asterisk users - Jul 2015

Friday July 31 2015
TimeRepliesSubject
6:10PM 0 showing sip number insted of pri number
4:10PM 4 showing sip number insted of pri number
1:04PM 0 How to configure through the GUI 35 cisco ip phones, -spa502g (jg)
 
Thursday July 30 2015
TimeRepliesSubject
11:20PM 0 PJSIP T.38 issues
8:56AM 0 Insecure Meaning.
6:51AM 0 CEL eventtime incorrect, but CDR times are correct - 1.8.11.0
4:02AM 0 Windows Asterisk Help
12:56AM 1 How to configure through the GUI 35 cisco ip phones -spa502g
 
Wednesday July 29 2015
TimeRepliesSubject
10:40PM 0 Asterisk 1.8.22.0 built - encrypt authentication
8:30PM 3 Asterisk 1.8.22.0 built - encrypt authentication
4:53PM 0 Queues don't follow dialplan if no members are registered
4:47PM 2 Windows Asterisk Help
3:16PM 0 Windows Asterisk Help
2:26PM 2 Windows Asterisk Help
2:11PM 0 Windows Asterisk Help
1:59PM 3 Windows Asterisk Help
4:13AM 2 PJSIP T.38 issues
3:51AM 2 Queues don't follow dialplan if no members are registered
 
Tuesday July 28 2015
TimeRepliesSubject
9:26PM 0 Siren7 and Asterisk 13
5:12PM 0 Queues don't follow dialplan if no members are registered
4:58PM 2 Queues don't follow dialplan if no members are registered
10:05AM 1 re-invite update dialog
 
Monday July 27 2015
TimeRepliesSubject
6:38PM 1 Why no CentOS 7 repos?
6:19PM 0 Why no CentOS 7 repos?
5:51PM 2 Why no CentOS 7 repos?
4:54PM 0 No audio on SIP over WebRTC
4:30PM 0 Asterisk 11.19.0-rc1 Now Available
11:22AM 0 PJSIP T.38 issues
9:51AM 1 Filters
3:15AM 2 PJSIP T.38 issues
 
Friday July 24 2015
TimeRepliesSubject
12:41AM 0 Update of dialed number on sip phones
 
Thursday July 23 2015
TimeRepliesSubject
9:54AM 1 Centos 6.5 Asterisk 1.8.11.0 - starts in rc.local, but not contactible?
9:13AM 1 Centos 6.5 Asterisk 1.8.11.0 - starts in rc.local, but not contactible?
3:12AM 0 Cisco 7940 and PJSIP registration
12:50AM 2 Cisco 7940 and PJSIP registration
 
Wednesday July 22 2015
TimeRepliesSubject
11:29AM 0 Cisco 7940 and PJSIP registration
5:38AM 2 Cisco 7940 and PJSIP registration
1:45AM 0 Cisco 7940 and PJSIP registration
1:37AM 2 Cisco 7940 and PJSIP registration
 
Tuesday July 21 2015
TimeRepliesSubject
1:58PM 0 Queue handling : does every call have to be accounted twice?
8:43AM 1 asterisk segfault debian jessie asterisk 11.13
8:27AM 1 Always 486 Busy Here for anonymous calls
7:19AM 0 sip can not transmit fax receive from chan dahdi
 
Monday July 20 2015
TimeRepliesSubject
5:53AM 0 RES: Can I use PJSIP_HEADER to read the SIP 183 message header?
 
Friday July 17 2015
TimeRepliesSubject
2:09PM 1 Recording INCOMING calls
2:06PM 0 Recording INCOMING calls
 
Thursday July 16 2015
TimeRepliesSubject
2:09PM 0 Asterisk and Vitelity's vMobile service
1:54PM 0 How to create direct media with PJSIP.conf configurations in Asterisk 13?
1:49PM 2 How to create direct media with PJSIP.conf configurations in Asterisk 13?
9:06AM 1 How to enable group call
8:51AM 0 how to return a transfered call to the transferrer?
8:37AM 2 Recording INCOMING calls
 
Wednesday July 15 2015
TimeRepliesSubject
9:35PM 1 How to dial extensions asynchronous-sequentially ?
9:24PM 0 How to dial extensions asynchronous-sequentially ?
9:16PM 2 How to dial extensions asynchronous-sequentially ?
7:51PM 2 how to return a transfered call to the transferrer?
5:46PM 1 Problem "no voice"
5:41PM 0 Problem "no voice"
5:01PM 2 Problem "no voice"
4:57PM 0 How to call a group of peers all registered with the same login?
 
