Re: [asterisk-users] MOH don't work after update
Take a look here: http://blog.our-files.com/2012/07/format_mp3-so-building-for-asterisk-1-8-11-using-packages-asterisk-org/ Am 16.06.2013 09:43, schrieb Olivier CALVANO: Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c, Fermeture) in new stack [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for Fermeture -- Executing [334xx@Accueil_Phibee_HNO:4] Hangup(SIP/SIP05-000c, ) in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP05-000c' I understand that he search the file in .ulaw, but why i don't use the mp3 ? musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/moh [Horaires] mode=quietmp3 directory=/var/lib/asterisk/moh/Horaires ps fax: 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 find /var/lib/asterisk/moh/ /var/lib/asterisk/moh/Horaires/Fermeture.mp3 ll -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010 /var/lib/asterisk/moh/Horaires/Fermeture.mp3 -- _ -- 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] block certain numbers
Hi. i would like to manually create a list of numbers to block. these numbers are from spammers (advertizers). is there an easy way to send these particular numbers to busy or even drop the call? -- _ -- 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] block certain numbers
On Monday 17 June 2013, binary dreamer wrote:
Hi.
i would like to manually create a list of numbers to block.
these numbers are from spammers (advertizers).
is there an easy way to send these particular numbers to busy or even drop
the call?
Yes! Dead easy.
Use an external script, written in your favourite language, to look up the
number in some sort of database and return failure (exit 1) if it finds it
there, or success (exit 0) if not. Call this with System() in dialplan. If
the System() call succeeds (meaning the number was not found in the
database), Asterisk will move onto the next priority; if it fails (meaning
the number was in the database) then it will move on by an extra 100.
Alternatively, you can read the value of ${SYSTEMSTATUS} to get the exit code.
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] block certain numbers
hello
if you have just some numbers to block you can use the below code in your
dial plan
exten = 5xx,1,NoOp(Caller-ID: ${CALLERID(all)})
exten = 5xx,n,GotoIf($[${CALLERID(num)}=0661xx ]?3:4)
exten = 5xx,n,hangup
exten = 5xx,n,Dial(SIP/223, 30)
2013/6/17 A J Stiles asterisk_l...@earthshod.co.uk
On Monday 17 June 2013, binary dreamer wrote:
Hi.
i would like to manually create a list of numbers to block.
these numbers are from spammers (advertizers).
is there an easy way to send these particular numbers to busy or even
drop
the call?
Yes! Dead easy.
Use an external script, written in your favourite language, to look up the
number in some sort of database and return failure (exit 1) if it finds
it
there, or success (exit 0) if not. Call this with System() in dialplan.
If
the System() call succeeds (meaning the number was not found in the
database), Asterisk will move onto the next priority; if it fails
(meaning
the number was in the database) then it will move on by an extra 100.
Alternatively, you can read the value of ${SYSTEMSTATUS} to get the exit
code.
--
AJS
Answers come *after* questions.
--
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[asterisk-users] Asterisk / PHP-AGI / pthreads
Hi there, does anyone have experience with Asterisk-AGI-Scripts in PHP while using pthreads in PHP? Are there any limitations or problems known? Best regards -Thorsten- -- _ -- 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] Queue Limit Callers
Hi, I have a requirement, which I am not sure whether it can be implemented. I had done some searches but didnt find an answer to this. Kindly let me know if some one has an idea to implement this: I have two Queues - Sales Booking I have 12 Agents who are added to both the queues Suppose there are 12 calls in the Booking Queue, and 6 calls in the Sales Queue. Only 8 calls in the Booking Queue should hit the Agents and the other 4 calls should remain in hold. 4 calls in the Sales Queue should hit the other 4 agents and the other 2 call should be in hold. Means at a time a maximum of 8 Booking calls only should hit the agents and 4 Sales Calls only should hit the agents. If number of logged in agents are less, proportionally the number of call limit should be reduced. For example, if there are only 10 agents, 7 Booking Calls should hit and 3 Sales calls should hit. The idea is that all agents should be able to answer calls in both queues in rotation. Otherwise its possible to add some agents to booking queue and other agents to sales queue. But thats not what is required. Kindly help if there is some idea to implement this. Regards Shanavaz. -- _ -- 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] MOH don't work after update
