Re: [asterisk-users] Signaling incoming call
On Sun, 31 May 2015, Luca Bertoncello wrote: register = 004935:MYSECRET@pbxluca/004935 register = 0049351222:MYSECRET@pbxfax/0049351222 register = 0049351333:MYSECRET@pbxanika/0049351333 register = 44:MYVERYSECRET@messagenet/44 [pbxluca] type=peer defaultuser=004935 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=004935 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [pbxfax] type=peer defaultuser=0049351222 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=fax_incoming outboundproxy=172.16.34.132 port=5060 fromuser=0049351222 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [snip] Just a few 'stylistic' and 'ease of maintenance' suggestions: 1) Group the 'register' with the stanza. 2) Add a bit of whitespace to increase 'readability.' 3) Sort the parameters to make it easier to maintain. 4) Move 'common' settings to 'general.' Thus, I'd write your sip.conf as: [general] canreinvite = no context = luca_incoming dtmfmode= rfc2833 insecure= invite outboundproxy = 172.16.34.132 port= 5060 qualify = yes qualifyfreq = 600 usereqphone = yes [general](+) register= 004935:MYSECRET@pbxluca/004935 [pbxluca] defaultuser = 004935 fromdomain = 172.16.34.132 fromuser= 004935 host= 172.16.34.132 secret = MYSECRET type= peer [general](+) register= 0049351222:MYSECRET@pbxfax/0049351222 [pbxfax] defaultuser = 0049351222 fromdomain = 172.16.34.132 fromuser= 0049351222 host= 172.16.34.132 secret = MYSECRET type= peer 5) I'd try and move more of the common settings to general, but these were the ones listed on voip-info.org. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use TRUNK only if IAX fails?
I would especially look at the CHANUNAVAIL dial status Since it sounds like you are probably qualifying your IAX trunk, that status will be the quickest way to overflow from IAX to TDM. On Sat, May 30, 2015, 11:35 PM Ashwin Surendran ashwin.surend...@now-health.com wrote: Hi Matt, I was a bit concerned on the delay if there might be any when my iax link is down? It would be two dial steps right when my iax link is down. But I’m more than happy to try. Many Thanks, Ashwin. *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Matt Riddell (lists) *Sent:* 30 May 2015 16:55 *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] How to use TRUNK only if IAX fails? The command he gave you was in Asterisk. Why do you not want to call it to try it? Then you can fail over to the other trunk if the IAX link is down. Kind regards, Matt On May 30, 2015, at 2:03 AM, Ashwin Surendran ashwin.surend...@now-health.com wrote: Many Thanks Carlos, I was hoping to check whether the remote server is available before I issue the dial in my dial plan. Is there a better way to do it in asterisk without using unix commands? Many Thanks, Ashwin On 5/30/15, 2:06 AM, Carlos Chavez cur...@telecomabmex.com wrote: On 5/29/15 1:16 PM, Ashwin Surendran wrote: Hi, I have multiple Asterisk servers in various parts of the world all connected using dedicated VPN¹s. Each of these servers have iax and dahdi TRUNK configured on them. Occasionally the VPN¹s fail. What I want to be able to do is on my dial plan, use IAX if the asterisk server can reach the remote server using the internet OR, use TRUNK only if it can¹t use IAX. Any ideas on how this can be implemented on the dial plan? Check the DIALSTATUS variable to see if the IAX trunk failed and then dial via DAHDI. https://wiki.asterisk.org/wiki/display/AST/Dial+Channel+Variables -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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 This email (and any attachments or hyperlinks within it) may contain information that is confidential, legally privileged or otherwise protected from disclosure. If you are not the intended recipient of this email, you are not entitled to use, disclose, distribute, copy, print, disseminate or rely on this email in any way. If you have received this email in error, please notify the sender immediately by telephone or email and destroy it, and all copies of it. Thank you for your cooperation. -- _ -- 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 This email (and any attachments or hyperlinks within it) may contain information that is confidential, legally privileged or otherwise protected from disclosure. If you are not the intended recipient of this email, you are not entitled to use, disclose, distribute, copy, print, disseminate or rely on this email in any way. If you have received this email in error, please notify the sender immediately by telephone or email and destroy it, and all copies of it. Thank you for your cooperation. -- _ -- 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] Signaling incoming call
On Sun, 31 May 2015, Luca Bertoncello wrote: Now, it would be nice, if I can signaling on the phone which number will be called, so that, for example, if I receive a call for +4935 I get a message on the display or the phone ring with a particular tone, and if I receive a call for +49351222 the phone write something other on the display or ring with another tone. Is it possible? Maybe it depends from phone... I use a Thomson ST2022. You can fiddle with the caller ID to change what is displayed on the phone. You can fiddle with the ring tone by phone specific configuration and phone specific SIP headers (sipaddheader(Alert-Info: ...)). These seem relevant: http://www.voip-info.org/wiki/view/RTTTL+melodies+for+ST2030 (the discussion looks relevant as well). http://www.asteriskguru.com/tutorials/thomson_st2030.html http://www.freepbx.org/support/documentation/howtos/how-to-enable-distinctive-ringing-alert-info-for-calls-from-particular- -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to use TRUNK only if IAX fails?