Tuesday July 14 2015
TimeRepliesSubject
8:08PM 1 ConfBridge play message to all in conf
7:38PM 0 pjsip.conf question
7:34PM 2 pjsip.conf question
7:03PM 0 pjsip.conf question
6:56PM 2 pjsip.conf question
8:29AM 0 Dial L options and attended tranfer
 
Monday July 13 2015
TimeRepliesSubject
9:57PM 0 RES: RES: How to dial extensions asynchronous-sequentially ?
9:32PM 2 RES: RES: How to dial extensions asynchronous-sequentially ?
8:43PM 0 RES: How to dial extensions asynchronous-sequentially ?
7:51PM 3 RES: How to dial extensions asynchronous-sequentially ?
7:24PM 0 How to dial extensions asynchronous-sequentially ?
6:28PM 3 How to dial extensions asynchronous-sequentially ?
12:38PM 0 CEL eventtime drift
 
Sunday July 12 2015
TimeRepliesSubject
7:04AM 0 Does the asterisk support instance mes saging ?
7:03AM 0 Does the asterisk support instance messaging ?
7:02AM 0 Does the asterisk support instance messaging ?
 
Saturday July 11 2015
TimeRepliesSubject
12:11AM 0 Tdm4010p 4 port card
12:11AM 0 Tdm4010p 4 port card
12:10AM 0 Tdm4010p 4 port card
 
Friday July 10 2015
TimeRepliesSubject
6:36PM 2 RES: Can I use PJSIP_HEADER to read the SIP 183 message header?
6:29PM 0 Messages out of calls. Is it really possible?
6:14PM 0 Can I use PJSIP_HEADER to read the SIP 183 message header?
4:53PM 2 Can I use PJSIP_HEADER to read the SIP 183 message header?
4:51PM 2 Messages out of calls. Is it really possible?
3:37PM 3 Tdm4010p 4 port card
2:45PM 2 Sending E-Mail from voicemail with AND without attachment
11:09AM 0 11.18.0 patch against 11.17.0 running version failed to apply
11:09AM 0 11.18.0 patch against 11.17.0 running version failed to apply
9:22AM 0 Asterisk SMS
8:30AM 2 Asterisk SMS
7:26AM 2 Sending E-Mail from voicemail
 
Thursday July 9 2015
TimeRepliesSubject
7:51PM 0 What Dial Plan function can access the contents of the SDP ?
6:18PM 1 Can't install gmime22
3:34PM 0 Asterisk 13 / realtime voicemail creation
3:28PM 2 Asterisk 13 / realtime voicemail creation
3:05PM 0 11.18.0 patch against 11.17.0 running version failed to apply
8:02AM 0 Call Return
5:10AM 0 PJSIP, T.38 fax gateway
 
Wednesday July 8 2015
TimeRepliesSubject
10:28PM 2 11.18.0 patch against 11.17.0 running version failed to apply
7:11PM 0 tls on asterisk 13
7:09PM 0 tls on asterisk 13
7:05PM 6 tls on asterisk 13
6:58PM 0 How to handle multiple lines call
6:13PM 2 Call Return
4:02PM 1 How to handle multiple lines call
3:45PM 1 How to enable IM over the asterisk server
3:36PM 0 11.18.0 patch against 11.17.0 running version failed to apply
2:53PM 0 How may SIP 183 messages a caller receives when many callee rings?
2:51PM 0 Bug in ast_frame_adjust_volume in 12.2.0?
1:14PM 2 11.18.0 patch against 11.17.0 running version failed to apply
11:55AM 0 How to enable IM over the asterisk server
9:27AM 1 DTMF issue
9:21AM 0 Asterisk how to setup alarm too many outgoing calls from same user
8:24AM 3 How to enable IM over the asterisk server
5:43AM 0 How to enable IM over the asterisk server
 