i don't think's that it's the same problems, because me format_mp3.so is loaded: [root@ipbx Conf-Extensions]# asterisk -rx 'core show file formats' Format Name Extensions -- -- slin mp3mp3 gsmgsmgsm slin192sln192 sln192 slin96 sln96 sln96 slin48 sln48 sln48 slin44 sln44 sln44 slin32 sln32 sln32 slin24 sln24 sln24 slin16 sln16 sln16 slin12 sln12 sln12 slin slnsln|raw ilbc iLBC ilbc g723 g723sf g723|g723sf slin16 wav16 wav16 slin wavwav siren14siren14siren14 g719 g719 g719 h264 h264 h264 g726 g726-16g726-16 g726 g726-24g726-24 g726 g726-32g726-32 g726 g726-40g726-40 g729 g729 g729 siren7 siren7 siren7 gsmwav49 WAV|wav49 g722 g722 g722 ulaw au au alaw alaw alaw|al|alw ulaw pcmpcm|ulaw|ul|mu|ulw adpcm voxvox h263 h263 h263 31 file formats registered. i see the mp3 file format 2013/6/17 Thorsten Göllner t...@ovm-group.com Take a look here: http://blog.our-files.com/2012/07/format_mp3-so-building-for-asterisk-1-8-11-using-packages-asterisk-org/ Am 16.06.2013 09:43, schrieb Olivier CALVANO: Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c, Fermeture) in new stack [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for Fermeture -- Executing [334xx@Accueil_Phibee_HNO:4] Hangup(SIP/SIP05-000c, ) in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP05-000c' I understand that he search the file in .ulaw, but why i don't use the mp3 ? musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/moh [Horaires] mode=quietmp3 directory=/var/lib/asterisk/moh/Horaires ps fax: 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 find /var/lib/asterisk/moh/ /var/lib/asterisk/moh/Horaires/Fermeture.mp3 ll -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010 /var/lib/asterisk/moh/Horaires/Fermeture.mp3 -- _ -- 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] MOH don't work after update
If I read your log entries correctly, you are not playing any MOH at all. BackGround() normally plays sound files from the language dependent sound directory. 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] MOH don't work after update
Is the subdir Horaires readable/executable for User Asterisk/Asterisk? Did you try to convert it to wav? Am 17.06.2013 09:47, schrieb Thorsten Göllner: Take a look here: http://blog.our-files.com/2012/07/format_mp3-so-building-for-asterisk-1-8-11-using-packages-asterisk-org/ Am 16.06.2013 09:43, schrieb Olivier CALVANO: Hi we have a small problems. We have a Asterisk 1.6.1 old server with music on old. we have updated to AsteriskNow 11.4.0 and now, when we want play sound, we have a errors: -- Executing [334xx@Accueil_HNO:2] BackGround(SIP/SIP05-000c, Fermeture) in new stack [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:701 ast_openstream_full: File Fermeture does not exist in any format [Jun 16 07:35:06] WARNING[7634][C-0006]: file.c:1017 ast_streamfile: Unable to open Fermeture (format (alaw)): No such file or directory [Jun 16 07:35:06] WARNING[7634][C-0006]: pbx.c:11180 pbx_builtin_background: ast_streamfile failed on SIP/SIP05-000c for Fermeture -- Executing [334xx@Accueil_Phibee_HNO:4] Hangup(SIP/SIP05-000c, ) in new stack == Spawn extension (Accueil_HNO, 334xx, 4) exited non-zero on 'SIP/SIP05-000c' I understand that he search the file in .ulaw, but why i don't use the mp3 ? musiconhold.conf [default] mode=quietmp3 directory=/var/lib/asterisk/moh [Horaires] mode=quietmp3 directory=/var/lib/asterisk/moh/Horaires ps fax: 7555 pts/0S 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk 7558 pts/0Sl 0:06 \_ /usr/sbin/asterisk -f -U asterisk -G asterisk -vvvg -c 7578 pts/0S 0:00 \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 7580 pts/0S 0:00 | \_ mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 Fermeture.mp3 find /var/lib/asterisk/moh/ /var/lib/asterisk/moh/Horaires/Fermeture.mp3 ll -rw-r--r-- 1 asterisk asterisk 1396613 Nov 24 2010 /var/lib/asterisk/moh/Horaires/Fermeture.mp3 -- _ -- 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] block certain numbers
i am using asterisk's internal database to create a phonebook. i was
thinking to create something similar for the blocking, but i got stuck on
how to block the numbers.