Many Thanks Trey! That’s what I need. -Ashwin. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Trey Hilyard Sent: 01 June 2015 06:28 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to use TRUNK only if IAX fails? I would especially look at the CHANUNAVAIL dial status Since it sounds like you are probably qualifying your IAX trunk, that status will be the quickest way to overflow from IAX to TDM. On Sat, May 30, 2015, 11:35 PM Ashwin Surendran ashwin.surend...@now-health.commailto:ashwin.surend...@now-health.com wrote: Hi Matt, I was a bit concerned on the delay if there might be any when my iax link is down? It would be two dial steps right when my iax link is down. But I’m more than happy to try. Many Thanks, Ashwin. From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell (lists) Sent: 30 May 2015 16:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to use TRUNK only if IAX fails? The command he gave you was in Asterisk. Why do you not want to call it to try it? Then you can fail over to the other trunk if the IAX link is down. Kind regards, Matt On May 30, 2015, at 2:03 AM, Ashwin Surendran ashwin.surend...@now-health.commailto:ashwin.surend...@now-health.com wrote: Many Thanks Carlos, I was hoping to check whether the remote server is available before I issue the dial in my dial plan. Is there a better way to do it in asterisk without using unix commands? Many Thanks, Ashwin On 5/30/15, 2:06 AM, Carlos Chavez cur...@telecomabmex.commailto:cur...@telecomabmex.com wrote: On 5/29/15 1:16 PM, Ashwin Surendran wrote: Hi, I have multiple Asterisk servers in various parts of the world all connected using dedicated VPN¹s. Each of these servers have iax and dahdi TRUNK configured on them. Occasionally the VPN¹s fail. What I want to be able to do is on my dial plan, use IAX if the asterisk server can reach the remote server using the internet OR, use TRUNK only if it can¹t use IAX. Any ideas on how this can be implemented on the dial plan? Check the DIALSTATUS variable to see if the IAX trunk failed and then dial via DAHDI. https://wiki.asterisk.org/wiki/display/AST/Dial+Channel+Variables -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez +52 (55)9116-91161 -- _ -- 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 This email (and any attachments or hyperlinks within it) may contain information that is confidential, legally privileged or otherwise protected from disclosure. If you are not the intended recipient of this email, you are not entitled to use, disclose, distribute, copy, print, disseminate or rely on this email in any way. If you have received this email in error, please notify the sender immediately by telephone or email and destroy it, and all copies of it. Thank you for your cooperation. -- _ -- 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 This email (and any attachments or hyperlinks within it) may contain information that is confidential, legally privileged or otherwise protected from disclosure. If you are not the intended recipient of this email, you are not entitled to use, disclose, distribute, copy, print, disseminate or rely on this email in any way. If you have received this email in error, please notify the sender immediately by telephone or email and destroy it, and all copies of it. Thank you for your cooperation. -- _ -- 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 This email (and any attachments or hyperlinks within it) may contain information that is confidential, legally privileged or otherwise protected from disclosure. If you are not the intended recipient of this email, you are not entitled to use, disclose, distribute, copy, print,
Re: [asterisk-users] Signaling incoming call
Steve Edwards asterisk@sedwards.com schrieb: You can fiddle with the caller ID to change what is displayed on the phone. You can fiddle with the ring tone by phone specific configuration and phone specific SIP headers (sipaddheader(Alert-Info: ...)). These seem relevant: http://www.voip-info.org/wiki/view/RTTTL+melodies+for+ST2030 (the discussion looks relevant as well). http://www.asteriskguru.com/tutorials/thomson_st2030.html http://www.freepbx.org/support/documentation/howtos/how-to-enable-distinctive-ringing-alert-info-for-calls-from-particular- Thank you very much! I'll try it and report to the list. Regards Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Signaling incoming call
