Re: [asterisk-users] Obtaining high voice quality in dahdi channel
Asterisk uses echo cancellation to enhance audio. EC can be done in soft- or hardware. All major card manufacturers have variants of their cards with hardware echo cancellation support. Searching the internet gives a lot of information about this topic. If you install a Sangoma card without HWEC, then you do get the default software canceler. Depending on your needs, switching to the OSLEC canceler might or might not be a better choice. Generally, a card with HWEC gives the best results, especially when you also have a lot of facsimile messages. 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
[asterisk-users] Confbridge Dynamic video_mode
Hi All,
I want to set the video_mode of the confbridge dynamically in the dialplan.
SO say if 5 users join the conference with follow_talker mode, it should
work like that (and it does). But if 6th user changes the video_mode to
first_marked and gets marked in the dial plan and joins the conference, he
does not become the single video source of the conf. The video mode stays
follow_talker.
I have tried changing conf mode dynamically in dialplan but does not work
following is my settings and dialplan:
confbridge.conf
--
[common_bridge]
type=bridge
record_conference=yes
internal_sample_rate=auto
mixing_interval=20
video_mode = none
conference participants.
release_as_single_video_src
[default_user]
type=user
dsp_talking_threshold=128
dsp_silence_threshold=2000
talk_detection_events=yes
extensions.conf
---
exten = 200,1,Noop(Going to ConfBridge now)
same = n,SET(CONFBRIDGE(bridge,video_mode)=follow_talker)
same = n,Answer()
same = n,Confbridge(1234,common_bridge,,sample_user_menu)
same = n,NOOP(${CONFBRIDGESTATUS})
exten = 300,1,SET(CONFBRIDGE(user,marked)=yes)
same = n,SET(CONFBRIDGE(bridge,video_mode)=first_marked)
same = n,Answer()
same = n,Confbridge(1234,common_bridge,,sample_user_menu)
same = n,NOOP(${CONFBRIDGESTATUS})
Marked user dials 300 and all others dial 200.
--
Best Ragards
Rizwan H Qureshi
V: +971 (0) 528272154
linkedin.com/in/rhqureshi
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Re: [asterisk-users] hwo to stok variable wiith menu
hello list
i would your help please regarding this issue
with the below code i can store the call date and the callerid ,now i want
to store also the sip phone called 223
could you please see the code and tell me how can i add the sip phone in
my table 'Menu'
exten = 506,1,Ringing()
exten = 506,n,Dial(SIP/223, 30)
exten = 506,n,Goto(support,s,1)
[support]
exten = s,1,NoOp(User chose support option)
exten = s,n,MYSQL(Connect connid localhost aheevaccs aheevaccs aheevaccs)
exten = s,n,MYSQL(Query resultid ${connid} INSERT\ INTO\ menu\ SET\
callerid='${CALLERID(num)}'\, calldate=now())
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
thanks and regards
2011/12/1 salaheddine elharit salah.elharit...@gmail.com
Hi Noll,
all works perfectly thanks a lot for your help and support i really
appreciate it :)
Best Regards
2011/12/1 Dale Noll dn...@wi.rr.com
On 11/30/2011 11:13 AM, salaheddine elharit wrote:
i have last question regarding this thread
with exten = 3,n,MYSQL(Query resultid ${connid} insert into test (
option_name ) values ('${CALLERID(num)}'))
i can store the phone number without issue
i need also the date and hour fo call in the count coulum
could you please give me the syntex
best regards
The example table that I gave originally was before I knew what you were
looking to do. I assumed, incorrectly that you simply wanted to track how
many times an option was selected in the menu.
I would recommend that you create a table specifically for this
application.
That table may look like this. Please name the table and columns
appropriately for your application.
create table option_three (
calldatedatetime,
calleridvarchar(40)
)
Then the sql would look something like this...