On Mon, Jun 17, 2013 at 12:47 PM, Salaheddine Elharit
salah.elharit...@gmail.com wrote:
hello
if you have just some numbers to block you can use the below code in
your dial plan
exten = 5xx,1,NoOp(Caller-ID: ${CALLERID(all)})
exten = 5xx,n,GotoIf($[${CALLERID(num)}=0661xx ]?3:4)
exten = 5xx,n,hangup
exten = 5xx,n,Dial(SIP/223, 30)
2013/6/17 A J Stiles asterisk_l...@earthshod.co.uk
On Monday 17 June 2013, binary dreamer wrote:
Hi.
i would like to manually create a list of numbers to block.
these numbers are from spammers (advertizers).
is there an easy way to send these particular numbers to busy or even
drop
the call?
Yes! Dead easy.
Use an external script, written in your favourite language, to look up the
number in some sort of database and return failure (exit 1) if it finds
it
there, or success (exit 0) if not. Call this with System() in
dialplan. If
the System() call succeeds (meaning the number was not found in the
database), Asterisk will move onto the next priority; if it fails
(meaning
the number was in the database) then it will move on by an extra 100.
Alternatively, you can read the value of ${SYSTEMSTATUS} to get the exit
code.
--
AJS
Answers come *after* questions.
--
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[asterisk-users] Cisco SSCP to SIP
Hi all, I'm trying to convers some Cisco SSCP phones to the SIP formware. The phone boots, I see it tries to fetch a bunch of files on my TFTP: Jun 17 09:37:45 firewall dnsmasq-dhcp[21202]: DHCPACK(eth2) 192.168.10.103 6c:50:4d:da:f0:67 SEP6C504DDAF067 Jun 17 09:38:10 firewall in.tftpd[22666]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22666]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22667]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22667]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22668]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:10 firewall in.tftpd[22668]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22669]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22669]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22671]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22671]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22672]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22672]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22673]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22673]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22674]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:12 firewall in.tftpd[22674]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22675]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22675]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22676]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22676]: sending NAK (1, File not found) to 192.168.10.103 But none of those are the SIP firmware filename I downloaded... Any hints ? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk / PHP-AGI / pthreads
On Mon, 17 Jun 2013, Thorsten Göllner wrote:
does anyone have experience with Asterisk-AGI-Scripts in PHP while using
pthreads in PHP? Are there any limitations or problems known?
I've written 'pthread-ed' AGIs in C.
The only 'pthread related' limitation I stumbled into is that you can only
execute a single AGI request at a time -- which is kind of obvious if you
understand the AGI protocol.
My use case was playing a file ('Please wait while we authorize your
credit card') while processing the credit request.
Since our card processor almost always returned the credit response before
the end of the file, the 'user experience' was that the credit request was
instantaneous.
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000--
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Re: [asterisk-users] Problem with CEL logging and channel bridging
Il 13/06/2013 11:31, Fabio Moretti scrisse: Hi, I've already post this to the forum three days ago, sorry if it's sounds like a crosspost, but I've got no replies, so I'm trying other channels :) ok, definitely CEL is a big question mark for most of us. can someone point me to in deep CEL documentation or to an open source code that use it so I can study more? not asterisk code, please, I tried but I find really hard find how and then events are generated. thanks -- Fabio Moretti Gerente de Sistemas www.tecytal.com 0800 8780 (+598) 248 77921 -- _ -- 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] Cisco SSCP to SIP
