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Last Post
[asterisk-users] ConfBridge play message to all in conf
1
geisj at pagestation.com
1688
Tue Jul 14, 2015 3:15 pm Shishir.Pokharel at on...
[asterisk-users] pjsip.conf question
4
dan at amtelco.com
3021
Tue Jul 14, 2015 3:01 pm dan at amtelco.com
[asterisk-users] RES: RES: RES: How to dial extensions asynchronous-sequentially ?
0
pimenta at inatel.br
1636
Tue Jul 14, 2015 6:55 am pimenta at inatel.br
[asterisk-users] Dial L options and attended tranfer
0
sysadmin at reed-media...
1470
Tue Jul 14, 2015 3:30 am sysadmin at reed-media...
[asterisk-users] RES: RES: How to dial extensions asynchronous-sequentially ?
2
pimenta at inatel.br
2149
Mon Jul 13, 2015 4:57 pm govoiper at gmail.com
[asterisk-users] RES: How to dial extensions asynchronous-sequentially ?
0
govoiper at gmail.com
1575
Mon Jul 13, 2015 3:44 pm govoiper at gmail.com
[asterisk-users] RES: How to dial extensions asynchronous-sequentially ?
0
pimenta at inatel.br
1505
Mon Jul 13, 2015 2:51 pm pimenta at inatel.br
[asterisk-users] How to dial extensions asynchronous-sequentially ?
1
pimenta at inatel.br
1846
Mon Jul 13, 2015 2:24 pm govoiper at gmail.com
[asterisk-users] CEL eventtime drift
0
viljoens at verishare....
1541
Mon Jul 13, 2015 7:35 am viljoens at verishare....
[asterisk-users] Tdm4010p 4 port card
6
tvjudge at gmail.com
3394
Sun Jul 12, 2015 2:27 am tzafrir.cohen at xorco...
[asterisk-users] Does the asterisk support instance messaging ?
1
engthyda at gmail.com
1710
Sun Jul 12, 2015 2:04 am jrees at gmlnt.com
[asterisk-users] Does the asterisk support instance mes saging ?
0
jrees at gmlnt.com
1415
Sun Jul 12, 2015 2:04 am jrees at gmlnt.com
[asterisk-users] RES: Messages out of calls. Is it really possible?
0
pimenta at inatel.br
1633
Fri Jul 10, 2015 1:48 pm pimenta at inatel.br
[asterisk-users] Messages out of calls. Is it really possible?
1
pimenta at inatel.br
1986
Fri Jul 10, 2015 1:29 pm mjordan at digium.com
[asterisk-users] Can I use PJSIP_HEADER to read the SIP 183 message header?
1
pimenta at inatel.br
1928
Fri Jul 10, 2015 1:14 pm mmichelson at digium.com
[asterisk-users] Sending E-Mail from voicemail
2
lucabert at lucabert.de
2199
Fri Jul 10, 2015 11:37 am johnkiniston at gmail.com
[asterisk-users] Sending E-Mail from voicemail with AND without attachment
2
lucabert at lucabert.de
2017
Fri Jul 10, 2015 10:02 am motty.cruz at gmail.com
[asterisk-users] Asterisk SMS
2
engthyda at gmail.com
2054
Fri Jul 10, 2015 9:49 am BryantZ at zktech.com
[asterisk-users] 11.18.0 patch against 11.17.0 running version failed to apply
0
jrees at gmlnt.com
1342
Fri Jul 10, 2015 6:10 am jrees at gmlnt.com
[asterisk-users] 11.18.0 patch against 11.17.0 running version failed to apply
5
admin at tootai.net
2885
Fri Jul 10, 2015 6:09 am jrees at gmlnt.com
[asterisk-users] Can't install gmime22
2
harel at mayorcom.com
2067
Thu Jul 09, 2015 10:09 pm mjordan at digium.com
[asterisk-users] What Dial Plan function can access the contents of the SDP ?
0
pimenta at inatel.br
1454
Thu Jul 09, 2015 2:54 pm pimenta at inatel.br
[asterisk-users] Asterisk 13 / realtime voicemail creation
2
cedric at oceanet.com
2178
Thu Jul 09, 2015 10:40 am cedric at oceanet.com
[asterisk-users] Call Return
2
andrew at convergedgro...
2028
Thu Jul 09, 2015 9:06 am andrew at convergedgro...
[asterisk-users] PJSIP, T.38 fax gateway
0
jd.girard at sysnux.pf
1386
Thu Jul 09, 2015 12:10 am jd.girard at sysnux.pf
[asterisk-users] tls on asterisk 13
4
xserverlinux at gmail.com
2540
Wed Jul 08, 2015 3:34 pm xserverlinux at gmail.com
[asterisk-users] RES: How many SIP 183 messages a caller receives when many callee rings?