Hi list! Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with fooling... My phone will receive calls from 3 numbers. All that was done in my dialplan. Now, it would be nice, if I can signaling on the phone which number will be called, so that, for example, if I receive a call for +4935 I get a message on the display or the phone ring with a particular tone, and if I receive a call for +49351222 the phone write something other on the display or ring with another tone. Is it possible? Maybe it depends from phone... I use a Thomson ST2022. Thanks a lot Luca Bertoncello (lucab...@lucabert.de) -- _ -- 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] Signaling incoming call
Finally I got my Asterisk works with my two phones... It was a problem on my Firewall (for the phone of my wife) and on my Dialplan (for forwarding calls). Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Well, now I have some time to spend with fooling... My phone will receive calls from 3 numbers. All that was done in my dialplan. Now, it would be nice, if I can signaling on the phone which number will be called, so that, for example, if I receive a call for +4935 I get a message on the display or the phone ring with a particular tone, and if I receive a call for +49351222 the phone write something other on the display or ring with another tone. Is it possible? Maybe it depends from phone... I use a Thomson ST2022. I don't know your phones, but there are multiple ways to achieve that. By far the easiest method is to work with multiple SIP identities. You can adjust quite a few parameters, like display text, ring tone, timings, forwarding While you are busy with this, you can add additional accounts that operate as intercoms (baby monitors) so you don't have to wait for an answer. Interesting exercise, but might disturb peace in the house. If your phone supports only a single identity, then you have to adjust caller ids, etc with Asterisk. jg -- _ -- 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] Signaling incoming call
-BEGIN PGP SIGNED MESSAGE- Hash: SHA256 On 05/31/2015 02:31 PM, Luca Bertoncello wrote: Hi list! Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Don't worry, Asterisk works very well with Deutsche Telekom and there new ip-based connections ... - -- DavaoSOFT, the home of ERPel ERPel, das deutsche Warenwirtschaftssystem fuer LINUX http://www.davaosoft.com -BEGIN PGP SIGNATURE- Version: GnuPG v2 iQEcBAEBCAAGBQJVasmmAAoJEG6MZewV4LQAPf4H/iN9V52VMz1OMb6QI4WHC125 G1erQWVdUssq65/+/2tnQ7kQZwt56HhdlbieBZcLK0BpO0xnSVFZ2AGd0nzbWn9h 4NR8E6PRYfODXh/vs5INoNEhBL8s/EpnzfyseBpLCqhxJGFgtJ7FREJiRd1Y4J+j Y+vao2lLf+d0yOw8S9Z3Zklctx9+X2H0fzfkPYiBqXvj229KQHRquexi7PfaBI5Q B7Ol5GfkdZy7rSctZOtTCyOwwEcLh08931EQFN5SC8dpDnZFj5vfRSVMzRt3w3vm x44rekn8Qc2G46Y9cPWmtMTTVnQi5sAL/elnEP8qb1B6//0dPCJ88cXpFrhI+CQ= =XV7F -END PGP SIGNATURE- -- _ -- 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] Signaling incoming call
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Guenther Boelter gboel...@gmail.com schrieb: -BEGIN PGP SIGNED MESSAGE- Hash: SHA256 On 05/31/2015 02:31 PM, Luca Bertoncello wrote: Hi list! Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Don't worry, Asterisk works very well with Deutsche Telekom and there new ip-based connections ... That's a good news... Currenty I configured my sip.conf so: register = 004935:MYSECRET@pbxluca/004935 register = 0049351222:MYSECRET@pbxfax/0049351222 register = 0049351333:MYSECRET@pbxanika/0049351333 register = 44:MYVERYSECRET@messagenet/44 [pbxluca] type=peer defaultuser=004935 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=004935 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [pbxfax] type=peer defaultuser=0049351222 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=fax_incoming outboundproxy=172.16.34.132 port=5060 fromuser=0049351222 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [pbxanika] type=peer