exten = 3,n,MYSQL(Query resultid ${connid} insert into option_three (
calldate, callerid ) values ( now(), '${CALLERID(num)}'))
Dale
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Lyta Alexander - Babylon 5
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[asterisk-users] Transfer cmd via AsyncAGI
Hello, I am using Asterisk 11.0.1 and do not notice any changes regarding the Transfer on newer Asterisk 11.x versions. I am using AMI and controlling a channel via AsyncAGI. I send a Transfer cmd (such as the following) Action: AGI ActionID: C8 Channel: SIP/1004-0002 CommandID: C8 Command: EXEC Transfer SIP/1003 Destination phone starts ringing. If it answers the call, everything works fine. I am notified the agiexec completed successfully and given a TRANSFERSTATUS of SUCCESS. I am also notified when the call is hungup so that I can cleanup information regarding the call. Event: Hangup Privilege: call,all Channel: SIP/1004-0002 Uniqueid: 1367761382.0 CallerIDNum: 1004 CallerIDName: 1004 - Asterisk ConnectedLineNum: unknown ConnectedLineName: unknown AccountCode: 3 Cause: 16 Cause-txt: Normal Clearing However, if the destination does not answer the Transfer (SIP REFER) and I hangup the original call being transferred, it stops ringing the destination phone and partially notifies me of the completed call. However, I do not receive any Event: Hangup for the original Channel: SIP/1004-0002 In fact, I do not receive any information for the agiexec completing. The very last event notification I receive with this channel is the following Event: AGIExec Privilege: agi,all SubEvent: Start Channel: SIP/1004-0002 CommandId: 556226156 Command: EXEC Transfer SIP/1003 Any suggestions on what I may be doing wrong? Are there any known fixes for this? Or should I submit a bug to the developer list? Have a great day! Dan -- _ -- 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] No early media on 302 redirect via two servers
We have a situation where we get no early media in this call flow: VoIP origination provider Server1 (our server) Customer server Customer phone with call-forward set Server1 to dial the forward-to number Then there is no early media while the forward-to number is ringing. Our server is Asterisk 1.6 and theirs is 1.8. I tried promiscredir=yes and then the calls fail altogether because rather than using the local channel, it makes a SIP call that is not allowed. I don't think that the FORWARD_CONTEXT variable is used in these versions, because setting it doesn't impact the channel selection at all. -- 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] passing '302 moved temporarily' back to the SIP provider
On Tue, May 7, 2013 at 10:05 PM, Satish Barot satish4aster...@gmail.comwrote: promiscredir= yes in sip.conf should help you achieve your requirement. I haven't been able to get that to work in a similar situation, except we are the provider. It results in the new invite being from the CLID of the original caller, and fails. -- 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] passing '302 moved temporarily' back to the SIP provider
On 5/9/13, Carlos Alvarez car...@televolve.com wrote:
On Tue, May 7, 2013 at 10:05 PM, Satish Barot
satish4aster...@gmail.comwrote:
promiscredir= yes in sip.conf should help you achieve your requirement.
I haven't been able to get that to work in a similar situation, except we
are the provider. It results in the new invite being from the CLID of the
original caller, and fails.
--
Carlos Alvarez
TelEvolve
602-889-3003
Completely misunderstood the OP!
Revised solution:
Set promiscredir= no in sip.conf. I assume you land your dids in
[incoming-trunk] and here is the basic dialplan tested on 11 but
should work on 1.8.
[incoming-trunk]
;-- Handle Incoming DIDs. Mine start with 89 and are of 4 digits --;
exten =
_89XX,1,Noop(RDNIS=${CALLERID(rdnis)}::ANI=${CALLERID(ani)}::DNID=${CALLERID(dnid)})
same = n,Set(__ORIGCHANNEL=${CHANNEL})
same = n,Dial(SIP/${EXTEN},30)
;-- Dialplan to handle 302 Moved temporarily --;
exten =
_X..,1,Noop(ORIGCHANNEL=${ORIGCHANNEL}::RDNIS=${CALLERID(rdnis)}::ANI=${CALLERID(ani)}::DNID=${CALLERID(dnid)}::CHANNELTYPE=${CHANNEL(channeltype)})
same =
n,ExecIf($[${CALLERID(rdnis)}!=]?ChannelRedirect(${ORIGCHANNEL},back2provider,${EXTEN},1)
same = n,Hangup()
[back2provider]
;--Send 302 back to provider --;
exten = _X.,1,Transfer(${EXTEN})
same = n,NoOp(TRANSFERSTATUS=${TRANSFERSTATUS})
same = n,Hangup()
--Satish Barot
Ahmedabad, India
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Re: [asterisk-users] passing '302 moved temporarily' back to the SIP provider
On 5/9/13, Satish Barot satish4aster...@gmail.com wrote:
On 5/9/13, Carlos Alvarez car...@televolve.com wrote:
On Tue, May 7, 2013 at 10:05 PM, Satish Barot
satish4aster...@gmail.comwrote:
promiscredir= yes in sip.conf should help you achieve your requirement.
I haven't been able to get that to work in a similar situation, except we
are the provider. It results in the new invite being from the CLID of
the
original caller, and fails.
--
Carlos Alvarez
TelEvolve
602-889-3003
Completely misunderstood the OP!
Revised solution:
Set promiscredir= no in sip.conf. I assume you land your dids in
[incoming-trunk] and here is the basic dialplan tested on 11 but
should work on 1.8.
[incoming-trunk]
;-- Handle Incoming DIDs. Mine start with 89 and are of 4 digits --;
exten =
_89XX,1,Noop(RDNIS=${CALLERID(rdnis)}::ANI=${CALLERID(ani)}::DNID=${CALLERID(dnid)})
same = n,Set(__ORIGCHANNEL=${CHANNEL})
same = n,Dial(SIP/${EXTEN},30)
;-- Dialplan to handle 302 Moved temporarily --;
exten =
_X..,1,Noop(ORIGCHANNEL=${ORIGCHANNEL}::RDNIS=${CALLERID(rdnis)}::ANI=${CALLERID(ani)}::DNID=${CALLERID(dnid)}::CHANNELTYPE=${CHANNEL(channeltype)})
same =
n,ExecIf($[${CALLERID(rdnis)}!=]?ChannelRedirect(${ORIGCHANNEL},back2provider,${EXTEN},1)
same = n,Hangup()
[back2provider]
;--Send 302 back to provider --;
exten = _X.,1,Transfer(${EXTEN})
same = n,NoOp(TRANSFERSTATUS=${TRANSFERSTATUS})
same = n,Hangup()
--Satish Barot
Ahmedabad, India
[incoming-trunk] is also a context of my SIP extensions.
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