0
pimenta at inatel.br
1367
Wed Jul 08, 2015 3:27 pm pimenta at inatel.br
[asterisk-users] How to handle multiple lines call
2
engthyda at gmail.com
1801
Wed Jul 08, 2015 1:58 pm mcgrathr at mail2web.com
[asterisk-users] Bug in ast_frame_adjust_volume in 12.2.0?
2
kenner at gnat.com
1877
Wed Jul 08, 2015 10:57 am kenner at gnat.com
[asterisk-users] How to enable IM over the asterisk server
8
engthyda at gmail.com
3558
Wed Jul 08, 2015 10:51 am jcass78 at gmail.com
[asterisk-users] How may SIP 183 messages a caller receives when many callee rings?
1
pimenta at inatel.br
1565
Wed Jul 08, 2015 9:54 am jcolp at digium.com
[asterisk-users] Can I use ARI to update the builtin database, without executing the dial plan?
1
pimenta at inatel.br
1561
Wed Jul 08, 2015 9:51 am jcolp at digium.com
[asterisk-users] תשובה: Asterisk how to setup alarm too many outgoing calls from same user
0
isrlgb at gmail.com
1220
Wed Jul 08, 2015 4:33 am isrlgb at gmail.com
[asterisk-users] DTMF issue
8
jrees at gmlnt.com
3286
Wed Jul 08, 2015 4:27 am jrees at gmlnt.com
[asterisk-users] Asterisk how to setup alarm too many outgoing calls from same user
2
motty.cruz at gmail.com
2160
Wed Jul 08, 2015 4:22 am ish at pack-net.co.uk
[asterisk-users] Asterisk pin code for out-going international calls (safeguard against fraud)
5
motty.cruz at gmail.com
2629
Tue Jul 07, 2015 2:36 pm johnkiniston at gmail.com
[asterisk-users] RES: What database should I use, for simple data storing? SQLite or the buitin one?
1
pimenta at inatel.br
1535
Tue Jul 07, 2015 12:06 pm pimenta at inatel.br
[asterisk-users] RES: Fwd: What database should I use, for simple data storing? SQLite or the buitin one?
0
pimenta at inatel.br
1261
Tue Jul 07, 2015 12:03 pm pimenta at inatel.br
[asterisk-users] Voicemail: saycid without prefix
4
lucabert at lucabert.de
2263
Tue Jul 07, 2015 11:35 am johnkiniston at gmail.com
[asterisk-users] Issue call quality: Asterisk call quality on trunks
1
kvandenouweland at van...
1668
Tue Jul 07, 2015 10:33 am henry at usinternet.com
[asterisk-users] What database should I use, for simple data storing? SQLite or the buitin one?
2
pimenta at inatel.br
1863
Tue Jul 07, 2015 9:59 am asterisk at voipbusine...
[asterisk-users] Fwd: What database should I use, for simple data storing? SQLite or the buitin one?
0
support at telium.ca
1230
Tue Jul 07, 2015 9:52 am support at telium.ca
[asterisk-users] CDR in an MySQL-Database
5
lucabert at lucabert.de
3004
Tue Jul 07, 2015 12:15 am lucabert at lucabert.de
[asterisk-users] SIP/2.0 401 Unauthorized when calling from one SIP extension to another
0
jorgearturo at gmail.com
1372
Mon Jul 06, 2015 1:24 pm jorgearturo at gmail.com
[asterisk-users] Asterisk 13.4.0 - mixmonitor only records one side's perspective
0
viljoens at verishare....
1466
Mon Jul 06, 2015 9:15 am viljoens at verishare....
[asterisk-users] Unisteam not showing callerid
0
satskiy.a at gmail.com
1515
Mon Jul 06, 2015 5:20 am satskiy.a at gmail.com
[asterisk-users] Choosing codecs
10
lucabert at lucabert.de
4373
Mon Jul 06, 2015 4:59 am lucabert at lucabert.de
[asterisk-users] Action Originate in Asterisk 13 creates 2 calls in core show channels
2
abgenis at gmail.com
2107
Fri Jul 03, 2015 6:24 pm mjordan at digium.com
[asterisk-users] Asterisk 11 and pulseaudio setup as local user
7
geisj at pagestation.com
3566
Fri Jul 03, 2015 5:32 pm lists at venturevoip.com
[asterisk-users] confbridge play tone before speaking
0
geisj at pagestation.com
1402
Thu Jul 02, 2015 12:02 pm geisj at pagestation.com
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