Tuesday July 7 2015
TimeRepliesSubject
9:23PM 0 DTMF issue
8:53PM 2 DTMF issue
8:28PM 1 Can I use ARI to update the builtin database, without executing the dial plan?
7:44PM 0 DTMF issue
7:22PM 0 Asterisk pin code for out-going international calls (safeguard against fraud)
7:03PM 2 DTMF issue
6:14PM 0 DTMF issue
6:01PM 2 Bug in ast_frame_adjust_volume in 12.2.0?
4:09PM 2 Asterisk pin code for out-going international calls (safeguard against fraud)
2:58PM 0 What database should I use, for simple data storing? SQLite or the buitin one?
2:51PM 1 Fwd: What database should I use, for simple data storing? SQLite or the buitin one?
2:32PM 0 What database should I use, for simple data storing? SQLite or the buitin one?
2:26PM 4 What database should I use, for simple data storing? SQLite or the buitin one?
9:28AM 2 How to enable IM over the asterisk server
8:17AM 0 How to enable IM over the asterisk server
7:33AM 0 Voicemail: saycid without prefix
5:58AM 1 Issue call quality: Asterisk call quality on trunks
4:07AM 2 How to enable IM over the asterisk server
12:32AM 0 CDR in an MySQL-Database
 
Monday July 6 2015
TimeRepliesSubject
11:54PM 0 Asterisk pin code for out-going international calls (safeguard against fraud)
9:53PM 4 DTMF issue
9:43PM 3 Asterisk pin code for out-going international calls (safeguard against fraud)
8:20PM 1 CDR in an MySQL-Database
8:14PM 0 CDR in an MySQL-Database
7:22PM 4 CDR in an MySQL-Database
6:23PM 0 SIP/2.0 401 Unauthorized when calling from one SIP extension to another
4:25PM 2 Voicemail: saycid without prefix
4:23PM 0 Voicemail: saycid without prefix
3:06PM 0 Asterisk how to setup alarm too many outgoing calls from same user
2:27PM 2 Asterisk how to setup alarm too many outgoing calls from same user
2:17PM 0 Asterisk 13.4.0 - mixmonitor only records one side's perspective
1:26PM 2 How may SIP 183 messages a caller receives when many callee rings?
10:19AM 0 Unisteam not showing callerid
9:55AM 0 Choosing codecs
8:57AM 2 Choosing codecs
8:38AM 0 Choosing codecs
8:17AM 3 Choosing codecs
8:05AM 0 Choosing codecs
7:56AM 2 Choosing codecs
7:48AM 0 Choosing codecs
12:53AM 0 Can't install gmime22
 
Sunday July 5 2015
TimeRepliesSubject
8:57PM 0 Choosing codecs
7:34PM 2 Choosing codecs
 
Saturday July 4 2015
TimeRepliesSubject
6:06PM 0 from: rschroe@gmail.com
5:53AM 2 Voicemail: saycid without prefix
 
Friday July 3 2015
TimeRepliesSubject
6:46PM 2 Action Originate in Asterisk 13 creates 2 calls in core show channels
2:10PM 2 Asterisk 11 and pulseaudio setup as local user
11:17AM 0 Asterisk 11 and pulseaudio setup as local user
 
Thursday July 2 2015
TimeRepliesSubject
5:54PM 5 Asterisk 11 and pulseaudio setup as local user
5:02PM 0 confbridge play tone before speaking
3:28PM 0 multiple sip trunks with the same ITSP
12:53PM 0 Custom header when busy
12:49PM 3 Custom header when busy
12:31PM 0 Custom header when busy
12:05PM 2 Custom header when busy
12:02PM 0 asterisk email to fax
11:45AM 2 asterisk email to fax
11:02AM 0 Custom header when busy
8:29AM 0 For a failed retransmission - what were the IP addresses?
 
Wednesday July 1 2015
TimeRepliesSubject
6:44PM 0 Dell portability
6:36PM 2 Dell portability
12:43PM 0 Sip registrations question
9:46AM 2 Custom header when busy
7:10AM 0 Help With Physical Layer
3:03AM 1 Question on permit/deny