Even with the Cisco SIP firmware on the phones you still have to provide the XML configuration files to the phone via the TFTP. You need to have the XMLDefault.cnf.xml and SEPMAC ADDRESS.cnf.xml at the least... Jacob -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andre Courchesne Sent: Monday, June 17, 2013 8:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco SSCP to SIP Hi all, I'm trying to convers some Cisco SSCP phones to the SIP formware. The phone boots, I see it tries to fetch a bunch of files on my TFTP: Jun 17 09:37:45 firewall dnsmasq-dhcp[21202]: DHCPACK(eth2) 192.168.10.103 6c:50:4d:da:f0:67 SEP6C504DDAF067 Jun 17 09:38:10 firewall in.tftpd[22666]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22666]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22667]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22667]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22668]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:10 firewall in.tftpd[22668]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22669]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22669]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22671]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22671]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22672]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22672]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22673]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22673]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22674]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:12 firewall in.tftpd[22674]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22675]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22675]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22676]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22676]: sending NAK (1, File not found) to 192.168.10.103 But none of those are the SIP firmware filename I downloaded... Any hints ? Thanks. -- _ -- 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] Cisco SSCP to SIP
Yes I'm aware of the provisioning files, but first I need to have the freaking think update to an SIP firmware ;-) I found this how-to which is the best I found so far: http://www.adrianandgenese.com/blogger/2011/02/16/how-to-upgrade-or-convert-a-cisco-ip-79xx-7940-7960-794x-796x-797x-phone-to-sip-or-sccp/ However I see that the phone tries to fetch XMLDefault.cnf.xml.sgn and not XMLDefault.cnf.xml... Any idea what is the sgn extension ? Andre On 2013-06-17, at 9:48 AM, jacob.e.mi...@l-3com.com wrote: Even with the Cisco SIP firmware on the phones you still have to provide the XML configuration files to the phone via the TFTP. You need to have the XMLDefault.cnf.xml and SEPMAC ADDRESS.cnf.xml at the least... Jacob -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andre Courchesne Sent: Monday, June 17, 2013 8:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco SSCP to SIP Hi all, I'm trying to convers some Cisco SSCP phones to the SIP formware. The phone boots, I see it tries to fetch a bunch of files on my TFTP: Jun 17 09:37:45 firewall dnsmasq-dhcp[21202]: DHCPACK(eth2) 192.168.10.103 6c:50:4d:da:f0:67 SEP6C504DDAF067 Jun 17 09:38:10 firewall in.tftpd[22666]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22666]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22667]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22667]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22668]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:10 firewall in.tftpd[22668]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22669]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22669]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22671]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22671]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22672]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22672]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22673]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22673]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22674]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:12 firewall in.tftpd[22674]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22675]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22675]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22676]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22676]: sending NAK (1, File not found) to 192.168.10.103 But none of those are the SIP firmware filename I downloaded... Any hints ? Thanks. -- _ -- 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 -- _ -- 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] SIGTRAN Integration
Anyone? N. -- _ -- 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] block certain numbers
On Monday 17 June 2013, binary dreamer wrote:
i am using asterisk's internal database to create a phonebook. i was
thinking to create something similar for the blocking, but i got stuck on
how to block the numbers.
Well, once you've worked out whether the caller is welcome or not, then you
need to dial your phone if the caller is allowed; or just play a suitable
message and then Hangup() if they aren't.
exten = s,1,NoOp(Incoming call from ${CALLERID(num)})
exten = s,2,System(check_ban_db ${CALLERID(num)})
; goes to 3 if system cmd exited OK or 103 if exited non-zero
exten = s,3,Dial(${MY_PHONE})
exten = s,4,Hangup()