defaultuser=0049351333 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=anika_incoming outboundproxy=172.16.34.132 port=5060 fromuser=0049351333 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [messagenet] type=peer defaultuser=44 secret=MYVERYSECRET dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.it port=5061 fromuser=44 fromdomain=sip.messagenet.it usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=60 Am I right if I say, that I just have to change defaultuser, host, secret, outboundproxy and fromdomain with the data from Telekom and it works? I thinks, it should be: defaultuser=0049351333 secret= MYSECRET host=tel.t-online.de context=anika_incoming outboundproxy=tel.t-online.de port=5060 fromuser=0049351333 fromdomain=tel.t-online.de I'm not sure, where I should write my Login (from my DSL-Line)... I see this page (in German): http://hilfe.telekom.de/hsp/cms/content/HSP/de/3378/FAQ/theme-133631783/Auftrag/theme-82239611/IP-basierter-Anschluss/faq-350884716;jsessionid=A18F587E00F25C8FC26ACF3685481D72 Could you please help me? Thanks Luca Bertoncello (lucab...@lucabert.de) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.10 (GNU/Linux) iEYEARECAAYFAlVqzVAACgkQ8Ggznj+1EDifYwCgiQTeZQsUljAP5CNpteeFW5aV ugMAn0BnmlGJRHqBJA19DXPgqv0ZUqq1 =vt3E -END PGP SIGNATURE- -- _ -- 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] Signaling incoming call
Am 31. Mai 2015 10:58:56 MESZ, schrieb Luca Bertoncello lucab...@lucabert.de: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Guenther Boelter gboel...@gmail.com schrieb: -BEGIN PGP SIGNED MESSAGE- Hash: SHA256 On 05/31/2015 02:31 PM, Luca Bertoncello wrote: Hi list! Now all works as expected, at least in the simulation I did with AsteriskNOW. Hopefully it will work later, when Deutsche Telekom changes my ISDN to VoIP... Don't worry, Asterisk works very well with Deutsche Telekom and there new ip-based connections ... That's a good news... Currenty I configured my sip.conf so: register = 004935:MYSECRET@pbxluca/004935 register = 0049351222:MYSECRET@pbxfax/0049351222 register = 0049351333:MYSECRET@pbxanika/0049351333 register = 44:MYVERYSECRET@messagenet/44 [pbxluca] type=peer defaultuser=004935 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=004935 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [pbxfax] type=peer defaultuser=0049351222 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=fax_incoming outboundproxy=172.16.34.132 port=5060 fromuser=0049351222 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [pbxanika] type=peer defaultuser=0049351333 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=anika_incoming outboundproxy=172.16.34.132 port=5060 fromuser=0049351333 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [messagenet] type=peer defaultuser=44 secret=MYVERYSECRET dtmfmode=rfc2833 host=sip.messagenet.it context=messagenet_incoming outboundproxy=sip.messagenet.it port=5061 fromuser=44 fromdomain=sip.messagenet.it usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=60 Am I right if I say, that I just have to change defaultuser, host, secret, outboundproxy and fromdomain with the data from Telekom and it works? I thinks, it should be: defaultuser=0049351333 secret= MYSECRET host=tel.t-online.de context=anika_incoming outboundproxy=tel.t-online.de port=5060 fromuser=0049351333 fromdomain=tel.t-online.de I'm not sure, where I should write my Login (from my DSL-Line)... I see this page (in German): http://hilfe.telekom.de/hsp/cms/content/HSP/de/3378/FAQ/theme-133631783/Auftrag/theme-82239611/IP-basierter-Anschluss/faq-350884716;jsessionid=A18F587E00F25C8FC26ACF3685481D72 Could you please help me? Thanks Luca Bertoncello (lucab...@lucabert.de) -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.10 (GNU/Linux) iEYEARECAAYFAlVqzVAACgkQ8Ggznj+1EDifYwCgiQTeZQsUljAP5CNpteeFW5aV ugMAn0BnmlGJRHqBJA19DXPgqv0ZUqq1 =vt3E -END PGP SIGNATURE- Hi Luca, I had a discussion recently regarding Asterisk and your provider. The result you can basically find in this message: http://lists.digium.com/pipermail/asterisk-users/2015-April/286353.html Regards, Sebastian -- _ -- 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
