fax-confirm-tx:1
(Retry 1)
-- Requested transfer capability: 0x00 - SPEECH
-- Hungup 'DAHDI/1-1'
[Sep 26 20:55:17] NOTICE[14894]: pbx_spool.c:357 attempt_thread: Call
completed to DAHDI/g0/abbbccc
Channel DAHDI/2-1 was answered.
-- Executing [...@fax-confirm-tx:1] SendFAX(DAHDI/2-1,
/var/spool/asterisk/fax-tx-status-20090926_2055.tif) in new stack
-- Channel 'DAHDI/2-1' sending fax
'/var/spool/asterisk/fax-tx-status-20090926_2055.tif'
[Sep 26 20:55:26] ERROR[14912]: res_fax_digium.c:1761 dgm_fax_start: fax
handle: 0 failed to queue document
'/var/spool/asterisk/fax-tx-status-20090926_2055.tif'
[Sep 26 20:55:26] ERROR[14912]: res_fax.c:811 generic_fax_exec: channel
'DAHDI/2-1' fax session '1' failure, reason: 'failed to start fax session'
The file does exist:
file /var/spool/asterisk/fax-tx-status-20090926_2055.tif
/var/spool/asterisk/fax-tx-status-20090926_2055.tif: TIFF image data,
little-endian
ls -l /var/spool/asterisk/fax-tx-status-20090926_2055.tif
-rw-r--r-- 1 root root 480892 2009-09-26 20:55
/var/spool/asterisk/fax-tx-status-20090926_2055.tif
Any help
Re: [asterisk-users] VOIP solutions
On Sat, Sep 26, 2009 at 12:47 PM, Alex Balashov abalas...@evaristesys.com wrote: Don't put a SIP server behind destination NAT. Just don't. Why not? Mind to explain? ABBAS SHAKEEL wrote: Sorry My Question was not very clear. Asterisk System that is placed some where on local LAN (suppose in office A) A sip(or any other whose softphone is available) phone Client that is out side this local network (suppose at office B). now if I want the asterisk server to be avaiable for this sip phone. As Asterisk Server is also behind NAT. SIP phone is also in any other network. How can I make them communicate. As in LAN i can easily by giving asterisk server IP. On Sat, Sep 26, 2009 at 7:57 PM, Philipp Kempgen philipp.kemp...@amooma.de mailto:philipp.kemp...@amooma.de wrote: Abbas Shakeel wrote: I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. Depends on what your goals are. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configuration at router level and all things like that. http://www.voip-info.org/wiki/view/NAT+and+VOIP I also heard that in Pakistan VOIP is not allowed. We need to buy a liscense that is very expensive and so on ... What exactly is your question? http://catb.org/~esr/faqs/smart-questions.html#explicit http://catb.org/~esr/faqs/smart-questions.html#homework http://catb.org/~esr/faqs/smart-questions.html#keepcool *SCNR* Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VOIP solutions
Thanks Alex and Hans. The Discussion was Really helpful. Now a days i am using VPN for far SIP clients. But I will try what Hans Suggested and let you know if i found any problem. On Sun, Sep 27, 2009 at 9:34 AM, C F shma...@gmail.com wrote: On Sat, Sep 26, 2009 at 12:47 PM, Alex Balashov abalas...@evaristesys.com wrote: Don't put a SIP server behind destination NAT. Just don't. Why not? Mind to explain? ABBAS SHAKEEL wrote: Sorry My Question was not very clear. Asterisk System that is placed some where on local LAN (suppose in office A) A sip(or any other whose softphone is available) phone Client that is out side this local network (suppose at office B). now if I want the asterisk server to be avaiable for this sip phone. As Asterisk Server is also behind NAT. SIP phone is also in any other network. How can I make them communicate. As in LAN i can easily by giving asterisk server IP. On Sat, Sep 26, 2009 at 7:57 PM, Philipp Kempgen philipp.kemp...@amooma.de mailto:philipp.kemp...@amooma.de wrote: Abbas Shakeel wrote: I Recently completed an IVR application with Asterisk. Now we are moving towards VOIP. Please give a direction how to move forward. Depends on what your goals are. What i have studied so far I am confused with NAT issues. As i can have many SIP peers on local LAN it works but from internet it donts. We need to do configuration at router level and all things like that. http://www.voip-info.org/wiki/view/NAT+and+VOIP I also heard that in Pakistan VOIP is not allowed. We need to buy a liscense that is very expensive and so on ... What exactly is your question? http://catb.org/~esr/faqs/smart-questions.html#explicit http://catb.org/~esr/faqs/smart-questions.html#homework http://catb.org/~esr/faqs/smart-questions.html#keepcool *SCNR* Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards Shakeel Abbas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] New thread - SIP over VPN
Isn't an SSL based tunnel all TCP? Not in the case of OpenVPN. I'm not sure about the commercial offerings. Correct. My recollection is that OpenSSL uses TCP for the setup and management of the tunnel (e.g. authentication and key exchange) and uses UDP to carry the actual payload... each tunneled IP packet is wrapped in a UDP datagram. That way, the UDP transport mimics the basic characteristics of normal IP transport - it's best-efforts, order not guaranteed, and no built-in retries. The number of lost (or out-of-order) packets in such a tunneled connection shouldn't be significantly different than what you'd see if you weren't using a tunnel at all. There seems to be a good deal of feeling (and evidence) that trying to use TCP as the container for a tunnel is likely to cause more trouble than it solves. Yes, the TCP layer will make the tunnel reliable - but at the expense of adding unpredictable amounts of latency, due to TCP's built-in exponential-backoff retry timing. Things get *really* nasty if you try to wrap one TCP connection in another, because both connections will be independently retrying any lost or delayed packets - you'll end up retransmitting quite a bit more data than you would if you simply used TCP/IP (or TCP/IP wrapped in UDP/IP) and throughput will suffer. I've been using an OpenSSL tunnel to connect my Nokia N810 internet tablet to my Asterisk server, for about a year now. It works very nicely, eliminating NAT- related problems (no need to STUN) and allowing me to use VoIP from most WiFi networks I can log onto. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