; step 103 is where we deal with unwelcome callers
exten = s,103,Playback(f-off)
exten = s,104,Hangup()
You just need to make sure that you have a suitable message saved in
/var/lib/asterisk/sounds/f-off.wav .
--
AJS
Answers come *after* questions.
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Re: [asterisk-users] Problem with CEL logging and channel bridging
On Thu, Jun 13, 2013 at 9:31 AM, Fabio Moretti fmore...@tecytal.com wrote: Hi, I've already post this to the forum three days ago, sorry if it's sounds like a crosspost, but I've got no replies, so I'm trying other channels :) This is the link to the forum post if someone prefer to reply here: http://forums.asterisk.org/viewtopic.php?f=1t=86985 I'm using Asterisk 1.8.20.0 (the freepbx build) with CEL logging activated. I'm using CEL because in our pbx we have different queues and trunks serving different customers (we are an inbound call center) and we need to detect when and how we have to bill our customers. I'm facing an issue with the call transfer, for example I have: - call entering a queue - operator answer the call - operator make an outgoing call to reach the customer - operator put in communication the ingoing call with the outgoing this result in various channel to be created/destroyed, and I'm using bridge events to detect what is going on with the call. In this case I have (I've hidden CHAN_START,ANSWER and HANGUP events because they have no useful information in this case): ++---+-+---+-+--+-+-+--+ | id | eventtype | eventtime | exten | context | channame | appname | appdata | peer | ++---+-+---+-+--+-+-+--+ | 965224 | BRIDGE_START | 2013-06-10 10:15:18 | 20| ext-queues | DAHDI/i1/96034296-30a3 | Queue | 20,t,, | Local/1004@from-queue-00019c34;1 | | 965226 | BRIDGE_START | 2013-06-10 10:15:18 | s | macro-dial-one | Local/1004@from-queue-00019c34;2 | Dial| SIP/1004,,trM(auto-blkvm) | SIP/1004-40ce| | 965340 | BRIDGE_UPDATE | 2013-06-10 10:16:08 | s | macro-dialout-trunk | Local/1004@from-queue-00019c34;2 | Dial| IAX2/issuegroup/110,300,| IAX2/issuegroup-17175| | 965513 | BRIDGE_END| 2013-06-10 10:18:15 | 20| ext-queues | DAHDI/i1/96034296-30a3 | Queue | 20,t,, | Local/1004@from-queue-00019c34;1 | | 965515 | BRIDGE_END| 2013-06-10 10:18:15 | s | macro-dialout-trunk | Local/1004@from-queue-00019c34;2 | Dial| IAX2/issuegroup/110,300,| IAX2/issuegroup-17175| ++---+-+---+-+--+-+-+--+ The first BRIDGE_START is the connection between the inbound call (DAHDI/i1/96034296-30a3) and the local phone (Local/1004@from-queue-00019c34;1), the second BRIDGE_START is the connection between the local phone (Local/1004@from-queue-00019c34;2) and the outgoing call (SIP/1004-40ce) that is going out by a IAX trunk. After that I have a BRIGDE_UPDATE event where no field make me know which channel is being updated, I only have the channame (Local/1004@from-queue-00019c34;2) that is the channel being bridged out and the outgoing channel (IAX2/issuegroup-17175), but I have no information that in fact the ingoing call (DAHDI/i1/96034296-30a3) is being bridged to the outgoing channel. I have no other event (TRANSFER or something like that) to know what is going on. snip I think you have two questions here: what is the BRIDGE_UPDATE event telling you, and how do you know the DAHDI channel is communicating with the IAX trunk. A BRIDGE_UPDATE event occurs when a masquerade has happened and the participants in a bridge have been updated. In this particular case, the BRIDGE_UPDATE event is telling you that Local/1004@from-queue-00019c34;2 is no longer bridged with SIP/1004-40ce, but is in fact now bridged with the IAX trunk IAX2/issuegroup-17175. That is, the IAX trunk has taken the place of the SIP channel. Since you were already informed that a bridge started between that Local channel half and the SIP channel, the event only needs to tell you who got replaced - which is what it does. So, how do you know that Local/1004@from-queue-00019c34;2 is associated with DAHDI/i1/96034296-30a3? By definition, Local channels *always* exist in pairs - the two channels together make up one path of communication. The two halves are denoted by a common name with a suffix of ';1' and ';2' - the first half gets the ';1'; the second half gets the ';2'. When both halves are answered, you know that audio will be forwarded from one half to the other and vice versa. Since you know that DAHDI/i1/96034296-30a3 is in a bridge with Local/1004@from-queue-00019c34;1 and Local/1004@from-queue-00019c34;2* *is in a bridge with IAX2/issuegroup-17175, you automatically know that
[asterisk-users] VoIP call quality metrics: who cares?
Hi, How much do you care about call quality metrics to collect and analyze them? What metrics are of interest for you (of course packet loss, jitter, latency, but what else?). We have collected some for your review and would be happy to expand them with those you are using in your Asterisk systems. http://blog.sevana.fi/recommended-voip-call-quality-metrics/ Best Regards, Sevana http://www.sevana.fi -- _ -- 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] Cisco SSCP to SIP
This is for signed XML files, some of the newer models require signed files for security. Is there a reason to use SIP? There is a really good SCCP module for asterisk (chan-sccp-b http://sourceforge.net/projects/chan-sccp-b/ ). Usually you have to set in the SEPMAC ADDRESS.cnf.xml what firmware file to download, if it then finds the firmware file on the TFTP server it will upload and install the new firmware. This process can be difference depending on which Cisco phone you are using. When possible use Ciscos website for instructions on changing the firmware! Jacob From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andre Courchesne Sent: Monday, June 17, 2013 8:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco SSCP to SIP Yes I'm aware of the provisioning files, but first I need to have the freaking think update to an SIP firmware ;-) I found this how-to which is the best I found so far: http://www.adrianandgenese.com/blogger/2011/02/16/how-to-upgrade-or-conv ert-a-cisco-ip-79xx-7940-7960-794x-796x-797x-phone-to-sip-or-sccp/ However I see that the phone tries to fetch XMLDefault.cnf.xml.sgn and not XMLDefault.cnf.xml... Any idea what is the sgn extension ? Andre On 2013-06-17, at 9:48 AM, jacob.e.mi...@l-3com.com wrote: Even with the Cisco SIP firmware on the phones you still have to provide the XML configuration files to the phone via the TFTP. You need to have the XMLDefault.cnf.xml and SEPMAC ADDRESS.cnf.xml at the least... Jacob -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andre Courchesne Sent: Monday, June 17, 2013 8:42 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Cisco SSCP to SIP Hi all, I'm trying to convers some Cisco SSCP phones to the SIP formware. The phone boots, I see it tries to fetch a bunch of files on my TFTP: Jun 17 09:37:45 firewall dnsmasq-dhcp[21202]: DHCPACK(eth2) 192.168.10.103 6c:50:4d:da:f0:67 SEP6C504DDAF067 Jun 17 09:38:10 firewall in.tftpd[22666]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22666]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22667]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:10 firewall in.tftpd[22667]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:10 firewall in.tftpd[22668]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:10 firewall in.tftpd[22668]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22669]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22669]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:11 firewall in.tftpd[22671]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:11 firewall in.tftpd[22671]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22672]: RRQ from 192.168.10.103 filename CTLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22672]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22673]: RRQ from 192.168.10.103 filename ITLSEP6C504DDAF067.tlv Jun 17 09:38:12 firewall in.tftpd[22673]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22674]: RRQ from 192.168.10.103 filename ITLFile.tlv Jun 17 09:38:12 firewall in.tftpd[22674]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22675]: RRQ from 192.168.10.103 filename SEP6C504DDAF067.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22675]: sending NAK (1, File not found) to 192.168.10.103 Jun 17 09:38:12 firewall in.tftpd[22676]: RRQ from 192.168.10.103 filename XMLDefault.cnf.xml.sgn Jun 17 09:38:12 firewall in.tftpd[22676]: sending NAK (1, File not found) to 192.168.10.103 But none of those are the SIP firmware filename I downloaded... Any hints ? Thanks. -- _ -- 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] Problem with CEL logging and channel bridging
Il 17/06/2013 11:13, Matthew Jordan scrisse: Since you know that DAHDI/i1/96034296-30a3 is in a bridge with Local/1004@from-queue-00019c34;1 and Local/1004@from-queue-00019c34;2is in a bridge withIAX2/issuegroup-17175, you automatically know that DAHDI/i1/96034296-30a3 and IAX2/issuegroup-17175 can communicate (at least once everyone has Answered). The system you build on top of CEL has to understand the semantics of Local channels and tie the two together. Matt matt, thank you very much. in fact I was wondering if local-channel;1 and local-channel;2 have to be considered as "one" channel or not. Can I ask you if there's a in deep documentation of how channel and events are generated/destroyed? I'm trying to find the time to study, I'd like to generate a billing script based on CEL and a graphical interface for visualizing calls history. really thank you -- Fabio Moretti Gerente de Sistemas www.tecytal.com 0800 8780 (+598) 248 77921 -- _ -- 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] Has anyone succeeded in making a WebRTC call from Mozilla Nightly to Asterisk?
I am using Asterisk 11.3.0 and just updated Nightly to 24.0a1 (2013-06017) and get a SIP 488 Not Acceptable Here response. I have no problems using the same Asterisk configuration and the same page to make a call from Chrome. I have seen other people post a similar issue, but I have not seen a solution. If someone with good knowledge of this issue were to respond with this is a known issue or no, and this should be reported to Mozilla, that would be very helpful for me as well. Here is the error I see in the Asterisk console after it successfully parses the SDP a lines: Rejecting secure audio stream without encryption details: audio 62583 UDP/TLS/RTP/SAVPF 109 0 8 101 Trying to put 'SIP/2.0 488' onto WS socket destined for www.xxx.yyy.zzz:5060 No compatible codecs for this SIP call. Here is the sip.conf info. I have tried various permutations of the dtls and encryption parameters with no luck. I do have openssl and srtp built into Asterisk (that solved a different error dealing with the RTP engine). [webrtc-dtls] ; Add DTLS stuff for Mozilla Nightly (and eventually Firefox) type=user host=dynamic hassip=yes transport=ws,wss directmedia=no ; proxy the media icesupport=yes ; needed for webrtc avpf=yes; needed for webrtc context=default encryption=yes dtlsenable=yes dtlsverify=no dtlsrekey=60 dtlscafile=/opt/asterisk/keys/ca.crt dtlscertfile=/opt/asterisk/keys/asterisk.pem dtlssetup=actpass insecure=invite Here is the SDP offered by Nightly: v=0 o=Mozilla-SIPUA-24.0a1 25687 1 IN IP4 0.0.0.0 s=Doubango Telecom - firefox t=0 0 a=ice-ufrag:7194cbcc a=ice-pwd:e57c14491015e529b84c5a6baf6d7b67 a=fingerprint:sha-256 48:3E:0C:59:BA:EB:6C:F9:5D:65:BF:08:54:63:C3:EA:AF:A9:60:9D:39:47:A5:41:6B:E1:A8:EB:7C:06:BE:D4 m=audio 62583 UDP/TLS/RTP/SAVPF 109 0 8 101 c=IN IP4 www.xxx.yyy.zzz a=rtpmap:109 opus/48000/2 a=ptime:20 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=candidate:0 1 UDP 2111832319 192.168.1.109 62583 typ host a=candidate:1 1 UDP 1692467199 www.xxx.yyy.zzz 62583 typ srflx raddr 192.168.1.109 rport 62583 a=candidate:5 1 UDP 2111766783 192.168.56.1 62584 typ host a=candidate:0 2 UDP 2111832318 192.168.1.109 62585 typ host a=candidate:1 2 UDP 1692467198 www.xxx.yyy.zzz 62585 typ srflx raddr 192.168.1.109 rport 62585 a=candidate:5 2 UDP 2111766782 192.168.56.1 62586 typ host Thanks, - Joel -- _ -- 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] block certain numbers
That seems suitable.
for the cid name i am using asterisk's internal database for the lookup
such as
*database put cidname 222333 xyzwhoevername*
then how do you create the database for the check_ban_db?
On Monday 17 June 2013, binary wrote:
i am using asterisk's internal database to create a phonebook. i was
thinking to create something similar for the blocking, but i got stuck on
how to block the numbers.
Well, once you've worked out whether the caller is welcome or not, then you
need to dial your phone if the caller is allowed; or just play a suitable
message and then Hangup() if they aren't.
exten = s,1,NoOp(Incoming call from ${CALLERID(num)})
exten = s,2,System(check_ban_db ${CALLERID(num)})
; goes to 3 if system cmd exited OK or 103 if exited non-zero
exten = s,3,Dial(${MY_PHONE})
exten = s,4,Hangup()
; step 103 is where we deal with unwelcome callers
exten = s,103,Playback(f-off)
exten = s,104,Hangup()
You just need to make sure that you have a suitable message saved in
/var/lib/asterisk/sounds/f-off.wav .
--
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[asterisk-users] Is Asternic.net out of business (Flash Operator, Call Center Stats)?
We have licensed both products and sent a support request on 6/11, with zero reply or any activity on it at all so far. No replies to subsequent ticket updates or e-mails. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Is Asternic.net out of business (Flash Operator, Call Center Stats)?
No. Although Nicolas may have gone on holiday. I just purchased 2 licenses for fop2 a month or so ago. Carlos Alvarez car...@televolve.com wrote: We have licensed both products and sent a support request on 6/11, with zero reply or any activity on it at all so far. No replies to subsequent ticket updates or e-mails. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Is Asternic.net out of business (Flash Operator, Call Center Stats)?
I have never known them to not reply quickly. Email me offlist and I will give you non generic email addresses. -- ringfree.biz supp...@ringfree.biz 828-575-0030 On Jun 17, 2013, at 8:14 PM, Carlos Alvarez car...@televolve.com wrote: We have licensed both products and sent a support request on 6/11, with zero reply or any activity on it at all so far. No replies to subsequent ticket updates or e-mails. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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] Is Asternic.net out of business (Flash Operator, Call Center Stats)?
No vacation notice, nothing, other than the system auto-replying saying that the ticket will be closed because we didn't have any action on it. Rather distressing for our customers. On Mon, Jun 17, 2013 at 5:22 PM, Gregory Malsack gmals...@coastalacq.comwrote: No. Although Nicolas may have gone on holiday. I just purchased 2 licenses for fop2 a month or so ago. Carlos Alvarez car...@televolve.com wrote: We have licensed both products and sent a support request on 6/11, with zero reply or any activity on it at all so far. No replies to subsequent ticket updates or e-mails. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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 -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- 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
